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/MediaMetricsItem.h> 26 #include <media/Modulo.h> 27 #include <utils/threads.h> 28 29 #include <string> 30 31 #include "android/media/BnAudioTrackCallback.h" 32 #include "android/media/IAudioTrackCallback.h" 33 34 namespace android { 35 36 // ---------------------------------------------------------------------------- 37 38 struct audio_track_cblk_t; 39 class AudioTrackClientProxy; 40 class StaticAudioTrackClientProxy; 41 42 // ---------------------------------------------------------------------------- 43 44 class AudioTrack : public AudioSystem::AudioDeviceCallback 45 { 46 public: 47 48 /* Events used by AudioTrack callback function (callback_t). 49 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 50 */ 51 enum event_type { 52 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 53 // This event only occurs for TRANSFER_CALLBACK. 54 // If this event is delivered but the callback handler 55 // does not want to write more data, the handler must 56 // ignore the event by setting frameCount to zero. 57 // This might occur, for example, if the application is 58 // waiting for source data or is at the end of stream. 59 // 60 // For data filling, it is preferred that the callback 61 // does not block and instead returns a short count on 62 // the amount of data actually delivered 63 // (or 0, if no data is currently available). 64 EVENT_UNDERRUN = 1, // Buffer underrun occurred. This will not occur for 65 // static tracks. 66 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 67 // loop start if loop count was not 0 for a static track. 68 EVENT_MARKER = 3, // Playback head is at the specified marker position 69 // (See setMarkerPosition()). 70 EVENT_NEW_POS = 4, // Playback head is at a new position 71 // (See setPositionUpdatePeriod()). 72 EVENT_BUFFER_END = 5, // Playback has completed for a static track. 73 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 74 // voluntary invalidation by mediaserver, or mediaserver crash. 75 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 76 // back (after stop is called) for an offloaded track. 77 #if 0 // FIXME not yet implemented 78 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 79 // in the mapping from frame position to presentation time. 80 // See AudioTimestamp for the information included with event. 81 #endif 82 EVENT_CAN_WRITE_MORE_DATA = 9,// Notification that more data can be given by write() 83 // This event only occurs for TRANSFER_SYNC_NOTIF_CALLBACK. 84 }; 85 86 /* Client should declare a Buffer and pass the address to obtainBuffer() 87 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 88 */ 89 90 class Buffer 91 { 92 public: 93 // FIXME use m prefix 94 size_t frameCount; // number of sample frames corresponding to size; 95 // on input to obtainBuffer() it is the number of frames desired, 96 // on output from obtainBuffer() it is the number of available 97 // [empty slots for] frames to be filled 98 // on input to releaseBuffer() it is currently ignored 99 100 size_t size; // input/output in bytes == frameCount * frameSize 101 // on input to obtainBuffer() it is ignored 102 // on output from obtainBuffer() it is the number of available 103 // [empty slots for] bytes to be filled, 104 // which is frameCount * frameSize 105 // on input to releaseBuffer() it is the number of bytes to 106 // release 107 // FIXME This is redundant with respect to frameCount. Consider 108 // removing size and making frameCount the primary field. 109 110 union { 111 void* raw; 112 int16_t* i16; // signed 16-bit 113 int8_t* i8; // unsigned 8-bit, offset by 0x80 114 }; // input to obtainBuffer(): unused, output: pointer to buffer 115 116 uint32_t sequence; // IAudioTrack instance sequence number, as of obtainBuffer(). 117 // It is set by obtainBuffer() and confirmed by releaseBuffer(). 118 // Not "user-serviceable". 119 // TODO Consider sp<IMemory> instead, or in addition to this. 120 }; 121 122 /* As a convenience, if a callback is supplied, a handler thread 123 * is automatically created with the appropriate priority. This thread 124 * invokes the callback when a new buffer becomes available or various conditions occur. 125 * Parameters: 126 * 127 * event: type of event notified (see enum AudioTrack::event_type). 128 * user: Pointer to context for use by the callback receiver. 129 * info: Pointer to optional parameter according to event type: 130 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 131 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 132 * written. 133 * - EVENT_UNDERRUN: unused. 134 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 135 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 136 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 137 * - EVENT_BUFFER_END: unused. 138 * - EVENT_NEW_IAUDIOTRACK: unused. 139 * - EVENT_STREAM_END: unused. 140 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 141 */ 142 143 typedef void (*callback_t)(int event, void* user, void *info); 144 145 /* Returns the minimum frame count required for the successful creation of 146 * an AudioTrack object. 147 * Returned status (from utils/Errors.h) can be: 148 * - NO_ERROR: successful operation 149 * - NO_INIT: audio server or audio hardware not initialized 150 * - BAD_VALUE: unsupported configuration 151 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 152 * and is undefined otherwise. 153 * FIXME This API assumes a route, and so should be deprecated. 154 */ 155 156 static status_t getMinFrameCount(size_t* frameCount, 157 audio_stream_type_t streamType, 158 uint32_t sampleRate); 159 160 /* Check if direct playback is possible for the given audio configuration and attributes. 161 * Return true if output is possible for the given parameters. Otherwise returns false. 162 */ 163 static bool isDirectOutputSupported(const audio_config_base_t& config, 164 const audio_attributes_t& attributes); 165 166 /* How data is transferred to AudioTrack 167 */ 168 enum transfer_type { 169 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 170 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 171 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 172 TRANSFER_SYNC, // synchronous write() 173 TRANSFER_SHARED, // shared memory 174 TRANSFER_SYNC_NOTIF_CALLBACK, // synchronous write(), notif EVENT_CAN_WRITE_MORE_DATA 175 }; 176 177 /* Constructs an uninitialized AudioTrack. No connection with 178 * AudioFlinger takes place. Use set() after this. 179 */ 180 AudioTrack(); 181 182 AudioTrack(const std::string& opPackageName); 183 184 /* Creates an AudioTrack object and registers it with AudioFlinger. 