AudioTrack.h revision 2301acc6a9c7a3af4ad01f3d1d0f76f13eca7350
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 30struct audio_track_cblk_t; 31class AudioTrackClientProxy; 32class StaticAudioTrackClientProxy; 33 34// ---------------------------------------------------------------------------- 35 36class AudioTrack : public RefBase 37{ 38public: 39 40 /* Events used by AudioTrack callback function (callback_t). 41 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 42 */ 43 enum event_type { 44 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 45 // If this event is delivered but the callback handler 46 // does not want to write more data, the handler must explicitly 47 // ignore the event by setting frameCount to zero. 48 EVENT_UNDERRUN = 1, // Buffer underrun occurred. 49 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 50 // loop start if loop count was not 0. 51 EVENT_MARKER = 3, // Playback head is at the specified marker position 52 // (See setMarkerPosition()). 53 EVENT_NEW_POS = 4, // Playback head is at a new position 54 // (See setPositionUpdatePeriod()). 55 EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer. 56 // Not currently used by android.media.AudioTrack. 57 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 58 // voluntary invalidation by mediaserver, or mediaserver crash. 59 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 60 // back (after stop is called) 61 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 62 // in the mapping from frame position to presentation time. 63 // See AudioTimestamp for the information included with event. 64 }; 65 66 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 67 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 68 */ 69 70 class Buffer 71 { 72 public: 73 // FIXME use m prefix 74 size_t frameCount; // number of sample frames corresponding to size; 75 // on input it is the number of frames desired, 76 // on output is the number of frames actually filled 77 // (currently ignored, but will make the primary field in future) 78 79 size_t size; // input/output in bytes == frameCount * frameSize 80 // on input it is unused 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 }; // input: unused, output: pointer to buffer 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 * - EVENT_STREAM_END: unused. 111 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 112 */ 113 114 typedef void (*callback_t)(int event, void* user, void *info); 115 116 /* Returns the minimum frame count required for the successful creation of 117 * an AudioTrack object. 118 * Returned status (from utils/Errors.h) can be: 119 * - NO_ERROR: successful operation 120 * - NO_INIT: audio server or audio hardware not initialized 121 * - BAD_VALUE: unsupported configuration 122 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 123 * and is undefined otherwise. 124 */ 125 126 static status_t getMinFrameCount(size_t* frameCount, 127 audio_stream_type_t streamType, 128 uint32_t sampleRate); 129 130 /* How data is transferred to AudioTrack 131 */ 132 enum transfer_type { 133 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 134 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 135 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 136 TRANSFER_SYNC, // synchronous write() 137 TRANSFER_SHARED, // shared memory 138 }; 139 140 /* Constructs an uninitialized AudioTrack. No connection with 141 * AudioFlinger takes place. Use set() after this. 142 */ 143 AudioTrack(); 144 145 /* Creates an AudioTrack object and registers it with AudioFlinger. 146 * Once created, the track needs to be started before it can be used. 147 * Unspecified values are set to appropriate default values. 148 * With this constructor, the track is configured for streaming mode. 149 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 150 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 151 * 152 * Parameters: 153 * 154 * streamType: Select the type of audio stream this track is attached to 155 * (e.g. AUDIO_STREAM_MUSIC). 156 * sampleRate: Data source sampling rate in Hz. 157 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 158 * 16 bits per sample). 159 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 160 * frameCount: Minimum size of track PCM buffer in frames. This defines the 161 * application's contribution to the 162 * latency of the track. The actual size selected by the AudioTrack could be 163 * larger if the requested size is not compatible with current audio HAL 164 * configuration. Zero means to use a default value. 165 * flags: See comments on audio_output_flags_t in <system/audio.h>. 166 * cbf: Callback function. If not null, this function is called periodically 167 * to provide new data and inform of marker, position updates, etc. 168 * user: Context for use by the callback receiver. 169 * notificationFrames: The callback function is called each time notificationFrames PCM 170 * frames have been consumed from track input buffer. 171 * This is expressed in units of frames at the initial source sample rate. 172 * sessionId: Specific session ID, or zero to use default. 173 * transferType: How data is transferred to AudioTrack. 