1/* 2 * Copyright (C) 2007 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17#ifndef ANDROID_AUDIOTRACK_H 18#define ANDROID_AUDIOTRACK_H 19 20#include <cutils/sched_policy.h> 21#include <media/AudioSystem.h> 22#include <media/AudioTimestamp.h> 23#include <media/IAudioTrack.h> 24#include <utils/threads.h> 25 26namespace android { 27 28// ---------------------------------------------------------------------------- 29 30class audio_track_cblk_t; 31class AudioTrackClientProxy; 32class StaticAudioTrackClientProxy; 33 34// ---------------------------------------------------------------------------- 35 36class AudioTrack : public RefBase 37{ 38public: 39 enum channel_index { 40 MONO = 0, 41 LEFT = 0, 42 RIGHT = 1 43 }; 44 45 /* Events used by AudioTrack callback function (callback_t). 46 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 47 */ 48 enum event_type { 49 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 50 // If this event is delivered but the callback handler 51 // does not want to write more data, the handler must explicitly 52 // ignore the event by setting frameCount to zero. 53 EVENT_UNDERRUN = 1, // Buffer underrun occurred. 54 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 55 // loop start if loop count was not 0. 56 EVENT_MARKER = 3, // Playback head is at the specified marker position 57 // (See setMarkerPosition()). 58 EVENT_NEW_POS = 4, // Playback head is at a new position 59 // (See setPositionUpdatePeriod()). 60 EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer. 61 // Not currently used by android.media.AudioTrack. 62 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 63 // voluntary invalidation by mediaserver, or mediaserver crash. 64 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 65 // back (after stop is called) 66 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 67 // in the mapping from frame position to presentation time. 68 // See AudioTimestamp for the information included with event. 69 }; 70 71 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 72 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 73 */ 74 75 class Buffer 76 { 77 public: 78 // FIXME use m prefix 79 size_t frameCount; // number of sample frames corresponding to size; 80 // on input it is the number of frames desired, 81 // on output is the number of frames actually filled 82 // (currently ignored, but will make the primary field in future) 83 84 size_t size; // input/output in bytes == frameCount * frameSize 85 // on output is the number of bytes actually filled 86 // FIXME this is redundant with respect to frameCount, 87 // and TRANSFER_OBTAIN mode is broken for 8-bit data 88 // since we don't define the frame format 89 90 union { 91 void* raw; 92 short* i16; // signed 16-bit 93 int8_t* i8; // unsigned 8-bit, offset by 0x80 94 }; 95 }; 96 97 /* As a convenience, if a callback is supplied, a handler thread 98 * is automatically created with the appropriate priority. This thread 99 * invokes the callback when a new buffer becomes available or various conditions occur. 100 * Parameters: 101 * 102 * event: type of event notified (see enum AudioTrack::event_type). 103 * user: Pointer to context for use by the callback receiver. 104 * info: Pointer to optional parameter according to event type: 105 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 106 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 107 * written. 108 * - EVENT_UNDERRUN: unused. 109 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 110 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 111 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 112 * - EVENT_BUFFER_END: unused. 113 * - EVENT_NEW_IAUDIOTRACK: unused. 114 * - EVENT_STREAM_END: unused. 115 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 116 */ 117 118 typedef void (*callback_t)(int event, void* user, void *info); 119 120 /* Returns the minimum frame count required for the successful creation of 121 * an AudioTrack object. 122 * Returned status (from utils/Errors.h) can be: 123 * - NO_ERROR: successful operation 124 * - NO_INIT: audio server or audio hardware not initialized 125 * - BAD_VALUE: unsupported configuration 126 */ 127 128 static status_t getMinFrameCount(size_t* frameCount, 129 audio_stream_type_t streamType, 130 uint32_t sampleRate); 131 132 /* How data is transferred to AudioTrack 133 */ 134 enum transfer_type { 135 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 136 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 137 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 138 TRANSFER_SYNC, // synchronous write() 139 TRANSFER_SHARED, // shared memory 140 }; 141 142 /* Constructs an uninitialized AudioTrack. No connection with 143 * AudioFlinger takes place. Use set() after this. 144 */ 145 AudioTrack(); 146 147 /* Creates an AudioTrack object and registers it with AudioFlinger. 