AudioTrack.h revision c5a17425986b4ce3384e6956762c86018b49c4a0
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 <cutils/sched_policy.h> 21#include <media/AudioSystem.h> 22#include <media/AudioTimestamp.h> 23#include <media/IAudioTrack.h> 24#include <utils/threads.h> 25 26namespace android { 27 28// ---------------------------------------------------------------------------- 29 30struct audio_track_cblk_t; 31class AudioTrackClientProxy; 32class StaticAudioTrackClientProxy; 33 34// ---------------------------------------------------------------------------- 35 36class AudioTrack : public RefBase 37{ 38public: 39 enum channel_index { 40 MONO = 0, 41 LEFT = 0, 42 RIGHT = 1 43 }; 44 45 /* Events used by AudioTrack callback function (callback_t). 46 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 47 */ 48 enum event_type { 49 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 50 // If this event is delivered but the callback handler 51 // does not want to write more data, the handler must explicitly 52 // ignore the event by setting frameCount to zero. 53 EVENT_UNDERRUN = 1, // Buffer underrun occurred. 54 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 55 // loop start if loop count was not 0. 56 EVENT_MARKER = 3, // Playback head is at the specified marker position 57 // (See setMarkerPosition()). 58 EVENT_NEW_POS = 4, // Playback head is at a new position 59 // (See setPositionUpdatePeriod()). 60 EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer. 61 // Not currently used by android.media.AudioTrack. 62 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 63 // voluntary invalidation by mediaserver, or mediaserver crash. 64 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 65 // back (after stop is called) 66 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 67 // in the mapping from frame position to presentation time. 68 // See AudioTimestamp for the information included with event. 69 }; 70 71 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 72 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 73 */ 74 75 class Buffer 76 { 77 public: 78 // FIXME use m prefix 79 size_t frameCount; // number of sample frames corresponding to size; 80 // on input it is the number of frames desired, 81 // on output is the number of frames actually filled 82 // (currently ignored, but will make the primary field in future) 83 84 size_t size; // input/output in bytes == frameCount * frameSize 85 // on output is the number of bytes actually filled 86 // FIXME this is redundant with respect to frameCount, 87 // and TRANSFER_OBTAIN mode is broken for 8-bit data 88 // since we don't define the frame format 89 90 union { 91 void* raw; 92 short* i16; // signed 16-bit 93 int8_t* i8; // unsigned 8-bit, offset by 0x80 94 }; 95 }; 96 97 /* As a convenience, if a callback is supplied, a handler thread 98 * is automatically created with the appropriate priority. This thread 99 * invokes the callback when a new buffer becomes available or various conditions occur. 100 * Parameters: 101 * 102 * event: type of event notified (see enum AudioTrack::event_type). 103 * user: Pointer to context for use by the callback receiver. 104 * info: Pointer to optional parameter according to event type: 105 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 106 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 107 * written. 108 * - EVENT_UNDERRUN: unused. 109 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 110 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 111 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 112 * - EVENT_BUFFER_END: unused. 113 * - EVENT_NEW_IAUDIOTRACK: unused. 114 * - EVENT_STREAM_END: unused. 115 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 116 */ 117 118 typedef void (*callback_t)(int event, void* user, void *info); 119 120 /* Returns the minimum frame count required for the successful creation of 121 * an AudioTrack object. 122 * Returned status (from utils/Errors.h) can be: 123 * - NO_ERROR: successful operation 124 * - NO_INIT: audio server or audio hardware not initialized 125 * - BAD_VALUE: unsupported configuration 126 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 127 * and is undefined otherwise. 128 */ 129 130 static status_t getMinFrameCount(size_t* frameCount, 131 audio_stream_type_t streamType, 132 uint32_t sampleRate); 133 134 /* How data is transferred to AudioTrack 135 */ 136 enum transfer_type { 137 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 138 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 139 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 140 TRANSFER_SYNC, // synchronous write() 141 TRANSFER_SHARED, // shared memory 142 }; 143 144 /* Constructs an uninitialized AudioTrack. No connection with 145 * AudioFlinger takes place. Use set() after this. 146 */ 147 AudioTrack(); 148 149 /* Creates an AudioTrack object and registers it with AudioFlinger. 150 * Once created, the track needs to be started before it can be used. 151 * Unspecified values are set to appropriate default values. 