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 26 namespace android { 27 28 // ---------------------------------------------------------------------------- 29 30 struct audio_track_cblk_t; 31 class AudioTrackClientProxy; 32 class StaticAudioTrackClientProxy; 33 34 // ---------------------------------------------------------------------------- 35 36 class AudioTrack : public RefBase 37 { 38 public: 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 const audio_attributes_t* pAttributes = NULL); 193 194 /* Creates an audio track and registers it with AudioFlinger. 195 * With this constructor, the track is configured for static buffer mode. 196 * The format must not be 8-bit linear PCM. 197 * Data to be rendered is passed in a shared memory buffer 198 * identified by the argument sharedBuffer, which must be non-0. 199 * The memory should be initialized to the desired data before calling start(). 200 * The write() method is not supported in this case. 201 * It is recommended to pass a callback function to be notified of playback end by an 202 * EVENT_UNDERRUN event. 203 */ 204 205 AudioTrack( audio_stream_type_t streamType, 206 uint32_t sampleRate, 207 audio_format_t format, 208 audio_channel_mask_t channelMask, 209 const sp<IMemory>& sharedBuffer, 210 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 211 callback_t cbf = NULL, 212 void* user = NULL, 213 uint32_t notificationFrames = 0, 214 int sessionId = AUDIO_SESSION_ALLOCATE, 215 transfer_type transferType = TRANSFER_DEFAULT, 216 const audio_offload_info_t *offloadInfo = NULL, 217 int uid = -1, 218 pid_t pid = -1, 219 const audio_attributes_t* pAttributes = NULL); 220 221 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 222 * Also destroys all resources associated with the AudioTrack. 223 */ 224 protected: 225 virtual ~AudioTrack(); 226 public: 227 228 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 229 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 230 * Returned status (from utils/Errors.h) can be: 231 * - NO_ERROR: successful initialization 232 * - INVALID_OPERATION: AudioTrack is already initialized 233 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 234 * - NO_INIT: audio server or audio hardware not initialized 235 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 236 * If sharedBuffer is non-0, the frameCount parameter is ignored and 237 * replaced by the shared buffer's total allocated size in frame units. 238 * 239 * Parameters not listed in the AudioTrack constructors above: 240 * 241 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 242 */ 243 status_t set(audio_stream_type_t streamType, 244 uint32_t sampleRate, 245 audio_format_t format, 246 audio_channel_mask_t channelMask, 247 size_t frameCount = 0, 248 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 249 callback_t cbf = NULL, 250 void* user = NULL, 251 uint32_t notificationFrames = 0, 252 const sp<IMemory>& sharedBuffer = 0, 253 bool threadCanCallJava = false, 254 int sessionId = AUDIO_SESSION_ALLOCATE, 255 transfer_type transferType = TRANSFER_DEFAULT, 256 const audio_offload_info_t *offloadInfo = NULL, 257 int uid = -1, 258 pid_t pid = -1, 259 const audio_attributes_t* pAttributes = NULL); 260 261 /* Result of constructing the AudioTrack. This must be checked for successful initialization 262 * before using any AudioTrack API (except for set()), because using 263 * an uninitialized AudioTrack produces undefined results. 264 * See set() method above for possible return codes. 265 */ 266 status_t initCheck() const { return mStatus; } 267 268 /* Returns this track's estimated latency in milliseconds. 269 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 270 * and audio hardware driver. 271 */ 272 uint32_t latency() const { return mLatency; } 273 274 /* getters, see constructors and set() */ 275 276 audio_stream_type_t streamType() const { return mStreamType; } 277 audio_format_t format() const { return mFormat; } 278 279 /* Return frame size in bytes, which for linear PCM is 280 * channelCount * (bit depth per channel / 8). 281 * channelCount is determined from channelMask, and bit depth comes from format. 282 * For non-linear formats, the frame size is typically 1 byte. 283 */ 284 size_t frameSize() const { return mFrameSize; } 285 286 uint32_t channelCount() const { return mChannelCount; } 287 size_t frameCount() const { return mFrameCount; } 288 289 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 290 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 291 292 /* After it's created the track is not active. Call start() to 293 * make it active. If set, the callback will start being called. 294 * If the track was previously paused, volume is ramped up over the first mix buffer. 295 */ 296 status_t start(); 297 298 /* Stop a track. 299 * In static buffer mode, the track is stopped immediately. 300 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 301 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 302 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 303 * is first drained, mixed, and output, and only then is the track marked as stopped. 