AudioTrack.h revision 6dc365f0d5a92084517f0c3846e4f07fc7206bab
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 */ 127 128 static status_t getMinFrameCount(size_t* frameCount, 129 audio_stream_type_t streamType, 130 uint32_t sampleRate); 131 132 /* How data is transferred to AudioTrack 133 */ 134 enum transfer_type { 135 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 136 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 137 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 138 TRANSFER_SYNC, // synchronous write() 139 TRANSFER_SHARED, // shared memory 140 }; 141 142 /* Constructs an uninitialized AudioTrack. No connection with 143 * AudioFlinger takes place. Use set() after this. 144 */ 145 AudioTrack(); 146 147 /* Creates an AudioTrack object and registers it with AudioFlinger. 148 * Once created, the track needs to be started before it can be used. 149 * Unspecified values are set to appropriate default values. 150 * With this constructor, the track is configured for streaming mode. 151 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 152 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 153 * 154 * Parameters: 155 * 156 * streamType: Select the type of audio stream this track is attached to 157 * (e.g. AUDIO_STREAM_MUSIC). 158 * sampleRate: Data source sampling rate in Hz. 159 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 160 * 16 bits per sample). 161 * channelMask: Channel mask. 162 * frameCount: Minimum size of track PCM buffer in frames. This defines the 163 * application's contribution to the 164 * latency of the track. The actual size selected by the AudioTrack could be 165 * larger if the requested size is not compatible with current audio HAL 166 * configuration. Zero means to use a default value. 167 * flags: See comments on audio_output_flags_t in <system/audio.h>. 168 * cbf: Callback function. If not null, this function is called periodically 169 * to provide new data and inform of marker, position updates, etc. 170 * user: Context for use by the callback receiver. 171 * notificationFrames: The callback function is called each time notificationFrames PCM 172 * frames have been consumed from track input buffer. 173 * This is expressed in units of frames at the initial source sample rate. 174 * sessionId: Specific session ID, or zero to use default. 175 * transferType: How data is transferred to AudioTrack. 176 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 177 */ 178 179 AudioTrack( audio_stream_type_t streamType, 180 uint32_t sampleRate, 181 audio_format_t format, 182 audio_channel_mask_t, 183 int frameCount = 0, 184 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 185 callback_t cbf = NULL, 186 void* user = NULL, 187 int notificationFrames = 0, 188 int sessionId = 0, 189 transfer_type transferType = TRANSFER_DEFAULT, 190 const audio_offload_info_t *offloadInfo = NULL); 191 192 /* Creates an audio track and registers it with AudioFlinger. 193 * With this constructor, the track is configured for static buffer mode. 194 * The format must not be 8-bit linear PCM. 195 * Data to be rendered is passed in a shared memory buffer 196 * identified by the argument sharedBuffer, which must be non-0. 197 * The memory should be initialized to the desired data before calling start(). 198 * The write() method is not supported in this case. 199 * It is recommended to pass a callback function to be notified of playback end by an 200 * EVENT_UNDERRUN event. 201 */ 202 203 AudioTrack( audio_stream_type_t streamType, 204 uint32_t sampleRate, 205 audio_format_t format, 206 audio_channel_mask_t channelMask, 207 const sp<IMemory>& sharedBuffer, 208 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 209 callback_t cbf = NULL, 210 void* user = NULL, 211 int notificationFrames = 0, 212 int sessionId = 0, 213 transfer_type transferType = TRANSFER_DEFAULT, 214 const audio_offload_info_t *offloadInfo = NULL); 215 216 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 217 * Also destroys all resources associated with the AudioTrack. 218 */ 219protected: 220 virtual ~AudioTrack(); 221public: 222 223 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 224 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 225 * Returned status (from utils/Errors.h) can be: 226 * - NO_ERROR: successful initialization 227 * - INVALID_OPERATION: AudioTrack is already initialized 228 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 229 * - NO_INIT: audio server or audio hardware not initialized 230 * If sharedBuffer is non-0, the frameCount parameter is ignored and 231 * replaced by the shared buffer's total allocated size in frame units. 