AudioTrack.h revision 6991ed25d8b5c2f75325ccd7cc522762da6f2b09
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 * FIXME This API assumes a route, and so should be deprecated. 129 */ 130 131 static status_t getMinFrameCount(size_t* frameCount, 132 audio_stream_type_t streamType, 133 uint32_t sampleRate); 134 135 /* How data is transferred to AudioTrack 136 */ 137 enum transfer_type { 138 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 139 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 140 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 141 TRANSFER_SYNC, // synchronous write() 142 TRANSFER_SHARED, // shared memory 143 }; 144 145 /* Constructs an uninitialized AudioTrack. No connection with 146 * AudioFlinger takes place. Use set() after this. 147 */ 148 AudioTrack(); 149 150 /* Creates an AudioTrack object and registers it with AudioFlinger. 151 * Once created, the track needs to be started before it can be used. 152 * Unspecified values are set to appropriate default values. 153 * With this constructor, the track is configured for streaming mode. 154 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 155 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 156 * 157 * Parameters: 158 * 159 * streamType: Select the type of audio stream this track is attached to 160 * (e.g. AUDIO_STREAM_MUSIC). 161 * sampleRate: Data source sampling rate in Hz. 162 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 163 * For direct and offloaded tracks, the possible format(s) depends on the 164 * output sink. 165 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 166 * frameCount: Minimum size of track PCM buffer in frames. This defines the 167 * application's contribution to the 168 * latency of the track. The actual size selected by the AudioTrack could be 169 * larger if the requested size is not compatible with current audio HAL 170 * configuration. Zero means to use a default value. 171 * flags: See comments on audio_output_flags_t in <system/audio.h>. 172 * cbf: Callback function. If not null, this function is called periodically 173 * to provide new data and inform of marker, position updates, etc. 174 * user: Context for use by the callback receiver. 175 * notificationFrames: The callback function is called each time notificationFrames PCM 176 * frames have been consumed from track input buffer. 177 * This is expressed in units of frames at the initial source sample rate. 178 * sessionId: Specific session ID, or zero to use default. 179 * transferType: How data is transferred to AudioTrack. 180 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 181 */ 182 183 AudioTrack( audio_stream_type_t streamType, 184 uint32_t sampleRate, 185 audio_format_t format, 186 audio_channel_mask_t channelMask, 187 size_t frameCount = 0, 188 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 189 callback_t cbf = NULL, 190 void* user = NULL, 191 uint32_t notificationFrames = 0, 192 int sessionId = AUDIO_SESSION_ALLOCATE, 193 transfer_type transferType = TRANSFER_DEFAULT, 194 const audio_offload_info_t *offloadInfo = NULL, 195 int uid = -1, 196 pid_t pid = -1, 197 const audio_attributes_t* pAttributes = NULL); 198 199 /* Creates an audio track and registers it with AudioFlinger. 200 * With this constructor, the track is configured for static buffer mode. 201 * Data to be rendered is passed in a shared memory buffer 202 * identified by the argument sharedBuffer, which must be non-0. 203 * The memory should be initialized to the desired data before calling start(). 204 * The write() method is not supported in this case. 205 * It is recommended to pass a callback function to be notified of playback end by an 206 * EVENT_UNDERRUN event. 207 */ 208 209 AudioTrack( audio_stream_type_t streamType, 210 uint32_t sampleRate, 211 audio_format_t format, 212 audio_channel_mask_t channelMask, 213 const sp<IMemory>& sharedBuffer, 214 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 215 callback_t cbf = NULL, 216 void* user = NULL, 217 uint32_t notificationFrames = 0, 218 int sessionId = AUDIO_SESSION_ALLOCATE, 219 transfer_type transferType = TRANSFER_DEFAULT, 220 const audio_offload_info_t *offloadInfo = NULL, 221 int uid = -1, 222 pid_t pid = -1, 223 const audio_attributes_t* pAttributes = NULL); 224 225 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 226 * Also destroys all resources associated with the AudioTrack. 227 */ 228protected: 229 virtual ~AudioTrack(); 230public: 231 232 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 233 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 234 * Returned status (from utils/Errors.h) can be: 235 * - NO_ERROR: successful initialization 236 * - INVALID_OPERATION: AudioTrack is already initialized 237 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 238 * - NO_INIT: audio server or audio hardware not initialized 239 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 240 * If sharedBuffer is non-0, the frameCount parameter is ignored and 241 * replaced by the shared buffer's total allocated size in frame units. 