304 */ 305 void stop(); 306 bool stopped() const; 307 308 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 309 * This has the effect of draining the buffers without mixing or output. 310 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 311 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 312 */ 313 void flush(); 314 315 /* Pause a track. After pause, the callback will cease being called and 316 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 317 * and will fill up buffers until the pool is exhausted. 318 * Volume is ramped down over the next mix buffer following the pause request, 319 * and then the track is marked as paused. It can be resumed with ramp up by start(). 320 */ 321 void pause(); 322 323 /* Set volume for this track, mostly used for games' sound effects 324 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 325 * This is the older API. New applications should use setVolume(float) when possible. 326 */ 327 status_t setVolume(float left, float right); 328 329 /* Set volume for all channels. This is the preferred API for new applications, 330 * especially for multi-channel content. 331 */ 332 status_t setVolume(float volume); 333 334 /* Set the send level for this track. An auxiliary effect should be attached 335 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 336 */ 337 status_t setAuxEffectSendLevel(float level); 338 void getAuxEffectSendLevel(float* level) const; 339 340 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 341 */ 342 status_t setSampleRate(uint32_t sampleRate); 343 344 /* Return current source sample rate in Hz */ 345 uint32_t getSampleRate() const; 346 347 /* Enables looping and sets the start and end points of looping. 348 * Only supported for static buffer mode. 349 * 350 * Parameters: 351 * 352 * loopStart: loop start in frames relative to start of buffer. 353 * loopEnd: loop end in frames relative to start of buffer. 354 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 355 * pending or active loop. loopCount == -1 means infinite looping. 356 * 357 * For proper operation the following condition must be respected: 358 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 359 * 360 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 361 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 362 * 363 */ 364 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 365 366 /* Sets marker position. When playback reaches the number of frames specified, a callback with 367 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 368 * notification callback. To set a marker at a position which would compute as 0, 369 * a workaround is to set the marker at a nearby position such as ~0 or 1. 370 * If the AudioTrack has been opened with no callback function associated, the operation will 371 * fail. 372 * 373 * Parameters: 374 * 375 * marker: marker position expressed in wrapping (overflow) frame units, 376 * like the return value of getPosition(). 377 * 378 * Returned status (from utils/Errors.h) can be: 379 * - NO_ERROR: successful operation 380 * - INVALID_OPERATION: the AudioTrack has no callback installed. 381 */ 382 status_t setMarkerPosition(uint32_t marker); 383 status_t getMarkerPosition(uint32_t *marker) const; 384 385 /* Sets position update period. Every time the number of frames specified has been played, 386 * a callback with event type EVENT_NEW_POS is called. 387 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 388 * callback. 389 * If the AudioTrack has been opened with no callback function associated, the operation will 390 * fail. 391 * Extremely small values may be rounded up to a value the implementation can support. 392 * 393 * Parameters: 394 * 395 * updatePeriod: position update notification period expressed in frames. 396 * 397 * Returned status (from utils/Errors.h) can be: 398 * - NO_ERROR: successful operation 399 * - INVALID_OPERATION: the AudioTrack has no callback installed. 400 */ 401 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 402 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 403 404 /* Sets playback head position. 405 * Only supported for static buffer mode. 406 * 407 * Parameters: 408 * 409 * position: New playback head position in frames relative to start of buffer. 410 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 411 * but will result in an immediate underrun if started. 412 * 413 * Returned status (from utils/Errors.h) can be: 414 * - NO_ERROR: successful operation 415 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 416 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 417 * buffer 418 */ 419 status_t setPosition(uint32_t position); 420 421 /* Return the total number of frames played since playback start. 422 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 423 * It is reset to zero by flush(), reload(), and stop(). 424 * 425 * Parameters: 426 * 427 * position: Address where to return play head position. 