232 * 233 * Parameters not listed in the AudioTrack constructors above: 234 * 235 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 236 */ 237 status_t set(audio_stream_type_t streamType, 238 uint32_t sampleRate, 239 audio_format_t format, 240 audio_channel_mask_t channelMask, 241 int frameCount = 0, 242 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 243 callback_t cbf = NULL, 244 void* user = NULL, 245 int notificationFrames = 0, 246 const sp<IMemory>& sharedBuffer = 0, 247 bool threadCanCallJava = false, 248 int sessionId = 0, 249 transfer_type transferType = TRANSFER_DEFAULT, 250 const audio_offload_info_t *offloadInfo = NULL); 251 252 /* Result of constructing the AudioTrack. This must be checked 253 * before using any AudioTrack API (except for set()), because using 254 * an uninitialized AudioTrack produces undefined results. 255 * See set() method above for possible return codes. 256 */ 257 status_t initCheck() const { return mStatus; } 258 259 /* Returns this track's estimated latency in milliseconds. 260 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 261 * and audio hardware driver. 262 */ 263 uint32_t latency() const { return mLatency; } 264 265 /* getters, see constructors and set() */ 266 267 audio_stream_type_t streamType() const { return mStreamType; } 268 audio_format_t format() const { return mFormat; } 269 270 /* Return frame size in bytes, which for linear PCM is 271 * channelCount * (bit depth per channel / 8). 272 * channelCount is determined from channelMask, and bit depth comes from format. 273 * For non-linear formats, the frame size is typically 1 byte. 274 */ 275 size_t frameSize() const { return mFrameSize; } 276 277 uint32_t channelCount() const { return mChannelCount; } 278 uint32_t frameCount() const { return mFrameCount; } 279 280 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 281 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 282 283 /* After it's created the track is not active. Call start() to 284 * make it active. If set, the callback will start being called. 285 * If the track was previously paused, volume is ramped up over the first mix buffer. 286 */ 287 status_t start(); 288 289 /* Stop a track. 290 * In static buffer mode, the track is stopped immediately. 291 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 292 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 293 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 294 * is first drained, mixed, and output, and only then is the track marked as stopped. 295 */ 296 void stop(); 297 bool stopped() const; 298 299 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 300 * This has the effect of draining the buffers without mixing or output. 301 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 302 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 303 */ 304 void flush(); 305 306 /* Pause a track. After pause, the callback will cease being called and 307 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 308 * and will fill up buffers until the pool is exhausted. 309 * Volume is ramped down over the next mix buffer following the pause request, 310 * and then the track is marked as paused. It can be resumed with ramp up by start(). 311 */ 312 void pause(); 313 314 /* Set volume for this track, mostly used for games' sound effects 315 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 316 * This is the older API. New applications should use setVolume(float) when possible. 317 */ 318 status_t setVolume(float left, float right); 319 320 /* Set volume for all channels. This is the preferred API for new applications, 321 * especially for multi-channel content. 322 */ 323 status_t setVolume(float volume); 324 325 /* Set the send level for this track. An auxiliary effect should be attached 326 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 327 */ 328 status_t setAuxEffectSendLevel(float level); 329 void getAuxEffectSendLevel(float* level) const; 330 331 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 332 */ 333 status_t setSampleRate(uint32_t sampleRate); 334 335 /* Return current source sample rate in Hz, or 0 if unknown */ 336 uint32_t getSampleRate() const; 337 338 /* Enables looping and sets the start and end points of looping. 339 * Only supported for static buffer mode. 340 * 341 * Parameters: 342 * 343 * loopStart: loop start in frames relative to start of buffer. 344 * loopEnd: loop end in frames relative to start of buffer. 345 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 346 * pending or active loop. loopCount == -1 means infinite looping. 347 * 348 * For proper operation the following condition must be respected: 349 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 350 * 351 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 352 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 353 * 354 */ 355 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 356 357 /* Sets marker position. When playback reaches the number of frames specified, a callback with 358 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 359 * notification callback. To set a marker at a position which would compute as 0, 360 * a workaround is to the set the marker at a nearby position such as ~0 or 1. 361 * If the AudioTrack has been opened with no callback function associated, the operation will 362 * fail. 363 * 364 * Parameters: 365 * 366 * marker: marker position expressed in wrapping (overflow) frame units, 367 * like the return value of getPosition(). 