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 class audio_track_cblk_t; 31 class AudioTrackClientProxy; 32 class StaticAudioTrackClientProxy; 33 34 // ---------------------------------------------------------------------------- 35 36 class AudioTrack : public RefBase 37 { 38 public: 39 enum channel_index { 40 MONO = 0, 41 LEFT = 0, 42 RIGHT = 1 43 }; 44 45 /* Events used by AudioTrack callback function (callback_t). 46 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 47 */ 48 enum event_type { 49 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 50 // If this event is delivered but the callback handler 51 // does not want to write more data, the handler must explicitly 52 // ignore the event by setting frameCount to zero. 53 EVENT_UNDERRUN = 1, // Buffer underrun occurred. 54 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 55 // loop start if loop count was not 0. 56 EVENT_MARKER = 3, // Playback head is at the specified marker position 57 // (See setMarkerPosition()). 58 EVENT_NEW_POS = 4, // Playback head is at a new position 59 // (See setPositionUpdatePeriod()). 60 EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer. 61 // Not currently used by android.media.AudioTrack. 62 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 63 // voluntary invalidation by mediaserver, or mediaserver crash. 64 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 65 // back (after stop is called) 66 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 67 // in the mapping from frame position to presentation time. 68 // See AudioTimestamp for the information included with event. 69 }; 70 71 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 72 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 73 */ 74 75 class Buffer 76 { 77 public: 78 // FIXME use m prefix 79 size_t frameCount; // number of sample frames corresponding to size; 80 // on input it is the number of frames desired, 81 // on output is the number of frames actually filled 82 // (currently ignored, but will make the primary field in future) 83 84 size_t size; // input/output in bytes == frameCount * frameSize 85 // on output is the number of bytes actually filled 86 // FIXME this is redundant with respect to frameCount, 87 // and TRANSFER_OBTAIN mode is broken for 8-bit data 88 // since we don't define the frame format 89 90 union { 91 void* raw; 92 short* i16; // signed 16-bit 93 int8_t* i8; // unsigned 8-bit, offset by 0x80 94 }; 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 */ 127 128 static status_t getMinFrameCount(size_t* frameCount, 129 audio_stream_type_t streamType, 130 uint32_t sampleRate); 131 132 /* How data is transferred to AudioTrack 133 */ 134 enum transfer_type { 135 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 136 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 137 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 138 TRANSFER_SYNC, // synchronous write() 139 TRANSFER_SHARED, // shared memory 140 }; 141 142 /* Constructs an uninitialized AudioTrack. No connection with 143 * AudioFlinger takes place. Use set() after this. 144 */ 145 AudioTrack(); 146 147 /* Creates an AudioTrack object and registers it with AudioFlinger. 148 * Once created, the track needs to be started before it can be used. 149 * Unspecified values are set to appropriate default values. 150 * With this constructor, the track is configured for streaming mode. 151 * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. 152 * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. 153 * 154 * Parameters: 155 * 156 * streamType: Select the type of audio stream this track is attached to 157 * (e.g. AUDIO_STREAM_MUSIC). 158 * sampleRate: Data source sampling rate in Hz. 159 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 160 * 16 bits per sample). 161 * channelMask: Channel mask. 162 * frameCount: Minimum size of track PCM buffer in frames. This defines the 163 * application's contribution to the 164 * latency of the track. The actual size selected by the AudioTrack could be 165 * larger if the requested size is not compatible with current audio HAL 166 * configuration. Zero means to use a default value. 167 * flags: See comments on audio_output_flags_t in <system/audio.h>. 168 * cbf: Callback function. If not null, this function is called periodically 169 * to provide new data and inform of marker, position updates, etc. 170 * user: Context for use by the callback receiver. 171 * notificationFrames: The callback function is called each time notificationFrames PCM 172 * frames have been consumed from track input buffer. 173 * This is expressed in units of frames at the initial source sample rate. 174 * sessionId: Specific session ID, or zero to use default. 175 * transferType: How data is transferred to AudioTrack. 176 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 177 */ 178 179 AudioTrack( audio_stream_type_t streamType, 180 uint32_t sampleRate, 181 audio_format_t format, 182 audio_channel_mask_t, 183 int frameCount = 0, 184 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 185 callback_t cbf = NULL, 186 void* user = NULL, 187 int notificationFrames = 0, 188 int sessionId = 0, 189 transfer_type transferType = TRANSFER_DEFAULT, 190 const audio_offload_info_t *offloadInfo = NULL, 191 int uid = -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 int notificationFrames = 0, 213 int sessionId = 0, 214 transfer_type transferType = TRANSFER_DEFAULT, 215 const audio_offload_info_t *offloadInfo = NULL, 216 int uid = -1); 217 218 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 219 * Also destroys all resources associated with the AudioTrack. 