AudioTrack.java revision 948c2e6ff46d65942277f2e0e9ce0c038972b9d8
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.ref.WeakReference; 20 21import android.os.Handler; 22import android.os.Looper; 23import android.os.Message; 24import android.util.Log; 25 26 27/** 28 * The AudioTrack class manages and plays a single audio resource for Java applications. 29 * It allows streaming of PCM audio buffers to the audio sink for playback. This is 30 * achieved by "pushing" the data to the AudioTrack object using one of the 31 * {@link #write(byte[], int, int)} and {@link #write(short[], int, int)} methods. 32 * 33 * <p>An AudioTrack instance can operate under two modes: static or streaming.<br> 34 * In Streaming mode, the application writes a continuous stream of data to the AudioTrack, using 35 * one of the {@code write()} methods. These are blocking and return when the data has been 36 * transferred from the Java layer to the native layer and queued for playback. The streaming 37 * mode is most useful when playing blocks of audio data that for instance are: 38 * 39 * <ul> 40 * <li>too big to fit in memory because of the duration of the sound to play,</li> 41 * <li>too big to fit in memory because of the characteristics of the audio data 42 * (high sampling rate, bits per sample ...)</li> 43 * <li>received or generated while previously queued audio is playing.</li> 44 * </ul> 45 * 46 * The static mode should be chosen when dealing with short sounds that fit in memory and 47 * that need to be played with the smallest latency possible. The static mode will 48 * therefore be preferred for UI and game sounds that are played often, and with the 49 * smallest overhead possible. 50 * 51 * <p>Upon creation, an AudioTrack object initializes its associated audio buffer. 52 * The size of this buffer, specified during the construction, determines how long an AudioTrack 53 * can play before running out of data.<br> 54 * For an AudioTrack using the static mode, this size is the maximum size of the sound that can 55 * be played from it.<br> 56 * For the streaming mode, data will be written to the audio sink in chunks of 57 * sizes less than or equal to the total buffer size. 58 * 59 * AudioTrack is not final and thus permits subclasses, but such use is not recommended. 60 */ 61public class AudioTrack 62{ 63 //--------------------------------------------------------- 64 // Constants 65 //-------------------- 66 /** Minimum value for a channel volume */ 67 private static final float VOLUME_MIN = 0.0f; 68 /** Maximum value for a channel volume */ 69 private static final float VOLUME_MAX = 1.0f; 70 71 /** Minimum value for sample rate */ 72 private static final int SAMPLE_RATE_HZ_MIN = 4000; 73 /** Maximum value for sample rate */ 74 private static final int SAMPLE_RATE_HZ_MAX = 48000; 75 76 /** indicates AudioTrack state is stopped */ 77 public static final int PLAYSTATE_STOPPED = 1; // matches SL_PLAYSTATE_STOPPED 78 /** indicates AudioTrack state is paused */ 79 public static final int PLAYSTATE_PAUSED = 2; // matches SL_PLAYSTATE_PAUSED 80 /** indicates AudioTrack state is playing */ 81 public static final int PLAYSTATE_PLAYING = 3; // matches SL_PLAYSTATE_PLAYING 82 83 // keep these values in sync with android_media_AudioTrack.cpp 84 /** 85 * Creation mode where audio data is transferred from Java to the native layer 86 * only once before the audio starts playing. 87 */ 88 public static final int MODE_STATIC = 0; 89 /** 90 * Creation mode where audio data is streamed from Java to the native layer 91 * as the audio is playing. 92 */ 93 public static final int MODE_STREAM = 1; 94 95 /** 96 * State of an AudioTrack that was not successfully initialized upon creation. 97 */ 98 public static final int STATE_UNINITIALIZED = 0; 99 /** 100 * State of an AudioTrack that is ready to be used. 101 */ 102 public static final int STATE_INITIALIZED = 1; 103 /** 104 * State of a successfully initialized AudioTrack that uses static data, 105 * but that hasn't received that data yet. 106 */ 107 public static final int STATE_NO_STATIC_DATA = 2; 108 109 // Error codes: 110 // to keep in sync with frameworks/base/core/jni/android_media_AudioTrack.cpp 111 /** 112 * Denotes a successful operation. 113 */ 114 public static final int SUCCESS = 0; 115 /** 116 * Denotes a generic operation failure. 117 */ 118 public static final int ERROR = -1; 119 /** 120 * Denotes a failure due to the use of an invalid value. 121 */ 122 public static final int ERROR_BAD_VALUE = -2; 123 /** 124 * Denotes a failure due to the improper use of a method. 125 */ 126 public static final int ERROR_INVALID_OPERATION = -3; 127 128 private static final int ERROR_NATIVESETUP_AUDIOSYSTEM = -16; 129 private static final int ERROR_NATIVESETUP_INVALIDCHANNELMASK = -17; 130 private static final int ERROR_NATIVESETUP_INVALIDFORMAT = -18; 131 private static final int ERROR_NATIVESETUP_INVALIDSTREAMTYPE = -19; 132 private static final int ERROR_NATIVESETUP_NATIVEINITFAILED = -20; 133 134 // Events: 135 // to keep in sync with frameworks/av/include/media/AudioTrack.h 136 /** 137 * Event id denotes when playback head has reached a previously set marker. 138 */ 139 private static final int NATIVE_EVENT_MARKER = 3; 140 /** 141 * Event id denotes when previously set update period has elapsed during playback. 142 */ 143 private static final int NATIVE_EVENT_NEW_POS = 4; 144 145 private final static String TAG = "android.media.AudioTrack"; 146 147 148 //-------------------------------------------------------------------------- 149 // Member variables 150 //-------------------- 151 /** 152 * Indicates the state of the AudioTrack instance. 153 */ 154 private int mState = STATE_UNINITIALIZED; 155 /** 156 * Indicates the play state of the AudioTrack instance. 157 */ 158 private int mPlayState = PLAYSTATE_STOPPED; 159 /** 160 * Lock to make sure mPlayState updates are reflecting the actual state of the object. 161 */ 162 private final Object mPlayStateLock = new Object(); 163 /** 164 * Sizes of the native audio buffer. 165 */ 166 private int mNativeBufferSizeInBytes = 0; 167 private int mNativeBufferSizeInFrames = 0; 168 /** 169 * Handler for events coming from the native code. 170 */ 171 private NativeEventHandlerDelegate mEventHandlerDelegate; 172 /** 173 * Looper associated with the thread that creates the AudioTrack instance. 174 */ 175 private final Looper mInitializationLooper; 176 /** 177 * The audio data source sampling rate in Hz. 178 */ 179 private int mSampleRate; // initialized by all constructors 180 /** 181 * The number of audio output channels (1 is mono, 2 is stereo). 