AudioTrack.java revision ff0d9f098e51c54e1a030ed21fd980680cb7b405
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.nio.ByteBuffer; 23import java.nio.NioUtils; 24 25import android.annotation.IntDef; 26import android.app.ActivityThread; 27import android.app.AppOpsManager; 28import android.content.Context; 29import android.os.Handler; 30import android.os.IBinder; 31import android.os.Looper; 32import android.os.Message; 33import android.os.Process; 34import android.os.RemoteException; 35import android.os.ServiceManager; 36import android.util.Log; 37 38import com.android.internal.app.IAppOpsService; 39 40 41/** 42 * The AudioTrack class manages and plays a single audio resource for Java applications. 43 * It allows streaming of PCM audio buffers to the audio sink for playback. This is 44 * achieved by "pushing" the data to the AudioTrack object using one of the 45 * {@link #write(byte[], int, int)}, {@link #write(short[], int, int)}, 46 * and {@link #write(float[], int, int, int)} methods. 47 * 48 * <p>An AudioTrack instance can operate under two modes: static or streaming.<br> 49 * In Streaming mode, the application writes a continuous stream of data to the AudioTrack, using 50 * one of the {@code write()} methods. These are blocking and return when the data has been 51 * transferred from the Java layer to the native layer and queued for playback. The streaming 52 * mode is most useful when playing blocks of audio data that for instance are: 53 * 54 * <ul> 55 * <li>too big to fit in memory because of the duration of the sound to play,</li> 56 * <li>too big to fit in memory because of the characteristics of the audio data 57 * (high sampling rate, bits per sample ...)</li> 58 * <li>received or generated while previously queued audio is playing.</li> 59 * </ul> 60 * 61 * The static mode should be chosen when dealing with short sounds that fit in memory and 62 * that need to be played with the smallest latency possible. The static mode will 63 * therefore be preferred for UI and game sounds that are played often, and with the 64 * smallest overhead possible. 65 * 66 * <p>Upon creation, an AudioTrack object initializes its associated audio buffer. 67 * The size of this buffer, specified during the construction, determines how long an AudioTrack 68 * can play before running out of data.<br> 69 * For an AudioTrack using the static mode, this size is the maximum size of the sound that can 70 * be played from it.<br> 71 * For the streaming mode, data will be written to the audio sink in chunks of 72 * sizes less than or equal to the total buffer size. 73 * 74 * AudioTrack is not final and thus permits subclasses, but such use is not recommended. 75 */ 76public class AudioTrack 77{ 78 //--------------------------------------------------------- 79 // Constants 80 //-------------------- 81 /** Minimum value for a linear gain or auxiliary effect level. 82 * This value must be exactly equal to 0.0f; do not change it. 83 */ 84 private static final float GAIN_MIN = 0.0f; 85 /** Maximum value for a linear gain or auxiliary effect level. 86 * This value must be greater than or equal to 1.0f. 87 */ 88 private static final float GAIN_MAX = 1.0f; 89 90 /** Minimum value for sample rate */ 91 private static final int SAMPLE_RATE_HZ_MIN = 4000; 92 /** Maximum value for sample rate */ 93 private static final int SAMPLE_RATE_HZ_MAX = 48000; 94 95 /** indicates AudioTrack state is stopped */ 96 public static final int PLAYSTATE_STOPPED = 1; // matches SL_PLAYSTATE_STOPPED 97 /** indicates AudioTrack state is paused */ 98 public static final int PLAYSTATE_PAUSED = 2; // matches SL_PLAYSTATE_PAUSED 99 /** indicates AudioTrack state is playing */ 100 public static final int PLAYSTATE_PLAYING = 3; // matches SL_PLAYSTATE_PLAYING 101 102 // keep these values in sync with android_media_AudioTrack.cpp 103 /** 104 * Creation mode where audio data is transferred from Java to the native layer 105 * only once before the audio starts playing. 106 */ 107 public static final int MODE_STATIC = 0; 108 /** 109 * Creation mode where audio data is streamed from Java to the native layer 110 * as the audio is playing. 111 */ 112 public static final int MODE_STREAM = 1; 113 114 /** 115 * State of an AudioTrack that was not successfully initialized upon creation. 116 */ 117 public static final int STATE_UNINITIALIZED = 0; 118 /** 119 * State of an AudioTrack that is ready to be used. 120 */ 121 public static final int STATE_INITIALIZED = 1; 122 /** 123 * State of a successfully initialized AudioTrack that uses static data, 124 * but that hasn't received that data yet. 125 */ 126 public static final int STATE_NO_STATIC_DATA = 2; 127 128 /** 129 * Denotes a successful operation. 130 */ 131 public static final int SUCCESS = AudioSystem.SUCCESS; 132 /** 133 * Denotes a generic operation failure. 134 */ 135 public static final int ERROR = AudioSystem.ERROR; 136 /** 137 * Denotes a failure due to the use of an invalid value. 138 */ 139 public static final int ERROR_BAD_VALUE = AudioSystem.BAD_VALUE; 140 /** 141 * Denotes a failure due to the improper use of a method. 142 */ 143 public static final int ERROR_INVALID_OPERATION = AudioSystem.INVALID_OPERATION; 144 145 // Error codes: 146 // to keep in sync with frameworks/base/core/jni/android_media_AudioTrack.cpp 147 private static final int ERROR_NATIVESETUP_AUDIOSYSTEM = -16; 148 private static final int ERROR_NATIVESETUP_INVALIDCHANNELMASK = -17; 149 private static final int ERROR_NATIVESETUP_INVALIDFORMAT = -18; 150 private static final int ERROR_NATIVESETUP_INVALIDSTREAMTYPE = -19; 151 private static final int ERROR_NATIVESETUP_NATIVEINITFAILED = -20; 152 153 // Events: 154 // to keep in sync with frameworks/av/include/media/AudioTrack.h 155 /** 156 * Event id denotes when playback head has reached a previously set marker. 157 */ 158 private static final int NATIVE_EVENT_MARKER = 3; 159 /** 160 * Event id denotes when previously set update period has elapsed during playback. 161 */ 162 private static final int NATIVE_EVENT_NEW_POS = 4; 163 164 private final static String TAG = "android.media.AudioTrack"; 165 166 167 /** @hide */ 168 @IntDef({ 169 WRITE_BLOCKING, 170 WRITE_NON_BLOCKING 171 }) 172 @Retention(RetentionPolicy.SOURCE) 173 public @interface WriteMode {} 174 175 /** 176 * The write mode indicating the write operation will block until all data has been written, 177 * to be used in {@link #write(ByteBuffer, int, int)} 178 */ 179 public final static int WRITE_BLOCKING = 0; 180 /** 181 * The write mode indicating the write operation will return immediately after 182 * queuing as much audio data for playback as possible without blocking, to be used in 183 * {@link #write(ByteBuffer, int, int)}. 184 */ 185 public final static int WRITE_NON_BLOCKING = 1; 186 187 //-------------------------------------------------------------------------- 188 // Member variables 189 //-------------------- 190 /** 191 * Indicates the state of the AudioTrack instance. 192 */ 193 private int mState = STATE_UNINITIALIZED; 194 /** 195 * Indicates the play state of the AudioTrack instance. 196 */ 197 private int mPlayState = PLAYSTATE_STOPPED; 198 /** 199 * Lock to make sure mPlayState updates are reflecting the actual state of the object. 200 */ 201 private final Object mPlayStateLock = new Object(); 202 /** 203 * Sizes of the native audio buffer. 204 */ 205 private int mNativeBufferSizeInBytes = 0; 206 private int mNativeBufferSizeInFrames = 0; 207 /** 208 * Handler for events coming from the native code. 209 */ 210 private NativeEventHandlerDelegate mEventHandlerDelegate; 211 /** 212 * Looper associated with the thread that creates the AudioTrack instance. 213 */ 214 private final Looper mInitializationLooper; 215 /** 216 * The audio data source sampling rate in Hz. 217 */ 218 private int mSampleRate; // initialized by all constructors 219 /** 220 * The number of audio output channels (1 is mono, 2 is stereo). 