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 uint32_t sequence; // IAudioTrack instance sequence number, as of obtainBuffer(). 112 // It is set by obtainBuffer() and confirmed by releaseBuffer(). 113 // Not "user-serviceable". 114 // TODO Consider sp<IMemory> instead, or in addition to this. 115 }; 116 117 /* As a convenience, if a callback is supplied, a handler thread 118 * is automatically created with the appropriate priority. This thread 119 * invokes the callback when a new buffer becomes available or various conditions occur. 120 * Parameters: 121 * 122 * event: type of event notified (see enum AudioTrack::event_type). 123 * user: Pointer to context for use by the callback receiver. 124 * info: Pointer to optional parameter according to event type: 125 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 126 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 127 * written. 128 * - EVENT_UNDERRUN: unused. 129 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 130 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 131 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 132 * - EVENT_BUFFER_END: unused. 133 * - EVENT_NEW_IAUDIOTRACK: unused. 134 * - EVENT_STREAM_END: unused. 135 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 136 */ 137 138 typedef void (*callback_t)(int event, void* user, void *info); 139 140 /* Returns the minimum frame count required for the successful creation of 141 * an AudioTrack object. 142 * Returned status (from utils/Errors.h) can be: 143 * - NO_ERROR: successful operation 144 * - NO_INIT: audio server or audio hardware not initialized 145 * - BAD_VALUE: unsupported configuration 146 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 147 * and is undefined otherwise. 148 * FIXME This API assumes a route, and so should be deprecated. 149 */ 150 151 static status_t getMinFrameCount(size_t* frameCount, 152 audio_stream_type_t streamType, 153 uint32_t sampleRate); 154 155 /* Check if direct playback is possible for the given audio configuration and attributes. 156 * Return true if output is possible for the given parameters. Otherwise returns false. 157 */ 158 static bool isDirectOutputSupported(const audio_config_base_t& config, 159 const audio_attributes_t& attributes); 160 161 /* How data is transferred to AudioTrack 162 */ 163 enum transfer_type { 164 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 165 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 166 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 167 TRANSFER_SYNC, // synchronous write() 168 TRANSFER_SHARED, // shared memory 169 TRANSFER_SYNC_NOTIF_CALLBACK, // synchronous write(), notif EVENT_CAN_WRITE_MORE_DATA 170 }; 171 172 /* Constructs an uninitialized AudioTrack. No connection with 173 * AudioFlinger takes place. Use set() after this. 174 */ 175 AudioTrack(); 176 177 /* Creates an AudioTrack object and registers it with AudioFlinger. 178 * Once created, the track needs to be started before it can be used. 179 * Unspecified values are set to appropriate default values. 180 * 181 * Parameters: 182 * 183 * streamType: Select the type of audio stream this track is attached to 184 * (e.g. AUDIO_STREAM_MUSIC). 185 * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate. 186 * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set. 187 * 0 will not work with current policy implementation for direct output 188 * selection where an exact match is needed for sampling rate. 189 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 190 * For direct and offloaded tracks, the possible format(s) depends on the 191 * output sink. 192 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 193 * frameCount: Minimum size of track PCM buffer in frames. This defines the 194 * application's contribution to the 195 * latency of the track. The actual size selected by the AudioTrack could be 196 * larger if the requested size is not compatible with current audio HAL 197 * configuration. Zero means to use a default value. 198 * flags: See comments on audio_output_flags_t in <system/audio.h>. 199 * cbf: Callback function. If not null, this function is called periodically 200 * to provide new data in TRANSFER_CALLBACK mode 201 * and inform of marker, position updates, etc. 202 * user: Context for use by the callback receiver. 203 * notificationFrames: The callback function is called each time notificationFrames PCM 204 * frames have been consumed from track input buffer by server. 205 * Zero means to use a default value, which is typically: 206 * - fast tracks: HAL buffer size, even if track frameCount is larger 207 * - normal tracks: 1/2 of track frameCount 208 * A positive value means that many frames at initial source sample rate. 209 * A negative value for this parameter specifies the negative of the 210 * requested number of notifications (sub-buffers) in the entire buffer. 211 * For fast tracks, the FastMixer will process one sub-buffer at a time. 212 * The size of each sub-buffer is determined by the HAL. 213 * To get "double buffering", for example, one should pass -2. 214 * The minimum number of sub-buffers is 1 (expressed as -1), 215 * and the maximum number of sub-buffers is 8 (expressed as -8). 216 * Negative is only permitted for fast tracks, and if frameCount is zero. 217 * TODO It is ugly to overload a parameter in this way depending on 218 * whether it is positive, negative, or zero. Consider splitting apart. 219 * sessionId: Specific session ID, or zero to use default. 220 * transferType: How data is transferred to AudioTrack. 221 * offloadInfo: If not NULL, provides offload parameters for 222 * AudioSystem::getOutputForAttr(). 223 * uid: User ID of the app which initially requested this AudioTrack 224 * for power management tracking, or -1 for current user ID. 225 * pid: Process ID of the app which initially requested this AudioTrack 226 * for power management tracking, or -1 for current process ID. 227 * pAttributes: If not NULL, supersedes streamType for use case selection. 