182 */ 183 private int mChannelCount = 1; 184 /** 185 * The audio channel mask. 186 */ 187 private int mChannels = AudioFormat.CHANNEL_OUT_MONO; 188 189 /** 190 * The type of the audio stream to play. See 191 * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, 192 * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, 193 * {@link AudioManager#STREAM_ALARM}, {@link AudioManager#STREAM_NOTIFICATION}, and 194 * {@link AudioManager#STREAM_DTMF}. 195 */ 196 private int mStreamType = AudioManager.STREAM_MUSIC; 197 /** 198 * The way audio is consumed by the audio sink, streaming or static. 199 */ 200 private int mDataLoadMode = MODE_STREAM; 201 /** 202 * The current audio channel configuration. 203 */ 204 private int mChannelConfiguration = AudioFormat.CHANNEL_OUT_MONO; 205 /** 206 * The encoding of the audio samples. 207 * @see AudioFormat#ENCODING_PCM_8BIT 208 * @see AudioFormat#ENCODING_PCM_16BIT 209 */ 210 private int mAudioFormat = AudioFormat.ENCODING_PCM_16BIT; 211 /** 212 * Audio session ID 213 */ 214 private int mSessionId = 0; 215 216 217 //-------------------------------- 218 // Used exclusively by native code 219 //-------------------- 220 /** 221 * Accessed by native methods: provides access to C++ AudioTrack object. 222 */ 223 @SuppressWarnings("unused") 224 private int mNativeTrackInJavaObj; 225 /** 226 * Accessed by native methods: provides access to the JNI data (i.e. resources used by 227 * the native AudioTrack object, but not stored in it). 228 */ 229 @SuppressWarnings("unused") 230 private int mJniData; 231 232 233 //-------------------------------------------------------------------------- 234 // Constructor, Finalize 235 //-------------------- 236 /** 237 * Class constructor. 238 * @param streamType the type of the audio stream. See 239 * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, 240 * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, 241 * {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}. 242 * @param sampleRateInHz the initial source sample rate expressed in Hz. 243 * @param channelConfig describes the configuration of the audio channels. 244 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 245 * {@link AudioFormat#CHANNEL_OUT_STEREO} 246 * @param audioFormat the format in which the audio data is represented. 247 * See {@link AudioFormat#ENCODING_PCM_16BIT} and 248 * {@link AudioFormat#ENCODING_PCM_8BIT} 249 * @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is 250 * read from for playback. 251 * If track's creation mode is {@link #MODE_STREAM}, you can write data into 252 * this buffer in chunks less than or equal to this size, and it is typical to use 253 * chunks of 1/2 of the total size to permit double-buffering. 254 * If the track's creation mode is {@link #MODE_STATIC}, 255 * this is the maximum length sample, or audio clip, that can be played by this instance. 256 * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size 257 * for the successful creation of an AudioTrack instance in streaming mode. Using values 258 * smaller than getMinBufferSize() will result in an initialization failure. 259 * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM} 260 * @throws java.lang.IllegalArgumentException 261 */ 262 public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, 263 int bufferSizeInBytes, int mode) 264 throws IllegalArgumentException { 265 this(streamType, sampleRateInHz, channelConfig, audioFormat, 266 bufferSizeInBytes, mode, 0 /*session*/); 267 } 268 269 /** 270 * Class constructor with audio session. Use this constructor when the AudioTrack must be 271 * attached to a particular audio session. The primary use of the audio session ID is to 272 * associate audio effects to a particular instance of AudioTrack: if an audio session ID 273 * is provided when creating an AudioEffect, this effect will be applied only to audio tracks 274 * and media players in the same session and not to the output mix. 275 * When an AudioTrack is created without specifying a session, it will create its own session 276 * which can be retrieved by calling the {@link #getAudioSessionId()} method. 277 * If a non-zero session ID is provided, this AudioTrack will share effects attached to this 278 * session 279 * with all other media players or audio tracks in the same session, otherwise a new session 280 * will be created for this track if none is supplied. 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} and 291 * {@link AudioFormat#ENCODING_PCM_8BIT} 292 * @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read 293 * from for playback. If using the AudioTrack in streaming mode, you can write data into 294 * this buffer in smaller chunks than this size. If using the AudioTrack in static mode, 295 * this is the maximum size of the sound that will be played for this instance. 296 * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size 297 * for the successful creation of an AudioTrack instance in streaming mode. Using values 298 * smaller than getMinBufferSize() will result in an initialization failure. 299 * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM} 300 * @param sessionId Id of audio session the AudioTrack must be attached to 301 * @throws java.lang.IllegalArgumentException 302 */ 303 public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, 304 int bufferSizeInBytes, int mode, int sessionId) 305 throws IllegalArgumentException { 306 // mState already == STATE_UNINITIALIZED 307 308 // remember which looper is associated with the AudioTrack instantiation 309 Looper looper; 310 if ((looper = Looper.myLooper()) == null) { 311 looper = Looper.getMainLooper(); 312 } 313 mInitializationLooper = looper; 314 315 audioParamCheck(streamType, sampleRateInHz, channelConfig, audioFormat, mode); 316 317 audioBuffSizeCheck(bufferSizeInBytes); 318 319 if (sessionId < 0) { 320 throw new IllegalArgumentException("Invalid audio session ID: "+sessionId); 321 } 322 323 int[] session = new int[1]; 324 session[0] = sessionId; 325 // native initialization 326 int initResult = native_setup(new WeakReference<AudioTrack>(this), 327 mStreamType, mSampleRate, mChannels, mAudioFormat, 328 mNativeBufferSizeInBytes, mDataLoadMode, session); 329 if (initResult != SUCCESS) { 330 loge("Error code "+initResult+" when initializing AudioTrack."); 331 return; // with mState == STATE_UNINITIALIZED 332 } 333 334 mSessionId = session[0]; 335 336 if (mDataLoadMode == MODE_STATIC) { 337 mState = STATE_NO_STATIC_DATA; 338 } else { 339 mState = STATE_INITIALIZED; 340 } 341 } 342 343 // mask of all the channels supported by this implementation 344 private static final int SUPPORTED_OUT_CHANNELS = 345 AudioFormat.CHANNEL_OUT_FRONT_LEFT | 346 AudioFormat.CHANNEL_OUT_FRONT_RIGHT | 347 AudioFormat.CHANNEL_OUT_FRONT_CENTER | 348 AudioFormat.CHANNEL_OUT_LOW_FREQUENCY | 349 AudioFormat.CHANNEL_OUT_BACK_LEFT | 350 AudioFormat.CHANNEL_OUT_BACK_RIGHT | 351 AudioFormat.CHANNEL_OUT_BACK_CENTER; 352 353 // Convenience method for the constructor's parameter checks. 