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 28 namespace android { 29 30 // ---------------------------------------------------------------------------- 31 32 struct audio_track_cblk_t; 33 class AudioTrackClientProxy; 34 class StaticAudioTrackClientProxy; 35 36 // ---------------------------------------------------------------------------- 37 38 class AudioTrack : public AudioSystem::AudioDeviceCallback 39 { 40 public: 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 */ 275 protected: 276 virtual ~AudioTrack(); 277 public: 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(); 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 /** 568 * @param transferType 569 * @return text string that matches the enum name 570 */ 571 static const char * convertTransferToText(transfer_type transferType); 572 573 /* Returns a handle on the audio output used by this AudioTrack. 574 * 575 * Parameters: 576 * none. 577 * 578 * Returned value: 579 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 580 * track needed to be re-created but that failed 581 */ 582 private: 583 audio_io_handle_t getOutput() const; 584 public: 585 586 /* Selects the audio device to use for output of this AudioTrack. A value of 587 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 588 * 589 * Parameters: 590 * The device ID of the selected device (as returned by the AudioDevicesManager API). 591 * 592 * Returned value: 593 * - NO_ERROR: successful operation 594 * TODO: what else can happen here? 595 */ 596 status_t setOutputDevice(audio_port_handle_t deviceId); 597 598 /* Returns the ID of the audio device selected for this AudioTrack. 599 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 600 * 601 * Parameters: 602 * none. 603 */ 604 audio_port_handle_t getOutputDevice(); 605 606 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 607 * attached. 608 * When the AudioTrack is inactive, the device ID returned can be either: 609 * - AUDIO_PORT_HANDLE_NONE if the AudioTrack is not attached to any output. 610 * - The device ID used before paused or stopped. 611 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioTrack 612 * has not been started yet. 613 * 614 * Parameters: 615 * none. 616 */ 617 audio_port_handle_t getRoutedDeviceId(); 618 619 /* Returns the unique session ID associated with this track. 620 * 621 * Parameters: 622 * none. 623 * 624 * Returned value: 625 * AudioTrack session ID. 626 */ 627 audio_session_t getSessionId() const { return mSessionId; } 628 629 /* Attach track auxiliary output to specified effect. Use effectId = 0 630 * to detach track from effect. 631 * 632 * Parameters: 633 * 634 * effectId: effectId obtained from AudioEffect::id(). 635 * 636 * Returned status (from utils/Errors.h) can be: 637 * - NO_ERROR: successful operation 638 * - INVALID_OPERATION: the effect is not an auxiliary effect. 639 * - BAD_VALUE: The specified effect ID is invalid 640 */ 641 status_t attachAuxEffect(int effectId); 642 643 /* Public API for TRANSFER_OBTAIN mode. 644 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 645 * After filling these slots with data, the caller should release them with releaseBuffer(). 646 * If the track buffer is not full, obtainBuffer() returns as many contiguous 647 * [empty slots for] frames as are available immediately. 648 * 649 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 650 * additional non-contiguous frames that are predicted to be available immediately, 651 * if the client were to release the first frames and then call obtainBuffer() again. 652 * This value is only a prediction, and needs to be confirmed. 653 * It will be set to zero for an error return. 654 * 655 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 656 * regardless of the value of waitCount. 657 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 658 * maximum timeout based on waitCount; see chart below. 659 * Buffers will be returned until the pool 660 * is exhausted, at which point obtainBuffer() will either block 661 * or return WOULD_BLOCK depending on the value of the "waitCount" 662 * parameter. 663 * 664 * Interpretation of waitCount: 665 * +n limits wait time to n * WAIT_PERIOD_MS, 666 * -1 causes an (almost) infinite wait time, 667 * 0 non-blocking. 668 * 669 * Buffer fields 670 * On entry: 671 * frameCount number of [empty slots for] frames requested 672 * size ignored 673 * raw ignored 674 * After error return: 675 * frameCount 0 676 * size 0 677 * raw undefined 678 * After successful return: 679 * frameCount actual number of [empty slots for] frames available, <= number requested 680 * size actual number of bytes available 681 * raw pointer to the buffer 682 */ 683 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 684 size_t *nonContig = NULL); 685 686 private: 687 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 688 * additional non-contiguous frames that are predicted to be available immediately, 689 * if the client were to release the first frames and then call obtainBuffer() again. 690 * This value is only a prediction, and needs to be confirmed. 691 * It will be set to zero for an error return. 