148 * Once created, the track needs to be started before it can be used. 149 * Unspecified values are set to appropriate default values. 150 * With this constructor, the track is configured for streaming mode. 151 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 152 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 153 * 154 * Parameters: 155 * 156 * streamType: Select the type of audio stream this track is attached to 157 * (e.g. AUDIO_STREAM_MUSIC). 158 * sampleRate: Data source sampling rate in Hz. 159 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 160 * 16 bits per sample). 161 * channelMask: Channel mask. 162 * frameCount: Minimum size of track PCM buffer in frames. This defines the 163 * application's contribution to the 164 * latency of the track. The actual size selected by the AudioTrack could be 165 * larger if the requested size is not compatible with current audio HAL 166 * configuration. Zero means to use a default value. 167 * flags: See comments on audio_output_flags_t in <system/audio.h>. 168 * cbf: Callback function. If not null, this function is called periodically 169 * to provide new data and inform of marker, position updates, etc. 170 * user: Context for use by the callback receiver. 171 * notificationFrames: The callback function is called each time notificationFrames PCM 172 * frames have been consumed from track input buffer. 173 * This is expressed in units of frames at the initial source sample rate. 174 * sessionId: Specific session ID, or zero to use default. 175 * transferType: How data is transferred to AudioTrack. 176 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 177 */ 178 179 AudioTrack( audio_stream_type_t streamType, 180 uint32_t sampleRate, 181 audio_format_t format, 182 audio_channel_mask_t, 183 int frameCount = 0, 184 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 185 callback_t cbf = NULL, 186 void* user = NULL, 187 int notificationFrames = 0, 188 int sessionId = 0, 189 transfer_type transferType = TRANSFER_DEFAULT, 190 const audio_offload_info_t *offloadInfo = NULL); 191 192 /* Creates an audio track and registers it with AudioFlinger. 193 * With this constructor, the track is configured for static buffer mode. 194 * The format must not be 8-bit linear PCM. 195 * Data to be rendered is passed in a shared memory buffer 196 * identified by the argument sharedBuffer, which must be non-0. 197 * The memory should be initialized to the desired data before calling start(). 198 * The write() method is not supported in this case. 199 * It is recommended to pass a callback function to be notified of playback end by an 200 * EVENT_UNDERRUN event. 201 */ 202 203 AudioTrack( audio_stream_type_t streamType, 204 uint32_t sampleRate, 205 audio_format_t format, 206 audio_channel_mask_t channelMask, 207 const sp<IMemory>& sharedBuffer, 208 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 209 callback_t cbf = NULL, 210 void* user = NULL, 211 int notificationFrames = 0, 212 int sessionId = 0, 213 transfer_type transferType = TRANSFER_DEFAULT, 214 const audio_offload_info_t *offloadInfo = NULL); 215 216 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 217 * Also destroys all resources associated with the AudioTrack. 218 */ 219protected: 220 virtual ~AudioTrack(); 221public: 222 223 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 224 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 225 * Returned status (from utils/Errors.h) can be: 226 * - NO_ERROR: successful initialization 227 * - INVALID_OPERATION: AudioTrack is already initialized 228 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 229 * - NO_INIT: audio server or audio hardware not initialized 230 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 231 * If sharedBuffer is non-0, the frameCount parameter is ignored and 232 * replaced by the shared buffer's total allocated size in frame units. 233 * 234 * Parameters not listed in the AudioTrack constructors above: 235 * 236 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 237 */ 238 status_t set(audio_stream_type_t streamType, 239 uint32_t sampleRate, 240 audio_format_t format, 241 audio_channel_mask_t channelMask, 242 int frameCount = 0, 243 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 244 callback_t cbf = NULL, 245 void* user = NULL, 246 int notificationFrames = 0, 247 const sp<IMemory>& sharedBuffer = 0, 248 bool threadCanCallJava = false, 249 int sessionId = 0, 250 transfer_type transferType = TRANSFER_DEFAULT, 251 const audio_offload_info_t *offloadInfo = NULL); 252 253 /* Result of constructing the AudioTrack. This must be checked for successful initialization 254 * before using any AudioTrack API (except for set()), because using 255 * an uninitialized AudioTrack produces undefined results. 256 * See set() method above for possible return codes. 257 */ 258 status_t initCheck() const { return mStatus; } 259 260 /* Returns this track's estimated latency in milliseconds. 