221 */ 222 private int mChannelCount = 1; 223 /** 224 * The audio channel mask. 225 */ 226 private int mChannels = AudioFormat.CHANNEL_OUT_MONO; 227 228 /** 229 * The type of the audio stream to play. See 230 * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, 231 * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, 232 * {@link AudioManager#STREAM_ALARM}, {@link AudioManager#STREAM_NOTIFICATION}, and 233 * {@link AudioManager#STREAM_DTMF}. 234 */ 235 private int mStreamType = AudioManager.STREAM_MUSIC; 236 /** 237 * The way audio is consumed by the audio sink, streaming or static. 238 */ 239 private int mDataLoadMode = MODE_STREAM; 240 /** 241 * The current audio channel configuration. 242 */ 243 private int mChannelConfiguration = AudioFormat.CHANNEL_OUT_MONO; 244 /** 245 * The encoding of the audio samples. 246 * @see AudioFormat#ENCODING_PCM_8BIT 247 * @see AudioFormat#ENCODING_PCM_16BIT 248 * @see AudioFormat#ENCODING_PCM_FLOAT 249 */ 250 private int mAudioFormat = AudioFormat.ENCODING_PCM_16BIT; 251 /** 252 * Audio session ID 253 */ 254 private int mSessionId = AudioSystem.AUDIO_SESSION_ALLOCATE; 255 /** 256 * Reference to the app-ops service. 257 */ 258 private final IAppOpsService mAppOps; 259 260 //-------------------------------- 261 // Used exclusively by native code 262 //-------------------- 263 /** 264 * Accessed by native methods: provides access to C++ AudioTrack object. 265 */ 266 @SuppressWarnings("unused") 267 private long mNativeTrackInJavaObj; 268 /** 269 * Accessed by native methods: provides access to the JNI data (i.e. resources used by 270 * the native AudioTrack object, but not stored in it). 271 */ 272 @SuppressWarnings("unused") 273 private long mJniData; 274 275 276 //-------------------------------------------------------------------------- 277 // Constructor, Finalize 278 //-------------------- 279 /** 280 * Class constructor. 281 * @param streamType the type of the audio stream. See 282 * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, 283 * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, 284 * {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}. 285 * @param sampleRateInHz the initial source sample rate expressed in Hz. 286 * @param channelConfig describes the configuration of the audio channels. 287 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 288 * {@link AudioFormat#CHANNEL_OUT_STEREO} 289 * @param audioFormat the format in which the audio data is represented. 290 * See {@link AudioFormat#ENCODING_PCM_16BIT}, 291 * {@link AudioFormat#ENCODING_PCM_8BIT}, 292 * and {@link AudioFormat#ENCODING_PCM_FLOAT}. 293 * @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is 294 * read from for playback. 295 * If track's creation mode is {@link #MODE_STREAM}, you can write data into 296 * this buffer in chunks less than or equal to this size, and it is typical to use 297 * chunks of 1/2 of the total size to permit double-buffering. 298 * If the track's creation mode is {@link #MODE_STATIC}, 299 * this is the maximum length sample, or audio clip, that can be played by this instance. 300 * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size 301 * for the successful creation of an AudioTrack instance in streaming mode. Using values 302 * smaller than getMinBufferSize() will result in an initialization failure. 303 * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM} 304 * @throws java.lang.IllegalArgumentException 305 */ 306 public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, 307 int bufferSizeInBytes, int mode) 308 throws IllegalArgumentException { 309 this(streamType, sampleRateInHz, channelConfig, audioFormat, 310 bufferSizeInBytes, mode, AudioSystem.AUDIO_SESSION_ALLOCATE); 311 } 312 313 /** 314 * Class constructor with audio session. Use this constructor when the AudioTrack must be 315 * attached to a particular audio session. The primary use of the audio session ID is to 316 * associate audio effects to a particular instance of AudioTrack: if an audio session ID 317 * is provided when creating an AudioEffect, this effect will be applied only to audio tracks 318 * and media players in the same session and not to the output mix. 319 * When an AudioTrack is created without specifying a session, it will create its own session 320 * which can be retrieved by calling the {@link #getAudioSessionId()} method. 321 * If a non-zero session ID is provided, this AudioTrack will share effects attached to this 322 * session 323 * with all other media players or audio tracks in the same session, otherwise a new session 324 * will be created for this track if none is supplied. 325 * @param streamType the type of the audio stream. See 326 * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, 327 * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, 328 * {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}. 329 * @param sampleRateInHz the initial source sample rate expressed in Hz. 330 * @param channelConfig describes the configuration of the audio channels. 331 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 332 * {@link AudioFormat#CHANNEL_OUT_STEREO} 333 * @param audioFormat the format in which the audio data is represented. 334 * See {@link AudioFormat#ENCODING_PCM_16BIT} and 335 * {@link AudioFormat#ENCODING_PCM_8BIT}, 336 * and {@link AudioFormat#ENCODING_PCM_FLOAT}. 337 * @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read 338 * from for playback. If using the AudioTrack in streaming mode, you can write data into 339 * this buffer in smaller chunks than this size. If using the AudioTrack in static mode, 340 * this is the maximum size of the sound that will be played for this instance. 341 * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size 342 * for the successful creation of an AudioTrack instance in streaming mode. Using values 343 * smaller than getMinBufferSize() will result in an initialization failure. 344 * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM} 345 * @param sessionId Id of audio session the AudioTrack must be attached to 346 * @throws java.lang.IllegalArgumentException 347 */ 348 public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, 349 int bufferSizeInBytes, int mode, int sessionId) 350 throws IllegalArgumentException { 351 // mState already == STATE_UNINITIALIZED 352 353 // remember which looper is associated with the AudioTrack instantiation 354 Looper looper; 355 if ((looper = Looper.myLooper()) == null) { 356 looper = Looper.getMainLooper(); 357 } 358 mInitializationLooper = looper; 359 360 audioParamCheck(streamType, sampleRateInHz, channelConfig, audioFormat, mode); 361 362 audioBuffSizeCheck(bufferSizeInBytes); 363 364 IBinder b = ServiceManager.getService(Context.APP_OPS_SERVICE); 365 mAppOps = IAppOpsService.Stub.asInterface(b); 366 367 if (sessionId < 0) { 368 throw new IllegalArgumentException("Invalid audio session ID: "+sessionId); 369 } 370 371 int[] session = new int[1]; 372 session[0] = sessionId; 373 // native initialization 374 int initResult = native_setup(new WeakReference<AudioTrack>(this), 375 mStreamType, mSampleRate, mChannels, mAudioFormat, 376 mNativeBufferSizeInBytes, mDataLoadMode, session); 377 if (initResult != SUCCESS) { 378 loge("Error code "+initResult+" when initializing AudioTrack."); 379 return; // with mState == STATE_UNINITIALIZED 380 } 381 382 mSessionId = session[0]; 383 384 if (mDataLoadMode == MODE_STATIC) { 385 mState = STATE_NO_STATIC_DATA; 386 } else { 387 mState = STATE_INITIALIZED; 388 } 389 } 390 391 // mask of all the channels supported by this implementation 392 private static final int SUPPORTED_OUT_CHANNELS = 393 AudioFormat.CHANNEL_OUT_FRONT_LEFT | 394 AudioFormat.CHANNEL_OUT_FRONT_RIGHT | 395 AudioFormat.CHANNEL_OUT_FRONT_CENTER | 396 AudioFormat.CHANNEL_OUT_LOW_FREQUENCY | 397 AudioFormat.CHANNEL_OUT_BACK_LEFT | 398 AudioFormat.CHANNEL_OUT_BACK_RIGHT | 399 AudioFormat.CHANNEL_OUT_BACK_CENTER; 400 401 // Convenience method for the constructor's parameter checks. 