185 * Once created, the track needs to be started before it can be used. 186 * Unspecified values are set to appropriate default values. 187 * 188 * Parameters: 189 * 190 * streamType: Select the type of audio stream this track is attached to 191 * (e.g. AUDIO_STREAM_MUSIC). 192 * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate. 193 * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set. 194 * 0 will not work with current policy implementation for direct output 195 * selection where an exact match is needed for sampling rate. 196 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 197 * For direct and offloaded tracks, the possible format(s) depends on the 198 * output sink. 199 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 200 * frameCount: Minimum size of track PCM buffer in frames. This defines the 201 * application's contribution to the 202 * latency of the track. The actual size selected by the AudioTrack could be 203 * larger if the requested size is not compatible with current audio HAL 204 * configuration. Zero means to use a default value. 205 * flags: See comments on audio_output_flags_t in <system/audio.h>. 206 * cbf: Callback function. If not null, this function is called periodically 207 * to provide new data in TRANSFER_CALLBACK mode 208 * and inform of marker, position updates, etc. 209 * user: Context for use by the callback receiver. 210 * notificationFrames: The callback function is called each time notificationFrames PCM 211 * frames have been consumed from track input buffer by server. 212 * Zero means to use a default value, which is typically: 213 * - fast tracks: HAL buffer size, even if track frameCount is larger 214 * - normal tracks: 1/2 of track frameCount 215 * A positive value means that many frames at initial source sample rate. 216 * A negative value for this parameter specifies the negative of the 217 * requested number of notifications (sub-buffers) in the entire buffer. 218 * For fast tracks, the FastMixer will process one sub-buffer at a time. 219 * The size of each sub-buffer is determined by the HAL. 220 * To get "double buffering", for example, one should pass -2. 221 * The minimum number of sub-buffers is 1 (expressed as -1), 222 * and the maximum number of sub-buffers is 8 (expressed as -8). 223 * Negative is only permitted for fast tracks, and if frameCount is zero. 224 * TODO It is ugly to overload a parameter in this way depending on 225 * whether it is positive, negative, or zero. Consider splitting apart. 226 * sessionId: Specific session ID, or zero to use default. 227 * transferType: How data is transferred to AudioTrack. 228 * offloadInfo: If not NULL, provides offload parameters for 229 * AudioSystem::getOutputForAttr(). 230 * uid: User ID of the app which initially requested this AudioTrack 231 * for power management tracking, or -1 for current user ID. 232 * pid: Process ID of the app which initially requested this AudioTrack 233 * for power management tracking, or -1 for current process ID. 234 * pAttributes: If not NULL, supersedes streamType for use case selection. 235 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 236 binder to AudioFlinger. 237 It will return an error instead. The application will recreate 238 the track based on offloading or different channel configuration, etc. 239 * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow 240 * maxRequiredSpeed playback. Values less than 1.0f and greater than 241 * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks 242 * and direct or offloaded tracks, this parameter is ignored. 243 * selectedDeviceId: Selected device id of the app which initially requested the AudioTrack 244 * to open with a specific device. 245 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 246 */ 247 248 AudioTrack( audio_stream_type_t streamType, 249 uint32_t sampleRate, 250 audio_format_t format, 251 audio_channel_mask_t channelMask, 252 size_t frameCount = 0, 253 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 254 callback_t cbf = NULL, 255 void* user = NULL, 256 int32_t notificationFrames = 0, 257 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 258 transfer_type transferType = TRANSFER_DEFAULT, 259 const audio_offload_info_t *offloadInfo = NULL, 260 uid_t uid = AUDIO_UID_INVALID, 261 pid_t pid = -1, 262 const audio_attributes_t* pAttributes = NULL, 263 bool doNotReconnect = false, 264 float maxRequiredSpeed = 1.0f, 265 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, 266 const std::string& opPackageName = ""); 267 268 /* Creates an audio track and registers it with AudioFlinger. 269 * With this constructor, the track is configured for static buffer mode. 270 * Data to be rendered is passed in a shared memory buffer 271 * identified by the argument sharedBuffer, which should be non-0. 272 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 273 * but without the ability to specify a non-zero value for the frameCount parameter. 274 * The memory should be initialized to the desired data before calling start(). 275 * The write() method is not supported in this case. 276 * It is recommended to pass a callback function to be notified of playback end by an 277 * EVENT_UNDERRUN event. 278 */ 279 280 AudioTrack( audio_stream_type_t streamType, 281 uint32_t sampleRate, 282 audio_format_t format, 283 audio_channel_mask_t channelMask, 284 const sp<IMemory>& sharedBuffer, 285 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 286 callback_t cbf = NULL, 287 void* user = NULL, 288 int32_t notificationFrames = 0, 289 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 290 transfer_type transferType = TRANSFER_DEFAULT, 291 const audio_offload_info_t *offloadInfo = NULL, 292 uid_t uid = AUDIO_UID_INVALID, 293 pid_t pid = -1, 294 const audio_attributes_t* pAttributes = NULL, 295 bool doNotReconnect = false, 296 float maxRequiredSpeed = 1.0f, 297 const std::string& opPackageName = ""); 298 299 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 300 * Also destroys all resources associated with the AudioTrack. 301 */ 302 protected: 303 virtual ~AudioTrack(); 304 public: 305 306 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 307 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 308 * set() is not multi-thread safe. 309 * Returned status (from utils/Errors.h) can be: 310 * - NO_ERROR: successful initialization 311 * - INVALID_OPERATION: AudioTrack is already initialized 312 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 313 * - NO_INIT: audio server or audio hardware not initialized 314 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 315 * If sharedBuffer is non-0, the frameCount parameter is ignored and 316 * replaced by the shared buffer's total allocated size in frame units. 