354 // This is where constructor IllegalArgumentException-s are thrown 355 // postconditions: 356 // mStreamType is valid 357 // mChannelCount is valid 358 // mChannels is valid 359 // mAudioFormat is valid 360 // mSampleRate is valid 361 // mDataLoadMode is valid 362 private void audioParamCheck(int streamType, int sampleRateInHz, 363 int channelConfig, int audioFormat, int mode) { 364 365 //-------------- 366 // stream type 367 if( (streamType != AudioManager.STREAM_ALARM) && (streamType != AudioManager.STREAM_MUSIC) 368 && (streamType != AudioManager.STREAM_RING) && (streamType != AudioManager.STREAM_SYSTEM) 369 && (streamType != AudioManager.STREAM_VOICE_CALL) 370 && (streamType != AudioManager.STREAM_NOTIFICATION) 371 && (streamType != AudioManager.STREAM_BLUETOOTH_SCO) 372 && (streamType != AudioManager.STREAM_DTMF)) { 373 throw new IllegalArgumentException("Invalid stream type."); 374 } 375 mStreamType = streamType; 376 377 //-------------- 378 // sample rate, note these values are subject to change 379 if ( (sampleRateInHz < 4000) || (sampleRateInHz > 48000) ) { 380 throw new IllegalArgumentException(sampleRateInHz 381 + "Hz is not a supported sample rate."); 382 } 383 mSampleRate = sampleRateInHz; 384 385 //-------------- 386 // channel config 387 mChannelConfiguration = channelConfig; 388 389 switch (channelConfig) { 390 case AudioFormat.CHANNEL_OUT_DEFAULT: //AudioFormat.CHANNEL_CONFIGURATION_DEFAULT 391 case AudioFormat.CHANNEL_OUT_MONO: 392 case AudioFormat.CHANNEL_CONFIGURATION_MONO: 393 mChannelCount = 1; 394 mChannels = AudioFormat.CHANNEL_OUT_MONO; 395 break; 396 case AudioFormat.CHANNEL_OUT_STEREO: 397 case AudioFormat.CHANNEL_CONFIGURATION_STEREO: 398 mChannelCount = 2; 399 mChannels = AudioFormat.CHANNEL_OUT_STEREO; 400 break; 401 default: 402 if (!isMultichannelConfigSupported(channelConfig)) { 403 // input channel configuration features unsupported channels 404 throw new IllegalArgumentException("Unsupported channel configuration."); 405 } 406 mChannels = channelConfig; 407 mChannelCount = Integer.bitCount(channelConfig); 408 } 409 410 //-------------- 411 // audio format 412 switch (audioFormat) { 413 case AudioFormat.ENCODING_DEFAULT: 414 mAudioFormat = AudioFormat.ENCODING_PCM_16BIT; 415 break; 416 case AudioFormat.ENCODING_PCM_16BIT: 417 case AudioFormat.ENCODING_PCM_8BIT: 418 mAudioFormat = audioFormat; 419 break; 420 default: 421 throw new IllegalArgumentException("Unsupported sample encoding." 422 + " Should be ENCODING_PCM_8BIT or ENCODING_PCM_16BIT."); 423 } 424 425 //-------------- 426 // audio load mode 427 if ( (mode != MODE_STREAM) && (mode != MODE_STATIC) ) { 428 throw new IllegalArgumentException("Invalid mode."); 429 } 430 mDataLoadMode = mode; 431 } 432 433 /** 434 * Convenience method to check that the channel configuration (a.k.a channel mask) is supported 435 * @param channelConfig the mask to validate 436 * @return false if the AudioTrack can't be used with such a mask 437 */ 438 private static boolean isMultichannelConfigSupported(int channelConfig) { 439 // check for unsupported channels 440 if ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) { 441 loge("Channel configuration features unsupported channels"); 442 return false; 443 } 444 // check for unsupported multichannel combinations: 445 // - FL/FR must be present 446 // - L/R channels must be paired (e.g. no single L channel) 447 final int frontPair = 448 AudioFormat.CHANNEL_OUT_FRONT_LEFT | AudioFormat.CHANNEL_OUT_FRONT_RIGHT; 449 if ((channelConfig & frontPair) != frontPair) { 450 loge("Front channels must be present in multichannel configurations"); 451 return false; 452 } 453 final int backPair = 454 AudioFormat.CHANNEL_OUT_BACK_LEFT | AudioFormat.CHANNEL_OUT_BACK_RIGHT; 455 if ((channelConfig & backPair) != 0) { 456 if ((channelConfig & backPair) != backPair) { 457 loge("Rear channels can't be used independently"); 458 return false; 459 } 460 } 461 return true; 462 } 463 464 465 // Convenience method for the constructor's audio buffer size check. 466 // preconditions: 467 // mChannelCount is valid 468 // mAudioFormat is valid 469 // postcondition: 470 // mNativeBufferSizeInBytes is valid (multiple of frame size, positive) 471 private void audioBuffSizeCheck(int audioBufferSize) { 472 // NB: this section is only valid with PCM data. 473 // To update when supporting compressed formats 474 int frameSizeInBytes = mChannelCount 475 * (mAudioFormat == AudioFormat.ENCODING_PCM_8BIT ? 1 : 2); 476 if ((audioBufferSize % frameSizeInBytes != 0) || (audioBufferSize < 1)) { 477 throw new IllegalArgumentException("Invalid audio buffer size."); 478 } 479 480 mNativeBufferSizeInBytes = audioBufferSize; 481 mNativeBufferSizeInFrames = audioBufferSize / frameSizeInBytes; 482 } 483 484 485 /** 486 * Releases the native AudioTrack resources. 487 */ 488 public void release() { 489 // even though native_release() stops the native AudioTrack, we need to stop 490 // AudioTrack subclasses too. 491 try { 492 stop(); 493 } catch(IllegalStateException ise) { 494 // don't raise an exception, we're releasing the resources. 495 } 496 native_release(); 497 mState = STATE_UNINITIALIZED; 498 } 499 500 @Override 501 protected void finalize() { 502 native_finalize(); 503 } 504 505 //-------------------------------------------------------------------------- 506 // Getters 507 //-------------------- 508 /** 509 * Returns the minimum valid volume value. Volume values set under this one will 510 * be clamped at this value. 511 * @return the minimum volume expressed as a linear attenuation. 512 */ 513 static public float getMinVolume() { 514 return VOLUME_MIN; 515 } 516 517 /** 518 * Returns the maximum valid volume value. Volume values set above this one will 519 * be clamped at this value. 520 * @return the maximum volume expressed as a linear attenuation. 