402 // This is where constructor IllegalArgumentException-s are thrown 403 // postconditions: 404 // mStreamType is valid 405 // mChannelCount is valid 406 // mChannels is valid 407 // mAudioFormat is valid 408 // mSampleRate is valid 409 // mDataLoadMode is valid 410 private void audioParamCheck(int streamType, int sampleRateInHz, 411 int channelConfig, int audioFormat, int mode) { 412 413 //-------------- 414 // stream type 415 if( (streamType != AudioManager.STREAM_ALARM) && (streamType != AudioManager.STREAM_MUSIC) 416 && (streamType != AudioManager.STREAM_RING) && (streamType != AudioManager.STREAM_SYSTEM) 417 && (streamType != AudioManager.STREAM_VOICE_CALL) 418 && (streamType != AudioManager.STREAM_NOTIFICATION) 419 && (streamType != AudioManager.STREAM_BLUETOOTH_SCO) 420 && (streamType != AudioManager.STREAM_DTMF)) { 421 throw new IllegalArgumentException("Invalid stream type."); 422 } 423 mStreamType = streamType; 424 425 //-------------- 426 // sample rate, note these values are subject to change 427 if ( (sampleRateInHz < 4000) || (sampleRateInHz > 48000) ) { 428 throw new IllegalArgumentException(sampleRateInHz 429 + "Hz is not a supported sample rate."); 430 } 431 mSampleRate = sampleRateInHz; 432 433 //-------------- 434 // channel config 435 mChannelConfiguration = channelConfig; 436 437 switch (channelConfig) { 438 case AudioFormat.CHANNEL_OUT_DEFAULT: //AudioFormat.CHANNEL_CONFIGURATION_DEFAULT 439 case AudioFormat.CHANNEL_OUT_MONO: 440 case AudioFormat.CHANNEL_CONFIGURATION_MONO: 441 mChannelCount = 1; 442 mChannels = AudioFormat.CHANNEL_OUT_MONO; 443 break; 444 case AudioFormat.CHANNEL_OUT_STEREO: 445 case AudioFormat.CHANNEL_CONFIGURATION_STEREO: 446 mChannelCount = 2; 447 mChannels = AudioFormat.CHANNEL_OUT_STEREO; 448 break; 449 default: 450 if (!isMultichannelConfigSupported(channelConfig)) { 451 // input channel configuration features unsupported channels 452 throw new IllegalArgumentException("Unsupported channel configuration."); 453 } 454 mChannels = channelConfig; 455 mChannelCount = Integer.bitCount(channelConfig); 456 } 457 458 //-------------- 459 // audio format 460 if (audioFormat == AudioFormat.ENCODING_DEFAULT) { 461 audioFormat = AudioFormat.ENCODING_PCM_16BIT; 462 } 463 464 if (!AudioFormat.isValidEncoding(audioFormat)) { 465 throw new IllegalArgumentException("Unsupported audio encoding."); 466 } 467 mAudioFormat = audioFormat; 468 469 //-------------- 470 // audio load mode 471 if (((mode != MODE_STREAM) && (mode != MODE_STATIC)) || 472 ((mode != MODE_STREAM) && !AudioFormat.isEncodingLinearPcm(mAudioFormat))) { 473 throw new IllegalArgumentException("Invalid mode."); 474 } 475 mDataLoadMode = mode; 476 } 477 478 /** 479 * Convenience method to check that the channel configuration (a.k.a channel mask) is supported 480 * @param channelConfig the mask to validate 481 * @return false if the AudioTrack can't be used with such a mask 482 */ 483 private static boolean isMultichannelConfigSupported(int channelConfig) { 484 // check for unsupported channels 485 if ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) { 486 loge("Channel configuration features unsupported channels"); 487 return false; 488 } 489 // check for unsupported multichannel combinations: 490 // - FL/FR must be present 491 // - L/R channels must be paired (e.g. no single L channel) 492 final int frontPair = 493 AudioFormat.CHANNEL_OUT_FRONT_LEFT | AudioFormat.CHANNEL_OUT_FRONT_RIGHT; 494 if ((channelConfig & frontPair) != frontPair) { 495 loge("Front channels must be present in multichannel configurations"); 496 return false; 497 } 498 final int backPair = 499 AudioFormat.CHANNEL_OUT_BACK_LEFT | AudioFormat.CHANNEL_OUT_BACK_RIGHT; 500 if ((channelConfig & backPair) != 0) { 501 if ((channelConfig & backPair) != backPair) { 502 loge("Rear channels can't be used independently"); 503 return false; 504 } 505 } 506 return true; 507 } 508 509 510 // Convenience method for the constructor's audio buffer size check. 511 // preconditions: 512 // mChannelCount is valid 513 // mAudioFormat is valid 514 // postcondition: 515 // mNativeBufferSizeInBytes is valid (multiple of frame size, positive) 516 private void audioBuffSizeCheck(int audioBufferSize) { 517 // NB: this section is only valid with PCM data. 518 // To update when supporting compressed formats 519 int frameSizeInBytes; 520 if (AudioFormat.isEncodingLinearPcm(mAudioFormat)) { 521 frameSizeInBytes = mChannelCount 522 * (AudioFormat.getBytesPerSample(mAudioFormat)); 523 } else { 524 frameSizeInBytes = 1; 525 } 526 if ((audioBufferSize % frameSizeInBytes != 0) || (audioBufferSize < 1)) { 527 throw new IllegalArgumentException("Invalid audio buffer size."); 528 } 529 530 mNativeBufferSizeInBytes = audioBufferSize; 531 mNativeBufferSizeInFrames = audioBufferSize / frameSizeInBytes; 532 } 533 534 535 /** 536 * Releases the native AudioTrack resources. 537 */ 538 public void release() { 539 // even though native_release() stops the native AudioTrack, we need to stop 540 // AudioTrack subclasses too. 541 try { 542 stop(); 543 } catch(IllegalStateException ise) { 544 // don't raise an exception, we're releasing the resources. 545 } 546 native_release(); 547 mState = STATE_UNINITIALIZED; 548 } 549 550 @Override 551 protected void finalize() { 552 native_finalize(); 553 } 554 555 //-------------------------------------------------------------------------- 556 // Getters 557 //-------------------- 558 /** 559 * Returns the minimum gain value, which is the constant 0.0. 560 * Gain values less than 0.0 will be clamped to 0.0. 561 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 562 * @return the minimum value, which is the constant 0.0. 563 */ 564 static public float getMinVolume() { 565 return GAIN_MIN; 566 } 567 568 /** 569 * Returns the maximum gain value, which is greater than or equal to 1.0. 570 * Gain values greater than the maximum will be clamped to the maximum. 571 * <p>The word "volume" in the API name is historical; this is actually a gain. 572 * expressed as a linear multiplier on sample values, where a maximum value of 1.0 573 * corresponds to a gain of 0 dB (sample values left unmodified). 574 * @return the maximum value, which is greater than or equal to 1.0. 575 */ 576 static public float getMaxVolume() { 577 return GAIN_MAX; 578 } 579 580 /** 581 * Returns the configured audio data sample rate in Hz 582 */ 583 public int getSampleRate() { 584 return mSampleRate; 585 } 586 587 /** 588 * Returns the current playback rate in Hz. 589 */ 590 public int getPlaybackRate() { 591 return native_get_playback_rate(); 592 } 593 594 /** 595 * Returns the configured audio data format. See {@link AudioFormat#ENCODING_PCM_16BIT} 596 * and {@link AudioFormat#ENCODING_PCM_8BIT}. 597 */ 598 public int getAudioFormat() { 599 return mAudioFormat; 600 } 601 602 /** 603 * Returns the type of audio stream this AudioTrack is configured for. 604 * Compare the result against {@link AudioManager#STREAM_VOICE_CALL}, 605 * {@link AudioManager#STREAM_SYSTEM}, {@link AudioManager#STREAM_RING}, 606 * {@link AudioManager#STREAM_MUSIC}, {@link AudioManager#STREAM_ALARM}, 607 * {@link AudioManager#STREAM_NOTIFICATION}, or {@link AudioManager#STREAM_DTMF}. 608 */ 609 public int getStreamType() { 610 return mStreamType; 611 } 612 613 /** 614 * Returns the configured channel configuration. 615 * See {@link AudioFormat#CHANNEL_OUT_MONO} 616 * and {@link AudioFormat#CHANNEL_OUT_STEREO}. 617 */ 618 public int getChannelConfiguration() { 619 return mChannelConfiguration; 620 } 621 622 /** 623 * Returns the configured number of channels. 624 */ 625 public int getChannelCount() { 626 return mChannelCount; 627 } 628 629 /** 630 * Returns the state of the AudioTrack instance. This is useful after the 631 * AudioTrack instance has been created to check if it was initialized 632 * properly. This ensures that the appropriate resources have been acquired. 