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 * Internal state post condition: 244 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 245 */ 246 status_t set(audio_stream_type_t streamType, 247 uint32_t sampleRate, 248 audio_format_t format, 249 audio_channel_mask_t channelMask, 250 size_t frameCount = 0, 251 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 252 callback_t cbf = NULL, 253 void* user = NULL, 254 uint32_t notificationFrames = 0, 255 const sp<IMemory>& sharedBuffer = 0, 256 bool threadCanCallJava = false, 257 int sessionId = AUDIO_SESSION_ALLOCATE, 258 transfer_type transferType = TRANSFER_DEFAULT, 259 const audio_offload_info_t *offloadInfo = NULL, 260 int uid = -1, 261 pid_t pid = -1, 262 const audio_attributes_t* pAttributes = NULL); 263 264 /* Result of constructing the AudioTrack. This must be checked for successful initialization 265 * before using any AudioTrack API (except for set()), because using 266 * an uninitialized AudioTrack produces undefined results. 267 * See set() method above for possible return codes. 268 */ initCheck()269 status_t initCheck() const { return mStatus; } 270 271 /* Returns this track's estimated latency in milliseconds. 272 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 273 * and audio hardware driver. 274 */ latency()275 uint32_t latency() const { return mLatency; } 276 277 /* getters, see constructors and set() */ 278 279 audio_stream_type_t streamType() const; format()280 audio_format_t format() const { return mFormat; } 281 282 /* Return frame size in bytes, which for linear PCM is 283 * channelCount * (bit depth per channel / 8). 284 * channelCount is determined from channelMask, and bit depth comes from format. 285 * For non-linear formats, the frame size is typically 1 byte. 286 */ frameSize()287 size_t frameSize() const { return mFrameSize; } 288 channelCount()289 uint32_t channelCount() const { return mChannelCount; } frameCount()290 size_t frameCount() const { return mFrameCount; } 291 292 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ sharedBuffer()293 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 294 295 /* After it's created the track is not active. Call start() to 296 * make it active. If set, the callback will start being called. 297 * If the track was previously paused, volume is ramped up over the first mix buffer. 298 */ 299 status_t start(); 300 301 /* Stop a track. 302 * In static buffer mode, the track is stopped immediately. 303 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 304 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 305 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 306 * is first drained, mixed, and output, and only then is the track marked as stopped. 307 */ 308 void stop(); 309 bool stopped() const; 310 311 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 312 * This has the effect of draining the buffers without mixing or output. 313 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 314 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 315 */ 316 void flush(); 317 318 /* Pause a track. After pause, the callback will cease being called and 319 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 320 * and will fill up buffers until the pool is exhausted. 321 * Volume is ramped down over the next mix buffer following the pause request, 322 * and then the track is marked as paused. It can be resumed with ramp up by start(). 323 */ 324 void pause(); 325 326 /* Set volume for this track, mostly used for games' sound effects 327 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 328 * This is the older API. New applications should use setVolume(float) when possible. 329 */ 330 status_t setVolume(float left, float right); 331 332 /* Set volume for all channels. This is the preferred API for new applications, 333 * especially for multi-channel content. 334 */ 335 status_t setVolume(float volume); 336 337 /* Set the send level for this track. An auxiliary effect should be attached 338 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 339 */ 340 status_t setAuxEffectSendLevel(float level); 341 void getAuxEffectSendLevel(float* level) const; 342 343 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 344 */ 345 status_t setSampleRate(uint32_t sampleRate); 346 347 /* Return current source sample rate in Hz */ 348 uint32_t getSampleRate() const; 349 350 /* Enables looping and sets the start and end points of looping. 351 * Only supported for static buffer mode. 352 * 353 * Parameters: 354 * 355 * loopStart: loop start in frames relative to start of buffer. 356 * loopEnd: loop end in frames relative to start of buffer. 357 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 358 * pending or active loop. loopCount == -1 means infinite looping. 