152 * With this constructor, the track is configured for streaming mode. 153 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 154 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 155 * 156 * Parameters: 157 * 158 * streamType: Select the type of audio stream this track is attached to 159 * (e.g. AUDIO_STREAM_MUSIC). 160 * sampleRate: Data source sampling rate in Hz. 161 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 162 * 16 bits per sample). 163 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 164 * frameCount: Minimum size of track PCM buffer in frames. This defines the 165 * application's contribution to the 166 * latency of the track. The actual size selected by the AudioTrack could be 167 * larger if the requested size is not compatible with current audio HAL 168 * configuration. Zero means to use a default value. 169 * flags: See comments on audio_output_flags_t in <system/audio.h>. 170 * cbf: Callback function. If not null, this function is called periodically 171 * to provide new data and inform of marker, position updates, etc. 172 * user: Context for use by the callback receiver. 173 * notificationFrames: The callback function is called each time notificationFrames PCM 174 * frames have been consumed from track input buffer. 175 * This is expressed in units of frames at the initial source sample rate. 176 * sessionId: Specific session ID, or zero to use default. 177 * transferType: How data is transferred to AudioTrack. 178 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 179 */ 180 181 AudioTrack( audio_stream_type_t streamType, 182 uint32_t sampleRate, 183 audio_format_t format, 184 audio_channel_mask_t, 185 size_t frameCount = 0, 186 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 187 callback_t cbf = NULL, 188 void* user = NULL, 189 uint32_t notificationFrames = 0, 190 int sessionId = AUDIO_SESSION_ALLOCATE, 191 transfer_type transferType = TRANSFER_DEFAULT, 192 const audio_offload_info_t *offloadInfo = NULL, 193 int uid = -1, 194 pid_t pid = -1); 195 196 /* Creates an audio track and registers it with AudioFlinger. 197 * With this constructor, the track is configured for static buffer mode. 198 * The format must not be 8-bit linear PCM. 199 * Data to be rendered is passed in a shared memory buffer 200 * identified by the argument sharedBuffer, which must be non-0. 201 * The memory should be initialized to the desired data before calling start(). 202 * The write() method is not supported in this case. 203 * It is recommended to pass a callback function to be notified of playback end by an 204 * EVENT_UNDERRUN event. 205 */ 206 207 AudioTrack( audio_stream_type_t streamType, 208 uint32_t sampleRate, 209 audio_format_t format, 210 audio_channel_mask_t channelMask, 211 const sp<IMemory>& sharedBuffer, 212 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 213 callback_t cbf = NULL, 214 void* user = NULL, 215 uint32_t notificationFrames = 0, 216 int sessionId = AUDIO_SESSION_ALLOCATE, 217 transfer_type transferType = TRANSFER_DEFAULT, 218 const audio_offload_info_t *offloadInfo = NULL, 219 int uid = -1, 220 pid_t pid = -1); 221 222 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 223 * Also destroys all resources associated with the AudioTrack. 224 */ 225protected: 226 virtual ~AudioTrack(); 227public: 228 229 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 230 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 231 * Returned status (from utils/Errors.h) can be: 232 * - NO_ERROR: successful initialization 233 * - INVALID_OPERATION: AudioTrack is already initialized 234 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 235 * - NO_INIT: audio server or audio hardware not initialized 236 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 237 * If sharedBuffer is non-0, the frameCount parameter is ignored and 238 * replaced by the shared buffer's total allocated size in frame units. 239 * 240 * Parameters not listed in the AudioTrack constructors above: 241 * 242 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 243 */ 244 status_t set(audio_stream_type_t streamType, 245 uint32_t sampleRate, 246 audio_format_t format, 247 audio_channel_mask_t channelMask, 248 size_t frameCount = 0, 249 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 250 callback_t cbf = NULL, 251 void* user = NULL, 252 uint32_t notificationFrames = 0, 253 const sp<IMemory>& sharedBuffer = 0, 254 bool threadCanCallJava = false, 255 int sessionId = AUDIO_SESSION_ALLOCATE, 256 transfer_type transferType = TRANSFER_DEFAULT, 257 const audio_offload_info_t *offloadInfo = NULL, 258 int uid = -1, 259 pid_t pid = -1); 260 261 /* Result of constructing the AudioTrack. This must be checked for successful initialization 262 * before using any AudioTrack API (except for set()), because using 263 * an uninitialized AudioTrack produces undefined results. 264 * See set() method above for possible return codes. 265 */ 266 status_t initCheck() const { return mStatus; } 267 268 /* Returns this track's estimated latency in milliseconds. 