174 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 175 */ 176 177 AudioTrack( audio_stream_type_t streamType, 178 uint32_t sampleRate, 179 audio_format_t format, 180 audio_channel_mask_t, 181 size_t frameCount = 0, 182 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 183 callback_t cbf = NULL, 184 void* user = NULL, 185 uint32_t notificationFrames = 0, 186 int sessionId = AUDIO_SESSION_ALLOCATE, 187 transfer_type transferType = TRANSFER_DEFAULT, 188 const audio_offload_info_t *offloadInfo = NULL, 189 int uid = -1, 190 pid_t pid = -1); 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 uint32_t notificationFrames = 0, 212 int sessionId = AUDIO_SESSION_ALLOCATE, 213 transfer_type transferType = TRANSFER_DEFAULT, 214 const audio_offload_info_t *offloadInfo = NULL, 215 int uid = -1, 216 pid_t pid = -1); 217 218 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 219 * Also destroys all resources associated with the AudioTrack. 220 */ 221protected: 222 virtual ~AudioTrack(); 223public: 224 225 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 226 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 227 * Returned status (from utils/Errors.h) can be: 228 * - NO_ERROR: successful initialization 229 * - INVALID_OPERATION: AudioTrack is already initialized 230 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 231 * - NO_INIT: audio server or audio hardware not initialized 232 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 233 * If sharedBuffer is non-0, the frameCount parameter is ignored and 234 * replaced by the shared buffer's total allocated size in frame units. 235 * 236 * Parameters not listed in the AudioTrack constructors above: 237 * 238 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 239 */ 240 status_t set(audio_stream_type_t streamType, 241 uint32_t sampleRate, 242 audio_format_t format, 243 audio_channel_mask_t channelMask, 244 size_t frameCount = 0, 245 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 246 callback_t cbf = NULL, 247 void* user = NULL, 248 uint32_t notificationFrames = 0, 249 const sp<IMemory>& sharedBuffer = 0, 250 bool threadCanCallJava = false, 251 int sessionId = AUDIO_SESSION_ALLOCATE, 252 transfer_type transferType = TRANSFER_DEFAULT, 253 const audio_offload_info_t *offloadInfo = NULL, 254 int uid = -1, 255 pid_t pid = -1); 256 257 /* Result of constructing the AudioTrack. This must be checked for successful initialization 258 * before using any AudioTrack API (except for set()), because using 259 * an uninitialized AudioTrack produces undefined results. 260 * See set() method above for possible return codes. 261 */ 262 status_t initCheck() const { return mStatus; } 263 264 /* Returns this track's estimated latency in milliseconds. 265 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 266 * and audio hardware driver. 267 */ 268 uint32_t latency() const { return mLatency; } 269 270 /* getters, see constructors and set() */ 271 272 audio_stream_type_t streamType() const { return mStreamType; } 273 audio_format_t format() const { return mFormat; } 274 275 /* Return frame size in bytes, which for linear PCM is 276 * channelCount * (bit depth per channel / 8). 277 * channelCount is determined from channelMask, and bit depth comes from format. 278 * For non-linear formats, the frame size is typically 1 byte. 279 */ 280 size_t frameSize() const { return mFrameSize; } 281 282 uint32_t channelCount() const { return mChannelCount; } 283 size_t frameCount() const { return mFrameCount; } 284 285 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 286 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 287 288 /* After it's created the track is not active. Call start() to 289 * make it active. If set, the callback will start being called. 290 * If the track was previously paused, volume is ramped up over the first mix buffer. 291 */ 292 status_t start(); 293 294 /* Stop a track. 295 * In static buffer mode, the track is stopped immediately. 296 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 297 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 298 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 299 * is first drained, mixed, and output, and only then is the track marked as stopped. 300 */ 301 void stop(); 302 bool stopped() const; 303 304 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 305 * This has the effect of draining the buffers without mixing or output. 306 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 307 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 308 */ 309 void flush(); 310 311 /* Pause a track. After pause, the callback will cease being called and 312 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 313 * and will fill up buffers until the pool is exhausted. 314 * Volume is ramped down over the next mix buffer following the pause request, 315 * and then the track is marked as paused. It can be resumed with ramp up by start(). 316 */ 317 void pause(); 318 319 /* Set volume for this track, mostly used for games' sound effects 320 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 321 * This is the older API. New applications should use setVolume(float) when possible. 322 */ 323 status_t setVolume(float left, float right); 324 325 /* Set volume for all channels. This is the preferred API for new applications, 326 * especially for multi-channel content. 