228 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 229 binder to AudioFlinger. 230 It will return an error instead. The application will recreate 231 the track based on offloading or different channel configuration, etc. 232 * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow 233 * maxRequiredSpeed playback. Values less than 1.0f and greater than 234 * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks 235 * and direct or offloaded tracks, this parameter is ignored. 236 * selectedDeviceId: Selected device id of the app which initially requested the AudioTrack 237 * to open with a specific device. 238 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 239 */ 240 241 AudioTrack( audio_stream_type_t streamType, 242 uint32_t sampleRate, 243 audio_format_t format, 244 audio_channel_mask_t channelMask, 245 size_t frameCount = 0, 246 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 247 callback_t cbf = NULL, 248 void* user = NULL, 249 int32_t notificationFrames = 0, 250 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 251 transfer_type transferType = TRANSFER_DEFAULT, 252 const audio_offload_info_t *offloadInfo = NULL, 253 uid_t uid = AUDIO_UID_INVALID, 254 pid_t pid = -1, 255 const audio_attributes_t* pAttributes = NULL, 256 bool doNotReconnect = false, 257 float maxRequiredSpeed = 1.0f, 258 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 259 260 /* Creates an audio track and registers it with AudioFlinger. 261 * With this constructor, the track is configured for static buffer mode. 262 * Data to be rendered is passed in a shared memory buffer 263 * identified by the argument sharedBuffer, which should be non-0. 264 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 265 * but without the ability to specify a non-zero value for the frameCount parameter. 266 * The memory should be initialized to the desired data before calling start(). 267 * The write() method is not supported in this case. 268 * It is recommended to pass a callback function to be notified of playback end by an 269 * EVENT_UNDERRUN event. 270 */ 271 272 AudioTrack( audio_stream_type_t streamType, 273 uint32_t sampleRate, 274 audio_format_t format, 275 audio_channel_mask_t channelMask, 276 const sp<IMemory>& sharedBuffer, 277 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 278 callback_t cbf = NULL, 279 void* user = NULL, 280 int32_t notificationFrames = 0, 281 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 282 transfer_type transferType = TRANSFER_DEFAULT, 283 const audio_offload_info_t *offloadInfo = NULL, 284 uid_t uid = AUDIO_UID_INVALID, 285 pid_t pid = -1, 286 const audio_attributes_t* pAttributes = NULL, 287 bool doNotReconnect = false, 288 float maxRequiredSpeed = 1.0f); 289 290 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 291 * Also destroys all resources associated with the AudioTrack. 292 */ 293 protected: 294 virtual ~AudioTrack(); 295 public: 296 297 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 298 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 299 * set() is not multi-thread safe. 300 * Returned status (from utils/Errors.h) can be: 301 * - NO_ERROR: successful initialization 302 * - INVALID_OPERATION: AudioTrack is already initialized 303 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 304 * - NO_INIT: audio server or audio hardware not initialized 305 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 306 * If sharedBuffer is non-0, the frameCount parameter is ignored and 307 * replaced by the shared buffer's total allocated size in frame units. 308 * 309 * Parameters not listed in the AudioTrack constructors above: 310 * 311 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 312 * Only set to true when AudioTrack object is used for a java android.media.AudioTrack 313 * in its JNI code. 314 * 315 * Internal state post condition: 316 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 317 */ 318 status_t set(audio_stream_type_t streamType, 319 uint32_t sampleRate, 320 audio_format_t format, 321 audio_channel_mask_t channelMask, 322 size_t frameCount = 0, 323 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 324 callback_t cbf = NULL, 325 void* user = NULL, 326 int32_t notificationFrames = 0, 327 const sp<IMemory>& sharedBuffer = 0, 328 bool threadCanCallJava = false, 329 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 330 transfer_type transferType = TRANSFER_DEFAULT, 331 const audio_offload_info_t *offloadInfo = NULL, 332 uid_t uid = AUDIO_UID_INVALID, 333 pid_t pid = -1, 334 const audio_attributes_t* pAttributes = NULL, 335 bool doNotReconnect = false, 336 float maxRequiredSpeed = 1.0f, 337 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 338 339 /* Result of constructing the AudioTrack. This must be checked for successful initialization 340 * before using any AudioTrack API (except for set()), because using 341 * an uninitialized AudioTrack produces undefined results. 342 * See set() method above for possible return codes. 343 */ initCheck()344 status_t initCheck() const { return mStatus; } 345 346 /* Returns this track's estimated latency in milliseconds. 347 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 348 * and audio hardware driver. 349 */ 350 uint32_t latency(); 351 352 /* Returns the number of application-level buffer underruns 353 * since the AudioTrack was created. 354 */ 355 uint32_t getUnderrunCount() const; 356 357 /* getters, see constructors and set() */ 358 359 audio_stream_type_t streamType() const; format()360 audio_format_t format() const { return mFormat; } 361 362 /* Return frame size in bytes, which for linear PCM is 363 * channelCount * (bit depth per channel / 8). 364 * channelCount is determined from channelMask, and bit depth comes from format. 365 * For non-linear formats, the frame size is typically 1 byte. 366 */ frameSize()367 size_t frameSize() const { return mFrameSize; } 368 channelCount()369 uint32_t channelCount() const { return mChannelCount; } frameCount()370 size_t frameCount() const { return mFrameCount; } 371 372 /* 373 * Return the period of the notification callback in frames. 