269 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 270 * and audio hardware driver. 271 */ 272 uint32_t latency() const { return mLatency; } 273 274 /* getters, see constructors and set() */ 275 276 audio_stream_type_t streamType() const { return mStreamType; } 277 audio_format_t format() const { return mFormat; } 278 279 /* Return frame size in bytes, which for linear PCM is 280 * channelCount * (bit depth per channel / 8). 281 * channelCount is determined from channelMask, and bit depth comes from format. 282 * For non-linear formats, the frame size is typically 1 byte. 283 */ 284 size_t frameSize() const { return mFrameSize; } 285 286 uint32_t channelCount() const { return mChannelCount; } 287 size_t frameCount() const { return mFrameCount; } 288 289 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 290 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 291 292 /* After it's created the track is not active. Call start() to 293 * make it active. If set, the callback will start being called. 294 * If the track was previously paused, volume is ramped up over the first mix buffer. 295 */ 296 status_t start(); 297 298 /* Stop a track. 299 * In static buffer mode, the track is stopped immediately. 300 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 301 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 302 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 303 * is first drained, mixed, and output, and only then is the track marked as stopped. 304 */ 305 void stop(); 306 bool stopped() const; 307 308 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 309 * This has the effect of draining the buffers without mixing or output. 310 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 311 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 312 */ 313 void flush(); 314 315 /* Pause a track. After pause, the callback will cease being called and 316 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 317 * and will fill up buffers until the pool is exhausted. 318 * Volume is ramped down over the next mix buffer following the pause request, 319 * and then the track is marked as paused. It can be resumed with ramp up by start(). 320 */ 321 void pause(); 322 323 /* Set volume for this track, mostly used for games' sound effects 324 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 325 * This is the older API. New applications should use setVolume(float) when possible. 326 */ 327 status_t setVolume(float left, float right); 328 329 /* Set volume for all channels. This is the preferred API for new applications, 330 * especially for multi-channel content. 331 */ 332 status_t setVolume(float volume); 333 334 /* Set the send level for this track. An auxiliary effect should be attached 335 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 336 */ 337 status_t setAuxEffectSendLevel(float level); 338 void getAuxEffectSendLevel(float* level) const; 339 340 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 341 */ 342 status_t setSampleRate(uint32_t sampleRate); 343 344 /* Return current source sample rate in Hz */ 345 uint32_t getSampleRate() const; 346 347 /* Enables looping and sets the start and end points of looping. 348 * Only supported for static buffer mode. 349 * 350 * Parameters: 351 * 352 * loopStart: loop start in frames relative to start of buffer. 353 * loopEnd: loop end in frames relative to start of buffer. 354 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 355 * pending or active loop. loopCount == -1 means infinite looping. 356 * 357 * For proper operation the following condition must be respected: 358 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 359 * 360 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 361 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 362 * 363 */ 364 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 365 366 /* Sets marker position. When playback reaches the number of frames specified, a callback with 367 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 368 * notification callback. To set a marker at a position which would compute as 0, 369 * a workaround is to set the marker at a nearby position such as ~0 or 1. 370 * If the AudioTrack has been opened with no callback function associated, the operation will 371 * fail. 372 * 373 * Parameters: 374 * 375 * marker: marker position expressed in wrapping (overflow) frame units, 376 * like the return value of getPosition(). 377 * 378 * Returned status (from utils/Errors.h) can be: 379 * - NO_ERROR: successful operation 380 * - INVALID_OPERATION: the AudioTrack has no callback installed. 381 */ 382 status_t setMarkerPosition(uint32_t marker); 383 status_t getMarkerPosition(uint32_t *marker) const; 384 385 /* Sets position update period. Every time the number of frames specified has been played, 386 * a callback with event type EVENT_NEW_POS is called. 387 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 388 * callback. 