428 * 429 * Returned status (from utils/Errors.h) can be: 430 * - NO_ERROR: successful operation 431 * - BAD_VALUE: position is NULL 432 */ 433 status_t getPosition(uint32_t *position); 434 435 /* For static buffer mode only, this returns the current playback position in frames 436 * relative to start of buffer. It is analogous to the position units used by 437 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 438 */ 439 status_t getBufferPosition(uint32_t *position); 440 441 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 442 * rewriting the buffer before restarting playback after a stop. 443 * This method must be called with the AudioTrack in paused or stopped state. 444 * Not allowed in streaming mode. 445 * 446 * Returned status (from utils/Errors.h) can be: 447 * - NO_ERROR: successful operation 448 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 449 */ 450 status_t reload(); 451 452 /* Returns a handle on the audio output used by this AudioTrack. 453 * 454 * Parameters: 455 * none. 456 * 457 * Returned value: 458 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 459 * track needed to be re-created but that failed 460 */ 461 audio_io_handle_t getOutput() const; 462 463 /* Returns the unique session ID associated with this track. 464 * 465 * Parameters: 466 * none. 467 * 468 * Returned value: 469 * AudioTrack session ID. 470 */ 471 int getSessionId() const { return mSessionId; } 472 473 /* Attach track auxiliary output to specified effect. Use effectId = 0 474 * to detach track from effect. 475 * 476 * Parameters: 477 * 478 * effectId: effectId obtained from AudioEffect::id(). 479 * 480 * Returned status (from utils/Errors.h) can be: 481 * - NO_ERROR: successful operation 482 * - INVALID_OPERATION: the effect is not an auxiliary effect. 483 * - BAD_VALUE: The specified effect ID is invalid 484 */ 485 status_t attachAuxEffect(int effectId); 486 487 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 488 * After filling these slots with data, the caller should release them with releaseBuffer(). 489 * If the track buffer is not full, obtainBuffer() returns as many contiguous 490 * [empty slots for] frames as are available immediately. 491 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 492 * regardless of the value of waitCount. 493 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 494 * maximum timeout based on waitCount; see chart below. 495 * Buffers will be returned until the pool 496 * is exhausted, at which point obtainBuffer() will either block 497 * or return WOULD_BLOCK depending on the value of the "waitCount" 498 * parameter. 499 * Each sample is 16-bit signed PCM. 500 * 501 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 502 * which should use write() or callback EVENT_MORE_DATA instead. 503 * 504 * Interpretation of waitCount: 505 * +n limits wait time to n * WAIT_PERIOD_MS, 506 * -1 causes an (almost) infinite wait time, 507 * 0 non-blocking. 508 * 509 * Buffer fields 510 * On entry: 511 * frameCount number of frames requested 512 * After error return: 513 * frameCount 0 514 * size 0 515 * raw undefined 516 * After successful return: 517 * frameCount actual number of frames available, <= number requested 518 * size actual number of bytes available 519 * raw pointer to the buffer 520 */ 521 522 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 523 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 524 __attribute__((__deprecated__)); 525 526 private: 527 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 528 * additional non-contiguous frames that are available immediately. 529 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 530 * in case the requested amount of frames is in two or more non-contiguous regions. 531 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 532 */ 533 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 534 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 535 public: 536 537 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 538 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 539 void releaseBuffer(Buffer* audioBuffer); 540 541 /* As a convenience we provide a write() interface to the audio buffer. 542 * Input parameter 'size' is in byte units. 543 * This is implemented on top of obtainBuffer/releaseBuffer. For best 544 * performance use callbacks. Returns actual number of bytes written >= 0, 545 * or one of the following negative status codes: 546 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 547 * BAD_VALUE size is invalid 548 * WOULD_BLOCK when obtainBuffer() returns same, or 549 * AudioTrack was stopped during the write 550 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 551 * Default behavior is to only return until all data has been transferred. Set 'blocking' to 552 * false for the method to return immediately without waiting to try multiple times to write 553 * the full content of the buffer. 554 */ 555 ssize_t write(const void* buffer, size_t size, bool blocking = true); 556 557 /* 558 * Dumps the state of an audio track. 559 */ 560 status_t dump(int fd, const Vector<String16>& args) const; 561 562 /* 563 * Return the total number of frames which AudioFlinger desired but were unavailable, 564 * and thus which resulted in an underrun. Reset to zero by stop(). 