521 */ 522 static public float getMaxVolume() { 523 return VOLUME_MAX; 524 } 525 526 /** 527 * Returns the configured audio data sample rate in Hz 528 */ 529 public int getSampleRate() { 530 return mSampleRate; 531 } 532 533 /** 534 * Returns the current playback rate in Hz. 535 */ 536 public int getPlaybackRate() { 537 return native_get_playback_rate(); 538 } 539 540 /** 541 * Returns the configured audio data format. See {@link AudioFormat#ENCODING_PCM_16BIT} 542 * and {@link AudioFormat#ENCODING_PCM_8BIT}. 543 */ 544 public int getAudioFormat() { 545 return mAudioFormat; 546 } 547 548 /** 549 * Returns the type of audio stream this AudioTrack is configured for. 550 * Compare the result against {@link AudioManager#STREAM_VOICE_CALL}, 551 * {@link AudioManager#STREAM_SYSTEM}, {@link AudioManager#STREAM_RING}, 552 * {@link AudioManager#STREAM_MUSIC}, {@link AudioManager#STREAM_ALARM}, 553 * {@link AudioManager#STREAM_NOTIFICATION}, or {@link AudioManager#STREAM_DTMF}. 554 */ 555 public int getStreamType() { 556 return mStreamType; 557 } 558 559 /** 560 * Returns the configured channel configuration. 561 * See {@link AudioFormat#CHANNEL_OUT_MONO} 562 * and {@link AudioFormat#CHANNEL_OUT_STEREO}. 563 */ 564 public int getChannelConfiguration() { 565 return mChannelConfiguration; 566 } 567 568 /** 569 * Returns the configured number of channels. 570 */ 571 public int getChannelCount() { 572 return mChannelCount; 573 } 574 575 /** 576 * Returns the state of the AudioTrack instance. This is useful after the 577 * AudioTrack instance has been created to check if it was initialized 578 * properly. This ensures that the appropriate resources have been acquired. 579 * @see #STATE_INITIALIZED 580 * @see #STATE_NO_STATIC_DATA 581 * @see #STATE_UNINITIALIZED 582 */ 583 public int getState() { 584 return mState; 585 } 586 587 /** 588 * Returns the playback state of the AudioTrack instance. 589 * @see #PLAYSTATE_STOPPED 590 * @see #PLAYSTATE_PAUSED 591 * @see #PLAYSTATE_PLAYING 592 */ 593 public int getPlayState() { 594 synchronized (mPlayStateLock) { 595 return mPlayState; 596 } 597 } 598 599 /** 600 * Returns the "native frame count", derived from the bufferSizeInBytes specified at 601 * creation time and converted to frame units. 602 * If track's creation mode is {@link #MODE_STATIC}, 603 * it is equal to the specified bufferSizeInBytes converted to frame units. 604 * If track's creation mode is {@link #MODE_STREAM}, 605 * it is typically greater than or equal to the specified bufferSizeInBytes converted to frame 606 * units; it may be rounded up to a larger value if needed by the target device implementation. 607 * @deprecated Only accessible by subclasses, which are not recommended for AudioTrack. 608 * See {@link AudioManager#getProperty(String)} for key 609 * {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}. 610 */ 611 @Deprecated 612 protected int getNativeFrameCount() { 613 return native_get_native_frame_count(); 614 } 615 616 /** 617 * Returns marker position expressed in frames. 618 * @return marker position in wrapping frame units similar to {@link #getPlaybackHeadPosition}, 619 * or zero if marker is disabled. 620 */ 621 public int getNotificationMarkerPosition() { 622 return native_get_marker_pos(); 623 } 624 625 /** 626 * Returns the notification update period expressed in frames. 627 * Zero means that no position update notifications are being delivered. 628 */ 629 public int getPositionNotificationPeriod() { 630 return native_get_pos_update_period(); 631 } 632 633 /** 634 * Returns the playback head position expressed in frames. 635 * Though the "int" type is signed 32-bits, the value should be reinterpreted as if it is 636 * unsigned 32-bits. That is, the next position after 0x7FFFFFFF is (int) 0x80000000. 637 * This is a continuously advancing counter. It will wrap (overflow) periodically, 638 * for example approximately once every 27:03:11 hours:minutes:seconds at 44.1 kHz. 639 * It is reset to zero by flush(), reload(), and stop(). 640 */ 641 public int getPlaybackHeadPosition() { 642 return native_get_position(); 643 } 644 645 /** 646 * Returns this track's estimated latency in milliseconds. This includes the latency due 647 * to AudioTrack buffer size, AudioMixer (if any) and audio hardware driver. 648 * 649 * DO NOT UNHIDE. The existing approach for doing A/V sync has too many problems. We need 650 * a better solution. 651 * @hide 652 */ 653 public int getLatency() { 654 return native_get_latency(); 655 } 656 657 /** 658 * Returns the output sample rate in Hz for the specified stream type. 659 */ 660 static public int getNativeOutputSampleRate(int streamType) { 661 return native_get_output_sample_rate(streamType); 662 } 663 664 /** 665 * Returns the minimum buffer size required for the successful creation of an AudioTrack 666 * object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't 667 * guarantee a smooth playback under load, and higher values should be chosen according to 668 * the expected frequency at which the buffer will be refilled with additional data to play. 669 * For example, if you intend to dynamically set the source sample rate of an AudioTrack 670 * to a higher value than the initial source sample rate, be sure to configure the buffer size 671 * based on the highest planned sample rate. 672 * @param sampleRateInHz the source sample rate expressed in Hz. 673 * @param channelConfig describes the configuration of the audio channels. 674 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 675 * {@link AudioFormat#CHANNEL_OUT_STEREO} 676 * @param audioFormat the format in which the audio data is represented. 677 * See {@link AudioFormat#ENCODING_PCM_16BIT} and 678 * {@link AudioFormat#ENCODING_PCM_8BIT} 679 * @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed, 680 * or {@link #ERROR} if unable to query for output properties, 681 * or the minimum buffer size expressed in bytes. 