AudioTrack.h revision 9b26847ed122a601ba857e0ef22ee2c818641120
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 * - BAD_VALUE: unsupported configuration 120 */ 121 122 static status_t getMinFrameCount(size_t* frameCount, 123 audio_stream_type_t streamType, 124 uint32_t sampleRate); 125 126 /* How data is transferred to AudioTrack 127 */ 128 enum transfer_type { 129 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 130 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 131 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 132 TRANSFER_SYNC, // synchronous write() 133 TRANSFER_SHARED, // shared memory 134 }; 135 136 /* Constructs an uninitialized AudioTrack. No connection with 137 * AudioFlinger takes place. Use set() after this. 138 */ 139 AudioTrack(); 140 141 /* Creates an AudioTrack object and registers it with AudioFlinger. 142 * Once created, the track needs to be started before it can be used. 143 * Unspecified values are set to appropriate default values. 144 * With this constructor, the track is configured for streaming mode. 145 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 146 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 147 * 148 * Parameters: 149 * 150 * streamType: Select the type of audio stream this track is attached to 151 * (e.g. AUDIO_STREAM_MUSIC). 152 * sampleRate: Data source sampling rate in Hz. 153 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 154 * 16 bits per sample). 155 * channelMask: Channel mask. 156 * frameCount: Minimum size of track PCM buffer in frames. This defines the 157 * application's contribution to the 158 * latency of the track. The actual size selected by the AudioTrack could be 159 * larger if the requested size is not compatible with current audio HAL 160 * configuration. Zero means to use a default value. 161 * flags: See comments on audio_output_flags_t in <system/audio.h>. 162 * cbf: Callback function. If not null, this function is called periodically 163 * to provide new data and inform of marker, position updates, etc. 164 * user: Context for use by the callback receiver. 165 * notificationFrames: The callback function is called each time notificationFrames PCM 166 * frames have been consumed from track input buffer. 167 * This is expressed in units of frames at the initial source sample rate. 168 * sessionId: Specific session ID, or zero to use default. 169 * transferType: How data is transferred to AudioTrack. 170 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 171 */ 172 173 AudioTrack( audio_stream_type_t streamType, 174 uint32_t sampleRate, 175 audio_format_t format, 176 audio_channel_mask_t, 177 int frameCount = 0, 178 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 179 callback_t cbf = NULL, 180 void* user = NULL, 181 int notificationFrames = 0, 182 int sessionId = 0, 183 transfer_type transferType = TRANSFER_DEFAULT, 184 const audio_offload_info_t *offloadInfo = NULL); 185 186 /* Creates an audio track and registers it with AudioFlinger. 187 * With this constructor, the track is configured for static buffer mode. 188 * The format must not be 8-bit linear PCM. 189 * Data to be rendered is passed in a shared memory buffer 190 * identified by the argument sharedBuffer, which must be non-0. 191 * The memory should be initialized to the desired data before calling start(). 192 * The write() method is not supported in this case. 193 * It is recommended to pass a callback function to be notified of playback end by an 194 * EVENT_UNDERRUN event. 195 */ 196 197 AudioTrack( audio_stream_type_t streamType, 198 uint32_t sampleRate, 199 audio_format_t format, 200 audio_channel_mask_t channelMask, 201 const sp<IMemory>& sharedBuffer, 202 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 203 callback_t cbf = NULL, 204 void* user = NULL, 205 int notificationFrames = 0, 206 int sessionId = 0, 207 transfer_type transferType = TRANSFER_DEFAULT, 208 const audio_offload_info_t *offloadInfo = NULL); 209 210 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 211 * Also destroys all resources associated with the AudioTrack. 212 */ 213protected: 214 virtual ~AudioTrack(); 215public: 216 217 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 218 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 219 * Returned status (from utils/Errors.h) can be: 220 * - NO_ERROR: successful initialization 221 * - INVALID_OPERATION: AudioTrack is already initialized 222 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 223 * - NO_INIT: audio server or audio hardware not initialized 224 * If sharedBuffer is non-0, the frameCount parameter is ignored and 225 * replaced by the shared buffer's total allocated size in frame units. 