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/MediaAnalyticsItem.h> 26 #include <media/Modulo.h> 27 #include <utils/threads.h> 28 29 namespace android { 30 31 // ---------------------------------------------------------------------------- 32 33 struct audio_track_cblk_t; 34 class AudioTrackClientProxy; 35 class StaticAudioTrackClientProxy; 36 37 // ---------------------------------------------------------------------------- 38 39 class AudioTrack : public AudioSystem::AudioDeviceCallback 40 { 41 public: 42 43 /* Events used by AudioTrack callback function (callback_t). 44 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 45 */ 46 enum event_type { 47 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 48 // This event only occurs for TRANSFER_CALLBACK. 49 // If this event is delivered but the callback handler 50 // does not want to write more data, the handler must 51 // ignore the event by setting frameCount to zero. 52 // This might occur, for example, if the application is 53 // waiting for source data or is at the end of stream. 54 // 55 // For data filling, it is preferred that the callback 56 // does not block and instead returns a short count on 57 // the amount of data actually delivered 58 // (or 0, if no data is currently available). 59 EVENT_UNDERRUN = 1, // Buffer underrun occurred. This will not occur for 60 // static tracks. 61 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 62 // loop start if loop count was not 0 for a static track. 63 EVENT_MARKER = 3, // Playback head is at the specified marker position 64 // (See setMarkerPosition()). 65 EVENT_NEW_POS = 4, // Playback head is at a new position 66 // (See setPositionUpdatePeriod()). 67 EVENT_BUFFER_END = 5, // Playback has completed for a static track. 68 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 69 // voluntary invalidation by mediaserver, or mediaserver crash. 70 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 71 // back (after stop is called) for an offloaded track. 72 #if 0 // FIXME not yet implemented 73 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 74 // in the mapping from frame position to presentation time. 75 // See AudioTimestamp for the information included with event. 76 #endif 77 EVENT_CAN_WRITE_MORE_DATA = 9,// Notification that more data can be given by write() 78 // This event only occurs for TRANSFER_SYNC_NOTIF_CALLBACK. 79 }; 80 81 /* Client should declare a Buffer and pass the address to obtainBuffer() 82 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 83 */ 84 85 class Buffer 86 { 87 public: 88 // FIXME use m prefix 89 size_t frameCount; // number of sample frames corresponding to size; 90 // on input to obtainBuffer() it is the number of frames desired, 91 // on output from obtainBuffer() it is the number of available 92 // [empty slots for] frames to be filled 93 // on input to releaseBuffer() it is currently ignored 94 95 size_t size; // input/output in bytes == frameCount * frameSize 96 // on input to obtainBuffer() it is ignored 97 // on output from obtainBuffer() it is the number of available 98 // [empty slots for] bytes to be filled, 99 // which is frameCount * frameSize 100 // on input to releaseBuffer() it is the number of bytes to 101 // release 102 // FIXME This is redundant with respect to frameCount. Consider 103 // removing size and making frameCount the primary field. 104 105 union { 106 void* raw; 107 int16_t* i16; // signed 16-bit 108 int8_t* i8; // unsigned 8-bit, offset by 0x80 109 }; // input to obtainBuffer(): unused, output: pointer to buffer 110 }; 111 112 /* As a convenience, if a callback is supplied, a handler thread 113 * is automatically created with the appropriate priority. This thread 114 * invokes the callback when a new buffer becomes available or various conditions occur. 115 * Parameters: 116 * 117 * event: type of event notified (see enum AudioTrack::event_type). 118 * user: Pointer to context for use by the callback receiver. 119 * info: Pointer to optional parameter according to event type: 120 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 121 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 122 * written. 123 * - EVENT_UNDERRUN: unused. 124 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 125 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 126 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 127 * - EVENT_BUFFER_END: unused. 128 * - EVENT_NEW_IAUDIOTRACK: unused. 129 * - EVENT_STREAM_END: unused. 130 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 131 */ 132 133 typedef void (*callback_t)(int event, void* user, void *info); 134 135 /* Returns the minimum frame count required for the successful creation of 136 * an AudioTrack object. 137 * Returned status (from utils/Errors.h) can be: 138 * - NO_ERROR: successful operation 139 * - NO_INIT: audio server or audio hardware not initialized 140 * - BAD_VALUE: unsupported configuration 141 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 142 * and is undefined otherwise. 143 * FIXME This API assumes a route, and so should be deprecated. 144 */ 145 146 static status_t getMinFrameCount(size_t* frameCount, 147 audio_stream_type_t streamType, 148 uint32_t sampleRate); 149 150 /* Check if direct playback is possible for the given audio configuration and attributes. 151 * Return true if output is possible for the given parameters. Otherwise returns false. 152 */ 153 static bool isDirectOutputSupported(const audio_config_base_t& config, 154 const audio_attributes_t& attributes); 155 156 /* How data is transferred to AudioTrack 157 */ 158 enum transfer_type { 159 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 160 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 161 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 162 TRANSFER_SYNC, // synchronous write() 163 TRANSFER_SHARED, // shared memory 164 TRANSFER_SYNC_NOTIF_CALLBACK, // synchronous write(), notif EVENT_CAN_WRITE_MORE_DATA 165 }; 166 167 /* Constructs an uninitialized AudioTrack. No connection with 168 * AudioFlinger takes place. Use set() after this. 169 */ 170 AudioTrack(); 171 172 /* Creates an AudioTrack object and registers it with AudioFlinger. 173 * Once created, the track needs to be started before it can be used. 174 * Unspecified values are set to appropriate default values. 