AudioTrack.h revision 25a9e5561a7f14e79b04f713a515a9464b9ea077
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 <media/Modulo.h> 26#include <utils/threads.h> 27 28namespace android { 29 30// ---------------------------------------------------------------------------- 31 32struct audio_track_cblk_t; 33class AudioTrackClientProxy; 34class StaticAudioTrackClientProxy; 35 36// ---------------------------------------------------------------------------- 37 38class AudioTrack : public RefBase 39{ 40public: 41 42 /* Events used by AudioTrack callback function (callback_t). 43 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 44 */ 45 enum event_type { 46 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 47 // This event only occurs for TRANSFER_CALLBACK. 48 // If this event is delivered but the callback handler 49 // does not want to write more data, the handler must 50 // ignore the event by setting frameCount to zero. 51 // This might occur, for example, if the application is 52 // waiting for source data or is at the end of stream. 53 // 54 // For data filling, it is preferred that the callback 55 // does not block and instead returns a short count on 56 // the amount of data actually delivered 57 // (or 0, if no data is currently available). 58 EVENT_UNDERRUN = 1, // Buffer underrun occurred. This will not occur for 59 // static tracks. 60 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 61 // loop start if loop count was not 0 for a static track. 62 EVENT_MARKER = 3, // Playback head is at the specified marker position 63 // (See setMarkerPosition()). 64 EVENT_NEW_POS = 4, // Playback head is at a new position 65 // (See setPositionUpdatePeriod()). 66 EVENT_BUFFER_END = 5, // Playback has completed for a static track. 67 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 68 // voluntary invalidation by mediaserver, or mediaserver crash. 69 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 70 // back (after stop is called) for an offloaded track. 71#if 0 // FIXME not yet implemented 72 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 73 // in the mapping from frame position to presentation time. 74 // See AudioTimestamp for the information included with event. 75#endif 76 }; 77 78 /* Client should declare a Buffer and pass the address to obtainBuffer() 79 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 80 */ 81 82 class Buffer 83 { 84 public: 85 // FIXME use m prefix 86 size_t frameCount; // number of sample frames corresponding to size; 87 // on input to obtainBuffer() it is the number of frames desired, 88 // on output from obtainBuffer() it is the number of available 89 // [empty slots for] frames to be filled 90 // on input to releaseBuffer() it is currently ignored 91 92 size_t size; // input/output in bytes == frameCount * frameSize 93 // on input to obtainBuffer() it is ignored 94 // on output from obtainBuffer() it is the number of available 95 // [empty slots for] bytes to be filled, 96 // which is frameCount * frameSize 97 // on input to releaseBuffer() it is the number of bytes to 98 // release 99 // FIXME This is redundant with respect to frameCount. Consider 100 // removing size and making frameCount the primary field. 101 102 union { 103 void* raw; 104 short* i16; // signed 16-bit 105 int8_t* i8; // unsigned 8-bit, offset by 0x80 106 }; // input to obtainBuffer(): unused, output: pointer to buffer 107 }; 108 109 /* As a convenience, if a callback is supplied, a handler thread 110 * is automatically created with the appropriate priority. This thread 111 * invokes the callback when a new buffer becomes available or various conditions occur. 112 * Parameters: 113 * 114 * event: type of event notified (see enum AudioTrack::event_type). 115 * user: Pointer to context for use by the callback receiver. 116 * info: Pointer to optional parameter according to event type: 117 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 118 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 119 * written. 120 * - EVENT_UNDERRUN: unused. 121 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 122 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 123 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 124 * - EVENT_BUFFER_END: unused. 125 * - EVENT_NEW_IAUDIOTRACK: unused. 126 * - EVENT_STREAM_END: unused. 127 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 128 */ 129 130 typedef void (*callback_t)(int event, void* user, void *info); 131 132 /* Returns the minimum frame count required for the successful creation of 133 * an AudioTrack object. 134 * Returned status (from utils/Errors.h) can be: 135 * - NO_ERROR: successful operation 136 * - NO_INIT: audio server or audio hardware not initialized 137 * - BAD_VALUE: unsupported configuration 138 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 139 * and is undefined otherwise. 140 * FIXME This API assumes a route, and so should be deprecated. 141 */ 142 143 static status_t getMinFrameCount(size_t* frameCount, 144 audio_stream_type_t streamType, 145 uint32_t sampleRate); 146 147 /* How data is transferred to AudioTrack 148 */ 149 enum transfer_type { 150 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 151 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 152 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 153 TRANSFER_SYNC, // synchronous write() 154 TRANSFER_SHARED, // shared memory 155 }; 156 157 /* Constructs an uninitialized AudioTrack. No connection with 158 * AudioFlinger takes place. Use set() after this. 159 */ 160 AudioTrack(); 161 162 /* Creates an AudioTrack object and registers it with AudioFlinger. 163 * Once created, the track needs to be started before it can be used. 164 * Unspecified values are set to appropriate default values. 165 * 166 * Parameters: 167 * 168 * streamType: Select the type of audio stream this track is attached to 169 * (e.g. AUDIO_STREAM_MUSIC). 