AudioTrack.h revision b46f394a85d704dd05287cf9bb77cf86e3c02a38
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 a Buffer and pass the 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 to obtainBuffer() it is the number of frames desired, 76 // on output from obtainBuffer() it is the number of available 77 // [empty slots for] frames to be filled 78 // on input to releaseBuffer() it is currently ignored 79 80 size_t size; // input/output in bytes == frameCount * frameSize 81 // on input to obtainBuffer() it is ignored 82 // on output from obtainBuffer() it is the number of available 83 // [empty slots for] bytes to be filled, 84 // which is frameCount * frameSize 85 // on input to releaseBuffer() it is the number of bytes to 86 // release 87 // FIXME This is redundant with respect to frameCount. Consider 88 // removing size and making frameCount the primary field. 89 90 union { 91 void* raw; 92 short* i16; // signed 16-bit 93 int8_t* i8; // unsigned 8-bit, offset by 0x80 94 }; // input: unused, output: pointer to buffer 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 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 127 * and is undefined otherwise. 128 */ 129 130 static status_t getMinFrameCount(size_t* frameCount, 131 audio_stream_type_t streamType, 132 uint32_t sampleRate); 133 134 /* How data is transferred to AudioTrack 135 */ 136 enum transfer_type { 137 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 138 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 139 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 140 TRANSFER_SYNC, // synchronous write() 141 TRANSFER_SHARED, // shared memory 142 }; 143 144 /* Constructs an uninitialized AudioTrack. No connection with 145 * AudioFlinger takes place. Use set() after this. 146 */ 147 AudioTrack(); 148 149 /* Creates an AudioTrack object and registers it with AudioFlinger. 150 * Once created, the track needs to be started before it can be used. 151 * Unspecified values are set to appropriate default values. 152 * With this constructor, the track is configured for streaming mode. 153 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 154 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 155 * 156 * Parameters: 157 * 158 * streamType: Select the type of audio stream this track is attached to 159 * (e.g. AUDIO_STREAM_MUSIC). 160 * sampleRate: Data source sampling rate in Hz. 161 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 162 * For direct and offloaded tracks, the possible format(s) depends on the 163 * output sink. 164 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 165 * frameCount: Minimum size of track PCM buffer in frames. This defines the 166 * application's contribution to the 167 * latency of the track. The actual size selected by the AudioTrack could be 168 * larger if the requested size is not compatible with current audio HAL 169 * configuration. Zero means to use a default value. 170 * flags: See comments on audio_output_flags_t in <system/audio.h>. 171 * cbf: Callback function. If not null, this function is called periodically 172 * to provide new data and inform of marker, position updates, etc. 173 * user: Context for use by the callback receiver. 174 * notificationFrames: The callback function is called each time notificationFrames PCM 175 * frames have been consumed from track input buffer. 176 * This is expressed in units of frames at the initial source sample rate. 177 * sessionId: Specific session ID, or zero to use default. 178 * transferType: How data is transferred to AudioTrack. 179 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 180 */ 181 182 AudioTrack( audio_stream_type_t streamType, 183 uint32_t sampleRate, 184 audio_format_t format, 185 audio_channel_mask_t, 186 size_t frameCount = 0, 187 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 188 callback_t cbf = NULL, 189 void* user = NULL, 190 uint32_t notificationFrames = 0, 191 int sessionId = AUDIO_SESSION_ALLOCATE, 192 transfer_type transferType = TRANSFER_DEFAULT, 193 const audio_offload_info_t *offloadInfo = NULL, 194 int uid = -1, 195 pid_t pid = -1, 196 const audio_attributes_t* pAttributes = NULL); 197 198 /* Creates an audio track and registers it with AudioFlinger. 199 * With this constructor, the track is configured for static buffer mode. 200 * Data to be rendered is passed in a shared memory buffer 201 * identified by the argument sharedBuffer, which must be non-0. 202 * The memory should be initialized to the desired data before calling start(). 203 * The write() method is not supported in this case. 204 * It is recommended to pass a callback function to be notified of playback end by an 205 * EVENT_UNDERRUN event. 