AudioTrack.h revision 3a474aa67fc31505740526dd249d96204c08bf79
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 <media/AudioResamplerPublic.h> 25#include <utils/threads.h> 26 27namespace android { 28 29// ---------------------------------------------------------------------------- 30 31struct audio_track_cblk_t; 32class AudioTrackClientProxy; 33class StaticAudioTrackClientProxy; 34 35// ---------------------------------------------------------------------------- 36 37class AudioTrack : public RefBase 38{ 39public: 40 41 /* Events used by AudioTrack callback function (callback_t). 42 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 43 */ 44 enum event_type { 45 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 46 // If this event is delivered but the callback handler 47 // does not want to write more data, the handler must explicitly 48 // ignore the event by setting frameCount to zero. 49 EVENT_UNDERRUN = 1, // Buffer underrun occurred. 50 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 51 // loop start if loop count was not 0. 52 EVENT_MARKER = 3, // Playback head is at the specified marker position 53 // (See setMarkerPosition()). 54 EVENT_NEW_POS = 4, // Playback head is at a new position 55 // (See setPositionUpdatePeriod()). 56 EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer. 57 // Not currently used by android.media.AudioTrack. 58 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 59 // voluntary invalidation by mediaserver, or mediaserver crash. 60 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 61 // back (after stop is called) 62 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 63 // in the mapping from frame position to presentation time. 64 // See AudioTimestamp for the information included with event. 65 }; 66 67 /* Client should declare a Buffer and pass the address to obtainBuffer() 68 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 69 */ 70 71 class Buffer 72 { 73 public: 74 // FIXME use m prefix 75 size_t frameCount; // number of sample frames corresponding to size; 76 // on input to obtainBuffer() it is the number of frames desired, 77 // on output from obtainBuffer() it is the number of available 78 // [empty slots for] frames to be filled 79 // on input to releaseBuffer() it is currently ignored 80 81 size_t size; // input/output in bytes == frameCount * frameSize 82 // on input to obtainBuffer() it is ignored 83 // on output from obtainBuffer() it is the number of available 84 // [empty slots for] bytes to be filled, 85 // which is frameCount * frameSize 86 // on input to releaseBuffer() it is the number of bytes to 87 // release 88 // FIXME This is redundant with respect to frameCount. Consider 89 // removing size and making frameCount the primary field. 90 91 union { 92 void* raw; 93 short* i16; // signed 16-bit 94 int8_t* i8; // unsigned 8-bit, offset by 0x80 95 }; // input to obtainBuffer(): unused, output: pointer to buffer 96 }; 97 98 /* As a convenience, if a callback is supplied, a handler thread 99 * is automatically created with the appropriate priority. This thread 100 * invokes the callback when a new buffer becomes available or various conditions occur. 101 * Parameters: 102 * 103 * event: type of event notified (see enum AudioTrack::event_type). 104 * user: Pointer to context for use by the callback receiver. 105 * info: Pointer to optional parameter according to event type: 106 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 107 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 108 * written. 109 * - EVENT_UNDERRUN: unused. 110 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 111 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 112 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 113 * - EVENT_BUFFER_END: unused. 114 * - EVENT_NEW_IAUDIOTRACK: unused. 115 * - EVENT_STREAM_END: unused. 116 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 117 */ 118 119 typedef void (*callback_t)(int event, void* user, void *info); 120 121 /* Returns the minimum frame count required for the successful creation of 122 * an AudioTrack object. 123 * Returned status (from utils/Errors.h) can be: 124 * - NO_ERROR: successful operation 125 * - NO_INIT: audio server or audio hardware not initialized 126 * - BAD_VALUE: unsupported configuration 127 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 128 * and is undefined otherwise. 129 * FIXME This API assumes a route, and so should be deprecated. 130 */ 131 132 static status_t getMinFrameCount(size_t* frameCount, 133 audio_stream_type_t streamType, 134 uint32_t sampleRate); 135 136 /* How data is transferred to AudioTrack 137 */ 138 enum transfer_type { 139 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 140 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 141 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 142 TRANSFER_SYNC, // synchronous write() 143 TRANSFER_SHARED, // shared memory 144 }; 145 146 /* Constructs an uninitialized AudioTrack. No connection with 147 * AudioFlinger takes place. Use set() after this. 148 */ 149 AudioTrack(); 150 151 /* Creates an AudioTrack object and registers it with AudioFlinger. 