175 * 176 * Parameters: 177 * 178 * streamType: Select the type of audio stream this track is attached to 179 * (e.g. AUDIO_STREAM_MUSIC). 180 * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate. 181 * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set. 182 * 0 will not work with current policy implementation for direct output 183 * selection where an exact match is needed for sampling rate. 184 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 185 * For direct and offloaded tracks, the possible format(s) depends on the 186 * output sink. 187 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 188 * frameCount: Minimum size of track PCM buffer in frames. This defines the 189 * application's contribution to the 190 * latency of the track. The actual size selected by the AudioTrack could be 191 * larger if the requested size is not compatible with current audio HAL 192 * configuration. Zero means to use a default value. 193 * flags: See comments on audio_output_flags_t in <system/audio.h>. 194 * cbf: Callback function. If not null, this function is called periodically 195 * to provide new data in TRANSFER_CALLBACK mode 196 * and inform of marker, position updates, etc. 197 * user: Context for use by the callback receiver. 198 * notificationFrames: The callback function is called each time notificationFrames PCM 199 * frames have been consumed from track input buffer by server. 200 * Zero means to use a default value, which is typically: 201 * - fast tracks: HAL buffer size, even if track frameCount is larger 202 * - normal tracks: 1/2 of track frameCount 203 * A positive value means that many frames at initial source sample rate. 204 * A negative value for this parameter specifies the negative of the 205 * requested number of notifications (sub-buffers) in the entire buffer. 206 * For fast tracks, the FastMixer will process one sub-buffer at a time. 207 * The size of each sub-buffer is determined by the HAL. 208 * To get "double buffering", for example, one should pass -2. 209 * The minimum number of sub-buffers is 1 (expressed as -1), 210 * and the maximum number of sub-buffers is 8 (expressed as -8). 211 * Negative is only permitted for fast tracks, and if frameCount is zero. 212 * TODO It is ugly to overload a parameter in this way depending on 213 * whether it is positive, negative, or zero. Consider splitting apart. 214 * sessionId: Specific session ID, or zero to use default. 215 * transferType: How data is transferred to AudioTrack. 216 * offloadInfo: If not NULL, provides offload parameters for 217 * AudioSystem::getOutputForAttr(). 218 * uid: User ID of the app which initially requested this AudioTrack 219 * for power management tracking, or -1 for current user ID. 220 * pid: Process ID of the app which initially requested this AudioTrack 221 * for power management tracking, or -1 for current process ID. 222 * pAttributes: If not NULL, supersedes streamType for use case selection. 223 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 224 binder to AudioFlinger. 225 It will return an error instead. The application will recreate 226 the track based on offloading or different channel configuration, etc. 227 * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow 228 * maxRequiredSpeed playback. Values less than 1.0f and greater than 229 * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks 230 * and direct or offloaded tracks, this parameter is ignored. 231 * selectedDeviceId: Selected device id of the app which initially requested the AudioTrack 232 * to open with a specific device. 233 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 234 */ 235 236 AudioTrack( audio_stream_type_t streamType, 237 uint32_t sampleRate, 238 audio_format_t format, 239 audio_channel_mask_t channelMask, 240 size_t frameCount = 0, 241 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 242 callback_t cbf = NULL, 243 void* user = NULL, 244 int32_t notificationFrames = 0, 245 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 246 transfer_type transferType = TRANSFER_DEFAULT, 247 const audio_offload_info_t *offloadInfo = NULL, 248 uid_t uid = AUDIO_UID_INVALID, 249 pid_t pid = -1, 250 const audio_attributes_t* pAttributes = NULL, 251 bool doNotReconnect = false, 252 float maxRequiredSpeed = 1.0f, 253 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 254 255 /* Creates an audio track and registers it with AudioFlinger. 256 * With this constructor, the track is configured for static buffer mode. 257 * Data to be rendered is passed in a shared memory buffer 258 * identified by the argument sharedBuffer, which should be non-0. 259 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 260 * but without the ability to specify a non-zero value for the frameCount parameter. 261 * The memory should be initialized to the desired data before calling start(). 262 * The write() method is not supported in this case. 263 * It is recommended to pass a callback function to be notified of playback end by an 264 * EVENT_UNDERRUN event. 265 */ 266 267 AudioTrack( audio_stream_type_t streamType, 268 uint32_t sampleRate, 269 audio_format_t format, 270 audio_channel_mask_t channelMask, 271 const sp<IMemory>& sharedBuffer, 272 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 273 callback_t cbf = NULL, 274 void* user = NULL, 275 int32_t notificationFrames = 0, 276 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 277 transfer_type transferType = TRANSFER_DEFAULT, 278 const audio_offload_info_t *offloadInfo = NULL, 279 uid_t uid = AUDIO_UID_INVALID, 280 pid_t pid = -1, 281 const audio_attributes_t* pAttributes = NULL, 282 bool doNotReconnect = false, 283 float maxRequiredSpeed = 1.0f); 284 285 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 286 * Also destroys all resources associated with the AudioTrack. 287 */ 288 protected: 289 virtual ~AudioTrack(); 290 public: 291 292 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 293 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 294 * set() is not multi-thread safe. 295 * Returned status (from utils/Errors.h) can be: 296 * - NO_ERROR: successful initialization 297 * - INVALID_OPERATION: AudioTrack is already initialized 298 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 299 * - NO_INIT: audio server or audio hardware not initialized 300 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 301 * If sharedBuffer is non-0, the frameCount parameter is ignored and 302 * replaced by the shared buffer's total allocated size in frame units. 