AudioTrack.h revision faabb51ceef13bf1e3f692219ac410c1cd75d0de
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. For mixed tracks, any PCM format supported by server is OK 158 * or AUDIO_FORMAT_PCM_8_BIT which is handled on client side. For direct 159 * and offloaded tracks, the possible format(s) depends on the output sink. 160 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 161 * frameCount: Minimum size of track PCM buffer in frames. This defines the 162 * application's contribution to the 163 * latency of the track. The actual size selected by the AudioTrack could be 164 * larger if the requested size is not compatible with current audio HAL 165 * configuration. Zero means to use a default value. 166 * flags: See comments on audio_output_flags_t in <system/audio.h>. 167 * cbf: Callback function. If not null, this function is called periodically 168 * to provide new data and inform of marker, position updates, etc. 169 * user: Context for use by the callback receiver. 170 * notificationFrames: The callback function is called each time notificationFrames PCM 171 * frames have been consumed from track input buffer. 172 * This is expressed in units of frames at the initial source sample rate. 173 * sessionId: Specific session ID, or zero to use default. 174 * transferType: How data is transferred to AudioTrack. 175 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 176 */ 177 178 AudioTrack( audio_stream_type_t streamType, 179 uint32_t sampleRate, 180 audio_format_t format, 181 audio_channel_mask_t, 182 size_t frameCount = 0, 183 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 184 callback_t cbf = NULL, 185 void* user = NULL, 186 uint32_t notificationFrames = 0, 187 int sessionId = AUDIO_SESSION_ALLOCATE, 188 transfer_type transferType = TRANSFER_DEFAULT, 189 const audio_offload_info_t *offloadInfo = NULL, 190 int uid = -1, 191 pid_t pid = -1); 192 193 /* Creates an audio track and registers it with AudioFlinger. 194 * With this constructor, the track is configured for static buffer mode. 195 * The format must not be 8-bit linear PCM. 196 * Data to be rendered is passed in a shared memory buffer 197 * identified by the argument sharedBuffer, which must be non-0. 198 * The memory should be initialized to the desired data before calling start(). 199 * The write() method is not supported in this case. 200 * It is recommended to pass a callback function to be notified of playback end by an 201 * EVENT_UNDERRUN event. 202 */ 203 204 AudioTrack( audio_stream_type_t streamType, 205 uint32_t sampleRate, 206 audio_format_t format, 207 audio_channel_mask_t channelMask, 208 const sp<IMemory>& sharedBuffer, 209 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 210 callback_t cbf = NULL, 211 void* user = NULL, 212 uint32_t notificationFrames = 0, 213 int sessionId = AUDIO_SESSION_ALLOCATE, 214 transfer_type transferType = TRANSFER_DEFAULT, 215 const audio_offload_info_t *offloadInfo = NULL, 216 int uid = -1, 217 pid_t pid = -1); 218 219 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 220 * Also destroys all resources associated with the AudioTrack. 221 */ 222protected: 223 virtual ~AudioTrack(); 224public: 225 226 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 227 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 228 * Returned status (from utils/Errors.h) can be: 229 * - NO_ERROR: successful initialization 230 * - INVALID_OPERATION: AudioTrack is already initialized 231 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 232 * - NO_INIT: audio server or audio hardware not initialized 233 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 234 * If sharedBuffer is non-0, the frameCount parameter is ignored and 235 * replaced by the shared buffer's total allocated size in frame units. 236 * 237 * Parameters not listed in the AudioTrack constructors above: 238 * 239 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 240 */ 241 status_t set(audio_stream_type_t streamType, 242 uint32_t sampleRate, 243 audio_format_t format, 244 audio_channel_mask_t channelMask, 245 size_t frameCount = 0, 246 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 247 callback_t cbf = NULL, 248 void* user = NULL, 249 uint32_t notificationFrames = 0, 250 const sp<IMemory>& sharedBuffer = 0, 251 bool threadCanCallJava = false, 252 int sessionId = AUDIO_SESSION_ALLOCATE, 253 transfer_type transferType = TRANSFER_DEFAULT, 254 const audio_offload_info_t *offloadInfo = NULL, 255 int uid = -1, 256 pid_t pid = -1, 257 audio_attributes_t* pAttributes = NULL); 258 259 /* Result of constructing the AudioTrack. This must be checked for successful initialization 260 * before using any AudioTrack API (except for set()), because using 261 * an uninitialized AudioTrack produces undefined results. 262 * See set() method above for possible return codes. 263 */ 264 status_t initCheck() const { return mStatus; } 265 266 /* Returns this track's estimated latency in milliseconds. 267 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 268 * and audio hardware driver. 