152 * Once created, the track needs to be started before it can be used. 153 * Unspecified values are set to appropriate default values. 154 * 155 * Parameters: 156 * 157 * streamType: Select the type of audio stream this track is attached to 158 * (e.g. AUDIO_STREAM_MUSIC). 159 * sampleRate: Data source sampling rate in Hz. 160 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 161 * For direct and offloaded tracks, the possible format(s) depends on the 162 * output sink. 163 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 164 * frameCount: Minimum size of track PCM buffer in frames. This defines the 165 * application's contribution to the 166 * latency of the track. The actual size selected by the AudioTrack could be 167 * larger if the requested size is not compatible with current audio HAL 168 * configuration. Zero means to use a default value. 169 * flags: See comments on audio_output_flags_t in <system/audio.h>. 170 * cbf: Callback function. If not null, this function is called periodically 171 * to provide new data in TRANSFER_CALLBACK mode 172 * and inform of marker, position updates, etc. 173 * user: Context for use by the callback receiver. 174 * notificationFrames: The callback function is called each time notificationFrames PCM 175 * frames have been consumed from track input buffer. 176 * This is expressed in units of frames at the initial source sample rate. 177 * sessionId: Specific session ID, or zero to use default. 178 * transferType: How data is transferred to AudioTrack. 179 * offloadInfo: If not NULL, provides offload parameters for 180 * AudioSystem::getOutputForAttr(). 181 * uid: User ID of the app which initially requested this AudioTrack 182 * for power management tracking, or -1 for current user ID. 183 * pid: Process ID of the app which initially requested this AudioTrack 184 * for power management tracking, or -1 for current process ID. 185 * pAttributes: If not NULL, supersedes streamType for use case selection. 186 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 187 */ 188 189 AudioTrack( audio_stream_type_t streamType, 190 uint32_t sampleRate, 191 audio_format_t format, 192 audio_channel_mask_t channelMask, 193 size_t frameCount = 0, 194 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 195 callback_t cbf = NULL, 196 void* user = NULL, 197 uint32_t notificationFrames = 0, 198 int sessionId = AUDIO_SESSION_ALLOCATE, 199 transfer_type transferType = TRANSFER_DEFAULT, 200 const audio_offload_info_t *offloadInfo = NULL, 201 int uid = -1, 202 pid_t pid = -1, 203 const audio_attributes_t* pAttributes = NULL); 204 205 /* Creates an audio track and registers it with AudioFlinger. 206 * With this constructor, the track is configured for static buffer mode. 207 * Data to be rendered is passed in a shared memory buffer 208 * identified by the argument sharedBuffer, which should be non-0. 209 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 210 * but without the ability to specify a non-zero value for the frameCount parameter. 211 * The memory should be initialized to the desired data before calling start(). 212 * The write() method is not supported in this case. 213 * It is recommended to pass a callback function to be notified of playback end by an 214 * EVENT_UNDERRUN event. 215 */ 216 217 AudioTrack( audio_stream_type_t streamType, 218 uint32_t sampleRate, 219 audio_format_t format, 220 audio_channel_mask_t channelMask, 221 const sp<IMemory>& sharedBuffer, 222 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 223 callback_t cbf = NULL, 224 void* user = NULL, 225 uint32_t notificationFrames = 0, 226 int sessionId = AUDIO_SESSION_ALLOCATE, 227 transfer_type transferType = TRANSFER_DEFAULT, 228 const audio_offload_info_t *offloadInfo = NULL, 229 int uid = -1, 230 pid_t pid = -1, 231 const audio_attributes_t* pAttributes = NULL); 232 233 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 234 * Also destroys all resources associated with the AudioTrack. 235 */ 236protected: 237 virtual ~AudioTrack(); 238public: 239 240 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 241 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 242 * set() is not multi-thread safe. 243 * Returned status (from utils/Errors.h) can be: 244 * - NO_ERROR: successful initialization 245 * - INVALID_OPERATION: AudioTrack is already initialized 246 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 247 * - NO_INIT: audio server or audio hardware not initialized 248 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 249 * If sharedBuffer is non-0, the frameCount parameter is ignored and 250 * replaced by the shared buffer's total allocated size in frame units. 