368 * 369 * Returned status (from utils/Errors.h) can be: 370 * - NO_ERROR: successful operation 371 * - INVALID_OPERATION: the AudioTrack has no callback installed. 372 */ 373 status_t setMarkerPosition(uint32_t marker); 374 status_t getMarkerPosition(uint32_t *marker) const; 375 376 /* Sets position update period. Every time the number of frames specified has been played, 377 * a callback with event type EVENT_NEW_POS is called. 378 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 379 * callback. 380 * If the AudioTrack has been opened with no callback function associated, the operation will 381 * fail. 382 * Extremely small values may be rounded up to a value the implementation can support. 383 * 384 * Parameters: 385 * 386 * updatePeriod: position update notification period expressed in frames. 387 * 388 * Returned status (from utils/Errors.h) can be: 389 * - NO_ERROR: successful operation 390 * - INVALID_OPERATION: the AudioTrack has no callback installed. 391 */ 392 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 393 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 394 395 /* Sets playback head position. 396 * Only supported for static buffer mode. 397 * 398 * Parameters: 399 * 400 * position: New playback head position in frames relative to start of buffer. 401 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 402 * but will result in an immediate underrun if started. 403 * 404 * Returned status (from utils/Errors.h) can be: 405 * - NO_ERROR: successful operation 406 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 407 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 408 * buffer 409 */ 410 status_t setPosition(uint32_t position); 411 412 /* Return the total number of frames played since playback start. 413 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 414 * It is reset to zero by flush(), reload(), and stop(). 415 * 416 * Parameters: 417 * 418 * position: Address where to return play head position. 419 * 420 * Returned status (from utils/Errors.h) can be: 421 * - NO_ERROR: successful operation 422 * - BAD_VALUE: position is NULL 423 */ 424 status_t getPosition(uint32_t *position) const; 425 426 /* For static buffer mode only, this returns the current playback position in frames 427 * relative to start of buffer. It is analogous to the position units used by 428 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 429 */ 430 status_t getBufferPosition(uint32_t *position); 431 432 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 433 * rewriting the buffer before restarting playback after a stop. 434 * This method must be called with the AudioTrack in paused or stopped state. 435 * Not allowed in streaming mode. 436 * 437 * Returned status (from utils/Errors.h) can be: 438 * - NO_ERROR: successful operation 439 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 440 */ 441 status_t reload(); 442 443 /* Returns a handle on the audio output used by this AudioTrack. 444 * 445 * Parameters: 446 * none. 447 * 448 * Returned value: 449 * handle on audio hardware output 450 */ 451 audio_io_handle_t getOutput(); 452 453 /* Returns the unique session ID associated with this track. 454 * 455 * Parameters: 456 * none. 457 * 458 * Returned value: 459 * AudioTrack session ID. 460 */ 461 int getSessionId() const { return mSessionId; } 462 463 /* Attach track auxiliary output to specified effect. Use effectId = 0 464 * to detach track from effect. 465 * 466 * Parameters: 467 * 468 * effectId: effectId obtained from AudioEffect::id(). 469 * 470 * Returned status (from utils/Errors.h) can be: 471 * - NO_ERROR: successful operation 472 * - INVALID_OPERATION: the effect is not an auxiliary effect. 473 * - BAD_VALUE: The specified effect ID is invalid 474 */ 475 status_t attachAuxEffect(int effectId); 476 477 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 478 * After filling these slots with data, the caller should release them with releaseBuffer(). 479 * If the track buffer is not full, obtainBuffer() returns as many contiguous 480 * [empty slots for] frames as are available immediately. 481 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 482 * regardless of the value of waitCount. 483 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 484 * maximum timeout based on waitCount; see chart below. 485 * Buffers will be returned until the pool 486 * is exhausted, at which point obtainBuffer() will either block 487 * or return WOULD_BLOCK depending on the value of the "waitCount" 488 * parameter. 489 * Each sample is 16-bit signed PCM. 490 * 491 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 492 * which should use write() or callback EVENT_MORE_DATA instead. 493 * 494 * Interpretation of waitCount: 495 * +n limits wait time to n * WAIT_PERIOD_MS, 496 * -1 causes an (almost) infinite wait time, 497 * 0 non-blocking. 