AudioTrack.h revision 396fabdb6efcdac5aea3d9f559d1beedf6a4cedc
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 30class 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. 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 int frameCount = 0, 186 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 187 callback_t cbf = NULL, 188 void* user = NULL, 189 int notificationFrames = 0, 190 int sessionId = 0, 191 transfer_type transferType = TRANSFER_DEFAULT, 192 const audio_offload_info_t *offloadInfo = NULL, 193 int uid = -1); 194 195 /* Creates an audio track and registers it with AudioFlinger. 196 * With this constructor, the track is configured for static buffer mode. 197 * The format must not be 8-bit linear PCM. 198 * Data to be rendered is passed in a shared memory buffer 199 * identified by the argument sharedBuffer, which must be non-0. 200 * The memory should be initialized to the desired data before calling start(). 201 * The write() method is not supported in this case. 202 * It is recommended to pass a callback function to be notified of playback end by an 203 * EVENT_UNDERRUN event. 204 */ 205 206 AudioTrack( audio_stream_type_t streamType, 207 uint32_t sampleRate, 208 audio_format_t format, 209 audio_channel_mask_t channelMask, 210 const sp<IMemory>& sharedBuffer, 211 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 212 callback_t cbf = NULL, 213 void* user = NULL, 214 int notificationFrames = 0, 215 int sessionId = 0, 216 transfer_type transferType = TRANSFER_DEFAULT, 217 const audio_offload_info_t *offloadInfo = NULL, 218 int uid = -1); 219 220 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 221 * Also destroys all resources associated with the AudioTrack. 222 */ 223protected: 224 virtual ~AudioTrack(); 225public: 226 227 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 228 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 229 * Returned status (from utils/Errors.h) can be: 230 * - NO_ERROR: successful initialization 231 * - INVALID_OPERATION: AudioTrack is already initialized 232 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 233 * - NO_INIT: audio server or audio hardware not initialized 234 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 235 * If sharedBuffer is non-0, the frameCount parameter is ignored and 236 * replaced by the shared buffer's total allocated size in frame units. 237 * 238 * Parameters not listed in the AudioTrack constructors above: 239 * 240 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 241 */ 242 status_t set(audio_stream_type_t streamType, 243 uint32_t sampleRate, 244 audio_format_t format, 245 audio_channel_mask_t channelMask, 246 int frameCount = 0, 247 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 248 callback_t cbf = NULL, 249 void* user = NULL, 250 int notificationFrames = 0, 251 const sp<IMemory>& sharedBuffer = 0, 252 bool threadCanCallJava = false, 253 int sessionId = 0, 254 transfer_type transferType = TRANSFER_DEFAULT, 255 const audio_offload_info_t *offloadInfo = NULL, 256 int uid = -1); 257 258 /* Result of constructing the AudioTrack. This must be checked for successful initialization 259 * before using any AudioTrack API (except for set()), because using 260 * an uninitialized AudioTrack produces undefined results. 261 * See set() method above for possible return codes. 262 */ 263 status_t initCheck() const { return mStatus; } 264 265 /* Returns this track's estimated latency in milliseconds. 266 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 267 * and audio hardware driver. 268 */ 269 uint32_t latency() const { return mLatency; } 270 271 /* getters, see constructors and set() */ 272 273 audio_stream_type_t streamType() const { return mStreamType; } 274 audio_format_t format() const { return mFormat; } 275 276 /* Return frame size in bytes, which for linear PCM is 277 * channelCount * (bit depth per channel / 8). 278 * channelCount is determined from channelMask, and bit depth comes from format. 279 * For non-linear formats, the frame size is typically 1 byte. 280 */ 281 size_t frameSize() const { return mFrameSize; } 282 283 uint32_t channelCount() const { return mChannelCount; } 284 uint32_t frameCount() const { return mFrameCount; } 285 286 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 287 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 288 289 /* After it's created the track is not active. Call start() to 290 * make it active. If set, the callback will start being called. 291 * If the track was previously paused, volume is ramped up over the first mix buffer. 292 */ 293 status_t start(); 294 295 /* Stop a track. 296 * In static buffer mode, the track is stopped immediately. 297 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 298 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 299 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 300 * is first drained, mixed, and output, and only then is the track marked as stopped. 301 */ 302 void stop(); 303 bool stopped() const; 304 305 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 306 * This has the effect of draining the buffers without mixing or output. 307 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 308 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 309 */ 310 void flush(); 311 312 /* Pause a track. After pause, the callback will cease being called and 313 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 314 * and will fill up buffers until the pool is exhausted. 315 * Volume is ramped down over the next mix buffer following the pause request, 316 * and then the track is marked as paused. It can be resumed with ramp up by start(). 