692 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 693 * in case the requested amount of frames is in two or more non-contiguous regions. 694 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 695 */ 696 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 697 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 698 public: 699 700 /* Public API for TRANSFER_OBTAIN mode. 701 * Release a filled buffer of frames for AudioFlinger to process. 702 * 703 * Buffer fields: 704 * frameCount currently ignored but recommend to set to actual number of frames filled 705 * size actual number of bytes filled, must be multiple of frameSize 706 * raw ignored 707 */ 708 void releaseBuffer(const Buffer* audioBuffer); 709 710 /* As a convenience we provide a write() interface to the audio buffer. 711 * Input parameter 'size' is in byte units. 712 * This is implemented on top of obtainBuffer/releaseBuffer. For best 713 * performance use callbacks. Returns actual number of bytes written >= 0, 714 * or one of the following negative status codes: 715 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 716 * BAD_VALUE size is invalid 717 * WOULD_BLOCK when obtainBuffer() returns same, or 718 * AudioTrack was stopped during the write 719 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 720 * the track cannot be automatically restored. 721 * The application needs to recreate the AudioTrack 722 * because the audio device changed or AudioFlinger died. 723 * This typically occurs for direct or offload tracks 724 * or if mDoNotReconnect is true. 725 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 726 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 727 * false for the method to return immediately without waiting to try multiple times to write 728 * the full content of the buffer. 729 */ 730 ssize_t write(const void* buffer, size_t size, bool blocking = true); 731 732 /* 733 * Dumps the state of an audio track. 734 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 735 */ 736 status_t dump(int fd, const Vector<String16>& args) const; 737 738 /* 739 * Return the total number of frames which AudioFlinger desired but were unavailable, 740 * and thus which resulted in an underrun. Reset to zero by stop(). 741 */ 742 uint32_t getUnderrunFrames() const; 743 744 /* Get the flags */ 745 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 746 747 /* Set parameters - only possible when using direct output */ 748 status_t setParameters(const String8& keyValuePairs); 749 750 /* Sets the volume shaper object */ 751 VolumeShaper::Status applyVolumeShaper( 752 const sp<VolumeShaper::Configuration>& configuration, 753 const sp<VolumeShaper::Operation>& operation); 754 755 /* Gets the volume shaper state */ 756 sp<VolumeShaper::State> getVolumeShaperState(int id); 757 758 /* Get parameters */ 759 String8 getParameters(const String8& keys); 760 761 /* Poll for a timestamp on demand. 762 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 763 * or if you need to get the most recent timestamp outside of the event callback handler. 764 * Caution: calling this method too often may be inefficient; 765 * if you need a high resolution mapping between frame position and presentation time, 766 * consider implementing that at application level, based on the low resolution timestamps. 767 * Returns NO_ERROR if timestamp is valid. 768 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 769 * start/ACTIVE, when the number of frames consumed is less than the 770 * overall hardware latency to physical output. In WOULD_BLOCK cases, 771 * one might poll again, or use getPosition(), or use 0 position and 772 * current time for the timestamp. 773 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 774 * the track cannot be automatically restored. 775 * The application needs to recreate the AudioTrack 776 * because the audio device changed or AudioFlinger died. 777 * This typically occurs for direct or offload tracks 778 * or if mDoNotReconnect is true. 779 * INVALID_OPERATION wrong state, or some other error. 780 * 781 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 782 */ 783 status_t getTimestamp(AudioTimestamp& timestamp); 784 private: 785 status_t getTimestamp_l(AudioTimestamp& timestamp); 786 public: 787 788 /* Return the extended timestamp, with additional timebase info and improved drain behavior. 789 * 790 * This is similar to the AudioTrack.java API: 791 * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) 792 * 793 * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method 794 * 795 * 1. stop() by itself does not reset the frame position. 796 * A following start() resets the frame position to 0. 797 * 2. flush() by itself does not reset the frame position. 798 * The frame position advances by the number of frames flushed, 799 * when the first frame after flush reaches the audio sink. 800 * 3. BOOTTIME clock offsets are provided to help synchronize with 801 * non-audio streams, e.g. sensor data. 802 * 4. Position is returned with 64 bits of resolution. 803 * 804 * Parameters: 805 * timestamp: A pointer to the caller allocated ExtendedTimestamp. 