1 /* 2 * Copyright (C) 2007 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 17 #ifndef ANDROID_AUDIOTRACK_H 18 #define ANDROID_AUDIOTRACK_H 19 20 #include <stdint.h> 21 #include <sys/types.h> 22 23 #include <media/IAudioFlinger.h> 24 #include <media/IAudioTrack.h> 25 #include <media/AudioSystem.h> 26 27 #include <utils/RefBase.h> 28 #include <utils/Errors.h> 29 #include <binder/IInterface.h> 30 #include <binder/IMemory.h> 31 #include <cutils/sched_policy.h> 32 #include <utils/threads.h> 33 34 namespace android { 35 36 // ---------------------------------------------------------------------------- 37 38 class audio_track_cblk_t; 39 class AudioTrackClientProxy; 40 41 // ---------------------------------------------------------------------------- 42 43 class AudioTrack : virtual public RefBase 44 { 45 public: 46 enum channel_index { 47 MONO = 0, 48 LEFT = 0, 49 RIGHT = 1 50 }; 51 52 /* Events used by AudioTrack callback function (audio_track_cblk_t). 53 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 54 */ 55 enum event_type { 56 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 57 // If this event is delivered but the callback handler 58 // does not want to write more data, the handler must explicitly 59 // ignore the event by setting frameCount to zero. 60 EVENT_UNDERRUN = 1, // Buffer underrun occurred. 61 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 62 // loop start if loop count was not 0. 63 EVENT_MARKER = 3, // Playback head is at the specified marker position 64 // (See setMarkerPosition()). 65 EVENT_NEW_POS = 4, // Playback head is at a new position 66 // (See setPositionUpdatePeriod()). 67 EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer. 68 }; 69 70 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 71 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 72 */ 73 74 class Buffer 75 { 76 public: 77 size_t frameCount; // number of sample frames corresponding to size; 78 // on input it is the number of frames desired, 79 // on output is the number of frames actually filled 80 81 size_t size; // input/output in byte units 82 union { 83 void* raw; 84 short* i16; // signed 16-bit 85 int8_t* i8; // unsigned 8-bit, offset by 0x80 86 }; 87 }; 88 89 90 /* As a convenience, if a callback is supplied, a handler thread 91 * is automatically created with the appropriate priority. This thread 92 * invokes the callback when a new buffer becomes available or various conditions occur. 93 * Parameters: 94 * 95 * event: type of event notified (see enum AudioTrack::event_type). 96 * user: Pointer to context for use by the callback receiver. 97 * info: Pointer to optional parameter according to event type: 98 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 99 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 100 * written. 101 * - EVENT_UNDERRUN: unused. 102 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 103 * - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames. 104 * - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames. 105 * - EVENT_BUFFER_END: unused. 106 */ 107 108 typedef void (*callback_t)(int event, void* user, void *info); 109 110 /* Returns the minimum frame count required for the successful creation of 111 * an AudioTrack object. 112 * Returned status (from utils/Errors.h) can be: 113 * - NO_ERROR: successful operation 114 * - NO_INIT: audio server or audio hardware not initialized 115 */ 116 117 static status_t getMinFrameCount(size_t* frameCount, 118 audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, 119 uint32_t sampleRate = 0); 120 121 /* Constructs an uninitialized AudioTrack. No connection with 122 * AudioFlinger takes place. Use set() after this. 123 */ 124 AudioTrack(); 125 126 /* Creates an AudioTrack object and registers it with AudioFlinger. 127 * Once created, the track needs to be started before it can be used. 128 * Unspecified values are set to appropriate default values. 129 * With this constructor, the track is configured for streaming mode. 130 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 131 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is deprecated. 132 * 133 * Parameters: 134 * 135 * streamType: Select the type of audio stream this track is attached to 136 * (e.g. AUDIO_STREAM_MUSIC). 137 * sampleRate: Track sampling rate in Hz. 138 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 139 * 16 bits per sample). 140 * channelMask: Channel mask. 141 * frameCount: Minimum size of track PCM buffer in frames. This defines the 142 * application's contribution to the 143 * latency of the track. The actual size selected by the AudioTrack could be 144 * larger if the requested size is not compatible with current audio HAL 145 * configuration. Zero means to use a default value. 