565 */ 566 uint32_t getUnderrunFrames() const; 567 568 /* Get the flags */ 569 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 570 571 /* Set parameters - only possible when using direct output */ 572 status_t setParameters(const String8& keyValuePairs); 573 574 /* Get parameters */ 575 String8 getParameters(const String8& keys); 576 577 /* Poll for a timestamp on demand. 578 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 579 * or if you need to get the most recent timestamp outside of the event callback handler. 580 * Caution: calling this method too often may be inefficient; 581 * if you need a high resolution mapping between frame position and presentation time, 582 * consider implementing that at application level, based on the low resolution timestamps. 583 * Returns NO_ERROR if timestamp is valid. 584 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 585 * start/ACTIVE, when the number of frames consumed is less than the 586 * overall hardware latency to physical output. In WOULD_BLOCK cases, 587 * one might poll again, or use getPosition(), or use 0 position and 588 * current time for the timestamp. 589 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 590 * 591 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 592 */ 593 status_t getTimestamp(AudioTimestamp& timestamp); 594 595 protected: 596 /* copying audio tracks is not allowed */ 597 AudioTrack(const AudioTrack& other); 598 AudioTrack& operator = (const AudioTrack& other); 599 600 void setAttributesFromStreamType(audio_stream_type_t streamType); 601 void setStreamTypeFromAttributes(audio_attributes_t& aa); 602 /* paa is guaranteed non-NULL */ 603 bool isValidAttributes(const audio_attributes_t *paa); 604 605 /* a small internal class to handle the callback */ 606 class AudioTrackThread : public Thread 607 { 608 public: 609 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 610 611 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 612 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 613 virtual void requestExit(); 614 615 void pause(); // suspend thread from execution at next loop boundary 616 void resume(); // allow thread to execute, if not requested to exit 617 618 private: 619 void pauseInternal(nsecs_t ns = 0LL); 620 // like pause(), but only used internally within thread 621 622 friend class AudioTrack; 623 virtual bool threadLoop(); 624 AudioTrack& mReceiver; 625 virtual ~AudioTrackThread(); 626 Mutex mMyLock; // Thread::mLock is private 627 Condition mMyCond; // Thread::mThreadExitedCondition is private 628 bool mPaused; // whether thread is requested to pause at next loop entry 629 bool mPausedInt; // whether thread internally requests pause 630 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 631 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 632 }; 633 634 // body of AudioTrackThread::threadLoop() 635 // returns the maximum amount of time before we would like to run again, where: 636 // 0 immediately 637 // > 0 no later than this many nanoseconds from now 638 // NS_WHENEVER still active but no particular deadline 639 // NS_INACTIVE inactive so don't run again until re-started 640 // NS_NEVER never again 641 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 642 nsecs_t processAudioBuffer(); 643 644 bool isOffloaded() const; 645 bool isDirect() const; 646 bool isOffloadedOrDirect() const; 647 648 // caller must hold lock on mLock for all _l methods 649 650 status_t createTrack_l(); 651 652 // can only be called when mState != STATE_ACTIVE 653 void flush_l(); 654 655 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 656 657 // FIXME enum is faster than strcmp() for parameter 'from' 658 status_t restoreTrack_l(const char *from); 659 660 bool isOffloaded_l() const 661 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 662 663 bool isOffloadedOrDirect_l() const 664 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 665 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 666 667 bool isDirect_l() const 668 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 669 670 // increment mPosition by the delta of mServer, and return new value of mPosition 671 uint32_t updateAndGetPosition_l(); 672 673 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 674 sp<IAudioTrack> mAudioTrack; 675 sp<IMemory> mCblkMemory; 676 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 677 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 678 679 sp<AudioTrackThread> mAudioTrackThread; 680 681 float mVolume[2]; 682 float mSendLevel; 683 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. 