317 */ 318 void pause(); 319 320 /* Set volume for this track, mostly used for games' sound effects 321 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 322 * This is the older API. New applications should use setVolume(float) when possible. 323 */ 324 status_t setVolume(float left, float right); 325 326 /* Set volume for all channels. This is the preferred API for new applications, 327 * especially for multi-channel content. 328 */ 329 status_t setVolume(float volume); 330 331 /* Set the send level for this track. An auxiliary effect should be attached 332 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 333 */ 334 status_t setAuxEffectSendLevel(float level); 335 void getAuxEffectSendLevel(float* level) const; 336 337 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 338 */ 339 status_t setSampleRate(uint32_t sampleRate); 340 341 /* Return current source sample rate in Hz, or 0 if unknown */ 342 uint32_t getSampleRate() const; 343 344 /* Enables looping and sets the start and end points of looping. 345 * Only supported for static buffer mode. 346 * 347 * Parameters: 348 * 349 * loopStart: loop start in frames relative to start of buffer. 350 * loopEnd: loop end in frames relative to start of buffer. 351 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 352 * pending or active loop. loopCount == -1 means infinite looping. 353 * 354 * For proper operation the following condition must be respected: 355 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 356 * 357 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 358 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 359 * 360 */ 361 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 362 363 /* Sets marker position. When playback reaches the number of frames specified, a callback with 364 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 365 * notification callback. To set a marker at a position which would compute as 0, 366 * a workaround is to the set the marker at a nearby position such as ~0 or 1. 367 * If the AudioTrack has been opened with no callback function associated, the operation will 368 * fail. 369 * 370 * Parameters: 371 * 372 * marker: marker position expressed in wrapping (overflow) frame units, 373 * like the return value of getPosition(). 374 * 375 * Returned status (from utils/Errors.h) can be: 376 * - NO_ERROR: successful operation 377 * - INVALID_OPERATION: the AudioTrack has no callback installed. 378 */ 379 status_t setMarkerPosition(uint32_t marker); 380 status_t getMarkerPosition(uint32_t *marker) const; 381 382 /* Sets position update period. Every time the number of frames specified has been played, 383 * a callback with event type EVENT_NEW_POS is called. 384 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 385 * callback. 386 * If the AudioTrack has been opened with no callback function associated, the operation will 387 * fail. 388 * Extremely small values may be rounded up to a value the implementation can support. 389 * 390 * Parameters: 391 * 392 * updatePeriod: position update notification period expressed in frames. 393 * 394 * Returned status (from utils/Errors.h) can be: 395 * - NO_ERROR: successful operation 396 * - INVALID_OPERATION: the AudioTrack has no callback installed. 397 */ 398 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 399 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 400 401 /* Sets playback head position. 402 * Only supported for static buffer mode. 403 * 404 * Parameters: 405 * 406 * position: New playback head position in frames relative to start of buffer. 407 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 408 * but will result in an immediate underrun if started. 409 * 410 * Returned status (from utils/Errors.h) can be: 411 * - NO_ERROR: successful operation 412 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 413 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 414 * buffer 415 */ 416 status_t setPosition(uint32_t position); 417 418 /* Return the total number of frames played since playback start. 419 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 420 * It is reset to zero by flush(), reload(), and stop(). 421 * 422 * Parameters: 423 * 424 * position: Address where to return play head position. 425 * 426 * Returned status (from utils/Errors.h) can be: 427 * - NO_ERROR: successful operation 428 * - BAD_VALUE: position is NULL 429 */ 430 status_t getPosition(uint32_t *position) const; 431 432 /* For static buffer mode only, this returns the current playback position in frames 433 * relative to start of buffer. It is analogous to the position units used by 434 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 435 */ 436 status_t getBufferPosition(uint32_t *position); 437 438 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 439 * rewriting the buffer before restarting playback after a stop. 440 * This method must be called with the AudioTrack in paused or stopped state. 441 * Not allowed in streaming mode. 442 * 443 * Returned status (from utils/Errors.h) can be: 444 * - NO_ERROR: successful operation 445 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 446 */ 447 status_t reload(); 448 449 /* Returns a handle on the audio output used by this AudioTrack. 450 * 451 * Parameters: 452 * none. 453 * 454 * Returned value: 455 * handle on audio hardware output 456 */ 457 audio_io_handle_t getOutput(); 458 459 /* Returns the unique session ID associated with this track. 460 * 461 * Parameters: 462 * none. 463 * 464 * Returned value: 465 * AudioTrack session ID. 466 */ 467 int getSessionId() const { return mSessionId; } 468 469 /* Attach track auxiliary output to specified effect. Use effectId = 0 470 * to detach track from effect. 