806 * 807 * Returns NO_ERROR on success; timestamp is filled with valid data. 808 * BAD_VALUE if timestamp is NULL. 809 * WOULD_BLOCK if called immediately after start() when the number 810 * of frames consumed is less than the 811 * overall hardware latency to physical output. In WOULD_BLOCK cases, 812 * one might poll again, or use getPosition(), or use 0 position and 813 * current time for the timestamp. 814 * If WOULD_BLOCK is returned, the timestamp is still 815 * modified with the LOCATION_CLIENT portion filled. 816 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 817 * the track cannot be automatically restored. 818 * The application needs to recreate the AudioTrack 819 * because the audio device changed or AudioFlinger died. 820 * This typically occurs for direct or offloaded tracks 821 * or if mDoNotReconnect is true. 822 * INVALID_OPERATION if called on a offloaded or direct track. 823 * Use getTimestamp(AudioTimestamp& timestamp) instead. 824 */ 825 status_t getTimestamp(ExtendedTimestamp *timestamp); 826 private: 827 status_t getTimestamp_l(ExtendedTimestamp *timestamp); 828 public: 829 830 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 831 * AudioTrack is routed is updated. 832 * Replaces any previously installed callback. 833 * Parameters: 834 * callback: The callback interface 835 * Returns NO_ERROR if successful. 836 * INVALID_OPERATION if the same callback is already installed. 837 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 838 * BAD_VALUE if the callback is NULL 839 */ 840 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 841 842 /* remove an AudioDeviceCallback. 843 * Parameters: 844 * callback: The callback interface 845 * Returns NO_ERROR if successful. 846 * INVALID_OPERATION if the callback is not installed 847 * BAD_VALUE if the callback is NULL 848 */ 849 status_t removeAudioDeviceCallback( 850 const sp<AudioSystem::AudioDeviceCallback>& callback); 851 852 // AudioSystem::AudioDeviceCallback> virtuals 853 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 854 audio_port_handle_t deviceId); 855 856 857 858 /* Obtain the pending duration in milliseconds for playback of pure PCM 859 * (mixable without embedded timing) data remaining in AudioTrack. 860 * 861 * This is used to estimate the drain time for the client-server buffer 862 * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. 863 * One may optionally request to find the duration to play through the HAL 864 * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, 865 * INVALID_OPERATION may be returned if the kernel location is unavailable. 866 * 867 * Returns NO_ERROR if successful. 868 * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained 869 * or the AudioTrack does not contain pure PCM data. 870 * BAD_VALUE if msec is nullptr or location is invalid. 871 */ 872 status_t pendingDuration(int32_t *msec, 873 ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); 874 875 /* hasStarted() is used to determine if audio is now audible at the device after 876 * a start() command. The underlying implementation checks a nonzero timestamp position 877 * or increment for the audible assumption. 878 * 879 * hasStarted() returns true if the track has been started() and audio is audible 880 * and no subsequent pause() or flush() has been called. Immediately after pause() or 881 * flush() hasStarted() will return false. 882 * 883 * If stop() has been called, hasStarted() will return true if audio is still being 884 * delivered or has finished delivery (even if no audio was written) for both offloaded 885 * and normal tracks. This property removes a race condition in checking hasStarted() 886 * for very short clips, where stop() must be called to finish drain. 887 * 888 * In all cases, hasStarted() may turn false briefly after a subsequent start() is called 889 * until audio becomes audible again. 890 */ 891 bool hasStarted(); // not const 892 893 bool isPlaying() { 894 AutoMutex lock(mLock); 895 return mState == STATE_ACTIVE || mState == STATE_STOPPING; 896 } 897 protected: 898 /* copying audio tracks is not allowed */ 899 AudioTrack(const AudioTrack& other); 900 AudioTrack& operator = (const AudioTrack& other); 901 902 /* a small internal class to handle the callback */ 903 class AudioTrackThread : public Thread 904 { 905 public: 906 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 907 908 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 909 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 910 virtual void requestExit(); 911 912 void pause(); // suspend thread from execution at next loop boundary 913 void resume(); // allow thread to execute, if not requested to exit 914 void wake(); // wake to handle changed notification conditions. 915 916 private: 917 void pauseInternal(nsecs_t ns = 0LL); 918 // like pause(), but only used internally within thread 919 920 friend class AudioTrack; 921 virtual bool threadLoop(); 922 AudioTrack& mReceiver; 923 virtual ~AudioTrackThread(); 924 Mutex mMyLock; // Thread::mLock is private 925 Condition mMyCond; // Thread::mThreadExitedCondition is private 926 bool mPaused; // whether thread is requested to pause at next loop entry 927 bool mPausedInt; // whether thread internally requests pause 928 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 929 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 930 // to processAudioBuffer() as state may have changed 931 // since pause time calculated. 