226 * 227 * Parameters not listed in the AudioTrack constructors above: 228 * 229 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 230 */ 231 status_t set(audio_stream_type_t streamType, 232 uint32_t sampleRate, 233 audio_format_t format, 234 audio_channel_mask_t channelMask, 235 int frameCount = 0, 236 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 237 callback_t cbf = NULL, 238 void* user = NULL, 239 int notificationFrames = 0, 240 const sp<IMemory>& sharedBuffer = 0, 241 bool threadCanCallJava = false, 242 int sessionId = 0, 243 transfer_type transferType = TRANSFER_DEFAULT, 244 const audio_offload_info_t *offloadInfo = NULL); 245 246 /* Result of constructing the AudioTrack. This must be checked 247 * before using any AudioTrack API (except for set()), because using 248 * an uninitialized AudioTrack produces undefined results. 249 * See set() method above for possible return codes. 250 */ 251 status_t initCheck() const { return mStatus; } 252 253 /* Returns this track's estimated latency in milliseconds. 254 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 255 * and audio hardware driver. 256 */ 257 uint32_t latency() const { return mLatency; } 258 259 /* getters, see constructors and set() */ 260 261 audio_stream_type_t streamType() const { return mStreamType; } 262 audio_format_t format() const { return mFormat; } 263 264 /* Return frame size in bytes, which for linear PCM is 265 * channelCount * (bit depth per channel / 8). 266 * channelCount is determined from channelMask, and bit depth comes from format. 267 * For non-linear formats, the frame size is typically 1 byte. 268 */ 269 size_t frameSize() const { return mFrameSize; } 270 271 uint32_t channelCount() const { return mChannelCount; } 272 uint32_t frameCount() const { return mFrameCount; } 273 274 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 275 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 276 277 /* After it's created the track is not active. Call start() to 278 * make it active. If set, the callback will start being called. 279 * If the track was previously paused, volume is ramped up over the first mix buffer. 280 */ 281 status_t start(); 282 283 /* Stop a track. 284 * In static buffer mode, the track is stopped immediately. 285 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 286 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 287 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 288 * is first drained, mixed, and output, and only then is the track marked as stopped. 289 */ 290 void stop(); 291 bool stopped() const; 292 293 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 294 * This has the effect of draining the buffers without mixing or output. 295 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 296 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 297 */ 298 void flush(); 299 300 /* Pause a track. After pause, the callback will cease being called and 301 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 302 * and will fill up buffers until the pool is exhausted. 303 * Volume is ramped down over the next mix buffer following the pause request, 304 * and then the track is marked as paused. It can be resumed with ramp up by start(). 305 */ 306 void pause(); 307 308 /* Set volume for this track, mostly used for games' sound effects 309 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 310 * This is the older API. New applications should use setVolume(float) when possible. 311 */ 312 status_t setVolume(float left, float right); 313 314 /* Set volume for all channels. This is the preferred API for new applications, 315 * especially for multi-channel content. 316 */ 317 status_t setVolume(float volume); 318 319 /* Set the send level for this track. An auxiliary effect should be attached 320 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 321 */ 322 status_t setAuxEffectSendLevel(float level); 323 void getAuxEffectSendLevel(float* level) const; 324 325 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 326 */ 327 status_t setSampleRate(uint32_t sampleRate); 328 329 /* Return current source sample rate in Hz, or 0 if unknown */ 330 uint32_t getSampleRate() const; 331 332 /* Enables looping and sets the start and end points of looping. 333 * Only supported for static buffer mode. 334 * 335 * FIXME The comments below are for the new planned interpretation which is not yet implemented. 336 * Currently the legacy behavior is still implemented, where loopStart and loopEnd 337 * are in wrapping (overflow) frame units like the return value of getPosition(). 338 * The plan is to fix all callers to use the new version at same time implementation changes. 339 * 340 * Parameters: 341 * 342 * loopStart: loop start in frames relative to start of buffer. 343 * loopEnd: loop end in frames relative to start of buffer. 344 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 345 * pending or active loop. loopCount == -1 means infinite looping. 