170 * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate. 171 * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set. 172 * 0 will not work with current policy implementation for direct output 173 * selection where an exact match is needed for sampling rate. 174 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 175 * For direct and offloaded tracks, the possible format(s) depends on the 176 * output sink. 177 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 178 * frameCount: Minimum size of track PCM buffer in frames. This defines the 179 * application's contribution to the 180 * latency of the track. The actual size selected by the AudioTrack could be 181 * larger if the requested size is not compatible with current audio HAL 182 * configuration. Zero means to use a default value. 183 * flags: See comments on audio_output_flags_t in <system/audio.h>. 184 * cbf: Callback function. If not null, this function is called periodically 185 * to provide new data in TRANSFER_CALLBACK mode 186 * and inform of marker, position updates, etc. 187 * user: Context for use by the callback receiver. 188 * notificationFrames: The callback function is called each time notificationFrames PCM 189 * frames have been consumed from track input buffer by server. 190 * Zero means to use a default value, which is typically: 191 * - fast tracks: HAL buffer size, even if track frameCount is larger 192 * - normal tracks: 1/2 of track frameCount 193 * A positive value means that many frames at initial source sample rate. 194 * A negative value for this parameter specifies the negative of the 195 * requested number of notifications (sub-buffers) in the entire buffer. 196 * For fast tracks, the FastMixer will process one sub-buffer at a time. 197 * The size of each sub-buffer is determined by the HAL. 198 * To get "double buffering", for example, one should pass -2. 199 * The minimum number of sub-buffers is 1 (expressed as -1), 200 * and the maximum number of sub-buffers is 8 (expressed as -8). 201 * Negative is only permitted for fast tracks, and if frameCount is zero. 202 * TODO It is ugly to overload a parameter in this way depending on 203 * whether it is positive, negative, or zero. Consider splitting apart. 204 * sessionId: Specific session ID, or zero to use default. 205 * transferType: How data is transferred to AudioTrack. 206 * offloadInfo: If not NULL, provides offload parameters for 207 * AudioSystem::getOutputForAttr(). 208 * uid: User ID of the app which initially requested this AudioTrack 209 * for power management tracking, or -1 for current user ID. 210 * pid: Process ID of the app which initially requested this AudioTrack 211 * for power management tracking, or -1 for current process ID. 212 * pAttributes: If not NULL, supersedes streamType for use case selection. 213 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 214 binder to AudioFlinger. 215 It will return an error instead. The application will recreate 216 the track based on offloading or different channel configuration, etc. 217 * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow 218 * maxRequiredSpeed playback. Values less than 1.0f and greater than 219 * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks 220 * and direct or offloaded tracks, this parameter is ignored. 221 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 222 */ 223 224 AudioTrack( audio_stream_type_t streamType, 225 uint32_t sampleRate, 226 audio_format_t format, 227 audio_channel_mask_t channelMask, 228 size_t frameCount = 0, 229 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 230 callback_t cbf = NULL, 231 void* user = NULL, 232 int32_t notificationFrames = 0, 233 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 234 transfer_type transferType = TRANSFER_DEFAULT, 235 const audio_offload_info_t *offloadInfo = NULL, 236 uid_t uid = AUDIO_UID_INVALID, 237 pid_t pid = -1, 238 const audio_attributes_t* pAttributes = NULL, 239 bool doNotReconnect = false, 240 float maxRequiredSpeed = 1.0f); 241 242 /* Creates an audio track and registers it with AudioFlinger. 243 * With this constructor, the track is configured for static buffer mode. 244 * Data to be rendered is passed in a shared memory buffer 245 * identified by the argument sharedBuffer, which should be non-0. 246 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 247 * but without the ability to specify a non-zero value for the frameCount parameter. 248 * The memory should be initialized to the desired data before calling start(). 249 * The write() method is not supported in this case. 250 * It is recommended to pass a callback function to be notified of playback end by an 251 * EVENT_UNDERRUN event. 252 */ 253 254 AudioTrack( audio_stream_type_t streamType, 255 uint32_t sampleRate, 256 audio_format_t format, 257 audio_channel_mask_t channelMask, 258 const sp<IMemory>& sharedBuffer, 259 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 260 callback_t cbf = NULL, 261 void* user = NULL, 262 int32_t notificationFrames = 0, 263 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 264 transfer_type transferType = TRANSFER_DEFAULT, 265 const audio_offload_info_t *offloadInfo = NULL, 266 uid_t uid = AUDIO_UID_INVALID, 267 pid_t pid = -1, 268 const audio_attributes_t* pAttributes = NULL, 269 bool doNotReconnect = false, 270 float maxRequiredSpeed = 1.0f); 271 272 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 273 * Also destroys all resources associated with the AudioTrack. 274 */ 275protected: 276 virtual ~AudioTrack(); 277public: 278 279 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 280 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 281 * set() is not multi-thread safe. 282 * Returned status (from utils/Errors.h) can be: 283 * - NO_ERROR: successful initialization 284 * - INVALID_OPERATION: AudioTrack is already initialized 285 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 286 * - NO_INIT: audio server or audio hardware not initialized 287 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 288 * If sharedBuffer is non-0, the frameCount parameter is ignored and 289 * replaced by the shared buffer's total allocated size in frame units. 