242 * 243 * Parameters not listed in the AudioTrack constructors above: 244 * 245 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 246 * 247 * Internal state post condition: 248 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 249 */ 250 status_t set(audio_stream_type_t streamType, 251 uint32_t sampleRate, 252 audio_format_t format, 253 audio_channel_mask_t channelMask, 254 size_t frameCount = 0, 255 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 256 callback_t cbf = NULL, 257 void* user = NULL, 258 uint32_t notificationFrames = 0, 259 const sp<IMemory>& sharedBuffer = 0, 260 bool threadCanCallJava = false, 261 int sessionId = AUDIO_SESSION_ALLOCATE, 262 transfer_type transferType = TRANSFER_DEFAULT, 263 const audio_offload_info_t *offloadInfo = NULL, 264 int uid = -1, 265 pid_t pid = -1, 266 const audio_attributes_t* pAttributes = NULL); 267 268 /* Result of constructing the AudioTrack. This must be checked for successful initialization 269 * before using any AudioTrack API (except for set()), because using 270 * an uninitialized AudioTrack produces undefined results. 271 * See set() method above for possible return codes. 272 */ 273 status_t initCheck() const { return mStatus; } 274 275 /* Returns this track's estimated latency in milliseconds. 276 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 277 * and audio hardware driver. 278 */ 279 uint32_t latency() const { return mLatency; } 280 281 /* getters, see constructors and set() */ 282 283 audio_stream_type_t streamType() const; 284 audio_format_t format() const { return mFormat; } 285 286 /* Return frame size in bytes, which for linear PCM is 287 * channelCount * (bit depth per channel / 8). 288 * channelCount is determined from channelMask, and bit depth comes from format. 289 * For non-linear formats, the frame size is typically 1 byte. 290 */ 291 size_t frameSize() const { return mFrameSize; } 292 293 uint32_t channelCount() const { return mChannelCount; } 294 size_t frameCount() const { return mFrameCount; } 295 296 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 297 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 298 299 /* After it's created the track is not active. Call start() to 300 * make it active. If set, the callback will start being called. 301 * If the track was previously paused, volume is ramped up over the first mix buffer. 302 */ 303 status_t start(); 304 305 /* Stop a track. 306 * In static buffer mode, the track is stopped immediately. 307 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 308 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 309 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 310 * is first drained, mixed, and output, and only then is the track marked as stopped. 311 */ 312 void stop(); 313 bool stopped() const; 314 315 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 316 * This has the effect of draining the buffers without mixing or output. 317 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 318 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 319 */ 320 void flush(); 321 322 /* Pause a track. After pause, the callback will cease being called and 323 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 324 * and will fill up buffers until the pool is exhausted. 325 * Volume is ramped down over the next mix buffer following the pause request, 326 * and then the track is marked as paused. It can be resumed with ramp up by start(). 327 */ 328 void pause(); 329 330 /* Set volume for this track, mostly used for games' sound effects 331 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 332 * This is the older API. New applications should use setVolume(float) when possible. 333 */ 334 status_t setVolume(float left, float right); 335 336 /* Set volume for all channels. This is the preferred API for new applications, 337 * especially for multi-channel content. 338 */ 339 status_t setVolume(float volume); 340 341 /* Set the send level for this track. An auxiliary effect should be attached 342 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 343 */ 344 status_t setAuxEffectSendLevel(float level); 345 void getAuxEffectSendLevel(float* level) const; 346 347 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 348 */ 349 status_t setSampleRate(uint32_t sampleRate); 350 351 /* Return current source sample rate in Hz */ 352 uint32_t getSampleRate() const; 353 354 /* Enables looping and sets the start and end points of looping. 355 * Only supported for static buffer mode. 356 * 357 * Parameters: 358 * 359 * loopStart: loop start in frames relative to start of buffer. 360 * loopEnd: loop end in frames relative to start of buffer. 