374 * This value is set when the AudioTrack is constructed. 375 * It can be modified if the AudioTrack is rerouted. 376 */ getNotificationPeriodInFrames()377 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 378 379 /* Return effective size of audio buffer that an application writes to 380 * or a negative error if the track is uninitialized. 381 */ 382 ssize_t getBufferSizeInFrames(); 383 384 /* Returns the buffer duration in microseconds at current playback rate. 385 */ 386 status_t getBufferDurationInUs(int64_t *duration); 387 388 /* Set the effective size of audio buffer that an application writes to. 389 * This is used to determine the amount of available room in the buffer, 390 * which determines when a write will block. 391 * This allows an application to raise and lower the audio latency. 392 * The requested size may be adjusted so that it is 393 * greater or equal to the absolute minimum and 394 * less than or equal to the getBufferCapacityInFrames(). 395 * It may also be adjusted slightly for internal reasons. 396 * 397 * Return the final size or a negative error if the track is unitialized 398 * or does not support variable sizes. 399 */ 400 ssize_t setBufferSizeInFrames(size_t size); 401 402 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ sharedBuffer()403 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 404 405 /* 406 * return metrics information for the current track. 407 */ 408 status_t getMetrics(MediaAnalyticsItem * &item); 409 410 /* After it's created the track is not active. Call start() to 411 * make it active. If set, the callback will start being called. 412 * If the track was previously paused, volume is ramped up over the first mix buffer. 413 */ 414 status_t start(); 415 416 /* Stop a track. 417 * In static buffer mode, the track is stopped immediately. 418 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 419 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 420 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 421 * is first drained, mixed, and output, and only then is the track marked as stopped. 422 */ 423 void stop(); 424 bool stopped() const; 425 426 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 427 * This has the effect of draining the buffers without mixing or output. 428 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 429 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 430 */ 431 void flush(); 432 433 /* Pause a track. After pause, the callback will cease being called and 434 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 435 * and will fill up buffers until the pool is exhausted. 436 * Volume is ramped down over the next mix buffer following the pause request, 437 * and then the track is marked as paused. It can be resumed with ramp up by start(). 438 */ 439 void pause(); 440 441 /* Set volume for this track, mostly used for games' sound effects 442 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 443 * This is the older API. New applications should use setVolume(float) when possible. 444 */ 445 status_t setVolume(float left, float right); 446 447 /* Set volume for all channels. This is the preferred API for new applications, 448 * especially for multi-channel content. 449 */ 450 status_t setVolume(float volume); 451 452 /* Set the send level for this track. An auxiliary effect should be attached 453 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 454 */ 455 status_t setAuxEffectSendLevel(float level); 456 void getAuxEffectSendLevel(float* level) const; 457 458 /* Set source sample rate for this track in Hz, mostly used for games' sound effects. 459 * Zero is not permitted. 460 */ 461 status_t setSampleRate(uint32_t sampleRate); 462 463 /* Return current source sample rate in Hz. 464 * If specified as zero in constructor or set(), this will be the sink sample rate. 465 */ 466 uint32_t getSampleRate() const; 467 468 /* Return the original source sample rate in Hz. This corresponds to the sample rate 469 * if playback rate had normal speed and pitch. 470 */ 471 uint32_t getOriginalSampleRate() const; 472 473 /* Set source playback rate for timestretch 474 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 475 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 476 * 477 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 478 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 479 * 480 * Speed increases the playback rate of media, but does not alter pitch. 481 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 482 */ 483 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 484 485 /* Return current playback rate */ 486 const AudioPlaybackRate& getPlaybackRate() const; 487 488 /* Enables looping and sets the start and end points of looping. 489 * Only supported for static buffer mode. 490 * 491 * Parameters: 492 * 493 * loopStart: loop start in frames relative to start of buffer. 494 * loopEnd: loop end in frames relative to start of buffer. 495 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 496 * pending or active loop. loopCount == -1 means infinite looping. 497 * 498 * For proper operation the following condition must be respected: 499 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 500 * 501 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 502 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 503 * 504 */ 505 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 506 507 /* Sets marker position. When playback reaches the number of frames specified, a callback with 508 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 509 * notification callback. To set a marker at a position which would compute as 0, 510 * a workaround is to set the marker at a nearby position such as ~0 or 1. 511 * If the AudioTrack has been opened with no callback function associated, the operation will 512 * fail. 513 * 514 * Parameters: 515 * 516 * marker: marker position expressed in wrapping (overflow) frame units, 517 * like the return value of getPosition(). 