290 * 291 * Parameters not listed in the AudioTrack constructors above: 292 * 293 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 294 * 295 * Internal state post condition: 296 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 297 */ 298 status_t set(audio_stream_type_t streamType, 299 uint32_t sampleRate, 300 audio_format_t format, 301 audio_channel_mask_t channelMask, 302 size_t frameCount = 0, 303 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 304 callback_t cbf = NULL, 305 void* user = NULL, 306 int32_t notificationFrames = 0, 307 const sp<IMemory>& sharedBuffer = 0, 308 bool threadCanCallJava = false, 309 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 310 transfer_type transferType = TRANSFER_DEFAULT, 311 const audio_offload_info_t *offloadInfo = NULL, 312 uid_t uid = AUDIO_UID_INVALID, 313 pid_t pid = -1, 314 const audio_attributes_t* pAttributes = NULL, 315 bool doNotReconnect = false, 316 float maxRequiredSpeed = 1.0f); 317 318 /* Result of constructing the AudioTrack. This must be checked for successful initialization 319 * before using any AudioTrack API (except for set()), because using 320 * an uninitialized AudioTrack produces undefined results. 321 * See set() method above for possible return codes. 322 */ 323 status_t initCheck() const { return mStatus; } 324 325 /* Returns this track's estimated latency in milliseconds. 326 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 327 * and audio hardware driver. 328 */ 329 uint32_t latency() const { return mLatency; } 330 331 /* Returns the number of application-level buffer underruns 332 * since the AudioTrack was created. 333 */ 334 uint32_t getUnderrunCount() const; 335 336 /* getters, see constructors and set() */ 337 338 audio_stream_type_t streamType() const; 339 audio_format_t format() const { return mFormat; } 340 341 /* Return frame size in bytes, which for linear PCM is 342 * channelCount * (bit depth per channel / 8). 343 * channelCount is determined from channelMask, and bit depth comes from format. 344 * For non-linear formats, the frame size is typically 1 byte. 345 */ 346 size_t frameSize() const { return mFrameSize; } 347 348 uint32_t channelCount() const { return mChannelCount; } 349 size_t frameCount() const { return mFrameCount; } 350 351 /* 352 * Return the period of the notification callback in frames. 353 * This value is set when the AudioTrack is constructed. 354 * It can be modified if the AudioTrack is rerouted. 355 */ 356 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 357 358 /* Return effective size of audio buffer that an application writes to 359 * or a negative error if the track is uninitialized. 360 */ 361 ssize_t getBufferSizeInFrames(); 362 363 /* Returns the buffer duration in microseconds at current playback rate. 364 */ 365 status_t getBufferDurationInUs(int64_t *duration); 366 367 /* Set the effective size of audio buffer that an application writes to. 368 * This is used to determine the amount of available room in the buffer, 369 * which determines when a write will block. 370 * This allows an application to raise and lower the audio latency. 371 * The requested size may be adjusted so that it is 372 * greater or equal to the absolute minimum and 373 * less than or equal to the getBufferCapacityInFrames(). 374 * It may also be adjusted slightly for internal reasons. 375 * 376 * Return the final size or a negative error if the track is unitialized 377 * or does not support variable sizes. 378 */ 379 ssize_t setBufferSizeInFrames(size_t size); 380 381 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 382 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 383 384 /* After it's created the track is not active. Call start() to 385 * make it active. If set, the callback will start being called. 386 * If the track was previously paused, volume is ramped up over the first mix buffer. 387 */ 388 status_t start(); 389 390 /* Stop a track. 391 * In static buffer mode, the track is stopped immediately. 392 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 393 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 394 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 395 * is first drained, mixed, and output, and only then is the track marked as stopped. 396 */ 397 void stop(); 398 bool stopped() const; 399 400 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 401 * This has the effect of draining the buffers without mixing or output. 402 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 403 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 404 */ 405 void flush(); 406 407 /* Pause a track. After pause, the callback will cease being called and 408 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 409 * and will fill up buffers until the pool is exhausted. 410 * Volume is ramped down over the next mix buffer following the pause request, 411 * and then the track is marked as paused. It can be resumed with ramp up by start(). 412 */ 413 void pause(); 414 415 /* Set volume for this track, mostly used for games' sound effects 416 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 417 * This is the older API. New applications should use setVolume(float) when possible. 418 */ 419 status_t setVolume(float left, float right); 420 421 /* Set volume for all channels. This is the preferred API for new applications, 422 * especially for multi-channel content. 