633 * @see #STATE_INITIALIZED 634 * @see #STATE_NO_STATIC_DATA 635 * @see #STATE_UNINITIALIZED 636 */ 637 public int getState() { 638 return mState; 639 } 640 641 /** 642 * Returns the playback state of the AudioTrack instance. 643 * @see #PLAYSTATE_STOPPED 644 * @see #PLAYSTATE_PAUSED 645 * @see #PLAYSTATE_PLAYING 646 */ 647 public int getPlayState() { 648 synchronized (mPlayStateLock) { 649 return mPlayState; 650 } 651 } 652 653 /** 654 * Returns the "native frame count", derived from the bufferSizeInBytes specified at 655 * creation time and converted to frame units. 656 * If track's creation mode is {@link #MODE_STATIC}, 657 * it is equal to the specified bufferSizeInBytes converted to frame units. 658 * If track's creation mode is {@link #MODE_STREAM}, 659 * it is typically greater than or equal to the specified bufferSizeInBytes converted to frame 660 * units; it may be rounded up to a larger value if needed by the target device implementation. 661 * @deprecated Only accessible by subclasses, which are not recommended for AudioTrack. 662 * See {@link AudioManager#getProperty(String)} for key 663 * {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}. 664 */ 665 @Deprecated 666 protected int getNativeFrameCount() { 667 return native_get_native_frame_count(); 668 } 669 670 /** 671 * Returns marker position expressed in frames. 672 * @return marker position in wrapping frame units similar to {@link #getPlaybackHeadPosition}, 673 * or zero if marker is disabled. 674 */ 675 public int getNotificationMarkerPosition() { 676 return native_get_marker_pos(); 677 } 678 679 /** 680 * Returns the notification update period expressed in frames. 681 * Zero means that no position update notifications are being delivered. 682 */ 683 public int getPositionNotificationPeriod() { 684 return native_get_pos_update_period(); 685 } 686 687 /** 688 * Returns the playback head position expressed in frames. 689 * Though the "int" type is signed 32-bits, the value should be reinterpreted as if it is 690 * unsigned 32-bits. That is, the next position after 0x7FFFFFFF is (int) 0x80000000. 691 * This is a continuously advancing counter. It will wrap (overflow) periodically, 692 * for example approximately once every 27:03:11 hours:minutes:seconds at 44.1 kHz. 693 * It is reset to zero by flush(), reload(), and stop(). 694 */ 695 public int getPlaybackHeadPosition() { 696 return native_get_position(); 697 } 698 699 /** 700 * Returns this track's estimated latency in milliseconds. This includes the latency due 701 * to AudioTrack buffer size, AudioMixer (if any) and audio hardware driver. 702 * 703 * DO NOT UNHIDE. The existing approach for doing A/V sync has too many problems. We need 704 * a better solution. 705 * @hide 706 */ 707 public int getLatency() { 708 return native_get_latency(); 709 } 710 711 /** 712 * Returns the output sample rate in Hz for the specified stream type. 713 */ 714 static public int getNativeOutputSampleRate(int streamType) { 715 return native_get_output_sample_rate(streamType); 716 } 717 718 /** 719 * Returns the minimum buffer size required for the successful creation of an AudioTrack 720 * object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't 721 * guarantee a smooth playback under load, and higher values should be chosen according to 722 * the expected frequency at which the buffer will be refilled with additional data to play. 723 * For example, if you intend to dynamically set the source sample rate of an AudioTrack 724 * to a higher value than the initial source sample rate, be sure to configure the buffer size 725 * based on the highest planned sample rate. 726 * @param sampleRateInHz the source sample rate expressed in Hz. 727 * @param channelConfig describes the configuration of the audio channels. 728 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 729 * {@link AudioFormat#CHANNEL_OUT_STEREO} 730 * @param audioFormat the format in which the audio data is represented. 731 * See {@link AudioFormat#ENCODING_PCM_16BIT} and 732 * {@link AudioFormat#ENCODING_PCM_8BIT}, 733 * and {@link AudioFormat#ENCODING_PCM_FLOAT}. 734 * @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed, 735 * or {@link #ERROR} if unable to query for output properties, 736 * or the minimum buffer size expressed in bytes. 737 */ 738 static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) { 739 int channelCount = 0; 740 switch(channelConfig) { 741 case AudioFormat.CHANNEL_OUT_MONO: 742 case AudioFormat.CHANNEL_CONFIGURATION_MONO: 743 channelCount = 1; 744 break; 745 case AudioFormat.CHANNEL_OUT_STEREO: 746 case AudioFormat.CHANNEL_CONFIGURATION_STEREO: 747 channelCount = 2; 748 break; 749 default: 750 if ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) { 751 // input channel configuration features unsupported channels 752 loge("getMinBufferSize(): Invalid channel configuration."); 753 return ERROR_BAD_VALUE; 754 } else { 755 channelCount = Integer.bitCount(channelConfig); 756 } 757 } 758 759 if (!AudioFormat.isValidEncoding(audioFormat)) { 760 loge("getMinBufferSize(): Invalid audio format."); 761 return ERROR_BAD_VALUE; 762 } 763 764 // sample rate, note these values are subject to change 765 if ( (sampleRateInHz < SAMPLE_RATE_HZ_MIN) || (sampleRateInHz > SAMPLE_RATE_HZ_MAX) ) { 766 loge("getMinBufferSize(): " + sampleRateInHz + " Hz is not a supported sample rate."); 767 return ERROR_BAD_VALUE; 768 } 769 770 int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat); 771 if (size <= 0) { 772 loge("getMinBufferSize(): error querying hardware"); 773 return ERROR; 774 } 775 else { 776 return size; 777 } 778 } 779 780 /** 781 * Returns the audio session ID. 782 * 783 * @return the ID of the audio session this AudioTrack belongs to. 784 */ 785 public int getAudioSessionId() { 786 return mSessionId; 787 } 788 789 /** 790 * Poll for a timestamp on demand. 791 * <p> 792 * If you need to track timestamps during initial warmup or after a routing or mode change, 793 * you should request a new timestamp once per second until the reported timestamps 794 * show that the audio clock is stable. 795 * Thereafter, query for a new timestamp approximately once every 10 seconds to once per minute. 796 * Calling this method more often is inefficient. 797 * It is also counter-productive to call this method more often than recommended, 798 * because the short-term differences between successive timestamp reports are not meaningful. 799 * If you need a high-resolution mapping between frame position and presentation time, 800 * consider implementing that at application level, based on low-resolution timestamps. 801 * <p> 802 * The audio data at the returned position may either already have been 803 * presented, or may have not yet been presented but is committed to be presented. 804 * It is not possible to request the time corresponding to a particular position, 805 * or to request the (fractional) position corresponding to a particular time. 806 * If you need such features, consider implementing them at application level. 807 * 808 * @param timestamp a reference to a non-null AudioTimestamp instance allocated 809 * and owned by caller. 810 * @return true if a timestamp is available, or false if no timestamp is available. 811 * If a timestamp if available, 812 * the AudioTimestamp instance is filled in with a position in frame units, together 813 * with the estimated time when that frame was presented or is committed to 814 * be presented. 815 * In the case that no timestamp is available, any supplied instance is left unaltered. 816 * A timestamp may be temporarily unavailable while the audio clock is stabilizing, 817 * or during and immediately after a route change. 818 */ 819 // Add this text when the "on new timestamp" API is added: 820 // Use if you need to get the most recent timestamp outside of the event callback handler. 