518 * 519 * Returned status (from utils/Errors.h) can be: 520 * - NO_ERROR: successful operation 521 * - INVALID_OPERATION: the AudioTrack has no callback installed. 522 */ 523 status_t setMarkerPosition(uint32_t marker); 524 status_t getMarkerPosition(uint32_t *marker) const; 525 526 /* Sets position update period. Every time the number of frames specified has been played, 527 * a callback with event type EVENT_NEW_POS is called. 528 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 529 * callback. 530 * If the AudioTrack has been opened with no callback function associated, the operation will 531 * fail. 532 * Extremely small values may be rounded up to a value the implementation can support. 533 * 534 * Parameters: 535 * 536 * updatePeriod: position update notification period expressed in frames. 537 * 538 * Returned status (from utils/Errors.h) can be: 539 * - NO_ERROR: successful operation 540 * - INVALID_OPERATION: the AudioTrack has no callback installed. 541 */ 542 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 543 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 544 545 /* Sets playback head position. 546 * Only supported for static buffer mode. 547 * 548 * Parameters: 549 * 550 * position: New playback head position in frames relative to start of buffer. 551 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 552 * but will result in an immediate underrun if started. 553 * 554 * Returned status (from utils/Errors.h) can be: 555 * - NO_ERROR: successful operation 556 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 557 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 558 * buffer 559 */ 560 status_t setPosition(uint32_t position); 561 562 /* Return the total number of frames played since playback start. 563 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 564 * It is reset to zero by flush(), reload(), and stop(). 565 * 566 * Parameters: 567 * 568 * position: Address where to return play head position. 569 * 570 * Returned status (from utils/Errors.h) can be: 571 * - NO_ERROR: successful operation 572 * - BAD_VALUE: position is NULL 573 */ 574 status_t getPosition(uint32_t *position); 575 576 /* For static buffer mode only, this returns the current playback position in frames 577 * relative to start of buffer. It is analogous to the position units used by 578 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 579 */ 580 status_t getBufferPosition(uint32_t *position); 581 582 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 583 * rewriting the buffer before restarting playback after a stop. 584 * This method must be called with the AudioTrack in paused or stopped state. 585 * Not allowed in streaming mode. 586 * 587 * Returned status (from utils/Errors.h) can be: 588 * - NO_ERROR: successful operation 589 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 590 */ 591 status_t reload(); 592 593 /** 594 * @param transferType 595 * @return text string that matches the enum name 596 */ 597 static const char * convertTransferToText(transfer_type transferType); 598 599 /* Returns a handle on the audio output used by this AudioTrack. 600 * 601 * Parameters: 602 * none. 603 * 604 * Returned value: 605 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 606 * track needed to be re-created but that failed 607 */ 608 private: 609 audio_io_handle_t getOutput() const; 610 public: 611 612 /* Selects the audio device to use for output of this AudioTrack. A value of 613 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 614 * 615 * Parameters: 616 * The device ID of the selected device (as returned by the AudioDevicesManager API). 617 * 618 * Returned value: 619 * - NO_ERROR: successful operation 620 * TODO: what else can happen here? 621 */ 622 status_t setOutputDevice(audio_port_handle_t deviceId); 623 624 /* Returns the ID of the audio device selected for this AudioTrack. 625 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 626 * 627 * Parameters: 628 * none. 629 */ 630 audio_port_handle_t getOutputDevice(); 631 632 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 633 * attached. 634 * When the AudioTrack is inactive, the device ID returned can be either: 635 * - AUDIO_PORT_HANDLE_NONE if the AudioTrack is not attached to any output. 636 * - The device ID used before paused or stopped. 637 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioTrack 638 * has not been started yet. 639 * 640 * Parameters: 641 * none. 642 */ 643 audio_port_handle_t getRoutedDeviceId(); 644 645 /* Returns the unique session ID associated with this track. 646 * 647 * Parameters: 648 * none. 649 * 650 * Returned value: 651 * AudioTrack session ID. 652 */ getSessionId()653 audio_session_t getSessionId() const { return mSessionId; } 654 655 /* Attach track auxiliary output to specified effect. Use effectId = 0 656 * to detach track from effect. 657 * 658 * Parameters: 659 * 660 * effectId: effectId obtained from AudioEffect::id(). 661 * 662 * Returned status (from utils/Errors.h) can be: 663 * - NO_ERROR: successful operation 664 * - INVALID_OPERATION: the effect is not an auxiliary effect. 665 * - BAD_VALUE: The specified effect ID is invalid 666 */ 667 status_t attachAuxEffect(int effectId); 668 669 /* Public API for TRANSFER_OBTAIN mode. 670 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 671 * After filling these slots with data, the caller should release them with releaseBuffer(). 672 * If the track buffer is not full, obtainBuffer() returns as many contiguous 673 * [empty slots for] frames as are available immediately. 674 * 675 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 676 * additional non-contiguous frames that are predicted to be available immediately, 677 * if the client were to release the first frames and then call obtainBuffer() again. 678 * This value is only a prediction, and needs to be confirmed. 679 * It will be set to zero for an error return. 