AudioTrack.java revision 9e29086d5df800532e736d8f31e2b9159b102524
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, 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 // Java channel mask definitions below match those 705 // in /system/core/include/system/audio.h in the JNI code of AudioTrack. 706 707 // internal maximum size for bits parameter, not part of public API 708 private static final int AUDIO_CHANNEL_BITS_LOG2 = 30; 709 710 // log(2) of maximum number of representations, not part of public API 711 private static final int AUDIO_CHANNEL_REPRESENTATION_LOG2 = 2; 712 713 // used to create a channel index mask or channel position mask 714 // with getChannelMaskFromRepresentationAndBits(); 715 private static final int CHANNEL_OUT_REPRESENTATION_POSITION = 0; 716 private static final int CHANNEL_OUT_REPRESENTATION_INDEX = 2; 717 718 /** 719 * Return the channel mask from its representation and bits. 720 * 721 * This creates a channel mask for mChannels which combines a 722 * representation field and a bits field. This is for internal 723 * communication to native code, not part of the public API. 724 * 725 * @param representation the type of channel mask, 726 * either CHANNEL_OUT_REPRESENTATION_POSITION 727 * or CHANNEL_OUT_REPRESENTATION_INDEX 728 * @param bits is the channel bits specifying occupancy 729 * @return the channel mask 730 * @throws java.lang.IllegalArgumentException if representation is not recognized or 731 * the bits field is not acceptable for that representation 732 */ 733 private static int getChannelMaskFromRepresentationAndBits(int representation, int bits) { 734 switch (representation) { 735 case CHANNEL_OUT_REPRESENTATION_POSITION: 736 case CHANNEL_OUT_REPRESENTATION_INDEX: 737 if ((bits & ~((1 << AUDIO_CHANNEL_BITS_LOG2) - 1)) != 0) { 738 throw new IllegalArgumentException("invalid bits " + bits); 739 } 740 return representation << AUDIO_CHANNEL_BITS_LOG2 | bits; 741 default: 742 throw new IllegalArgumentException("invalid representation " + representation); 743 } 744 } 745 746 // Convenience method for the constructor's parameter checks. 747 // This is where constructor IllegalArgumentException-s are thrown 748 // postconditions: 749 // mChannelCount is valid 750 // mChannels is valid 751 // mAudioFormat is valid 752 // mSampleRate is valid 753 // mDataLoadMode is valid 754 private void audioParamCheck(int sampleRateInHz, int channelConfig, int channelIndexMask, 755 int audioFormat, int mode) { 756 //-------------- 757 // sample rate, note these values are subject to change 758 if (sampleRateInHz < SAMPLE_RATE_HZ_MIN || sampleRateInHz > SAMPLE_RATE_HZ_MAX) { 759 throw new IllegalArgumentException(sampleRateInHz 760 + "Hz is not a supported sample rate."); 761 } 762 mSampleRate = sampleRateInHz; 763 764 //-------------- 765 // channel config 766 mChannelConfiguration = channelConfig; 767 768 switch (channelConfig) { 769 case AudioFormat.CHANNEL_OUT_DEFAULT: //AudioFormat.CHANNEL_CONFIGURATION_DEFAULT 770 case AudioFormat.CHANNEL_OUT_MONO: 771 case AudioFormat.CHANNEL_CONFIGURATION_MONO: 772 mChannelCount = 1; 773 mChannels = AudioFormat.CHANNEL_OUT_MONO; 774 break; 775 case AudioFormat.CHANNEL_OUT_STEREO: 776 case AudioFormat.CHANNEL_CONFIGURATION_STEREO: 777 mChannelCount = 2; 778 mChannels = AudioFormat.CHANNEL_OUT_STEREO; 779 break; 780 default: 781 if (channelConfig == AudioFormat.CHANNEL_INVALID && channelIndexMask != 0) { 782 mChannelCount = 0; 783 break; // channel index configuration only 784 } 785 if (!isMultichannelConfigSupported(channelConfig)) { 786 // input channel configuration features unsupported channels 787 throw new IllegalArgumentException("Unsupported channel configuration."); 788 } 789 mChannels = channelConfig; 790 mChannelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig); 791 } 792 // check the channel index configuration (if present) 793 mChannelIndexMask = channelIndexMask; 794 if (mChannelIndexMask != 0) { 795 // restrictive: indexMask could allow up to AUDIO_CHANNEL_BITS_LOG2 796 final int indexMask = (1 << CHANNEL_COUNT_MAX) - 1; 797 if ((channelIndexMask & ~indexMask) != 0) { 798 throw new IllegalArgumentException("Unsupported channel index configuration " 799 + channelIndexMask); 800 } 801 int channelIndexCount = Integer.bitCount(channelIndexMask); 802 if (mChannelCount == 0) { 803 mChannelCount = channelIndexCount; 804 } else if (mChannelCount != channelIndexCount) { 805 throw new IllegalArgumentException("Channel count must match"); 806 } 807 808 // AudioTrack prefers to use the channel index configuration 809 // over the channel position configuration if both are specified. 810 mChannels = getChannelMaskFromRepresentationAndBits( 811 CHANNEL_OUT_REPRESENTATION_INDEX, mChannelIndexMask); 812 } 813 814 //-------------- 815 // audio format 816 if (audioFormat == AudioFormat.ENCODING_DEFAULT) { 817 audioFormat = AudioFormat.ENCODING_PCM_16BIT; 818 } 819 820 if (!AudioFormat.isValidEncoding(audioFormat)) { 821 throw new IllegalArgumentException("Unsupported audio encoding."); 822 } 823 mAudioFormat = audioFormat; 824 825 //-------------- 826 // audio load mode 827 if (((mode != MODE_STREAM) && (mode != MODE_STATIC)) || 828 ((mode != MODE_STREAM) && !AudioFormat.isEncodingLinearPcm(mAudioFormat))) { 829 throw new IllegalArgumentException("Invalid mode."); 830 } 831 mDataLoadMode = mode; 832 } 833 834 /** 835 * Convenience method to check that the channel configuration (a.k.a channel mask) is supported 836 * @param channelConfig the mask to validate 837 * @return false if the AudioTrack can't be used with such a mask 838 */ 839 private static boolean isMultichannelConfigSupported(int channelConfig) { 840 // check for unsupported channels 841 if ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) { 842 loge("Channel configuration features unsupported channels"); 843 return false; 844 } 845 final int channelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig); 846 if (channelCount > CHANNEL_COUNT_MAX) { 847 loge("Channel configuration contains too many channels " + 848 channelCount + ">" + CHANNEL_COUNT_MAX); 849 return false; 850 } 851 // check for unsupported multichannel combinations: 852 // - FL/FR must be present 853 // - L/R channels must be paired (e.g. no single L channel) 854 final int frontPair = 855 AudioFormat.CHANNEL_OUT_FRONT_LEFT | AudioFormat.CHANNEL_OUT_FRONT_RIGHT; 856 if ((channelConfig & frontPair) != frontPair) { 857 loge("Front channels must be present in multichannel configurations"); 858 return false; 859 } 860 final int backPair = 861 AudioFormat.CHANNEL_OUT_BACK_LEFT | AudioFormat.CHANNEL_OUT_BACK_RIGHT; 862 if ((channelConfig & backPair) != 0) { 863 if ((channelConfig & backPair) != backPair) { 864 loge("Rear channels can't be used independently"); 865 return false; 866 } 867 } 868 final int sidePair = 869 AudioFormat.CHANNEL_OUT_SIDE_LEFT | AudioFormat.CHANNEL_OUT_SIDE_RIGHT; 870 if ((channelConfig & sidePair) != 0 871 && (channelConfig & sidePair) != sidePair) { 872 loge("Side channels can't be used independently"); 873 return false; 874 } 875 return true; 876 } 877 878 879 // Convenience method for the constructor's audio buffer size check. 880 // preconditions: 881 // mChannelCount is valid 882 // mAudioFormat is valid 883 // postcondition: 884 // mNativeBufferSizeInBytes is valid (multiple of frame size, positive) 885 private void audioBuffSizeCheck(int audioBufferSize) { 886 // NB: this section is only valid with PCM data. 887 // To update when supporting compressed formats 888 int frameSizeInBytes; 889 if (AudioFormat.isEncodingLinearPcm(mAudioFormat)) { 890 frameSizeInBytes = mChannelCount 891 * (AudioFormat.getBytesPerSample(mAudioFormat)); 892 } else { 893 frameSizeInBytes = 1; 894 } 895 if ((audioBufferSize % frameSizeInBytes != 0) || (audioBufferSize < 1)) { 896 throw new IllegalArgumentException("Invalid audio buffer size."); 897 } 898 899 mNativeBufferSizeInBytes = audioBufferSize; 900 mNativeBufferSizeInFrames = audioBufferSize / frameSizeInBytes; 901 } 902 903 904 /** 905 * Releases the native AudioTrack resources. 906 */ 907 public void release() { 908 // even though native_release() stops the native AudioTrack, we need to stop 909 // AudioTrack subclasses too. 910 try { 911 stop(); 912 } catch(IllegalStateException ise) { 913 // don't raise an exception, we're releasing the resources. 