389 * If the AudioTrack has been opened with no callback function associated, the operation will 390 * fail. 391 * Extremely small values may be rounded up to a value the implementation can support. 392 * 393 * Parameters: 394 * 395 * updatePeriod: position update notification period expressed in frames. 396 * 397 * Returned status (from utils/Errors.h) can be: 398 * - NO_ERROR: successful operation 399 * - INVALID_OPERATION: the AudioTrack has no callback installed. 400 */ 401 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 402 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 403 404 /* Sets playback head position. 405 * Only supported for static buffer mode. 406 * 407 * Parameters: 408 * 409 * position: New playback head position in frames relative to start of buffer. 410 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 411 * but will result in an immediate underrun if started. 412 * 413 * Returned status (from utils/Errors.h) can be: 414 * - NO_ERROR: successful operation 415 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 416 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 417 * buffer 418 */ 419 status_t setPosition(uint32_t position); 420 421 /* Return the total number of frames played since playback start. 422 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 423 * It is reset to zero by flush(), reload(), and stop(). 424 * 425 * Parameters: 426 * 427 * position: Address where to return play head position. 428 * 429 * Returned status (from utils/Errors.h) can be: 430 * - NO_ERROR: successful operation 431 * - BAD_VALUE: position is NULL 432 */ 433 status_t getPosition(uint32_t *position) const; 434 435 /* For static buffer mode only, this returns the current playback position in frames 436 * relative to start of buffer. It is analogous to the position units used by 437 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 438 */ 439 status_t getBufferPosition(uint32_t *position); 440 441 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 442 * rewriting the buffer before restarting playback after a stop. 443 * This method must be called with the AudioTrack in paused or stopped state. 444 * Not allowed in streaming mode. 445 * 446 * Returned status (from utils/Errors.h) can be: 447 * - NO_ERROR: successful operation 448 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 449 */ 450 status_t reload(); 451 452 /* Returns a handle on the audio output used by this AudioTrack. 453 * 454 * Parameters: 455 * none. 456 * 457 * Returned value: 458 * handle on audio hardware output 459 */ 460 audio_io_handle_t getOutput() const; 461 462 /* Returns the unique session ID associated with this track. 463 * 464 * Parameters: 465 * none. 466 * 467 * Returned value: 468 * AudioTrack session ID. 469 */ 470 int getSessionId() const { return mSessionId; } 471 472 /* Attach track auxiliary output to specified effect. Use effectId = 0 473 * to detach track from effect. 474 * 475 * Parameters: 476 * 477 * effectId: effectId obtained from AudioEffect::id(). 478 * 479 * Returned status (from utils/Errors.h) can be: 480 * - NO_ERROR: successful operation 481 * - INVALID_OPERATION: the effect is not an auxiliary effect. 482 * - BAD_VALUE: The specified effect ID is invalid 483 */ 484 status_t attachAuxEffect(int effectId); 485 486 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 487 * After filling these slots with data, the caller should release them with releaseBuffer(). 488 * If the track buffer is not full, obtainBuffer() returns as many contiguous 489 * [empty slots for] frames as are available immediately. 490 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 491 * regardless of the value of waitCount. 492 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 493 * maximum timeout based on waitCount; see chart below. 494 * Buffers will be returned until the pool 495 * is exhausted, at which point obtainBuffer() will either block 496 * or return WOULD_BLOCK depending on the value of the "waitCount" 497 * parameter. 498 * Each sample is 16-bit signed PCM. 499 * 500 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 501 * which should use write() or callback EVENT_MORE_DATA instead. 502 * 503 * Interpretation of waitCount: 504 * +n limits wait time to n * WAIT_PERIOD_MS, 505 * -1 causes an (almost) infinite wait time, 506 * 0 non-blocking. 507 * 508 * Buffer fields 509 * On entry: 510 * frameCount number of frames requested 511 * After error return: 512 * frameCount 0 513 * size 0 514 * raw undefined 515 * After successful return: 516 * frameCount actual number of frames available, <= number requested 517 * size actual number of bytes available 518 * raw pointer to the buffer 519 */ 520 521 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 522 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 523 __attribute__((__deprecated__)); 524 525private: 526 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 527 * additional non-contiguous frames that are available immediately. 