AudioTrack.h revision 02de89293b74ab1e9a77ce2367c5c499ab038968
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 status_t 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 * Parameters: 335 * 336 * loopStart: loop start in frames relative to start of buffer. 337 * loopEnd: loop end in frames relative to start of buffer. 338 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 339 * pending or active loop. loopCount == -1 means infinite looping. 340 * 341 * For proper operation the following condition must be respected: 342 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 343 * 344 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 345 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 346 * 347 */ 348 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 349 350 /* Sets marker position. When playback reaches the number of frames specified, a callback with 351 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 352 * notification callback. To set a marker at a position which would compute as 0, 353 * a workaround is to the set the marker at a nearby position such as ~0 or 1. 354 * If the AudioTrack has been opened with no callback function associated, the operation will 355 * fail. 356 * 357 * Parameters: 358 * 359 * marker: marker position expressed in wrapping (overflow) frame units, 360 * like the return value of getPosition(). 361 * 362 * Returned status (from utils/Errors.h) can be: 363 * - NO_ERROR: successful operation 364 * - INVALID_OPERATION: the AudioTrack has no callback installed. 365 */ 366 status_t setMarkerPosition(uint32_t marker); 367 status_t getMarkerPosition(uint32_t *marker) const; 368 369 /* Sets position update period. Every time the number of frames specified has been played, 370 * a callback with event type EVENT_NEW_POS is called. 371 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 372 * callback. 373 * If the AudioTrack has been opened with no callback function associated, the operation will 374 * fail. 375 * Extremely small values may be rounded up to a value the implementation can support. 376 * 377 * Parameters: 378 * 379 * updatePeriod: position update notification period expressed in frames. 380 * 381 * Returned status (from utils/Errors.h) can be: 382 * - NO_ERROR: successful operation 383 * - INVALID_OPERATION: the AudioTrack has no callback installed. 384 */ 385 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 386 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 387 388 /* Sets playback head position. 389 * Only supported for static buffer mode. 390 * 391 * Parameters: 392 * 393 * position: New playback head position in frames relative to start of buffer. 394 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 395 * but will result in an immediate underrun if started. 396 * 397 * Returned status (from utils/Errors.h) can be: 398 * - NO_ERROR: successful operation 399 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 400 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 401 * buffer 402 */ 403 status_t setPosition(uint32_t position); 404 405 /* Return the total number of frames played since playback start. 406 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 407 * It is reset to zero by flush(), reload(), and stop(). 408 * 409 * Parameters: 410 * 411 * position: Address where to return play head position. 412 * 413 * Returned status (from utils/Errors.h) can be: 414 * - NO_ERROR: successful operation 415 * - BAD_VALUE: position is NULL 416 */ 417 status_t getPosition(uint32_t *position) const; 418 419 /* For static buffer mode only, this returns the current playback position in frames 420 * relative to start of buffer. It is analogous to the position units used by 421 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 422 */ 423 status_t getBufferPosition(uint32_t *position); 424 425 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 426 * rewriting the buffer before restarting playback after a stop. 427 * This method must be called with the AudioTrack in paused or stopped state. 428 * Not allowed in streaming mode. 429 * 430 * Returned status (from utils/Errors.h) can be: 431 * - NO_ERROR: successful operation 432 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 433 */ 434 status_t reload(); 435 436 /* Returns a handle on the audio output used by this AudioTrack. 437 * 438 * Parameters: 439 * none. 440 * 441 * Returned value: 442 * handle on audio hardware output 443 */ 444 audio_io_handle_t getOutput(); 445 446 /* Returns the unique session ID associated with this track. 447 * 448 * Parameters: 449 * none. 450 * 451 * Returned value: 452 * AudioTrack session ID. 453 */ 454 int getSessionId() const { return mSessionId; } 455 456 /* Attach track auxiliary output to specified effect. Use effectId = 0 457 * to detach track from effect. 458 * 459 * Parameters: 460 * 461 * effectId: effectId obtained from AudioEffect::id(). 462 * 463 * Returned status (from utils/Errors.h) can be: 464 * - NO_ERROR: successful operation 465 * - INVALID_OPERATION: the effect is not an auxiliary effect. 466 * - BAD_VALUE: The specified effect ID is invalid 467 */ 468 status_t attachAuxEffect(int effectId); 469 470 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 471 * After filling these slots with data, the caller should release them with releaseBuffer(). 