361 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 362 * pending or active loop. loopCount == -1 means infinite looping. 363 * 364 * For proper operation the following condition must be respected: 365 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 366 * 367 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 368 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 369 * 370 */ 371 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 372 373 /* Sets marker position. When playback reaches the number of frames specified, a callback with 374 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 375 * notification callback. To set a marker at a position which would compute as 0, 376 * a workaround is to set the marker at a nearby position such as ~0 or 1. 377 * If the AudioTrack has been opened with no callback function associated, the operation will 378 * fail. 379 * 380 * Parameters: 381 * 382 * marker: marker position expressed in wrapping (overflow) frame units, 383 * like the return value of getPosition(). 384 * 385 * Returned status (from utils/Errors.h) can be: 386 * - NO_ERROR: successful operation 387 * - INVALID_OPERATION: the AudioTrack has no callback installed. 388 */ 389 status_t setMarkerPosition(uint32_t marker); 390 status_t getMarkerPosition(uint32_t *marker) const; 391 392 /* Sets position update period. Every time the number of frames specified has been played, 393 * a callback with event type EVENT_NEW_POS is called. 394 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 395 * callback. 396 * If the AudioTrack has been opened with no callback function associated, the operation will 397 * fail. 398 * Extremely small values may be rounded up to a value the implementation can support. 399 * 400 * Parameters: 401 * 402 * updatePeriod: position update notification period expressed in frames. 403 * 404 * Returned status (from utils/Errors.h) can be: 405 * - NO_ERROR: successful operation 406 * - INVALID_OPERATION: the AudioTrack has no callback installed. 407 */ 408 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 409 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 410 411 /* Sets playback head position. 412 * Only supported for static buffer mode. 413 * 414 * Parameters: 415 * 416 * position: New playback head position in frames relative to start of buffer. 417 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 418 * but will result in an immediate underrun if started. 419 * 420 * Returned status (from utils/Errors.h) can be: 421 * - NO_ERROR: successful operation 422 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 423 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 424 * buffer 425 */ 426 status_t setPosition(uint32_t position); 427 428 /* Return the total number of frames played since playback start. 429 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 430 * It is reset to zero by flush(), reload(), and stop(). 431 * 432 * Parameters: 433 * 434 * position: Address where to return play head position. 435 * 436 * Returned status (from utils/Errors.h) can be: 437 * - NO_ERROR: successful operation 438 * - BAD_VALUE: position is NULL 439 */ 440 status_t getPosition(uint32_t *position); 441 442 /* For static buffer mode only, this returns the current playback position in frames 443 * relative to start of buffer. It is analogous to the position units used by 444 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 445 */ 446 status_t getBufferPosition(uint32_t *position); 447 448 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 449 * rewriting the buffer before restarting playback after a stop. 450 * This method must be called with the AudioTrack in paused or stopped state. 451 * Not allowed in streaming mode. 452 * 453 * Returned status (from utils/Errors.h) can be: 454 * - NO_ERROR: successful operation 455 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 456 */ 457 status_t reload(); 458 459 /* Returns a handle on the audio output used by this AudioTrack. 460 * 461 * Parameters: 462 * none. 463 * 464 * Returned value: 465 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 466 * track needed to be re-created but that failed 467 */ 468 audio_io_handle_t getOutput() const; 469 470 /* Returns the unique session ID associated with this track. 471 * 472 * Parameters: 473 * none. 474 * 475 * Returned value: 476 * AudioTrack session ID. 477 */ 478 int getSessionId() const { return mSessionId; } 479 480 /* Attach track auxiliary output to specified effect. Use effectId = 0 481 * to detach track from effect. 482 * 483 * Parameters: 484 * 485 * effectId: effectId obtained from AudioEffect::id(). 486 * 487 * Returned status (from utils/Errors.h) can be: 488 * - NO_ERROR: successful operation 489 * - INVALID_OPERATION: the effect is not an auxiliary effect. 