346 * 347 * For proper operation the following condition must be respected: 348 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 349 * 350 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 351 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 352 * 353 */ 354 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 355 356 /* Sets marker position. When playback reaches the number of frames specified, a callback with 357 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 358 * notification callback. To set a marker at a position which would compute as 0, 359 * a workaround is to the set the marker at a nearby position such as ~0 or 1. 360 * If the AudioTrack has been opened with no callback function associated, the operation will 361 * fail. 362 * 363 * Parameters: 364 * 365 * marker: marker position expressed in wrapping (overflow) frame units, 366 * like the return value of getPosition(). 367 * 368 * Returned status (from utils/Errors.h) can be: 369 * - NO_ERROR: successful operation 370 * - INVALID_OPERATION: the AudioTrack has no callback installed. 371 */ 372 status_t setMarkerPosition(uint32_t marker); 373 status_t getMarkerPosition(uint32_t *marker) const; 374 375 /* Sets position update period. Every time the number of frames specified has been played, 376 * a callback with event type EVENT_NEW_POS is called. 377 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 378 * callback. 379 * If the AudioTrack has been opened with no callback function associated, the operation will 380 * fail. 381 * Extremely small values may be rounded up to a value the implementation can support. 382 * 383 * Parameters: 384 * 385 * updatePeriod: position update notification period expressed in frames. 386 * 387 * Returned status (from utils/Errors.h) can be: 388 * - NO_ERROR: successful operation 389 * - INVALID_OPERATION: the AudioTrack has no callback installed. 390 */ 391 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 392 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 393 394 /* Sets playback head position. 395 * Only supported for static buffer mode. 396 * 397 * FIXME The comments below are for the new planned interpretation which is not yet implemented. 398 * Currently the legacy behavior is still implemented, where the new position 399 * is in wrapping (overflow) frame units like the return value of getPosition(). 400 * The plan is to fix all callers to use the new version at same time implementation changes. 401 * 402 * Parameters: 403 * 404 * position: New playback head position in frames relative to start of buffer. 405 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 406 * but will result in an immediate underrun if started. 407 * 408 * Returned status (from utils/Errors.h) can be: 409 * - NO_ERROR: successful operation 410 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 411 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 412 * buffer 413 */ 414 status_t setPosition(uint32_t position); 415 416 /* Return the total number of frames played since playback start. 417 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 418 * It is reset to zero by flush(), reload(), and stop(). 419 * 420 * Parameters: 421 * 422 * position: Address where to return play head position. 423 * 424 * Returned status (from utils/Errors.h) can be: 425 * - NO_ERROR: successful operation 426 * - BAD_VALUE: position is NULL 427 */ 428 status_t getPosition(uint32_t *position) const; 429 430 /* For static buffer mode only, this returns the current playback position in frames 431 * relative to start of buffer. It is analogous to the new API for 432 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 433 */ 434 status_t getBufferPosition(uint32_t *position); 435 436 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 437 * rewriting the buffer before restarting playback after a stop. 438 * This method must be called with the AudioTrack in paused or stopped state. 439 * Not allowed in streaming mode. 440 * 441 * Returned status (from utils/Errors.h) can be: 442 * - NO_ERROR: successful operation 443 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 444 */ 445 status_t reload(); 446 447 /* Returns a handle on the audio output used by this AudioTrack. 448 * 449 * Parameters: 450 * none. 451 * 452 * Returned value: 453 * handle on audio hardware output 454 */ 455 audio_io_handle_t getOutput(); 456 457 /* Returns the unique session ID associated with this track. 458 * 459 * Parameters: 460 * none. 461 * 462 * Returned value: 463 * AudioTrack session ID. 464 */ 465 int getSessionId() const { return mSessionId; } 466 467 /* Attach track auxiliary output to specified effect. Use effectId = 0 468 * to detach track from effect. 469 * 470 * Parameters: 471 * 472 * effectId: effectId obtained from AudioEffect::id(). 473 * 474 * Returned status (from utils/Errors.h) can be: 475 * - NO_ERROR: successful operation 476 * - INVALID_OPERATION: the effect is not an auxiliary effect. 477 * - BAD_VALUE: The specified effect ID is invalid 478 */ 479 status_t attachAuxEffect(int effectId); 480 481 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 482 * After filling these slots with data, the caller should release them with releaseBuffer(). 