682 */ 683 static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) { 684 int channelCount = 0; 685 switch(channelConfig) { 686 case AudioFormat.CHANNEL_OUT_MONO: 687 case AudioFormat.CHANNEL_CONFIGURATION_MONO: 688 channelCount = 1; 689 break; 690 case AudioFormat.CHANNEL_OUT_STEREO: 691 case AudioFormat.CHANNEL_CONFIGURATION_STEREO: 692 channelCount = 2; 693 break; 694 default: 695 if ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) { 696 // input channel configuration features unsupported channels 697 loge("getMinBufferSize(): Invalid channel configuration."); 698 return ERROR_BAD_VALUE; 699 } else { 700 channelCount = Integer.bitCount(channelConfig); 701 } 702 } 703 704 if ((audioFormat != AudioFormat.ENCODING_PCM_16BIT) 705 && (audioFormat != AudioFormat.ENCODING_PCM_8BIT)) { 706 loge("getMinBufferSize(): Invalid audio format."); 707 return ERROR_BAD_VALUE; 708 } 709 710 // sample rate, note these values are subject to change 711 if ( (sampleRateInHz < SAMPLE_RATE_HZ_MIN) || (sampleRateInHz > SAMPLE_RATE_HZ_MAX) ) { 712 loge("getMinBufferSize(): " + sampleRateInHz + " Hz is not a supported sample rate."); 713 return ERROR_BAD_VALUE; 714 } 715 716 int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat); 717 if (size <= 0) { 718 loge("getMinBufferSize(): error querying hardware"); 719 return ERROR; 720 } 721 else { 722 return size; 723 } 724 } 725 726 /** 727 * Returns the audio session ID. 728 * 729 * @return the ID of the audio session this AudioTrack belongs to. 730 */ 731 public int getAudioSessionId() { 732 return mSessionId; 733 } 734 735 /** 736 * Poll for a timestamp on demand. 737 * 738 * Use if {@link TimestampListener} is not delivered often enough for your needs, 739 * or if you need to get the most recent timestamp outside of the event callback handler. 740 * Calling this method too often may be inefficient; 741 * if you need a high-resolution mapping between frame position and presentation time, 742 * consider implementing that at application level, based on low-resolution timestamps. 743 * The audio data at the returned position may either already have been 744 * presented, or may have not yet been presented but is committed to be presented. 745 * It is not possible to request the time corresponding to a particular position, 746 * or to request the (fractional) position corresponding to a particular time. 747 * If you need such features, consider implementing them at application level. 748 * 749 * @param timestamp a reference to a non-null AudioTimestamp instance allocated 750 * and owned by caller, or null. 751 * @return that same instance if timestamp parameter is non-null and a timestamp is available, 752 * or a reference to a new AudioTimestamp instance which is now owned by caller 753 * if timestamp parameter is null and a timestamp is available, 754 * or null if no timestamp is available. In either successful case, 755 * the AudioTimestamp instance is filled in with a position in frame units, together 756 * with the estimated time when that frame was presented or is committed to 757 * be presented. 758 * In the case that no timestamp is available, any supplied instance is left unaltered. 759 * 760 * @hide 761 */ 762 public AudioTimestamp getTimestamp(AudioTimestamp timestamp) 763 { 764 // It's unfortunate, but we have to either create garbage every time or use synchronized 765 long[] longArray = new long[2]; 766 int ret = native_get_timestamp(longArray); 767 if (ret == SUCCESS) { 768 if (timestamp == null) { 769 timestamp = new AudioTimestamp(); 770 } 771 timestamp.framePosition = longArray[0]; 772 timestamp.nanoTime = longArray[1]; 773 } else { 774 timestamp = null; 775 } 776 return timestamp; 777 } 778 779 780 //-------------------------------------------------------------------------- 781 // Initialization / configuration 782 //-------------------- 783 /** 784 * Sets the listener the AudioTrack notifies when a previously set marker is reached or 785 * for each periodic playback head position update. 786 * Notifications will be received in the same thread as the one in which the AudioTrack 787 * instance was created. 788 * @param listener 789 */ 790 public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener) { 791 setPlaybackPositionUpdateListener(listener, null); 792 } 793 794 /** 795 * Sets the listener the AudioTrack notifies when a previously set marker is reached or 796 * for each periodic playback head position update. 797 * Use this method to receive AudioTrack events in the Handler associated with another 798 * thread than the one in which you created the AudioTrack instance. 799 * @param listener 800 * @param handler the Handler that will receive the event notification messages. 801 */ 802 public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener, 803 Handler handler) { 804 if (listener != null) { 805 mEventHandlerDelegate = new NativeEventHandlerDelegate(this, listener, handler); 806 } else { 807 mEventHandlerDelegate = null; 808 } 809 } 810 811 812 813 /** 814 * Sets the specified left/right output volume values on the AudioTrack. Values are clamped 815 * to the ({@link #getMinVolume()}, {@link #getMaxVolume()}) interval if outside this range. 816 * @param leftVolume output attenuation for the left channel. A value of 0.0f is silence, 817 * a value of 1.0f is no attenuation. 818 * @param rightVolume output attenuation for the right channel 819 * @return error code or success, see {@link #SUCCESS}, 820 * {@link #ERROR_INVALID_OPERATION} 821 */ 822 public int setStereoVolume(float leftVolume, float rightVolume) { 823 if (mState == STATE_UNINITIALIZED) { 824 return ERROR_INVALID_OPERATION; 825 } 826 827 // clamp the volumes 828 if (leftVolume < getMinVolume()) { 829 leftVolume = getMinVolume(); 830 } 831 if (leftVolume > getMaxVolume()) { 832 leftVolume = getMaxVolume(); 833 } 834 if (rightVolume < getMinVolume()) { 835 rightVolume = getMinVolume(); 836 } 837 if (rightVolume > getMaxVolume()) { 838 rightVolume = getMaxVolume(); 839 } 840 841 native_setVolume(leftVolume, rightVolume); 842 843 return SUCCESS; 844 } 845 846 847 /** 848 * Similar, except set volume of all channels to same value. 849 * @hide 850 */ 851 public int setVolume(float volume) { 852 return setStereoVolume(volume, volume); 853 } 854 855 856 /** 857 * Sets the playback sample rate for this track. This sets the sampling rate at which 858 * the audio data will be consumed and played back 859 * (as set by the sampleRateInHz parameter in the 860 * {@link #AudioTrack(int, int, int, int, int, int)} constructor), 861 * not the original sampling rate of the 862 * content. For example, setting it to half the sample rate of the content will cause the 863 * playback to last twice as long, but will also result in a pitch shift down by one octave. 864 * The valid sample rate range is from 1 Hz to twice the value returned by 865 * {@link #getNativeOutputSampleRate(int)}. 