317 * 318 * Parameters not listed in the AudioTrack constructors above: 319 * 320 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 321 * Only set to true when AudioTrack object is used for a java android.media.AudioTrack 322 * in its JNI code. 323 * 324 * Internal state post condition: 325 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 326 */ 327 status_t set(audio_stream_type_t streamType, 328 uint32_t sampleRate, 329 audio_format_t format, 330 audio_channel_mask_t channelMask, 331 size_t frameCount = 0, 332 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 333 callback_t cbf = NULL, 334 void* user = NULL, 335 int32_t notificationFrames = 0, 336 const sp<IMemory>& sharedBuffer = 0, 337 bool threadCanCallJava = false, 338 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 339 transfer_type transferType = TRANSFER_DEFAULT, 340 const audio_offload_info_t *offloadInfo = NULL, 341 uid_t uid = AUDIO_UID_INVALID, 342 pid_t pid = -1, 343 const audio_attributes_t* pAttributes = NULL, 344 bool doNotReconnect = false, 345 float maxRequiredSpeed = 1.0f, 346 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 347 348 /* Result of constructing the AudioTrack. This must be checked for successful initialization 349 * before using any AudioTrack API (except for set()), because using 350 * an uninitialized AudioTrack produces undefined results. 351 * See set() method above for possible return codes. 352 */ initCheck()353 status_t initCheck() const { return mStatus; } 354 355 /* Returns this track's estimated latency in milliseconds. 356 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 357 * and audio hardware driver. 358 */ 359 uint32_t latency(); 360 361 /* Returns the number of application-level buffer underruns 362 * since the AudioTrack was created. 363 */ 364 uint32_t getUnderrunCount() const; 365 366 /* getters, see constructors and set() */ 367 368 audio_stream_type_t streamType() const; format()369 audio_format_t format() const { return mFormat; } 370 371 /* Return frame size in bytes, which for linear PCM is 372 * channelCount * (bit depth per channel / 8). 373 * channelCount is determined from channelMask, and bit depth comes from format. 374 * For non-linear formats, the frame size is typically 1 byte. 375 */ frameSize()376 size_t frameSize() const { return mFrameSize; } 377 channelCount()378 uint32_t channelCount() const { return mChannelCount; } frameCount()379 size_t frameCount() const { return mFrameCount; } 380 381 /* 382 * Return the period of the notification callback in frames. 383 * This value is set when the AudioTrack is constructed. 384 * It can be modified if the AudioTrack is rerouted. 385 */ getNotificationPeriodInFrames()386 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 387 388 /* Return effective size of audio buffer that an application writes to 389 * or a negative error if the track is uninitialized. 390 */ 391 ssize_t getBufferSizeInFrames(); 392 393 /* Returns the buffer duration in microseconds at current playback rate. 394 */ 395 status_t getBufferDurationInUs(int64_t *duration); 396 397 /* Set the effective size of audio buffer that an application writes to. 398 * This is used to determine the amount of available room in the buffer, 399 * which determines when a write will block. 400 * This allows an application to raise and lower the audio latency. 401 * The requested size may be adjusted so that it is 402 * greater or equal to the absolute minimum and 403 * less than or equal to the getBufferCapacityInFrames(). 404 * It may also be adjusted slightly for internal reasons. 405 * 406 * Return the final size or a negative error if the track is unitialized 407 * or does not support variable sizes. 408 */ 409 ssize_t setBufferSizeInFrames(size_t size); 410 411 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ sharedBuffer()412 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 413 414 /* 415 * return metrics information for the current track. 416 */ 417 status_t getMetrics(mediametrics::Item * &item); 418 419 /* 420 * Set name of API that is using this object. 421 * For example "aaudio" or "opensles". 422 * This may be logged or reported as part of MediaMetrics. 423 */ setCallerName(const std::string & name)424 void setCallerName(const std::string &name) { 425 mCallerName = name; 426 } 427 getCallerName()428 std::string getCallerName() const { 429 return mCallerName; 430 }; 431 432 /* After it's created the track is not active. Call start() to 433 * make it active. If set, the callback will start being called. 434 * If the track was previously paused, volume is ramped up over the first mix buffer. 435 */ 436 status_t start(); 437 438 /* Stop a track. 439 * In static buffer mode, the track is stopped immediately. 440 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 441 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 442 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 443 * is first drained, mixed, and output, and only then is the track marked as stopped. 444 */ 445 void stop(); 446 bool stopped() const; 447 448 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 449 * This has the effect of draining the buffers without mixing or output. 450 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 451 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 452 */ 453 void flush(); 454 455 /* Pause a track. After pause, the callback will cease being called and 456 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 457 * and will fill up buffers until the pool is exhausted. 458 * Volume is ramped down over the next mix buffer following the pause request, 459 * and then the track is marked as paused. It can be resumed with ramp up by start(). 460 */ 461 void pause(); 462 463 /* Set volume for this track, mostly used for games' sound effects 464 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 465 * This is the older API. New applications should use setVolume(float) when possible. 466 */ 467 status_t setVolume(float left, float right); 468 469 /* Set volume for all channels. This is the preferred API for new applications, 470 * especially for multi-channel content. 471 */ 472 status_t setVolume(float volume); 473 474 /* Set the send level for this track. An auxiliary effect should be attached 475 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 476 */ 477 status_t setAuxEffectSendLevel(float level); 478 void getAuxEffectSendLevel(float* level) const; 479 480 /* Set source sample rate for this track in Hz, mostly used for games' sound effects. 