932 }; 933 934 // body of AudioTrackThread::threadLoop() 935 // returns the maximum amount of time before we would like to run again, where: 936 // 0 immediately 937 // > 0 no later than this many nanoseconds from now 938 // NS_WHENEVER still active but no particular deadline 939 // NS_INACTIVE inactive so don't run again until re-started 940 // NS_NEVER never again 941 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 942 nsecs_t processAudioBuffer(); 943 944 // caller must hold lock on mLock for all _l methods 945 946 void updateLatency_l(); // updates mAfLatency and mLatency from AudioSystem cache 947 948 status_t createTrack_l(); 949 950 // can only be called when mState != STATE_ACTIVE 951 void flush_l(); 952 953 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 954 955 // FIXME enum is faster than strcmp() for parameter 'from' 956 status_t restoreTrack_l(const char *from); 957 958 uint32_t getUnderrunCount_l() const; 959 960 bool isOffloaded() const; 961 bool isDirect() const; 962 bool isOffloadedOrDirect() const; 963 964 bool isOffloaded_l() const 965 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 966 967 bool isOffloadedOrDirect_l() const 968 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 969 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 970 971 bool isDirect_l() const 972 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 973 974 // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) 975 bool isPurePcmData_l() const 976 { return audio_is_linear_pcm(mFormat) 977 && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } 978 979 // increment mPosition by the delta of mServer, and return new value of mPosition 980 Modulo<uint32_t> updateAndGetPosition_l(); 981 982 // check sample rate and speed is compatible with AudioTrack 983 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed); 984 985 void restartIfDisabled(); 986 987 void updateRoutedDeviceId_l(); 988 989 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 990 sp<IAudioTrack> mAudioTrack; 991 sp<IMemory> mCblkMemory; 992 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 993 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 994 995 sp<AudioTrackThread> mAudioTrackThread; 996 bool mThreadCanCallJava; 997 998 float mVolume[2]; 999 float mSendLevel; 1000 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 1001 uint32_t mOriginalSampleRate; 1002 AudioPlaybackRate mPlaybackRate; 1003 float mMaxRequiredSpeed; // use PCM buffer size to allow this speed 1004 1005 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 1006 // This allocated buffer size is maintained by the proxy. 1007 size_t mFrameCount; // maximum size of buffer 1008 1009 size_t mReqFrameCount; // frame count to request the first or next time 1010 // a new IAudioTrack is needed, non-decreasing 1011 1012 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 1013 // These values can be used for informational purposes until the track is invalidated, 1014 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 1015 uint32_t mAfLatency; // AudioFlinger latency in ms 1016 size_t mAfFrameCount; // AudioFlinger frame count 1017 uint32_t mAfSampleRate; // AudioFlinger sample rate 1018 1019 // constant after constructor or set() 1020 audio_format_t mFormat; // as requested by client, not forced to 16-bit 1021 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 1022 // this AudioTrack has valid attributes 1023 uint32_t mChannelCount; 1024 audio_channel_mask_t mChannelMask; 1025 sp<IMemory> mSharedBuffer; 1026 transfer_type mTransfer; 1027 audio_offload_info_t mOffloadInfoCopy; 1028 const audio_offload_info_t* mOffloadInfo; 1029 audio_attributes_t mAttributes; 1030 1031 size_t mFrameSize; // frame size in bytes 1032 1033 status_t mStatus; 1034 1035 // can change dynamically when IAudioTrack invalidated 1036 uint32_t mLatency; // in ms 1037 1038 // Indicates the current track state. Protected by mLock. 1039 enum State { 1040 STATE_ACTIVE, 1041 STATE_STOPPED, 1042 STATE_PAUSED, 1043 STATE_PAUSED_STOPPING, 1044 STATE_FLUSHED, 1045 STATE_STOPPING, 1046 } mState; 1047 1048 // for client callback handler 1049 callback_t mCbf; // callback handler for events, or NULL 1050 void* mUserData; 1051 1052 // for notification APIs 1053 1054 // next 2 fields are const after constructor or set() 1055 uint32_t mNotificationFramesReq; // requested number of frames between each 1056 // notification callback, 1057 // at initial source sample rate 1058 uint32_t mNotificationsPerBufferReq; 1059 // requested number of notifications per buffer, 1060 // currently only used for fast tracks with 1061 // default track buffer size 1062 1063 uint32_t mNotificationFramesAct; // actual number of frames between each 1064 // notification callback, 1065 // at initial source sample rate 1066 bool mRefreshRemaining; // processAudioBuffer() should refresh 1067 // mRemainingFrames and mRetryOnPartialBuffer 1068 1069 // used for static track cbf and restoration 1070 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 1071 uint32_t mLoopStart; // last setLoop loopStart 1072 uint32_t mLoopEnd; // last setLoop loopEnd 1073 int32_t mLoopCountNotified; // the last loopCount notified by callback. 