914 } 915 native_release(); 916 mState = STATE_UNINITIALIZED; 917 } 918 919 @Override 920 protected void finalize() { 921 native_finalize(); 922 } 923 924 //-------------------------------------------------------------------------- 925 // Getters 926 //-------------------- 927 /** 928 * Returns the minimum gain value, which is the constant 0.0. 929 * Gain values less than 0.0 will be clamped to 0.0. 930 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 931 * @return the minimum value, which is the constant 0.0. 932 */ 933 static public float getMinVolume() { 934 return GAIN_MIN; 935 } 936 937 /** 938 * Returns the maximum gain value, which is greater than or equal to 1.0. 939 * Gain values greater than the maximum will be clamped to the maximum. 940 * <p>The word "volume" in the API name is historical; this is actually a gain. 941 * expressed as a linear multiplier on sample values, where a maximum value of 1.0 942 * corresponds to a gain of 0 dB (sample values left unmodified). 943 * @return the maximum value, which is greater than or equal to 1.0. 944 */ 945 static public float getMaxVolume() { 946 return GAIN_MAX; 947 } 948 949 /** 950 * Returns the configured audio data sample rate in Hz 951 */ 952 public int getSampleRate() { 953 return mSampleRate; 954 } 955 956 /** 957 * Returns the current playback sample rate rate in Hz. 958 */ 959 public int getPlaybackRate() { 960 return native_get_playback_rate(); 961 } 962 963 /** 964 * Returns the current playback settings. 965 * See {@link #setPlaybackSettings(PlaybackSettings)} to set playback settings 966 * @return current {@link PlaybackSettings}. 967 * @throws IllegalStateException if track is not initialized. 968 */ 969 public @NonNull PlaybackSettings getPlaybackSettings() { 970 float[] floatArray = new float[2]; 971 int[] intArray = new int[2]; 972 native_get_playback_settings(floatArray, intArray); 973 return new PlaybackSettings() 974 .setSpeed(floatArray[0]) 975 .setPitch(floatArray[1]) 976 .setAudioFallbackMode(intArray[0]) 977 .setAudioStretchMode(intArray[1]); 978 } 979 980 /** 981 * Returns the configured audio data format. See {@link AudioFormat#ENCODING_PCM_16BIT} 982 * and {@link AudioFormat#ENCODING_PCM_8BIT}. 983 */ 984 public int getAudioFormat() { 985 return mAudioFormat; 986 } 987 988 /** 989 * Returns the type of audio stream this AudioTrack is configured for. 990 * Compare the result against {@link AudioManager#STREAM_VOICE_CALL}, 991 * {@link AudioManager#STREAM_SYSTEM}, {@link AudioManager#STREAM_RING}, 992 * {@link AudioManager#STREAM_MUSIC}, {@link AudioManager#STREAM_ALARM}, 993 * {@link AudioManager#STREAM_NOTIFICATION}, or {@link AudioManager#STREAM_DTMF}. 994 */ 995 public int getStreamType() { 996 return mStreamType; 997 } 998 999 /** 1000 * Returns the configured channel position mask. 1001 * For example, refer to {@link AudioFormat#CHANNEL_OUT_MONO}, 1002 * {@link AudioFormat#CHANNEL_OUT_STEREO}, {@link AudioFormat#CHANNEL_OUT_5POINT1}. 1003 */ 1004 public int getChannelConfiguration() { 1005 return mChannelConfiguration; 1006 } 1007 1008 /** 1009 * Returns the configured number of channels. 1010 */ 1011 public int getChannelCount() { 1012 return mChannelCount; 1013 } 1014 1015 /** 1016 * Returns the state of the AudioTrack instance. This is useful after the 1017 * AudioTrack instance has been created to check if it was initialized 1018 * properly. This ensures that the appropriate resources have been acquired. 1019 * @see #STATE_INITIALIZED 1020 * @see #STATE_NO_STATIC_DATA 1021 * @see #STATE_UNINITIALIZED 1022 */ 1023 public int getState() { 1024 return mState; 1025 } 1026 1027 /** 1028 * Returns the playback state of the AudioTrack instance. 1029 * @see #PLAYSTATE_STOPPED 1030 * @see #PLAYSTATE_PAUSED 1031 * @see #PLAYSTATE_PLAYING 1032 */ 1033 public int getPlayState() { 1034 synchronized (mPlayStateLock) { 1035 return mPlayState; 1036 } 1037 } 1038 1039 /** 1040 * Returns the "native frame count" of the <code>AudioTrack</code> buffer. 1041 * <p> If the track's creation mode is {@link #MODE_STATIC}, 1042 * it is equal to the specified bufferSizeInBytes on construction, converted to frame units. 1043 * A static track's native frame count will not change. 1044 * <p> If the track's creation mode is {@link #MODE_STREAM}, 1045 * it is greater than or equal to the specified bufferSizeInBytes converted to frame units. 1046 * For streaming tracks, this value may be rounded up to a larger value if needed by 1047 * the target output sink, and 1048 * if the track is subsequently routed to a different output sink, the native 1049 * frame count may enlarge to accommodate. 1050 * See also {@link AudioManager#getProperty(String)} for key 1051 * {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}. 1052 * @return current size in frames of the audio track buffer. 1053 * @throws IllegalStateException 1054 */ 1055 public int getNativeFrameCount() throws IllegalStateException { 1056 return native_get_native_frame_count(); 1057 } 1058 1059 /** 1060 * Returns marker position expressed in frames. 1061 * @return marker position in wrapping frame units similar to {@link #getPlaybackHeadPosition}, 1062 * or zero if marker is disabled. 1063 */ 1064 public int getNotificationMarkerPosition() { 1065 return native_get_marker_pos(); 1066 } 1067 1068 /** 1069 * Returns the notification update period expressed in frames. 1070 * Zero means that no position update notifications are being delivered. 1071 */ 1072 public int getPositionNotificationPeriod() { 1073 return native_get_pos_update_period(); 1074 } 1075 1076 /** 1077 * Returns the playback head position expressed in frames. 1078 * Though the "int" type is signed 32-bits, the value should be reinterpreted as if it is 1079 * unsigned 32-bits. That is, the next position after 0x7FFFFFFF is (int) 0x80000000. 1080 * This is a continuously advancing counter. It will wrap (overflow) periodically, 1081 * for example approximately once every 27:03:11 hours:minutes:seconds at 44.1 kHz. 1082 * It is reset to zero by {@link #flush()}, {@link #reloadStaticData()}, and {@link #stop()}. 1083 * If the track's creation mode is {@link #MODE_STATIC}, the return value indicates 1084 * the total number of frames played since reset, 1085 * <i>not</i> the current offset within the buffer. 1086 */ 1087 public int getPlaybackHeadPosition() { 1088 return native_get_position(); 1089 } 1090 1091 /** 1092 * Returns this track's estimated latency in milliseconds. This includes the latency due 1093 * to AudioTrack buffer size, AudioMixer (if any) and audio hardware driver. 1094 * 1095 * DO NOT UNHIDE. The existing approach for doing A/V sync has too many problems. We need 1096 * a better solution. 1097 * @hide 1098 */ 1099 public int getLatency() { 1100 return native_get_latency(); 1101 } 1102 1103 /** 1104 * Returns the output sample rate in Hz for the specified stream type. 1105 */ 1106 static public int getNativeOutputSampleRate(int streamType) { 1107 return native_get_output_sample_rate(streamType); 1108 } 1109 1110 /** 1111 * Returns the minimum buffer size required for the successful creation of an AudioTrack 1112 * object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't 1113 * guarantee a smooth playback under load, and higher values should be chosen according to 1114 * the expected frequency at which the buffer will be refilled with additional data to play. 1115 * For example, if you intend to dynamically set the source sample rate of an AudioTrack 1116 * to a higher value than the initial source sample rate, be sure to configure the buffer size 1117 * based on the highest planned sample rate. 1118 * @param sampleRateInHz the source sample rate expressed in Hz. 1119 * @param channelConfig describes the configuration of the audio channels. 1120 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 1121 * {@link AudioFormat#CHANNEL_OUT_STEREO} 1122 * @param audioFormat the format in which the audio data is represented. 1123 * See {@link AudioFormat#ENCODING_PCM_16BIT} and 1124 * {@link AudioFormat#ENCODING_PCM_8BIT}, 1125 * and {@link AudioFormat#ENCODING_PCM_FLOAT}. 1126 * @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed, 1127 * or {@link #ERROR} if unable to query for output properties, 1128 * or the minimum buffer size expressed in bytes. 