821 public boolean getTimestamp(AudioTimestamp timestamp) 822 { 823 if (timestamp == null) { 824 throw new IllegalArgumentException(); 825 } 826 // It's unfortunate, but we have to either create garbage every time or use synchronized 827 long[] longArray = new long[2]; 828 int ret = native_get_timestamp(longArray); 829 if (ret != SUCCESS) { 830 return false; 831 } 832 timestamp.framePosition = longArray[0]; 833 timestamp.nanoTime = longArray[1]; 834 return true; 835 } 836 837 838 //-------------------------------------------------------------------------- 839 // Initialization / configuration 840 //-------------------- 841 /** 842 * Sets the listener the AudioTrack notifies when a previously set marker is reached or 843 * for each periodic playback head position update. 844 * Notifications will be received in the same thread as the one in which the AudioTrack 845 * instance was created. 846 * @param listener 847 */ 848 public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener) { 849 setPlaybackPositionUpdateListener(listener, null); 850 } 851 852 /** 853 * Sets the listener the AudioTrack notifies when a previously set marker is reached or 854 * for each periodic playback head position update. 855 * Use this method to receive AudioTrack events in the Handler associated with another 856 * thread than the one in which you created the AudioTrack instance. 857 * @param listener 858 * @param handler the Handler that will receive the event notification messages. 859 */ 860 public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener, 861 Handler handler) { 862 if (listener != null) { 863 mEventHandlerDelegate = new NativeEventHandlerDelegate(this, listener, handler); 864 } else { 865 mEventHandlerDelegate = null; 866 } 867 } 868 869 870 private static float clampGainOrLevel(float gainOrLevel) { 871 if (Float.isNaN(gainOrLevel)) { 872 throw new IllegalArgumentException(); 873 } 874 if (gainOrLevel < GAIN_MIN) { 875 gainOrLevel = GAIN_MIN; 876 } else if (gainOrLevel > GAIN_MAX) { 877 gainOrLevel = GAIN_MAX; 878 } 879 return gainOrLevel; 880 } 881 882 883 /** 884 * Sets the specified left and right output gain values on the AudioTrack. 885 * <p>Gain values are clamped to the closed interval [0.0, max] where 886 * max is the value of {@link #getMaxVolume}. 887 * A value of 0.0 results in zero gain (silence), and 888 * a value of 1.0 means unity gain (signal unchanged). 889 * The default value is 1.0 meaning unity gain. 890 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 891 * @param leftGain output gain for the left channel. 892 * @param rightGain output gain for the right channel 893 * @return error code or success, see {@link #SUCCESS}, 894 * {@link #ERROR_INVALID_OPERATION} 895 * @deprecated Applications should use {@link #setVolume} instead, as it 896 * more gracefully scales down to mono, and up to multi-channel content beyond stereo. 897 */ 898 public int setStereoVolume(float leftGain, float rightGain) { 899 if (isRestricted()) { 900 return SUCCESS; 901 } 902 if (mState == STATE_UNINITIALIZED) { 903 return ERROR_INVALID_OPERATION; 904 } 905 906 leftGain = clampGainOrLevel(leftGain); 907 rightGain = clampGainOrLevel(rightGain); 908 909 native_setVolume(leftGain, rightGain); 910 911 return SUCCESS; 912 } 913 914 915 /** 916 * Sets the specified output gain value on all channels of this track. 917 * <p>Gain values are clamped to the closed interval [0.0, max] where 918 * max is the value of {@link #getMaxVolume}. 919 * A value of 0.0 results in zero gain (silence), and 920 * a value of 1.0 means unity gain (signal unchanged). 921 * The default value is 1.0 meaning unity gain. 922 * <p>This API is preferred over {@link #setStereoVolume}, as it 923 * more gracefully scales down to mono, and up to multi-channel content beyond stereo. 924 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 925 * @param gain output gain for all channels. 926 * @return error code or success, see {@link #SUCCESS}, 927 * {@link #ERROR_INVALID_OPERATION} 928 */ 929 public int setVolume(float gain) { 930 return setStereoVolume(gain, gain); 931 } 932 933 934 /** 935 * Sets the playback sample rate for this track. This sets the sampling rate at which 936 * the audio data will be consumed and played back 937 * (as set by the sampleRateInHz parameter in the 938 * {@link #AudioTrack(int, int, int, int, int, int)} constructor), 939 * not the original sampling rate of the 940 * content. For example, setting it to half the sample rate of the content will cause the 941 * playback to last twice as long, but will also result in a pitch shift down by one octave. 942 * The valid sample rate range is from 1 Hz to twice the value returned by 943 * {@link #getNativeOutputSampleRate(int)}. 944 * @param sampleRateInHz the sample rate expressed in Hz 945 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 946 * {@link #ERROR_INVALID_OPERATION} 947 */ 948 public int setPlaybackRate(int sampleRateInHz) { 949 if (mState != STATE_INITIALIZED) { 950 return ERROR_INVALID_OPERATION; 951 } 952 if (sampleRateInHz <= 0) { 953 return ERROR_BAD_VALUE; 954 } 955 return native_set_playback_rate(sampleRateInHz); 956 } 957 958 959 /** 960 * Sets the position of the notification marker. At most one marker can be active. 961 * @param markerInFrames marker position in wrapping frame units similar to 962 * {@link #getPlaybackHeadPosition}, or zero to disable the marker. 963 * To set a marker at a position which would appear as zero due to wraparound, 964 * a workaround is to use a non-zero position near zero, such as -1 or 1. 965 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 966 * {@link #ERROR_INVALID_OPERATION} 967 */ 968 public int setNotificationMarkerPosition(int markerInFrames) { 969 if (mState == STATE_UNINITIALIZED) { 970 return ERROR_INVALID_OPERATION; 971 } 972 return native_set_marker_pos(markerInFrames); 973 } 974 975 976 /** 977 * Sets the period for the periodic notification event. 978 * @param periodInFrames update period expressed in frames 979 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_INVALID_OPERATION} 980 */ 981 public int setPositionNotificationPeriod(int periodInFrames) { 982 if (mState == STATE_UNINITIALIZED) { 983 return ERROR_INVALID_OPERATION; 984 } 985 return native_set_pos_update_period(periodInFrames); 986 } 987 988 989 /** 990 * Sets the playback head position. 991 * The track must be stopped or paused for the position to be changed, 992 * and must use the {@link #MODE_STATIC} mode. 993 * @param positionInFrames playback head position expressed in frames 994 * Zero corresponds to start of buffer. 995 * The position must not be greater than the buffer size in frames, or negative. 996 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 997 * {@link #ERROR_INVALID_OPERATION} 998 */ 999 public int setPlaybackHeadPosition(int positionInFrames) { 1000 if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED || 1001 getPlayState() == PLAYSTATE_PLAYING) { 1002 return ERROR_INVALID_OPERATION; 1003 } 1004 if (!(0 <= positionInFrames && positionInFrames <= mNativeBufferSizeInFrames)) { 1005 return ERROR_BAD_VALUE; 1006 } 1007 return native_set_position(positionInFrames); 1008 } 1009 1010 /** 1011 * Sets the loop points and the loop count. The loop can be infinite. 1012 * Similarly to setPlaybackHeadPosition, 1013 * the track must be stopped or paused for the loop points to be changed, 1014 * and must use the {@link #MODE_STATIC} mode. 1015 * @param startInFrames loop start marker expressed in frames 1016 * Zero corresponds to start of buffer. 1017 * The start marker must not be greater than or equal to the buffer size in frames, or negative. 1018 * @param endInFrames loop end marker expressed in frames 1019 * The total buffer size in frames corresponds to end of buffer. 1020 * The end marker must not be greater than the buffer size in frames. 