303 * 304 * Parameters not listed in the AudioTrack constructors above: 305 * 306 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 307 * Only set to true when AudioTrack object is used for a java android.media.AudioTrack 308 * in its JNI code. 309 * 310 * Internal state post condition: 311 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 312 */ 313 status_t set(audio_stream_type_t streamType, 314 uint32_t sampleRate, 315 audio_format_t format, 316 audio_channel_mask_t channelMask, 317 size_t frameCount = 0, 318 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 319 callback_t cbf = NULL, 320 void* user = NULL, 321 int32_t notificationFrames = 0, 322 const sp<IMemory>& sharedBuffer = 0, 323 bool threadCanCallJava = false, 324 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 325 transfer_type transferType = TRANSFER_DEFAULT, 326 const audio_offload_info_t *offloadInfo = NULL, 327 uid_t uid = AUDIO_UID_INVALID, 328 pid_t pid = -1, 329 const audio_attributes_t* pAttributes = NULL, 330 bool doNotReconnect = false, 331 float maxRequiredSpeed = 1.0f, 332 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 333 334 /* Result of constructing the AudioTrack. This must be checked for successful initialization 335 * before using any AudioTrack API (except for set()), because using 336 * an uninitialized AudioTrack produces undefined results. 337 * See set() method above for possible return codes. 338 */ 339 status_t initCheck() const { return mStatus; } 340 341 /* Returns this track's estimated latency in milliseconds. 342 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 343 * and audio hardware driver. 344 */ 345 uint32_t latency(); 346 347 /* Returns the number of application-level buffer underruns 348 * since the AudioTrack was created. 349 */ 350 uint32_t getUnderrunCount() const; 351 352 /* getters, see constructors and set() */ 353 354 audio_stream_type_t streamType() const; 355 audio_format_t format() const { return mFormat; } 356 357 /* Return frame size in bytes, which for linear PCM is 358 * channelCount * (bit depth per channel / 8). 359 * channelCount is determined from channelMask, and bit depth comes from format. 360 * For non-linear formats, the frame size is typically 1 byte. 361 */ 362 size_t frameSize() const { return mFrameSize; } 363 364 uint32_t channelCount() const { return mChannelCount; } 365 size_t frameCount() const { return mFrameCount; } 366 367 /* 368 * Return the period of the notification callback in frames. 369 * This value is set when the AudioTrack is constructed. 370 * It can be modified if the AudioTrack is rerouted. 371 */ 372 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 373 374 /* Return effective size of audio buffer that an application writes to 375 * or a negative error if the track is uninitialized. 376 */ 377 ssize_t getBufferSizeInFrames(); 378 379 /* Returns the buffer duration in microseconds at current playback rate. 380 */ 381 status_t getBufferDurationInUs(int64_t *duration); 382 383 /* Set the effective size of audio buffer that an application writes to. 384 * This is used to determine the amount of available room in the buffer, 385 * which determines when a write will block. 386 * This allows an application to raise and lower the audio latency. 387 * The requested size may be adjusted so that it is 388 * greater or equal to the absolute minimum and 389 * less than or equal to the getBufferCapacityInFrames(). 390 * It may also be adjusted slightly for internal reasons. 391 * 392 * Return the final size or a negative error if the track is unitialized 393 * or does not support variable sizes. 394 */ 395 ssize_t setBufferSizeInFrames(size_t size); 396 397 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 398 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 399 400 /* 401 * return metrics information for the current track. 402 */ 403 status_t getMetrics(MediaAnalyticsItem * &item); 404 405 /* After it's created the track is not active. Call start() to 406 * make it active. If set, the callback will start being called. 407 * If the track was previously paused, volume is ramped up over the first mix buffer. 408 */ 409 status_t start(); 410 411 /* Stop a track. 412 * In static buffer mode, the track is stopped immediately. 413 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 414 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 415 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 416 * is first drained, mixed, and output, and only then is the track marked as stopped. 417 */ 418 void stop(); 419 bool stopped() const; 420 421 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 422 * This has the effect of draining the buffers without mixing or output. 423 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 424 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 425 */ 426 void flush(); 427 428 /* Pause a track. After pause, the callback will cease being called and 429 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 430 * and will fill up buffers until the pool is exhausted. 431 * Volume is ramped down over the next mix buffer following the pause request, 432 * and then the track is marked as paused. It can be resumed with ramp up by start(). 433 */ 434 void pause(); 435 436 /* Set volume for this track, mostly used for games' sound effects 437 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 438 * This is the older API. New applications should use setVolume(float) when possible. 439 */ 440 status_t setVolume(float left, float right); 441 442 /* Set volume for all channels. This is the preferred API for new applications, 443 * especially for multi-channel content. 444 */ 445 status_t setVolume(float volume); 446 447 /* Set the send level for this track. An auxiliary effect should be attached 448 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 449 */ 450 status_t setAuxEffectSendLevel(float level); 451 void getAuxEffectSendLevel(float* level) const; 452 453 /* Set source sample rate for this track in Hz, mostly used for games' sound effects. 454 * Zero is not permitted. 