206 */ 207 208 AudioTrack( audio_stream_type_t streamType, 209 uint32_t sampleRate, 210 audio_format_t format, 211 audio_channel_mask_t channelMask, 212 const sp<IMemory>& sharedBuffer, 213 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 214 callback_t cbf = NULL, 215 void* user = NULL, 216 uint32_t notificationFrames = 0, 217 int sessionId = AUDIO_SESSION_ALLOCATE, 218 transfer_type transferType = TRANSFER_DEFAULT, 219 const audio_offload_info_t *offloadInfo = NULL, 220 int uid = -1, 221 pid_t pid = -1, 222 const audio_attributes_t* pAttributes = NULL); 223 224 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 225 * Also destroys all resources associated with the AudioTrack. 226 */ 227protected: 228 virtual ~AudioTrack(); 229public: 230 231 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 232 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 233 * Returned status (from utils/Errors.h) can be: 234 * - NO_ERROR: successful initialization 235 * - INVALID_OPERATION: AudioTrack is already initialized 236 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 237 * - NO_INIT: audio server or audio hardware not initialized 238 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 239 * If sharedBuffer is non-0, the frameCount parameter is ignored and 240 * replaced by the shared buffer's total allocated size in frame units. 241 * 242 * Parameters not listed in the AudioTrack constructors above: 243 * 244 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 245 * 246 * Internal state post condition: 247 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 248 */ 249 status_t set(audio_stream_type_t streamType, 250 uint32_t sampleRate, 251 audio_format_t format, 252 audio_channel_mask_t channelMask, 253 size_t frameCount = 0, 254 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 255 callback_t cbf = NULL, 256 void* user = NULL, 257 uint32_t notificationFrames = 0, 258 const sp<IMemory>& sharedBuffer = 0, 259 bool threadCanCallJava = false, 260 int sessionId = AUDIO_SESSION_ALLOCATE, 261 transfer_type transferType = TRANSFER_DEFAULT, 262 const audio_offload_info_t *offloadInfo = NULL, 263 int uid = -1, 264 pid_t pid = -1, 265 const audio_attributes_t* pAttributes = NULL); 266 267 /* Result of constructing the AudioTrack. This must be checked for successful initialization 268 * before using any AudioTrack API (except for set()), because using 269 * an uninitialized AudioTrack produces undefined results. 270 * See set() method above for possible return codes. 271 */ 272 status_t initCheck() const { return mStatus; } 273 274 /* Returns this track's estimated latency in milliseconds. 275 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 276 * and audio hardware driver. 277 */ 278 uint32_t latency() const { return mLatency; } 279 280 /* getters, see constructors and set() */ 281 282 audio_stream_type_t streamType() const; 283 audio_format_t format() const { return mFormat; } 284 285 /* Return frame size in bytes, which for linear PCM is 286 * channelCount * (bit depth per channel / 8). 287 * channelCount is determined from channelMask, and bit depth comes from format. 288 * For non-linear formats, the frame size is typically 1 byte. 289 */ 290 size_t frameSize() const { return mFrameSize; } 291 292 uint32_t channelCount() const { return mChannelCount; } 293 size_t frameCount() const { return mFrameCount; } 294 295 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 296 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 297 298 /* After it's created the track is not active. Call start() to 299 * make it active. If set, the callback will start being called. 300 * If the track was previously paused, volume is ramped up over the first mix buffer. 301 */ 302 status_t start(); 303 304 /* Stop a track. 305 * In static buffer mode, the track is stopped immediately. 306 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 307 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 308 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 309 * is first drained, mixed, and output, and only then is the track marked as stopped. 310 */ 311 void stop(); 312 bool stopped() const; 313 314 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 315 * This has the effect of draining the buffers without mixing or output. 316 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 317 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 318 */ 319 void flush(); 320 321 /* Pause a track. After pause, the callback will cease being called and 322 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 323 * and will fill up buffers until the pool is exhausted. 324 * Volume is ramped down over the next mix buffer following the pause request, 325 * and then the track is marked as paused. It can be resumed with ramp up by start(). 326 */ 327 void pause(); 328 329 /* Set volume for this track, mostly used for games' sound effects 330 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 331 * This is the older API. New applications should use setVolume(float) when possible. 332 */ 333 status_t setVolume(float left, float right); 334 335 /* Set volume for all channels. This is the preferred API for new applications, 336 * especially for multi-channel content. 