866 * @param sampleRateInHz the sample rate expressed in Hz 867 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 868 * {@link #ERROR_INVALID_OPERATION} 869 */ 870 public int setPlaybackRate(int sampleRateInHz) { 871 if (mState != STATE_INITIALIZED) { 872 return ERROR_INVALID_OPERATION; 873 } 874 if (sampleRateInHz <= 0) { 875 return ERROR_BAD_VALUE; 876 } 877 return native_set_playback_rate(sampleRateInHz); 878 } 879 880 881 /** 882 * Sets the position of the notification marker. At most one marker can be active. 883 * @param markerInFrames marker position in wrapping frame units similar to 884 * {@link #getPlaybackHeadPosition}, or zero to disable the marker. 885 * To set a marker at a position which would appear as zero due to wraparound, 886 * a workaround is to use a non-zero position near zero, such as -1 or 1. 887 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 888 * {@link #ERROR_INVALID_OPERATION} 889 */ 890 public int setNotificationMarkerPosition(int markerInFrames) { 891 if (mState == STATE_UNINITIALIZED) { 892 return ERROR_INVALID_OPERATION; 893 } 894 return native_set_marker_pos(markerInFrames); 895 } 896 897 898 /** 899 * Sets the period for the periodic notification event. 900 * @param periodInFrames update period expressed in frames 901 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_INVALID_OPERATION} 902 */ 903 public int setPositionNotificationPeriod(int periodInFrames) { 904 if (mState == STATE_UNINITIALIZED) { 905 return ERROR_INVALID_OPERATION; 906 } 907 return native_set_pos_update_period(periodInFrames); 908 } 909 910 911 /** 912 * Sets the playback head position. 913 * The track must be stopped or paused for the position to be changed, 914 * and must use the {@link #MODE_STATIC} mode. 915 * @param positionInFrames playback head position expressed in frames 916 * Zero corresponds to start of buffer. 917 * The position must not be greater than the buffer size in frames, or negative. 918 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 919 * {@link #ERROR_INVALID_OPERATION} 920 */ 921 public int setPlaybackHeadPosition(int positionInFrames) { 922 if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED || 923 getPlayState() == PLAYSTATE_PLAYING) { 924 return ERROR_INVALID_OPERATION; 925 } 926 if (!(0 <= positionInFrames && positionInFrames <= mNativeBufferSizeInFrames)) { 927 return ERROR_BAD_VALUE; 928 } 929 return native_set_position(positionInFrames); 930 } 931 932 /** 933 * Sets the loop points and the loop count. The loop can be infinite. 934 * Similarly to setPlaybackHeadPosition, 935 * the track must be stopped or paused for the loop points to be changed, 936 * and must use the {@link #MODE_STATIC} mode. 937 * @param startInFrames loop start marker expressed in frames 938 * Zero corresponds to start of buffer. 939 * The start marker must not be greater than or equal to the buffer size in frames, or negative. 940 * @param endInFrames loop end marker expressed in frames 941 * The total buffer size in frames corresponds to end of buffer. 942 * The end marker must not be greater than the buffer size in frames. 943 * For looping, the end marker must not be less than or equal to the start marker, 944 * but to disable looping 945 * it is permitted for start marker, end marker, and loop count to all be 0. 946 * @param loopCount the number of times the loop is looped. 947 * A value of -1 means infinite looping, and 0 disables looping. 948 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 949 * {@link #ERROR_INVALID_OPERATION} 950 */ 951 public int setLoopPoints(int startInFrames, int endInFrames, int loopCount) { 952 if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED || 953 getPlayState() == PLAYSTATE_PLAYING) { 954 return ERROR_INVALID_OPERATION; 955 } 956 if (loopCount == 0) { 957 ; // explicitly allowed as an exception to the loop region range check 958 } else if (!(0 <= startInFrames && startInFrames < mNativeBufferSizeInFrames && 959 startInFrames < endInFrames && endInFrames <= mNativeBufferSizeInFrames)) { 960 return ERROR_BAD_VALUE; 961 } 962 return native_set_loop(startInFrames, endInFrames, loopCount); 963 } 964 965 /** 966 * Sets the initialization state of the instance. This method was originally intended to be used 967 * in an AudioTrack subclass constructor to set a subclass-specific post-initialization state. 968 * However, subclasses of AudioTrack are no longer recommended, so this method is obsolete. 969 * @param state the state of the AudioTrack instance 970 * @deprecated Only accessible by subclasses, which are not recommended for AudioTrack. 971 */ 972 @Deprecated 973 protected void setState(int state) { 974 mState = state; 975 } 976 977 978 //--------------------------------------------------------- 979 // Transport control methods 980 //-------------------- 981 /** 982 * Starts playing an AudioTrack. 983 * If track's creation mode is {@link #MODE_STATIC}, you must have called write() prior. 984 * 985 * @throws IllegalStateException 986 */ 987 public void play() 988 throws IllegalStateException { 989 if (mState != STATE_INITIALIZED) { 990 throw new IllegalStateException("play() called on uninitialized AudioTrack."); 991 } 992 993 synchronized(mPlayStateLock) { 994 native_start(); 995 mPlayState = PLAYSTATE_PLAYING; 996 } 997 } 998 999 /** 1000 * Stops playing the audio data. 1001 * When used on an instance created in {@link #MODE_STREAM} mode, audio will stop playing 1002 * after the last buffer that was written has been played. For an immediate stop, use 1003 * {@link #pause()}, followed by {@link #flush()} to discard audio data that hasn't been played 1004 * back yet. 1005 * @throws IllegalStateException 1006 */ 1007 public void stop() 1008 throws IllegalStateException { 1009 if (mState != STATE_INITIALIZED) { 1010 throw new IllegalStateException("stop() called on uninitialized AudioTrack."); 1011 } 1012 1013 // stop playing 1014 synchronized(mPlayStateLock) { 1015 native_stop(); 1016 mPlayState = PLAYSTATE_STOPPED; 1017 } 1018 } 1019 1020 /** 1021 * Pauses the playback of the audio data. Data that has not been played 1022 * back will not be discarded. Subsequent calls to {@link #play} will play 1023 * this data back. See {@link #flush()} to discard this data. 1024 * 1025 * @throws IllegalStateException 1026 */ 1027 public void pause() 1028 throws IllegalStateException { 1029 if (mState != STATE_INITIALIZED) { 1030 throw new IllegalStateException("pause() called on uninitialized AudioTrack."); 1031 } 1032 //logd("pause()"); 1033 1034 // pause playback 1035 synchronized(mPlayStateLock) { 1036 native_pause(); 1037 mPlayState = PLAYSTATE_PAUSED; 1038 } 1039 } 1040 1041 1042 //--------------------------------------------------------- 1043 // Audio data supply 1044 //-------------------- 1045 1046 /** 1047 * Flushes the audio data currently queued for playback. Any data that has 1048 * not been played back will be discarded. No-op if not stopped or paused, 1049 * or if the track's creation mode is not {@link #MODE_STREAM}. 