1129 */ 1130 static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) { 1131 int channelCount = 0; 1132 switch(channelConfig) { 1133 case AudioFormat.CHANNEL_OUT_MONO: 1134 case AudioFormat.CHANNEL_CONFIGURATION_MONO: 1135 channelCount = 1; 1136 break; 1137 case AudioFormat.CHANNEL_OUT_STEREO: 1138 case AudioFormat.CHANNEL_CONFIGURATION_STEREO: 1139 channelCount = 2; 1140 break; 1141 default: 1142 if (!isMultichannelConfigSupported(channelConfig)) { 1143 loge("getMinBufferSize(): Invalid channel configuration."); 1144 return ERROR_BAD_VALUE; 1145 } else { 1146 channelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig); 1147 } 1148 } 1149 1150 if (!AudioFormat.isValidEncoding(audioFormat)) { 1151 loge("getMinBufferSize(): Invalid audio format."); 1152 return ERROR_BAD_VALUE; 1153 } 1154 1155 // sample rate, note these values are subject to change 1156 if ( (sampleRateInHz < SAMPLE_RATE_HZ_MIN) || (sampleRateInHz > SAMPLE_RATE_HZ_MAX) ) { 1157 loge("getMinBufferSize(): " + sampleRateInHz + " Hz is not a supported sample rate."); 1158 return ERROR_BAD_VALUE; 1159 } 1160 1161 int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat); 1162 if (size <= 0) { 1163 loge("getMinBufferSize(): error querying hardware"); 1164 return ERROR; 1165 } 1166 else { 1167 return size; 1168 } 1169 } 1170 1171 /** 1172 * Returns the audio session ID. 1173 * 1174 * @return the ID of the audio session this AudioTrack belongs to. 1175 */ 1176 public int getAudioSessionId() { 1177 return mSessionId; 1178 } 1179 1180 /** 1181 * Poll for a timestamp on demand. 1182 * <p> 1183 * If you need to track timestamps during initial warmup or after a routing or mode change, 1184 * you should request a new timestamp once per second until the reported timestamps 1185 * show that the audio clock is stable. 1186 * Thereafter, query for a new timestamp approximately once every 10 seconds to once per minute. 1187 * Calling this method more often is inefficient. 1188 * It is also counter-productive to call this method more often than recommended, 1189 * because the short-term differences between successive timestamp reports are not meaningful. 1190 * If you need a high-resolution mapping between frame position and presentation time, 1191 * consider implementing that at application level, based on low-resolution timestamps. 1192 * <p> 1193 * The audio data at the returned position may either already have been 1194 * presented, or may have not yet been presented but is committed to be presented. 1195 * It is not possible to request the time corresponding to a particular position, 1196 * or to request the (fractional) position corresponding to a particular time. 1197 * If you need such features, consider implementing them at application level. 1198 * 1199 * @param timestamp a reference to a non-null AudioTimestamp instance allocated 1200 * and owned by caller. 1201 * @return true if a timestamp is available, or false if no timestamp is available. 1202 * If a timestamp if available, 1203 * the AudioTimestamp instance is filled in with a position in frame units, together 1204 * with the estimated time when that frame was presented or is committed to 1205 * be presented. 1206 * In the case that no timestamp is available, any supplied instance is left unaltered. 1207 * A timestamp may be temporarily unavailable while the audio clock is stabilizing, 1208 * or during and immediately after a route change. 1209 */ 1210 // Add this text when the "on new timestamp" API is added: 1211 // Use if you need to get the most recent timestamp outside of the event callback handler. 1212 public boolean getTimestamp(AudioTimestamp timestamp) 1213 { 1214 if (timestamp == null) { 1215 throw new IllegalArgumentException(); 1216 } 1217 // It's unfortunate, but we have to either create garbage every time or use synchronized 1218 long[] longArray = new long[2]; 1219 int ret = native_get_timestamp(longArray); 1220 if (ret != SUCCESS) { 1221 return false; 1222 } 1223 timestamp.framePosition = longArray[0]; 1224 timestamp.nanoTime = longArray[1]; 1225 return true; 1226 } 1227 1228 1229 //-------------------------------------------------------------------------- 1230 // Initialization / configuration 1231 //-------------------- 1232 /** 1233 * Sets the listener the AudioTrack notifies when a previously set marker is reached or 1234 * for each periodic playback head position update. 1235 * Notifications will be received in the same thread as the one in which the AudioTrack 1236 * instance was created. 1237 * @param listener 1238 */ 1239 public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener) { 1240 setPlaybackPositionUpdateListener(listener, null); 1241 } 1242 1243 /** 1244 * Sets the listener the AudioTrack notifies when a previously set marker is reached or 1245 * for each periodic playback head position update. 1246 * Use this method to receive AudioTrack events in the Handler associated with another 1247 * thread than the one in which you created the AudioTrack instance. 1248 * @param listener 1249 * @param handler the Handler that will receive the event notification messages. 1250 */ 1251 public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener, 1252 Handler handler) { 1253 if (listener != null) { 1254 mEventHandlerDelegate = new NativePositionEventHandlerDelegate(this, listener, handler); 1255 } else { 1256 mEventHandlerDelegate = null; 1257 } 1258 } 1259 1260 1261 private static float clampGainOrLevel(float gainOrLevel) { 1262 if (Float.isNaN(gainOrLevel)) { 1263 throw new IllegalArgumentException(); 1264 } 1265 if (gainOrLevel < GAIN_MIN) { 1266 gainOrLevel = GAIN_MIN; 1267 } else if (gainOrLevel > GAIN_MAX) { 1268 gainOrLevel = GAIN_MAX; 1269 } 1270 return gainOrLevel; 1271 } 1272 1273 1274 /** 1275 * Sets the specified left and right output gain values on the AudioTrack. 1276 * <p>Gain values are clamped to the closed interval [0.0, max] where 1277 * max is the value of {@link #getMaxVolume}. 1278 * A value of 0.0 results in zero gain (silence), and 1279 * a value of 1.0 means unity gain (signal unchanged). 1280 * The default value is 1.0 meaning unity gain. 1281 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 1282 * @param leftGain output gain for the left channel. 1283 * @param rightGain output gain for the right channel 1284 * @return error code or success, see {@link #SUCCESS}, 1285 * {@link #ERROR_INVALID_OPERATION} 1286 * @deprecated Applications should use {@link #setVolume} instead, as it 1287 * more gracefully scales down to mono, and up to multi-channel content beyond stereo. 1288 */ 1289 public int setStereoVolume(float leftGain, float rightGain) { 1290 if (isRestricted()) { 1291 return SUCCESS; 1292 } 1293 if (mState == STATE_UNINITIALIZED) { 1294 return ERROR_INVALID_OPERATION; 1295 } 1296 1297 leftGain = clampGainOrLevel(leftGain); 1298 rightGain = clampGainOrLevel(rightGain); 1299 1300 native_setVolume(leftGain, rightGain); 1301 1302 return SUCCESS; 1303 } 1304 1305 1306 /** 1307 * Sets the specified output gain value on all channels of this track. 1308 * <p>Gain values are clamped to the closed interval [0.0, max] where 1309 * max is the value of {@link #getMaxVolume}. 1310 * A value of 0.0 results in zero gain (silence), and 1311 * a value of 1.0 means unity gain (signal unchanged). 1312 * The default value is 1.0 meaning unity gain. 1313 * <p>This API is preferred over {@link #setStereoVolume}, as it 1314 * more gracefully scales down to mono, and up to multi-channel content beyond stereo. 1315 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 1316 * @param gain output gain for all channels. 1317 * @return error code or success, see {@link #SUCCESS}, 1318 * {@link #ERROR_INVALID_OPERATION} 1319 */ 1320 public int setVolume(float gain) { 1321 return setStereoVolume(gain, gain); 1322 } 1323 1324 1325 /** 1326 * Sets the playback sample rate for this track. This sets the sampling rate at which 1327 * the audio data will be consumed and played back 1328 * (as set by the sampleRateInHz parameter in the 1329 * {@link #AudioTrack(int, int, int, int, int, int)} constructor), 1330 * not the original sampling rate of the 1331 * content. For example, setting it to half the sample rate of the content will cause the 1332 * playback to last twice as long, but will also result in a pitch shift down by one octave. 1333 * The valid sample rate range is from 1 Hz to twice the value returned by 1334 * {@link #getNativeOutputSampleRate(int)}. 1335 * Use {@link #setPlaybackSettings(PlaybackSettings)} for speed control. 1336 * @param sampleRateInHz the sample rate expressed in Hz 1337 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1338 * {@link #ERROR_INVALID_OPERATION} 1339 */ 1340 public int setPlaybackRate(int sampleRateInHz) { 1341 if (mState != STATE_INITIALIZED) { 1342 return ERROR_INVALID_OPERATION; 1343 } 1344 if (sampleRateInHz <= 0) { 1345 return ERROR_BAD_VALUE; 1346 } 1347 return native_set_playback_rate(sampleRateInHz); 1348 } 1349 1350 1351 /** 1352 * Sets the playback settings. 