472 * If the track buffer is not full, obtainBuffer() returns as many contiguous 473 * [empty slots for] frames as are available immediately. 474 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 475 * regardless of the value of waitCount. 476 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 477 * maximum timeout based on waitCount; see chart below. 478 * Buffers will be returned until the pool 479 * is exhausted, at which point obtainBuffer() will either block 480 * or return WOULD_BLOCK depending on the value of the "waitCount" 481 * parameter. 482 * Each sample is 16-bit signed PCM. 483 * 484 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 485 * which should use write() or callback EVENT_MORE_DATA instead. 486 * 487 * Interpretation of waitCount: 488 * +n limits wait time to n * WAIT_PERIOD_MS, 489 * -1 causes an (almost) infinite wait time, 490 * 0 non-blocking. 491 * 492 * Buffer fields 493 * On entry: 494 * frameCount number of frames requested 495 * After error return: 496 * frameCount 0 497 * size 0 498 * raw undefined 499 * After successful return: 500 * frameCount actual number of frames available, <= number requested 501 * size actual number of bytes available 502 * raw pointer to the buffer 503 */ 504 505 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 506 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 507 __attribute__((__deprecated__)); 508 509private: 510 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 511 * additional non-contiguous frames that are available immediately. 512 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 513 * in case the requested amount of frames is in two or more non-contiguous regions. 514 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 515 */ 516 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 517 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 518public: 519 520//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy 521// enum { 522// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 523// TEAR_DOWN = 0x80000002, 524// STOPPED = 1, 525// STREAM_END_WAIT, 526// STREAM_END 527// }; 528 529 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 530 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 531 void releaseBuffer(Buffer* audioBuffer); 532 533 /* As a convenience we provide a write() interface to the audio buffer. 534 * Input parameter 'size' is in byte units. 535 * This is implemented on top of obtainBuffer/releaseBuffer. For best 536 * performance use callbacks. Returns actual number of bytes written >= 0, 537 * or one of the following negative status codes: 538 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 539 * BAD_VALUE size is invalid 540 * WOULD_BLOCK when obtainBuffer() returns same, or 541 * AudioTrack was stopped during the write 542 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 543 */ 544 ssize_t write(const void* buffer, size_t size); 545 546 /* 547 * Dumps the state of an audio track. 548 */ 549 status_t dump(int fd, const Vector<String16>& args) const; 550 551 /* 552 * Return the total number of frames which AudioFlinger desired but were unavailable, 553 * and thus which resulted in an underrun. Reset to zero by stop(). 554 */ 555 uint32_t getUnderrunFrames() const; 556 557 /* Get the flags */ 558 audio_output_flags_t getFlags() const { return mFlags; } 559 560 /* Set parameters - only possible when using direct output */ 561 status_t setParameters(const String8& keyValuePairs); 562 563 /* Get parameters */ 564 String8 getParameters(const String8& keys); 565 566protected: 567 /* copying audio tracks is not allowed */ 568 AudioTrack(const AudioTrack& other); 569 AudioTrack& operator = (const AudioTrack& other); 570 571 /* a small internal class to handle the callback */ 572 class AudioTrackThread : public Thread 573 { 574 public: 575 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 576 577 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 578 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 579 virtual void requestExit(); 580 581 void pause(); // suspend thread from execution at next loop boundary 582 void resume(); // allow thread to execute, if not requested to exit 583 void pauseConditional(); 584 // like pause(), but only if prior resume() wasn't latched 585 586 private: 587 friend class AudioTrack; 588 virtual bool threadLoop(); 589 AudioTrack& mReceiver; 590 virtual ~AudioTrackThread(); 591 Mutex mMyLock; // Thread::mLock is private 592 Condition mMyCond; // Thread::mThreadExitedCondition is private 593 bool mPaused; // whether thread is currently paused 594 bool mResumeLatch; // whether next pauseConditional() will be a nop 595 }; 596 597 // body of AudioTrackThread::threadLoop() 598 // returns the maximum amount of time before we would like to run again, where: 599 // 0 immediately 600 // > 0 no later than this many nanoseconds from now 601 // NS_WHENEVER still active but no particular deadline 602 // NS_INACTIVE inactive so don't run again until re-started 603 // NS_NEVER never again 604 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 605 nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread); 606 status_t processStreamEnd(int32_t waitCount); 607 608 609 // caller must hold lock on mLock for all _l methods 