359 * 360 * For proper operation the following condition must be respected: 361 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 362 * 363 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 364 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 365 * 366 */ 367 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 368 369 /* Sets marker position. When playback reaches the number of frames specified, a callback with 370 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 371 * notification callback. To set a marker at a position which would compute as 0, 372 * a workaround is to set the marker at a nearby position such as ~0 or 1. 373 * If the AudioTrack has been opened with no callback function associated, the operation will 374 * fail. 375 * 376 * Parameters: 377 * 378 * marker: marker position expressed in wrapping (overflow) frame units, 379 * like the return value of getPosition(). 380 * 381 * Returned status (from utils/Errors.h) can be: 382 * - NO_ERROR: successful operation 383 * - INVALID_OPERATION: the AudioTrack has no callback installed. 384 */ 385 status_t setMarkerPosition(uint32_t marker); 386 status_t getMarkerPosition(uint32_t *marker) const; 387 388 /* Sets position update period. Every time the number of frames specified has been played, 389 * a callback with event type EVENT_NEW_POS is called. 390 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 391 * callback. 392 * If the AudioTrack has been opened with no callback function associated, the operation will 393 * fail. 394 * Extremely small values may be rounded up to a value the implementation can support. 395 * 396 * Parameters: 397 * 398 * updatePeriod: position update notification period expressed in frames. 399 * 400 * Returned status (from utils/Errors.h) can be: 401 * - NO_ERROR: successful operation 402 * - INVALID_OPERATION: the AudioTrack has no callback installed. 403 */ 404 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 405 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 406 407 /* Sets playback head position. 408 * Only supported for static buffer mode. 409 * 410 * Parameters: 411 * 412 * position: New playback head position in frames relative to start of buffer. 413 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 414 * but will result in an immediate underrun if started. 415 * 416 * Returned status (from utils/Errors.h) can be: 417 * - NO_ERROR: successful operation 418 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 419 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 420 * buffer 421 */ 422 status_t setPosition(uint32_t position); 423 424 /* Return the total number of frames played since playback start. 425 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 426 * It is reset to zero by flush(), reload(), and stop(). 427 * 428 * Parameters: 429 * 430 * position: Address where to return play head position. 431 * 432 * Returned status (from utils/Errors.h) can be: 433 * - NO_ERROR: successful operation 434 * - BAD_VALUE: position is NULL 435 */ 436 status_t getPosition(uint32_t *position); 437 438 /* For static buffer mode only, this returns the current playback position in frames 439 * relative to start of buffer. It is analogous to the position units used by 440 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 441 */ 442 status_t getBufferPosition(uint32_t *position); 443 444 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 445 * rewriting the buffer before restarting playback after a stop. 446 * This method must be called with the AudioTrack in paused or stopped state. 447 * Not allowed in streaming mode. 448 * 449 * Returned status (from utils/Errors.h) can be: 450 * - NO_ERROR: successful operation 451 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 452 */ 453 status_t reload(); 454 455 /* Returns a handle on the audio output used by this AudioTrack. 456 * 457 * Parameters: 458 * none. 459 * 460 * Returned value: 461 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 462 * track needed to be re-created but that failed 463 */ 464 audio_io_handle_t getOutput() const; 465 466 /* Returns the unique session ID associated with this track. 467 * 468 * Parameters: 469 * none. 470 * 471 * Returned value: 472 * AudioTrack session ID. 473 */ getSessionId()474 int getSessionId() const { return mSessionId; } 475 476 /* Attach track auxiliary output to specified effect. Use effectId = 0 477 * to detach track from effect. 478 * 479 * Parameters: 480 * 481 * effectId: effectId obtained from AudioEffect::id(). 482 * 483 * Returned status (from utils/Errors.h) can be: 484 * - NO_ERROR: successful operation 485 * - INVALID_OPERATION: the effect is not an auxiliary effect. 486 * - BAD_VALUE: The specified effect ID is invalid 487 */ 488 status_t attachAuxEffect(int effectId); 489 490 /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 491 * After filling these slots with data, the caller should release them with releaseBuffer(). 