680 * 681 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 682 * regardless of the value of waitCount. 683 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 684 * maximum timeout based on waitCount; see chart below. 685 * Buffers will be returned until the pool 686 * is exhausted, at which point obtainBuffer() will either block 687 * or return WOULD_BLOCK depending on the value of the "waitCount" 688 * parameter. 689 * 690 * Interpretation of waitCount: 691 * +n limits wait time to n * WAIT_PERIOD_MS, 692 * -1 causes an (almost) infinite wait time, 693 * 0 non-blocking. 694 * 695 * Buffer fields 696 * On entry: 697 * frameCount number of [empty slots for] frames requested 698 * size ignored 699 * raw ignored 700 * sequence ignored 701 * After error return: 702 * frameCount 0 703 * size 0 704 * raw undefined 705 * sequence undefined 706 * After successful return: 707 * frameCount actual number of [empty slots for] frames available, <= number requested 708 * size actual number of bytes available 709 * raw pointer to the buffer 710 * sequence IAudioTrack instance sequence number, as of obtainBuffer() 711 */ 712 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 713 size_t *nonContig = NULL); 714 715 private: 716 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 717 * additional non-contiguous frames that are predicted to be available immediately, 718 * if the client were to release the first frames and then call obtainBuffer() again. 719 * This value is only a prediction, and needs to be confirmed. 720 * It will be set to zero for an error return. 721 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 722 * in case the requested amount of frames is in two or more non-contiguous regions. 723 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 724 */ 725 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 726 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 727 public: 728 729 /* Public API for TRANSFER_OBTAIN mode. 730 * Release a filled buffer of frames for AudioFlinger to process. 731 * 732 * Buffer fields: 733 * frameCount currently ignored but recommend to set to actual number of frames filled 734 * size actual number of bytes filled, must be multiple of frameSize 735 * raw ignored 736 */ 737 void releaseBuffer(const Buffer* audioBuffer); 738 739 /* As a convenience we provide a write() interface to the audio buffer. 740 * Input parameter 'size' is in byte units. 741 * This is implemented on top of obtainBuffer/releaseBuffer. For best 742 * performance use callbacks. Returns actual number of bytes written >= 0, 743 * or one of the following negative status codes: 744 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 745 * BAD_VALUE size is invalid 746 * WOULD_BLOCK when obtainBuffer() returns same, or 747 * AudioTrack was stopped during the write 748 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 749 * the track cannot be automatically restored. 750 * The application needs to recreate the AudioTrack 751 * because the audio device changed or AudioFlinger died. 752 * This typically occurs for direct or offload tracks 753 * or if mDoNotReconnect is true. 754 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 755 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 756 * false for the method to return immediately without waiting to try multiple times to write 757 * the full content of the buffer. 758 */ 759 ssize_t write(const void* buffer, size_t size, bool blocking = true); 760 761 /* 762 * Dumps the state of an audio track. 763 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 764 */ 765 status_t dump(int fd, const Vector<String16>& args) const; 766 767 /* 768 * Return the total number of frames which AudioFlinger desired but were unavailable, 769 * and thus which resulted in an underrun. Reset to zero by stop(). 770 */ 771 uint32_t getUnderrunFrames() const; 772 773 /* Get the flags */ getFlags()774 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 775 776 /* Set parameters - only possible when using direct output */ 777 status_t setParameters(const String8& keyValuePairs); 778 779 /* Sets the volume shaper object */ 780 media::VolumeShaper::Status applyVolumeShaper( 781 const sp<media::VolumeShaper::Configuration>& configuration, 782 const sp<media::VolumeShaper::Operation>& operation); 783 784 /* Gets the volume shaper state */ 785 sp<media::VolumeShaper::State> getVolumeShaperState(int id); 786 787 /* Selects the presentation (if available) */ 788 status_t selectPresentation(int presentationId, int programId); 789 790 /* Get parameters */ 791 String8 getParameters(const String8& keys); 792 793 /* Poll for a timestamp on demand. 794 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 795 * or if you need to get the most recent timestamp outside of the event callback handler. 796 * Caution: calling this method too often may be inefficient; 797 * if you need a high resolution mapping between frame position and presentation time, 798 * consider implementing that at application level, based on the low resolution timestamps. 799 * Returns NO_ERROR if timestamp is valid. 800 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 801 * start/ACTIVE, when the number of frames consumed is less than the 802 * overall hardware latency to physical output. In WOULD_BLOCK cases, 803 * one might poll again, or use getPosition(), or use 0 position and 804 * current time for the timestamp. 805 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 806 * the track cannot be automatically restored. 807 * The application needs to recreate the AudioTrack 808 * because the audio device changed or AudioFlinger died. 809 * This typically occurs for direct or offload tracks 810 * or if mDoNotReconnect is true. 811 * INVALID_OPERATION wrong state, or some other error. 812 * 813 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 814 */ 815 status_t getTimestamp(AudioTimestamp& timestamp); 816 private: 817 status_t getTimestamp_l(AudioTimestamp& timestamp); 818 public: 819 820 /* Return the extended timestamp, with additional timebase info and improved drain behavior. 