AudioTrack.h revision fb1fdc9d6603aa228362e7349451f6455c9849c2
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/IAudioTrack.h> 23#include <utils/threads.h> 24 25namespace android { 26 27// ---------------------------------------------------------------------------- 28 29class audio_track_cblk_t; 30class AudioTrackClientProxy; 31class StaticAudioTrackClientProxy; 32 33// ---------------------------------------------------------------------------- 34 35class AudioTrack : public RefBase 36{ 37public: 38 enum channel_index { 39 MONO = 0, 40 LEFT = 0, 41 RIGHT = 1 42 }; 43 44 /* Events used by AudioTrack callback function (callback_t). 45 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 46 */ 47 enum event_type { 48 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 49 // If this event is delivered but the callback handler 50 // does not want to write more data, the handler must explicitly 51 // ignore the event by setting frameCount to zero. 52 EVENT_UNDERRUN = 1, // Buffer underrun occurred. 53 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 54 // loop start if loop count was not 0. 55 EVENT_MARKER = 3, // Playback head is at the specified marker position 56 // (See setMarkerPosition()). 57 EVENT_NEW_POS = 4, // Playback head is at a new position 58 // (See setPositionUpdatePeriod()). 59 EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer. 60 // Not currently used by android.media.AudioTrack. 61 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 62 // voluntary invalidation by mediaserver, or mediaserver crash. 63 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 64 // back (after stop is called) 65 }; 66 67 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 68 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 69 */ 70 71 class Buffer 72 { 73 public: 74 // FIXME use m prefix 75 size_t frameCount; // number of sample frames corresponding to size; 76 // on input it is the number of frames desired, 77 // on output is the number of frames actually filled 78 // (currently ignored, but will make the primary field in future) 79 80 size_t size; // input/output in bytes == frameCount * frameSize 81 // on output is the number of bytes actually filled 82 // FIXME this is redundant with respect to frameCount, 83 // and TRANSFER_OBTAIN mode is broken for 8-bit data 84 // since we don't define the frame format 85 86 union { 87 void* raw; 88 short* i16; // signed 16-bit 89 int8_t* i8; // unsigned 8-bit, offset by 0x80 90 }; 91 }; 92 93 /* As a convenience, if a callback is supplied, a handler thread 94 * is automatically created with the appropriate priority. This thread 95 * invokes the callback when a new buffer becomes available or various conditions occur. 96 * Parameters: 97 * 98 * event: type of event notified (see enum AudioTrack::event_type). 99 * user: Pointer to context for use by the callback receiver. 100 * info: Pointer to optional parameter according to event type: 101 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 102 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 103 * written. 104 * - EVENT_UNDERRUN: unused. 105 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 106 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 107 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 108 * - EVENT_BUFFER_END: unused. 109 * - EVENT_NEW_IAUDIOTRACK: unused. 110 */ 111 112 typedef void (*callback_t)(int event, void* user, void *info); 113 114 /* Returns the minimum frame count required for the successful creation of 115 * an AudioTrack object. 116 * Returned status (from utils/Errors.h) can be: 117 * - NO_ERROR: successful operation 118 * - NO_INIT: audio server or audio hardware not initialized 119 */ 120 121 static status_t getMinFrameCount(size_t* frameCount, 122 audio_stream_type_t streamType, 123 uint32_t sampleRate); 124 125 /* How data is transferred to AudioTrack 126 */ 127 enum transfer_type { 128 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 129 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 130 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 131 TRANSFER_SYNC, // synchronous write() 132 TRANSFER_SHARED, // shared memory 133 }; 134 135 /* Constructs an uninitialized AudioTrack. No connection with 136 * AudioFlinger takes place. Use set() after this. 137 */ 138 AudioTrack(); 139 140 /* Creates an AudioTrack object and registers it with AudioFlinger. 141 * Once created, the track needs to be started before it can be used. 142 * Unspecified values are set to appropriate default values. 143 * With this constructor, the track is configured for streaming mode. 144 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 145 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 146 * 147 * Parameters: 148 * 149 * streamType: Select the type of audio stream this track is attached to 150 * (e.g. AUDIO_STREAM_MUSIC). 