269 */ 270 uint32_t latency() const { return mLatency; } 271 272 /* getters, see constructors and set() */ 273 274 audio_stream_type_t streamType() const { return mStreamType; } 275 audio_format_t format() const { return mFormat; } 276 277 /* Return frame size in bytes, which for linear PCM is 278 * channelCount * (bit depth per channel / 8). 279 * channelCount is determined from channelMask, and bit depth comes from format. 280 * For non-linear formats, the frame size is typically 1 byte. 281 */ 282 size_t frameSize() const { return mFrameSize; } 283 284 uint32_t channelCount() const { return mChannelCount; } 285 size_t frameCount() const { return mFrameCount; } 286 287 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 288 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 289 290 /* After it's created the track is not active. Call start() to 291 * make it active. If set, the callback will start being called. 292 * If the track was previously paused, volume is ramped up over the first mix buffer. 293 */ 294 status_t start(); 295 296 /* Stop a track. 297 * In static buffer mode, the track is stopped immediately. 298 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 299 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 300 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 301 * is first drained, mixed, and output, and only then is the track marked as stopped. 302 */ 303 void stop(); 304 bool stopped() const; 305 306 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 307 * This has the effect of draining the buffers without mixing or output. 308 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 309 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 310 */ 311 void flush(); 312 313 /* Pause a track. After pause, the callback will cease being called and 314 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 315 * and will fill up buffers until the pool is exhausted. 316 * Volume is ramped down over the next mix buffer following the pause request, 317 * and then the track is marked as paused. It can be resumed with ramp up by start(). 318 */ 319 void pause(); 320 321 /* Set volume for this track, mostly used for games' sound effects 322 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 323 * This is the older API. New applications should use setVolume(float) when possible. 324 */ 325 status_t setVolume(float left, float right); 326 327 /* Set volume for all channels. This is the preferred API for new applications, 328 * especially for multi-channel content. 329 */ 330 status_t setVolume(float volume); 331 332 /* Set the send level for this track. An auxiliary effect should be attached 333 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 334 */ 335 status_t setAuxEffectSendLevel(float level); 336 void getAuxEffectSendLevel(float* level) const; 337 338 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 339 */ 340 status_t setSampleRate(uint32_t sampleRate); 341 342 /* Return current source sample rate in Hz */ 343 uint32_t getSampleRate() const; 344 345 /* Enables looping and sets the start and end points of looping. 346 * Only supported for static buffer mode. 347 * 348 * Parameters: 349 * 350 * loopStart: loop start in frames relative to start of buffer. 351 * loopEnd: loop end in frames relative to start of buffer. 352 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 353 * pending or active loop. loopCount == -1 means infinite looping. 354 * 355 * For proper operation the following condition must be respected: 356 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 357 * 358 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 359 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 360 * 361 */ 362 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 363 364 /* Sets marker position. When playback reaches the number of frames specified, a callback with 365 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 366 * notification callback. To set a marker at a position which would compute as 0, 367 * a workaround is to set the marker at a nearby position such as ~0 or 1. 368 * If the AudioTrack has been opened with no callback function associated, the operation will 369 * fail. 370 * 371 * Parameters: 372 * 373 * marker: marker position expressed in wrapping (overflow) frame units, 374 * like the return value of getPosition(). 375 * 376 * Returned status (from utils/Errors.h) can be: 377 * - NO_ERROR: successful operation 378 * - INVALID_OPERATION: the AudioTrack has no callback installed. 379 */ 380 status_t setMarkerPosition(uint32_t marker); 381 status_t getMarkerPosition(uint32_t *marker) const; 382 383 /* Sets position update period. Every time the number of frames specified has been played, 384 * a callback with event type EVENT_NEW_POS is called. 385 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 386 * callback. 387 * If the AudioTrack has been opened with no callback function associated, the operation will 388 * fail. 389 * Extremely small values may be rounded up to a value the implementation can support. 