483 * If the track buffer is not full, obtainBuffer() returns as many contiguous 484 * [empty slots for] frames as are available immediately. 485 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 486 * regardless of the value of waitCount. 487 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 488 * maximum timeout based on waitCount; see chart below. 489 * Buffers will be returned until the pool 490 * is exhausted, at which point obtainBuffer() will either block 491 * or return WOULD_BLOCK depending on the value of the "waitCount" 492 * parameter. 493 * Each sample is 16-bit signed PCM. 494 * 495 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 496 * which should use write() or callback EVENT_MORE_DATA instead. 497 * 498 * Interpretation of waitCount: 499 * +n limits wait time to n * WAIT_PERIOD_MS, 500 * -1 causes an (almost) infinite wait time, 501 * 0 non-blocking. 502 * 503 * Buffer fields 504 * On entry: 505 * frameCount number of frames requested 506 * After error return: 507 * frameCount 0 508 * size 0 509 * raw undefined 510 * After successful return: 511 * frameCount actual number of frames available, <= number requested 512 * size actual number of bytes available 513 * raw pointer to the buffer 514 */ 515 516 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 517 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 518 __attribute__((__deprecated__)); 519 520private: 521 /* New internal API 522 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 523 * additional non-contiguous frames that are available immediately. 524 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 525 * in case the requested amount of frames is in two or more non-contiguous regions. 526 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 527 */ 528 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 529 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 530public: 531 532//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy 533// enum { 534// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 535// TEAR_DOWN = 0x80000002, 536// STOPPED = 1, 537// STREAM_END_WAIT, 538// STREAM_END 539// }; 540 541 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 542 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 543 void releaseBuffer(Buffer* audioBuffer); 544 545 /* As a convenience we provide a write() interface to the audio buffer. 546 * Input parameter 'size' is in byte units. 547 * This is implemented on top of obtainBuffer/releaseBuffer. For best 548 * performance use callbacks. Returns actual number of bytes written >= 0, 549 * or one of the following negative status codes: 550 * INVALID_OPERATION AudioTrack is configured for shared buffer mode 551 * BAD_VALUE size is invalid 552 * WOULD_BLOCK when obtainBuffer() returns same, or 553 * AudioTrack was stopped during the write 554 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 555 * Not supported for static buffer mode. 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 579protected: 580 /* copying audio tracks is not allowed */ 581 AudioTrack(const AudioTrack& other); 582 AudioTrack& operator = (const AudioTrack& other); 583 584 /* a small internal class to handle the callback */ 585 class AudioTrackThread : public Thread 586 { 587 public: 588 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 589 590 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 591 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 592 virtual void requestExit(); 593 594 void pause(); // suspend thread from execution at next loop boundary 595 void resume(); // allow thread to execute, if not requested to exit 596 void pauseConditional(); 597 // like pause(), but only if prior resume() wasn't latched 598 599 private: 600 friend class AudioTrack; 601 virtual bool threadLoop(); 602 AudioTrack& mReceiver; 603 virtual ~AudioTrackThread(); 604 Mutex mMyLock; // Thread::mLock is private 605 Condition mMyCond; // Thread::mThreadExitedCondition is private 606 bool mPaused; // whether thread is currently paused 607 bool mResumeLatch; // whether next pauseConditional() will be a nop 608 }; 609 610 // body of AudioTrackThread::threadLoop() 611 // returns the maximum amount of time before we would like to run again, where: 612 // 0 immediately 613 // > 0 no later than this many nanoseconds from now 614 // NS_WHENEVER still active but no particular deadline 615 // NS_INACTIVE inactive so don't run again until re-started 616 // NS_NEVER never again 617 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 618 nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread); 619 status_t processStreamEnd(int32_t waitCount); 620 621 622 // caller must hold lock on mLock for all _l methods 623 624 status_t createTrack_l(audio_stream_type_t streamType, 625 uint32_t sampleRate, 626 audio_format_t format, 627 size_t frameCount, 628 audio_output_flags_t flags, 629 const sp<IMemory>& sharedBuffer, 630 audio_io_handle_t output, 631 size_t epoch); 632 633 // can only be called when mState != STATE_ACTIVE 634 void