455 */ 456 status_t setSampleRate(uint32_t sampleRate); 457 458 /* Return current source sample rate in Hz. 459 * If specified as zero in constructor or set(), this will be the sink sample rate. 460 */ 461 uint32_t getSampleRate() const; 462 463 /* Return the original source sample rate in Hz. This corresponds to the sample rate 464 * if playback rate had normal speed and pitch. 465 */ 466 uint32_t getOriginalSampleRate() const; 467 468 /* Set source playback rate for timestretch 469 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 470 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 471 * 472 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 473 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 474 * 475 * Speed increases the playback rate of media, but does not alter pitch. 476 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 477 */ 478 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 479 480 /* Return current playback rate */ 481 const AudioPlaybackRate& getPlaybackRate() const; 482 483 /* Enables looping and sets the start and end points of looping. 484 * Only supported for static buffer mode. 485 * 486 * Parameters: 487 * 488 * loopStart: loop start in frames relative to start of buffer. 489 * loopEnd: loop end in frames relative to start of buffer. 490 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 491 * pending or active loop. loopCount == -1 means infinite looping. 492 * 493 * For proper operation the following condition must be respected: 494 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 495 * 496 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 497 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 498 * 499 */ 500 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 501 502 /* Sets marker position. When playback reaches the number of frames specified, a callback with 503 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 504 * notification callback. To set a marker at a position which would compute as 0, 505 * a workaround is to set the marker at a nearby position such as ~0 or 1. 506 * If the AudioTrack has been opened with no callback function associated, the operation will 507 * fail. 508 * 509 * Parameters: 510 * 511 * marker: marker position expressed in wrapping (overflow) frame units, 512 * like the return value of getPosition(). 513 * 514 * Returned status (from utils/Errors.h) can be: 515 * - NO_ERROR: successful operation 516 * - INVALID_OPERATION: the AudioTrack has no callback installed. 517 */ 518 status_t setMarkerPosition(uint32_t marker); 519 status_t getMarkerPosition(uint32_t *marker) const; 520 521 /* Sets position update period. Every time the number of frames specified has been played, 522 * a callback with event type EVENT_NEW_POS is called. 523 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 524 * callback. 525 * If the AudioTrack has been opened with no callback function associated, the operation will 526 * fail. 527 * Extremely small values may be rounded up to a value the implementation can support. 528 * 529 * Parameters: 530 * 531 * updatePeriod: position update notification period expressed in frames. 532 * 533 * Returned status (from utils/Errors.h) can be: 534 * - NO_ERROR: successful operation 535 * - INVALID_OPERATION: the AudioTrack has no callback installed. 536 */ 537 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 538 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 539 540 /* Sets playback head position. 541 * Only supported for static buffer mode. 542 * 543 * Parameters: 544 * 545 * position: New playback head position in frames relative to start of buffer. 546 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 547 * but will result in an immediate underrun if started. 548 * 549 * Returned status (from utils/Errors.h) can be: 550 * - NO_ERROR: successful operation 551 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 552 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 553 * buffer 554 */ 555 status_t setPosition(uint32_t position); 556 557 /* Return the total number of frames played since playback start. 558 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 559 * It is reset to zero by flush(), reload(), and stop(). 560 * 561 * Parameters: 562 * 563 * position: Address where to return play head position. 564 * 565 * Returned status (from utils/Errors.h) can be: 566 * - NO_ERROR: successful operation 567 * - BAD_VALUE: position is NULL 568 */ 569 status_t getPosition(uint32_t *position); 570 571 /* For static buffer mode only, this returns the current playback position in frames 572 * relative to start of buffer. It is analogous to the position units used by 573 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 574 */ 575 status_t getBufferPosition(uint32_t *position); 576 577 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 578 * rewriting the buffer before restarting playback after a stop. 579 * This method must be called with the AudioTrack in paused or stopped state. 580 * Not allowed in streaming mode. 581 * 582 * Returned status (from utils/Errors.h) can be: 583 * - NO_ERROR: successful operation 584 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 585 */ 586 status_t reload(); 587 588 /** 589 * @param transferType 590 * @return text string that matches the enum name 591 */ 592 static const char * convertTransferToText(transfer_type transferType); 593 594 /* Returns a handle on the audio output used by this AudioTrack. 595 * 596 * Parameters: 597 * none. 598 * 599 * Returned value: 600 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 601 * track needed to be re-created but that failed 602 */ 603 private: 604 audio_io_handle_t getOutput() const; 605 public: 606 607 /* Selects the audio device to use for output of this AudioTrack. A value of 608 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 609 * 610 * Parameters: 611 * The device ID of the selected device (as returned by the AudioDevicesManager API). 612 * 613 * Returned value: 614 * - NO_ERROR: successful operation 615 * TODO: what else can happen here? 616 */ 617 status_t setOutputDevice(audio_port_handle_t deviceId); 618 619 /* Returns the ID of the audio device selected for this AudioTrack. 620 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 621 * 622 * Parameters: 623 * none. 624 */ 625 audio_port_handle_t getOutputDevice(); 626 627 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 628 * attached. 