390 * 391 * Parameters: 392 * 393 * updatePeriod: position update notification period expressed in frames. 394 * 395 * Returned status (from utils/Errors.h) can be: 396 * - NO_ERROR: successful operation 397 * - INVALID_OPERATION: the AudioTrack has no callback installed. 398 */ 399 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 400 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 401 402 /* Sets playback head position. 403 * Only supported for static buffer mode. 404 * 405 * Parameters: 406 * 407 * position: New playback head position in frames relative to start of buffer. 408 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 409 * but will result in an immediate underrun if started. 410 * 411 * Returned status (from utils/Errors.h) can be: 412 * - NO_ERROR: successful operation 413 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 414 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 415 * buffer 416 */ 417 status_t setPosition(uint32_t position); 418 419 /* Return the total number of frames played since playback start. 420 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 421 * It is reset to zero by flush(), reload(), and stop(). 422 * 423 * Parameters: 424 * 425 * position: Address where to return play head position. 426 * 427 * Returned status (from utils/Errors.h) can be: 428 * - NO_ERROR: successful operation 429 * - BAD_VALUE: position is NULL 430 */ 431 status_t getPosition(uint32_t *position) const; 432 433 /* For static buffer mode only, this returns the current playback position in frames 434 * relative to start of buffer. It is analogous to the position units used by 435 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 436 */ 437 status_t getBufferPosition(uint32_t *position); 438 439 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 440 * rewriting the buffer before restarting playback after a stop. 441 * This method must be called with the AudioTrack in paused or stopped state. 442 * Not allowed in streaming mode. 443 * 444 * Returned status (from utils/Errors.h) can be: 445 * - NO_ERROR: successful operation 446 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 447 */ 448 status_t reload(); 449 450 /* Returns a handle on the audio output used by this AudioTrack. 451 * 452 * Parameters: 453 * none. 454 * 455 * Returned value: 456 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 457 * track needed to be re-created but that failed 458 */ 459 audio_io_handle_t getOutput() const; 460 461 /* Returns the unique session ID associated with this track. 462 * 463 * Parameters: 464 * none. 465 * 466 * Returned value: 467 * AudioTrack session ID. 468 */ 469 int getSessionId() const { return mSessionId; } 470 471 /* Attach track auxiliary output to specified effect. Use effectId = 0 472 * to detach track from effect. 473 * 474 * Parameters: 475 * 476 * effectId: effectId obtained from AudioEffect::id(). 477 * 478 * Returned status (from utils/Errors.h) can be: 479 * - NO_ERROR: successful operation 480 * - INVALID_OPERATION: the effect is not an auxiliary effect. 481 * - BAD_VALUE: The specified effect ID is invalid 482 */ 483 status_t attachAuxEffect(int effectId); 484 485 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 486 * After filling these slots with data, the caller should release them with releaseBuffer(). 487 * If the track buffer is not full, obtainBuffer() returns as many contiguous 488 * [empty slots for] frames as are available immediately. 489 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 490 * regardless of the value of waitCount. 491 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 492 * maximum timeout based on waitCount; see chart below. 493 * Buffers will be returned until the pool 494 * is exhausted, at which point obtainBuffer() will either block 495 * or return WOULD_BLOCK depending on the value of the "waitCount" 496 * parameter. 497 * Each sample is 16-bit signed PCM. 498 * 499 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 500 * which should use write() or callback EVENT_MORE_DATA instead. 501 * 502 * Interpretation of waitCount: 503 * +n limits wait time to n * WAIT_PERIOD_MS, 504 * -1 causes an (almost) infinite wait time, 505 * 0 non-blocking. 506 * 507 * Buffer fields 508 * On entry: 509 * frameCount number of frames requested 510 * After error return: 511 * frameCount 0 512 * size 0 513 * raw undefined 514 * After successful return: 515 * frameCount actual number of frames available, <= number requested 516 * size actual number of bytes available 517 * raw pointer to the buffer 518 */ 519 520 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 521 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 522 __attribute__((__deprecated__)); 523 524private: 525 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 526 * additional non-contiguous frames that are available immediately. 527 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 528 * in case the requested amount of frames is in two or more non-contiguous regions. 