821 * 822 * This is similar to the AudioTrack.java API: 823 * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) 824 * 825 * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method 826 * 827 * 1. stop() by itself does not reset the frame position. 828 * A following start() resets the frame position to 0. 829 * 2. flush() by itself does not reset the frame position. 830 * The frame position advances by the number of frames flushed, 831 * when the first frame after flush reaches the audio sink. 832 * 3. BOOTTIME clock offsets are provided to help synchronize with 833 * non-audio streams, e.g. sensor data. 834 * 4. Position is returned with 64 bits of resolution. 835 * 836 * Parameters: 837 * timestamp: A pointer to the caller allocated ExtendedTimestamp. 838 * 839 * Returns NO_ERROR on success; timestamp is filled with valid data. 840 * BAD_VALUE if timestamp is NULL. 841 * WOULD_BLOCK if called immediately after start() when the number 842 * of frames consumed is less than the 843 * overall hardware latency to physical output. In WOULD_BLOCK cases, 844 * one might poll again, or use getPosition(), or use 0 position and 845 * current time for the timestamp. 846 * If WOULD_BLOCK is returned, the timestamp is still 847 * modified with the LOCATION_CLIENT portion filled. 848 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 849 * the track cannot be automatically restored. 850 * The application needs to recreate the AudioTrack 851 * because the audio device changed or AudioFlinger died. 852 * This typically occurs for direct or offloaded tracks 853 * or if mDoNotReconnect is true. 854 * INVALID_OPERATION if called on a offloaded or direct track. 855 * Use getTimestamp(AudioTimestamp& timestamp) instead. 856 */ 857 status_t getTimestamp(ExtendedTimestamp *timestamp); 858 private: 859 status_t getTimestamp_l(ExtendedTimestamp *timestamp); 860 public: 861 862 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 863 * AudioTrack is routed is updated. 864 * Replaces any previously installed callback. 865 * Parameters: 866 * callback: The callback interface 867 * Returns NO_ERROR if successful. 868 * INVALID_OPERATION if the same callback is already installed. 869 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 870 * BAD_VALUE if the callback is NULL 871 */ 872 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 873 874 /* remove an AudioDeviceCallback. 875 * Parameters: 876 * callback: The callback interface 877 * Returns NO_ERROR if successful. 878 * INVALID_OPERATION if the callback is not installed 879 * BAD_VALUE if the callback is NULL 880 */ 881 status_t removeAudioDeviceCallback( 882 const sp<AudioSystem::AudioDeviceCallback>& callback); 883 884 // AudioSystem::AudioDeviceCallback> virtuals 885 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 886 audio_port_handle_t deviceId); 887 888 889 890 /* Obtain the pending duration in milliseconds for playback of pure PCM 891 * (mixable without embedded timing) data remaining in AudioTrack. 892 * 893 * This is used to estimate the drain time for the client-server buffer 894 * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. 895 * One may optionally request to find the duration to play through the HAL 896 * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, 897 * INVALID_OPERATION may be returned if the kernel location is unavailable. 898 * 899 * Returns NO_ERROR if successful. 900 * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained 901 * or the AudioTrack does not contain pure PCM data. 902 * BAD_VALUE if msec is nullptr or location is invalid. 903 */ 904 status_t pendingDuration(int32_t *msec, 905 ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); 906 907 /* hasStarted() is used to determine if audio is now audible at the device after 908 * a start() command. The underlying implementation checks a nonzero timestamp position 909 * or increment for the audible assumption. 910 * 911 * hasStarted() returns true if the track has been started() and audio is audible 912 * and no subsequent pause() or flush() has been called. Immediately after pause() or 913 * flush() hasStarted() will return false. 914 * 915 * If stop() has been called, hasStarted() will return true if audio is still being 916 * delivered or has finished delivery (even if no audio was written) for both offloaded 917 * and normal tracks. This property removes a race condition in checking hasStarted() 918 * for very short clips, where stop() must be called to finish drain. 919 * 920 * In all cases, hasStarted() may turn false briefly after a subsequent start() is called 921 * until audio becomes audible again. 922 */ 923 bool hasStarted(); // not const 924 isPlaying()925 bool isPlaying() { 926 AutoMutex lock(mLock); 927 return mState == STATE_ACTIVE || mState == STATE_STOPPING; 928 } 929 930 /* Get the unique port ID assigned to this AudioTrack instance by audio policy manager. 931 * The ID is unique across all audioserver clients and can change during the life cycle 932 * of a given AudioTrack instance if the connection to audioserver is restored. 933 */ getPortId()934 audio_port_handle_t getPortId() const { return mPortId; }; 935 936 protected: 937 /* copying audio tracks is not allowed */ 938 AudioTrack(const AudioTrack& other); 939 AudioTrack& operator = (const AudioTrack& other); 940 941 /* a small internal class to handle the callback */ 942 class AudioTrackThread : public Thread 943 { 944 public: 945 AudioTrackThread(AudioTrack& receiver); 946 947 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 948 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 949 virtual void requestExit(); 950 951 void pause(); // suspend thread from execution at next loop boundary 952 void resume(); // allow thread to execute, if not requested to exit 953 void wake(); // wake to handle changed notification conditions. 