481 * Zero is not permitted. 482 */ 483 status_t setSampleRate(uint32_t sampleRate); 484 485 /* Return current source sample rate in Hz. 486 * If specified as zero in constructor or set(), this will be the sink sample rate. 487 */ 488 uint32_t getSampleRate() const; 489 490 /* Return the original source sample rate in Hz. This corresponds to the sample rate 491 * if playback rate had normal speed and pitch. 492 */ 493 uint32_t getOriginalSampleRate() const; 494 495 /* Set source playback rate for timestretch 496 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 497 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 498 * 499 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 500 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 501 * 502 * Speed increases the playback rate of media, but does not alter pitch. 503 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 504 */ 505 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 506 507 /* Return current playback rate */ 508 const AudioPlaybackRate& getPlaybackRate() const; 509 510 /* Enables looping and sets the start and end points of looping. 511 * Only supported for static buffer mode. 512 * 513 * Parameters: 514 * 515 * loopStart: loop start in frames relative to start of buffer. 516 * loopEnd: loop end in frames relative to start of buffer. 517 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 518 * pending or active loop. loopCount == -1 means infinite looping. 519 * 520 * For proper operation the following condition must be respected: 521 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 522 * 523 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 524 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 525 * 526 */ 527 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 528 529 /* Sets marker position. When playback reaches the number of frames specified, a callback with 530 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 531 * notification callback. To set a marker at a position which would compute as 0, 532 * a workaround is to set the marker at a nearby position such as ~0 or 1. 533 * If the AudioTrack has been opened with no callback function associated, the operation will 534 * fail. 535 * 536 * Parameters: 537 * 538 * marker: marker position expressed in wrapping (overflow) frame units, 539 * like the return value of getPosition(). 540 * 541 * Returned status (from utils/Errors.h) can be: 542 * - NO_ERROR: successful operation 543 * - INVALID_OPERATION: the AudioTrack has no callback installed. 544 */ 545 status_t setMarkerPosition(uint32_t marker); 546 status_t getMarkerPosition(uint32_t *marker) const; 547 548 /* Sets position update period. Every time the number of frames specified has been played, 549 * a callback with event type EVENT_NEW_POS is called. 550 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 551 * callback. 552 * If the AudioTrack has been opened with no callback function associated, the operation will 553 * fail. 554 * Extremely small values may be rounded up to a value the implementation can support. 555 * 556 * Parameters: 557 * 558 * updatePeriod: position update notification period expressed in frames. 559 * 560 * Returned status (from utils/Errors.h) can be: 561 * - NO_ERROR: successful operation 562 * - INVALID_OPERATION: the AudioTrack has no callback installed. 563 */ 564 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 565 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 566 567 /* Sets playback head position. 568 * Only supported for static buffer mode. 569 * 570 * Parameters: 571 * 572 * position: New playback head position in frames relative to start of buffer. 573 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 574 * but will result in an immediate underrun if started. 575 * 576 * Returned status (from utils/Errors.h) can be: 577 * - NO_ERROR: successful operation 578 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 579 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 580 * buffer 581 */ 582 status_t setPosition(uint32_t position); 583 584 /* Return the total number of frames played since playback start. 585 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 586 * It is reset to zero by flush(), reload(), and stop(). 587 * 588 * Parameters: 589 * 590 * position: Address where to return play head position. 591 * 592 * Returned status (from utils/Errors.h) can be: 593 * - NO_ERROR: successful operation 594 * - BAD_VALUE: position is NULL 595 */ 596 status_t getPosition(uint32_t *position); 597 598 /* For static buffer mode only, this returns the current playback position in frames 599 * relative to start of buffer. It is analogous to the position units used by 600 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 601 */ 602 status_t getBufferPosition(uint32_t *position); 603 604 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 605 * rewriting the buffer before restarting playback after a stop. 606 * This method must be called with the AudioTrack in paused or stopped state. 607 * Not allowed in streaming mode. 608 * 609 * Returned status (from utils/Errors.h) can be: 610 * - NO_ERROR: successful operation 611 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 612 */ 613 status_t reload(); 614 615 /** 616 * @param transferType 617 * @return text string that matches the enum name 618 */ 619 static const char * convertTransferToText(transfer_type transferType); 620 621 /* Returns a handle on the audio output used by this AudioTrack. 622 * 623 * Parameters: 624 * none. 625 * 626 * Returned value: 627 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 628 * track needed to be re-created but that failed 629 */ 630 private: 631 audio_io_handle_t getOutput() const; 632 public: 633 634 /* Selects the audio device to use for output of this AudioTrack. A value of 635 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 636 * 637 * Parameters: 638 * The device ID of the selected device (as returned by the AudioDevicesManager API). 639 * 640 * Returned value: 641 * - NO_ERROR: successful operation 642 * TODO: what else can happen here? 643 */ 644 status_t setOutputDevice(audio_port_handle_t deviceId); 645 646 /* Returns the ID of the audio device selected for this AudioTrack. 647 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 648 * 649 * Parameters: 650 * none. 651 */ 652 audio_port_handle_t getOutputDevice(); 653 654 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 655 * attached. 656 * When the AudioTrack is inactive, the device ID returned can be either: 657 * - AUDIO_PORT_HANDLE_NONE if the AudioTrack is not attached to any output. 