1353 * This method returns failure if it cannot apply the playback settings. 1354 * One possible cause is that the parameters for speed or pitch are out of range. 1355 * Another possible cause is that the <code>AudioTrack</code> is streaming 1356 * (see {@link #MODE_STREAM}) and the 1357 * buffer size is too small. For speeds greater than 1.0f, the <code>AudioTrack</code> buffer 1358 * on configuration must be larger than the speed multiplied by the minimum size 1359 * {@link #getMinBufferSize(int, int, int)}) to allow proper playback. 1360 * @param settings see {@link PlaybackSettings}. In particular, 1361 * speed, pitch, and audio mode should be set. 1362 * @throws IllegalArgumentException if the settings are invalid or not accepted. 1363 * @throws IllegalStateException if track is not initialized. 1364 */ 1365 public void setPlaybackSettings(@NonNull PlaybackSettings settings) { 1366 if (settings == null) { 1367 throw new IllegalArgumentException("settings is null"); 1368 } 1369 float[] floatArray; 1370 int[] intArray; 1371 try { 1372 floatArray = new float[] { 1373 settings.getSpeed(), 1374 settings.getPitch(), 1375 }; 1376 intArray = new int[] { 1377 settings.getAudioFallbackMode(), 1378 settings.getAudioStretchMode(), 1379 }; 1380 } catch (IllegalStateException e) { 1381 throw new IllegalArgumentException(e); 1382 } 1383 native_set_playback_settings(floatArray, intArray); 1384 } 1385 1386 1387 /** 1388 * Sets the position of the notification marker. At most one marker can be active. 1389 * @param markerInFrames marker position in wrapping frame units similar to 1390 * {@link #getPlaybackHeadPosition}, or zero to disable the marker. 1391 * To set a marker at a position which would appear as zero due to wraparound, 1392 * a workaround is to use a non-zero position near zero, such as -1 or 1. 1393 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1394 * {@link #ERROR_INVALID_OPERATION} 1395 */ 1396 public int setNotificationMarkerPosition(int markerInFrames) { 1397 if (mState == STATE_UNINITIALIZED) { 1398 return ERROR_INVALID_OPERATION; 1399 } 1400 return native_set_marker_pos(markerInFrames); 1401 } 1402 1403 1404 /** 1405 * Sets the period for the periodic notification event. 1406 * @param periodInFrames update period expressed in frames. 1407 * Zero period means no position updates. A negative period is not allowed. 1408 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_INVALID_OPERATION} 1409 */ 1410 public int setPositionNotificationPeriod(int periodInFrames) { 1411 if (mState == STATE_UNINITIALIZED) { 1412 return ERROR_INVALID_OPERATION; 1413 } 1414 return native_set_pos_update_period(periodInFrames); 1415 } 1416 1417 1418 /** 1419 * Sets the playback head position within the static buffer. 1420 * The track must be stopped or paused for the position to be changed, 1421 * and must use the {@link #MODE_STATIC} mode. 1422 * @param positionInFrames playback head position within buffer, expressed in frames. 1423 * Zero corresponds to start of buffer. 1424 * The position must not be greater than the buffer size in frames, or negative. 1425 * Though this method and {@link #getPlaybackHeadPosition()} have similar names, 1426 * the position values have different meanings. 1427 * <br> 1428 * If looping is currently enabled and the new position is greater than or equal to the 1429 * loop end marker, the behavior varies by API level: for API level 22 and above, 1430 * the looping is first disabled and then the position is set. 1431 * For earlier API levels, the behavior is unspecified. 1432 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1433 * {@link #ERROR_INVALID_OPERATION} 1434 */ 1435 public int setPlaybackHeadPosition(int positionInFrames) { 1436 if (mDataLoadMode == MODE_STREAM || mState == STATE_UNINITIALIZED || 1437 getPlayState() == PLAYSTATE_PLAYING) { 1438 return ERROR_INVALID_OPERATION; 1439 } 1440 if (!(0 <= positionInFrames && positionInFrames <= mNativeBufferSizeInFrames)) { 1441 return ERROR_BAD_VALUE; 1442 } 1443 return native_set_position(positionInFrames); 1444 } 1445 1446 /** 1447 * Sets the loop points and the loop count. The loop can be infinite. 1448 * Similarly to setPlaybackHeadPosition, 1449 * the track must be stopped or paused for the loop points to be changed, 1450 * and must use the {@link #MODE_STATIC} mode. 1451 * @param startInFrames loop start marker expressed in frames. 1452 * Zero corresponds to start of buffer. 1453 * The start marker must not be greater than or equal to the buffer size in frames, or negative. 1454 * @param endInFrames loop end marker expressed in frames. 1455 * The total buffer size in frames corresponds to end of buffer. 1456 * The end marker must not be greater than the buffer size in frames. 1457 * For looping, the end marker must not be less than or equal to the start marker, 1458 * but to disable looping 1459 * it is permitted for start marker, end marker, and loop count to all be 0. 1460 * If any input parameters are out of range, this method returns {@link #ERROR_BAD_VALUE}. 1461 * If the loop period (endInFrames - startInFrames) is too small for the implementation to 1462 * support, 1463 * {@link #ERROR_BAD_VALUE} is returned. 1464 * The loop range is the interval [startInFrames, endInFrames). 1465 * <br> 1466 * For API level 22 and above, the position is left unchanged, 1467 * unless it is greater than or equal to the loop end marker, in which case 1468 * it is forced to the loop start marker. 1469 * For earlier API levels, the effect on position is unspecified. 1470 * @param loopCount the number of times the loop is looped; must be greater than or equal to -1. 1471 * A value of -1 means infinite looping, and 0 disables looping. 1472 * A value of positive N means to "loop" (go back) N times. For example, 1473 * a value of one means to play the region two times in total. 1474 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1475 * {@link #ERROR_INVALID_OPERATION} 1476 */ 1477 public int setLoopPoints(int startInFrames, int endInFrames, int loopCount) { 1478 if (mDataLoadMode == MODE_STREAM || mState == STATE_UNINITIALIZED || 1479 getPlayState() == PLAYSTATE_PLAYING) { 1480 return ERROR_INVALID_OPERATION; 1481 } 1482 if (loopCount == 0) { 1483 ; // explicitly allowed as an exception to the loop region range check 1484 } else if (!(0 <= startInFrames && startInFrames < mNativeBufferSizeInFrames && 1485 startInFrames < endInFrames && endInFrames <= mNativeBufferSizeInFrames)) { 1486 return ERROR_BAD_VALUE; 1487 } 1488 return native_set_loop(startInFrames, endInFrames, loopCount); 1489 } 1490 1491 /** 1492 * Sets the initialization state of the instance. This method was originally intended to be used 1493 * in an AudioTrack subclass constructor to set a subclass-specific post-initialization state. 1494 * However, subclasses of AudioTrack are no longer recommended, so this method is obsolete. 1495 * @param state the state of the AudioTrack instance 1496 * @deprecated Only accessible by subclasses, which are not recommended for AudioTrack. 1497 */ 1498 @Deprecated 1499 protected void setState(int state) { 1500 mState = state; 1501 } 1502 1503 1504 //--------------------------------------------------------- 1505 // Transport control methods 1506 //-------------------- 1507 /** 1508 * Starts playing an AudioTrack. 1509 * If track's creation mode is {@link #MODE_STATIC}, you must have called one of 1510 * the {@link #write(byte[], int, int)}, {@link #write(short[], int, int)}, 1511 * or {@link #write(float[], int, int, int)} methods. 1512 * If the mode is {@link #MODE_STREAM}, you can optionally prime the 1513 * output buffer by writing up to bufferSizeInBytes (from constructor) before starting. 1514 * This priming will avoid an immediate underrun, but is not required. 