1050 */ 1051 public void flush() { 1052 if (mState == STATE_INITIALIZED) { 1053 // flush the data in native layer 1054 native_flush(); 1055 } 1056 1057 } 1058 1059 /** 1060 * Writes the audio data to the audio sink for playback (streaming mode), 1061 * or copies audio data for later playback (static buffer mode). 1062 * In streaming mode, will block until all data has been written to the audio sink. 1063 * In static buffer mode, copies the data to the buffer starting at offset 0. 1064 * Note that the actual playback of this data might occur after this function 1065 * returns. This function is thread safe with respect to {@link #stop} calls, 1066 * in which case all of the specified data might not be written to the audio sink. 1067 * 1068 * @param audioData the array that holds the data to play. 1069 * @param offsetInBytes the offset expressed in bytes in audioData where the data to play 1070 * starts. 1071 * @param sizeInBytes the number of bytes to read in audioData after the offset. 1072 * @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION} 1073 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1074 * the parameters don't resolve to valid data and indexes. 1075 */ 1076 1077 public int write(byte[] audioData, int offsetInBytes, int sizeInBytes) { 1078 1079 if (mState == STATE_UNINITIALIZED) { 1080 return ERROR_INVALID_OPERATION; 1081 } 1082 1083 if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0) 1084 || (offsetInBytes + sizeInBytes < 0) // detect integer overflow 1085 || (offsetInBytes + sizeInBytes > audioData.length)) { 1086 return ERROR_BAD_VALUE; 1087 } 1088 1089 int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat); 1090 1091 if ((mDataLoadMode == MODE_STATIC) 1092 && (mState == STATE_NO_STATIC_DATA) 1093 && (ret > 0)) { 1094 // benign race with respect to other APIs that read mState 1095 mState = STATE_INITIALIZED; 1096 } 1097 1098 return ret; 1099 } 1100 1101 1102 /** 1103 * Writes the audio data to the audio sink for playback (streaming mode), 1104 * or copies audio data for later playback (static buffer mode). 1105 * In streaming mode, will block until all data has been written to the audio sink. 1106 * In static buffer mode, copies the data to the buffer starting at offset 0. 1107 * Note that the actual playback of this data might occur after this function 1108 * returns. This function is thread safe with respect to {@link #stop} calls, 1109 * in which case all of the specified data might not be written to the audio sink. 1110 * 1111 * @param audioData the array that holds the data to play. 1112 * @param offsetInShorts the offset expressed in shorts in audioData where the data to play 1113 * starts. 1114 * @param sizeInShorts the number of shorts to read in audioData after the offset. 1115 * @return the number of shorts that were written or {@link #ERROR_INVALID_OPERATION} 1116 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1117 * the parameters don't resolve to valid data and indexes. 1118 */ 1119 1120 public int write(short[] audioData, int offsetInShorts, int sizeInShorts) { 1121 1122 if (mState == STATE_UNINITIALIZED) { 1123 return ERROR_INVALID_OPERATION; 1124 } 1125 1126 if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0) 1127 || (offsetInShorts + sizeInShorts < 0) // detect integer overflow 1128 || (offsetInShorts + sizeInShorts > audioData.length)) { 1129 return ERROR_BAD_VALUE; 1130 } 1131 1132 int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat); 1133 1134 if ((mDataLoadMode == MODE_STATIC) 1135 && (mState == STATE_NO_STATIC_DATA) 1136 && (ret > 0)) { 1137 // benign race with respect to other APIs that read mState 1138 mState = STATE_INITIALIZED; 1139 } 1140 1141 return ret; 1142 } 1143 1144 1145 /** 1146 * Notifies the native resource to reuse the audio data already loaded in the native 1147 * layer, that is to rewind to start of buffer. 1148 * The track's creation mode must be {@link #MODE_STATIC}. 1149 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1150 * {@link #ERROR_INVALID_OPERATION} 1151 */ 1152 public int reloadStaticData() { 1153 if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED) { 1154 return ERROR_INVALID_OPERATION; 1155 } 1156 return native_reload_static(); 1157 } 1158 1159 //-------------------------------------------------------------------------- 1160 // Audio effects management 1161 //-------------------- 1162 1163 /** 1164 * Attaches an auxiliary effect to the audio track. A typical auxiliary 1165 * effect is a reverberation effect which can be applied on any sound source 1166 * that directs a certain amount of its energy to this effect. This amount 1167 * is defined by setAuxEffectSendLevel(). 1168 * {@see #setAuxEffectSendLevel(float)}. 1169 * <p>After creating an auxiliary effect (e.g. 1170 * {@link android.media.audiofx.EnvironmentalReverb}), retrieve its ID with 1171 * {@link android.media.audiofx.AudioEffect#getId()} and use it when calling 1172 * this method to attach the audio track to the effect. 1173 * <p>To detach the effect from the audio track, call this method with a 1174 * null effect id. 1175 * 1176 * @param effectId system wide unique id of the effect to attach 1177 * @return error code or success, see {@link #SUCCESS}, 1178 * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR_BAD_VALUE} 1179 */ 1180 public int attachAuxEffect(int effectId) { 1181 if (mState == STATE_UNINITIALIZED) { 1182 return ERROR_INVALID_OPERATION; 1183 } 1184 return native_attachAuxEffect(effectId); 1185 } 1186 1187 /** 1188 * Sets the send level of the audio track to the attached auxiliary effect 1189 * {@link #attachAuxEffect(int)}. The level value range is 0.0f to 1.0f. 1190 * Values are clamped to the (0.0f, 1.0f) interval if outside this range. 1191 * <p>By default the send level is 0.0f, so even if an effect is attached to the player 1192 * this method must be called for the effect to be applied. 