658 * - The device ID used before paused or stopped. 659 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioTrack 660 * has not been started yet. 661 * 662 * Parameters: 663 * none. 664 */ 665 audio_port_handle_t getRoutedDeviceId(); 666 667 /* Returns the unique session ID associated with this track. 668 * 669 * Parameters: 670 * none. 671 * 672 * Returned value: 673 * AudioTrack session ID. 674 */ getSessionId()675 audio_session_t getSessionId() const { return mSessionId; } 676 677 /* Attach track auxiliary output to specified effect. Use effectId = 0 678 * to detach track from effect. 679 * 680 * Parameters: 681 * 682 * effectId: effectId obtained from AudioEffect::id(). 683 * 684 * Returned status (from utils/Errors.h) can be: 685 * - NO_ERROR: successful operation 686 * - INVALID_OPERATION: the effect is not an auxiliary effect. 687 * - BAD_VALUE: The specified effect ID is invalid 688 */ 689 status_t attachAuxEffect(int effectId); 690 691 /* Public API for TRANSFER_OBTAIN mode. 692 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 693 * After filling these slots with data, the caller should release them with releaseBuffer(). 694 * If the track buffer is not full, obtainBuffer() returns as many contiguous 695 * [empty slots for] frames as are available immediately. 696 * 697 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 698 * additional non-contiguous frames that are predicted to be available immediately, 699 * if the client were to release the first frames and then call obtainBuffer() again. 700 * This value is only a prediction, and needs to be confirmed. 701 * It will be set to zero for an error return. 702 * 703 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 704 * regardless of the value of waitCount. 705 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 706 * maximum timeout based on waitCount; see chart below. 707 * Buffers will be returned until the pool 708 * is exhausted, at which point obtainBuffer() will either block 709 * or return WOULD_BLOCK depending on the value of the "waitCount" 710 * parameter. 711 * 712 * Interpretation of waitCount: 713 * +n limits wait time to n * WAIT_PERIOD_MS, 714 * -1 causes an (almost) infinite wait time, 715 * 0 non-blocking. 716 * 717 * Buffer fields 718 * On entry: 719 * frameCount number of [empty slots for] frames requested 720 * size ignored 721 * raw ignored 722 * sequence ignored 723 * After error return: 724 * frameCount 0 725 * size 0 726 * raw undefined 727 * sequence undefined 728 * After successful return: 729 * frameCount actual number of [empty slots for] frames available, <= number requested 730 * size actual number of bytes available 731 * raw pointer to the buffer 732 * sequence IAudioTrack instance sequence number, as of obtainBuffer() 733 */ 734 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 735 size_t *nonContig = NULL); 736 737 private: 738 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 739 * additional non-contiguous frames that are predicted to be available immediately, 740 * if the client were to release the first frames and then call obtainBuffer() again. 741 * This value is only a prediction, and needs to be confirmed. 742 * It will be set to zero for an error return. 743 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 744 * in case the requested amount of frames is in two or more non-contiguous regions. 745 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 746 */ 747 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 748 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 749 public: 750 751 /* Public API for TRANSFER_OBTAIN mode. 752 * Release a filled buffer of frames for AudioFlinger to process. 753 * 754 * Buffer fields: 755 * frameCount currently ignored but recommend to set to actual number of frames filled 756 * size actual number of bytes filled, must be multiple of frameSize 757 * raw ignored 758 */ 759 void releaseBuffer(const Buffer* audioBuffer); 760 761 /* As a convenience we provide a write() interface to the audio buffer. 762 * Input parameter 'size' is in byte units. 763 * This is implemented on top of obtainBuffer/releaseBuffer. For best 764 * performance use callbacks. Returns actual number of bytes written >= 0, 765 * or one of the following negative status codes: 766 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 767 * BAD_VALUE size is invalid 768 * WOULD_BLOCK when obtainBuffer() returns same, or 769 * AudioTrack was stopped during the write 770 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 771 * the track cannot be automatically restored. 772 * The application needs to recreate the AudioTrack 773 * because the audio device changed or AudioFlinger died. 774 * This typically occurs for direct or offload tracks 775 * or if mDoNotReconnect is true. 776 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 777 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 778 * false for the method to return immediately without waiting to try multiple times to write 779 * the full content of the buffer. 780 */ 781 ssize_t write(const void* buffer, size_t size, bool blocking = true); 782 783 /* 784 * Dumps the state of an audio track. 785 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 786 */ 787 status_t dump(int fd, const Vector<String16>& args) const; 788 789 /* 790 * Return the total number of frames which AudioFlinger desired but were unavailable, 791 * and thus which resulted in an underrun. Reset to zero by stop(). 792 */ 793 uint32_t getUnderrunFrames() const; 794 795 /* Get the flags */ getFlags()796 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 797 798 /* Set parameters - only possible when using direct output */ 799 status_t setParameters(const String8& keyValuePairs); 800 801 /* Sets the volume shaper object */ 802 media::VolumeShaper::Status applyVolumeShaper( 803 const sp<media::VolumeShaper::Configuration>& configuration, 804 const sp<media::VolumeShaper::Operation>& operation); 805 806 /* Gets the volume shaper state */ 807 sp<media::VolumeShaper::State> getVolumeShaperState(int id); 808 809 /* Selects the presentation (if available) */ 810 status_t selectPresentation(int presentationId, int programId); 811 812 /* Get parameters */ 813 String8 getParameters(const String8& keys); 814 815 /* Poll for a timestamp on demand. 816 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 817 * or if you need to get the most recent timestamp outside of the event callback handler. 818 * Caution: calling this method too often may be inefficient; 819 * if you need a high resolution mapping between frame position and presentation time, 820 * consider implementing that at application level, based on the low resolution timestamps. 