261 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 262 * and audio hardware driver. 263 */ 264 uint32_t latency() const { return mLatency; } 265 266 /* getters, see constructors and set() */ 267 268 audio_stream_type_t streamType() const { return mStreamType; } 269 audio_format_t format() const { return mFormat; } 270 271 /* Return frame size in bytes, which for linear PCM is 272 * channelCount * (bit depth per channel / 8). 273 * channelCount is determined from channelMask, and bit depth comes from format. 274 * For non-linear formats, the frame size is typically 1 byte. 275 */ 276 size_t frameSize() const { return mFrameSize; } 277 278 uint32_t channelCount() const { return mChannelCount; } 279 uint32_t frameCount() const { return mFrameCount; } 280 281 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 282 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 283 284 /* After it's created the track is not active. Call start() to 285 * make it active. If set, the callback will start being called. 286 * If the track was previously paused, volume is ramped up over the first mix buffer. 287 */ 288 status_t start(); 289 290 /* Stop a track. 291 * In static buffer mode, the track is stopped immediately. 292 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 293 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 294 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 295 * is first drained, mixed, and output, and only then is the track marked as stopped. 296 */ 297 void stop(); 298 bool stopped() const; 299 300 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 301 * This has the effect of draining the buffers without mixing or output. 302 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 303 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 304 */ 305 void flush(); 306 307 /* Pause a track. After pause, the callback will cease being called and 308 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 309 * and will fill up buffers until the pool is exhausted. 310 * Volume is ramped down over the next mix buffer following the pause request, 311 * and then the track is marked as paused. It can be resumed with ramp up by start(). 312 */ 313 void pause(); 314 315 /* Set volume for this track, mostly used for games' sound effects 316 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 317 * This is the older API. New applications should use setVolume(float) when possible. 318 */ 319 status_t setVolume(float left, float right); 320 321 /* Set volume for all channels. This is the preferred API for new applications, 322 * especially for multi-channel content. 323 */ 324 status_t setVolume(float volume); 325 326 /* Set the send level for this track. An auxiliary effect should be attached 327 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 328 */ 329 status_t setAuxEffectSendLevel(float level); 330 void getAuxEffectSendLevel(float* level) const; 331 332 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 333 */ 334 status_t setSampleRate(uint32_t sampleRate); 335 336 /* Return current source sample rate in Hz, or 0 if unknown */ 337 uint32_t getSampleRate() const; 338 339 /* Enables looping and sets the start and end points of looping. 340 * Only supported for static buffer mode. 341 * 342 * Parameters: 343 * 344 * loopStart: loop start in frames relative to start of buffer. 345 * loopEnd: loop end in frames relative to start of buffer. 346 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 347 * pending or active loop. loopCount == -1 means infinite looping. 348 * 349 * For proper operation the following condition must be respected: 350 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 351 * 352 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 353 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 354 * 355 */ 356 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 357 358 /* Sets marker position. When playback reaches the number of frames specified, a callback with 359 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 360 * notification callback. To set a marker at a position which would compute as 0, 361 * a workaround is to the set the marker at a nearby position such as ~0 or 1. 362 * If the AudioTrack has been opened with no callback function associated, the operation will 363 * fail. 364 * 365 * Parameters: 366 * 367 * marker: marker position expressed in wrapping (overflow) frame units, 368 * like the return value of getPosition(). 369 * 370 * Returned status (from utils/Errors.h) can be: 371 * - NO_ERROR: successful operation 372 * - INVALID_OPERATION: the AudioTrack has no callback installed. 373 */ 374 status_t setMarkerPosition(uint32_t marker); 375 status_t getMarkerPosition(uint32_t *marker) const; 376 377 /* Sets position update period. Every time the number of frames specified has been played, 378 * a callback with event type EVENT_NEW_POS is called. 