1074 // mLoopCountNotified counts down, matching 1075 // the remaining loop count for static track 1076 // playback. 1077 1078 // These are private to processAudioBuffer(), and are not protected by a lock 1079 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 1080 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 1081 uint32_t mObservedSequence; // last observed value of mSequence 1082 1083 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 1084 bool mMarkerReached; 1085 Modulo<uint32_t> mNewPosition; // in frames 1086 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 1087 1088 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 1089 // which is count of frames consumed by server, 1090 // reset by new IAudioTrack, 1091 // whether it is reset by stop() is TBD 1092 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 1093 // monotonically after new IAudioTrack, 1094 // and could be easily widened to uint64_t 1095 Modulo<uint32_t> mReleased; // count of frames released to server 1096 // but not necessarily consumed by server, 1097 // reset by stop() but continues monotonically 1098 // after new IAudioTrack to restore mPosition, 1099 // and could be easily widened to uint64_t 1100 int64_t mStartFromZeroUs; // the start time after flush or stop, 1101 // when position should be 0. 1102 // only used for offloaded and direct tracks. 1103 int64_t mStartNs; // the time when start() is called. 1104 ExtendedTimestamp mStartEts; // Extended timestamp at start for normal 1105 // AudioTracks. 1106 AudioTimestamp mStartTs; // Timestamp at start for offloaded or direct 1107 // AudioTracks. 1108 1109 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 1110 bool mTimestampStartupGlitchReported; // reduce log spam 1111 bool mRetrogradeMotionReported; // reduce log spam 1112 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 1113 ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp 1114 1115 uint32_t mUnderrunCountOffset; // updated when restoring tracks 1116 1117 int64_t mFramesWritten; // total frames written. reset to zero after 1118 // the start() following stop(). It is not 1119 // changed after restoring the track or 1120 // after flush. 1121 int64_t mFramesWrittenServerOffset; // An offset to server frames due to 1122 // restoring AudioTrack, or stop/start. 1123 // This offset is also used for static tracks. 1124 int64_t mFramesWrittenAtRestore; // Frames written at restore point (or frames 1125 // delivered for static tracks). 1126 // -1 indicates no previous restore point. 1127 1128 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 1129 // be denied by client or server, such as 1130 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 1131 // held to read or write those bits reliably. 1132 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 1133 1134 bool mDoNotReconnect; 1135 1136 audio_session_t mSessionId; 1137 int mAuxEffectId; 1138 1139 mutable Mutex mLock; 1140 1141 int mPreviousPriority; // before start() 1142 SchedPolicy mPreviousSchedulingGroup; 1143 bool mAwaitBoost; // thread should wait for priority boost before running 1144 1145 // The proxy should only be referenced while a lock is held because the proxy isn't 1146 // multi-thread safe, especially the SingleStateQueue part of the proxy. 1147 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 1148 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 1149 // them around in case they are replaced during the obtainBuffer(). 1150 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 1151 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 1152 1153 bool mInUnderrun; // whether track is currently in underrun state 1154 uint32_t mPausedPosition; 1155 1156 // For Device Selection API 1157 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 1158 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 1159 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 1160 // May not match the app selection depending on other 1161 // activity and connected devices. 1162 1163 sp<VolumeHandler> mVolumeHandler; 1164 1165 private: 1166 class DeathNotifier : public IBinder::DeathRecipient { 1167 public: 1168 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 1169 protected: 1170 virtual void binderDied(const wp<IBinder>& who); 1171 private: 1172 const wp<AudioTrack> mAudioTrack; 1173 }; 1174 1175 sp<DeathNotifier> mDeathNotifier; 1176 uint32_t mSequence; // incremented for each new IAudioTrack attempt 1177 uid_t mClientUid; 1178 pid_t mClientPid; 1179 1180 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1181 audio_port_handle_t mPortId; // unique ID allocated by audio policy 1182 }; 1183 1184 }; // namespace android 1185 1186 #endif // ANDROID_AUDIOTRACK_H 1187