629 * When the AudioTrack is inactive, the device ID returned can be either: 630 * - AUDIO_PORT_HANDLE_NONE if the AudioTrack is not attached to any output. 631 * - The device ID used before paused or stopped. 632 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioTrack 633 * has not been started yet. 634 * 635 * Parameters: 636 * none. 637 */ 638 audio_port_handle_t getRoutedDeviceId(); 639 640 /* Returns the unique session ID associated with this track. 641 * 642 * Parameters: 643 * none. 644 * 645 * Returned value: 646 * AudioTrack session ID. 647 */ 648 audio_session_t getSessionId() const { return mSessionId; } 649 650 /* Attach track auxiliary output to specified effect. Use effectId = 0 651 * to detach track from effect. 652 * 653 * Parameters: 654 * 655 * effectId: effectId obtained from AudioEffect::id(). 656 * 657 * Returned status (from utils/Errors.h) can be: 658 * - NO_ERROR: successful operation 659 * - INVALID_OPERATION: the effect is not an auxiliary effect. 660 * - BAD_VALUE: The specified effect ID is invalid 661 */ 662 status_t attachAuxEffect(int effectId); 663 664 /* Public API for TRANSFER_OBTAIN mode. 665 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 666 * After filling these slots with data, the caller should release them with releaseBuffer(). 667 * If the track buffer is not full, obtainBuffer() returns as many contiguous 668 * [empty slots for] frames as are available immediately. 669 * 670 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 671 * additional non-contiguous frames that are predicted to be available immediately, 672 * if the client were to release the first frames and then call obtainBuffer() again. 673 * This value is only a prediction, and needs to be confirmed. 674 * It will be set to zero for an error return. 675 * 676 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 677 * regardless of the value of waitCount. 678 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 679 * maximum timeout based on waitCount; see chart below. 680 * Buffers will be returned until the pool 681 * is exhausted, at which point obtainBuffer() will either block 682 * or return WOULD_BLOCK depending on the value of the "waitCount" 683 * parameter. 684 * 685 * Interpretation of waitCount: 686 * +n limits wait time to n * WAIT_PERIOD_MS, 687 * -1 causes an (almost) infinite wait time, 688 * 0 non-blocking. 689 * 690 * Buffer fields 691 * On entry: 692 * frameCount number of [empty slots for] frames requested 693 * size ignored 694 * raw ignored 695 * After error return: 696 * frameCount 0 697 * size 0 698 * raw undefined 699 * After successful return: 700 * frameCount actual number of [empty slots for] frames available, <= number requested 701 * size actual number of bytes available 702 * raw pointer to the buffer 703 */ 704 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 705 size_t *nonContig = NULL); 706 707 private: 708 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 709 * additional non-contiguous frames that are predicted to be available immediately, 710 * if the client were to release the first frames and then call obtainBuffer() again. 711 * This value is only a prediction, and needs to be confirmed. 712 * It will be set to zero for an error return. 713 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 714 * in case the requested amount of frames is in two or more non-contiguous regions. 715 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 716 */ 717 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 718 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 719 public: 720 721 /* Public API for TRANSFER_OBTAIN mode. 722 * Release a filled buffer of frames for AudioFlinger to process. 723 * 724 * Buffer fields: 725 * frameCount currently ignored but recommend to set to actual number of frames filled 726 * size actual number of bytes filled, must be multiple of frameSize 727 * raw ignored 728 */ 729 void releaseBuffer(const Buffer* audioBuffer); 730 731 /* As a convenience we provide a write() interface to the audio buffer. 732 * Input parameter 'size' is in byte units. 733 * This is implemented on top of obtainBuffer/releaseBuffer. For best 734 * performance use callbacks. Returns actual number of bytes written >= 0, 735 * or one of the following negative status codes: 736 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 737 * BAD_VALUE size is invalid 738 * WOULD_BLOCK when obtainBuffer() returns same, or 739 * AudioTrack was stopped during the write 740 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 741 * the track cannot be automatically restored. 742 * The application needs to recreate the AudioTrack 743 * because the audio device changed or AudioFlinger died. 744 * This typically occurs for direct or offload tracks 745 * or if mDoNotReconnect is true. 746 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 747 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 748 * false for the method to return immediately without waiting to try multiple times to write 749 * the full content of the buffer. 750 */ 751 ssize_t write(const void* buffer, size_t size, bool blocking = true); 752 753 /* 754 * Dumps the state of an audio track. 755 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 756 */ 757 status_t dump(int fd, const Vector<String16>& args) const; 758 759 /* 760 * Return the total number of frames which AudioFlinger desired but were unavailable, 761 * and thus which resulted in an underrun. Reset to zero by stop(). 762 */ 763 uint32_t getUnderrunFrames() const; 764 765 /* Get the flags */ 766 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 767 768 /* Set parameters - only possible when using direct output */ 769 status_t setParameters(const String8& keyValuePairs); 770 771 /* Sets the volume shaper object */ 772 media::VolumeShaper::Status applyVolumeShaper( 773 const sp<media::VolumeShaper::Configuration>& configuration, 774 const sp<media::VolumeShaper::Operation>& operation); 775 776 /* Gets the volume shaper state */ 777 sp<media::VolumeShaper::State> getVolumeShaperState(int id); 778 779 /* Selects the presentation (if available) */ 780 status_t selectPresentation(int presentationId, int programId); 781 782 /* Get parameters */ 783 String8 getParameters(const String8& keys); 784 785 /* Poll for a timestamp on demand. 786 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 787 * or if you need to get the most recent timestamp outside of the event callback handler. 