423 */ 424 status_t setVolume(float volume); 425 426 /* Set the send level for this track. An auxiliary effect should be attached 427 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 428 */ 429 status_t setAuxEffectSendLevel(float level); 430 void getAuxEffectSendLevel(float* level) const; 431 432 /* Set source sample rate for this track in Hz, mostly used for games' sound effects. 433 * Zero is not permitted. 434 */ 435 status_t setSampleRate(uint32_t sampleRate); 436 437 /* Return current source sample rate in Hz. 438 * If specified as zero in constructor or set(), this will be the sink sample rate. 439 */ 440 uint32_t getSampleRate() const; 441 442 /* Return the original source sample rate in Hz. This corresponds to the sample rate 443 * if playback rate had normal speed and pitch. 444 */ 445 uint32_t getOriginalSampleRate() const; 446 447 /* Set source playback rate for timestretch 448 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 449 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 450 * 451 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 452 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 453 * 454 * Speed increases the playback rate of media, but does not alter pitch. 455 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 456 */ 457 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 458 459 /* Return current playback rate */ 460 const AudioPlaybackRate& getPlaybackRate() const; 461 462 /* Enables looping and sets the start and end points of looping. 463 * Only supported for static buffer mode. 464 * 465 * Parameters: 466 * 467 * loopStart: loop start in frames relative to start of buffer. 468 * loopEnd: loop end in frames relative to start of buffer. 469 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 470 * pending or active loop. loopCount == -1 means infinite looping. 471 * 472 * For proper operation the following condition must be respected: 473 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 474 * 475 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 476 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 477 * 478 */ 479 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 480 481 /* Sets marker position. When playback reaches the number of frames specified, a callback with 482 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 483 * notification callback. To set a marker at a position which would compute as 0, 484 * a workaround is to set the marker at a nearby position such as ~0 or 1. 485 * If the AudioTrack has been opened with no callback function associated, the operation will 486 * fail. 487 * 488 * Parameters: 489 * 490 * marker: marker position expressed in wrapping (overflow) frame units, 491 * like the return value of getPosition(). 492 * 493 * Returned status (from utils/Errors.h) can be: 494 * - NO_ERROR: successful operation 495 * - INVALID_OPERATION: the AudioTrack has no callback installed. 496 */ 497 status_t setMarkerPosition(uint32_t marker); 498 status_t getMarkerPosition(uint32_t *marker) const; 499 500 /* Sets position update period. Every time the number of frames specified has been played, 501 * a callback with event type EVENT_NEW_POS is called. 502 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 503 * callback. 504 * If the AudioTrack has been opened with no callback function associated, the operation will 505 * fail. 506 * Extremely small values may be rounded up to a value the implementation can support. 507 * 508 * Parameters: 509 * 510 * updatePeriod: position update notification period expressed in frames. 511 * 512 * Returned status (from utils/Errors.h) can be: 513 * - NO_ERROR: successful operation 514 * - INVALID_OPERATION: the AudioTrack has no callback installed. 515 */ 516 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 517 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 518 519 /* Sets playback head position. 520 * Only supported for static buffer mode. 521 * 522 * Parameters: 523 * 524 * position: New playback head position in frames relative to start of buffer. 525 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 526 * but will result in an immediate underrun if started. 527 * 528 * Returned status (from utils/Errors.h) can be: 529 * - NO_ERROR: successful operation 530 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 531 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 532 * buffer 533 */ 534 status_t setPosition(uint32_t position); 535 536 /* Return the total number of frames played since playback start. 537 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 538 * It is reset to zero by flush(), reload(), and stop(). 539 * 540 * Parameters: 541 * 542 * position: Address where to return play head position. 543 * 544 * Returned status (from utils/Errors.h) can be: 545 * - NO_ERROR: successful operation 546 * - BAD_VALUE: position is NULL 547 */ 548 status_t getPosition(uint32_t *position); 549 550 /* For static buffer mode only, this returns the current playback position in frames 551 * relative to start of buffer. It is analogous to the position units used by 552 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 553 */ 554 status_t getBufferPosition(uint32_t *position); 555 556 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 557 * rewriting the buffer before restarting playback after a stop. 558 * This method must be called with the AudioTrack in paused or stopped state. 559 * Not allowed in streaming mode. 560 * 561 * Returned status (from utils/Errors.h) can be: 562 * - NO_ERROR: successful operation 563 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 564 */ 565 status_t reload(); 566 567 /* Returns a handle on the audio output used by this AudioTrack. 568 * 569 * Parameters: 570 * none. 571 * 572 * Returned value: 573 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 574 * track needed to be re-created but that failed 575 */ 576private: 577 audio_io_handle_t getOutput() const; 578public: 579 580 /* Selects the audio device to use for output of this AudioTrack. A value of 581 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 582 * 583 * Parameters: 584 * The device ID of the selected device (as returned by the AudioDevicesManager API). 