1515 * 1516 * @throws IllegalStateException 1517 */ 1518 public void play() 1519 throws IllegalStateException { 1520 if (mState != STATE_INITIALIZED) { 1521 throw new IllegalStateException("play() called on uninitialized AudioTrack."); 1522 } 1523 if (isRestricted()) { 1524 setVolume(0); 1525 } 1526 synchronized(mPlayStateLock) { 1527 native_start(); 1528 mPlayState = PLAYSTATE_PLAYING; 1529 } 1530 } 1531 1532 private boolean isRestricted() { 1533 if ((mAttributes.getFlags() & AudioAttributes.FLAG_BYPASS_INTERRUPTION_POLICY) != 0) { 1534 return false; 1535 } 1536 try { 1537 final int usage = AudioAttributes.usageForLegacyStreamType(mStreamType); 1538 final int mode = mAppOps.checkAudioOperation(AppOpsManager.OP_PLAY_AUDIO, usage, 1539 Process.myUid(), ActivityThread.currentPackageName()); 1540 return mode != AppOpsManager.MODE_ALLOWED; 1541 } catch (RemoteException e) { 1542 return false; 1543 } 1544 } 1545 1546 /** 1547 * Stops playing the audio data. 1548 * When used on an instance created in {@link #MODE_STREAM} mode, audio will stop playing 1549 * after the last buffer that was written has been played. For an immediate stop, use 1550 * {@link #pause()}, followed by {@link #flush()} to discard audio data that hasn't been played 1551 * back yet. 1552 * @throws IllegalStateException 1553 */ 1554 public void stop() 1555 throws IllegalStateException { 1556 if (mState != STATE_INITIALIZED) { 1557 throw new IllegalStateException("stop() called on uninitialized AudioTrack."); 1558 } 1559 1560 // stop playing 1561 synchronized(mPlayStateLock) { 1562 native_stop(); 1563 mPlayState = PLAYSTATE_STOPPED; 1564 mAvSyncHeader = null; 1565 mAvSyncBytesRemaining = 0; 1566 } 1567 } 1568 1569 /** 1570 * Pauses the playback of the audio data. Data that has not been played 1571 * back will not be discarded. Subsequent calls to {@link #play} will play 1572 * this data back. See {@link #flush()} to discard this data. 1573 * 1574 * @throws IllegalStateException 1575 */ 1576 public void pause() 1577 throws IllegalStateException { 1578 if (mState != STATE_INITIALIZED) { 1579 throw new IllegalStateException("pause() called on uninitialized AudioTrack."); 1580 } 1581 //logd("pause()"); 1582 1583 // pause playback 1584 synchronized(mPlayStateLock) { 1585 native_pause(); 1586 mPlayState = PLAYSTATE_PAUSED; 1587 } 1588 } 1589 1590 1591 //--------------------------------------------------------- 1592 // Audio data supply 1593 //-------------------- 1594 1595 /** 1596 * Flushes the audio data currently queued for playback. Any data that has 1597 * been written but not yet presented will be discarded. No-op if not stopped or paused, 1598 * or if the track's creation mode is not {@link #MODE_STREAM}. 1599 * <BR> Note that although data written but not yet presented is discarded, there is no 1600 * guarantee that all of the buffer space formerly used by that data 1601 * is available for a subsequent write. 1602 * For example, a call to {@link #write(byte[], int, int)} with <code>sizeInBytes</code> 1603 * less than or equal to the total buffer size 1604 * may return a short actual transfer count. 1605 */ 1606 public void flush() { 1607 if (mState == STATE_INITIALIZED) { 1608 // flush the data in native layer 1609 native_flush(); 1610 mAvSyncHeader = null; 1611 mAvSyncBytesRemaining = 0; 1612 } 1613 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 * @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION} 1632 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1633 * the parameters don't resolve to valid data and indexes, or 1634 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1635 * needs to be recreated. 1636 */ 1637 public int write(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes) { 1638 return write(audioData, offsetInBytes, sizeInBytes, WRITE_BLOCKING); 1639 } 1640 1641 /** 1642 * Writes the audio data to the audio sink for playback (streaming mode), 1643 * or copies audio data for later playback (static buffer mode). 1644 * The format specified in the AudioTrack constructor should be 1645 * {@link AudioFormat#ENCODING_PCM_8BIT} to correspond to the data in the array. 1646 * In streaming mode, will block until all data has been written to the audio sink. 1647 * In static buffer mode, copies the data to the buffer starting at offset 0. 1648 * Note that the actual playback of this data might occur after this function 1649 * returns. This function is thread safe with respect to {@link #stop} calls, 1650 * in which case all of the specified data might not be written to the audio sink. 1651 * 1652 * @param audioData the array that holds the data to play. 1653 * @param offsetInBytes the offset expressed in bytes in audioData where the data to play 1654 * starts. 1655 * @param sizeInBytes the number of bytes to read in audioData after the offset. 1656 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1657 * effect in static mode. 1658 * <br>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1659 * to the audio sink. 1660 * <br>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1661 * queuing as much audio data for playback as possible without blocking. 1662 * @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION} 1663 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1664 * the parameters don't resolve to valid data and indexes, or 1665 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1666 * needs to be recreated. 1667 */ 1668 public int write(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes, 1669 @WriteMode int writeMode) { 1670 1671 if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) { 1672 return ERROR_INVALID_OPERATION; 1673 } 1674 1675 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1676 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1677 return ERROR_BAD_VALUE; 1678 } 1679 1680 if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0) 1681 || (offsetInBytes + sizeInBytes < 0) // detect integer overflow 1682 || (offsetInBytes + sizeInBytes > audioData.length)) { 1683 return ERROR_BAD_VALUE; 1684 } 1685 1686 int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat, 1687 writeMode == WRITE_BLOCKING); 1688 1689 if ((mDataLoadMode == MODE_STATIC) 1690 && (mState == STATE_NO_STATIC_DATA) 1691 && (ret > 0)) { 1692 // benign race with respect to other APIs that read mState 1693 mState = STATE_INITIALIZED; 1694 } 1695 1696 return ret; 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 * @return the number of shorts that were written or {@link #ERROR_INVALID_OPERATION} 1715 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1716 * the parameters don't resolve to valid data and indexes, or 1717 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1718 * needs to be recreated. 1719 */ 1720 public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts) { 1721 return write(audioData, offsetInShorts, sizeInShorts, WRITE_BLOCKING); 1722 } 1723 1724 /** 1725 * Writes the audio data to the audio sink for playback (streaming mode), 1726 * or copies audio data for later playback (static buffer mode). 1727 * The format specified in the AudioTrack constructor should be 1728 * {@link AudioFormat#ENCODING_PCM_16BIT} to correspond to the data in the array. 1729 * In streaming mode, will block until all data has been written to the audio sink. 1730 * In static buffer mode, copies the data to the buffer starting at offset 0. 1731 * Note that the actual playback of this data might occur after this function 1732 * returns. This function is thread safe with respect to {@link #stop} calls, 1733 * in which case all of the specified data might not be written to the audio sink. 1734 * 1735 * @param audioData the array that holds the data to play. 1736 * @param offsetInShorts the offset expressed in shorts in audioData where the data to play 1737 * starts. 1738 * @param sizeInShorts the number of shorts to read in audioData after the offset. 1739 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1740 * effect in static mode. 1741 * <br>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1742 * to the audio sink. 1743 * <br>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1744 * queuing as much audio data for playback as possible without blocking. 1745 * @return the number of shorts that were written or {@link #ERROR_INVALID_OPERATION} 1746 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1747 * the parameters don't resolve to valid data and indexes, or 1748 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1749 * needs to be recreated. 