492 * If the track buffer is not full, obtainBuffer() returns as many contiguous 493 * [empty slots for] frames as are available immediately. 494 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 495 * regardless of the value of waitCount. 496 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 497 * maximum timeout based on waitCount; see chart below. 498 * Buffers will be returned until the pool 499 * is exhausted, at which point obtainBuffer() will either block 500 * or return WOULD_BLOCK depending on the value of the "waitCount" 501 * parameter. 502 * Each sample is 16-bit signed PCM. 503 * 504 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 505 * which should use write() or callback EVENT_MORE_DATA instead. 506 * 507 * Interpretation of waitCount: 508 * +n limits wait time to n * WAIT_PERIOD_MS, 509 * -1 causes an (almost) infinite wait time, 510 * 0 non-blocking. 511 * 512 * Buffer fields 513 * On entry: 514 * frameCount number of frames requested 515 * After error return: 516 * frameCount 0 517 * size 0 518 * raw undefined 519 * After successful return: 520 * frameCount actual number of frames available, <= number requested 521 * size actual number of bytes available 522 * raw pointer to the buffer 523 */ 524 525 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 526 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 527 __attribute__((__deprecated__)); 528 529 private: 530 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 531 * additional non-contiguous frames that are available immediately. 532 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 533 * in case the requested amount of frames is in two or more non-contiguous regions. 534 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 535 */ 536 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 537 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 538 public: 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 * Default behavior is to only return until all data has been transferred. Set 'blocking' to 555 * false for the method to return immediately without waiting to try multiple times to write 556 * the full content of the buffer. 557 */ 558 ssize_t write(const void* buffer, size_t size, bool blocking = true); 559 560 /* 561 * Dumps the state of an audio track. 562 */ 563 status_t dump(int fd, const Vector<String16>& args) const; 564 565 /* 566 * Return the total number of frames which AudioFlinger desired but were unavailable, 567 * and thus which resulted in an underrun. Reset to zero by stop(). 568 */ 569 uint32_t getUnderrunFrames() const; 570 571 /* Get the flags */ getFlags()572 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 573 574 /* Set parameters - only possible when using direct output */ 575 status_t setParameters(const String8& keyValuePairs); 576 577 /* Get parameters */ 578 String8 getParameters(const String8& keys); 579 580 /* Poll for a timestamp on demand. 581 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 582 * or if you need to get the most recent timestamp outside of the event callback handler. 583 * Caution: calling this method too often may be inefficient; 584 * if you need a high resolution mapping between frame position and presentation time, 585 * consider implementing that at application level, based on the low resolution timestamps. 586 * Returns NO_ERROR if timestamp is valid. 587 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 588 * start/ACTIVE, when the number of frames consumed is less than the 589 * overall hardware latency to physical output. In WOULD_BLOCK cases, 590 * one might poll again, or use getPosition(), or use 0 position and 591 * current time for the timestamp. 592 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 593 * 594 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 595 */ 596 status_t getTimestamp(AudioTimestamp& timestamp); 597 598 protected: 599 /* copying audio tracks is not allowed */ 600 AudioTrack(const AudioTrack& other); 601 AudioTrack& operator = (const AudioTrack& other); 602 603 void setAttributesFromStreamType(audio_stream_type_t streamType); 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 isOffloaded_l()660 bool isOffloaded_l() const 661 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 662 isOffloadedOrDirect_l()663 bool isOffloadedOrDirect_l() const 664 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 665 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 666 isDirect_l()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; // mStreamType == AUDIO_STREAM_DEFAULT implies 692 // this AudioTrack has valid attributes 693 uint32_t mChannelCount; 694 audio_channel_mask_t mChannelMask; 695 sp<IMemory> mSharedBuffer; 696 transfer_type mTransfer; 697 audio_offload_info_t mOffloadInfoCopy; 698 const audio_offload_info_t* mOffloadInfo; 699 audio_attributes_t mAttributes; 700 701 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's 702 // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. 