821 * Returns NO_ERROR if timestamp is valid. 822 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 823 * start/ACTIVE, when the number of frames consumed is less than the 824 * overall hardware latency to physical output. In WOULD_BLOCK cases, 825 * one might poll again, or use getPosition(), or use 0 position and 826 * current time for the timestamp. 827 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 828 * the track cannot be automatically restored. 829 * The application needs to recreate the AudioTrack 830 * because the audio device changed or AudioFlinger died. 831 * This typically occurs for direct or offload tracks 832 * or if mDoNotReconnect is true. 833 * INVALID_OPERATION wrong state, or some other error. 834 * 835 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 836 */ 837 status_t getTimestamp(AudioTimestamp& timestamp); 838 private: 839 status_t getTimestamp_l(AudioTimestamp& timestamp); 840 public: 841 842 /* Return the extended timestamp, with additional timebase info and improved drain behavior. 843 * 844 * This is similar to the AudioTrack.java API: 845 * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) 846 * 847 * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method 848 * 849 * 1. stop() by itself does not reset the frame position. 850 * A following start() resets the frame position to 0. 851 * 2. flush() by itself does not reset the frame position. 852 * The frame position advances by the number of frames flushed, 853 * when the first frame after flush reaches the audio sink. 854 * 3. BOOTTIME clock offsets are provided to help synchronize with 855 * non-audio streams, e.g. sensor data. 856 * 4. Position is returned with 64 bits of resolution. 857 * 858 * Parameters: 859 * timestamp: A pointer to the caller allocated ExtendedTimestamp. 860 * 861 * Returns NO_ERROR on success; timestamp is filled with valid data. 862 * BAD_VALUE if timestamp is NULL. 863 * WOULD_BLOCK if called immediately after start() when the number 864 * of frames consumed is less than the 865 * overall hardware latency to physical output. In WOULD_BLOCK cases, 866 * one might poll again, or use getPosition(), or use 0 position and 867 * current time for the timestamp. 868 * If WOULD_BLOCK is returned, the timestamp is still 869 * modified with the LOCATION_CLIENT portion filled. 870 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 871 * the track cannot be automatically restored. 872 * The application needs to recreate the AudioTrack 873 * because the audio device changed or AudioFlinger died. 874 * This typically occurs for direct or offloaded tracks 875 * or if mDoNotReconnect is true. 876 * INVALID_OPERATION if called on a offloaded or direct track. 877 * Use getTimestamp(AudioTimestamp& timestamp) instead. 878 */ 879 status_t getTimestamp(ExtendedTimestamp *timestamp); 880 private: 881 status_t getTimestamp_l(ExtendedTimestamp *timestamp); 882 public: 883 884 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 885 * AudioTrack is routed is updated. 886 * Replaces any previously installed callback. 887 * Parameters: 888 * callback: The callback interface 889 * Returns NO_ERROR if successful. 890 * INVALID_OPERATION if the same callback is already installed. 891 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 892 * BAD_VALUE if the callback is NULL 893 */ 894 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 895 896 /* remove an AudioDeviceCallback. 897 * Parameters: 898 * callback: The callback interface 899 * Returns NO_ERROR if successful. 900 * INVALID_OPERATION if the callback is not installed 901 * BAD_VALUE if the callback is NULL 902 */ 903 status_t removeAudioDeviceCallback( 904 const sp<AudioSystem::AudioDeviceCallback>& callback); 905 906 // AudioSystem::AudioDeviceCallback> virtuals 907 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 908 audio_port_handle_t deviceId); 909 910 /* Obtain the pending duration in milliseconds for playback of pure PCM 911 * (mixable without embedded timing) data remaining in AudioTrack. 912 * 913 * This is used to estimate the drain time for the client-server buffer 914 * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. 915 * One may optionally request to find the duration to play through the HAL 916 * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, 917 * INVALID_OPERATION may be returned if the kernel location is unavailable. 918 * 919 * Returns NO_ERROR if successful. 920 * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained 921 * or the AudioTrack does not contain pure PCM data. 922 * BAD_VALUE if msec is nullptr or location is invalid. 923 */ 924 status_t pendingDuration(int32_t *msec, 925 ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); 926 927 /* hasStarted() is used to determine if audio is now audible at the device after 928 * a start() command. The underlying implementation checks a nonzero timestamp position 929 * or increment for the audible assumption. 930 * 931 * hasStarted() returns true if the track has been started() and audio is audible 932 * and no subsequent pause() or flush() has been called. Immediately after pause() or 933 * flush() hasStarted() will return false. 934 * 935 * If stop() has been called, hasStarted() will return true if audio is still being 936 * delivered or has finished delivery (even if no audio was written) for both offloaded 937 * and normal tracks. This property removes a race condition in checking hasStarted() 938 * for very short clips, where stop() must be called to finish drain. 939 * 940 * In all cases, hasStarted() may turn false briefly after a subsequent start() is called 941 * until audio becomes audible again. 942 */ 943 bool hasStarted(); // not const 944 isPlaying()945 bool isPlaying() { 946 AutoMutex lock(mLock); 947 return mState == STATE_ACTIVE || mState == STATE_STOPPING; 948 } 949 950 /* Get the unique port ID assigned to this AudioTrack instance by audio policy manager. 951 * The ID is unique across all audioserver clients and can change during the life cycle 952 * of a given AudioTrack instance if the connection to audioserver is restored. 