1750 */ 1751 public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts, 1752 @WriteMode int writeMode) { 1753 1754 if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) { 1755 return ERROR_INVALID_OPERATION; 1756 } 1757 1758 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1759 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1760 return ERROR_BAD_VALUE; 1761 } 1762 1763 if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0) 1764 || (offsetInShorts + sizeInShorts < 0) // detect integer overflow 1765 || (offsetInShorts + sizeInShorts > audioData.length)) { 1766 return ERROR_BAD_VALUE; 1767 } 1768 1769 int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat, 1770 writeMode == WRITE_BLOCKING); 1771 1772 if ((mDataLoadMode == MODE_STATIC) 1773 && (mState == STATE_NO_STATIC_DATA) 1774 && (ret > 0)) { 1775 // benign race with respect to other APIs that read mState 1776 mState = STATE_INITIALIZED; 1777 } 1778 1779 return ret; 1780 } 1781 1782 /** 1783 * Writes the audio data to the audio sink for playback (streaming mode), 1784 * or copies audio data for later playback (static buffer mode). 1785 * The format specified in the AudioTrack constructor should be 1786 * {@link AudioFormat#ENCODING_PCM_FLOAT} to correspond to the data in the array. 1787 * In static buffer mode, copies the data to the buffer starting at offset 0, 1788 * and the write mode is ignored. 1789 * In streaming mode, the blocking behavior will depend on the write mode. 1790 * <p> 1791 * Note that the actual playback of this data might occur after this function 1792 * returns. This function is thread safe with respect to {@link #stop} calls, 1793 * in which case all of the specified data might not be written to the audio sink. 1794 * <p> 1795 * @param audioData the array that holds the data to play. 1796 * The implementation does not clip for sample values within the nominal range 1797 * [-1.0f, 1.0f], provided that all gains in the audio pipeline are 1798 * less than or equal to unity (1.0f), and in the absence of post-processing effects 1799 * that could add energy, such as reverb. For the convenience of applications 1800 * that compute samples using filters with non-unity gain, 1801 * sample values +3 dB beyond the nominal range are permitted. 1802 * However such values may eventually be limited or clipped, depending on various gains 1803 * and later processing in the audio path. Therefore applications are encouraged 1804 * to provide samples values within the nominal range. 1805 * @param offsetInFloats the offset, expressed as a number of floats, 1806 * in audioData where the data to play starts. 1807 * @param sizeInFloats the number of floats to read in audioData after the offset. 1808 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1809 * effect in static mode. 1810 * <br>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1811 * to the audio sink. 1812 * <br>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1813 * queuing as much audio data for playback as possible without blocking. 1814 * @return the number of floats that were written, or {@link #ERROR_INVALID_OPERATION} 1815 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1816 * the parameters don't resolve to valid data and indexes, or 1817 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1818 * needs to be recreated. 1819 */ 1820 public int write(@NonNull float[] audioData, int offsetInFloats, int sizeInFloats, 1821 @WriteMode int writeMode) { 1822 1823 if (mState == STATE_UNINITIALIZED) { 1824 Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); 1825 return ERROR_INVALID_OPERATION; 1826 } 1827 1828 if (mAudioFormat != AudioFormat.ENCODING_PCM_FLOAT) { 1829 Log.e(TAG, "AudioTrack.write(float[] ...) requires format ENCODING_PCM_FLOAT"); 1830 return ERROR_INVALID_OPERATION; 1831 } 1832 1833 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1834 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1835 return ERROR_BAD_VALUE; 1836 } 1837 1838 if ( (audioData == null) || (offsetInFloats < 0 ) || (sizeInFloats < 0) 1839 || (offsetInFloats + sizeInFloats < 0) // detect integer overflow 1840 || (offsetInFloats + sizeInFloats > audioData.length)) { 1841 Log.e(TAG, "AudioTrack.write() called with invalid array, offset, or size"); 1842 return ERROR_BAD_VALUE; 1843 } 1844 1845 int ret = native_write_float(audioData, offsetInFloats, sizeInFloats, mAudioFormat, 1846 writeMode == WRITE_BLOCKING); 1847 1848 if ((mDataLoadMode == MODE_STATIC) 1849 && (mState == STATE_NO_STATIC_DATA) 1850 && (ret > 0)) { 1851 // benign race with respect to other APIs that read mState 1852 mState = STATE_INITIALIZED; 1853 } 1854 1855 return ret; 1856 } 1857 1858 1859 /** 1860 * Writes the audio data to the audio sink for playback (streaming mode), 1861 * or copies audio data for later playback (static buffer mode). 1862 * In static buffer mode, copies the data to the buffer starting at its 0 offset, and the write 1863 * mode is ignored. 1864 * In streaming mode, the blocking behavior will depend on the write mode. 1865 * @param audioData the buffer that holds the data to play, starting at the position reported 1866 * by <code>audioData.position()</code>. 1867 * <BR>Note that upon return, the buffer position (<code>audioData.position()</code>) will 1868 * have been advanced to reflect the amount of data that was successfully written to 1869 * the AudioTrack. 1870 * @param sizeInBytes number of bytes to write. 1871 * <BR>Note this may differ from <code>audioData.remaining()</code>, but cannot exceed it. 1872 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1873 * effect in static mode. 1874 * <BR>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1875 * to the audio sink. 1876 * <BR>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1877 * queuing as much audio data for playback as possible without blocking. 1878 * @return 0 or a positive number of bytes that were written, or 1879 * {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION}, or 1880 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1881 * needs to be recreated. 1882 */ 1883 public int write(@NonNull ByteBuffer audioData, int sizeInBytes, 1884 @WriteMode int writeMode) { 1885 1886 if (mState == STATE_UNINITIALIZED) { 1887 Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); 1888 return ERROR_INVALID_OPERATION; 1889 } 1890 1891 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1892 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1893 return ERROR_BAD_VALUE; 1894 } 1895 1896 if ( (audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) { 1897 Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value"); 1898 return ERROR_BAD_VALUE; 1899 } 1900 1901 int ret = 0; 1902 if (audioData.isDirect()) { 1903 ret = native_write_native_bytes(audioData, 1904 audioData.position(), sizeInBytes, mAudioFormat, 1905 writeMode == WRITE_BLOCKING); 1906 } else { 1907 ret = native_write_byte(NioUtils.unsafeArray(audioData), 1908 NioUtils.unsafeArrayOffset(audioData) + audioData.position(), 1909 sizeInBytes, mAudioFormat, 1910 writeMode == WRITE_BLOCKING); 1911 } 1912 1913 if ((mDataLoadMode == MODE_STATIC) 1914 && (mState == STATE_NO_STATIC_DATA) 1915 && (ret > 0)) { 1916 // benign race with respect to other APIs that read mState 1917 mState = STATE_INITIALIZED; 1918 } 1919 1920 if (ret > 0) { 1921 audioData.position(audioData.position() + ret); 1922 } 1923 1924 return ret; 1925 } 1926 1927 /** 1928 * Writes the audio data to the audio sink for playback (streaming mode) on a HW_AV_SYNC track. 1929 * In streaming mode, the blocking behavior will depend on the write mode. 1930 * @param audioData the buffer that holds the data to play, starting at the position reported 1931 * by <code>audioData.position()</code>. 1932 * <BR>Note that upon return, the buffer position (<code>audioData.position()</code>) will 1933 * have been advanced to reflect the amount of data that was successfully written to 1934 * the AudioTrack. 1935 * @param sizeInBytes number of bytes to write. 1936 * <BR>Note this may differ from <code>audioData.remaining()</code>, but cannot exceed it. 1937 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. 1938 * <BR>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1939 * to the audio sink. 1940 * <BR>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1941 * queuing as much audio data for playback as possible without blocking. 1942 * @param timestamp The timestamp of the first decodable audio frame in the provided audioData. 1943 * @return 0 or a positive number of bytes that were written, or 1944 * {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION}, or 1945 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1946 * needs to be recreated. 