251 * 252 * Parameters not listed in the AudioTrack constructors above: 253 * 254 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 255 * 256 * Internal state post condition: 257 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 258 */ 259 status_t set(audio_stream_type_t streamType, 260 uint32_t sampleRate, 261 audio_format_t format, 262 audio_channel_mask_t channelMask, 263 size_t frameCount = 0, 264 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 265 callback_t cbf = NULL, 266 void* user = NULL, 267 uint32_t notificationFrames = 0, 268 const sp<IMemory>& sharedBuffer = 0, 269 bool threadCanCallJava = false, 270 int sessionId = AUDIO_SESSION_ALLOCATE, 271 transfer_type transferType = TRANSFER_DEFAULT, 272 const audio_offload_info_t *offloadInfo = NULL, 273 int uid = -1, 274 pid_t pid = -1, 275 const audio_attributes_t* pAttributes = NULL); 276 277 /* Result of constructing the AudioTrack. This must be checked for successful initialization 278 * before using any AudioTrack API (except for set()), because using 279 * an uninitialized AudioTrack produces undefined results. 280 * See set() method above for possible return codes. 281 */ 282 status_t initCheck() const { return mStatus; } 283 284 /* Returns this track's estimated latency in milliseconds. 285 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 286 * and audio hardware driver. 287 */ 288 uint32_t latency() const { return mLatency; } 289 290 /* getters, see constructors and set() */ 291 292 audio_stream_type_t streamType() const; 293 audio_format_t format() const { return mFormat; } 294 295 /* Return frame size in bytes, which for linear PCM is 296 * channelCount * (bit depth per channel / 8). 297 * channelCount is determined from channelMask, and bit depth comes from format. 298 * For non-linear formats, the frame size is typically 1 byte. 299 */ 300 size_t frameSize() const { return mFrameSize; } 301 302 uint32_t channelCount() const { return mChannelCount; } 303 size_t frameCount() const { return mFrameCount; } 304 305 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 306 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 307 308 /* After it's created the track is not active. Call start() to 309 * make it active. If set, the callback will start being called. 310 * If the track was previously paused, volume is ramped up over the first mix buffer. 311 */ 312 status_t start(); 313 314 /* Stop a track. 315 * In static buffer mode, the track is stopped immediately. 316 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 317 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 318 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 319 * is first drained, mixed, and output, and only then is the track marked as stopped. 320 */ 321 void stop(); 322 bool stopped() const; 323 324 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 325 * This has the effect of draining the buffers without mixing or output. 326 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 327 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 328 */ 329 void flush(); 330 331 /* Pause a track. After pause, the callback will cease being called and 332 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 333 * and will fill up buffers until the pool is exhausted. 334 * Volume is ramped down over the next mix buffer following the pause request, 335 * and then the track is marked as paused. It can be resumed with ramp up by start(). 336 */ 337 void pause(); 338 339 /* Set volume for this track, mostly used for games' sound effects 340 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 341 * This is the older API. New applications should use setVolume(float) when possible. 342 */ 343 status_t setVolume(float left, float right); 344 345 /* Set volume for all channels. This is the preferred API for new applications, 346 * especially for multi-channel content. 347 */ 348 status_t setVolume(float volume); 349 350 /* Set the send level for this track. An auxiliary effect should be attached 351 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 352 */ 353 status_t setAuxEffectSendLevel(float level); 354 void getAuxEffectSendLevel(float* level) const; 355 356 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 357 */ 358 status_t setSampleRate(uint32_t sampleRate); 359 360 /* Return current source sample rate in Hz */ 361 uint32_t getSampleRate() const; 362 363 /* Return the original source sample rate in Hz. This corresponds to the sample rate 364 * if playback rate had normal speed and pitch. 365 */ 366 uint32_t getOriginalSampleRate() const; 367 368 /* Set source playback rate for timestretch 369 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 370 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 371 * 372 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 373 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 374 * 375 * Speed increases the playback rate of media, but does not alter pitch. 376 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 377 */ 378 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 379 380 /* Return current playback rate */ 381 const AudioPlaybackRate& getPlaybackRate() const; 382 383 /* Enables looping and sets the start and end points of looping. 384 * Only supported for static buffer mode. 385 * 386 * Parameters: 387 * 388 * loopStart: loop start in frames relative to start of buffer. 