220 */ 221 protected: 222 virtual ~AudioTrack(); 223 public: 224 225 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 226 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 227 * Returned status (from utils/Errors.h) can be: 228 * - NO_ERROR: successful initialization 229 * - INVALID_OPERATION: AudioTrack is already initialized 230 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 231 * - NO_INIT: audio server or audio hardware not initialized 232 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 233 * If sharedBuffer is non-0, the frameCount parameter is ignored and 234 * replaced by the shared buffer's total allocated size in frame units. 235 * 236 * Parameters not listed in the AudioTrack constructors above: 237 * 238 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 239 */ 240 status_t set(audio_stream_type_t streamType, 241 uint32_t sampleRate, 242 audio_format_t format, 243 audio_channel_mask_t channelMask, 244 int frameCount = 0, 245 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 246 callback_t cbf = NULL, 247 void* user = NULL, 248 int notificationFrames = 0, 249 const sp<IMemory>& sharedBuffer = 0, 250 bool threadCanCallJava = false, 251 int sessionId = 0, 252 transfer_type transferType = TRANSFER_DEFAULT, 253 const audio_offload_info_t *offloadInfo = NULL, 254 int uid = -1); 255 256 /* Result of constructing the AudioTrack. This must be checked for successful initialization 257 * before using any AudioTrack API (except for set()), because using 258 * an uninitialized AudioTrack produces undefined results. 259 * See set() method above for possible return codes. 260 */ initCheck()261 status_t initCheck() const { return mStatus; } 262 263 /* Returns this track's estimated latency in milliseconds. 264 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 265 * and audio hardware driver. 266 */ latency()267 uint32_t latency() const { return mLatency; } 268 269 /* getters, see constructors and set() */ 270 streamType()271 audio_stream_type_t streamType() const { return mStreamType; } format()272 audio_format_t format() const { return mFormat; } 273 274 /* Return frame size in bytes, which for linear PCM is 275 * channelCount * (bit depth per channel / 8). 276 * channelCount is determined from channelMask, and bit depth comes from format. 277 * For non-linear formats, the frame size is typically 1 byte. 278 */ frameSize()279 size_t frameSize() const { return mFrameSize; } 280 channelCount()281 uint32_t channelCount() const { return mChannelCount; } frameCount()282 uint32_t frameCount() const { return mFrameCount; } 283 284 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ sharedBuffer()285 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 286 287 /* After it's created the track is not active. Call start() to 288 * make it active. If set, the callback will start being called. 289 * If the track was previously paused, volume is ramped up over the first mix buffer. 290 */ 291 status_t start(); 292 293 /* Stop a track. 294 * In static buffer mode, the track is stopped immediately. 295 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 296 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 297 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 298 * is first drained, mixed, and output, and only then is the track marked as stopped. 299 */ 300 void stop(); 301 bool stopped() const; 302 303 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 304 * This has the effect of draining the buffers without mixing or output. 305 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 306 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 307 */ 308 void flush(); 309 310 /* Pause a track. After pause, the callback will cease being called and 311 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 312 * and will fill up buffers until the pool is exhausted. 313 * Volume is ramped down over the next mix buffer following the pause request, 314 * and then the track is marked as paused. It can be resumed with ramp up by start(). 315 */ 316 void pause(); 317 318 /* Set volume for this track, mostly used for games' sound effects 319 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 320 * This is the older API. New applications should use setVolume(float) when possible. 321 */ 322 status_t setVolume(float left, float right); 323 324 /* Set volume for all channels. This is the preferred API for new applications, 325 * especially for multi-channel content. 326 */ 327 status_t setVolume(float volume); 328 329 /* Set the send level for this track. An auxiliary effect should be attached 330 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 331 */ 332 status_t setAuxEffectSendLevel(float level); 333 void getAuxEffectSendLevel(float* level) const; 334 335 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 336 */ 337 status_t setSampleRate(uint32_t sampleRate); 338 339 /* Return current source sample rate in Hz, or 0 if unknown */ 340 uint32_t getSampleRate() const; 341 342 /* Enables looping and sets the start and end points of looping. 