1193 * <p>Note that the passed level value is a raw scalar. UI controls should be scaled 1194 * logarithmically: the gain applied by audio framework ranges from -72dB to 0dB, 1195 * so an appropriate conversion from linear UI input x to level is: 1196 * x == 0 -> level = 0 1197 * 0 < x <= R -> level = 10^(72*(x-R)/20/R) 1198 * 1199 * @param level send level scalar 1200 * @return error code or success, see {@link #SUCCESS}, 1201 * {@link #ERROR_INVALID_OPERATION} 1202 */ 1203 public int setAuxEffectSendLevel(float level) { 1204 if (mState == STATE_UNINITIALIZED) { 1205 return ERROR_INVALID_OPERATION; 1206 } 1207 // clamp the level 1208 if (level < getMinVolume()) { 1209 level = getMinVolume(); 1210 } 1211 if (level > getMaxVolume()) { 1212 level = getMaxVolume(); 1213 } 1214 native_setAuxEffectSendLevel(level); 1215 return SUCCESS; 1216 } 1217 1218 //--------------------------------------------------------- 1219 // Interface definitions 1220 //-------------------- 1221 /** 1222 * Interface definition for a callback to be invoked when the playback head position of 1223 * an AudioTrack has reached a notification marker or has increased by a certain period. 1224 */ 1225 public interface OnPlaybackPositionUpdateListener { 1226 /** 1227 * Called on the listener to notify it that the previously set marker has been reached 1228 * by the playback head. 1229 */ 1230 void onMarkerReached(AudioTrack track); 1231 1232 /** 1233 * Called on the listener to periodically notify it that the playback head has reached 1234 * a multiple of the notification period. 1235 */ 1236 void onPeriodicNotification(AudioTrack track); 1237 } 1238 1239 1240 //--------------------------------------------------------- 1241 // Inner classes 1242 //-------------------- 1243 /** 1244 * Helper class to handle the forwarding of native events to the appropriate listener 1245 * (potentially) handled in a different thread 1246 */ 1247 private class NativeEventHandlerDelegate { 1248 private final Handler mHandler; 1249 1250 NativeEventHandlerDelegate(final AudioTrack track, 1251 final OnPlaybackPositionUpdateListener listener, 1252 Handler handler) { 1253 // find the looper for our new event handler 1254 Looper looper; 1255 if (handler != null) { 1256 looper = handler.getLooper(); 1257 } else { 1258 // no given handler, use the looper the AudioTrack was created in 1259 looper = mInitializationLooper; 1260 } 1261 1262 // construct the event handler with this looper 1263 if (looper != null) { 1264 // implement the event handler delegate 1265 mHandler = new Handler(looper) { 1266 @Override 1267 public void handleMessage(Message msg) { 1268 if (track == null) { 1269 return; 1270 } 1271 switch(msg.what) { 1272 case NATIVE_EVENT_MARKER: 1273 if (listener != null) { 1274 listener.onMarkerReached(track); 1275 } 1276 break; 1277 case NATIVE_EVENT_NEW_POS: 1278 if (listener != null) { 1279 listener.onPeriodicNotification(track); 1280 } 1281 break; 1282 default: 1283 loge("Unknown native event type: " + msg.what); 1284 break; 1285 } 1286 } 1287 }; 1288 } else { 1289 mHandler = null; 1290 } 1291 } 1292 1293 Handler getHandler() { 1294 return mHandler; 1295 } 1296 } 1297 1298 1299 //--------------------------------------------------------- 1300 // Java methods called from the native side 1301 //-------------------- 1302 @SuppressWarnings("unused") 1303 private static void postEventFromNative(Object audiotrack_ref, 1304 int what, int arg1, int arg2, Object obj) { 1305 //logd("Event posted from the native side: event="+ what + " args="+ arg1+" "+arg2); 1306 AudioTrack track = (AudioTrack)((WeakReference)audiotrack_ref).get(); 1307 if (track == null) { 1308 return; 1309 } 1310 1311 NativeEventHandlerDelegate delegate = track.mEventHandlerDelegate; 1312 if (delegate != null) { 1313 Handler handler = delegate.getHandler(); 1314 if (handler != null) { 1315 Message m = handler.obtainMessage(what, arg1, arg2, obj); 1316 handler.sendMessage(m); 1317 } 1318 } 1319 1320 } 1321 1322 1323 //--------------------------------------------------------- 1324 // Native methods called from the Java side 1325 //-------------------- 1326 1327 private native final int native_setup(Object audiotrack_this, 1328 int streamType, int sampleRate, int nbChannels, int audioFormat, 1329 int buffSizeInBytes, int mode, int[] sessionId); 1330 1331 private native final void native_finalize(); 1332 1333 private native final void native_release(); 1334 1335 private native final void native_start(); 1336 1337 private native final void native_stop(); 1338 1339 private native final void native_pause(); 1340 1341 private native final void native_flush(); 1342 1343 private native final int native_write_byte(byte[] audioData, 1344 int offsetInBytes, int sizeInBytes, int format); 1345 1346 private native final int native_write_short(short[] audioData, 1347 int offsetInShorts, int sizeInShorts, int format); 1348 1349 private native final int native_reload_static(); 1350 1351 private native final int native_get_native_frame_count(); 1352 1353 private native final void native_setVolume(float leftVolume, float rightVolume); 1354 1355 private native final int native_set_playback_rate(int sampleRateInHz); 1356 private native final int native_get_playback_rate(); 1357 1358 private native final int native_set_marker_pos(int marker); 1359 private native final int native_get_marker_pos(); 1360 1361 private native final int native_set_pos_update_period(int updatePeriod); 1362 private native final int native_get_pos_update_period(); 1363 1364 private native final int native_set_position(int position); 1365 private native final int native_get_position(); 1366 1367 private native final int native_get_latency(); 1368 1369 // longArray must be a non-null array of length >= 2 1370 // [0] is assigned the frame position 1371 // [1] is assigned the time in CLOCK_MONOTONIC nanoseconds 1372 private native final int native_get_timestamp(long[] longArray); 1373 1374 private native final int native_set_loop(int start, int end, int loopCount); 1375 1376 static private native final int native_get_output_sample_rate(int streamType); 1377 static private native final int native_get_min_buff_size( 1378 int sampleRateInHz, int channelConfig, int audioFormat); 1379 1380 private native final int native_attachAuxEffect(int effectId); 1381 private native final void native_setAuxEffectSendLevel(float level); 1382 1383 //--------------------------------------------------------- 1384 // Utility methods 1385 //------------------ 1386 1387 private static void logd(String msg) { 1388 Log.d(TAG, msg); 1389 } 1390 1391 private static void loge(String msg) { 1392 Log.e(TAG, msg); 1393 } 1394 1395} 1396