flush_l(); 635 636 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 637 audio_io_handle_t getOutput_l(); 638 639 // FIXME enum is faster than strcmp() for parameter 'from' 640 status_t restoreTrack_l(const char *from); 641 642 bool isOffloaded() const 643 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 644 645 // may be changed if IAudioTrack is re-created 646 sp<IAudioTrack> mAudioTrack; 647 sp<IMemory> mCblkMemory; 648 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 649 650 sp<AudioTrackThread> mAudioTrackThread; 651 float mVolume[2]; 652 float mSendLevel; 653 uint32_t mSampleRate; 654 size_t mFrameCount; // corresponds to current IAudioTrack 655 size_t mReqFrameCount; // frame count to request the next time a new 656 // IAudioTrack is needed 657 658 659 // constant after constructor or set() 660 audio_format_t mFormat; // as requested by client, not forced to 16-bit 661 audio_stream_type_t mStreamType; 662 uint32_t mChannelCount; 663 audio_channel_mask_t mChannelMask; 664 transfer_type mTransfer; 665 666 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 667 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 668 size_t mFrameSize; // app-level frame size 669 size_t mFrameSizeAF; // AudioFlinger frame size 670 671 status_t mStatus; 672 673 // can change dynamically when IAudioTrack invalidated 674 uint32_t mLatency; // in ms 675 676 // Indicates the current track state. Protected by mLock. 677 enum State { 678 STATE_ACTIVE, 679 STATE_STOPPED, 680 STATE_PAUSED, 681 STATE_PAUSED_STOPPING, 682 STATE_FLUSHED, 683 STATE_STOPPING, 684 } mState; 685 686 // for client callback handler 687 callback_t mCbf; // callback handler for events, or NULL 688 void* mUserData; 689 690 // for notification APIs 691 uint32_t mNotificationFramesReq; // requested number of frames between each 692 // notification callback, 693 // at initial source sample rate 694 uint32_t mNotificationFramesAct; // actual number of frames between each 695 // notification callback, 696 // at initial source sample rate 697 bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 698 699 // These are private to processAudioBuffer(), and are not protected by a lock 700 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 701 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 702 uint32_t mObservedSequence; // last observed value of mSequence 703 704 sp<IMemory> mSharedBuffer; 705 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 706 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 707 bool mMarkerReached; 708 uint32_t mNewPosition; // in frames 709 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 710 711 audio_output_flags_t mFlags; 712 int mSessionId; 713 int mAuxEffectId; 714 715 mutable Mutex mLock; 716 717 bool mIsTimed; 718 int mPreviousPriority; // before start() 719 SchedPolicy mPreviousSchedulingGroup; 720 bool mAwaitBoost; // thread should wait for priority boost before running 721 722 // The proxy should only be referenced while a lock is held because the proxy isn't 723 // multi-thread safe, especially the SingleStateQueue part of the proxy. 724 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 725 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 726 // them around in case they are replaced during the obtainBuffer(). 727 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 728 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 729 730 bool mInUnderrun; // whether track is currently in underrun state 731 String8 mName; // server's name for this IAudioTrack 732 733private: 734 class DeathNotifier : public IBinder::DeathRecipient { 735 public: 736 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 737 protected: 738 virtual void binderDied(const wp<IBinder>& who); 739 private: 740 const wp<AudioTrack> mAudioTrack; 741 }; 742 743 sp<DeathNotifier> mDeathNotifier; 744 uint32_t mSequence; // incremented for each new IAudioTrack attempt 745 audio_io_handle_t mOutput; // cached output io handle 746}; 747 748class TimedAudioTrack : public AudioTrack 749{ 750public: 751 TimedAudioTrack(); 752 753 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 754 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 755 756 /* queue a buffer obtained via allocateTimedBuffer for playback at the 757 given timestamp. PTS units are microseconds on the media time timeline. 758 The media time transform (set with setMediaTimeTransform) set by the 759 audio producer will handle converting from media time to local time 760 (perhaps going through the common time timeline in the case of 761 synchronized multiroom audio case) */ 762 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 763 764 /* define a transform between media time and either common time or 765 local time */ 766 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 767 status_t setMediaTimeTransform(const LinearTransform& xform, 768 TargetTimeline target); 769}; 770 771}; // namespace android 772 773#endif // ANDROID_AUDIOTRACK_H 774