389 * loopEnd: loop end in frames relative to start of buffer. 390 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 391 * pending or active loop. loopCount == -1 means infinite looping. 392 * 393 * For proper operation the following condition must be respected: 394 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 395 * 396 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 397 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 398 * 399 */ 400 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 401 402 /* Sets marker position. When playback reaches the number of frames specified, a callback with 403 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 404 * notification callback. To set a marker at a position which would compute as 0, 405 * a workaround is to set the marker at a nearby position such as ~0 or 1. 406 * If the AudioTrack has been opened with no callback function associated, the operation will 407 * fail. 408 * 409 * Parameters: 410 * 411 * marker: marker position expressed in wrapping (overflow) frame units, 412 * like the return value of getPosition(). 413 * 414 * Returned status (from utils/Errors.h) can be: 415 * - NO_ERROR: successful operation 416 * - INVALID_OPERATION: the AudioTrack has no callback installed. 417 */ 418 status_t setMarkerPosition(uint32_t marker); 419 status_t getMarkerPosition(uint32_t *marker) const; 420 421 /* Sets position update period. Every time the number of frames specified has been played, 422 * a callback with event type EVENT_NEW_POS is called. 423 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 424 * callback. 425 * If the AudioTrack has been opened with no callback function associated, the operation will 426 * fail. 427 * Extremely small values may be rounded up to a value the implementation can support. 428 * 429 * Parameters: 430 * 431 * updatePeriod: position update notification period expressed in frames. 432 * 433 * Returned status (from utils/Errors.h) can be: 434 * - NO_ERROR: successful operation 435 * - INVALID_OPERATION: the AudioTrack has no callback installed. 436 */ 437 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 438 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 439 440 /* Sets playback head position. 441 * Only supported for static buffer mode. 442 * 443 * Parameters: 444 * 445 * position: New playback head position in frames relative to start of buffer. 446 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 447 * but will result in an immediate underrun if started. 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 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 453 * buffer 454 */ 455 status_t setPosition(uint32_t position); 456 457 /* Return the total number of frames played since playback start. 458 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 459 * It is reset to zero by flush(), reload(), and stop(). 460 * 461 * Parameters: 462 * 463 * position: Address where to return play head position. 464 * 465 * Returned status (from utils/Errors.h) can be: 466 * - NO_ERROR: successful operation 467 * - BAD_VALUE: position is NULL 468 */ 469 status_t getPosition(uint32_t *position); 470 471 /* For static buffer mode only, this returns the current playback position in frames 472 * relative to start of buffer. It is analogous to the position units used by 473 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 474 */ 475 status_t getBufferPosition(uint32_t *position); 476 477 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 478 * rewriting the buffer before restarting playback after a stop. 479 * This method must be called with the AudioTrack in paused or stopped state. 480 * Not allowed in streaming mode. 481 * 482 * Returned status (from utils/Errors.h) can be: 483 * - NO_ERROR: successful operation 484 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 485 */ 486 status_t reload(); 487 488 /* Returns a handle on the audio output used by this AudioTrack. 489 * 490 * Parameters: 491 * none. 492 * 493 * Returned value: 494 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 495 * track needed to be re-created but that failed 496 */ 497private: 498 audio_io_handle_t getOutput() const; 499public: 500 501 /* Selects the audio device to use for output of this AudioTrack. A value of 502 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 503 * 504 * Parameters: 505 * The device ID of the selected device (as returned by the AudioDevicesManager API). 506 * 507 * Returned value: 508 * - NO_ERROR: successful operation 509 * TODO: what else can happen here? 510 */ 511 status_t setOutputDevice(audio_port_handle_t deviceId); 512 513 /* Returns the ID of the audio device used for output of this AudioTrack. 514 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 515 * 516 * Parameters: 517 * none. 518 */ 519 audio_port_handle_t getOutputDevice(); 520 521 /* Returns the unique session ID associated with this track. 522 * 523 * Parameters: 524 * none. 525 * 526 * Returned value: 527 * AudioTrack session ID. 528 */ 529 int getSessionId() const { return mSessionId; } 530 531 /* Attach track auxiliary output to specified effect. Use effectId = 0 532 * to detach track from effect. 