529 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 530 */ 531 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 532 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 533public: 534 535 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 536 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 537 void releaseBuffer(Buffer* audioBuffer); 538 539 /* As a convenience we provide a write() interface to the audio buffer. 540 * Input parameter 'size' is in byte units. 541 * This is implemented on top of obtainBuffer/releaseBuffer. For best 542 * performance use callbacks. Returns actual number of bytes written >= 0, 543 * or one of the following negative status codes: 544 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 545 * BAD_VALUE size is invalid 546 * WOULD_BLOCK when obtainBuffer() returns same, or 547 * AudioTrack was stopped during the write 548 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 549 * Default behavior is to only return until all data has been transferred. Set 'blocking' to 550 * false for the method to return immediately without waiting to try multiple times to write 551 * the full content of the buffer. 552 */ 553 ssize_t write(const void* buffer, size_t size, bool blocking = true); 554 555 /* 556 * Dumps the state of an audio track. 557 */ 558 status_t dump(int fd, const Vector<String16>& args) const; 559 560 /* 561 * Return the total number of frames which AudioFlinger desired but were unavailable, 562 * and thus which resulted in an underrun. Reset to zero by stop(). 563 */ 564 uint32_t getUnderrunFrames() const; 565 566 /* Get the flags */ 567 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 568 569 /* Set parameters - only possible when using direct output */ 570 status_t setParameters(const String8& keyValuePairs); 571 572 /* Get parameters */ 573 String8 getParameters(const String8& keys); 574 575 /* Poll for a timestamp on demand. 576 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 577 * or if you need to get the most recent timestamp outside of the event callback handler. 578 * Caution: calling this method too often may be inefficient; 579 * if you need a high resolution mapping between frame position and presentation time, 580 * consider implementing that at application level, based on the low resolution timestamps. 581 * Returns NO_ERROR if timestamp is valid. 582 */ 583 status_t getTimestamp(AudioTimestamp& timestamp); 584 585protected: 586 /* copying audio tracks is not allowed */ 587 AudioTrack(const AudioTrack& other); 588 AudioTrack& operator = (const AudioTrack& other); 589 590 void setAttributesFromStreamType(audio_stream_type_t streamType); 591 void setStreamTypeFromAttributes(audio_attributes_t& aa); 592 /* paa is guaranteed non-NULL */ 593 bool isValidAttributes(const audio_attributes_t *paa); 594 595 /* a small internal class to handle the callback */ 596 class AudioTrackThread : public Thread 597 { 598 public: 599 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 600 601 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 602 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 603 virtual void requestExit(); 604 605 void pause(); // suspend thread from execution at next loop boundary 606 void resume(); // allow thread to execute, if not requested to exit 607 608 private: 609 void pauseInternal(nsecs_t ns = 0LL); 610 // like pause(), but only used internally within thread 611 612 friend class AudioTrack; 613 virtual bool threadLoop(); 614 AudioTrack& mReceiver; 615 virtual ~AudioTrackThread(); 616 Mutex mMyLock; // Thread::mLock is private 617 Condition mMyCond; // Thread::mThreadExitedCondition is private 618 bool mPaused; // whether thread is requested to pause at next loop entry 619 bool mPausedInt; // whether thread internally requests pause 620 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 621 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 622 }; 623 624 // body of AudioTrackThread::threadLoop() 625 // returns the maximum amount of time before we would like to run again, where: 626 // 0 immediately 627 // > 0 no later than this many nanoseconds from now 628 // NS_WHENEVER still active but no particular deadline 629 // NS_INACTIVE inactive so don't run again until re-started 630 // NS_NEVER never again 631 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 632 nsecs_t processAudioBuffer(); 633 634 bool isOffloaded() const; 635 636 // caller must hold lock on mLock for all _l methods 637 638 status_t createTrack_l(size_t epoch); 639 640 // can only be called when mState != STATE_ACTIVE 641 void flush_l(); 642 643 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 644 645 // FIXME enum is faster than strcmp() for parameter 'from' 646 status_t restoreTrack_l(const char *from); 647 648 bool isOffloaded_l() const 649 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 650 651 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 652 sp<IAudioTrack> mAudioTrack; 653 sp<IMemory> mCblkMemory; 654 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 655 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 656 657 sp<AudioTrackThread> mAudioTrackThread; 658 659 float mVolume[2]; 660 float mSendLevel; 661 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. 