498 * 499 * Buffer fields 500 * On entry: 501 * frameCount number of frames requested 502 * After error return: 503 * frameCount 0 504 * size 0 505 * raw undefined 506 * After successful return: 507 * frameCount actual number of frames available, <= number requested 508 * size actual number of bytes available 509 * raw pointer to the buffer 510 */ 511 512 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 513 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 514 __attribute__((__deprecated__)); 515 516private: 517 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 518 * additional non-contiguous frames that are available immediately. 519 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 520 * in case the requested amount of frames is in two or more non-contiguous regions. 521 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 522 */ 523 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 524 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 525public: 526 527//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy 528// enum { 529// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 530// TEAR_DOWN = 0x80000002, 531// STOPPED = 1, 532// STREAM_END_WAIT, 533// STREAM_END 534// }; 535 536 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 537 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 538 void releaseBuffer(Buffer* audioBuffer); 539 540 /* As a convenience we provide a write() interface to the audio buffer. 541 * Input parameter 'size' is in byte units. 542 * This is implemented on top of obtainBuffer/releaseBuffer. For best 543 * performance use callbacks. Returns actual number of bytes written >= 0, 544 * or one of the following negative status codes: 545 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 546 * BAD_VALUE size is invalid 547 * WOULD_BLOCK when obtainBuffer() returns same, or 548 * AudioTrack was stopped during the write 549 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 550 */ 551 ssize_t write(const void* buffer, size_t size); 552 553 /* 554 * Dumps the state of an audio track. 555 */ 556 status_t dump(int fd, const Vector<String16>& args) const; 557 558 /* 559 * Return the total number of frames which AudioFlinger desired but were unavailable, 560 * and thus which resulted in an underrun. Reset to zero by stop(). 561 */ 562 uint32_t getUnderrunFrames() const; 563 564 /* Get the flags */ 565 audio_output_flags_t getFlags() const { return mFlags; } 566 567 /* Set parameters - only possible when using direct output */ 568 status_t setParameters(const String8& keyValuePairs); 569 570 /* Get parameters */ 571 String8 getParameters(const String8& keys); 572 573 /* Poll for a timestamp on demand. 574 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 575 * or if you need to get the most recent timestamp outside of the event callback handler. 576 * Caution: calling this method too often may be inefficient; 577 * if you need a high resolution mapping between frame position and presentation time, 578 * consider implementing that at application level, based on the low resolution timestamps. 579 * Returns NO_ERROR if timestamp is valid. 580 */ 581 status_t getTimestamp(AudioTimestamp& timestamp); 582 583protected: 584 /* copying audio tracks is not allowed */ 585 AudioTrack(const AudioTrack& other); 586 AudioTrack& operator = (const AudioTrack& other); 587 588 /* a small internal class to handle the callback */ 589 class AudioTrackThread : public Thread 590 { 591 public: 592 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 593 594 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 595 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 596 virtual void requestExit(); 597 598 void pause(); // suspend thread from execution at next loop boundary 599 void resume(); // allow thread to execute, if not requested to exit 600 void pauseConditional(); 601 // like pause(), but only if prior resume() wasn't latched 602 603 private: 604 friend class AudioTrack; 605 virtual bool threadLoop(); 606 AudioTrack& mReceiver; 607 virtual ~AudioTrackThread(); 608 Mutex mMyLock; // Thread::mLock is private 609 Condition mMyCond; // Thread::mThreadExitedCondition is private 610 bool mPaused; // whether thread is currently paused 611 bool mResumeLatch; // whether next pauseConditional() will be a nop 612 }; 613 614 // body of AudioTrackThread::threadLoop() 615 // returns the maximum amount of time before we would like to run again, where: 616 // 0 immediately 617 // > 0 no later than this many nanoseconds from now 618 // NS_WHENEVER still active but no particular deadline 619 // NS_INACTIVE inactive so don't run again until re-started 620 // NS_NEVER never again 621 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 622 nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread); 623 status_t processStreamEnd(int32_t waitCount); 624 625 626 // caller must hold lock on mLock for all _l methods 627 628 status_t createTrack_l(audio_stream_type_t streamType, 629 uint32_t sampleRate, 630 audio_format_t format, 631 size_t frameCount, 632 audio_output_flags_t flags, 633 const sp<IMemory>& sharedBuffer, 634 audio_io_handle_t output, 635 size_t epoch); 636 637 // can only be called when mState != STATE_ACTIVE 638 void flush_l(); 639 640 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 641 audio_io_handle_t getOutput_l(); 642 643 // FIXME enum is faster than strcmp() for parameter 'from' 644 status_t restoreTrack_l(const char *from); 645 646 bool isOffloaded() const 647 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 648 649 // may be changed if IAudioTrack is re-created 650 sp<IAudioTrack> mAudioTrack; 651 sp<IMemory> mCblkMemory; 652 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 653 654 sp<AudioTrackThread> mAudioTrackThread; 655 float mVolume[2]; 656 float mSendLevel; 657 uint32_t mSampleRate; 658 size_t mFrameCount; // corresponds to current IAudioTrack 659 size_t mReqFrameCount; // frame count to request the next time a new 660 // IAudioTrack is needed 661 662 // constant after constructor or set() 663 audio_format_t mFormat; // as requested by client, not forced to 16-bit 664 audio_stream_type_t mStreamType; 665 uint32_t mChannelCount; 666 audio_channel_mask_t mChannelMask; 667 transfer_type mTransfer; 668 669 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 670 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 671 size_t mFrameSize; // app-level frame size 672 size_t mFrameSizeAF; // AudioFlinger frame size 673 674 status_t mStatus; 675 676 // can change dynamically when IAudioTrack invalidated 677 uint32_t mLatency; // in ms 678 679 // Indicates the current track state. Protected by mLock. 680 enum State { 681 STATE_ACTIVE, 682 STATE_STOPPED, 683 STATE_PAUSED, 684 STATE_PAUSED_STOPPING, 685 STATE_FLUSHED, 686 STATE_STOPPING, 687 } mState; 688 689 // for client callback handler 690 callback_t mCbf; // callback handler for events, or NULL 691 void* mUserData; 692 693 // for notification APIs 694 uint32_t mNotificationFramesReq; // requested number of frames between each 695 // notification callback, 696 // at initial source sample rate 697 uint32_t mNotificationFramesAct; // actual number of frames between each 698 // notification callback, 699 // at initial source sample rate 700 bool mRefreshRemaining; // processAudioBuffer() should refresh 701 // mRemainingFrames and mRetryOnPartialBuffer 702 703 // These are private to processAudioBuffer(), and are not protected by a lock 704 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 705 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 706 uint32_t mObservedSequence; // last observed value of mSequence 707 708 sp<IMemory> mSharedBuffer; 709 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 710 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 711 bool mMarkerReached; 712 uint32_t mNewPosition; // in frames 713 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 714 715 audio_output_flags_t mFlags; 716 int mSessionId; 717 int mAuxEffectId; 718 719 mutable Mutex mLock; 720 721 bool mIsTimed; 722 int mPreviousPriority; // before start() 723 SchedPolicy mPreviousSchedulingGroup; 724 bool mAwaitBoost; // thread should wait for priority boost before running 725 726 // The proxy should only be referenced while a lock is held because the proxy isn't 727 // multi-thread safe, especially the SingleStateQueue part of the proxy. 728 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 729 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 730 // them around in case they are replaced during the obtainBuffer(). 731 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 732 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 733 734 bool mInUnderrun; // whether track is currently in underrun state 735 String8 mName; // server's name for this IAudioTrack 736 737private: 738 class DeathNotifier : public IBinder::DeathRecipient { 739 public: 740 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 741 protected: 742 virtual void binderDied(const wp<IBinder>& who); 743 private: 744 const wp<AudioTrack> mAudioTrack; 745 }; 746 747 sp<DeathNotifier> mDeathNotifier; 748 uint32_t mSequence; // incremented for each new IAudioTrack attempt 749 audio_io_handle_t mOutput; // cached output io handle 750}; 751 752class TimedAudioTrack : public AudioTrack 753{ 754public: 755 TimedAudioTrack(); 756 757 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 758 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 759 760 /* queue a buffer obtained via allocateTimedBuffer for playback at the 761 given timestamp. PTS units are microseconds on the media time timeline. 762 The media time transform (set with setMediaTimeTransform) set by the 763 audio producer will handle converting from media time to local time 764 (perhaps going through the common time timeline in the case of 765 synchronized multiroom audio case) */ 766 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 767 768 /* define a transform between media time and either common time or 769 local time */ 770 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 771 status_t setMediaTimeTransform(const LinearTransform& xform, 772 TargetTimeline target); 773}; 774 775}; // namespace android 776 777#endif // ANDROID_AUDIOTRACK_H 778