1021 * For looping, the end marker must not be less than or equal to the start marker, 1022 * but to disable looping 1023 * it is permitted for start marker, end marker, and loop count to all be 0. 1024 * @param loopCount the number of times the loop is looped. 1025 * A value of -1 means infinite looping, and 0 disables looping. 1026 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1027 * {@link #ERROR_INVALID_OPERATION} 1028 */ 1029 public int setLoopPoints(int startInFrames, int endInFrames, int loopCount) { 1030 if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED || 1031 getPlayState() == PLAYSTATE_PLAYING) { 1032 return ERROR_INVALID_OPERATION; 1033 } 1034 if (loopCount == 0) { 1035 ; // explicitly allowed as an exception to the loop region range check 1036 } else if (!(0 <= startInFrames && startInFrames < mNativeBufferSizeInFrames && 1037 startInFrames < endInFrames && endInFrames <= mNativeBufferSizeInFrames)) { 1038 return ERROR_BAD_VALUE; 1039 } 1040 return native_set_loop(startInFrames, endInFrames, loopCount); 1041 } 1042 1043 /** 1044 * Sets the initialization state of the instance. This method was originally intended to be used 1045 * in an AudioTrack subclass constructor to set a subclass-specific post-initialization state. 1046 * However, subclasses of AudioTrack are no longer recommended, so this method is obsolete. 1047 * @param state the state of the AudioTrack instance 1048 * @deprecated Only accessible by subclasses, which are not recommended for AudioTrack. 1049 */ 1050 @Deprecated 1051 protected void setState(int state) { 1052 mState = state; 1053 } 1054 1055 1056 //--------------------------------------------------------- 1057 // Transport control methods 1058 //-------------------- 1059 /** 1060 * Starts playing an AudioTrack. 1061 * If track's creation mode is {@link #MODE_STATIC}, you must have called write() prior. 1062 * 1063 * @throws IllegalStateException 1064 */ 1065 public void play() 1066 throws IllegalStateException { 1067 if (mState != STATE_INITIALIZED) { 1068 throw new IllegalStateException("play() called on uninitialized AudioTrack."); 1069 } 1070 if (isRestricted()) { 1071 setVolume(0); 1072 } 1073 synchronized(mPlayStateLock) { 1074 native_start(); 1075 mPlayState = PLAYSTATE_PLAYING; 1076 } 1077 } 1078 1079 private boolean isRestricted() { 1080 try { 1081 final int mode = mAppOps.checkAudioOperation(AppOpsManager.OP_PLAY_AUDIO, mStreamType, 1082 Process.myUid(), ActivityThread.currentPackageName()); 1083 return mode != AppOpsManager.MODE_ALLOWED; 1084 } catch (RemoteException e) { 1085 return false; 1086 } 1087 } 1088 1089 /** 1090 * Stops playing the audio data. 1091 * When used on an instance created in {@link #MODE_STREAM} mode, audio will stop playing 1092 * after the last buffer that was written has been played. For an immediate stop, use 1093 * {@link #pause()}, followed by {@link #flush()} to discard audio data that hasn't been played 1094 * back yet. 1095 * @throws IllegalStateException 1096 */ 1097 public void stop() 1098 throws IllegalStateException { 1099 if (mState != STATE_INITIALIZED) { 1100 throw new IllegalStateException("stop() called on uninitialized AudioTrack."); 1101 } 1102 1103 // stop playing 1104 synchronized(mPlayStateLock) { 1105 native_stop(); 1106 mPlayState = PLAYSTATE_STOPPED; 1107 } 1108 } 1109 1110 /** 1111 * Pauses the playback of the audio data. Data that has not been played 1112 * back will not be discarded. Subsequent calls to {@link #play} will play 1113 * this data back. See {@link #flush()} to discard this data. 1114 * 1115 * @throws IllegalStateException 1116 */ 1117 public void pause() 1118 throws IllegalStateException { 1119 if (mState != STATE_INITIALIZED) { 1120 throw new IllegalStateException("pause() called on uninitialized AudioTrack."); 1121 } 1122 //logd("pause()"); 1123 1124 // pause playback 1125 synchronized(mPlayStateLock) { 1126 native_pause(); 1127 mPlayState = PLAYSTATE_PAUSED; 1128 } 1129 } 1130 1131 1132 //--------------------------------------------------------- 1133 // Audio data supply 1134 //-------------------- 1135 1136 /** 1137 * Flushes the audio data currently queued for playback. Any data that has 1138 * not been played back will be discarded. No-op if not stopped or paused, 1139 * or if the track's creation mode is not {@link #MODE_STREAM}. 1140 */ 1141 public void flush() { 1142 if (mState == STATE_INITIALIZED) { 1143 // flush the data in native layer 1144 native_flush(); 1145 } 1146 1147 } 1148 1149 /** 1150 * Writes the audio data to the audio sink for playback (streaming mode), 1151 * or copies audio data for later playback (static buffer mode). 1152 * In streaming mode, will block until all data has been written to the audio sink. 1153 * In static buffer mode, copies the data to the buffer starting at offset 0. 1154 * Note that the actual playback of this data might occur after this function 1155 * returns. This function is thread safe with respect to {@link #stop} calls, 1156 * in which case all of the specified data might not be written to the audio sink. 1157 * 1158 * @param audioData the array that holds the data to play. 1159 * @param offsetInBytes the offset expressed in bytes in audioData where the data to play 1160 * starts. 1161 * @param sizeInBytes the number of bytes to read in audioData after the offset. 1162 * @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION} 1163 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1164 * the parameters don't resolve to valid data and indexes, or 1165 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1166 * needs to be recreated. 1167 */ 1168 1169 public int write(byte[] audioData, int offsetInBytes, int sizeInBytes) { 1170 1171 if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) { 1172 return ERROR_INVALID_OPERATION; 1173 } 1174 1175 if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0) 1176 || (offsetInBytes + sizeInBytes < 0) // detect integer overflow 1177 || (offsetInBytes + sizeInBytes > audioData.length)) { 1178 return ERROR_BAD_VALUE; 1179 } 1180 1181 int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat, 1182 true /*isBlocking*/); 1183 1184 if ((mDataLoadMode == MODE_STATIC) 1185 && (mState == STATE_NO_STATIC_DATA) 1186 && (ret > 0)) { 1187 // benign race with respect to other APIs that read mState 1188 mState = STATE_INITIALIZED; 1189 } 1190 1191 return ret; 1192 } 1193 1194 1195 /** 1196 * Writes the audio data to the audio sink for playback (streaming mode), 1197 * or copies audio data for later playback (static buffer mode). 1198 * In streaming mode, will block until all data has been written to the audio sink. 1199 * In static buffer mode, copies the data to the buffer starting at offset 0. 1200 * Note that the actual playback of this data might occur after this function 1201 * returns. This function is thread safe with respect to {@link #stop} calls, 1202 * in which case all of the specified data might not be written to the audio sink. 1203 * 1204 * @param audioData the array that holds the data to play. 1205 * @param offsetInShorts the offset expressed in shorts in audioData where the data to play 1206 * starts. 1207 * @param sizeInShorts the number of shorts to read in audioData after the offset. 1208 * @return the number of shorts that were written or {@link #ERROR_INVALID_OPERATION} 1209 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1210 * the parameters don't resolve to valid data and indexes. 1211 */ 1212 1213 public int write(short[] audioData, int offsetInShorts, int sizeInShorts) { 1214 1215 if (mState == STATE_UNINITIALIZED || mAudioFormat != AudioFormat.ENCODING_PCM_16BIT) { 1216 return ERROR_INVALID_OPERATION; 1217 } 1218 1219 if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0) 1220 || (offsetInShorts + sizeInShorts < 0) // detect integer overflow 1221 || (offsetInShorts + sizeInShorts > audioData.length)) { 1222 return ERROR_BAD_VALUE; 1223 } 1224 1225 int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat); 1226 1227 if ((mDataLoadMode == MODE_STATIC) 1228 && (mState == STATE_NO_STATIC_DATA) 1229 && (ret > 0)) { 1230 // benign race with respect to other APIs that read mState 1231 mState = STATE_INITIALIZED; 1232 } 1233 1234 return ret; 1235 } 1236 1237 1238 /** 1239 * Writes the audio data to the audio sink for playback (streaming mode), 1240 * or copies audio data for later playback (static buffer mode). 