146 * flags: See comments on audio_output_flags_t in <system/audio.h>. 147 * cbf: Callback function. If not null, this function is called periodically 148 * to provide new data and inform of marker, position updates, etc. 149 * user: Context for use by the callback receiver. 150 * notificationFrames: The callback function is called each time notificationFrames PCM 151 * frames have been consumed from track input buffer. 152 * sessionId: Specific session ID, or zero to use default. 153 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 154 * If not present in parameter list, then fixed at false. 155 */ 156 157 AudioTrack( audio_stream_type_t streamType, 158 uint32_t sampleRate = 0, 159 audio_format_t format = AUDIO_FORMAT_DEFAULT, 160 audio_channel_mask_t channelMask = 0, 161 int frameCount = 0, 162 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 163 callback_t cbf = NULL, 164 void* user = NULL, 165 int notificationFrames = 0, 166 int sessionId = 0); 167 168 /* Creates an audio track and registers it with AudioFlinger. 169 * With this constructor, the track is configured for static buffer mode. 170 * The format must not be 8-bit linear PCM. 171 * Data to be rendered is passed in a shared memory buffer 172 * identified by the argument sharedBuffer, which must be non-0. 173 * The memory should be initialized to the desired data before calling start(). 174 * The write() method is not supported in this case. 175 * It is recommended to pass a callback function to be notified of playback end by an 176 * EVENT_UNDERRUN event. 177 * FIXME EVENT_MORE_DATA still occurs; it must be ignored. 178 */ 179 180 AudioTrack( audio_stream_type_t streamType, 181 uint32_t sampleRate = 0, 182 audio_format_t format = AUDIO_FORMAT_DEFAULT, 183 audio_channel_mask_t channelMask = 0, 184 const sp<IMemory>& sharedBuffer = 0, 185 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 186 callback_t cbf = NULL, 187 void* user = NULL, 188 int notificationFrames = 0, 189 int sessionId = 0); 190 191 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 192 * Also destroys all resources associated with the AudioTrack. 193 */ 194 ~AudioTrack(); 195 196 /* Initialize an uninitialized AudioTrack. 197 * Returned status (from utils/Errors.h) can be: 198 * - NO_ERROR: successful initialization 199 * - INVALID_OPERATION: AudioTrack is already initialized 200 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 201 * - NO_INIT: audio server or audio hardware not initialized 202 * If sharedBuffer is non-0, the frameCount parameter is ignored and 203 * replaced by the shared buffer's total allocated size in frame units. 204 */ 205 status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, 206 uint32_t sampleRate = 0, 207 audio_format_t format = AUDIO_FORMAT_DEFAULT, 208 audio_channel_mask_t channelMask = 0, 209 int frameCount = 0, 210 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 211 callback_t cbf = NULL, 212 void* user = NULL, 213 int notificationFrames = 0, 214 const sp<IMemory>& sharedBuffer = 0, 215 bool threadCanCallJava = false, 216 int sessionId = 0); 217 218 /* Result of constructing the AudioTrack. This must be checked 219 * before using any AudioTrack API (except for set()), because using 220 * an uninitialized AudioTrack produces undefined results. 221 * See set() method above for possible return codes. 222 */ 223 status_t initCheck() const { return mStatus; } 224 225 /* Returns this track's estimated latency in milliseconds. 226 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 227 * and audio hardware driver. 228 */ 229 uint32_t latency() const { return mLatency; } 230 231 /* getters, see constructors and set() */ 232 233 audio_stream_type_t streamType() const { return mStreamType; } 234 audio_format_t format() const { return mFormat; } 235 236 /* Return frame size in bytes, which for linear PCM is channelCount * (bit depth per channel / 8). 237 * channelCount is determined from channelMask, and bit depth comes from format. 238 * For non-linear formats, the frame size is typically 1 byte. 239 */ 240 uint32_t channelCount() const { return mChannelCount; } 241 242 uint32_t frameCount() const { return mFrameCount; } 243 size_t frameSize() const { return mFrameSize; } 244 245 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 246 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 247 248 /* After it's created the track is not active. Call start() to 249 * make it active. If set, the callback will start being called. 250 * If the track was previously paused, volume is ramped up over the first mix buffer. 251 */ 252 void start(); 253 254 /* Stop a track. 255 * In static buffer mode, the track is stopped immediately. 