151 * sampleRate: Data source sampling rate in Hz. 152 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 153 * 16 bits per sample). 154 * channelMask: Channel mask. 155 * frameCount: Minimum size of track PCM buffer in frames. This defines the 156 * application's contribution to the 157 * latency of the track. The actual size selected by the AudioTrack could be 158 * larger if the requested size is not compatible with current audio HAL 159 * configuration. Zero means to use a default value. 160 * flags: See comments on audio_output_flags_t in <system/audio.h>. 161 * cbf: Callback function. If not null, this function is called periodically 162 * to provide new data and inform of marker, position updates, etc. 163 * user: Context for use by the callback receiver. 164 * notificationFrames: The callback function is called each time notificationFrames PCM 165 * frames have been consumed from track input buffer. 166 * This is expressed in units of frames at the initial source sample rate. 167 * sessionId: Specific session ID, or zero to use default. 168 * transferType: How data is transferred to AudioTrack. 169 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 170 */ 171 172 AudioTrack( audio_stream_type_t streamType, 173 uint32_t sampleRate = 0, 174 audio_format_t format = AUDIO_FORMAT_DEFAULT, 175 audio_channel_mask_t channelMask = 0, 176 int frameCount = 0, 177 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 178 callback_t cbf = NULL, 179 void* user = NULL, 180 int notificationFrames = 0, 181 int sessionId = 0, 182 transfer_type transferType = TRANSFER_DEFAULT, 183 const audio_offload_info_t *offloadInfo = NULL); 184 185 /* Creates an audio track and registers it with AudioFlinger. 186 * With this constructor, the track is configured for static buffer mode. 187 * The format must not be 8-bit linear PCM. 188 * Data to be rendered is passed in a shared memory buffer 189 * identified by the argument sharedBuffer, which must be non-0. 190 * The memory should be initialized to the desired data before calling start(). 191 * The write() method is not supported in this case. 192 * It is recommended to pass a callback function to be notified of playback end by an 193 * EVENT_UNDERRUN event. 194 */ 195 196 AudioTrack( audio_stream_type_t streamType, 197 uint32_t sampleRate = 0, 198 audio_format_t format = AUDIO_FORMAT_DEFAULT, 199 audio_channel_mask_t channelMask = 0, 200 const sp<IMemory>& sharedBuffer = 0, 201 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 202 callback_t cbf = NULL, 203 void* user = NULL, 204 int notificationFrames = 0, 205 int sessionId = 0, 206 transfer_type transferType = TRANSFER_DEFAULT, 207 const audio_offload_info_t *offloadInfo = NULL); 208 209 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 210 * Also destroys all resources associated with the AudioTrack. 211 */ 212protected: 213 virtual ~AudioTrack(); 214public: 215 216 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 217 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 218 * Returned status (from utils/Errors.h) can be: 219 * - NO_ERROR: successful initialization 220 * - INVALID_OPERATION: AudioTrack is already initialized 221 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 222 * - NO_INIT: audio server or audio hardware not initialized 223 * If sharedBuffer is non-0, the frameCount parameter is ignored and 224 * replaced by the shared buffer's total allocated size in frame units. 225 * 226 * Parameters not listed in the AudioTrack constructors above: 227 * 228 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 229 */ 230 status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, 231 uint32_t sampleRate = 0, 232 audio_format_t format = AUDIO_FORMAT_DEFAULT, 233 audio_channel_mask_t channelMask = 0, 234 int frameCount = 0, 235 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 236 callback_t cbf = NULL, 237 void* user = NULL, 238 int notificationFrames = 0, 239 const sp<IMemory>& sharedBuffer = 0, 240 bool threadCanCallJava = false, 241 int sessionId = 0, 242 transfer_type transferType = TRANSFER_DEFAULT, 243 const audio_offload_info_t *offloadInfo = NULL); 244 245 /* Result of constructing the AudioTrack. This must be checked 246 * before using any AudioTrack API (except for set()), because using 247 * an uninitialized AudioTrack produces undefined results. 248 * See set() method above for possible return codes. 249 */ 250 status_t initCheck() const { return mStatus; } 251 252 /* Returns this track's estimated latency in milliseconds. 253 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 254 * and audio hardware driver. 255 */ 256 uint32_t latency() const { return mLatency; } 257 258 /* getters, see constructors and set() */ 259 260 audio_stream_type_t streamType() const { return mStreamType; } 261 audio_format_t format() const { return mFormat; } 262 263 /* Return frame size in bytes, which for linear PCM is 264 * channelCount * (bit depth per channel / 8). 