379 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 380 * callback. 381 * If the AudioTrack has been opened with no callback function associated, the operation will 382 * fail. 383 * Extremely small values may be rounded up to a value the implementation can support. 384 * 385 * Parameters: 386 * 387 * updatePeriod: position update notification period expressed in frames. 388 * 389 * Returned status (from utils/Errors.h) can be: 390 * - NO_ERROR: successful operation 391 * - INVALID_OPERATION: the AudioTrack has no callback installed. 392 */ 393 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 394 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 395 396 /* Sets playback head position. 397 * Only supported for static buffer mode. 398 * 399 * Parameters: 400 * 401 * position: New playback head position in frames relative to start of buffer. 402 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 403 * but will result in an immediate underrun if started. 404 * 405 * Returned status (from utils/Errors.h) can be: 406 * - NO_ERROR: successful operation 407 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 408 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 409 * buffer 410 */ 411 status_t setPosition(uint32_t position); 412 413 /* Return the total number of frames played since playback start. 414 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 415 * It is reset to zero by flush(), reload(), and stop(). 416 * 417 * Parameters: 418 * 419 * position: Address where to return play head position. 420 * 421 * Returned status (from utils/Errors.h) can be: 422 * - NO_ERROR: successful operation 423 * - BAD_VALUE: position is NULL 424 */ 425 status_t getPosition(uint32_t *position) const; 426 427 /* For static buffer mode only, this returns the current playback position in frames 428 * relative to start of buffer. It is analogous to the position units used by 429 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 430 */ 431 status_t getBufferPosition(uint32_t *position); 432 433 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 434 * rewriting the buffer before restarting playback after a stop. 435 * This method must be called with the AudioTrack in paused or stopped state. 436 * Not allowed in streaming mode. 437 * 438 * Returned status (from utils/Errors.h) can be: 439 * - NO_ERROR: successful operation 440 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 441 */ 442 status_t reload(); 443 444 /* Returns a handle on the audio output used by this AudioTrack. 445 * 446 * Parameters: 447 * none. 448 * 449 * Returned value: 450 * handle on audio hardware output 451 */ 452 audio_io_handle_t getOutput(); 453 454 /* Returns the unique session ID associated with this track. 455 * 456 * Parameters: 457 * none. 458 * 459 * Returned value: 460 * AudioTrack session ID. 461 */ 462 int getSessionId() const { return mSessionId; } 463 464 /* Attach track auxiliary output to specified effect. Use effectId = 0 465 * to detach track from effect. 466 * 467 * Parameters: 468 * 469 * effectId: effectId obtained from AudioEffect::id(). 470 * 471 * Returned status (from utils/Errors.h) can be: 472 * - NO_ERROR: successful operation 473 * - INVALID_OPERATION: the effect is not an auxiliary effect. 474 * - BAD_VALUE: The specified effect ID is invalid 475 */ 476 status_t attachAuxEffect(int effectId); 477 478 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 479 * After filling these slots with data, the caller should release them with releaseBuffer(). 480 * If the track buffer is not full, obtainBuffer() returns as many contiguous 481 * [empty slots for] frames as are available immediately. 482 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 483 * regardless of the value of waitCount. 484 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 485 * maximum timeout based on waitCount; see chart below. 486 * Buffers will be returned until the pool 487 * is exhausted, at which point obtainBuffer() will either block 488 * or return WOULD_BLOCK depending on the value of the "waitCount" 489 * parameter. 490 * Each sample is 16-bit signed PCM. 491 * 492 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 493 * which should use write() or callback EVENT_MORE_DATA instead. 494 * 495 * Interpretation of waitCount: 496 * +n limits wait time to n * WAIT_PERIOD_MS, 497 * -1 causes an (almost) infinite wait time, 498 * 0 non-blocking. 499 * 500 * Buffer fields 501 * On entry: 502 * frameCount number of frames requested 503 * After error return: 504 * frameCount 0 505 * size 0 506 * raw undefined 507 * After successful return: 508 * frameCount actual number of frames available, <= number requested 509 * size actual number of bytes available 510 * raw pointer to the buffer 511 */ 512 513 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 514 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 515 __attribute__((__deprecated__)); 516 517private: 518 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 519 * additional non-contiguous frames that are available immediately. 