471 * 472 * Parameters: 473 * 474 * effectId: effectId obtained from AudioEffect::id(). 475 * 476 * Returned status (from utils/Errors.h) can be: 477 * - NO_ERROR: successful operation 478 * - INVALID_OPERATION: the effect is not an auxiliary effect. 479 * - BAD_VALUE: The specified effect ID is invalid 480 */ 481 status_t attachAuxEffect(int effectId); 482 483 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 484 * After filling these slots with data, the caller should release them with releaseBuffer(). 485 * If the track buffer is not full, obtainBuffer() returns as many contiguous 486 * [empty slots for] frames as are available immediately. 487 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 488 * regardless of the value of waitCount. 489 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 490 * maximum timeout based on waitCount; see chart below. 491 * Buffers will be returned until the pool 492 * is exhausted, at which point obtainBuffer() will either block 493 * or return WOULD_BLOCK depending on the value of the "waitCount" 494 * parameter. 495 * Each sample is 16-bit signed PCM. 496 * 497 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 498 * which should use write() or callback EVENT_MORE_DATA instead. 499 * 500 * Interpretation of waitCount: 501 * +n limits wait time to n * WAIT_PERIOD_MS, 502 * -1 causes an (almost) infinite wait time, 503 * 0 non-blocking. 504 * 505 * Buffer fields 506 * On entry: 507 * frameCount number of frames requested 508 * After error return: 509 * frameCount 0 510 * size 0 511 * raw undefined 512 * After successful return: 513 * frameCount actual number of frames available, <= number requested 514 * size actual number of bytes available 515 * raw pointer to the buffer 516 */ 517 518 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 519 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 520 __attribute__((__deprecated__)); 521 522private: 523 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 524 * additional non-contiguous frames that are available immediately. 525 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 526 * in case the requested amount of frames is in two or more non-contiguous regions. 527 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 528 */ 529 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 530 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 531public: 532 533//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy 534// enum { 535// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 536// TEAR_DOWN = 0x80000002, 537// STOPPED = 1, 538// STREAM_END_WAIT, 539// STREAM_END 540// }; 541 542 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 543 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 544 void releaseBuffer(Buffer* audioBuffer); 545 546 /* As a convenience we provide a write() interface to the audio buffer. 547 * Input parameter 'size' is in byte units. 548 * This is implemented on top of obtainBuffer/releaseBuffer. For best 549 * performance use callbacks. Returns actual number of bytes written >= 0, 550 * or one of the following negative status codes: 551 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 552 * BAD_VALUE size is invalid 553 * WOULD_BLOCK when obtainBuffer() returns same, or 554 * AudioTrack was stopped during the write 555 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 556 */ 557 ssize_t write(const void* buffer, size_t size); 558 559 /* 560 * Dumps the state of an audio track. 561 */ 562 status_t dump(int fd, const Vector<String16>& args) const; 563 564 /* 565 * Return the total number of frames which AudioFlinger desired but were unavailable, 566 * and thus which resulted in an underrun. Reset to zero by stop(). 567 */ 568 uint32_t getUnderrunFrames() const; 569 570 /* Get the flags */ 571 audio_output_flags_t getFlags() const { return mFlags; } 572 573 /* Set parameters - only possible when using direct output */ 574 status_t setParameters(const String8& keyValuePairs); 575 576 /* Get parameters */ 577 String8 getParameters(const String8& keys); 578 579 /* Poll for a timestamp on demand. 580 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 581 * or if you need to get the most recent timestamp outside of the event callback handler. 582 * Caution: calling this method too often may be inefficient; 583 * if you need a high resolution mapping between frame position and presentation time, 584 * consider implementing that at application level, based on the low resolution timestamps. 585 * Returns NO_ERROR if timestamp is valid. 586 */ 587 status_t getTimestamp(AudioTimestamp& timestamp); 588 589protected: 590 /* copying audio tracks is not allowed */ 591 AudioTrack(const AudioTrack& other); 592 AudioTrack& operator = (const AudioTrack& other); 593 594 /* a small internal class to handle the callback */ 595 class AudioTrackThread : public Thread 596 { 597 public: 598 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 599 600 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 601 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 602 virtual void requestExit(); 603 604 void pause(); // suspend thread from execution at next loop boundary 605 void resume(); // allow thread to execute, if not requested to exit 606 607 private: 608 void pauseInternal(nsecs_t ns = 0LL); 609 // like pause(), but only used internally within thread 610 611 friend class AudioTrack; 612 virtual bool threadLoop(); 613 AudioTrack& mReceiver; 614 virtual ~AudioTrackThread(); 615 Mutex mMyLock; // Thread::mLock is private 616 Condition mMyCond; // Thread::mThreadExitedCondition is private 617 bool mPaused; // whether thread is requested to pause at next loop entry 618 bool mPausedInt; // whether