585 * 586 * Returned value: 587 * - NO_ERROR: successful operation 588 * TODO: what else can happen here? 589 */ 590 status_t setOutputDevice(audio_port_handle_t deviceId); 591 592 /* Returns the ID of the audio device selected for this AudioTrack. 593 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 594 * 595 * Parameters: 596 * none. 597 */ 598 audio_port_handle_t getOutputDevice(); 599 600 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 601 * attached. 602 * A value of AUDIO_PORT_HANDLE_NONE indicates the audio track is not attached to any output. 603 * 604 * Parameters: 605 * none. 606 */ 607 audio_port_handle_t getRoutedDeviceId(); 608 609 /* Returns the unique session ID associated with this track. 610 * 611 * Parameters: 612 * none. 613 * 614 * Returned value: 615 * AudioTrack session ID. 616 */ 617 audio_session_t getSessionId() const { return mSessionId; } 618 619 /* Attach track auxiliary output to specified effect. Use effectId = 0 620 * to detach track from effect. 621 * 622 * Parameters: 623 * 624 * effectId: effectId obtained from AudioEffect::id(). 625 * 626 * Returned status (from utils/Errors.h) can be: 627 * - NO_ERROR: successful operation 628 * - INVALID_OPERATION: the effect is not an auxiliary effect. 629 * - BAD_VALUE: The specified effect ID is invalid 630 */ 631 status_t attachAuxEffect(int effectId); 632 633 /* Public API for TRANSFER_OBTAIN mode. 634 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 635 * After filling these slots with data, the caller should release them with releaseBuffer(). 636 * If the track buffer is not full, obtainBuffer() returns as many contiguous 637 * [empty slots for] frames as are available immediately. 638 * 639 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 640 * additional non-contiguous frames that are predicted to be available immediately, 641 * if the client were to release the first frames and then call obtainBuffer() again. 642 * This value is only a prediction, and needs to be confirmed. 643 * It will be set to zero for an error return. 644 * 645 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 646 * regardless of the value of waitCount. 647 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 648 * maximum timeout based on waitCount; see chart below. 649 * Buffers will be returned until the pool 650 * is exhausted, at which point obtainBuffer() will either block 651 * or return WOULD_BLOCK depending on the value of the "waitCount" 652 * parameter. 653 * 654 * Interpretation of waitCount: 655 * +n limits wait time to n * WAIT_PERIOD_MS, 656 * -1 causes an (almost) infinite wait time, 657 * 0 non-blocking. 658 * 659 * Buffer fields 660 * On entry: 661 * frameCount number of [empty slots for] frames requested 662 * size ignored 663 * raw ignored 664 * After error return: 665 * frameCount 0 666 * size 0 667 * raw undefined 668 * After successful return: 669 * frameCount actual number of [empty slots for] frames available, <= number requested 670 * size actual number of bytes available 671 * raw pointer to the buffer 672 */ 673 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 674 size_t *nonContig = NULL); 675 676private: 677 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 678 * additional non-contiguous frames that are predicted to be available immediately, 679 * if the client were to release the first frames and then call obtainBuffer() again. 680 * This value is only a prediction, and needs to be confirmed. 681 * It will be set to zero for an error return. 682 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 683 * in case the requested amount of frames is in two or more non-contiguous regions. 684 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 685 */ 686 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 687 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 688public: 689 690 /* Public API for TRANSFER_OBTAIN mode. 691 * Release a filled buffer of frames for AudioFlinger to process. 692 * 693 * Buffer fields: 694 * frameCount currently ignored but recommend to set to actual number of frames filled 695 * size actual number of bytes filled, must be multiple of frameSize 696 * raw ignored 697 */ 698 void releaseBuffer(const Buffer* audioBuffer); 699 700 /* As a convenience we provide a write() interface to the audio buffer. 701 * Input parameter 'size' is in byte units. 702 * This is implemented on top of obtainBuffer/releaseBuffer. For best 703 * performance use callbacks. Returns actual number of bytes written >= 0, 704 * or one of the following negative status codes: 705 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 706 * BAD_VALUE size is invalid 707 * WOULD_BLOCK when obtainBuffer() returns same, or 708 * AudioTrack was stopped during the write 709 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 710 * the track cannot be automatically restored. 711 * The application needs to recreate the AudioTrack 712 * because the audio device changed or AudioFlinger died. 713 * This typically occurs for direct or offload tracks 714 * or if mDoNotReconnect is true. 715 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 716 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 717 * false for the method to return immediately without waiting to try multiple times to write 718 * the full content of the buffer. 719 */ 720 ssize_t write(const void* buffer, size_t size, bool blocking = true); 721 722 /* 723 * Dumps the state of an audio track. 724 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 725 */ 726 status_t dump(int fd, const Vector<String16>& args) const; 727 728 /* 729 * Return the total number of frames which AudioFlinger desired but were unavailable, 730 * and thus which resulted in an underrun. Reset to zero by stop(). 