337 */ 338 status_t setVolume(float volume); 339 340 /* Set the send level for this track. An auxiliary effect should be attached 341 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 342 */ 343 status_t setAuxEffectSendLevel(float level); 344 void getAuxEffectSendLevel(float* level) const; 345 346 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 347 */ 348 status_t setSampleRate(uint32_t sampleRate); 349 350 /* Return current source sample rate in Hz */ 351 uint32_t getSampleRate() const; 352 353 /* Enables looping and sets the start and end points of looping. 354 * Only supported for static buffer mode. 355 * 356 * Parameters: 357 * 358 * loopStart: loop start in frames relative to start of buffer. 359 * loopEnd: loop end in frames relative to start of buffer. 360 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 361 * pending or active loop. loopCount == -1 means infinite looping. 362 * 363 * For proper operation the following condition must be respected: 364 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 365 * 366 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 367 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 368 * 369 */ 370 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 371 372 /* Sets marker position. When playback reaches the number of frames specified, a callback with 373 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 374 * notification callback. To set a marker at a position which would compute as 0, 375 * a workaround is to set the marker at a nearby position such as ~0 or 1. 376 * If the AudioTrack has been opened with no callback function associated, the operation will 377 * fail. 378 * 379 * Parameters: 380 * 381 * marker: marker position expressed in wrapping (overflow) frame units, 382 * like the return value of getPosition(). 383 * 384 * Returned status (from utils/Errors.h) can be: 385 * - NO_ERROR: successful operation 386 * - INVALID_OPERATION: the AudioTrack has no callback installed. 387 */ 388 status_t setMarkerPosition(uint32_t marker); 389 status_t getMarkerPosition(uint32_t *marker) const; 390 391 /* Sets position update period. Every time the number of frames specified has been played, 392 * a callback with event type EVENT_NEW_POS is called. 393 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 394 * callback. 395 * If the AudioTrack has been opened with no callback function associated, the operation will 396 * fail. 397 * Extremely small values may be rounded up to a value the implementation can support. 398 * 399 * Parameters: 400 * 401 * updatePeriod: position update notification period expressed in frames. 402 * 403 * Returned status (from utils/Errors.h) can be: 404 * - NO_ERROR: successful operation 405 * - INVALID_OPERATION: the AudioTrack has no callback installed. 406 */ 407 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 408 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 409 410 /* Sets playback head position. 411 * Only supported for static buffer mode. 412 * 413 * Parameters: 414 * 415 * position: New playback head position in frames relative to start of buffer. 416 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 417 * but will result in an immediate underrun if started. 418 * 419 * Returned status (from utils/Errors.h) can be: 420 * - NO_ERROR: successful operation 421 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 422 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 423 * buffer 424 */ 425 status_t setPosition(uint32_t position); 426 427 /* Return the total number of frames played since playback start. 428 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 429 * It is reset to zero by flush(), reload(), and stop(). 430 * 431 * Parameters: 432 * 433 * position: Address where to return play head position. 434 * 435 * Returned status (from utils/Errors.h) can be: 436 * - NO_ERROR: successful operation 437 * - BAD_VALUE: position is NULL 438 */ 439 status_t getPosition(uint32_t *position); 440 441 /* For static buffer mode only, this returns the current playback position in frames 442 * relative to start of buffer. It is analogous to the position units used by 443 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 444 */ 445 status_t getBufferPosition(uint32_t *position); 446 447 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 448 * rewriting the buffer before restarting playback after a stop. 449 * This method must be called with the AudioTrack in paused or stopped state. 450 * Not allowed in streaming mode. 451 * 452 * Returned status (from utils/Errors.h) can be: 453 * - NO_ERROR: successful operation 454 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 455 */ 456 status_t reload(); 457 458 /* Returns a handle on the audio output used by this AudioTrack. 459 * 460 * Parameters: 461 * none. 462 * 463 * Returned value: 464 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 465 * track needed to be re-created but that failed 466 */ 467 audio_io_handle_t getOutput() const; 468 469 /* Returns the unique session ID associated with this track. 