1241 * In static buffer mode, copies the data to the buffer starting at offset 0, 1242 * and the write mode is ignored. 1243 * In streaming mode, the blocking behavior will depend on the write mode. 1244 * <p> 1245 * Note that the actual playback of this data might occur after this function 1246 * returns. This function is thread safe with respect to {@link #stop} calls, 1247 * in which case all of the specified data might not be written to the audio sink. 1248 * <p> 1249 * @param audioData the array that holds the data to play. 1250 * The implementation does not clip for sample values within the nominal range 1251 * [-1.0f, 1.0f], provided that all gains in the audio pipeline are 1252 * less than or equal to unity (1.0f), and in the absence of post-processing effects 1253 * that could add energy, such as reverb. For the convenience of applications 1254 * that compute samples using filters with non-unity gain, 1255 * sample values +3 dB beyond the nominal range are permitted. 1256 * However such values may eventually be limited or clipped, depending on various gains 1257 * and later processing in the audio path. Therefore applications are encouraged 1258 * to provide samples values within the nominal range. 1259 * @param offsetInFloats the offset, expressed as a number of floats, 1260 * in audioData where the data to play starts. 1261 * @param sizeInFloats the number of floats to read in audioData after the offset. 1262 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1263 * effect in static mode. 1264 * <BR>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1265 * to the audio sink. 1266 * <BR>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1267 * queuing as much audio data for playback as possible without blocking. 1268 * @return the number of floats that were written, or {@link #ERROR_INVALID_OPERATION} 1269 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1270 * the parameters don't resolve to valid data and indexes. 1271 */ 1272 public int write(float[] audioData, int offsetInFloats, int sizeInFloats, 1273 @WriteMode int writeMode) { 1274 1275 if (mState == STATE_UNINITIALIZED) { 1276 Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); 1277 return ERROR_INVALID_OPERATION; 1278 } 1279 1280 if (mAudioFormat != AudioFormat.ENCODING_PCM_FLOAT) { 1281 Log.e(TAG, "AudioTrack.write(float[] ...) requires format ENCODING_PCM_FLOAT"); 1282 return ERROR_INVALID_OPERATION; 1283 } 1284 1285 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1286 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1287 return ERROR_BAD_VALUE; 1288 } 1289 1290 if ( (audioData == null) || (offsetInFloats < 0 ) || (sizeInFloats < 0) 1291 || (offsetInFloats + sizeInFloats < 0) // detect integer overflow 1292 || (offsetInFloats + sizeInFloats > audioData.length)) { 1293 Log.e(TAG, "AudioTrack.write() called with invalid array, offset, or size"); 1294 return ERROR_BAD_VALUE; 1295 } 1296 1297 int ret = native_write_float(audioData, offsetInFloats, sizeInFloats, mAudioFormat, 1298 writeMode == WRITE_BLOCKING); 1299 1300 if ((mDataLoadMode == MODE_STATIC) 1301 && (mState == STATE_NO_STATIC_DATA) 1302 && (ret > 0)) { 1303 // benign race with respect to other APIs that read mState 1304 mState = STATE_INITIALIZED; 1305 } 1306 1307 return ret; 1308 } 1309 1310 1311 /** 1312 * Writes the audio data to the audio sink for playback (streaming mode), 1313 * or copies audio data for later playback (static buffer mode). 1314 * In static buffer mode, copies the data to the buffer starting at its 0 offset, and the write 1315 * mode is ignored. 1316 * In streaming mode, the blocking behavior will depend on the write mode. 1317 * @param audioData the buffer that holds the data to play, starting at the position reported 1318 * by <code>audioData.position()</code>. 1319 * <BR>Note that upon return, the buffer position (<code>audioData.position()</code>) will 1320 * have been advanced to reflect the amount of data that was successfully written to 1321 * the AudioTrack. 1322 * @param sizeInBytes number of bytes to write. 1323 * <BR>Note this may differ from <code>audioData.remaining()</code>, but cannot exceed it. 1324 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1325 * effect in static mode. 1326 * <BR>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1327 * to the audio sink. 1328 * <BR>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1329 * queuing as much audio data for playback as possible without blocking. 1330 * @return 0 or a positive number of bytes that were written, or 1331 * {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION} 1332 */ 1333 public int write(ByteBuffer audioData, int sizeInBytes, 1334 @WriteMode int writeMode) { 1335 1336 if (mState == STATE_UNINITIALIZED) { 1337 Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); 1338 return ERROR_INVALID_OPERATION; 1339 } 1340 1341 if (mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) { 1342 Log.e(TAG, "AudioTrack.write(ByteBuffer ...) not yet supported for ENCODING_PCM_FLOAT"); 1343 return ERROR_INVALID_OPERATION; 1344 } 1345 1346 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1347 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1348 return ERROR_BAD_VALUE; 1349 } 1350 1351 if ( (audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) { 1352 Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value"); 1353 return ERROR_BAD_VALUE; 1354 } 1355 1356 int ret = 0; 1357 if (audioData.isDirect()) { 1358 ret = native_write_native_bytes(audioData, 1359 audioData.position(), sizeInBytes, mAudioFormat, 1360 writeMode == WRITE_BLOCKING); 1361 } else { 1362 ret = native_write_byte(NioUtils.unsafeArray(audioData), 1363 NioUtils.unsafeArrayOffset(audioData) + audioData.position(), 1364 sizeInBytes, mAudioFormat, 1365 writeMode == WRITE_BLOCKING); 1366 } 1367 1368 if ((mDataLoadMode == MODE_STATIC) 1369 && (mState == STATE_NO_STATIC_DATA) 1370 && (ret > 0)) { 1371 // benign race with respect to other APIs that read mState 1372 mState = STATE_INITIALIZED; 1373 } 1374 1375 if (ret > 0) { 1376 audioData.position(audioData.position() + ret); 1377 } 1378 1379 return ret; 1380 } 1381 1382 /** 1383 * Notifies the native resource to reuse the audio data already loaded in the native 1384 * layer, that is to rewind to start of buffer. 1385 * The track's creation mode must be {@link #MODE_STATIC}. 1386 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1387 * {@link #ERROR_INVALID_OPERATION} 1388 */ 1389 public int reloadStaticData() { 1390 if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED) { 1391 return ERROR_INVALID_OPERATION; 1392 } 1393 return native_reload_static(); 1394 } 1395 1396 //-------------------------------------------------------------------------- 1397 // Audio effects management 1398 //-------------------- 1399 1400 /** 1401 * Attaches an auxiliary effect to the audio track. A typical auxiliary 1402 * effect is a reverberation effect which can be applied on any sound source 1403 * that directs a certain amount of its energy to this effect. This amount 1404 * is defined by setAuxEffectSendLevel(). 1405 * {@see #setAuxEffectSendLevel(float)}. 1406 * <p>After creating an auxiliary effect (e.g. 1407 * {@link android.media.audiofx.EnvironmentalReverb}), retrieve its ID with 1408 * {@link android.media.audiofx.AudioEffect#getId()} and use it when calling 1409 * this method to attach the audio track to the effect. 