954 955 private: 956 void pauseInternal(nsecs_t ns = 0LL); 957 // like pause(), but only used internally within thread 958 959 friend class AudioTrack; 960 virtual bool threadLoop(); 961 AudioTrack& mReceiver; 962 virtual ~AudioTrackThread(); 963 Mutex mMyLock; // Thread::mLock is private 964 Condition mMyCond; // Thread::mThreadExitedCondition is private 965 bool mPaused; // whether thread is requested to pause at next loop entry 966 bool mPausedInt; // whether thread internally requests pause 967 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 968 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 969 // to processAudioBuffer() as state may have changed 970 // since pause time calculated. 971 }; 972 973 // body of AudioTrackThread::threadLoop() 974 // returns the maximum amount of time before we would like to run again, where: 975 // 0 immediately 976 // > 0 no later than this many nanoseconds from now 977 // NS_WHENEVER still active but no particular deadline 978 // NS_INACTIVE inactive so don't run again until re-started 979 // NS_NEVER never again 980 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 981 nsecs_t processAudioBuffer(); 982 983 // caller must hold lock on mLock for all _l methods 984 985 void updateLatency_l(); // updates mAfLatency and mLatency from AudioSystem cache 986 987 status_t createTrack_l(); 988 989 // can only be called when mState != STATE_ACTIVE 990 void flush_l(); 991 992 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 993 994 // FIXME enum is faster than strcmp() for parameter 'from' 995 status_t restoreTrack_l(const char *from); 996 997 uint32_t getUnderrunCount_l() const; 998 999 bool isOffloaded() const; 1000 bool isDirect() const; 1001 bool isOffloadedOrDirect() const; 1002 isOffloaded_l()1003 bool isOffloaded_l() const 1004 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 1005 isOffloadedOrDirect_l()1006 bool isOffloadedOrDirect_l() const 1007 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 1008 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 1009 isDirect_l()1010 bool isDirect_l() const 1011 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 1012 1013 // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) isPurePcmData_l()1014 bool isPurePcmData_l() const 1015 { return audio_is_linear_pcm(mFormat) 1016 && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } 1017 1018 // increment mPosition by the delta of mServer, and return new value of mPosition 1019 Modulo<uint32_t> updateAndGetPosition_l(); 1020 1021 // check sample rate and speed is compatible with AudioTrack 1022 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed); 1023 1024 void restartIfDisabled(); 1025 1026 void updateRoutedDeviceId_l(); 1027 1028 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 1029 sp<IAudioTrack> mAudioTrack; 1030 sp<IMemory> mCblkMemory; 1031 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 1032 audio_io_handle_t mOutput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getOutputForAttr() 1033 1034 sp<AudioTrackThread> mAudioTrackThread; 1035 bool mThreadCanCallJava; 1036 1037 float mVolume[2]; 1038 float mSendLevel; 1039 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 1040 uint32_t mOriginalSampleRate; 1041 AudioPlaybackRate mPlaybackRate; 1042 float mMaxRequiredSpeed; // use PCM buffer size to allow this speed 1043 1044 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 1045 // This allocated buffer size is maintained by the proxy. 1046 size_t mFrameCount; // maximum size of buffer 1047 1048 size_t mReqFrameCount; // frame count to request the first or next time 1049 // a new IAudioTrack is needed, non-decreasing 1050 1051 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 1052 // These values can be used for informational purposes until the track is invalidated, 1053 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 1054 uint32_t mAfLatency; // AudioFlinger latency in ms 1055 size_t mAfFrameCount; // AudioFlinger frame count 1056 uint32_t mAfSampleRate; // AudioFlinger sample rate 1057 1058 // constant after constructor or set() 1059 audio_format_t mFormat; // as requested by client, not forced to 16-bit 1060 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 1061 // this AudioTrack has valid attributes 1062 uint32_t mChannelCount; 1063 audio_channel_mask_t mChannelMask; 1064 sp<IMemory> mSharedBuffer; 1065 transfer_type mTransfer; 1066 audio_offload_info_t mOffloadInfoCopy; 1067 const audio_offload_info_t* mOffloadInfo; 1068 audio_attributes_t mAttributes; 1069 1070 size_t mFrameSize; // frame size in bytes 1071 1072 status_t mStatus; 1073 1074 // can change dynamically when IAudioTrack invalidated 1075 uint32_t mLatency; // in ms 1076 1077 // Indicates the current track state. Protected by mLock. 1078 enum State { 1079 STATE_ACTIVE, 1080 STATE_STOPPED, 1081 STATE_PAUSED, 1082 STATE_PAUSED_STOPPING, 1083 STATE_FLUSHED, 1084 STATE_STOPPING, 1085 } mState; 1086 stateToString(State state)1087 static constexpr const char *stateToString(State state) 1088 { 1089 switch (state) { 1090 case STATE_ACTIVE: return "STATE_ACTIVE"; 1091 case STATE_STOPPED: return "STATE_STOPPED"; 1092 case STATE_PAUSED: return "STATE_PAUSED"; 1093 case STATE_PAUSED_STOPPING: return "STATE_PAUSED_STOPPING"; 1094 case STATE_FLUSHED: return "STATE_FLUSHED"; 1095 case STATE_STOPPING: return "STATE_STOPPING"; 1096 default: return "UNKNOWN"; 1097 } 1098 } 1099 1100 // for client callback handler 1101 callback_t mCbf; // callback handler for events, or NULL 1102 void* mUserData; 1103 1104 // for notification APIs 1105 1106 // next 2 fields are const after constructor or set() 1107 uint32_t mNotificationFramesReq; // requested number of frames between each 1108 // notification callback, 1109 // at initial source sample rate 1110 uint32_t mNotificationsPerBufferReq; 1111 // requested number of notifications per buffer, 1112 // currently only used for fast tracks with 1113 // default track buffer size 1114 1115 uint32_t mNotificationFramesAct; // actual number of frames between each 1116 // notification callback, 1117 // at initial source sample rate 1118 bool mRefreshRemaining; // processAudioBuffer() should refresh 1119 // mRemainingFrames and mRetryOnPartialBuffer 1120 1121 // used for static track cbf and restoration 1122 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 1123 uint32_t mLoopStart; // last setLoop loopStart 1124 uint32_t mLoopEnd; // last setLoop loopEnd 1125 int32_t mLoopCountNotified; // the last loopCount notified by callback. 