520 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 521 * in case the requested amount of frames is in two or more non-contiguous regions. 522 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 523 */ 524 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 525 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 526public: 527 528//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy 529// enum { 530// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 531// TEAR_DOWN = 0x80000002, 532// STOPPED = 1, 533// STREAM_END_WAIT, 534// STREAM_END 535// }; 536 537 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 538 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 539 void releaseBuffer(Buffer* audioBuffer); 540 541 /* As a convenience we provide a write() interface to the audio buffer. 542 * Input parameter 'size' is in byte units. 543 * This is implemented on top of obtainBuffer/releaseBuffer. For best 544 * performance use callbacks. Returns actual number of bytes written >= 0, 545 * or one of the following negative status codes: 546 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 547 * BAD_VALUE size is invalid 548 * WOULD_BLOCK when obtainBuffer() returns same, or 549 * AudioTrack was stopped during the write 550 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 551 */ 552 ssize_t write(const void* buffer, size_t size); 553 554 /* 555 * Dumps the state of an audio track. 556 */ 557 status_t dump(int fd, const Vector<String16>& args) const; 558 559 /* 560 * Return the total number of frames which AudioFlinger desired but were unavailable, 561 * and thus which resulted in an underrun. Reset to zero by stop(). 562 */ 563 uint32_t getUnderrunFrames() const; 564 565 /* Get the flags */ 566 audio_output_flags_t getFlags() const { return mFlags; } 567 568 /* Set parameters - only possible when using direct output */ 569 status_t setParameters(const String8& keyValuePairs); 570 571 /* Get parameters */ 572 String8 getParameters(const String8& keys); 573 574 /* Poll for a timestamp on demand. 575 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 576 * or if you need to get the most recent timestamp outside of the event callback handler. 577 * Caution: calling this method too often may be inefficient; 578 * if you need a high resolution mapping between frame position and presentation time, 579 * consider implementing that at application level, based on the low resolution timestamps. 580 * Returns NO_ERROR if timestamp is valid. 581 */ 582 status_t getTimestamp(AudioTimestamp& timestamp); 583 584protected: 585 /* copying audio tracks is not allowed */ 586 AudioTrack(const AudioTrack& other); 587 AudioTrack& operator = (const AudioTrack& other); 588 589 /* a small internal class to handle the callback */ 590 class AudioTrackThread : public Thread 591 { 592 public: 593 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 594 595 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 596 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 597 virtual void requestExit(); 598 599 void pause(); // suspend thread from execution at next loop boundary 600 void resume(); // allow thread to execute, if not requested to exit 601 602 private: 603 void pauseInternal(nsecs_t ns = 0LL); 604 // like pause(), but only used internally within thread 605 606 friend class AudioTrack; 607 virtual bool threadLoop(); 608 AudioTrack& mReceiver; 609 virtual ~AudioTrackThread(); 610 Mutex mMyLock; // Thread::mLock is private 611 Condition mMyCond; // Thread::mThreadExitedCondition is private 612 bool mPaused; // whether thread is requested to pause at next loop entry 613 bool mPausedInt; // whether thread internally requests pause 614 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 615 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 616 }; 617 618 // body of AudioTrackThread::threadLoop() 619 // returns the maximum amount of time before we would like to run again, where: 620 // 0 immediately 621 // > 0 no later than this many nanoseconds from now 622 // NS_WHENEVER still active but no particular deadline 623 // NS_INACTIVE inactive so don't run again until re-started 624 // NS_NEVER never again 625 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 626 nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread); 627 status_t processStreamEnd(int32_t waitCount); 628 629 630 // caller must hold lock on mLock for all _l methods 631 632 status_t createTrack_l(audio_stream_type_t streamType, 633 uint32_t sampleRate, 634 audio_format_t format, 635 size_t frameCount, 636 audio_output_flags_t flags, 637 const sp<IMemory>& sharedBuffer, 638 audio_io_handle_t output, 639 size_t epoch); 640 641 // can only be called when mState != STATE_ACTIVE 642 void flush_l(); 643 644 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 