265 * channelCount is determined from channelMask, and bit depth comes from format. 266 * For non-linear formats, the frame size is typically 1 byte. 267 */ 268 size_t frameSize() const { return mFrameSize; } 269 270 uint32_t channelCount() const { return mChannelCount; } 271 uint32_t frameCount() const { return mFrameCount; } 272 273 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 274 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 275 276 /* After it's created the track is not active. Call start() to 277 * make it active. If set, the callback will start being called. 278 * If the track was previously paused, volume is ramped up over the first mix buffer. 279 */ 280 void start(); 281 282 /* Stop a track. 283 * In static buffer mode, the track is stopped immediately. 284 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 285 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 286 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 287 * is first drained, mixed, and output, and only then is the track marked as stopped. 288 */ 289 void stop(); 290 bool stopped() const; 291 292 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 293 * This has the effect of draining the buffers without mixing or output. 294 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 295 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 296 */ 297 void flush(); 298 299 /* Pause a track. After pause, the callback will cease being called and 300 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 301 * and will fill up buffers until the pool is exhausted. 302 * Volume is ramped down over the next mix buffer following the pause request, 303 * and then the track is marked as paused. It can be resumed with ramp up by start(). 304 */ 305 void pause(); 306 307 /* Set volume for this track, mostly used for games' sound effects 308 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 309 * This is the older API. New applications should use setVolume(float) when possible. 310 */ 311 status_t setVolume(float left, float right); 312 313 /* Set volume for all channels. This is the preferred API for new applications, 314 * especially for multi-channel content. 315 */ 316 status_t setVolume(float volume); 317 318 /* Set the send level for this track. An auxiliary effect should be attached 319 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 320 */ 321 status_t setAuxEffectSendLevel(float level); 322 void getAuxEffectSendLevel(float* level) const; 323 324 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 325 */ 326 status_t setSampleRate(uint32_t sampleRate); 327 328 /* Return current source sample rate in Hz, or 0 if unknown */ 329 uint32_t getSampleRate() const; 330 331 /* Enables looping and sets the start and end points of looping. 332 * Only supported for static buffer mode. 333 * 334 * FIXME The comments below are for the new planned interpretation which is not yet implemented. 335 * Currently the legacy behavior is still implemented, where loopStart and loopEnd 336 * are in wrapping (overflow) frame units like the return value of getPosition(). 337 * The plan is to fix all callers to use the new version at same time implementation changes. 338 * 339 * Parameters: 340 * 341 * loopStart: loop start in frames relative to start of buffer. 342 * loopEnd: loop end in frames relative to start of buffer. 343 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 344 * pending or active loop. loopCount == -1 means infinite looping. 345 * 346 * For proper operation the following condition must be respected: 347 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 348 * 349 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 350 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 351 * 352 */ 353 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 354 355 /* Sets marker position. When playback reaches the number of frames specified, a callback with 356 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 357 * notification callback. To set a marker at a position which would compute as 0, 358 * a workaround is to the set the marker at a nearby position such as ~0 or 1. 359 * If the AudioTrack has been opened with no callback function associated, the operation will 360 * fail. 361 * 362 * Parameters: 363 * 364 * marker: marker position expressed in wrapping (overflow) frame units, 365 * like the return value of getPosition(). 366 * 367 * Returned status (from utils/Errors.h) can be: 368 * - NO_ERROR: successful operation 369 * - INVALID_OPERATION: the AudioTrack has no callback installed. 