788 * Caution: calling this method too often may be inefficient; 789 * if you need a high resolution mapping between frame position and presentation time, 790 * consider implementing that at application level, based on the low resolution timestamps. 791 * Returns NO_ERROR if timestamp is valid. 792 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 793 * start/ACTIVE, when the number of frames consumed is less than the 794 * overall hardware latency to physical output. In WOULD_BLOCK cases, 795 * one might poll again, or use getPosition(), or use 0 position and 796 * current time for the timestamp. 797 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 798 * the track cannot be automatically restored. 799 * The application needs to recreate the AudioTrack 800 * because the audio device changed or AudioFlinger died. 801 * This typically occurs for direct or offload tracks 802 * or if mDoNotReconnect is true. 803 * INVALID_OPERATION wrong state, or some other error. 804 * 805 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 806 */ 807 status_t getTimestamp(AudioTimestamp& timestamp); 808 private: 809 status_t getTimestamp_l(AudioTimestamp& timestamp); 810 public: 811 812 /* Return the extended timestamp, with additional timebase info and improved drain behavior. 813 * 814 * This is similar to the AudioTrack.java API: 815 * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) 816 * 817 * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method 818 * 819 * 1. stop() by itself does not reset the frame position. 820 * A following start() resets the frame position to 0. 821 * 2. flush() by itself does not reset the frame position. 822 * The frame position advances by the number of frames flushed, 823 * when the first frame after flush reaches the audio sink. 824 * 3. BOOTTIME clock offsets are provided to help synchronize with 825 * non-audio streams, e.g. sensor data. 826 * 4. Position is returned with 64 bits of resolution. 827 * 828 * Parameters: 829 * timestamp: A pointer to the caller allocated ExtendedTimestamp. 830 * 831 * Returns NO_ERROR on success; timestamp is filled with valid data. 832 * BAD_VALUE if timestamp is NULL. 833 * WOULD_BLOCK if called immediately after start() when the number 834 * of frames consumed is less than the 835 * overall hardware latency to physical output. In WOULD_BLOCK cases, 836 * one might poll again, or use getPosition(), or use 0 position and 837 * current time for the timestamp. 838 * If WOULD_BLOCK is returned, the timestamp is still 839 * modified with the LOCATION_CLIENT portion filled. 840 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 841 * the track cannot be automatically restored. 842 * The application needs to recreate the AudioTrack 843 * because the audio device changed or AudioFlinger died. 844 * This typically occurs for direct or offloaded tracks 845 * or if mDoNotReconnect is true. 846 * INVALID_OPERATION if called on a offloaded or direct track. 847 * Use getTimestamp(AudioTimestamp& timestamp) instead. 848 */ 849 status_t getTimestamp(ExtendedTimestamp *timestamp); 850 private: 851 status_t getTimestamp_l(ExtendedTimestamp *timestamp); 852 public: 853 854 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 855 * AudioTrack is routed is updated. 856 * Replaces any previously installed callback. 857 * Parameters: 858 * callback: The callback interface 859 * Returns NO_ERROR if successful. 860 * INVALID_OPERATION if the same callback is already installed. 861 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 862 * BAD_VALUE if the callback is NULL 863 */ 864 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 865 866 /* remove an AudioDeviceCallback. 867 * Parameters: 868 * callback: The callback interface 869 * Returns NO_ERROR if successful. 870 * INVALID_OPERATION if the callback is not installed 871 * BAD_VALUE if the callback is NULL 872 */ 873 status_t removeAudioDeviceCallback( 874 const sp<AudioSystem::AudioDeviceCallback>& callback); 875 876 // AudioSystem::AudioDeviceCallback> virtuals 877 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 878 audio_port_handle_t deviceId); 879 880 881 882 /* Obtain the pending duration in milliseconds for playback of pure PCM 883 * (mixable without embedded timing) data remaining in AudioTrack. 884 * 885 * This is used to estimate the drain time for the client-server buffer 886 * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. 887 * One may optionally request to find the duration to play through the HAL 888 * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, 889 * INVALID_OPERATION may be returned if the kernel location is unavailable. 890 * 891 * Returns NO_ERROR if successful. 892 * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained 893 * or the AudioTrack does not contain pure PCM data. 894 * BAD_VALUE if msec is nullptr or location is invalid. 895 */ 896 status_t pendingDuration(int32_t *msec, 897 ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); 898 899 /* hasStarted() is used to determine if audio is now audible at the device after 900 * a start() command. The underlying implementation checks a nonzero timestamp position 901 * or increment for the audible assumption. 902 * 903 * hasStarted() returns true if the track has been started() and audio is audible 904 * and no subsequent pause() or flush() has been called. Immediately after pause() or 905 * flush() hasStarted() will return false. 906 * 907 * If stop() has been called, hasStarted() will return true if audio is still being 908 * delivered or has finished delivery (even if no audio was written) for both offloaded 909 * and normal tracks. This property removes a race condition in checking hasStarted() 910 * for very short clips, where stop() must be called to finish drain. 911 * 912 * In all cases, hasStarted() may turn false briefly after a subsequent start() is called 913 * until audio becomes audible again. 914 */ 915 bool hasStarted(); // not const 916 917 bool isPlaying() { 918 AutoMutex lock(mLock); 919 return mState == STATE_ACTIVE || mState == STATE_STOPPING; 920 } 921 922 /* Get the unique port ID assigned to this AudioTrack instance by audio policy manager. 923 * The ID is unique across all audioserver clients and can change during the life cycle 924 * of a given AudioTrack instance if the connection to audioserver is restored. 