490 * - BAD_VALUE: The specified effect ID is invalid 491 */ 492 status_t attachAuxEffect(int effectId); 493 494 /* Public API for TRANSFER_OBTAIN mode. 495 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 496 * After filling these slots with data, the caller should release them with releaseBuffer(). 497 * If the track buffer is not full, obtainBuffer() returns as many contiguous 498 * [empty slots for] frames as are available immediately. 499 * 500 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 501 * additional non-contiguous frames that are predicted to be available immediately, 502 * if the client were to release the first frames and then call obtainBuffer() again. 503 * This value is only a prediction, and needs to be confirmed. 504 * It will be set to zero for an error return. 505 * 506 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 507 * regardless of the value of waitCount. 508 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 509 * maximum timeout based on waitCount; see chart below. 510 * Buffers will be returned until the pool 511 * is exhausted, at which point obtainBuffer() will either block 512 * or return WOULD_BLOCK depending on the value of the "waitCount" 513 * parameter. 514 * 515 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 516 * which should use write() or callback EVENT_MORE_DATA instead. 517 * 518 * Interpretation of waitCount: 519 * +n limits wait time to n * WAIT_PERIOD_MS, 520 * -1 causes an (almost) infinite wait time, 521 * 0 non-blocking. 522 * 523 * Buffer fields 524 * On entry: 525 * frameCount number of [empty slots for] frames requested 526 * size ignored 527 * raw ignored 528 * After error return: 529 * frameCount 0 530 * size 0 531 * raw undefined 532 * After successful return: 533 * frameCount actual number of [empty slots for] frames available, <= number requested 534 * size actual number of bytes available 535 * raw pointer to the buffer 536 */ 537 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 538 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 539 size_t *nonContig = NULL) 540 __attribute__((__deprecated__)); 541 542private: 543 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 544 * additional non-contiguous frames that are predicted to be available immediately, 545 * if the client were to release the first frames and then call obtainBuffer() again. 546 * This value is only a prediction, and needs to be confirmed. 547 * It will be set to zero for an error return. 548 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 549 * in case the requested amount of frames is in two or more non-contiguous regions. 550 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 551 */ 552 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 553 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 554public: 555 556 /* Public API for TRANSFER_OBTAIN mode. 557 * Release a filled buffer of frames for AudioFlinger to process. 558 * 559 * Buffer fields: 560 * frameCount currently ignored but recommend to set to actual number of frames filled 561 * size actual number of bytes filled, must be multiple of frameSize 562 * raw ignored 563 * 564 */ 565 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 566 void releaseBuffer(const Buffer* audioBuffer); 567 568 /* As a convenience we provide a write() interface to the audio buffer. 569 * Input parameter 'size' is in byte units. 570 * This is implemented on top of obtainBuffer/releaseBuffer. For best 571 * performance use callbacks. Returns actual number of bytes written >= 0, 572 * or one of the following negative status codes: 573 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 574 * BAD_VALUE size is invalid 575 * WOULD_BLOCK when obtainBuffer() returns same, or 576 * AudioTrack was stopped during the write 577 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 578 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 579 * false for the method to return immediately without waiting to try multiple times to write 580 * the full content of the buffer. 581 */ 582 ssize_t write(const void* buffer, size_t size, bool blocking = true); 583 584 /* 585 * Dumps the state of an audio track. 586 */ 587 status_t dump(int fd, const Vector<String16>& args) const; 588 589 /* 590 * Return the total number of frames which AudioFlinger desired but were unavailable, 591 * and thus which resulted in an underrun. Reset to zero by stop(). 592 */ 593 uint32_t getUnderrunFrames() const; 594 595 /* Get the flags */ 596 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 597 598 /* Set parameters - only possible when using direct output */ 599 status_t setParameters(const String8& keyValuePairs); 600 601 /* Get parameters */ 602 String8 getParameters(const String8& keys); 603 604 /* Poll for a timestamp on demand. 605 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 606 * or if you need to get the most recent timestamp outside of the event callback handler. 