731 */ 732 uint32_t getUnderrunFrames() const; 733 734 /* Get the flags */ 735 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 736 737 /* Set parameters - only possible when using direct output */ 738 status_t setParameters(const String8& keyValuePairs); 739 740 /* Sets the volume shaper object */ 741 VolumeShaper::Status applyVolumeShaper( 742 const sp<VolumeShaper::Configuration>& configuration, 743 const sp<VolumeShaper::Operation>& operation); 744 745 /* Gets the volume shaper state */ 746 sp<VolumeShaper::State> getVolumeShaperState(int id); 747 748 /* Get parameters */ 749 String8 getParameters(const String8& keys); 750 751 /* Poll for a timestamp on demand. 752 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 753 * or if you need to get the most recent timestamp outside of the event callback handler. 754 * Caution: calling this method too often may be inefficient; 755 * if you need a high resolution mapping between frame position and presentation time, 756 * consider implementing that at application level, based on the low resolution timestamps. 757 * Returns NO_ERROR if timestamp is valid. 758 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 759 * start/ACTIVE, when the number of frames consumed is less than the 760 * overall hardware latency to physical output. In WOULD_BLOCK cases, 761 * one might poll again, or use getPosition(), or use 0 position and 762 * current time for the timestamp. 763 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 764 * the track cannot be automatically restored. 765 * The application needs to recreate the AudioTrack 766 * because the audio device changed or AudioFlinger died. 767 * This typically occurs for direct or offload tracks 768 * or if mDoNotReconnect is true. 769 * INVALID_OPERATION wrong state, or some other error. 770 * 771 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 772 */ 773 status_t getTimestamp(AudioTimestamp& timestamp); 774private: 775 status_t getTimestamp_l(AudioTimestamp& timestamp); 776public: 777 778 /* Return the extended timestamp, with additional timebase info and improved drain behavior. 779 * 780 * This is similar to the AudioTrack.java API: 781 * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) 782 * 783 * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method 784 * 785 * 1. stop() by itself does not reset the frame position. 786 * A following start() resets the frame position to 0. 787 * 2. flush() by itself does not reset the frame position. 788 * The frame position advances by the number of frames flushed, 789 * when the first frame after flush reaches the audio sink. 790 * 3. BOOTTIME clock offsets are provided to help synchronize with 791 * non-audio streams, e.g. sensor data. 792 * 4. Position is returned with 64 bits of resolution. 793 * 794 * Parameters: 795 * timestamp: A pointer to the caller allocated ExtendedTimestamp. 796 * 797 * Returns NO_ERROR on success; timestamp is filled with valid data. 798 * BAD_VALUE if timestamp is NULL. 799 * WOULD_BLOCK if called immediately after start() when the number 800 * of frames consumed is less than the 801 * overall hardware latency to physical output. In WOULD_BLOCK cases, 802 * one might poll again, or use getPosition(), or use 0 position and 803 * current time for the timestamp. 804 * If WOULD_BLOCK is returned, the timestamp is still 805 * modified with the LOCATION_CLIENT portion filled. 806 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 807 * the track cannot be automatically restored. 808 * The application needs to recreate the AudioTrack 809 * because the audio device changed or AudioFlinger died. 810 * This typically occurs for direct or offloaded tracks 811 * or if mDoNotReconnect is true. 812 * INVALID_OPERATION if called on a offloaded or direct track. 813 * Use getTimestamp(AudioTimestamp& timestamp) instead. 814 */ 815 status_t getTimestamp(ExtendedTimestamp *timestamp); 816private: 817 status_t getTimestamp_l(ExtendedTimestamp *timestamp); 818public: 819 820 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 821 * AudioTrack is routed is updated. 822 * Replaces any previously installed callback. 823 * Parameters: 824 * callback: The callback interface 825 * Returns NO_ERROR if successful. 826 * INVALID_OPERATION if the same callback is already installed. 827 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 828 * BAD_VALUE if the callback is NULL 829 */ 830 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 831 832 /* remove an AudioDeviceCallback. 833 * Parameters: 834 * callback: The callback interface 835 * Returns NO_ERROR if successful. 836 * INVALID_OPERATION if the callback is not installed 837 * BAD_VALUE if the callback is NULL 838 */ 839 status_t removeAudioDeviceCallback( 840 const sp<AudioSystem::AudioDeviceCallback>& callback); 841 842 /* Obtain the pending duration in milliseconds for playback of pure PCM 843 * (mixable without embedded timing) data remaining in AudioTrack. 844 * 845 * This is used to estimate the drain time for the client-server buffer 846 * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. 847 * One may optionally request to find the duration to play through the HAL 848 * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, 849 * INVALID_OPERATION may be returned if the kernel location is unavailable. 850 * 851 * Returns NO_ERROR if successful. 852 * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained 853 * or the AudioTrack does not contain pure PCM data. 854 * BAD_VALUE if msec is nullptr or location is invalid. 855 */ 856 status_t pendingDuration(int32_t *msec, 857 ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); 858 859 /* hasStarted() is used to determine if audio is now audible at the device after 860 * a start() command. The underlying implementation checks a nonzero timestamp position 861 * or increment for the audible assumption. 