470 * 471 * Parameters: 472 * none. 473 * 474 * Returned value: 475 * AudioTrack session ID. 476 */ 477 int getSessionId() const { return mSessionId; } 478 479 /* Attach track auxiliary output to specified effect. Use effectId = 0 480 * to detach track from effect. 481 * 482 * Parameters: 483 * 484 * effectId: effectId obtained from AudioEffect::id(). 485 * 486 * Returned status (from utils/Errors.h) can be: 487 * - NO_ERROR: successful operation 488 * - INVALID_OPERATION: the effect is not an auxiliary effect. 489 * - BAD_VALUE: The specified effect ID is invalid 490 */ 491 status_t attachAuxEffect(int effectId); 492 493 /* Public API for TRANSFER_OBTAIN mode. 494 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 495 * After filling these slots with data, the caller should release them with releaseBuffer(). 496 * If the track buffer is not full, obtainBuffer() returns as many contiguous 497 * [empty slots for] frames as are available immediately. 498 * 499 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 500 * additional non-contiguous frames that are predicted to be available immediately, 501 * if the client were to release the first frames and then call obtainBuffer() again. 502 * This value is only a prediction, and needs to be confirmed. 503 * It will be set to zero for an error return. 504 * 505 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 506 * regardless of the value of waitCount. 507 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 508 * maximum timeout based on waitCount; see chart below. 509 * Buffers will be returned until the pool 510 * is exhausted, at which point obtainBuffer() will either block 511 * or return WOULD_BLOCK depending on the value of the "waitCount" 512 * parameter. 513 * 514 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 515 * which should use write() or callback EVENT_MORE_DATA instead. 516 * 517 * Interpretation of waitCount: 518 * +n limits wait time to n * WAIT_PERIOD_MS, 519 * -1 causes an (almost) infinite wait time, 520 * 0 non-blocking. 521 * 522 * Buffer fields 523 * On entry: 524 * frameCount number of [empty slots for] frames requested 525 * size ignored 526 * raw ignored 527 * After error return: 528 * frameCount 0 529 * size 0 530 * raw undefined 531 * After successful return: 532 * frameCount actual number of [empty slots for] frames available, <= number requested 533 * size actual number of bytes available 534 * raw pointer to the buffer 535 */ 536 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 537 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 538 size_t *nonContig = NULL) 539 __attribute__((__deprecated__)); 540 541private: 542 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 543 * additional non-contiguous frames that are predicted to be available immediately, 544 * if the client were to release the first frames and then call obtainBuffer() again. 545 * This value is only a prediction, and needs to be confirmed. 546 * It will be set to zero for an error return. 547 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 548 * in case the requested amount of frames is in two or more non-contiguous regions. 549 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 550 */ 551 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 552 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 553public: 554 555 /* Public API for TRANSFER_OBTAIN mode. 556 * Release a filled buffer of frames for AudioFlinger to process. 557 * 558 * Buffer fields: 559 * frameCount currently ignored but recommend to set to actual number of frames filled 560 * size actual number of bytes filled, must be multiple of frameSize 561 * raw ignored 562 * 563 */ 564 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 565 void releaseBuffer(const Buffer* audioBuffer); 566 567 /* As a convenience we provide a write() interface to the audio buffer. 568 * Input parameter 'size' is in byte units. 569 * This is implemented on top of obtainBuffer/releaseBuffer. For best 570 * performance use callbacks. Returns actual number of bytes written >= 0, 571 * or one of the following negative status codes: 572 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 573 * BAD_VALUE size is invalid 574 * WOULD_BLOCK when obtainBuffer() returns same, or 575 * AudioTrack was stopped during the write 576 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 577 * Default behavior is to only return until all data has been transferred. Set 'blocking' to 578 * false for the method to return immediately without waiting to try multiple times to write 579 * the full content of the buffer. 580 */ 581 ssize_t write(const void* buffer, size_t size, bool blocking = true); 582 583 /* 584 * Dumps the state of an audio track. 585 */ 586 status_t dump(int fd, const Vector<String16>& args) const; 587 588 /* 589 * Return the total number of frames which AudioFlinger desired but were unavailable, 590 * and thus which resulted in an underrun. Reset to zero by stop(). 591 */ 592 uint32_t getUnderrunFrames() const; 593 594 /* Get the flags */ 595 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 596 597 /* Set parameters - only possible when using direct output */ 598 status_t setParameters(const String8& keyValuePairs); 599 600 /* Get parameters */ 601 String8 getParameters(const String8& keys); 602 603 /* Poll for a timestamp on demand. 