370 */ 371 status_t setMarkerPosition(uint32_t marker); 372 status_t getMarkerPosition(uint32_t *marker) const; 373 374 /* Sets position update period. Every time the number of frames specified has been played, 375 * a callback with event type EVENT_NEW_POS is called. 376 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 377 * callback. 378 * If the AudioTrack has been opened with no callback function associated, the operation will 379 * fail. 380 * Extremely small values may be rounded up to a value the implementation can support. 381 * 382 * Parameters: 383 * 384 * updatePeriod: position update notification period expressed in frames. 385 * 386 * Returned status (from utils/Errors.h) can be: 387 * - NO_ERROR: successful operation 388 * - INVALID_OPERATION: the AudioTrack has no callback installed. 389 */ 390 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 391 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 392 393 /* Sets playback head position. 394 * Only supported for static buffer mode. 395 * 396 * FIXME The comments below are for the new planned interpretation which is not yet implemented. 397 * Currently the legacy behavior is still implemented, where the new position 398 * is in wrapping (overflow) frame units like the return value of getPosition(). 399 * The plan is to fix all callers to use the new version at same time implementation changes. 400 * 401 * Parameters: 402 * 403 * position: New playback head position in frames relative to start of buffer. 404 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 405 * but will result in an immediate underrun if started. 406 * 407 * Returned status (from utils/Errors.h) can be: 408 * - NO_ERROR: successful operation 409 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 410 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 411 * buffer 412 */ 413 status_t setPosition(uint32_t position); 414 415 /* Return the total number of frames played since playback start. 416 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 417 * It is reset to zero by flush(), reload(), and stop(). 418 * 419 * Parameters: 420 * 421 * position: Address where to return play head position. 422 * 423 * Returned status (from utils/Errors.h) can be: 424 * - NO_ERROR: successful operation 425 * - BAD_VALUE: position is NULL 426 */ 427 status_t getPosition(uint32_t *position) const; 428 429 /* For static buffer mode only, this returns the current playback position in frames 430 * relative to start of buffer. It is analogous to the new API for 431 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 432 */ 433 status_t getBufferPosition(uint32_t *position); 434 435 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 436 * rewriting the buffer before restarting playback after a stop. 437 * This method must be called with the AudioTrack in paused or stopped state. 438 * Not allowed in streaming mode. 439 * 440 * Returned status (from utils/Errors.h) can be: 441 * - NO_ERROR: successful operation 442 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 443 */ 444 status_t reload(); 445 446 /* Returns a handle on the audio output used by this AudioTrack. 447 * 448 * Parameters: 449 * none. 450 * 451 * Returned value: 452 * handle on audio hardware output 453 */ 454 audio_io_handle_t getOutput(); 455 456 /* Returns the unique session ID associated with this track. 457 * 458 * Parameters: 459 * none. 460 * 461 * Returned value: 462 * AudioTrack session ID. 463 */ 464 int getSessionId() const { return mSessionId; } 465 466 /* Attach track auxiliary output to specified effect. Use effectId = 0 467 * to detach track from effect. 468 * 469 * Parameters: 470 * 471 * effectId: effectId obtained from AudioEffect::id(). 472 * 473 * Returned status (from utils/Errors.h) can be: 474 * - NO_ERROR: successful operation 475 * - INVALID_OPERATION: the effect is not an auxiliary effect. 476 * - BAD_VALUE: The specified effect ID is invalid 477 */ 478 status_t attachAuxEffect(int effectId); 479 480 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 481 * After filling these slots with data, the caller should release them with releaseBuffer(). 482 * If the track buffer is not full, obtainBuffer() returns as many contiguous 483 * [empty slots for] frames as are available immediately. 484 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 485 * regardless of the value of waitCount. 486 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 487 * maximum timeout based on waitCount; see chart below. 488 * Buffers will be returned until the pool 489 * is exhausted, at which point obtainBuffer() will either block 490 * or return WOULD_BLOCK depending on the value of the "waitCount" 491 * parameter. 492 * Each sample is 16-bit signed PCM. 493 * 494 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 495 * which should use write() or callback EVENT_MORE_DATA instead. 