533 * 534 * Parameters: 535 * 536 * effectId: effectId obtained from AudioEffect::id(). 537 * 538 * Returned status (from utils/Errors.h) can be: 539 * - NO_ERROR: successful operation 540 * - INVALID_OPERATION: the effect is not an auxiliary effect. 541 * - BAD_VALUE: The specified effect ID is invalid 542 */ 543 status_t attachAuxEffect(int effectId); 544 545 /* Public API for TRANSFER_OBTAIN mode. 546 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 547 * After filling these slots with data, the caller should release them with releaseBuffer(). 548 * If the track buffer is not full, obtainBuffer() returns as many contiguous 549 * [empty slots for] frames as are available immediately. 550 * 551 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 552 * additional non-contiguous frames that are predicted to be available immediately, 553 * if the client were to release the first frames and then call obtainBuffer() again. 554 * This value is only a prediction, and needs to be confirmed. 555 * It will be set to zero for an error return. 556 * 557 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 558 * regardless of the value of waitCount. 559 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 560 * maximum timeout based on waitCount; see chart below. 561 * Buffers will be returned until the pool 562 * is exhausted, at which point obtainBuffer() will either block 563 * or return WOULD_BLOCK depending on the value of the "waitCount" 564 * parameter. 565 * 566 * Interpretation of waitCount: 567 * +n limits wait time to n * WAIT_PERIOD_MS, 568 * -1 causes an (almost) infinite wait time, 569 * 0 non-blocking. 570 * 571 * Buffer fields 572 * On entry: 573 * frameCount number of [empty slots for] frames requested 574 * size ignored 575 * raw ignored 576 * After error return: 577 * frameCount 0 578 * size 0 579 * raw undefined 580 * After successful return: 581 * frameCount actual number of [empty slots for] frames available, <= number requested 582 * size actual number of bytes available 583 * raw pointer to the buffer 584 */ 585 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 586 size_t *nonContig = NULL); 587 588private: 589 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 590 * additional non-contiguous frames that are predicted to be available immediately, 591 * if the client were to release the first frames and then call obtainBuffer() again. 592 * This value is only a prediction, and needs to be confirmed. 593 * It will be set to zero for an error return. 594 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 595 * in case the requested amount of frames is in two or more non-contiguous regions. 596 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 597 */ 598 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 599 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 600public: 601 602 /* Public API for TRANSFER_OBTAIN mode. 603 * Release a filled buffer of frames for AudioFlinger to process. 604 * 605 * Buffer fields: 606 * frameCount currently ignored but recommend to set to actual number of frames filled 607 * size actual number of bytes filled, must be multiple of frameSize 608 * raw ignored 609 */ 610 void releaseBuffer(const Buffer* audioBuffer); 611 612 /* As a convenience we provide a write() interface to the audio buffer. 613 * Input parameter 'size' is in byte units. 614 * This is implemented on top of obtainBuffer/releaseBuffer. For best 615 * performance use callbacks. Returns actual number of bytes written >= 0, 616 * or one of the following negative status codes: 617 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 618 * BAD_VALUE size is invalid 619 * WOULD_BLOCK when obtainBuffer() returns same, or 620 * AudioTrack was stopped during the write 621 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 622 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 623 * false for the method to return immediately without waiting to try multiple times to write 624 * the full content of the buffer. 625 */ 626 ssize_t write(const void* buffer, size_t size, bool blocking = true); 627 628 /* 629 * Dumps the state of an audio track. 630 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 631 */ 632 status_t dump(int fd, const Vector<String16>& args) const; 633 634 /* 635 * Return the total number of frames which AudioFlinger desired but were unavailable, 636 * and thus which resulted in an underrun. Reset to zero by stop(). 637 */ 638 uint32_t getUnderrunFrames() const; 639 640 /* Get the flags */ 641 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 642 643 /* Set parameters - only possible when using direct output */ 644 status_t setParameters(const String8& keyValuePairs); 645 646 /* Get parameters */ 647 String8 getParameters(const String8& keys); 648 649 /* Poll for a timestamp on demand. 650 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 651 * or if you need to get the most recent timestamp outside of the event callback handler. 