1410 * <p>To detach the effect from the audio track, call this method with a 1411 * null effect id. 1412 * 1413 * @param effectId system wide unique id of the effect to attach 1414 * @return error code or success, see {@link #SUCCESS}, 1415 * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR_BAD_VALUE} 1416 */ 1417 public int attachAuxEffect(int effectId) { 1418 if (mState == STATE_UNINITIALIZED) { 1419 return ERROR_INVALID_OPERATION; 1420 } 1421 return native_attachAuxEffect(effectId); 1422 } 1423 1424 /** 1425 * Sets the send level of the audio track to the attached auxiliary effect 1426 * {@link #attachAuxEffect(int)}. Effect levels 1427 * are clamped to the closed interval [0.0, max] where 1428 * max is the value of {@link #getMaxVolume}. 1429 * A value of 0.0 results in no effect, and a value of 1.0 is full send. 1430 * <p>By default the send level is 0.0f, so even if an effect is attached to the player 1431 * this method must be called for the effect to be applied. 1432 * <p>Note that the passed level value is a linear scalar. UI controls should be scaled 1433 * logarithmically: the gain applied by audio framework ranges from -72dB to at least 0dB, 1434 * so an appropriate conversion from linear UI input x to level is: 1435 * x == 0 -> level = 0 1436 * 0 < x <= R -> level = 10^(72*(x-R)/20/R) 1437 * 1438 * @param level linear send level 1439 * @return error code or success, see {@link #SUCCESS}, 1440 * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR} 1441 */ 1442 public int setAuxEffectSendLevel(float level) { 1443 if (isRestricted()) { 1444 return SUCCESS; 1445 } 1446 if (mState == STATE_UNINITIALIZED) { 1447 return ERROR_INVALID_OPERATION; 1448 } 1449 level = clampGainOrLevel(level); 1450 int err = native_setAuxEffectSendLevel(level); 1451 return err == 0 ? SUCCESS : ERROR; 1452 } 1453 1454 //--------------------------------------------------------- 1455 // Interface definitions 1456 //-------------------- 1457 /** 1458 * Interface definition for a callback to be invoked when the playback head position of 1459 * an AudioTrack has reached a notification marker or has increased by a certain period. 1460 */ 1461 public interface OnPlaybackPositionUpdateListener { 1462 /** 1463 * Called on the listener to notify it that the previously set marker has been reached 1464 * by the playback head. 1465 */ 1466 void onMarkerReached(AudioTrack track); 1467 1468 /** 1469 * Called on the listener to periodically notify it that the playback head has reached 1470 * a multiple of the notification period. 1471 */ 1472 void onPeriodicNotification(AudioTrack track); 1473 } 1474 1475 //--------------------------------------------------------- 1476 // Inner classes 1477 //-------------------- 1478 /** 1479 * Helper class to handle the forwarding of native events to the appropriate listener 1480 * (potentially) handled in a different thread 1481 */ 1482 private class NativeEventHandlerDelegate { 1483 private final Handler mHandler; 1484 1485 NativeEventHandlerDelegate(final AudioTrack track, 1486 final OnPlaybackPositionUpdateListener listener, 1487 Handler handler) { 1488 // find the looper for our new event handler 1489 Looper looper; 1490 if (handler != null) { 1491 looper = handler.getLooper(); 1492 } else { 1493 // no given handler, use the looper the AudioTrack was created in 1494 looper = mInitializationLooper; 1495 } 1496 1497 // construct the event handler with this looper 1498 if (looper != null) { 1499 // implement the event handler delegate 1500 mHandler = new Handler(looper) { 1501 @Override 1502 public void handleMessage(Message msg) { 1503 if (track == null) { 1504 return; 1505 } 1506 switch(msg.what) { 1507 case NATIVE_EVENT_MARKER: 1508 if (listener != null) { 1509 listener.onMarkerReached(track); 1510 } 1511 break; 1512 case NATIVE_EVENT_NEW_POS: 1513 if (listener != null) { 1514 listener.onPeriodicNotification(track); 1515 } 1516 break; 1517 default: 1518 loge("Unknown native event type: " + msg.what); 1519 break; 1520 } 1521 } 1522 }; 1523 } else { 1524 mHandler = null; 1525 } 1526 } 1527 1528 Handler getHandler() { 1529 return mHandler; 1530 } 1531 } 1532 1533 1534 //--------------------------------------------------------- 1535 // Java methods called from the native side 1536 //-------------------- 1537 @SuppressWarnings("unused") 1538 private static void postEventFromNative(Object audiotrack_ref, 1539 int what, int arg1, int arg2, Object obj) { 1540 //logd("Event posted from the native side: event="+ what + " args="+ arg1+" "+arg2); 1541 AudioTrack track = (AudioTrack)((WeakReference)audiotrack_ref).get(); 1542 if (track == null) { 1543 return; 1544 } 1545 1546 NativeEventHandlerDelegate delegate = track.mEventHandlerDelegate; 1547 if (delegate != null) { 1548 Handler handler = delegate.getHandler(); 1549 if (handler != null) { 1550 Message m = handler.obtainMessage(what, arg1, arg2, obj); 1551 handler.sendMessage(m); 1552 } 1553 } 1554 1555 } 1556 1557 1558 //--------------------------------------------------------- 1559 // Native methods called from the Java side 1560 //-------------------- 1561 1562 private native final int native_setup(Object audiotrack_this, 1563 int streamType, int sampleRate, int channelMask, int audioFormat, 1564 int buffSizeInBytes, int mode, int[] sessionId); 1565 1566 private native final void native_finalize(); 1567 1568 private native final void native_release(); 1569 1570 private native final void native_start(); 1571 1572 private native final void native_stop(); 1573 1574 private native final void native_pause(); 1575 1576 private native final void native_flush(); 1577 1578 private native final int native_write_byte(byte[] audioData, 1579 int offsetInBytes, int sizeInBytes, int format, 1580 boolean isBlocking); 1581 1582 private native final int native_write_short(short[] audioData, 1583 int offsetInShorts, int sizeInShorts, int format); 1584 1585 private native final int native_write_float(float[] audioData, 1586 int offsetInFloats, int sizeInFloats, int format, 1587 boolean isBlocking); 1588 1589 private native final int native_write_native_bytes(Object audioData, 1590 int positionInBytes, int sizeInBytes, int format, boolean blocking); 1591 1592 private native final int native_reload_static(); 1593 1594 private native final int native_get_native_frame_count(); 1595 1596 private native final void native_setVolume(float leftVolume, float rightVolume); 1597 1598 private native final int native_set_playback_rate(int sampleRateInHz); 1599 private native final int native_get_playback_rate(); 1600 1601 private native final int native_set_marker_pos(int marker); 1602 private native final int native_get_marker_pos(); 1603 1604 private native final int native_set_pos_update_period(int updatePeriod); 1605 private native final int native_get_pos_update_period(); 1606 1607 private native final int native_set_position(int position); 1608 private native final int native_get_position(); 1609 1610 private native final int native_get_latency(); 1611 1612 // longArray must be a non-null array of length >= 2 1613 // [0] is assigned the frame position 1614 // [1] is assigned the time in CLOCK_MONOTONIC nanoseconds 1615 private native final int native_get_timestamp(long[] longArray); 1616 1617 private native final int native_set_loop(int start, int end, int loopCount); 1618 1619 static private native final int native_get_output_sample_rate(int streamType); 1620 static private native final int native_get_min_buff_size( 1621 int sampleRateInHz, int channelConfig, int audioFormat); 1622 1623 private native final int native_attachAuxEffect(int effectId); 1624 private native final int native_setAuxEffectSendLevel(float level); 1625 1626 //--------------------------------------------------------- 1627 // Utility methods 1628 //------------------ 1629 1630 private static void logd(String msg) { 1631 Log.d(TAG, msg); 1632 } 1633 1634 private static void loge(String msg) { 1635 Log.e(TAG, msg); 1636 } 1637 1638} 1639