607 * Caution: calling this method too often may be inefficient; 608 * if you need a high resolution mapping between frame position and presentation time, 609 * consider implementing that at application level, based on the low resolution timestamps. 610 * Returns NO_ERROR if timestamp is valid. 611 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 612 * start/ACTIVE, when the number of frames consumed is less than the 613 * overall hardware latency to physical output. In WOULD_BLOCK cases, 614 * one might poll again, or use getPosition(), or use 0 position and 615 * current time for the timestamp. 616 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 617 * 618 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 619 */ 620 status_t getTimestamp(AudioTimestamp& timestamp); 621 622protected: 623 /* copying audio tracks is not allowed */ 624 AudioTrack(const AudioTrack& other); 625 AudioTrack& operator = (const AudioTrack& other); 626 627 void setAttributesFromStreamType(audio_stream_type_t streamType); 628 629 /* a small internal class to handle the callback */ 630 class AudioTrackThread : public Thread 631 { 632 public: 633 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 634 635 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 636 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 637 virtual void requestExit(); 638 639 void pause(); // suspend thread from execution at next loop boundary 640 void resume(); // allow thread to execute, if not requested to exit 641 void wake(); // wake to handle changed notification conditions. 642 643 private: 644 void pauseInternal(nsecs_t ns = 0LL); 645 // like pause(), but only used internally within thread 646 647 friend class AudioTrack; 648 virtual bool threadLoop(); 649 AudioTrack& mReceiver; 650 virtual ~AudioTrackThread(); 651 Mutex mMyLock; // Thread::mLock is private 652 Condition mMyCond; // Thread::mThreadExitedCondition is private 653 bool mPaused; // whether thread is requested to pause at next loop entry 654 bool mPausedInt; // whether thread internally requests pause 655 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 656 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 657 // to processAudioBuffer() as state may have changed 658 // since pause time calculated. 659 }; 660 661 // body of AudioTrackThread::threadLoop() 662 // returns the maximum amount of time before we would like to run again, where: 663 // 0 immediately 664 // > 0 no later than this many nanoseconds from now 665 // NS_WHENEVER still active but no particular deadline 666 // NS_INACTIVE inactive so don't run again until re-started 667 // NS_NEVER never again 668 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 669 nsecs_t processAudioBuffer(); 670 671 bool isOffloaded() const; 672 bool isDirect() const; 673 bool isOffloadedOrDirect() const; 674 675 // caller must hold lock on mLock for all _l methods 676 677 status_t createTrack_l(); 678 679 // can only be called when mState != STATE_ACTIVE 680 void flush_l(); 681 682 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 683 684 // FIXME enum is faster than strcmp() for parameter 'from' 685 status_t restoreTrack_l(const char *from); 686 687 bool isOffloaded_l() const 688 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 689 690 bool isOffloadedOrDirect_l() const 691 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 692 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 693 694 bool isDirect_l() const 695 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 696 697 // increment mPosition by the delta of mServer, and return new value of mPosition 698 uint32_t updateAndGetPosition_l(); 699 700 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 701 sp<IAudioTrack> mAudioTrack; 702 sp<IMemory> mCblkMemory; 703 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 704 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 705 706 sp<AudioTrackThread> mAudioTrackThread; 707 708 float mVolume[2]; 709 float mSendLevel; 710 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 711 size_t mFrameCount; // corresponds to current IAudioTrack, value is 712 // reported back by AudioFlinger to the client 713 size_t mReqFrameCount; // frame count to request the first or next time 714 // a new IAudioTrack is needed, non-decreasing 715 716 // constant after constructor or set() 717 audio_format_t mFormat; // as requested by client, not forced to 16-bit 718 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 719 // this AudioTrack has valid attributes 720 uint32_t mChannelCount; 721 audio_channel_mask_t mChannelMask; 722 sp<IMemory> mSharedBuffer; 723 transfer_type mTransfer; 724 audio_offload_info_t mOffloadInfoCopy; 725 const audio_offload_info_t* mOffloadInfo; 726 audio_attributes_t mAttributes; 727 728 size_t mFrameSize; // frame size in bytes 729 730 status_t mStatus; 731 732 // can change dynamically when IAudioTrack invalidated 733 uint32_t mLatency; // in ms 734 735 // Indicates the current track state. Protected by mLock. 