327 */ 328 status_t setVolume(float volume); 329 330 /* Set the send level for this track. An auxiliary effect should be attached 331 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 332 */ 333 status_t setAuxEffectSendLevel(float level); 334 void getAuxEffectSendLevel(float* level) const; 335 336 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 337 */ 338 status_t setSampleRate(uint32_t sampleRate); 339 340 /* Return current source sample rate in Hz */ 341 uint32_t getSampleRate() const; 342 343 /* Enables looping and sets the start and end points of looping. 344 * Only supported for static buffer mode. 345 * 346 * Parameters: 347 * 348 * loopStart: loop start in frames relative to start of buffer. 349 * loopEnd: loop end in frames relative to start of buffer. 350 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 351 * pending or active loop. loopCount == -1 means infinite looping. 352 * 353 * For proper operation the following condition must be respected: 354 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 355 * 356 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 357 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 358 * 359 */ 360 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 361 362 /* Sets marker position. When playback reaches the number of frames specified, a callback with 363 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 364 * notification callback. To set a marker at a position which would compute as 0, 365 * a workaround is to set the marker at a nearby position such as ~0 or 1. 366 * If the AudioTrack has been opened with no callback function associated, the operation will 367 * fail. 368 * 369 * Parameters: 370 * 371 * marker: marker position expressed in wrapping (overflow) frame units, 372 * like the return value of getPosition(). 373 * 374 * Returned status (from utils/Errors.h) can be: 375 * - NO_ERROR: successful operation 376 * - INVALID_OPERATION: the AudioTrack has no callback installed. 377 */ 378 status_t setMarkerPosition(uint32_t marker); 379 status_t getMarkerPosition(uint32_t *marker) const; 380 381 /* Sets position update period. Every time the number of frames specified has been played, 382 * a callback with event type EVENT_NEW_POS is called. 383 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 384 * callback. 385 * If the AudioTrack has been opened with no callback function associated, the operation will 386 * fail. 387 * Extremely small values may be rounded up to a value the implementation can support. 388 * 389 * Parameters: 390 * 391 * updatePeriod: position update notification period expressed in frames. 392 * 393 * Returned status (from utils/Errors.h) can be: 394 * - NO_ERROR: successful operation 395 * - INVALID_OPERATION: the AudioTrack has no callback installed. 396 */ 397 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 398 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 399 400 /* Sets playback head position. 401 * Only supported for static buffer mode. 402 * 403 * Parameters: 404 * 405 * position: New playback head position in frames relative to start of buffer. 406 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 407 * but will result in an immediate underrun if started. 408 * 409 * Returned status (from utils/Errors.h) can be: 410 * - NO_ERROR: successful operation 411 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 412 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 413 * buffer 414 */ 415 status_t setPosition(uint32_t position); 416 417 /* Return the total number of frames played since playback start. 418 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 419 * It is reset to zero by flush(), reload(), and stop(). 420 * 421 * Parameters: 422 * 423 * position: Address where to return play head position. 424 * 425 * Returned status (from utils/Errors.h) can be: 426 * - NO_ERROR: successful operation 427 * - BAD_VALUE: position is NULL 428 */ 429 status_t getPosition(uint32_t *position) const; 430 431 /* For static buffer mode only, this returns the current playback position in frames 432 * relative to start of buffer. It is analogous to the position units used by 433 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 434 */ 435 status_t getBufferPosition(uint32_t *position); 436 437 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 438 * rewriting the buffer before restarting playback after a stop. 439 * This method must be called with the AudioTrack in paused or stopped state. 440 * Not allowed in streaming mode. 441 * 442 * Returned status (from utils/Errors.h) can be: 443 * - NO_ERROR: successful operation 444 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 445 */ 446 status_t reload(); 447 448 /* Returns a handle on the audio output used by this AudioTrack. 449 * 450 * Parameters: 451 * none. 452 * 453 * Returned value: 454 * handle on audio hardware output 455 */ 456 audio_io_handle_t getOutput() const; 457 458 /* Returns the unique session ID associated with this track. 459 * 460 * Parameters: 461 * none. 462 * 463 * Returned value: 464 * AudioTrack session ID. 465 */ 466 int getSessionId() const { return mSessionId; } 467 468 /* Attach track auxiliary output to specified effect. Use effectId = 0 469 * to detach track from effect. 