604 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 605 * or if you need to get the most recent timestamp outside of the event callback handler. 606 * Caution: calling this method too often may be inefficient; 607 * if you need a high resolution mapping between frame position and presentation time, 608 * consider implementing that at application level, based on the low resolution timestamps. 609 * Returns NO_ERROR if timestamp is valid. 610 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 611 * start/ACTIVE, when the number of frames consumed is less than the 612 * overall hardware latency to physical output. In WOULD_BLOCK cases, 613 * one might poll again, or use getPosition(), or use 0 position and 614 * current time for the timestamp. 615 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 616 * 617 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 618 */ 619 status_t getTimestamp(AudioTimestamp& timestamp); 620 621protected: 622 /* copying audio tracks is not allowed */ 623 AudioTrack(const AudioTrack& other); 624 AudioTrack& operator = (const AudioTrack& other); 625 626 void setAttributesFromStreamType(audio_stream_type_t streamType); 627 628 /* a small internal class to handle the callback */ 629 class AudioTrackThread : public Thread 630 { 631 public: 632 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 633 634 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 635 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 636 virtual void requestExit(); 637 638 void pause(); // suspend thread from execution at next loop boundary 639 void resume(); // allow thread to execute, if not requested to exit 640 void wake(); // wake to handle changed notification conditions. 641 642 private: 643 void pauseInternal(nsecs_t ns = 0LL); 644 // like pause(), but only used internally within thread 645 646 friend class AudioTrack; 647 virtual bool threadLoop(); 648 AudioTrack& mReceiver; 649 virtual ~AudioTrackThread(); 650 Mutex mMyLock; // Thread::mLock is private 651 Condition mMyCond; // Thread::mThreadExitedCondition is private 652 bool mPaused; // whether thread is requested to pause at next loop entry 653 bool mPausedInt; // whether thread internally requests pause 654 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 655 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 656 // to processAudioBuffer() as state may have changed 657 // since pause time calculated. 658 }; 659 660 // body of AudioTrackThread::threadLoop() 661 // returns the maximum amount of time before we would like to run again, where: 662 // 0 immediately 663 // > 0 no later than this many nanoseconds from now 664 // NS_WHENEVER still active but no particular deadline 665 // NS_INACTIVE inactive so don't run again until re-started 666 // NS_NEVER never again 667 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 668 nsecs_t processAudioBuffer(); 669 670 bool isOffloaded() const; 671 bool isDirect() const; 672 bool isOffloadedOrDirect() const; 673 674 // caller must hold lock on mLock for all _l methods 675 676 status_t createTrack_l(); 677 678 // can only be called when mState != STATE_ACTIVE 679 void flush_l(); 680 681 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 682 683 // FIXME enum is faster than strcmp() for parameter 'from' 684 status_t restoreTrack_l(const char *from); 685 686 bool isOffloaded_l() const 687 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 688 689 bool isOffloadedOrDirect_l() const 690 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 691 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 692 693 bool isDirect_l() const 694 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 695 696 // increment mPosition by the delta of mServer, and return new value of mPosition 697 uint32_t updateAndGetPosition_l(); 698 699 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 700 sp<IAudioTrack> mAudioTrack; 701 sp<IMemory> mCblkMemory; 702 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 703 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 704 705 sp<AudioTrackThread> mAudioTrackThread; 706 707 float mVolume[2]; 708 float mSendLevel; 709 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 710 size_t mFrameCount; // corresponds to current IAudioTrack, value is 711 // reported back by AudioFlinger to the client 712 size_t mReqFrameCount; // frame count to request the first or next time 713 // a new IAudioTrack is needed, non-decreasing 714 715 // constant after constructor or set() 716 audio_format_t mFormat; // as requested by client, not forced to 16-bit 717 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 718 // this AudioTrack has valid attributes 719 uint32_t mChannelCount; 720 audio_channel_mask_t mChannelMask; 721 sp<IMemory> mSharedBuffer; 722 transfer_type mTransfer; 723 audio_offload_info_t mOffloadInfoCopy; 724 const audio_offload_info_t* mOffloadInfo; 725 audio_attributes_t mAttributes; 726 727 size_t mFrameSize; // frame size in bytes 728 729 status_t mStatus; 730 731 // can change dynamically when IAudioTrack invalidated 732 uint32_t mLatency; // in ms 733 734 // Indicates the current track state. Protected by mLock. 