1947 */ 1948 public int write(ByteBuffer audioData, int sizeInBytes, 1949 @WriteMode int writeMode, long timestamp) { 1950 1951 if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) == 0) { 1952 Log.d(TAG, "AudioTrack.write() called on a regular AudioTrack. Ignoring pts..."); 1953 return write(audioData, sizeInBytes, writeMode); 1954 } 1955 1956 if ((audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) { 1957 Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value"); 1958 return ERROR_BAD_VALUE; 1959 } 1960 1961 // create timestamp header if none exists 1962 if (mAvSyncHeader == null) { 1963 mAvSyncHeader = ByteBuffer.allocate(16); 1964 mAvSyncHeader.order(ByteOrder.BIG_ENDIAN); 1965 mAvSyncHeader.putInt(0x55550001); 1966 mAvSyncHeader.putInt(sizeInBytes); 1967 mAvSyncHeader.putLong(timestamp); 1968 mAvSyncHeader.position(0); 1969 mAvSyncBytesRemaining = sizeInBytes; 1970 } 1971 1972 // write timestamp header if not completely written already 1973 int ret = 0; 1974 if (mAvSyncHeader.remaining() != 0) { 1975 ret = write(mAvSyncHeader, mAvSyncHeader.remaining(), writeMode); 1976 if (ret < 0) { 1977 Log.e(TAG, "AudioTrack.write() could not write timestamp header!"); 1978 mAvSyncHeader = null; 1979 mAvSyncBytesRemaining = 0; 1980 return ret; 1981 } 1982 if (mAvSyncHeader.remaining() > 0) { 1983 Log.v(TAG, "AudioTrack.write() partial timestamp header written."); 1984 return 0; 1985 } 1986 } 1987 1988 // write audio data 1989 int sizeToWrite = Math.min(mAvSyncBytesRemaining, sizeInBytes); 1990 ret = write(audioData, sizeToWrite, writeMode); 1991 if (ret < 0) { 1992 Log.e(TAG, "AudioTrack.write() could not write audio data!"); 1993 mAvSyncHeader = null; 1994 mAvSyncBytesRemaining = 0; 1995 return ret; 1996 } 1997 1998 mAvSyncBytesRemaining -= ret; 1999 if (mAvSyncBytesRemaining == 0) { 2000 mAvSyncHeader = null; 2001 } 2002 2003 return ret; 2004 } 2005 2006 2007 /** 2008 * Sets the playback head position within the static buffer to zero, 2009 * that is it rewinds to start of static buffer. 2010 * The track must be stopped or paused, and 2011 * the track's creation mode must be {@link #MODE_STATIC}. 2012 * <p> 2013 * For API level 22 and above, also resets the value returned by 2014 * {@link #getPlaybackHeadPosition()} to zero. 2015 * For earlier API levels, the reset behavior is unspecified. 2016 * <p> 2017 * {@link #setPlaybackHeadPosition(int)} to zero 2018 * is recommended instead when the reset of {@link #getPlaybackHeadPosition} is not needed. 2019 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 2020 * {@link #ERROR_INVALID_OPERATION} 2021 */ 2022 public int reloadStaticData() { 2023 if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED) { 2024 return ERROR_INVALID_OPERATION; 2025 } 2026 return native_reload_static(); 2027 } 2028 2029 //-------------------------------------------------------------------------- 2030 // Audio effects management 2031 //-------------------- 2032 2033 /** 2034 * Attaches an auxiliary effect to the audio track. A typical auxiliary 2035 * effect is a reverberation effect which can be applied on any sound source 2036 * that directs a certain amount of its energy to this effect. This amount 2037 * is defined by setAuxEffectSendLevel(). 2038 * {@see #setAuxEffectSendLevel(float)}. 2039 * <p>After creating an auxiliary effect (e.g. 2040 * {@link android.media.audiofx.EnvironmentalReverb}), retrieve its ID with 2041 * {@link android.media.audiofx.AudioEffect#getId()} and use it when calling 2042 * this method to attach the audio track to the effect. 2043 * <p>To detach the effect from the audio track, call this method with a 2044 * null effect id. 2045 * 2046 * @param effectId system wide unique id of the effect to attach 2047 * @return error code or success, see {@link #SUCCESS}, 2048 * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR_BAD_VALUE} 2049 */ 2050 public int attachAuxEffect(int effectId) { 2051 if (mState == STATE_UNINITIALIZED) { 2052 return ERROR_INVALID_OPERATION; 2053 } 2054 return native_attachAuxEffect(effectId); 2055 } 2056 2057 /** 2058 * Sets the send level of the audio track to the attached auxiliary effect 2059 * {@link #attachAuxEffect(int)}. Effect levels 2060 * are clamped to the closed interval [0.0, max] where 2061 * max is the value of {@link #getMaxVolume}. 2062 * A value of 0.0 results in no effect, and a value of 1.0 is full send. 2063 * <p>By default the send level is 0.0f, so even if an effect is attached to the player 2064 * this method must be called for the effect to be applied. 2065 * <p>Note that the passed level value is a linear scalar. UI controls should be scaled 2066 * logarithmically: the gain applied by audio framework ranges from -72dB to at least 0dB, 2067 * so an appropriate conversion from linear UI input x to level is: 2068 * x == 0 -> level = 0 2069 * 0 < x <= R -> level = 10^(72*(x-R)/20/R) 2070 * 2071 * @param level linear send level 2072 * @return error code or success, see {@link #SUCCESS}, 2073 * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR} 2074 */ 2075 public int setAuxEffectSendLevel(float level) { 2076 if (isRestricted()) { 2077 return SUCCESS; 2078 } 2079 if (mState == STATE_UNINITIALIZED) { 2080 return ERROR_INVALID_OPERATION; 2081 } 2082 level = clampGainOrLevel(level); 2083 int err = native_setAuxEffectSendLevel(level); 2084 return err == 0 ? SUCCESS : ERROR; 2085 } 2086 2087 //-------------------------------------------------------------------------- 2088 // Explicit Routing 2089 //-------------------- 2090 private AudioDeviceInfo mPreferredDevice = null; 2091 2092 /** 2093 * Specifies an audio device (via an {@link AudioDeviceInfo} object) to route 2094 * the output from this AudioTrack. 2095 * @param deviceInfo The {@link AudioDeviceInfo} specifying the audio sink. 2096 * If deviceInfo is null, default routing is restored. 2097 * @return true if succesful, false if the specified {@link AudioDeviceInfo} is non-null and 2098 * does not correspond to a valid audio output device. 2099 */ 2100 public boolean setPreferredOutputDevice(AudioDeviceInfo deviceInfo) { 2101 // Do some validation.... 2102 if (deviceInfo != null && !deviceInfo.isSink()) { 2103 return false; 2104 } 2105 2106 mPreferredDevice = deviceInfo; 2107 int preferredDeviceId = mPreferredDevice != null ? deviceInfo.getId() : 0; 2108 2109 return native_setOutputDevice(preferredDeviceId); 2110 } 2111 2112 /** 2113 * Returns the selected output specified by {@link #setPreferredOutputDevice}. Note that this 2114 * is not guaranteed to correspond to the actual device being used for playback. 2115 */ 2116 public AudioDeviceInfo getPreferredOutputDevice() { 2117 return mPreferredDevice; 2118 } 2119 2120 //-------------------------------------------------------------------------- 2121 // (Re)Routing Info 2122 //-------------------- 2123 /** 2124 * Returns an {@link AudioDeviceInfo} identifying the current routing of this AudioTrack. 2125 */ 2126 public AudioDeviceInfo getRoutedDevice() { 2127 return null; 2128 } 2129 2130 /** 2131 * The message sent to apps when the routing of this AudioTrack changes if they provide 2132 * a {#link Handler} object to addOnAudioTrackRoutingListener(). 2133 */ 2134 private ArrayMap<OnAudioTrackRoutingListener, NativeRoutingEventHandlerDelegate> 2135 mRoutingChangeListeners = 2136 new ArrayMap<OnAudioTrackRoutingListener, NativeRoutingEventHandlerDelegate>(); 2137 2138 /** 2139 * Adds an {@link OnAudioTrackRoutingListener} to receive notifications of routing changes 2140 * on this AudioTrack. 2141 */ 2142 public void addOnAudioTrackRoutingListener(OnAudioTrackRoutingListener listener, 2143 android.os.Handler handler) { 2144 if (listener != null && !mRoutingChangeListeners.containsKey(listener)) { 2145 synchronized (mRoutingChangeListeners) { 2146 mRoutingChangeListeners.put( 2147 listener, new NativeRoutingEventHandlerDelegate(this, listener, handler)); 2148 } 2149 } 2150 } 2151 2152 /** 2153 * Removes an {@link OnAudioTrackRoutingListener} which has been previously added 2154 * to receive notifications of changes to the set of connected audio devices. 2155 */ 2156 public void removeOnAudioTrackRoutingListener(OnAudioTrackRoutingListener listener) { 2157 synchronized (mRoutingChangeListeners) { 2158 if (mRoutingChangeListeners.containsKey(listener)) { 2159 mRoutingChangeListeners.remove(listener); 2160 } 2161 } 2162 } 2163 2164 /** 2165 * Sends device list change notification to all listeners. 