645 audio_io_handle_t getOutput_l(); 646 647 // FIXME enum is faster than strcmp() for parameter 'from' 648 status_t restoreTrack_l(const char *from); 649 650 bool isOffloaded() const 651 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 652 653 // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0 654 sp<IAudioTrack> mAudioTrack; 655 sp<IMemory> mCblkMemory; 656 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 657 658 sp<AudioTrackThread> mAudioTrackThread; 659 float mVolume[2]; 660 float mSendLevel; 661 uint32_t mSampleRate; 662 size_t mFrameCount; // corresponds to current IAudioTrack 663 size_t mReqFrameCount; // frame count to request the next time a new 664 // IAudioTrack is needed 665 666 667 // constant after constructor or set() 668 audio_format_t mFormat; // as requested by client, not forced to 16-bit 669 audio_stream_type_t mStreamType; 670 uint32_t mChannelCount; 671 audio_channel_mask_t mChannelMask; 672 transfer_type mTransfer; 673 674 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 675 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 676 size_t mFrameSize; // app-level frame size 677 size_t mFrameSizeAF; // AudioFlinger frame size 678 679 status_t mStatus; 680 681 // can change dynamically when IAudioTrack invalidated 682 uint32_t mLatency; // in ms 683 684 // Indicates the current track state. Protected by mLock. 685 enum State { 686 STATE_ACTIVE, 687 STATE_STOPPED, 688 STATE_PAUSED, 689 STATE_PAUSED_STOPPING, 690 STATE_FLUSHED, 691 STATE_STOPPING, 692 } mState; 693 694 // for client callback handler 695 callback_t mCbf; // callback handler for events, or NULL 696 void* mUserData; 697 698 // for notification APIs 699 uint32_t mNotificationFramesReq; // requested number of frames between each 700 // notification callback, 701 // at initial source sample rate 702 uint32_t mNotificationFramesAct; // actual number of frames between each 703 // notification callback, 704 // at initial source sample rate 705 bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 706 707 // These are private to processAudioBuffer(), and are not protected by a lock 708 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 709 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 710 uint32_t mObservedSequence; // last observed value of mSequence 711 712 sp<IMemory> mSharedBuffer; 713 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 714 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 715 bool mMarkerReached; 716 uint32_t mNewPosition; // in frames 717 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 718 719 audio_output_flags_t mFlags; 720 int mSessionId; 721 int mAuxEffectId; 722 723 mutable Mutex mLock; 724 725 bool mIsTimed; 726 int mPreviousPriority; // before start() 727 SchedPolicy mPreviousSchedulingGroup; 728 bool mAwaitBoost; // thread should wait for priority boost before running 729 730 // The proxy should only be referenced while a lock is held because the proxy isn't 731 // multi-thread safe, especially the SingleStateQueue part of the proxy. 732 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 733 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 734 // them around in case they are replaced during the obtainBuffer(). 735 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 736 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 737 738 bool mInUnderrun; // whether track is currently in underrun state 739 String8 mName; // server's name for this IAudioTrack 740 741private: 742 class DeathNotifier : public IBinder::DeathRecipient { 743 public: 744 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 745 protected: 746 virtual void binderDied(const wp<IBinder>& who); 747 private: 748 const wp<AudioTrack> mAudioTrack; 749 }; 750 751 sp<DeathNotifier> mDeathNotifier; 752 uint32_t mSequence; // incremented for each new IAudioTrack attempt 753 audio_io_handle_t mOutput; // cached output io handle 754}; 755 756class TimedAudioTrack : public AudioTrack 757{ 758public: 759 TimedAudioTrack(); 760 761 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 762 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 763 764 /* queue a buffer obtained via allocateTimedBuffer for playback at the 765 given timestamp. PTS units are microseconds on the media time timeline. 766 The media time transform (set with setMediaTimeTransform) set by the 767 audio producer will handle converting from media time to local time 768 (perhaps going through the common time timeline in the case of 769 synchronized multiroom audio case) */ 770 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 771 772 /* define a transform between media time and either common time or 773 local time */ 774 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 775 status_t setMediaTimeTransform(const LinearTransform& xform, 776 TargetTimeline target); 777}; 778 779}; // namespace android 780 781#endif // ANDROID_AUDIOTRACK_H 782