343 * Only supported for static buffer mode. 344 * 345 * Parameters: 346 * 347 * loopStart: loop start in frames relative to start of buffer. 348 * loopEnd: loop end in frames relative to start of buffer. 349 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 350 * pending or active loop. loopCount == -1 means infinite looping. 351 * 352 * For proper operation the following condition must be respected: 353 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 354 * 355 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 356 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 357 * 358 */ 359 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 360 361 /* Sets marker position. When playback reaches the number of frames specified, a callback with 362 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 363 * notification callback. To set a marker at a position which would compute as 0, 364 * a workaround is to the set the marker at a nearby position such as ~0 or 1. 365 * If the AudioTrack has been opened with no callback function associated, the operation will 366 * fail. 367 * 368 * Parameters: 369 * 370 * marker: marker position expressed in wrapping (overflow) frame units, 371 * like the return value of getPosition(). 372 * 373 * Returned status (from utils/Errors.h) can be: 374 * - NO_ERROR: successful operation 375 * - INVALID_OPERATION: the AudioTrack has no callback installed. 376 */ 377 status_t setMarkerPosition(uint32_t marker); 378 status_t getMarkerPosition(uint32_t *marker) const; 379 380 /* Sets position update period. Every time the number of frames specified has been played, 381 * a callback with event type EVENT_NEW_POS is called. 382 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 383 * callback. 384 * If the AudioTrack has been opened with no callback function associated, the operation will 385 * fail. 386 * Extremely small values may be rounded up to a value the implementation can support. 387 * 388 * Parameters: 389 * 390 * updatePeriod: position update notification period expressed in frames. 391 * 392 * Returned status (from utils/Errors.h) can be: 393 * - NO_ERROR: successful operation 394 * - INVALID_OPERATION: the AudioTrack has no callback installed. 395 */ 396 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 397 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 398 399 /* Sets playback head position. 400 * Only supported for static buffer mode. 401 * 402 * Parameters: 403 * 404 * position: New playback head position in frames relative to start of buffer. 405 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 406 * but will result in an immediate underrun if started. 407 * 408 * Returned status (from utils/Errors.h) can be: 409 * - NO_ERROR: successful operation 410 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 411 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 412 * buffer 413 */ 414 status_t setPosition(uint32_t position); 415 416 /* Return the total number of frames played since playback start. 417 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 418 * It is reset to zero by flush(), reload(), and stop(). 419 * 420 * Parameters: 421 * 422 * position: Address where to return play head position. 423 * 424 * Returned status (from utils/Errors.h) can be: 425 * - NO_ERROR: successful operation 426 * - BAD_VALUE: position is NULL 427 */ 428 status_t getPosition(uint32_t *position) const; 429 430 /* For static buffer mode only, this returns the current playback position in frames 431 * relative to start of buffer. It is analogous to the position units used by 432 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 433 */ 434 status_t getBufferPosition(uint32_t *position); 435 436 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 437 * rewriting the buffer before restarting playback after a stop. 438 * This method must be called with the AudioTrack in paused or stopped state. 439 * Not allowed in streaming mode. 440 * 441 * Returned status (from utils/Errors.h) can be: 442 * - NO_ERROR: successful operation 443 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 444 */ 445 status_t reload(); 446 447 /* Returns a handle on the audio output used by this AudioTrack. 448 * 449 * Parameters: 450 * none. 451 * 452 * Returned value: 453 * handle on audio hardware output 454 */ 455 audio_io_handle_t getOutput(); 456 457 /* Returns the unique session ID associated with this track. 458 * 459 * Parameters: 460 * none. 461 * 462 * Returned value: 463 * AudioTrack session ID. 464 */ getSessionId()465 int getSessionId() const { return mSessionId; } 466 467 /* Attach track auxiliary output to specified effect. Use effectId = 0 468 * to detach track from effect. 469 * 470 * Parameters: 471 * 472 * effectId: effectId obtained from AudioEffect::id(). 473 * 474 * Returned status (from utils/Errors.h) can be: 475 * - NO_ERROR: successful operation 476 * - INVALID_OPERATION: the effect is not an auxiliary effect. 477 * - BAD_VALUE: The specified effect ID is invalid 478 */ 479 status_t attachAuxEffect(int effectId); 480 481 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 482 * After filling these slots with data, the caller should release them with releaseBuffer(). 