470 * 471 * Parameters: 472 * 473 * effectId: effectId obtained from AudioEffect::id(). 474 * 475 * Returned status (from utils/Errors.h) can be: 476 * - NO_ERROR: successful operation 477 * - INVALID_OPERATION: the effect is not an auxiliary effect. 478 * - BAD_VALUE: The specified effect ID is invalid 479 */ 480 status_t attachAuxEffect(int effectId); 481 482 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 483 * After filling these slots with data, the caller should release them with releaseBuffer(). 484 * If the track buffer is not full, obtainBuffer() returns as many contiguous 485 * [empty slots for] frames as are available immediately. 486 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 487 * regardless of the value of waitCount. 488 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 489 * maximum timeout based on waitCount; see chart below. 490 * Buffers will be returned until the pool 491 * is exhausted, at which point obtainBuffer() will either block 492 * or return WOULD_BLOCK depending on the value of the "waitCount" 493 * parameter. 494 * Each sample is 16-bit signed PCM. 495 * 496 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 497 * which should use write() or callback EVENT_MORE_DATA instead. 498 * 499 * Interpretation of waitCount: 500 * +n limits wait time to n * WAIT_PERIOD_MS, 501 * -1 causes an (almost) infinite wait time, 502 * 0 non-blocking. 503 * 504 * Buffer fields 505 * On entry: 506 * frameCount number of frames requested 507 * After error return: 508 * frameCount 0 509 * size 0 510 * raw undefined 511 * After successful return: 512 * frameCount actual number of frames available, <= number requested 513 * size actual number of bytes available 514 * raw pointer to the buffer 515 */ 516 517 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 518 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 519 __attribute__((__deprecated__)); 520 521private: 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 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 533 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 534 void releaseBuffer(Buffer* audioBuffer); 535 536 /* As a convenience we provide a write() interface to the audio buffer. 537 * Input parameter 'size' is in byte units. 538 * This is implemented on top of obtainBuffer/releaseBuffer. For best 539 * performance use callbacks. Returns actual number of bytes written >= 0, 540 * or one of the following negative status codes: 541 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 542 * BAD_VALUE size is invalid 543 * WOULD_BLOCK when obtainBuffer() returns same, or 544 * AudioTrack was stopped during the write 545 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 546 * Default behavior is to only return until all data has been transferred. Set 'blocking' to 547 * false for the method to return immediately without waiting to try multiple times to write 548 * the full content of the buffer. 549 */ 550 ssize_t write(const void* buffer, size_t size, bool blocking = true); 551 552 /* 553 * Dumps the state of an audio track. 554 */ 555 status_t dump(int fd, const Vector<String16>& args) const; 556 557 /* 558 * Return the total number of frames which AudioFlinger desired but were unavailable, 559 * and thus which resulted in an underrun. Reset to zero by stop(). 560 */ 561 uint32_t getUnderrunFrames() const; 562 563 /* Get the flags */ 564 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 565 566 /* Set parameters - only possible when using direct output */ 567 status_t setParameters(const String8& keyValuePairs); 568 569 /* Get parameters */ 570 String8 getParameters(const String8& keys); 571 572 /* Poll for a timestamp on demand. 573 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 574 * or if you need to get the most recent timestamp outside of the event callback handler. 575 * Caution: calling this method too often may be inefficient; 576 * if you need a high resolution mapping between frame position and presentation time, 577 * consider implementing that at application level, based on the low resolution timestamps. 578 * Returns NO_ERROR if timestamp is valid. 579 */ 580 status_t getTimestamp(AudioTimestamp& timestamp); 581 582protected: 583 /* copying audio tracks is not allowed */ 584 AudioTrack(const AudioTrack& other); 585 AudioTrack& operator = (const AudioTrack& other); 586 587 /* a small internal class to handle the callback */ 588 class AudioTrackThread : public Thread 589 { 590 public: 591 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 592 593 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 594 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 595 virtual void requestExit(); 596 597 void pause(); // suspend thread from execution at next loop boundary 598 void resume(); // allow thread to execute, if not requested to exit 599 600 private: 601 void pauseInternal(nsecs_t ns = 0LL); 602 // like pause(), but only used internally within thread 603 604 friend class AudioTrack; 605 virtual bool threadLoop(); 606 AudioTrack& mReceiver; 607 virtual ~AudioTrackThread(); 608 Mutex mMyLock; // Thread::mLock is