1126 // mLoopCountNotified counts down, matching 1127 // the remaining loop count for static track 1128 // playback. 1129 1130 // These are private to processAudioBuffer(), and are not protected by a lock 1131 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 1132 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 1133 uint32_t mObservedSequence; // last observed value of mSequence 1134 1135 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 1136 bool mMarkerReached; 1137 Modulo<uint32_t> mNewPosition; // in frames 1138 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 1139 1140 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 1141 // which is count of frames consumed by server, 1142 // reset by new IAudioTrack, 1143 // whether it is reset by stop() is TBD 1144 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 1145 // monotonically after new IAudioTrack, 1146 // and could be easily widened to uint64_t 1147 Modulo<uint32_t> mReleased; // count of frames released to server 1148 // but not necessarily consumed by server, 1149 // reset by stop() but continues monotonically 1150 // after new IAudioTrack to restore mPosition, 1151 // and could be easily widened to uint64_t 1152 int64_t mStartFromZeroUs; // the start time after flush or stop, 1153 // when position should be 0. 1154 // only used for offloaded and direct tracks. 1155 int64_t mStartNs; // the time when start() is called. 1156 ExtendedTimestamp mStartEts; // Extended timestamp at start for normal 1157 // AudioTracks. 1158 AudioTimestamp mStartTs; // Timestamp at start for offloaded or direct 1159 // AudioTracks. 1160 1161 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 1162 bool mTimestampStartupGlitchReported; // reduce log spam 1163 bool mTimestampRetrogradePositionReported; // reduce log spam 1164 bool mTimestampRetrogradeTimeReported; // reduce log spam 1165 bool mTimestampStallReported; // reduce log spam 1166 bool mTimestampStaleTimeReported; // reduce log spam 1167 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 1168 ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp 1169 1170 uint32_t mUnderrunCountOffset; // updated when restoring tracks 1171 1172 int64_t mFramesWritten; // total frames written. reset to zero after 1173 // the start() following stop(). It is not 1174 // changed after restoring the track or 1175 // after flush. 1176 int64_t mFramesWrittenServerOffset; // An offset to server frames due to 1177 // restoring AudioTrack, or stop/start. 1178 // This offset is also used for static tracks. 1179 int64_t mFramesWrittenAtRestore; // Frames written at restore point (or frames 1180 // delivered for static tracks). 1181 // -1 indicates no previous restore point. 1182 1183 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 1184 // be denied by client or server, such as 1185 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 1186 // held to read or write those bits reliably. 1187 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 1188 1189 bool mDoNotReconnect; 1190 1191 audio_session_t mSessionId; 1192 int mAuxEffectId; 1193 audio_port_handle_t mPortId; // Id from Audio Policy Manager 1194 1195 mutable Mutex mLock; 1196 1197 int mPreviousPriority; // before start() 1198 SchedPolicy mPreviousSchedulingGroup; 1199 bool mAwaitBoost; // thread should wait for priority boost before running 1200 1201 // The proxy should only be referenced while a lock is held because the proxy isn't 1202 // multi-thread safe, especially the SingleStateQueue part of the proxy. 1203 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 1204 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 1205 // them around in case they are replaced during the obtainBuffer(). 1206 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 1207 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 1208 1209 bool mInUnderrun; // whether track is currently in underrun state 1210 uint32_t mPausedPosition; 1211 1212 // For Device Selection API 1213 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 1214 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 1215 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 1216 // May not match the app selection depending on other 1217 // activity and connected devices. 1218 1219 sp<media::VolumeHandler> mVolumeHandler; 1220 1221 private: 1222 class DeathNotifier : public IBinder::DeathRecipient { 1223 public: DeathNotifier(AudioTrack * audioTrack)1224 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 1225 protected: 1226 virtual void binderDied(const wp<IBinder>& who); 1227 private: 1228 const wp<AudioTrack> mAudioTrack; 1229 }; 1230 1231 sp<DeathNotifier> mDeathNotifier; 1232 uint32_t mSequence; // incremented for each new IAudioTrack attempt 1233 uid_t mClientUid; 1234 pid_t mClientPid; 1235 1236 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1237 1238 private: 1239 class MediaMetrics { 1240 public: MediaMetrics()1241 MediaMetrics() : mAnalyticsItem(MediaAnalyticsItem::create("audiotrack")) { 1242 } ~MediaMetrics()1243 ~MediaMetrics() { 1244 // mAnalyticsItem alloc failure will be flagged in the constructor 1245 // don't log empty records 1246 if (mAnalyticsItem->count() > 0) { 1247 mAnalyticsItem->selfrecord(); 1248 } 1249 } 1250 void gather(const AudioTrack *track); dup()1251 MediaAnalyticsItem *dup() { return mAnalyticsItem->dup(); } 1252 private: 1253 std::unique_ptr<MediaAnalyticsItem> mAnalyticsItem; 1254 }; 1255 MediaMetrics mMediaMetrics; 1256 }; 1257 1258 }; // namespace android 1259 1260 #endif // ANDROID_AUDIOTRACK_H 1261