925 */ 926 audio_port_handle_t getPortId() const { return mPortId; }; 927 928 protected: 929 /* copying audio tracks is not allowed */ 930 AudioTrack(const AudioTrack& other); 931 AudioTrack& operator = (const AudioTrack& other); 932 933 /* a small internal class to handle the callback */ 934 class AudioTrackThread : public Thread 935 { 936 public: 937 AudioTrackThread(AudioTrack& receiver); 938 939 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 940 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 941 virtual void requestExit(); 942 943 void pause(); // suspend thread from execution at next loop boundary 944 void resume(); // allow thread to execute, if not requested to exit 945 void wake(); // wake to handle changed notification conditions. 946 947 private: 948 void pauseInternal(nsecs_t ns = 0LL); 949 // like pause(), but only used internally within thread 950 951 friend class AudioTrack; 952 virtual bool threadLoop(); 953 AudioTrack& mReceiver; 954 virtual ~AudioTrackThread(); 955 Mutex mMyLock; // Thread::mLock is private 956 Condition mMyCond; // Thread::mThreadExitedCondition is private 957 bool mPaused; // whether thread is requested to pause at next loop entry 958 bool mPausedInt; // whether thread internally requests pause 959 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 960 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 961 // to processAudioBuffer() as state may have changed 962 // since pause time calculated. 963 }; 964 965 // body of AudioTrackThread::threadLoop() 966 // returns the maximum amount of time before we would like to run again, where: 967 // 0 immediately 968 // > 0 no later than this many nanoseconds from now 969 // NS_WHENEVER still active but no particular deadline 970 // NS_INACTIVE inactive so don't run again until re-started 971 // NS_NEVER never again 972 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 973 nsecs_t processAudioBuffer(); 974 975 // caller must hold lock on mLock for all _l methods 976 977 void updateLatency_l(); // updates mAfLatency and mLatency from AudioSystem cache 978 979 status_t createTrack_l(); 980 981 // can only be called when mState != STATE_ACTIVE 982 void flush_l(); 983 984 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 985 986 // FIXME enum is faster than strcmp() for parameter 'from' 987 status_t restoreTrack_l(const char *from); 988 989 uint32_t getUnderrunCount_l() const; 990 991 bool isOffloaded() const; 992 bool isDirect() const; 993 bool isOffloadedOrDirect() const; 994 995 bool isOffloaded_l() const 996 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 997 998 bool isOffloadedOrDirect_l() const 999 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 1000 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 1001 1002 bool isDirect_l() const 1003 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 1004 1005 // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) 1006 bool isPurePcmData_l() const 1007 { return audio_is_linear_pcm(mFormat) 1008 && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } 1009 1010 // increment mPosition by the delta of mServer, and return new value of mPosition 1011 Modulo<uint32_t> updateAndGetPosition_l(); 1012 1013 // check sample rate and speed is compatible with AudioTrack 1014 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed); 1015 1016 void restartIfDisabled(); 1017 1018 void updateRoutedDeviceId_l(); 1019 1020 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 1021 sp<IAudioTrack> mAudioTrack; 1022 sp<IMemory> mCblkMemory; 1023 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 1024 audio_io_handle_t mOutput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getOutputForAttr() 1025 1026 sp<AudioTrackThread> mAudioTrackThread; 1027 bool mThreadCanCallJava; 1028 1029 float mVolume[2]; 1030 float mSendLevel; 1031 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 1032 uint32_t mOriginalSampleRate; 1033 AudioPlaybackRate mPlaybackRate; 1034 float mMaxRequiredSpeed; // use PCM buffer size to allow this speed 1035 1036 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 1037 // This allocated buffer size is maintained by the proxy. 1038 size_t mFrameCount; // maximum size of buffer 1039 1040 size_t mReqFrameCount; // frame count to request the first or next time 1041 // a new IAudioTrack is needed, non-decreasing 1042 1043 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 1044 // These values can be used for informational purposes until the track is invalidated, 1045 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 1046 uint32_t mAfLatency; // AudioFlinger latency in ms 1047 size_t mAfFrameCount; // AudioFlinger frame count 1048 uint32_t mAfSampleRate; // AudioFlinger sample rate 1049 1050 // constant after constructor or set() 1051 audio_format_t mFormat; // as requested by client, not forced to 16-bit 1052 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 1053 // this AudioTrack has valid attributes 1054 uint32_t mChannelCount; 1055 audio_channel_mask_t mChannelMask; 1056 sp<IMemory> mSharedBuffer; 1057 transfer_type mTransfer; 1058 audio_offload_info_t mOffloadInfoCopy; 1059 const audio_offload_info_t* mOffloadInfo; 1060 audio_attributes_t mAttributes; 1061 1062 size_t mFrameSize; // frame size in bytes 1063 1064 status_t mStatus; 1065 1066 // can change dynamically when IAudioTrack invalidated 1067 uint32_t mLatency; // in ms 1068 1069 // Indicates the current track state. Protected by mLock. 