256 * In streaming mode, the callback will cease being called and 257 * obtainBuffer returns STOPPED. Note that obtainBuffer() still works 258 * and will fill up buffers until the pool is exhausted. 259 * The stop does not occur immediately: any data remaining in the buffer 260 * is first drained, mixed, and output, and only then is the track marked as stopped. 261 */ 262 void stop(); 263 bool stopped() const; 264 265 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 266 * This has the effect of draining the buffers without mixing or output. 267 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 268 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 269 */ 270 void flush(); 271 272 /* Pause a track. After pause, the callback will cease being called and 273 * obtainBuffer returns STOPPED. Note that obtainBuffer() still works 274 * and will fill up buffers until the pool is exhausted. 275 * Volume is ramped down over the next mix buffer following the pause request, 276 * and then the track is marked as paused. It can be resumed with ramp up by start(). 277 */ 278 void pause(); 279 280 /* Set volume for this track, mostly used for games' sound effects 281 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 282 * This is the older API. New applications should use setVolume(float) when possible. 283 */ 284 status_t setVolume(float left, float right); 285 286 /* Set volume for all channels. This is the preferred API for new applications, 287 * especially for multi-channel content. 288 */ 289 status_t setVolume(float volume); 290 291 /* Set the send level for this track. An auxiliary effect should be attached 292 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 293 */ 294 status_t setAuxEffectSendLevel(float level); 295 void getAuxEffectSendLevel(float* level) const; 296 297 /* Set sample rate for this track in Hz, mostly used for games' sound effects 298 */ 299 status_t setSampleRate(uint32_t sampleRate); 300 301 /* Return current sample rate in Hz, or 0 if unknown */ 302 uint32_t getSampleRate() const; 303 304 /* Enables looping and sets the start and end points of looping. 305 * Only supported for static buffer mode. 306 * 307 * Parameters: 308 * 309 * loopStart: loop start expressed as the number of PCM frames played since AudioTrack start. 310 * loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start. 311 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 312 * pending or active loop. loopCount = -1 means infinite looping. 313 * 314 * For proper operation the following condition must be respected: 315 * (loopEnd-loopStart) <= framecount() 316 */ 317 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 318 319 /* Sets marker position. When playback reaches the number of frames specified, a callback with 320 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 321 * notification callback. To set a marker at a position which would compute as 0, 322 * a workaround is to the set the marker at a nearby position such as -1 or 1. 323 * If the AudioTrack has been opened with no callback function associated, the operation will 324 * fail. 325 * 326 * Parameters: 327 * 328 * marker: marker position expressed in wrapping (overflow) frame units, 329 * like the return value of getPosition(). 330 * 331 * Returned status (from utils/Errors.h) can be: 332 * - NO_ERROR: successful operation 333 * - INVALID_OPERATION: the AudioTrack has no callback installed. 334 */ 335 status_t setMarkerPosition(uint32_t marker); 336 status_t getMarkerPosition(uint32_t *marker) const; 337 338 /* Sets position update period. Every time the number of frames specified has been played, 339 * a callback with event type EVENT_NEW_POS is called. 340 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 341 * callback. 342 * If the AudioTrack has been opened with no callback function associated, the operation will 343 * fail. 344 * Extremely small values may be rounded up to a value the implementation can support. 345 * 346 * Parameters: 347 * 348 * updatePeriod: position update notification period expressed in frames. 349 * 350 * Returned status (from utils/Errors.h) can be: 351 * - NO_ERROR: successful operation 352 * - INVALID_OPERATION: the AudioTrack has no callback installed. 353 */ 354 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 355 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 356 357 /* Sets playback head position within AudioTrack buffer. The new position is specified 358 * in number of frames. 359 * This method must be called with the AudioTrack in paused or stopped state. 360 * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames. 361 * Therefore using this method makes sense only when playing a "static" audio buffer 362 * as opposed to streaming. 