thread internally requests pause 619 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 620 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 621 }; 622 623 // body of AudioTrackThread::threadLoop() 624 // returns the maximum amount of time before we would like to run again, where: 625 // 0 immediately 626 // > 0 no later than this many nanoseconds from now 627 // NS_WHENEVER still active but no particular deadline 628 // NS_INACTIVE inactive so don't run again until re-started 629 // NS_NEVER never again 630 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 631 nsecs_t processAudioBuffer(); 632 status_t processStreamEnd(int32_t waitCount); 633 634 635 // caller must hold lock on mLock for all _l methods 636 637 status_t createTrack_l(audio_stream_type_t streamType, 638 uint32_t sampleRate, 639 audio_format_t format, 640 size_t frameCount, 641 audio_output_flags_t flags, 642 const sp<IMemory>& sharedBuffer, 643 audio_io_handle_t output, 644 size_t epoch); 645 646 // can only be called when mState != STATE_ACTIVE 647 void flush_l(); 648 649 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 650 audio_io_handle_t getOutput_l(); 651 652 // FIXME enum is faster than strcmp() for parameter 'from' 653 status_t restoreTrack_l(const char *from); 654 655 bool isOffloaded() const 656 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 657 658 // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0 659 sp<IAudioTrack> mAudioTrack; 660 sp<IMemory> mCblkMemory; 661 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 662 663 sp<AudioTrackThread> mAudioTrackThread; 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 transfer_type mTransfer; 678 679 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 680 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 681 size_t mFrameSize; // app-level frame size 682 size_t mFrameSizeAF; // AudioFlinger frame size 683 684 status_t mStatus; 685 686 // can change dynamically when IAudioTrack invalidated 687 uint32_t mLatency; // in ms 688 689 // Indicates the current track state. Protected by mLock. 690 enum State { 691 STATE_ACTIVE, 692 STATE_STOPPED, 693 STATE_PAUSED, 694 STATE_PAUSED_STOPPING, 695 STATE_FLUSHED, 696 STATE_STOPPING, 697 } mState; 698 699 // for client callback handler 700 callback_t mCbf; // callback handler for events, or NULL 701 void* mUserData; 702 703 // for notification APIs 704 uint32_t mNotificationFramesReq; // requested number of frames between each 705 // notification callback, 706 // at initial source sample rate 707 uint32_t mNotificationFramesAct; // actual number of frames between each 708 // notification callback, 709 // at initial source sample rate 710 bool mRefreshRemaining; // processAudioBuffer() should refresh 711 // mRemainingFrames and mRetryOnPartialBuffer 712 713 // These are private to processAudioBuffer(), and are not protected by a lock 714 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 715 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 716 uint32_t mObservedSequence; // last observed value of mSequence 717 718 sp<IMemory> mSharedBuffer; 719 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 720 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 721 bool mMarkerReached; 722 uint32_t mNewPosition; // in frames 723 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 724 725 audio_output_flags_t mFlags; 726 int mSessionId; 727 int mAuxEffectId; 728 729 mutable Mutex mLock; 730 731 bool mIsTimed; 732 int mPreviousPriority; // before start() 733 SchedPolicy mPreviousSchedulingGroup; 734 bool mAwaitBoost; // thread should wait for priority boost before running 735 736 // The proxy should only be referenced while a lock is held because the proxy isn't 737 // multi-thread safe, especially the SingleStateQueue part of the proxy. 738 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 739 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 740 // them around in case they are replaced during the obtainBuffer(). 741 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 742 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 743 744 bool mInUnderrun; // whether track is currently in underrun state 745 String8 mName; // server's name for this IAudioTrack 746 747private: 748 class DeathNotifier : public IBinder::DeathRecipient { 749 public: 750 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 751 protected: 752 virtual void binderDied(const wp<IBinder>& who); 753 private: 754 const wp<AudioTrack> mAudioTrack; 755 }; 756 757 sp<DeathNotifier> mDeathNotifier; 758 uint32_t mSequence; // incremented for each new IAudioTrack attempt 759 audio_io_handle_t mOutput; // cached output io handle 760 int mClientUid; 761}; 762 763class TimedAudioTrack : public AudioTrack 764{ 765public: 766 TimedAudioTrack(); 767 768 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 769 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 770 771 /* queue a buffer obtained via allocateTimedBuffer for playback at the 772 given timestamp. PTS units are microseconds on the media time timeline. 773 The media time transform (set with setMediaTimeTransform) set by the 774 audio producer will handle converting from media time to local time 775 (perhaps going through the common time timeline in the case of 776 synchronized multiroom audio case) */ 777 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 778 779 /* define a transform between media time and either common time or 780 local time */ 781 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 782 status_t setMediaTimeTransform(const LinearTransform& xform, 783 TargetTimeline target); 784}; 785 786}; // namespace android 787 788#endif // ANDROID_AUDIOTRACK_H 789