1070 enum State { 1071 STATE_ACTIVE, 1072 STATE_STOPPED, 1073 STATE_PAUSED, 1074 STATE_PAUSED_STOPPING, 1075 STATE_FLUSHED, 1076 STATE_STOPPING, 1077 } mState; 1078 1079 static constexpr const char *stateToString(State state) 1080 { 1081 switch (state) { 1082 case STATE_ACTIVE: return "STATE_ACTIVE"; 1083 case STATE_STOPPED: return "STATE_STOPPED"; 1084 case STATE_PAUSED: return "STATE_PAUSED"; 1085 case STATE_PAUSED_STOPPING: return "STATE_PAUSED_STOPPING"; 1086 case STATE_FLUSHED: return "STATE_FLUSHED"; 1087 case STATE_STOPPING: return "STATE_STOPPING"; 1088 default: return "UNKNOWN"; 1089 } 1090 } 1091 1092 // for client callback handler 1093 callback_t mCbf; // callback handler for events, or NULL 1094 void* mUserData; 1095 1096 // for notification APIs 1097 1098 // next 2 fields are const after constructor or set() 1099 uint32_t mNotificationFramesReq; // requested number of frames between each 1100 // notification callback, 1101 // at initial source sample rate 1102 uint32_t mNotificationsPerBufferReq; 1103 // requested number of notifications per buffer, 1104 // currently only used for fast tracks with 1105 // default track buffer size 1106 1107 uint32_t mNotificationFramesAct; // actual number of frames between each 1108 // notification callback, 1109 // at initial source sample rate 1110 bool mRefreshRemaining; // processAudioBuffer() should refresh 1111 // mRemainingFrames and mRetryOnPartialBuffer 1112 1113 // used for static track cbf and restoration 1114 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 1115 uint32_t mLoopStart; // last setLoop loopStart 1116 uint32_t mLoopEnd; // last setLoop loopEnd 1117 int32_t mLoopCountNotified; // the last loopCount notified by callback. 1118 // mLoopCountNotified counts down, matching 1119 // the remaining loop count for static track 1120 // playback. 1121 1122 // These are private to processAudioBuffer(), and are not protected by a lock 1123 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 1124 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 1125 uint32_t mObservedSequence; // last observed value of mSequence 1126 1127 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 1128 bool mMarkerReached; 1129 Modulo<uint32_t> mNewPosition; // in frames 1130 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 1131 1132 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 1133 // which is count of frames consumed by server, 1134 // reset by new IAudioTrack, 1135 // whether it is reset by stop() is TBD 1136 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 1137 // monotonically after new IAudioTrack, 1138 // and could be easily widened to uint64_t 1139 Modulo<uint32_t> mReleased; // count of frames released to server 1140 // but not necessarily consumed by server, 1141 // reset by stop() but continues monotonically 1142 // after new IAudioTrack to restore mPosition, 1143 // and could be easily widened to uint64_t 1144 int64_t mStartFromZeroUs; // the start time after flush or stop, 1145 // when position should be 0. 1146 // only used for offloaded and direct tracks. 1147 int64_t mStartNs; // the time when start() is called. 1148 ExtendedTimestamp mStartEts; // Extended timestamp at start for normal 1149 // AudioTracks. 1150 AudioTimestamp mStartTs; // Timestamp at start for offloaded or direct 1151 // AudioTracks. 1152 1153 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 1154 bool mTimestampStartupGlitchReported; // reduce log spam 1155 bool mTimestampRetrogradePositionReported; // reduce log spam 1156 bool mTimestampRetrogradeTimeReported; // reduce log spam 1157 bool mTimestampStallReported; // reduce log spam 1158 bool mTimestampStaleTimeReported; // reduce log spam 1159 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 1160 ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp 1161 1162 uint32_t mUnderrunCountOffset; // updated when restoring tracks 1163 1164 int64_t mFramesWritten; // total frames written. reset to zero after 1165 // the start() following stop(). It is not 1166 // changed after restoring the track or 1167 // after flush. 1168 int64_t mFramesWrittenServerOffset; // An offset to server frames due to 1169 // restoring AudioTrack, or stop/start. 1170 // This offset is also used for static tracks. 1171 int64_t mFramesWrittenAtRestore; // Frames written at restore point (or frames 1172 // delivered for static tracks). 1173 // -1 indicates no previous restore point. 1174 1175 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 1176 // be denied by client or server, such as 1177 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 1178 // held to read or write those bits reliably. 1179 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 1180 1181 bool mDoNotReconnect; 1182 1183 audio_session_t mSessionId; 1184 int mAuxEffectId; 1185 audio_port_handle_t mPortId; // Id from Audio Policy Manager 1186 1187 mutable Mutex mLock; 1188 1189 int mPreviousPriority; // before start() 1190 SchedPolicy mPreviousSchedulingGroup; 1191 bool mAwaitBoost; // thread should wait for priority boost before running 1192 1193 // The proxy should only be referenced while a lock is held because the proxy isn't 1194 // multi-thread safe, especially the SingleStateQueue part of the proxy. 1195 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 1196 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 1197 // them around in case they are replaced during the obtainBuffer(). 1198 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 1199 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 1200 1201 bool mInUnderrun; // whether track is currently in underrun state 1202 uint32_t mPausedPosition; 1203 1204 // For Device Selection API 1205 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 1206 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 1207 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 1208 // May not match the app selection depending on other 1209 // activity and connected devices. 1210 1211 sp<media::VolumeHandler> mVolumeHandler; 1212 1213 private: 1214 class DeathNotifier : public IBinder::DeathRecipient { 1215 public: 1216 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 1217 protected: 1218 virtual void binderDied(const wp<IBinder>& who); 1219 private: 1220 const wp<AudioTrack> mAudioTrack; 1221 }; 1222 1223 sp<DeathNotifier> mDeathNotifier; 1224 uint32_t mSequence; // incremented for each new IAudioTrack attempt 1225 uid_t mClientUid; 1226 pid_t mClientPid; 1227 1228 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1229 1230 private: 1231 class MediaMetrics { 1232 public: 1233 MediaMetrics() : mAnalyticsItem(MediaAnalyticsItem::create("audiotrack")) { 1234 } 1235 ~MediaMetrics() { 1236 // mAnalyticsItem alloc failure will be flagged in the constructor 1237 // don't log empty records 1238 if (mAnalyticsItem->count() > 0) { 1239 mAnalyticsItem->selfrecord(); 1240 } 1241 } 1242 void gather(const AudioTrack *track); 1243 MediaAnalyticsItem *dup() { return mAnalyticsItem->dup(); } 1244 private: 1245 std::unique_ptr<MediaAnalyticsItem> mAnalyticsItem; 1246 }; 1247 MediaMetrics mMediaMetrics; 1248 }; 1249 1250 }; // namespace android 1251 1252 #endif // ANDROID_AUDIOTRACK_H 1253