483 * If the track buffer is not full, obtainBuffer() returns as many contiguous 484 * [empty slots for] frames as are available immediately. 485 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 486 * regardless of the value of waitCount. 487 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 488 * maximum timeout based on waitCount; see chart below. 489 * Buffers will be returned until the pool 490 * is exhausted, at which point obtainBuffer() will either block 491 * or return WOULD_BLOCK depending on the value of the "waitCount" 492 * parameter. 493 * Each sample is 16-bit signed PCM. 494 * 495 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 496 * which should use write() or callback EVENT_MORE_DATA instead. 497 * 498 * Interpretation of waitCount: 499 * +n limits wait time to n * WAIT_PERIOD_MS, 500 * -1 causes an (almost) infinite wait time, 501 * 0 non-blocking. 502 * 503 * Buffer fields 504 * On entry: 505 * frameCount number of frames requested 506 * After error return: 507 * frameCount 0 508 * size 0 509 * raw undefined 510 * After successful return: 511 * frameCount actual number of frames available, <= number requested 512 * size actual number of bytes available 513 * raw pointer to the buffer 514 */ 515 516 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 517 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 518 __attribute__((__deprecated__)); 519 520 private: 521 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 522 * additional non-contiguous frames that are available immediately. 523 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 524 * in case the requested amount of frames is in two or more non-contiguous regions. 525 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 526 */ 527 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 528 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 529 public: 530 531 //EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy 532 // enum { 533 // NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 534 // TEAR_DOWN = 0x80000002, 535 // STOPPED = 1, 536 // STREAM_END_WAIT, 537 // STREAM_END 538 // }; 539 540 /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ 541 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 542 void releaseBuffer(Buffer* audioBuffer); 543 544 /* As a convenience we provide a write() interface to the audio buffer. 545 * Input parameter 'size' is in byte units. 546 * This is implemented on top of obtainBuffer/releaseBuffer. For best 547 * performance use callbacks. Returns actual number of bytes written >= 0, 548 * or one of the following negative status codes: 549 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 550 * BAD_VALUE size is invalid 551 * WOULD_BLOCK when obtainBuffer() returns same, or 552 * AudioTrack was stopped during the write 553 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 554 */ 555 ssize_t write(const void* buffer, size_t size); 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 */ getFlags()569 audio_output_flags_t getFlags() const { 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 */ 585 status_t getTimestamp(AudioTimestamp& timestamp); 586 587 protected: 588 /* copying audio tracks is not allowed */ 589 AudioTrack(const AudioTrack& other); 590 AudioTrack& operator = (const AudioTrack& other); 591 592 /* a small internal class to handle the callback */ 593 class AudioTrackThread : public Thread 594 { 595 public: 596 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 597 598 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 599 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 600 virtual void requestExit(); 601 602 void pause(); // suspend thread from execution at next loop boundary 603 void resume(); // allow thread to execute, if not requested to exit 604 605 private: 606 void pauseInternal(nsecs_t ns = 0LL); 607 // like pause(), but only used internally within thread 608 609 friend class AudioTrack; 610 virtual bool threadLoop(); 611 AudioTrack& mReceiver; 612 virtual ~AudioTrackThread(); 613 Mutex mMyLock; // Thread::mLock is private 614 Condition mMyCond; // Thread::mThreadExitedCondition is private 615 bool mPaused; // whether thread is requested to pause at next loop entry 616 bool mPausedInt; // whether thread internally requests pause 617 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 618 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 619 }; 620 621 // body of AudioTrackThread::threadLoop() 622 // returns the maximum amount of time before we would like to run again, where: 623 // 0 immediately 624 // > 0 no later than this many nanoseconds from now 625 // NS_WHENEVER still active but no particular deadline 626 // NS_INACTIVE inactive so don't run again until re-started 627 // NS_NEVER never again 628 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 629 nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread); 630 status_t processStreamEnd(int32_t waitCount); 631 632 633 // caller must hold lock on mLock for all _l methods 634 635 status_t createTrack_l(audio_stream_type_t streamType, 636 uint32_t sampleRate, 637 audio_format_t format, 638 size_t frameCount, 639 audio_output_flags_t flags, 640 