528 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 529 * in case the requested amount of frames is in two or more non-contiguous regions. 530 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 531 */ 532 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 533 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 534public: 535 536//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy 537// enum { 538// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 539// TEAR_DOWN = 0x80000002, 540// STOPPED = 1, 541// STREAM_END_WAIT, 542// STREAM_END 543// }; 544 545 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 546 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 547 void releaseBuffer(Buffer* audioBuffer); 548 549 /* As a convenience we provide a write() interface to the audio buffer. 550 * Input parameter 'size' is in byte units. 551 * This is implemented on top of obtainBuffer/releaseBuffer. For best 552 * performance use callbacks. Returns actual number of bytes written >= 0, 553 * or one of the following negative status codes: 554 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 555 * BAD_VALUE size is invalid 556 * WOULD_BLOCK when obtainBuffer() returns same, or 557 * AudioTrack was stopped during the write 558 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 559 * Default behavior is to only return until all data has been transferred. Set 'blocking' to 560 * false for the method to return immediately without waiting to try multiple times to write 561 * the full content of the buffer. 562 */ 563 ssize_t write(const void* buffer, size_t size, bool blocking = true); 564 565 /* 566 * Dumps the state of an audio track. 567 */ 568 status_t dump(int fd, const Vector<String16>& args) const; 569 570 /* 571 * Return the total number of frames which AudioFlinger desired but were unavailable, 572 * and thus which resulted in an underrun. Reset to zero by stop(). 573 */ 574 uint32_t getUnderrunFrames() const; 575 576 /* Get the flags */ 577 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 578 579 /* Set parameters - only possible when using direct output */ 580 status_t setParameters(const String8& keyValuePairs); 581 582 /* Get parameters */ 583 String8 getParameters(const String8& keys); 584 585 /* Poll for a timestamp on demand. 586 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 587 * or if you need to get the most recent timestamp outside of the event callback handler. 588 * Caution: calling this method too often may be inefficient; 589 * if you need a high resolution mapping between frame position and presentation time, 590 * consider implementing that at application level, based on the low resolution timestamps. 591 * Returns NO_ERROR if timestamp is valid. 592 */ 593 status_t getTimestamp(AudioTimestamp& timestamp); 594 595protected: 596 /* copying audio tracks is not allowed */ 597 AudioTrack(const AudioTrack& other); 598 AudioTrack& operator = (const AudioTrack& other); 599 600 /* a small internal class to handle the callback */ 601 class AudioTrackThread : public Thread 602 { 603 public: 604 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 605 606 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 607 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 608 virtual void requestExit(); 609 610 void pause(); // suspend thread from execution at next loop boundary 611 void resume(); // allow thread to execute, if not requested to exit 612 613 private: 614 void pauseInternal(nsecs_t ns = 0LL); 615 // like pause(), but only used internally within thread 616 617 friend class AudioTrack; 618 virtual bool threadLoop(); 619 AudioTrack& mReceiver; 620 virtual ~AudioTrackThread(); 621 Mutex mMyLock; // Thread::mLock is private 622 Condition mMyCond; // Thread::mThreadExitedCondition is private 623 bool mPaused; // whether thread is requested to pause at next loop entry 624 bool mPausedInt; // whether thread internally requests pause 625 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 626 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 627 }; 628 629 // body of AudioTrackThread::threadLoop() 630 // returns the maximum amount of time before we would like to run again, where: 631 // 0 immediately 632 // > 0 no later than this many nanoseconds from now 633 // NS_WHENEVER still active but no particular deadline 634 // NS_INACTIVE inactive so don't run again until re-started 635 // NS_NEVER never again 636 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 637 nsecs_t processAudioBuffer(); 638 639 bool isOffloaded() const; 640 641 // caller must hold lock on mLock for all _l methods 642 643 status_t createTrack_l(size_t epoch); 644 645 // can only be called when mState != STATE_ACTIVE 646 void flush_l(); 647 648 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 649 650 // FIXME enum is faster than strcmp() for parameter 'from' 651 status_t restoreTrack_l(const char *from); 652 653 bool isOffloaded_l() const 654 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 655 656 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 657 sp<IAudioTrack> mAudioTrack; 658 sp<IMemory> mCblkMemory; 659 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 660 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 661 662 sp<AudioTrackThread> mAudioTrackThread; 663 664 float mVolume[2]; 665 float mSendLevel; 666 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. 