363 * The getPosition() method on the other hand returns the total number of frames played since 364 * playback start. 365 * 366 * Parameters: 367 * 368 * position: New playback head position within AudioTrack buffer. 369 * 370 * Returned status (from utils/Errors.h) can be: 371 * - NO_ERROR: successful operation 372 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 373 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 374 * buffer 375 */ 376 status_t setPosition(uint32_t position); 377 378 /* Return the total number of frames played since playback start. 379 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 380 * It is reset to zero by flush(), reload(), and stop(). 381 */ 382 status_t getPosition(uint32_t *position); 383 384 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 385 * rewriting the buffer before restarting playback after a stop. 386 * This method must be called with the AudioTrack in paused or stopped state. 387 * Not allowed in streaming mode. 388 * 389 * Returned status (from utils/Errors.h) can be: 390 * - NO_ERROR: successful operation 391 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 392 */ 393 status_t reload(); 394 395 /* Returns a handle on the audio output used by this AudioTrack. 396 * 397 * Parameters: 398 * none. 399 * 400 * Returned value: 401 * handle on audio hardware output 402 */ 403 audio_io_handle_t getOutput(); 404 405 /* Returns the unique session ID associated with this track. 406 * 407 * Parameters: 408 * none. 409 * 410 * Returned value: 411 * AudioTrack session ID. 412 */ 413 int getSessionId() const { return mSessionId; } 414 415 /* Attach track auxiliary output to specified effect. Use effectId = 0 416 * to detach track from effect. 417 * 418 * Parameters: 419 * 420 * effectId: effectId obtained from AudioEffect::id(). 421 * 422 * Returned status (from utils/Errors.h) can be: 423 * - NO_ERROR: successful operation 424 * - INVALID_OPERATION: the effect is not an auxiliary effect. 425 * - BAD_VALUE: The specified effect ID is invalid 426 */ 427 status_t attachAuxEffect(int effectId); 428 429 /* Obtains a buffer of "frameCount" frames. The buffer must be 430 * filled entirely, and then released with releaseBuffer(). 431 * If the track is stopped, obtainBuffer() returns 432 * STOPPED instead of NO_ERROR as long as there are buffers available, 433 * at which point NO_MORE_BUFFERS is returned. 434 * Buffers will be returned until the pool 435 * is exhausted, at which point obtainBuffer() will either block 436 * or return WOULD_BLOCK depending on the value of the "blocking" 437 * parameter. 438 * 439 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 440 * which should use write() or callback EVENT_MORE_DATA instead. 441 * 442 * Interpretation of waitCount: 443 * +n limits wait time to n * WAIT_PERIOD_MS, 444 * -1 causes an (almost) infinite wait time, 445 * 0 non-blocking. 446 * 447 * Buffer fields 448 * On entry: 449 * frameCount number of frames requested 450 * After error return: 451 * frameCount 0 452 * size 0 453 * raw undefined 454 * After successful return: 455 * frameCount actual number of frames available, <= number requested 456 * size actual number of bytes available 457 * raw pointer to the buffer 458 */ 459 460 enum { 461 NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 462 STOPPED = 1 463 }; 464 465 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); 466 467 /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */ 468 void releaseBuffer(Buffer* audioBuffer); 469 470 /* As a convenience we provide a write() interface to the audio buffer. 471 * This is implemented on top of obtainBuffer/releaseBuffer. For best 472 * performance use callbacks. Returns actual number of bytes written >= 0, 473 * or one of the following negative status codes: 474 * INVALID_OPERATION AudioTrack is configured for shared buffer mode 475 * BAD_VALUE size is invalid 476 * STOPPED AudioTrack was stopped during the write 477 * NO_MORE_BUFFERS when obtainBuffer() returns same 478 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 479 * Not supported for static buffer mode. 480 */ 481 ssize_t write(const void* buffer, size_t size); 482 483 /* 484 * Dumps the state of an audio track. 