862 * 863 * hasStarted() returns true if the track has been started() and audio is audible 864 * and no subsequent pause() or flush() has been called. Immediately after pause() or 865 * flush() hasStarted() will return false. 866 * 867 * If stop() has been called, hasStarted() will return true if audio is still being 868 * delivered or has finished delivery (even if no audio was written) for both offloaded 869 * and normal tracks. This property removes a race condition in checking hasStarted() 870 * for very short clips, where stop() must be called to finish drain. 871 * 872 * In all cases, hasStarted() may turn false briefly after a subsequent start() is called 873 * until audio becomes audible again. 874 */ 875 bool hasStarted(); // not const 876 877protected: 878 /* copying audio tracks is not allowed */ 879 AudioTrack(const AudioTrack& other); 880 AudioTrack& operator = (const AudioTrack& other); 881 882 /* a small internal class to handle the callback */ 883 class AudioTrackThread : public Thread 884 { 885 public: 886 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 887 888 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 889 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 890 virtual void requestExit(); 891 892 void pause(); // suspend thread from execution at next loop boundary 893 void resume(); // allow thread to execute, if not requested to exit 894 void wake(); // wake to handle changed notification conditions. 895 896 private: 897 void pauseInternal(nsecs_t ns = 0LL); 898 // like pause(), but only used internally within thread 899 900 friend class AudioTrack; 901 virtual bool threadLoop(); 902 AudioTrack& mReceiver; 903 virtual ~AudioTrackThread(); 904 Mutex mMyLock; // Thread::mLock is private 905 Condition mMyCond; // Thread::mThreadExitedCondition is private 906 bool mPaused; // whether thread is requested to pause at next loop entry 907 bool mPausedInt; // whether thread internally requests pause 908 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 909 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 910 // to processAudioBuffer() as state may have changed 911 // since pause time calculated. 912 }; 913 914 // body of AudioTrackThread::threadLoop() 915 // returns the maximum amount of time before we would like to run again, where: 916 // 0 immediately 917 // > 0 no later than this many nanoseconds from now 918 // NS_WHENEVER still active but no particular deadline 919 // NS_INACTIVE inactive so don't run again until re-started 920 // NS_NEVER never again 921 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 922 nsecs_t processAudioBuffer(); 923 924 // caller must hold lock on mLock for all _l methods 925 926 status_t createTrack_l(); 927 928 // can only be called when mState != STATE_ACTIVE 929 void flush_l(); 930 931 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 932 933 // FIXME enum is faster than strcmp() for parameter 'from' 934 status_t restoreTrack_l(const char *from); 935 936 uint32_t getUnderrunCount_l() const; 937 938 bool isOffloaded() const; 939 bool isDirect() const; 940 bool isOffloadedOrDirect() const; 941 942 bool isOffloaded_l() const 943 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 944 945 bool isOffloadedOrDirect_l() const 946 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 947 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 948 949 bool isDirect_l() const 950 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 951 952 // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) 953 bool isPurePcmData_l() const 954 { return audio_is_linear_pcm(mFormat) 955 && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } 956 957 // increment mPosition by the delta of mServer, and return new value of mPosition 958 Modulo<uint32_t> updateAndGetPosition_l(); 959 960 // check sample rate and speed is compatible with AudioTrack 961 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; 962 963 void restartIfDisabled(); 964 965 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 966 sp<IAudioTrack> mAudioTrack; 967 sp<IMemory> mCblkMemory; 968 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 969 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 970 971 sp<AudioTrackThread> mAudioTrackThread; 972 bool mThreadCanCallJava; 973 974 float mVolume[2]; 975 float mSendLevel; 976 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 977 uint32_t mOriginalSampleRate; 978 AudioPlaybackRate mPlaybackRate; 979 float mMaxRequiredSpeed; // use PCM buffer size to allow this speed 980 981 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 982 // This allocated buffer size is maintained by the proxy. 983 size_t mFrameCount; // maximum size of buffer 984 985 size_t mReqFrameCount; // frame count to request the first or next time 986 // a new IAudioTrack is needed, non-decreasing 987 988 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 989 // These values can be used for informational purposes until the track is invalidated, 990 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 991 uint32_t mAfLatency; // AudioFlinger latency in ms 992 size_t mAfFrameCount; // AudioFlinger frame count 993 uint32_t mAfSampleRate; // AudioFlinger sample rate 994 995 // constant after constructor or set() 996 audio_format_t mFormat; // as requested by client, not forced to 16-bit 997 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 998 // this AudioTrack has valid attributes 999 uint32_t mChannelCount; 1000 audio_channel_mask_t mChannelMask; 1001 sp<IMemory> mSharedBuffer; 1002 transfer_type mTransfer; 1003 audio_offload_info_t mOffloadInfoCopy; 1004 const audio_offload_info_t* mOffloadInfo; 1005 audio_attributes_t mAttributes; 1006 1007 size_t mFrameSize; // frame size in bytes 1008 1009 status_t mStatus; 1010 1011 // can change dynamically when IAudioTrack invalidated 1012 uint32_t mLatency; // in ms 1013 1014 // Indicates the current track state. Protected by mLock. 