610 611 status_t createTrack_l(audio_stream_type_t streamType, 612 uint32_t sampleRate, 613 audio_format_t format, 614 size_t frameCount, 615 audio_output_flags_t flags, 616 const sp<IMemory>& sharedBuffer, 617 audio_io_handle_t output, 618 size_t epoch); 619 620 // can only be called when mState != STATE_ACTIVE 621 void flush_l(); 622 623 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 624 audio_io_handle_t getOutput_l(); 625 626 // FIXME enum is faster than strcmp() for parameter 'from' 627 status_t restoreTrack_l(const char *from); 628 629 bool isOffloaded() const 630 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 631 632 // may be changed if IAudioTrack is re-created 633 sp<IAudioTrack> mAudioTrack; 634 sp<IMemory> mCblkMemory; 635 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 636 637 sp<AudioTrackThread> mAudioTrackThread; 638 float mVolume[2]; 639 float mSendLevel; 640 uint32_t mSampleRate; 641 size_t mFrameCount; // corresponds to current IAudioTrack 642 size_t mReqFrameCount; // frame count to request the next time a new 643 // IAudioTrack is needed 644 645 646 // constant after constructor or set() 647 audio_format_t mFormat; // as requested by client, not forced to 16-bit 648 audio_stream_type_t mStreamType; 649 uint32_t mChannelCount; 650 audio_channel_mask_t mChannelMask; 651 transfer_type mTransfer; 652 653 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 654 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 655 size_t mFrameSize; // app-level frame size 656 size_t mFrameSizeAF; // AudioFlinger frame size 657 658 status_t mStatus; 659 660 // can change dynamically when IAudioTrack invalidated 661 uint32_t mLatency; // in ms 662 663 // Indicates the current track state. Protected by mLock. 664 enum State { 665 STATE_ACTIVE, 666 STATE_STOPPED, 667 STATE_PAUSED, 668 STATE_PAUSED_STOPPING, 669 STATE_FLUSHED, 670 STATE_STOPPING, 671 } mState; 672 673 callback_t mCbf; // callback handler for events, or NULL 674 void* mUserData; // for client callback handler 675 676 // for notification APIs 677 uint32_t mNotificationFramesReq; // requested number of frames between each 678 // notification callback, 679 // at initial source sample rate 680 uint32_t mNotificationFramesAct; // actual number of frames between each 681 // notification callback, 682 // at initial source sample rate 683 bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 684 685 // These are private to processAudioBuffer(), and are not protected by a lock 686 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 687 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 688 uint32_t mObservedSequence; // last observed value of mSequence 689 690 sp<IMemory> mSharedBuffer; 691 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 692 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 693 bool mMarkerReached; 694 uint32_t mNewPosition; // in frames 695 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 696 697 audio_output_flags_t mFlags; 698 int mSessionId; 699 int mAuxEffectId; 700 701 mutable Mutex mLock; 702 703 bool mIsTimed; 704 int mPreviousPriority; // before start() 705 SchedPolicy mPreviousSchedulingGroup; 706 bool mAwaitBoost; // thread should wait for priority boost before running 707 708 // The proxy should only be referenced while a lock is held because the proxy isn't 709 // multi-thread safe, especially the SingleStateQueue part of the proxy. 710 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 711 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 712 // them around in case they are replaced during the obtainBuffer(). 713 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 714 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 715 716 bool mInUnderrun; // whether track is currently in underrun state 717 String8 mName; // server's name for this IAudioTrack 718 719private: 720 class DeathNotifier : public IBinder::DeathRecipient { 721 public: 722 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 723 protected: 724 virtual void binderDied(const wp<IBinder>& who); 725 private: 726 const wp<AudioTrack> mAudioTrack; 727 }; 728 729 sp<DeathNotifier> mDeathNotifier; 730 uint32_t mSequence; // incremented for each new IAudioTrack attempt 731 audio_io_handle_t mOutput; // cached output io handle 732}; 733 734class TimedAudioTrack : public AudioTrack 735{ 736public: 737 TimedAudioTrack(); 738 739 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 740 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 741 742 /* queue a buffer obtained via allocateTimedBuffer for playback at the 743 given timestamp. PTS units are microseconds on the media time timeline. 744 The media time transform (set with setMediaTimeTransform) set by the 745 audio producer will handle converting from media time to local time 746 (perhaps going through the common time timeline in the case of 747 synchronized multiroom audio case) */ 748 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 749 750 /* define a transform between media time and either common time or 751 local time */ 752 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 753 status_t setMediaTimeTransform(const LinearTransform& xform, 754 TargetTimeline target); 755}; 756 757}; // namespace android 758 759#endif // ANDROID_AUDIOTRACK_H 760