private 609 Condition mMyCond; // Thread::mThreadExitedCondition is private 610 bool mPaused; // whether thread is requested to pause at next loop entry 611 bool mPausedInt; // whether thread internally requests pause 612 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 613 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 614 }; 615 616 // body of AudioTrackThread::threadLoop() 617 // returns the maximum amount of time before we would like to run again, where: 618 // 0 immediately 619 // > 0 no later than this many nanoseconds from now 620 // NS_WHENEVER still active but no particular deadline 621 // NS_INACTIVE inactive so don't run again until re-started 622 // NS_NEVER never again 623 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 624 nsecs_t processAudioBuffer(); 625 626 bool isOffloaded() const; 627 628 // caller must hold lock on mLock for all _l methods 629 630 status_t createTrack_l(size_t epoch); 631 632 // can only be called when mState != STATE_ACTIVE 633 void flush_l(); 634 635 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 636 637 // FIXME enum is faster than strcmp() for parameter 'from' 638 status_t restoreTrack_l(const char *from); 639 640 bool isOffloaded_l() const 641 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 642 643 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 644 sp<IAudioTrack> mAudioTrack; 645 sp<IMemory> mCblkMemory; 646 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 647 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 648 649 sp<AudioTrackThread> mAudioTrackThread; 650 651 float mVolume[2]; 652 float mSendLevel; 653 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. 654 size_t mFrameCount; // corresponds to current IAudioTrack, value is 655 // reported back by AudioFlinger to the client 656 size_t mReqFrameCount; // frame count to request the first or next time 657 // a new IAudioTrack is needed, non-decreasing 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 sp<IMemory> mSharedBuffer; 665 transfer_type mTransfer; 666 audio_offload_info_t mOffloadInfoCopy; 667 const audio_offload_info_t* mOffloadInfo; 668 669 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 670 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 671 size_t mFrameSize; // app-level frame size 672 size_t mFrameSizeAF; // AudioFlinger frame size 673 674 status_t mStatus; 675 676 // can change dynamically when IAudioTrack invalidated 677 uint32_t mLatency; // in ms 678 679 // Indicates the current track state. Protected by mLock. 680 enum State { 681 STATE_ACTIVE, 682 STATE_STOPPED, 683 STATE_PAUSED, 684 STATE_PAUSED_STOPPING, 685 STATE_FLUSHED, 686 STATE_STOPPING, 687 } mState; 688 689 // for client callback handler 690 callback_t mCbf; // callback handler for events, or NULL 691 void* mUserData; 692 693 // for notification APIs 694 uint32_t mNotificationFramesReq; // requested number of frames between each 695 // notification callback, 696 // at initial source sample rate 697 uint32_t mNotificationFramesAct; // actual number of frames between each 698 // notification callback, 699 // at initial source sample rate 700 bool mRefreshRemaining; // processAudioBuffer() should refresh 701 // mRemainingFrames and mRetryOnPartialBuffer 702 703 // These are private to processAudioBuffer(), and are not protected by a lock 704 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 705 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 706 uint32_t mObservedSequence; // last observed value of mSequence 707 708 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 709 710 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 711 bool mMarkerReached; 712 uint32_t mNewPosition; // in frames 713 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 714 715 audio_output_flags_t mFlags; 716 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 717 // mLock must be held to read or write those bits reliably. 718 719 int mSessionId; 720 int mAuxEffectId; 721 722 mutable Mutex mLock; 723 724 bool mIsTimed; 725 int mPreviousPriority; // before start() 726 SchedPolicy mPreviousSchedulingGroup; 727 bool mAwaitBoost; // thread should wait for priority boost before running 728 729 // The proxy should only be referenced while a lock is held because the proxy isn't 730 // multi-thread safe, especially the SingleStateQueue part of the proxy. 731 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 732 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 733 // them around in case they are replaced during the obtainBuffer(). 734 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 735 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 736 737 bool mInUnderrun; // whether track is currently in underrun state 738 uint32_t mPausedPosition; 739 740private: 741 class DeathNotifier : public IBinder::DeathRecipient { 742 public: 743 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 744 protected: 745 virtual void binderDied(const wp<IBinder>& who); 746 private: 747 const wp<AudioTrack> mAudioTrack; 748 }; 749 750 sp<DeathNotifier> mDeathNotifier; 751 uint32_t mSequence; // incremented for each new IAudioTrack attempt 752 int mClientUid; 753 pid_t mClientPid; 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