1015 enum State { 1016 STATE_ACTIVE, 1017 STATE_STOPPED, 1018 STATE_PAUSED, 1019 STATE_PAUSED_STOPPING, 1020 STATE_FLUSHED, 1021 STATE_STOPPING, 1022 } mState; 1023 1024 // for client callback handler 1025 callback_t mCbf; // callback handler for events, or NULL 1026 void* mUserData; 1027 1028 // for notification APIs 1029 1030 // next 2 fields are const after constructor or set() 1031 uint32_t mNotificationFramesReq; // requested number of frames between each 1032 // notification callback, 1033 // at initial source sample rate 1034 uint32_t mNotificationsPerBufferReq; 1035 // requested number of notifications per buffer, 1036 // currently only used for fast tracks with 1037 // default track buffer size 1038 1039 uint32_t mNotificationFramesAct; // actual number of frames between each 1040 // notification callback, 1041 // at initial source sample rate 1042 bool mRefreshRemaining; // processAudioBuffer() should refresh 1043 // mRemainingFrames and mRetryOnPartialBuffer 1044 1045 // used for static track cbf and restoration 1046 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 1047 uint32_t mLoopStart; // last setLoop loopStart 1048 uint32_t mLoopEnd; // last setLoop loopEnd 1049 int32_t mLoopCountNotified; // the last loopCount notified by callback. 1050 // mLoopCountNotified counts down, matching 1051 // the remaining loop count for static track 1052 // playback. 1053 1054 // These are private to processAudioBuffer(), and are not protected by a lock 1055 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 1056 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 1057 uint32_t mObservedSequence; // last observed value of mSequence 1058 1059 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 1060 bool mMarkerReached; 1061 Modulo<uint32_t> mNewPosition; // in frames 1062 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 1063 1064 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 1065 // which is count of frames consumed by server, 1066 // reset by new IAudioTrack, 1067 // whether it is reset by stop() is TBD 1068 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 1069 // monotonically after new IAudioTrack, 1070 // and could be easily widened to uint64_t 1071 Modulo<uint32_t> mReleased; // count of frames released to server 1072 // but not necessarily consumed by server, 1073 // reset by stop() but continues monotonically 1074 // after new IAudioTrack to restore mPosition, 1075 // and could be easily widened to uint64_t 1076 int64_t mStartUs; // the start time after flush or stop. 1077 // only used for offloaded and direct tracks. 1078 ExtendedTimestamp mStartEts; // Extended timestamp at start for normal 1079 // AudioTracks. 1080 AudioTimestamp mStartTs; // Timestamp at start for offloaded or direct 1081 // AudioTracks. 1082 1083 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 1084 bool mTimestampStartupGlitchReported; // reduce log spam 1085 bool mRetrogradeMotionReported; // reduce log spam 1086 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 1087 ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp 1088 1089 uint32_t mUnderrunCountOffset; // updated when restoring tracks 1090 1091 int64_t mFramesWritten; // total frames written. reset to zero after 1092 // the start() following stop(). It is not 1093 // changed after restoring the track or 1094 // after flush. 1095 int64_t mFramesWrittenServerOffset; // An offset to server frames due to 1096 // restoring AudioTrack, or stop/start. 1097 // This offset is also used for static tracks. 1098 int64_t mFramesWrittenAtRestore; // Frames written at restore point (or frames 1099 // delivered for static tracks). 1100 // -1 indicates no previous restore point. 1101 1102 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 1103 // be denied by client or server, such as 1104 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 1105 // held to read or write those bits reliably. 1106 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 1107 1108 bool mDoNotReconnect; 1109 1110 audio_session_t mSessionId; 1111 int mAuxEffectId; 1112 1113 mutable Mutex mLock; 1114 1115 int mPreviousPriority; // before start() 1116 SchedPolicy mPreviousSchedulingGroup; 1117 bool mAwaitBoost; // thread should wait for priority boost before running 1118 1119 // The proxy should only be referenced while a lock is held because the proxy isn't 1120 // multi-thread safe, especially the SingleStateQueue part of the proxy. 1121 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 1122 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 1123 // them around in case they are replaced during the obtainBuffer(). 1124 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 1125 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 1126 1127 bool mInUnderrun; // whether track is currently in underrun state 1128 uint32_t mPausedPosition; 1129 1130 // For Device Selection API 1131 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 1132 audio_port_handle_t mSelectedDeviceId; 1133 1134 sp<VolumeHandler> mVolumeHandler; 1135 1136private: 1137 class DeathNotifier : public IBinder::DeathRecipient { 1138 public: 1139 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 1140 protected: 1141 virtual void binderDied(const wp<IBinder>& who); 1142 private: 1143 const wp<AudioTrack> mAudioTrack; 1144 }; 1145 1146 sp<DeathNotifier> mDeathNotifier; 1147 uint32_t mSequence; // incremented for each new IAudioTrack attempt 1148 uid_t mClientUid; 1149 pid_t mClientPid; 1150 1151 sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1152 audio_port_handle_t mPortId; // unique ID allocated by audio policy 1153}; 1154 1155}; // namespace android 1156 1157#endif // ANDROID_AUDIOTRACK_H 1158