652 * Caution: calling this method too often may be inefficient; 653 * if you need a high resolution mapping between frame position and presentation time, 654 * consider implementing that at application level, based on the low resolution timestamps. 655 * Returns NO_ERROR if timestamp is valid. 656 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 657 * start/ACTIVE, when the number of frames consumed is less than the 658 * overall hardware latency to physical output. In WOULD_BLOCK cases, 659 * one might poll again, or use getPosition(), or use 0 position and 660 * current time for the timestamp. 661 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 662 * 663 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 664 */ 665 status_t getTimestamp(AudioTimestamp& timestamp); 666 667protected: 668 /* copying audio tracks is not allowed */ 669 AudioTrack(const AudioTrack& other); 670 AudioTrack& operator = (const AudioTrack& other); 671 672 /* a small internal class to handle the callback */ 673 class AudioTrackThread : public Thread 674 { 675 public: 676 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 677 678 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 679 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 680 virtual void requestExit(); 681 682 void pause(); // suspend thread from execution at next loop boundary 683 void resume(); // allow thread to execute, if not requested to exit 684 void wake(); // wake to handle changed notification conditions. 685 686 private: 687 void pauseInternal(nsecs_t ns = 0LL); 688 // like pause(), but only used internally within thread 689 690 friend class AudioTrack; 691 virtual bool threadLoop(); 692 AudioTrack& mReceiver; 693 virtual ~AudioTrackThread(); 694 Mutex mMyLock; // Thread::mLock is private 695 Condition mMyCond; // Thread::mThreadExitedCondition is private 696 bool mPaused; // whether thread is requested to pause at next loop entry 697 bool mPausedInt; // whether thread internally requests pause 698 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 699 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 700 // to processAudioBuffer() as state may have changed 701 // since pause time calculated. 702 }; 703 704 // body of AudioTrackThread::threadLoop() 705 // returns the maximum amount of time before we would like to run again, where: 706 // 0 immediately 707 // > 0 no later than this many nanoseconds from now 708 // NS_WHENEVER still active but no particular deadline 709 // NS_INACTIVE inactive so don't run again until re-started 710 // NS_NEVER never again 711 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 712 nsecs_t processAudioBuffer(); 713 714 // caller must hold lock on mLock for all _l methods 715 716 status_t createTrack_l(); 717 718 // can only be called when mState != STATE_ACTIVE 719 void flush_l(); 720 721 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 722 723 // FIXME enum is faster than strcmp() for parameter 'from' 724 status_t restoreTrack_l(const char *from); 725 726 bool isOffloaded() const; 727 bool isDirect() const; 728 bool isOffloadedOrDirect() const; 729 730 bool isOffloaded_l() const 731 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 732 733 bool isOffloadedOrDirect_l() const 734 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 735 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 736 737 bool isDirect_l() const 738 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 739 740 // increment mPosition by the delta of mServer, and return new value of mPosition 741 uint32_t updateAndGetPosition_l(); 742 743 // check sample rate and speed is compatible with AudioTrack 744 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; 745 746 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 747 sp<IAudioTrack> mAudioTrack; 748 sp<IMemory> mCblkMemory; 749 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 750 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 751 752 sp<AudioTrackThread> mAudioTrackThread; 753 754 float mVolume[2]; 755 float mSendLevel; 756 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 757 uint32_t mOriginalSampleRate; 758 AudioPlaybackRate mPlaybackRate; 759 size_t mFrameCount; // corresponds to current IAudioTrack, value is 760 // reported back by AudioFlinger to the client 761 size_t mReqFrameCount; // frame count to request the first or next time 762 // a new IAudioTrack is needed, non-decreasing 763 764 // constant after constructor or set() 765 audio_format_t mFormat; // as requested by client, not forced to 16-bit 766 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 767 // this AudioTrack has valid attributes 768 uint32_t mChannelCount; 769 audio_channel_mask_t mChannelMask; 770 sp<IMemory> mSharedBuffer; 771 transfer_type mTransfer; 772 audio_offload_info_t mOffloadInfoCopy; 773 const audio_offload_info_t* mOffloadInfo; 774 audio_attributes_t mAttributes; 775 776 size_t mFrameSize; // frame size in bytes 777 778 status_t mStatus; 779 780 // can change dynamically when IAudioTrack invalidated 781 uint32_t mLatency; // in ms 782 783 // Indicates the current track state. Protected by mLock. 