684 size_t mFrameCount; // corresponds to current IAudioTrack, value is 685 // reported back by AudioFlinger to the client 686 size_t mReqFrameCount; // frame count to request the first or next time 687 // a new IAudioTrack is needed, non-decreasing 688 689 // constant after constructor or set() 690 audio_format_t mFormat; // as requested by client, not forced to 16-bit 691 audio_stream_type_t mStreamType; 692 uint32_t mChannelCount; 693 audio_channel_mask_t mChannelMask; 694 sp<IMemory> mSharedBuffer; 695 transfer_type mTransfer; 696 audio_offload_info_t mOffloadInfoCopy; 697 const audio_offload_info_t* mOffloadInfo; 698 audio_attributes_t mAttributes; 699 700 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 701 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 702 size_t mFrameSize; // app-level frame size 703 size_t mFrameSizeAF; // AudioFlinger frame size 704 705 status_t mStatus; 706 707 // can change dynamically when IAudioTrack invalidated 708 uint32_t mLatency; // in ms 709 710 // Indicates the current track state. Protected by mLock. 711 enum State { 712 STATE_ACTIVE, 713 STATE_STOPPED, 714 STATE_PAUSED, 715 STATE_PAUSED_STOPPING, 716 STATE_FLUSHED, 717 STATE_STOPPING, 718 } mState; 719 720 // for client callback handler 721 callback_t mCbf; // callback handler for events, or NULL 722 void* mUserData; 723 724 // for notification APIs 725 uint32_t mNotificationFramesReq; // requested number of frames between each 726 // notification callback, 727 // at initial source sample rate 728 uint32_t mNotificationFramesAct; // actual number of frames between each 729 // notification callback, 730 // at initial source sample rate 731 bool mRefreshRemaining; // processAudioBuffer() should refresh 732 // mRemainingFrames and mRetryOnPartialBuffer 733 734 // These are private to processAudioBuffer(), and are not protected by a lock 735 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 736 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 737 uint32_t mObservedSequence; // last observed value of mSequence 738 739 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 740 741 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 742 bool mMarkerReached; 743 uint32_t mNewPosition; // in frames 744 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 745 uint32_t mServer; // in frames, last known mProxy->getPosition() 746 // which is count of frames consumed by server, 747 // reset by new IAudioTrack, 748 // whether it is reset by stop() is TBD 749 uint32_t mPosition; // in frames, like mServer except continues 750 // monotonically after new IAudioTrack, 751 // and could be easily widened to uint64_t 752 uint32_t mReleased; // in frames, count of frames released to server 753 // but not necessarily consumed by server, 754 // reset by stop() but continues monotonically 755 // after new IAudioTrack to restore mPosition, 756 // and could be easily widened to uint64_t 757 int64_t mStartUs; // the start time after flush or stop. 758 // only used for offloaded and direct tracks. 759 760 audio_output_flags_t mFlags; 761 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 762 // mLock must be held to read or write those bits reliably. 763 764 int mSessionId; 765 int mAuxEffectId; 766 767 mutable Mutex mLock; 768 769 bool mIsTimed; 770 int mPreviousPriority; // before start() 771 SchedPolicy mPreviousSchedulingGroup; 772 bool mAwaitBoost; // thread should wait for priority boost before running 773 774 // The proxy should only be referenced while a lock is held because the proxy isn't 775 // multi-thread safe, especially the SingleStateQueue part of the proxy. 776 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 777 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 778 // them around in case they are replaced during the obtainBuffer(). 779 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 780 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 781 782 bool mInUnderrun; // whether track is currently in underrun state 783 uint32_t mPausedPosition; 784 785 private: 786 class DeathNotifier : public IBinder::DeathRecipient { 787 public: 788 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 789 protected: 790 virtual void binderDied(const wp<IBinder>& who); 791 private: 792 const wp<AudioTrack> mAudioTrack; 793 }; 794 795 sp<DeathNotifier> mDeathNotifier; 796 uint32_t mSequence; // incremented for each new IAudioTrack attempt 797 int mClientUid; 798 pid_t mClientPid; 799 }; 800 801 class TimedAudioTrack : public AudioTrack 802 { 803 public: 804 TimedAudioTrack(); 805 806 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 807 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 808 809 /* queue a buffer obtained via allocateTimedBuffer for playback at the 810 given timestamp. PTS units are microseconds on the media time timeline. 811 The media time transform (set with setMediaTimeTransform) set by the 812 audio producer will handle converting from media time to local time 813 (perhaps going through the common time timeline in the case of 814 synchronized multiroom audio case) */ 815 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 816 817 /* define a transform between media time and either common time or 818 local time */ 819 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 820 status_t setMediaTimeTransform(const LinearTransform& xform, 821 TargetTimeline target); 822 }; 823 824 }; // namespace android 825 826 #endif // ANDROID_AUDIOTRACK_H 827