496 * 497 * Interpretation of waitCount: 498 * +n limits wait time to n * WAIT_PERIOD_MS, 499 * -1 causes an (almost) infinite wait time, 500 * 0 non-blocking. 501 * 502 * Buffer fields 503 * On entry: 504 * frameCount number of frames requested 505 * After error return: 506 * frameCount 0 507 * size 0 508 * raw undefined 509 * After successful return: 510 * frameCount actual number of frames available, <= number requested 511 * size actual number of bytes available 512 * raw pointer to the buffer 513 */ 514 515 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 516 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 517 __attribute__((__deprecated__)); 518 519private: 520 /* New internal API 521 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 522 * additional non-contiguous frames that are available immediately. 523 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 524 * in case the requested amount of frames is in two or more non-contiguous regions. 525 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 526 */ 527 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 528 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 529public: 530 531//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy 532// enum { 533// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 534// TEAR_DOWN = 0x80000002, 535// STOPPED = 1, 536// STREAM_END_WAIT, 537// STREAM_END 538// }; 539 540 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 541 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 542 void releaseBuffer(Buffer* audioBuffer); 543 544 /* As a convenience we provide a write() interface to the audio buffer. 545 * Input parameter 'size' is in byte units. 546 * This is implemented on top of obtainBuffer/releaseBuffer. For best 547 * performance use callbacks. Returns actual number of bytes written >= 0, 548 * or one of the following negative status codes: 549 * INVALID_OPERATION AudioTrack is configured for shared buffer mode 550 * BAD_VALUE size is invalid 551 * WOULD_BLOCK when obtainBuffer() returns same, or 552 * AudioTrack was stopped during the write 553 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 554 * Not supported for static buffer mode. 555 */ 556 ssize_t write(const void* buffer, size_t size); 557 558 /* 559 * Dumps the state of an audio track. 560 */ 561 status_t dump(int fd, const Vector<String16>& args) const; 562 563 /* 564 * Return the total number of frames which AudioFlinger desired but were unavailable, 565 * and thus which resulted in an underrun. Reset to zero by stop(). 566 */ 567 uint32_t getUnderrunFrames() const; 568 569 /* Get the flags */ 570 audio_output_flags_t getFlags() const { return mFlags; } 571 572 /* Set parameters - only possible when using direct output */ 573 status_t setParameters(const String8& keyValuePairs); 574 575 /* Get parameters */ 576 String8 getParameters(const String8& keys); 577 578protected: 579 /* copying audio tracks is not allowed */ 580 AudioTrack(const AudioTrack& other); 581 AudioTrack& operator = (const AudioTrack& other); 582 583 /* a small internal class to handle the callback */ 584 class AudioTrackThread : public Thread 585 { 586 public: 587 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 588 589 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 590 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 591 virtual void requestExit(); 592 593 void pause(); // suspend thread from execution at next loop boundary 594 void resume(); // allow thread to execute, if not requested to exit 595 void pauseConditional(); 596 // like pause(), but only if prior resume() wasn't latched 597 598 private: 599 friend class AudioTrack; 600 virtual bool threadLoop(); 601 AudioTrack& mReceiver; 602 virtual ~AudioTrackThread(); 603 Mutex mMyLock; // Thread::mLock is private 604 Condition mMyCond; // Thread::mThreadExitedCondition is private 605 bool mPaused; // whether thread is currently paused 606 bool mResumeLatch; // whether next pauseConditional() will be a nop 607 }; 608 609 // body of AudioTrackThread::threadLoop() 610 // returns the maximum amount of time before we would like to run again, where: 611 // 0 immediately 612 // > 0 no later than this many nanoseconds from now 613 // NS_WHENEVER still active but no particular deadline 614 // NS_INACTIVE inactive so don't run again until re-started 615 // NS_NEVER never again 616 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 617 nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread); 618 status_t processStreamEnd(int32_t waitCount); 619 620 621 // caller must hold lock on mLock for all _l methods 622 623 status_t createTrack_l(audio_stream_type_t streamType, 624 uint32_t sampleRate, 625 audio_format_t format, 626 size_t frameCount, 627 audio_output_flags_t flags, 628 const sp<IMemory>& sharedBuffer, 629 audio_io_handle_t output, 630 size_t epoch); 631 