736 enum State { 737 STATE_ACTIVE, 738 STATE_STOPPED, 739 STATE_PAUSED, 740 STATE_PAUSED_STOPPING, 741 STATE_FLUSHED, 742 STATE_STOPPING, 743 } mState; 744 745 // for client callback handler 746 callback_t mCbf; // callback handler for events, or NULL 747 void* mUserData; 748 749 // for notification APIs 750 uint32_t mNotificationFramesReq; // requested number of frames between each 751 // notification callback, 752 // at initial source sample rate 753 uint32_t mNotificationFramesAct; // actual number of frames between each 754 // notification callback, 755 // at initial source sample rate 756 bool mRefreshRemaining; // processAudioBuffer() should refresh 757 // mRemainingFrames and mRetryOnPartialBuffer 758 759 // used for static track cbf and restoration 760 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 761 uint32_t mLoopStart; // last setLoop loopStart 762 uint32_t mLoopEnd; // last setLoop loopEnd 763 int32_t mLoopCountNotified; // the last loopCount notified by callback. 764 // mLoopCountNotified counts down, matching 765 // the remaining loop count for static track 766 // playback. 767 768 // These are private to processAudioBuffer(), and are not protected by a lock 769 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 770 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 771 uint32_t mObservedSequence; // last observed value of mSequence 772 773 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 774 bool mMarkerReached; 775 uint32_t mNewPosition; // in frames 776 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 777 uint32_t mServer; // in frames, last known mProxy->getPosition() 778 // which is count of frames consumed by server, 779 // reset by new IAudioTrack, 780 // whether it is reset by stop() is TBD 781 uint32_t mPosition; // in frames, like mServer except continues 782 // monotonically after new IAudioTrack, 783 // and could be easily widened to uint64_t 784 uint32_t mReleased; // in frames, count of frames released to server 785 // but not necessarily consumed by server, 786 // reset by stop() but continues monotonically 787 // after new IAudioTrack to restore mPosition, 788 // and could be easily widened to uint64_t 789 int64_t mStartUs; // the start time after flush or stop. 790 // only used for offloaded and direct tracks. 791 792 audio_output_flags_t mFlags; 793 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 794 // mLock must be held to read or write those bits reliably. 795 796 int mSessionId; 797 int mAuxEffectId; 798 799 mutable Mutex mLock; 800 801 bool mIsTimed; 802 int mPreviousPriority; // before start() 803 SchedPolicy mPreviousSchedulingGroup; 804 bool mAwaitBoost; // thread should wait for priority boost before running 805 806 // The proxy should only be referenced while a lock is held because the proxy isn't 807 // multi-thread safe, especially the SingleStateQueue part of the proxy. 808 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 809 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 810 // them around in case they are replaced during the obtainBuffer(). 811 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 812 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 813 814 bool mInUnderrun; // whether track is currently in underrun state 815 uint32_t mPausedPosition; 816 817private: 818 class DeathNotifier : public IBinder::DeathRecipient { 819 public: 820 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 821 protected: 822 virtual void binderDied(const wp<IBinder>& who); 823 private: 824 const wp<AudioTrack> mAudioTrack; 825 }; 826 827 sp<DeathNotifier> mDeathNotifier; 828 uint32_t mSequence; // incremented for each new IAudioTrack attempt 829 int mClientUid; 830 pid_t mClientPid; 831}; 832 833class TimedAudioTrack : public AudioTrack 834{ 835public: 836 TimedAudioTrack(); 837 838 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 839 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 840 841 /* queue a buffer obtained via allocateTimedBuffer for playback at the 842 given timestamp. PTS units are microseconds on the media time timeline. 843 The media time transform (set with setMediaTimeTransform) set by the 844 audio producer will handle converting from media time to local time 845 (perhaps going through the common time timeline in the case of 846 synchronized multiroom audio case) */ 847 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 848 849 /* define a transform between media time and either common time or 850 local time */ 851 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 852 status_t setMediaTimeTransform(const LinearTransform& xform, 853 TargetTimeline target); 854}; 855 856}; // namespace android 857 858#endif // ANDROID_AUDIOTRACK_H 859