667 size_t mFrameCount; // corresponds to current IAudioTrack, value is 668 // reported back by AudioFlinger to the client 669 size_t mReqFrameCount; // frame count to request the first or next time 670 // a new IAudioTrack is needed, non-decreasing 671 672 // constant after constructor or set() 673 audio_format_t mFormat; // as requested by client, not forced to 16-bit 674 audio_stream_type_t mStreamType; 675 uint32_t mChannelCount; 676 audio_channel_mask_t mChannelMask; 677 sp<IMemory> mSharedBuffer; 678 transfer_type mTransfer; 679 audio_offload_info_t mOffloadInfoCopy; 680 const audio_offload_info_t* mOffloadInfo; 681 682 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 683 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 684 size_t mFrameSize; // app-level frame size 685 size_t mFrameSizeAF; // AudioFlinger frame size 686 687 status_t mStatus; 688 689 // can change dynamically when IAudioTrack invalidated 690 uint32_t mLatency; // in ms 691 692 // Indicates the current track state. Protected by mLock. 693 enum State { 694 STATE_ACTIVE, 695 STATE_STOPPED, 696 STATE_PAUSED, 697 STATE_PAUSED_STOPPING, 698 STATE_FLUSHED, 699 STATE_STOPPING, 700 } mState; 701 702 // for client callback handler 703 callback_t mCbf; // callback handler for events, or NULL 704 void* mUserData; 705 706 // for notification APIs 707 uint32_t mNotificationFramesReq; // requested number of frames between each 708 // notification callback, 709 // at initial source sample rate 710 uint32_t mNotificationFramesAct; // actual number of frames between each 711 // notification callback, 712 // at initial source sample rate 713 bool mRefreshRemaining; // processAudioBuffer() should refresh 714 // mRemainingFrames and mRetryOnPartialBuffer 715 716 // These are private to processAudioBuffer(), and are not protected by a lock 717 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 718 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 719 uint32_t mObservedSequence; // last observed value of mSequence 720 721 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 722 723 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 724 bool mMarkerReached; 725 uint32_t mNewPosition; // in frames 726 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 727 728 audio_output_flags_t mFlags; 729 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 730 // mLock must be held to read or write those bits reliably. 731 732 int mSessionId; 733 int mAuxEffectId; 734 735 mutable Mutex mLock; 736 737 bool mIsTimed; 738 int mPreviousPriority; // before start() 739 SchedPolicy mPreviousSchedulingGroup; 740 bool mAwaitBoost; // thread should wait for priority boost before running 741 742 // The proxy should only be referenced while a lock is held because the proxy isn't 743 // multi-thread safe, especially the SingleStateQueue part of the proxy. 744 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 745 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 746 // them around in case they are replaced during the obtainBuffer(). 747 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 748 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 749 750 bool mInUnderrun; // whether track is currently in underrun state 751 uint32_t mPausedPosition; 752 753private: 754 class DeathNotifier : public IBinder::DeathRecipient { 755 public: 756 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 757 protected: 758 virtual void binderDied(const wp<IBinder>& who); 759 private: 760 const wp<AudioTrack> mAudioTrack; 761 }; 762 763 sp<DeathNotifier> mDeathNotifier; 764 uint32_t mSequence; // incremented for each new IAudioTrack attempt 765 int mClientUid; 766 pid_t mClientPid; 767}; 768 769class TimedAudioTrack : public AudioTrack 770{ 771public: 772 TimedAudioTrack(); 773 774 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 775 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 776 777 /* queue a buffer obtained via allocateTimedBuffer for playback at the 778 given timestamp. PTS units are microseconds on the media time timeline. 779 The media time transform (set with setMediaTimeTransform) set by the 780 audio producer will handle converting from media time to local time 781 (perhaps going through the common time timeline in the case of 782 synchronized multiroom audio case) */ 783 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 784 785 /* define a transform between media time and either common time or 786 local time */ 787 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 788 status_t setMediaTimeTransform(const LinearTransform& xform, 789 TargetTimeline target); 790}; 791 792}; // namespace android 793 794#endif // ANDROID_AUDIOTRACK_H 795