953 */ getPortId()954 audio_port_handle_t getPortId() const { return mPortId; }; 955 setAudioTrackCallback(const sp<media::IAudioTrackCallback> & callback)956 void setAudioTrackCallback(const sp<media::IAudioTrackCallback>& callback) { 957 mAudioTrackCallback->setAudioTrackCallback(callback); 958 } 959 960 protected: 961 /* copying audio tracks is not allowed */ 962 AudioTrack(const AudioTrack& other); 963 AudioTrack& operator = (const AudioTrack& other); 964 965 /* a small internal class to handle the callback */ 966 class AudioTrackThread : public Thread 967 { 968 public: 969 explicit AudioTrackThread(AudioTrack& receiver); 970 971 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 972 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 973 virtual void requestExit(); 974 975 void pause(); // suspend thread from execution at next loop boundary 976 void resume(); // allow thread to execute, if not requested to exit 977 void wake(); // wake to handle changed notification conditions. 978 979 private: 980 void pauseInternal(nsecs_t ns = 0LL); 981 // like pause(), but only used internally within thread 982 983 friend class AudioTrack; 984 virtual bool threadLoop(); 985 AudioTrack& mReceiver; 986 virtual ~AudioTrackThread(); 987 Mutex mMyLock; // Thread::mLock is private 988 Condition mMyCond; // Thread::mThreadExitedCondition is private 989 bool mPaused; // whether thread is requested to pause at next loop entry 990 bool mPausedInt; // whether thread internally requests pause 991 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 992 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 993 // to processAudioBuffer() as state may have changed 994 // since pause time calculated. 995 }; 996 997 // body of AudioTrackThread::threadLoop() 998 // returns the maximum amount of time before we would like to run again, where: 999 // 0 immediately 1000 // > 0 no later than this many nanoseconds from now 1001 // NS_WHENEVER still active but no particular deadline 1002 // NS_INACTIVE inactive so don't run again until re-started 1003 // NS_NEVER never again 1004 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 1005 nsecs_t processAudioBuffer(); 1006 1007 // caller must hold lock on mLock for all _l methods 1008 1009 void updateLatency_l(); // updates mAfLatency and mLatency from AudioSystem cache 1010 1011 status_t createTrack_l(); 1012 1013 // can only be called when mState != STATE_ACTIVE 1014 void flush_l(); 1015 1016 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 1017 1018 // FIXME enum is faster than strcmp() for parameter 'from' 1019 status_t restoreTrack_l(const char *from); 1020 1021 uint32_t getUnderrunCount_l() const; 1022 1023 bool isOffloaded() const; 1024 bool isDirect() const; 1025 bool isOffloadedOrDirect() const; 1026 isOffloaded_l()1027 bool isOffloaded_l() const 1028 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 1029 isOffloadedOrDirect_l()1030 bool isOffloadedOrDirect_l() const 1031 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 1032 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 1033 isDirect_l()1034 bool isDirect_l() const 1035 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 1036 1037 // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) isPurePcmData_l()1038 bool isPurePcmData_l() const 1039 { return audio_is_linear_pcm(mFormat) 1040 && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } 1041 1042 // increment mPosition by the delta of mServer, and return new value of mPosition 1043 Modulo<uint32_t> updateAndGetPosition_l(); 1044 1045 // check sample rate and speed is compatible with AudioTrack 1046 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed); 1047 1048 void restartIfDisabled(); 1049 1050 void updateRoutedDeviceId_l(); 1051 1052 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 1053 sp<IAudioTrack> mAudioTrack; 1054 sp<IMemory> mCblkMemory; 1055 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 1056 audio_io_handle_t mOutput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getOutputForAttr() 1057 1058 sp<AudioTrackThread> mAudioTrackThread; 1059 bool mThreadCanCallJava; 1060 1061 float mVolume[2]; 1062 float mSendLevel; 1063 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 1064 uint32_t mOriginalSampleRate; 1065 AudioPlaybackRate mPlaybackRate; 1066 float mMaxRequiredSpeed; // use PCM buffer size to allow this speed 1067 1068 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 1069 // This allocated buffer size is maintained by the proxy. 1070 size_t mFrameCount; // maximum size of buffer 1071 1072 size_t mReqFrameCount; // frame count to request the first or next time 1073 // a new IAudioTrack is needed, non-decreasing 1074 1075 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 1076 // These values can be used for informational purposes until the track is invalidated, 1077 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 1078 uint32_t mAfLatency; // AudioFlinger latency in ms 1079 size_t mAfFrameCount; // AudioFlinger frame count 1080 uint32_t mAfSampleRate; // AudioFlinger sample rate 1081 1082 // constant after constructor or set() 1083 audio_format_t mFormat; // as requested by client, not forced to 16-bit 1084 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 1085 // this AudioTrack has valid attributes 1086 uint32_t mChannelCount; 1087 audio_channel_mask_t mChannelMask; 1088 sp<IMemory> mSharedBuffer; 1089 transfer_type mTransfer; 1090 audio_offload_info_t mOffloadInfoCopy; 1091 const audio_offload_info_t* mOffloadInfo; 1092 audio_attributes_t mAttributes; 1093 1094 size_t mFrameSize; // frame size in bytes 1095 1096 status_t mStatus; 1097 1098 // can change dynamically when IAudioTrack invalidated 1099 uint32_t mLatency; // in ms 1100 1101 // Indicates the current track state. Protected by mLock. 1102 enum State { 1103 STATE_ACTIVE, 1104 STATE_STOPPED, 1105 STATE_PAUSED, 1106 STATE_PAUSED_STOPPING, 1107 STATE_FLUSHED, 1108 STATE_STOPPING, 1109 } mState; 1110 stateToString(State state)1111 static constexpr const char *stateToString(State state) 1112 { 1113 switch (state) { 1114 case STATE_ACTIVE: return "STATE_ACTIVE"; 1115 case STATE_STOPPED: return "STATE_STOPPED"; 1116 case STATE_PAUSED: return "STATE_PAUSED"; 1117 case STATE_PAUSED_STOPPING: return "STATE_PAUSED_STOPPING"; 1118 case STATE_FLUSHED: return "STATE_FLUSHED"; 1119 case STATE_STOPPING: return "STATE_STOPPING"; 1120 default: return "UNKNOWN"; 1121 } 1122 } 1123 1124 // for client callback handler 1125 callback_t mCbf; // callback handler for events, or NULL 1126 void* mUserData; 1127 1128 // for notification APIs 1129 1130 // next 2 fields are const after constructor or set() 1131 uint32_t mNotificationFramesReq; // requested number of frames between each 1132 // notification callback, 1133 // at initial source sample rate 1134 uint32_t mNotificationsPerBufferReq; 1135 // requested number of notifications per buffer, 1136 // currently only used for fast tracks with 1137 // default track buffer size 1138 1139 uint32_t mNotificationFramesAct; // actual number of frames between each 1140 // notification callback, 1141 // at initial source sample rate 1142 bool mRefreshRemaining; // processAudioBuffer() should refresh 1143 // mRemainingFrames and mRetryOnPartialBuffer 1144 1145 // used for static track cbf and restoration 1146 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 1147 uint32_t mLoopStart; // last setLoop loopStart 1148 uint32_t mLoopEnd; // last setLoop loopEnd 1149 int32_t mLoopCountNotified; // the last loopCount notified by callback. 