const sp<IMemory>& sharedBuffer, 641 audio_io_handle_t output, 642 size_t epoch); 643 644 // can only be called when mState != STATE_ACTIVE 645 void flush_l(); 646 647 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 648 audio_io_handle_t getOutput_l(); 649 650 // FIXME enum is faster than strcmp() for parameter 'from' 651 status_t restoreTrack_l(const char *from); 652 isOffloaded()653 bool isOffloaded() const 654 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 655 656 // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0 657 sp<IAudioTrack> mAudioTrack; 658 sp<IMemory> mCblkMemory; 659 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 660 661 sp<AudioTrackThread> mAudioTrackThread; 662 float mVolume[2]; 663 float mSendLevel; 664 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. 665 size_t mFrameCount; // corresponds to current IAudioTrack 666 size_t mReqFrameCount; // frame count to request the next time a new 667 // IAudioTrack is needed 668 669 670 // constant after constructor or set() 671 audio_format_t mFormat; // as requested by client, not forced to 16-bit 672 audio_stream_type_t mStreamType; 673 uint32_t mChannelCount; 674 audio_channel_mask_t mChannelMask; 675 transfer_type mTransfer; 676 677 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 678 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 679 size_t mFrameSize; // app-level frame size 680 size_t mFrameSizeAF; // AudioFlinger frame size 681 682 status_t mStatus; 683 684 // can change dynamically when IAudioTrack invalidated 685 uint32_t mLatency; // in ms 686 687 // Indicates the current track state. Protected by mLock. 688 enum State { 689 STATE_ACTIVE, 690 STATE_STOPPED, 691 STATE_PAUSED, 692 STATE_PAUSED_STOPPING, 693 STATE_FLUSHED, 694 STATE_STOPPING, 695 } mState; 696 697 // for client callback handler 698 callback_t mCbf; // callback handler for events, or NULL 699 void* mUserData; 700 701 // for notification APIs 702 uint32_t mNotificationFramesReq; // requested number of frames between each 703 // notification callback, 704 // at initial source sample rate 705 uint32_t mNotificationFramesAct; // actual number of frames between each 706 // notification callback, 707 // at initial source sample rate 708 bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 709 710 // These are private to processAudioBuffer(), and are not protected by a lock 711 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 712 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 713 uint32_t mObservedSequence; // last observed value of mSequence 714 715 sp<IMemory> mSharedBuffer; 716 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 717 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 718 bool mMarkerReached; 719 uint32_t mNewPosition; // in frames 720 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 721 722 audio_output_flags_t mFlags; 723 int mSessionId; 724 int mAuxEffectId; 725 726 mutable Mutex mLock; 727 728 bool mIsTimed; 729 int mPreviousPriority; // before start() 730 SchedPolicy mPreviousSchedulingGroup; 731 bool mAwaitBoost; // thread should wait for priority boost before running 732 733 // The proxy should only be referenced while a lock is held because the proxy isn't 734 // multi-thread safe, especially the SingleStateQueue part of the proxy. 735 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 736 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 737 // them around in case they are replaced during the obtainBuffer(). 738 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 739 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 740 741 bool mInUnderrun; // whether track is currently in underrun state 742 String8 mName; // server's name for this IAudioTrack 743 uint32_t mPausedPosition; 744 745 private: 746 class DeathNotifier : public IBinder::DeathRecipient { 747 public: DeathNotifier(AudioTrack * audioTrack)748 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 749 protected: 750 virtual void binderDied(const wp<IBinder>& who); 751 private: 752 const wp<AudioTrack> mAudioTrack; 753 }; 754 755 sp<DeathNotifier> mDeathNotifier; 756 uint32_t mSequence; // incremented for each new IAudioTrack attempt 757 audio_io_handle_t mOutput; // cached output io handle 758 int mClientUid; 759 }; 760 761 class TimedAudioTrack : public AudioTrack 762 { 763 public: 764 TimedAudioTrack(); 765 766 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 767 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 768 769 /* queue a buffer obtained via allocateTimedBuffer for playback at the 770 given timestamp. PTS units are microseconds on the media time timeline. 771 The media time transform (set with setMediaTimeTransform) set by the 772 audio producer will handle converting from media time to local time 773 (perhaps going through the common time timeline in the case of 774 synchronized multiroom audio case) */ 775 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 776 777 /* define a transform between media time and either common time or 778 local time */ 779 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 780 status_t setMediaTimeTransform(const LinearTransform& xform, 781 TargetTimeline target); 782 }; 783 784 }; // namespace android 785 786 #endif // ANDROID_AUDIOTRACK_H 787