703 size_t mFrameSize; // app-level frame size 704 size_t mFrameSizeAF; // AudioFlinger frame size 705 706 status_t mStatus; 707 708 // can change dynamically when IAudioTrack invalidated 709 uint32_t mLatency; // in ms 710 711 // Indicates the current track state. Protected by mLock. 712 enum State { 713 STATE_ACTIVE, 714 STATE_STOPPED, 715 STATE_PAUSED, 716 STATE_PAUSED_STOPPING, 717 STATE_FLUSHED, 718 STATE_STOPPING, 719 } mState; 720 721 // for client callback handler 722 callback_t mCbf; // callback handler for events, or NULL 723 void* mUserData; 724 725 // for notification APIs 726 uint32_t mNotificationFramesReq; // requested number of frames between each 727 // notification callback, 728 // at initial source sample rate 729 uint32_t mNotificationFramesAct; // actual number of frames between each 730 // notification callback, 731 // at initial source sample rate 732 bool mRefreshRemaining; // processAudioBuffer() should refresh 733 // mRemainingFrames and mRetryOnPartialBuffer 734 735 // These are private to processAudioBuffer(), and are not protected by a lock 736 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 737 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 738 uint32_t mObservedSequence; // last observed value of mSequence 739 740 uint32_t mLoopPeriod; // in frames, zero means looping is disabled 741 742 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 743 bool mMarkerReached; 744 uint32_t mNewPosition; // in frames 745 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 746 uint32_t mServer; // in frames, last known mProxy->getPosition() 747 // which is count of frames consumed by server, 748 // reset by new IAudioTrack, 749 // whether it is reset by stop() is TBD 750 uint32_t mPosition; // in frames, like mServer except continues 751 // monotonically after new IAudioTrack, 752 // and could be easily widened to uint64_t 753 uint32_t mReleased; // in frames, count of frames released to server 754 // but not necessarily consumed by server, 755 // reset by stop() but continues monotonically 756 // after new IAudioTrack to restore mPosition, 757 // and could be easily widened to uint64_t 758 int64_t mStartUs; // the start time after flush or stop. 759 // only used for offloaded and direct tracks. 760 761 audio_output_flags_t mFlags; 762 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 763 // mLock must be held to read or write those bits reliably. 764 765 int mSessionId; 766 int mAuxEffectId; 767 768 mutable Mutex mLock; 769 770 bool mIsTimed; 771 int mPreviousPriority; // before start() 772 SchedPolicy mPreviousSchedulingGroup; 773 bool mAwaitBoost; // thread should wait for priority boost before running 774 775 // The proxy should only be referenced while a lock is held because the proxy isn't 776 // multi-thread safe, especially the SingleStateQueue part of the proxy. 777 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 778 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 779 // them around in case they are replaced during the obtainBuffer(). 780 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 781 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 782 783 bool mInUnderrun; // whether track is currently in underrun state 784 uint32_t mPausedPosition; 785 786 private: 787 class DeathNotifier : public IBinder::DeathRecipient { 788 public: DeathNotifier(AudioTrack * audioTrack)789 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 790 protected: 791 virtual void binderDied(const wp<IBinder>& who); 792 private: 793 const wp<AudioTrack> mAudioTrack; 794 }; 795 796 sp<DeathNotifier> mDeathNotifier; 797 uint32_t mSequence; // incremented for each new IAudioTrack attempt 798 int mClientUid; 799 pid_t mClientPid; 800 }; 801 802 class TimedAudioTrack : public AudioTrack 803 { 804 public: 805 TimedAudioTrack(); 806 807 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 808 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 809 810 /* queue a buffer obtained via allocateTimedBuffer for playback at the 811 given timestamp. PTS units are microseconds on the media time timeline. 812 The media time transform (set with setMediaTimeTransform) set by the 813 audio producer will handle converting from media time to local time 814 (perhaps going through the common time timeline in the case of 815 synchronized multiroom audio case) */ 816 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 817 818 /* define a transform between media time and either common time or 819 local time */ 820 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 821 status_t setMediaTimeTransform(const LinearTransform& xform, 822 TargetTimeline target); 823 }; 824 825 }; // namespace android 826 827 #endif // ANDROID_AUDIOTRACK_H 828