2166 */ 2167 private void broadcastRoutingChange() { 2168 Collection<NativeRoutingEventHandlerDelegate> values; 2169 synchronized (mRoutingChangeListeners) { 2170 values = mRoutingChangeListeners.values(); 2171 } 2172 for(NativeRoutingEventHandlerDelegate delegate : values) { 2173 Handler handler = delegate.getHandler(); 2174 if (handler != null) { 2175 handler.sendEmptyMessage(NATIVE_EVENT_ROUTING_CHANGE); 2176 } 2177 } 2178 } 2179 2180 //--------------------------------------------------------- 2181 // Interface definitions 2182 //-------------------- 2183 /** 2184 * Interface definition for a callback to be invoked when the playback head position of 2185 * an AudioTrack has reached a notification marker or has increased by a certain period. 2186 */ 2187 public interface OnPlaybackPositionUpdateListener { 2188 /** 2189 * Called on the listener to notify it that the previously set marker has been reached 2190 * by the playback head. 2191 */ 2192 void onMarkerReached(AudioTrack track); 2193 2194 /** 2195 * Called on the listener to periodically notify it that the playback head has reached 2196 * a multiple of the notification period. 2197 */ 2198 void onPeriodicNotification(AudioTrack track); 2199 } 2200 2201 //--------------------------------------------------------- 2202 // Inner classes 2203 //-------------------- 2204 /** 2205 * Helper class to handle the forwarding of native events to the appropriate listener 2206 * (potentially) handled in a different thread 2207 */ 2208 private class NativePositionEventHandlerDelegate { 2209 private final Handler mHandler; 2210 2211 NativePositionEventHandlerDelegate(final AudioTrack track, 2212 final OnPlaybackPositionUpdateListener listener, 2213 Handler handler) { 2214 // find the looper for our new event handler 2215 Looper looper; 2216 if (handler != null) { 2217 looper = handler.getLooper(); 2218 } else { 2219 // no given handler, use the looper the AudioTrack was created in 2220 looper = mInitializationLooper; 2221 } 2222 2223 // construct the event handler with this looper 2224 if (looper != null) { 2225 // implement the event handler delegate 2226 mHandler = new Handler(looper) { 2227 @Override 2228 public void handleMessage(Message msg) { 2229 if (track == null) { 2230 return; 2231 } 2232 switch(msg.what) { 2233 case NATIVE_EVENT_MARKER: 2234 if (listener != null) { 2235 listener.onMarkerReached(track); 2236 } 2237 break; 2238 case NATIVE_EVENT_NEW_POS: 2239 if (listener != null) { 2240 listener.onPeriodicNotification(track); 2241 } 2242 break; 2243 default: 2244 loge("Unknown native event type: " + msg.what); 2245 break; 2246 } 2247 } 2248 }; 2249 } else { 2250 mHandler = null; 2251 } 2252 } 2253 2254 Handler getHandler() { 2255 return mHandler; 2256 } 2257 } 2258 2259 /** 2260 * Helper class to handle the forwarding of native events to the appropriate listener 2261 * (potentially) handled in a different thread 2262 */ 2263 private class NativeRoutingEventHandlerDelegate { 2264 private final Handler mHandler; 2265 2266 NativeRoutingEventHandlerDelegate(final AudioTrack track, 2267 final OnAudioTrackRoutingListener listener, 2268 Handler handler) { 2269 // find the looper for our new event handler 2270 Looper looper; 2271 if (handler != null) { 2272 looper = handler.getLooper(); 2273 } else { 2274 // no given handler, use the looper the AudioTrack was created in 2275 looper = mInitializationLooper; 2276 } 2277 2278 // construct the event handler with this looper 2279 if (looper != null) { 2280 // implement the event handler delegate 2281 mHandler = new Handler(looper) { 2282 @Override 2283 public void handleMessage(Message msg) { 2284 if (track == null) { 2285 return; 2286 } 2287 switch(msg.what) { 2288 case NATIVE_EVENT_ROUTING_CHANGE: 2289 if (listener != null) { 2290 listener.onAudioTrackRouting(track); 2291 } 2292 break; 2293 default: 2294 loge("Unknown native event type: " + msg.what); 2295 break; 2296 } 2297 } 2298 }; 2299 } else { 2300 mHandler = null; 2301 } 2302 } 2303 2304 Handler getHandler() { 2305 return mHandler; 2306 } 2307 } 2308 2309 //--------------------------------------------------------- 2310 // Java methods called from the native side 2311 //-------------------- 2312 @SuppressWarnings("unused") 2313 private static void postEventFromNative(Object audiotrack_ref, 2314 int what, int arg1, int arg2, Object obj) { 2315 //logd("Event posted from the native side: event="+ what + " args="+ arg1+" "+arg2); 2316 AudioTrack track = (AudioTrack)((WeakReference)audiotrack_ref).get(); 2317 if (track == null) { 2318 return; 2319 } 2320 2321 NativePositionEventHandlerDelegate delegate = track.mEventHandlerDelegate; 2322 if (delegate != null) { 2323 Handler handler = delegate.getHandler(); 2324 if (handler != null) { 2325 Message m = handler.obtainMessage(what, arg1, arg2, obj); 2326 handler.sendMessage(m); 2327 } 2328 } 2329 2330 } 2331 2332 2333 //--------------------------------------------------------- 2334 // Native methods called from the Java side 2335 //-------------------- 2336 2337 // post-condition: mStreamType is overwritten with a value 2338 // that reflects the audio attributes (e.g. an AudioAttributes object with a usage of 2339 // AudioAttributes.USAGE_MEDIA will map to AudioManager.STREAM_MUSIC 2340 private native final int native_setup(Object /*WeakReference<AudioTrack>*/ audiotrack_this, 2341 Object /*AudioAttributes*/ attributes, 2342 int sampleRate, int channelMask, int audioFormat, 2343 int buffSizeInBytes, int mode, int[] sessionId); 2344 2345 private native final void native_finalize(); 2346 2347 private native final void native_release(); 2348 2349 private native final void native_start(); 2350 2351 private native final void native_stop(); 2352 2353 private native final void native_pause(); 2354 2355 private native final void native_flush(); 2356 2357 private native final int native_write_byte(byte[] audioData, 2358 int offsetInBytes, int sizeInBytes, int format, 2359 boolean isBlocking); 2360 2361 private native final int native_write_short(short[] audioData, 2362 int offsetInShorts, int sizeInShorts, int format, 2363 boolean isBlocking); 2364 2365 private native final int native_write_float(float[] audioData, 2366 int offsetInFloats, int sizeInFloats, int format, 2367 boolean isBlocking); 2368 2369 private native final int native_write_native_bytes(Object audioData, 2370 int positionInBytes, int sizeInBytes, int format, boolean blocking); 2371 2372 private native final int native_reload_static(); 2373 2374 private native final int native_get_native_frame_count(); 2375 2376 private native final void native_setVolume(float leftVolume, float rightVolume); 2377 2378 private native final int native_set_playback_rate(int sampleRateInHz); 2379 private native final int native_get_playback_rate(); 2380 2381 // floatArray must be a non-null array of length >= 2 2382 // [0] is speed 2383 // [1] is pitch 2384 // intArray must be a non-null array of length >= 2 2385 // [0] is audio fallback mode 2386 // [1] is audio stretch mode 2387 private native final void native_set_playback_settings(float[] floatArray, int[] intArray); 2388 private native final void native_get_playback_settings(float[] floatArray, int[] intArray); 2389 2390 private native final int native_set_marker_pos(int marker); 2391 private native final int native_get_marker_pos(); 2392 2393 private native final int native_set_pos_update_period(int updatePeriod); 2394 private native final int native_get_pos_update_period(); 2395 2396 private native final int native_set_position(int position); 2397 private native final int native_get_position(); 2398 2399 private native final int native_get_latency(); 2400 2401 // longArray must be a non-null array of length >= 2 2402 // [0] is assigned the frame position 2403 // [1] is assigned the time in CLOCK_MONOTONIC nanoseconds 2404 private native final int native_get_timestamp(long[] longArray); 2405 2406 private native final int native_set_loop(int start, int end, int loopCount); 2407 2408 static private native final int native_get_output_sample_rate(int streamType); 2409 static private native final int native_get_min_buff_size( 2410 int sampleRateInHz, int channelConfig, int audioFormat); 2411 2412 private native final int native_attachAuxEffect(int effectId); 2413 private native final int native_setAuxEffectSendLevel(float level); 2414 2415 private native final boolean native_setOutputDevice(int deviceId); 2416 2417 //--------------------------------------------------------- 2418 // Utility methods 2419 //------------------ 2420 2421 private static void logd(String msg) { 2422 Log.d(TAG, msg); 2423 } 2424 2425 private static void loge(String msg) { 2426 Log.e(TAG, msg); 2427 } 2428} 2429