784 enum State { 785 STATE_ACTIVE, 786 STATE_STOPPED, 787 STATE_PAUSED, 788 STATE_PAUSED_STOPPING, 789 STATE_FLUSHED, 790 STATE_STOPPING, 791 } mState; 792 793 // for client callback handler 794 callback_t mCbf; // callback handler for events, or NULL 795 void* mUserData; 796 797 // for notification APIs 798 uint32_t mNotificationFramesReq; // requested number of frames between each 799 // notification callback, 800 // at initial source sample rate 801 uint32_t mNotificationFramesAct; // actual number of frames between each 802 // notification callback, 803 // at initial source sample rate 804 bool mRefreshRemaining; // processAudioBuffer() should refresh 805 // mRemainingFrames and mRetryOnPartialBuffer 806 807 // used for static track cbf and restoration 808 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 809 uint32_t mLoopStart; // last setLoop loopStart 810 uint32_t mLoopEnd; // last setLoop loopEnd 811 int32_t mLoopCountNotified; // the last loopCount notified by callback. 812 // mLoopCountNotified counts down, matching 813 // the remaining loop count for static track 814 // playback. 815 816 // These are private to processAudioBuffer(), and are not protected by a lock 817 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 818 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 819 uint32_t mObservedSequence; // last observed value of mSequence 820 821 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 822 bool mMarkerReached; 823 uint32_t mNewPosition; // in frames 824 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 825 826 uint32_t mServer; // in frames, last known mProxy->getPosition() 827 // which is count of frames consumed by server, 828 // reset by new IAudioTrack, 829 // whether it is reset by stop() is TBD 830 uint32_t mPosition; // in frames, like mServer except continues 831 // monotonically after new IAudioTrack, 832 // and could be easily widened to uint64_t 833 uint32_t mReleased; // in frames, count of frames released to server 834 // but not necessarily consumed by server, 835 // reset by stop() but continues monotonically 836 // after new IAudioTrack to restore mPosition, 837 // and could be easily widened to uint64_t 838 int64_t mStartUs; // the start time after flush or stop. 839 // only used for offloaded and direct tracks. 840 841 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 842 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 843 844 audio_output_flags_t mFlags; 845 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 846 // mLock must be held to read or write those bits reliably. 847 848 int mSessionId; 849 int mAuxEffectId; 850 851 mutable Mutex mLock; 852 853 bool mIsTimed; 854 int mPreviousPriority; // before start() 855 SchedPolicy mPreviousSchedulingGroup; 856 bool mAwaitBoost; // thread should wait for priority boost before running 857 858 // The proxy should only be referenced while a lock is held because the proxy isn't 859 // multi-thread safe, especially the SingleStateQueue part of the proxy. 860 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 861 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 862 // them around in case they are replaced during the obtainBuffer(). 863 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 864 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 865 866 bool mInUnderrun; // whether track is currently in underrun state 867 uint32_t mPausedPosition; 868 869 // For Device Selection API 870 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 871 audio_port_handle_t mSelectedDeviceId; 872 873private: 874 class DeathNotifier : public IBinder::DeathRecipient { 875 public: 876 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 877 protected: 878 virtual void binderDied(const wp<IBinder>& who); 879 private: 880 const wp<AudioTrack> mAudioTrack; 881 }; 882 883 sp<DeathNotifier> mDeathNotifier; 884 uint32_t mSequence; // incremented for each new IAudioTrack attempt 885 int mClientUid; 886 pid_t mClientPid; 887}; 888 889class TimedAudioTrack : public AudioTrack 890{ 891public: 892 TimedAudioTrack(); 893 894 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 895 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 896 897 /* queue a buffer obtained via allocateTimedBuffer for playback at the 898 given timestamp. PTS units are microseconds on the media time timeline. 899 The media time transform (set with setMediaTimeTransform) set by the 900 audio producer will handle converting from media time to local time 901 (perhaps going through the common time timeline in the case of 902 synchronized multiroom audio case) */ 903 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 904 905 /* define a transform between media time and either common time or 906 local time */ 907 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 908 status_t setMediaTimeTransform(const LinearTransform& xform, 909 TargetTimeline target); 910}; 911 912}; // namespace android 913 914#endif // ANDROID_AUDIOTRACK_H 915