485 */ 486 status_t dump(int fd, const Vector<String16>& args) const; 487 488 protected: 489 /* copying audio tracks is not allowed */ 490 AudioTrack(const AudioTrack& other); 491 AudioTrack& operator = (const AudioTrack& other); 492 493 /* a small internal class to handle the callback */ 494 class AudioTrackThread : public Thread 495 { 496 public: 497 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 498 499 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 500 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 501 virtual void requestExit(); 502 503 void pause(); // suspend thread from execution at next loop boundary 504 void resume(); // allow thread to execute, if not requested to exit 505 506 private: 507 friend class AudioTrack; 508 virtual bool threadLoop(); 509 AudioTrack& mReceiver; 510 ~AudioTrackThread(); 511 Mutex mMyLock; // Thread::mLock is private 512 Condition mMyCond; // Thread::mThreadExitedCondition is private 513 bool mPaused; // whether thread is currently paused 514 }; 515 516 // body of AudioTrackThread::threadLoop() 517 bool processAudioBuffer(const sp<AudioTrackThread>& thread); 518 519 // caller must hold lock on mLock for all _l methods 520 status_t createTrack_l(audio_stream_type_t streamType, 521 uint32_t sampleRate, 522 audio_format_t format, 523 size_t frameCount, 524 audio_output_flags_t flags, 525 const sp<IMemory>& sharedBuffer, 526 audio_io_handle_t output); 527 528 // can only be called when !mActive 529 void flush_l(); 530 531 status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 532 audio_io_handle_t getOutput_l(); 533 status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart); 534 bool stopped_l() const { return !mActive; } 535 536 sp<IAudioTrack> mAudioTrack; 537 sp<IMemory> mCblkMemory; 538 sp<AudioTrackThread> mAudioTrackThread; 539 540 float mVolume[2]; 541 float mSendLevel; 542 uint32_t mSampleRate; 543 size_t mFrameCount; // corresponds to current IAudioTrack 544 size_t mReqFrameCount; // frame count to request the next time a new 545 // IAudioTrack is needed 546 547 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 548 549 // Starting address of buffers in shared memory. If there is a shared buffer, mBuffers 550 // is the value of pointer() for the shared buffer, otherwise mBuffers points 551 // immediately after the control block. This address is for the mapping within client 552 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 553 void* mBuffers; 554 555 audio_format_t mFormat; // as requested by client, not forced to 16-bit 556 audio_stream_type_t mStreamType; 557 uint32_t mChannelCount; 558 audio_channel_mask_t mChannelMask; 559 560 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. 561 // For 8-bit PCM data, mFrameSizeAF is 562 // twice as large because data is expanded to 16-bit before being stored in buffer. 563 size_t mFrameSize; // app-level frame size 564 size_t mFrameSizeAF; // AudioFlinger frame size 565 566 status_t mStatus; 567 uint32_t mLatency; 568 569 bool mActive; // protected by mLock 570 571 callback_t mCbf; // callback handler for events, or NULL 572 void* mUserData; // for client callback handler 573 574 // for notification APIs 575 uint32_t mNotificationFramesReq; // requested number of frames between each 576 // notification callback 577 uint32_t mNotificationFramesAct; // actual number of frames between each 578 // notification callback 579 sp<IMemory> mSharedBuffer; 580 int mLoopCount; 581 uint32_t mRemainingFrames; 582 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 583 bool mMarkerReached; 584 uint32_t mNewPosition; // in frames 585 uint32_t mUpdatePeriod; // in frames 586 587 bool mFlushed; // FIXME will be made obsolete by making flush() synchronous 588 audio_output_flags_t mFlags; 589 int mSessionId; 590 int mAuxEffectId; 591 592 // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock: 593 // 1. mLock 594 // 2. mCblk->lock 595 // It is OK to lock only mCblk->lock. 596 mutable Mutex mLock; 597 598 bool mIsTimed; 599 int mPreviousPriority; // before start() 600 SchedPolicy mPreviousSchedulingGroup; 601 AudioTrackClientProxy* mProxy; 602 bool mAwaitBoost; // thread should wait for priority boost before running 603 }; 604 605 class TimedAudioTrack : public AudioTrack 606 { 607 public: 608 TimedAudioTrack(); 609 610 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 611 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 612 613 /* queue a buffer obtained via allocateTimedBuffer for playback at the 614 given timestamp. PTS units are microseconds on the media time timeline. 615 The media time transform (set with setMediaTimeTransform) set by the 616 audio producer will handle converting from media time to local time 617 (perhaps going through the common time timeline in the case of 618 synchronized multiroom audio case) */ 619 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 620 621 /* define a transform between media time and either common time or 622 local time */ 623 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 624 status_t setMediaTimeTransform(const LinearTransform& xform, 625 TargetTimeline target); 626 }; 627 628 }; // namespace android 629 630 #endif // ANDROID_AUDIOTRACK_H 631