735 enum State { 736 STATE_ACTIVE, 737 STATE_STOPPED, 738 STATE_PAUSED, 739 STATE_PAUSED_STOPPING, 740 STATE_FLUSHED, 741 STATE_STOPPING, 742 } mState; 743 744 // for client callback handler 745 callback_t mCbf; // callback handler for events, or NULL 746 void* mUserData; 747 748 // for notification APIs 749 uint32_t mNotificationFramesReq; // requested number of frames between each 750 // notification callback, 751 // at initial source sample rate 752 uint32_t mNotificationFramesAct; // actual number of frames between each 753 // notification callback, 754 // at initial source sample rate 755 bool mRefreshRemaining; // processAudioBuffer() should refresh 756 // mRemainingFrames and mRetryOnPartialBuffer 757 758 // used for static track cbf and restoration 759 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 760 uint32_t mLoopStart; // last setLoop loopStart 761 uint32_t mLoopEnd; // last setLoop loopEnd 762 int32_t mLoopCountNotified; // the last loopCount notified by callback. 763 // mLoopCountNotified counts down, matching 764 // the remaining loop count for static track 765 // playback. 766 767 // These are private to processAudioBuffer(), and are not protected by a lock 768 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 769 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 770 uint32_t mObservedSequence; // last observed value of mSequence 771 772 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 773 bool mMarkerReached; 774 uint32_t mNewPosition; // in frames 775 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 776 uint32_t mServer; // in frames, last known mProxy->getPosition() 777 // which is count of frames consumed by server, 778 // reset by new IAudioTrack, 779 // whether it is reset by stop() is TBD 780 uint32_t mPosition; // in frames, like mServer except continues 781 // monotonically after new IAudioTrack, 782 // and could be easily widened to uint64_t 783 uint32_t mReleased; // in frames, count of frames released to server 784 // but not necessarily consumed by server, 785 // reset by stop() but continues monotonically 786 // after new IAudioTrack to restore mPosition, 787 // and could be easily widened to uint64_t 788 int64_t mStartUs; // the start time after flush or stop. 789 // only used for offloaded and direct tracks. 790 791 audio_output_flags_t mFlags; 792 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 793 // mLock must be held to read or write those bits reliably. 794 795 int mSessionId; 796 int mAuxEffectId; 797 798 mutable Mutex mLock; 799 800 bool mIsTimed; 801 int mPreviousPriority; // before start() 802 SchedPolicy mPreviousSchedulingGroup; 803 bool mAwaitBoost; // thread should wait for priority boost before running 804 805 // The proxy should only be referenced while a lock is held because the proxy isn't 806 // multi-thread safe, especially the SingleStateQueue part of the proxy. 807 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 808 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 809 // them around in case they are replaced during the obtainBuffer(). 810 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 811 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 812 813 bool mInUnderrun; // whether track is currently in underrun state 814 uint32_t mPausedPosition; 815 816private: 817 class DeathNotifier : public IBinder::DeathRecipient { 818 public: 819 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 820 protected: 821 virtual void binderDied(const wp<IBinder>& who); 822 private: 823 const wp<AudioTrack> mAudioTrack; 824 }; 825 826 sp<DeathNotifier> mDeathNotifier; 827 uint32_t mSequence; // incremented for each new IAudioTrack attempt 828 int mClientUid; 829 pid_t mClientPid; 830}; 831 832class TimedAudioTrack : public AudioTrack 833{ 834public: 835 TimedAudioTrack(); 836 837 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 838 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 839 840 /* queue a buffer obtained via allocateTimedBuffer for playback at the 841 given timestamp. PTS units are microseconds on the media time timeline. 842 The media time transform (set with setMediaTimeTransform) set by the 843 audio producer will handle converting from media time to local time 844 (perhaps going through the common time timeline in the case of 845 synchronized multiroom audio case) */ 846 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 847 848 /* define a transform between media time and either common time or 849 local time */ 850 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 851 status_t setMediaTimeTransform(const LinearTransform& xform, 852 TargetTimeline target); 853}; 854 855}; // namespace android 856 857#endif // ANDROID_AUDIOTRACK_H 858