662 size_t mFrameCount; // corresponds to current IAudioTrack, value is 663 // reported back by AudioFlinger to the client 664 size_t mReqFrameCount; // frame count to request the first or next time 665 // a new IAudioTrack is needed, non-decreasing 666 667 // constant after constructor or set() 668 audio_format_t mFormat; // as requested by client, not forced to 16-bit 669 audio_stream_type_t mStreamType; 670 uint32_t mChannelCount; 671 audio_channel_mask_t mChannelMask; 672 sp<IMemory> mSharedBuffer; 673 transfer_type mTransfer; 674 audio_offload_info_t mOffloadInfoCopy; 675 const audio_offload_info_t* mOffloadInfo; 676 audio_attributes_t mAttributes; 677 678 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 679 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 680 size_t mFrameSize; // app-level frame size 681 size_t mFrameSizeAF; // AudioFlinger frame size 682 683 status_t mStatus; 684 685 // can change dynamically when IAudioTrack invalidated 686 uint32_t mLatency; // in ms 687 688 // Indicates the current track state. Protected by mLock. 689 enum State { 690 STATE_ACTIVE, 691 STATE_STOPPED, 692 STATE_PAUSED, 693 STATE_PAUSED_STOPPING, 694 STATE_FLUSHED, 695 STATE_STOPPING, 696 } mState; 697 698 // for client callback handler 699 callback_t mCbf; // callback handler for events, or NULL 700 void* mUserData; 701 702 // for notification APIs 703 uint32_t mNotificationFramesReq; // requested number of frames between each 704 // notification callback, 705 // at initial source sample rate 706 uint32_t mNotificationFramesAct; // actual number of frames between each 707 // notification callback, 708 // at initial source sample rate 709 bool mRefreshRemaining; // processAudioBuffer() should refresh 710 // mRemainingFrames and mRetryOnPartialBuffer 711 712 // These are private to processAudioBuffer(), and are not protected by a lock 713 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 714 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 715 uint32_t mObservedSequence; // last observed value of mSequence 716 717 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 718 719 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 720 bool mMarkerReached; 721 uint32_t mNewPosition; // in frames 722 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 723 724 audio_output_flags_t mFlags; 725 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 726 // mLock must be held to read or write those bits reliably. 727 728 int mSessionId; 729 int mAuxEffectId; 730 731 mutable Mutex mLock; 732 733 bool mIsTimed; 734 int mPreviousPriority; // before start() 735 SchedPolicy mPreviousSchedulingGroup; 736 bool mAwaitBoost; // thread should wait for priority boost before running 737 738 // The proxy should only be referenced while a lock is held because the proxy isn't 739 // multi-thread safe, especially the SingleStateQueue part of the proxy. 740 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 741 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 742 // them around in case they are replaced during the obtainBuffer(). 743 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 744 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 745 746 bool mInUnderrun; // whether track is currently in underrun state 747 uint32_t mPausedPosition; 748 749private: 750 class DeathNotifier : public IBinder::DeathRecipient { 751 public: 752 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 753 protected: 754 virtual void binderDied(const wp<IBinder>& who); 755 private: 756 const wp<AudioTrack> mAudioTrack; 757 }; 758 759 sp<DeathNotifier> mDeathNotifier; 760 uint32_t mSequence; // incremented for each new IAudioTrack attempt 761 int mClientUid; 762 pid_t mClientPid; 763}; 764 765class TimedAudioTrack : public AudioTrack 766{ 767public: 768 TimedAudioTrack(); 769 770 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 771 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 772 773 /* queue a buffer obtained via allocateTimedBuffer for playback at the 774 given timestamp. PTS units are microseconds on the media time timeline. 775 The media time transform (set with setMediaTimeTransform) set by the 776 audio producer will handle converting from media time to local time 777 (perhaps going through the common time timeline in the case of 778 synchronized multiroom audio case) */ 779 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 780 781 /* define a transform between media time and either common time or 782 local time */ 783 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 784 status_t setMediaTimeTransform(const LinearTransform& xform, 785 TargetTimeline target); 786}; 787 788}; // namespace android 789 790#endif // ANDROID_AUDIOTRACK_H 791