632 // can only be called when mState != STATE_ACTIVE 633 void flush_l(); 634 635 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 636 audio_io_handle_t getOutput_l(); 637 638 status_t getPosition_l(uint32_t *position); 639 640 // FIXME enum is faster than strcmp() for parameter 'from' 641 status_t restoreTrack_l(const char *from); 642 643 // may be changed if IAudioTrack is re-created 644 sp<IAudioTrack> mAudioTrack; 645 sp<IMemory> mCblkMemory; 646 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 647 648 sp<AudioTrackThread> mAudioTrackThread; 649 float mVolume[2]; 650 float mSendLevel; 651 uint32_t mSampleRate; 652 size_t mFrameCount; // corresponds to current IAudioTrack 653 size_t mReqFrameCount; // frame count to request the next time a new 654 // IAudioTrack is needed 655 656 657 // constant after constructor or set() 658 audio_format_t mFormat; // as requested by client, not forced to 16-bit 659 audio_stream_type_t mStreamType; 660 uint32_t mChannelCount; 661 audio_channel_mask_t mChannelMask; 662 transfer_type mTransfer; 663 664 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 665 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 666 size_t mFrameSize; // app-level frame size 667 size_t mFrameSizeAF; // AudioFlinger frame size 668 669 status_t mStatus; 670 671 // can change dynamically when IAudioTrack invalidated 672 uint32_t mLatency; // in ms 673 674 // Indicates the current track state. Protected by mLock. 675 enum State { 676 STATE_ACTIVE, 677 STATE_STOPPED, 678 STATE_PAUSED, 679 STATE_FLUSHED, 680 } mState; 681 682 callback_t mCbf; // callback handler for events, or NULL 683 void* mUserData; // for client callback handler 684 685 // for notification APIs 686 uint32_t mNotificationFramesReq; // requested number of frames between each 687 // notification callback, 688 // at initial source sample rate 689 uint32_t mNotificationFramesAct; // actual number of frames between each 690 // notification callback, 691 // at initial source sample rate 692 bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 693 694 // These are private to processAudioBuffer(), and are not protected by a lock 695 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 696 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 697 int mObservedSequence; // last observed value of mSequence 698 699 sp<IMemory> mSharedBuffer; 700 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 701 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 702 bool mMarkerReached; 703 uint32_t mNewPosition; // in frames 704 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 705 706 audio_output_flags_t mFlags; 707 int mSessionId; 708 int mAuxEffectId; 709 710 mutable Mutex mLock; 711 712 bool mIsTimed; 713 int mPreviousPriority; // before start() 714 SchedPolicy mPreviousSchedulingGroup; 715 bool mAwaitBoost; // thread should wait for priority boost before running 716 717 // The proxy should only be referenced while a lock is held because the proxy isn't 718 // multi-thread safe, especially the SingleStateQueue part of the proxy. 719 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 720 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 721 // them around in case they are replaced during the obtainBuffer(). 722 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 723 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 724 725 bool mInUnderrun; // whether track is currently in underrun state 726 727private: 728 class DeathNotifier : public IBinder::DeathRecipient { 729 public: 730 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 731 protected: 732 virtual void binderDied(const wp<IBinder>& who); 733 private: 734 const wp<AudioTrack> mAudioTrack; 735 }; 736 737 sp<DeathNotifier> mDeathNotifier; 738 uint32_t mSequence; // incremented for each new IAudioTrack attempt 739}; 740 741class TimedAudioTrack : public AudioTrack 742{ 743public: 744 TimedAudioTrack(); 745 746 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 747 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 748 749 /* queue a buffer obtained via allocateTimedBuffer for playback at the 750 given timestamp. PTS units are microseconds on the media time timeline. 751 The media time transform (set with setMediaTimeTransform) set by the 752 audio producer will handle converting from media time to local time 753 (perhaps going through the common time timeline in the case of 754 synchronized multiroom audio case) */ 755 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 756 757 /* define a transform between media time and either common time or 758 local time */ 759 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 760 status_t setMediaTimeTransform(const LinearTransform& xform, 761 TargetTimeline target); 762}; 763 764}; // namespace android 765 766#endif // ANDROID_AUDIOTRACK_H 767