1150 // mLoopCountNotified counts down, matching 1151 // the remaining loop count for static track 1152 // playback. 1153 1154 // These are private to processAudioBuffer(), and are not protected by a lock 1155 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 1156 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 1157 uint32_t mObservedSequence; // last observed value of mSequence 1158 1159 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 1160 bool mMarkerReached; 1161 Modulo<uint32_t> mNewPosition; // in frames 1162 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 1163 1164 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 1165 // which is count of frames consumed by server, 1166 // reset by new IAudioTrack, 1167 // whether it is reset by stop() is TBD 1168 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 1169 // monotonically after new IAudioTrack, 1170 // and could be easily widened to uint64_t 1171 Modulo<uint32_t> mReleased; // count of frames released to server 1172 // but not necessarily consumed by server, 1173 // reset by stop() but continues monotonically 1174 // after new IAudioTrack to restore mPosition, 1175 // and could be easily widened to uint64_t 1176 int64_t mStartFromZeroUs; // the start time after flush or stop, 1177 // when position should be 0. 1178 // only used for offloaded and direct tracks. 1179 int64_t mStartNs; // the time when start() is called. 1180 ExtendedTimestamp mStartEts; // Extended timestamp at start for normal 1181 // AudioTracks. 1182 AudioTimestamp mStartTs; // Timestamp at start for offloaded or direct 1183 // AudioTracks. 1184 1185 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 1186 bool mTimestampStartupGlitchReported; // reduce log spam 1187 bool mTimestampRetrogradePositionReported; // reduce log spam 1188 bool mTimestampRetrogradeTimeReported; // reduce log spam 1189 bool mTimestampStallReported; // reduce log spam 1190 bool mTimestampStaleTimeReported; // reduce log spam 1191 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 1192 ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp 1193 1194 uint32_t mUnderrunCountOffset; // updated when restoring tracks 1195 1196 int64_t mFramesWritten; // total frames written. reset to zero after 1197 // the start() following stop(). It is not 1198 // changed after restoring the track or 1199 // after flush. 1200 int64_t mFramesWrittenServerOffset; // An offset to server frames due to 1201 // restoring AudioTrack, or stop/start. 1202 // This offset is also used for static tracks. 1203 int64_t mFramesWrittenAtRestore; // Frames written at restore point (or frames 1204 // delivered for static tracks). 1205 // -1 indicates no previous restore point. 1206 1207 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 1208 // be denied by client or server, such as 1209 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 1210 // held to read or write those bits reliably. 1211 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 1212 1213 bool mDoNotReconnect; 1214 1215 audio_session_t mSessionId; 1216 int mAuxEffectId; 1217 audio_port_handle_t mPortId; // Id from Audio Policy Manager 1218 1219 mutable Mutex mLock; 1220 1221 int mPreviousPriority; // before start() 1222 SchedPolicy mPreviousSchedulingGroup; 1223 bool mAwaitBoost; // thread should wait for priority boost before running 1224 1225 // The proxy should only be referenced while a lock is held because the proxy isn't 1226 // multi-thread safe, especially the SingleStateQueue part of the proxy. 1227 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 1228 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 1229 // them around in case they are replaced during the obtainBuffer(). 1230 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 1231 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 1232 1233 bool mInUnderrun; // whether track is currently in underrun state 1234 uint32_t mPausedPosition; 1235 1236 // For Device Selection API 1237 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 1238 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 1239 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 1240 // May not match the app selection depending on other 1241 // activity and connected devices. 1242 1243 sp<media::VolumeHandler> mVolumeHandler; 1244 1245 const std::string mOpPackageName; 1246 1247 private: 1248 class DeathNotifier : public IBinder::DeathRecipient { 1249 public: DeathNotifier(AudioTrack * audioTrack)1250 explicit DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 1251 protected: 1252 virtual void binderDied(const wp<IBinder>& who); 1253 private: 1254 const wp<AudioTrack> mAudioTrack; 1255 }; 1256 1257 sp<DeathNotifier> mDeathNotifier; 1258 uint32_t mSequence; // incremented for each new IAudioTrack attempt 1259 uid_t mClientUid; 1260 pid_t mClientPid; 1261 1262 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1263 1264 private: 1265 class MediaMetrics { 1266 public: MediaMetrics()1267 MediaMetrics() : mMetricsItem(mediametrics::Item::create("audiotrack")) { 1268 } ~MediaMetrics()1269 ~MediaMetrics() { 1270 // mMetricsItem alloc failure will be flagged in the constructor 1271 // don't log empty records 1272 if (mMetricsItem->count() > 0) { 1273 mMetricsItem->selfrecord(); 1274 } 1275 } 1276 void gather(const AudioTrack *track); dup()1277 mediametrics::Item *dup() { return mMetricsItem->dup(); } 1278 private: 1279 std::unique_ptr<mediametrics::Item> mMetricsItem; 1280 }; 1281 MediaMetrics mMediaMetrics; 1282 std::string mMetricsId; // GUARDED_BY(mLock), could change in createTrack_l(). 1283 std::string mCallerName; // for example "aaudio" 1284 1285 private: 1286 class AudioTrackCallback : public media::BnAudioTrackCallback { 1287 public: 1288 binder::Status onCodecFormatChanged(const std::vector<uint8_t>& audioMetadata) override; 1289 1290 void setAudioTrackCallback(const sp<media::IAudioTrackCallback>& callback); 1291 private: 1292 Mutex mAudioTrackCbLock; 1293 wp<media::IAudioTrackCallback> mCallback; 1294 }; 1295 sp<AudioTrackCallback> mAudioTrackCallback; 1296 }; 1297 1298 }; // namespace android 1299 1300 #endif // ANDROID_AUDIOTRACK_H 1301