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 <binder/IMemory.h> 21 #include <cutils/sched_policy.h> 22 #include <media/AudioSystem.h> 23 #include <media/AudioTimestamp.h> 24 #include <media/AudioResamplerPublic.h> 25 #include <media/MediaMetricsItem.h> 26 #include <media/Modulo.h> 27 #include <media/VolumeShaper.h> 28 #include <utils/threads.h> 29 #include <android/content/AttributionSourceState.h> 30 31 #include <chrono> 32 #include <string> 33 34 #include "android/media/BnAudioTrackCallback.h" 35 #include "android/media/IAudioTrack.h" 36 #include "android/media/IAudioTrackCallback.h" 37 38 namespace android { 39 40 using content::AttributionSourceState; 41 42 // ---------------------------------------------------------------------------- 43 44 struct audio_track_cblk_t; 45 class AudioTrackClientProxy; 46 class StaticAudioTrackClientProxy; 47 48 // ---------------------------------------------------------------------------- 49 50 class AudioTrack : public AudioSystem::AudioDeviceCallback 51 { 52 public: 53 54 /* Events used by AudioTrack callback function (callback_t). 55 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 56 */ 57 enum event_type { 58 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 59 // This event only occurs for TRANSFER_CALLBACK. 60 // If this event is delivered but the callback handler 61 // does not want to write more data, the handler must 62 // ignore the event by setting frameCount to zero. 63 // This might occur, for example, if the application is 64 // waiting for source data or is at the end of stream. 65 // 66 // For data filling, it is preferred that the callback 67 // does not block and instead returns a short count on 68 // the amount of data actually delivered 69 // (or 0, if no data is currently available). 70 EVENT_UNDERRUN = 1, // Buffer underrun occurred. This will not occur for 71 // static tracks. 72 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 73 // loop start if loop count was not 0 for a static track. 74 EVENT_MARKER = 3, // Playback head is at the specified marker position 75 // (See setMarkerPosition()). 76 EVENT_NEW_POS = 4, // Playback head is at a new position 77 // (See setPositionUpdatePeriod()). 78 EVENT_BUFFER_END = 5, // Playback has completed for a static track. 79 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 80 // voluntary invalidation by mediaserver, or mediaserver crash. 81 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 82 // back (after stop is called) for an offloaded track. 83 #if 0 // FIXME not yet implemented 84 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 85 // in the mapping from frame position to presentation time. 86 // See AudioTimestamp for the information included with event. 87 #endif 88 EVENT_CAN_WRITE_MORE_DATA = 9,// Notification that more data can be given by write() 89 // This event only occurs for TRANSFER_SYNC_NOTIF_CALLBACK. 90 }; 91 92 /* Client should declare a Buffer and pass the address to obtainBuffer() 93 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 94 */ 95 96 class Buffer 97 { 98 friend AudioTrack; 99 public: size()100 size_t size() const { return mSize; } getFrameCount()101 size_t getFrameCount() const { return frameCount; } data()102 uint8_t * data() const { return ui8; } 103 // Leaving public for now to ease refactoring. This class will be 104 // replaced 105 size_t frameCount; // number of sample frames corresponding to size; 106 // on input to obtainBuffer() it is the number of frames desired, 107 // on output from obtainBuffer() it is the number of available 108 // [empty slots for] frames to be filled 109 // on input to releaseBuffer() it is currently ignored 110 private: 111 size_t mSize; // input/output in bytes == frameCount * frameSize 112 // on input to obtainBuffer() it is ignored 113 // on output from obtainBuffer() it is the number of available 114 // [empty slots for] bytes to be filled, 115 // which is frameCount * frameSize 116 // on input to releaseBuffer() it is the number of bytes to 117 // release 118 119 union { 120 void* raw; 121 int16_t* i16; // signed 16-bit 122 uint8_t* ui8; // unsigned 8-bit, offset by 0x80 123 }; // input to obtainBuffer(): unused, output: pointer to buffer 124 125 uint32_t sequence; // IAudioTrack instance sequence number, as of obtainBuffer(). 126 // It is set by obtainBuffer() and confirmed by releaseBuffer(). 127 // Not "user-serviceable". 128 }; 129 130 /* As a convenience, if a callback is supplied, a handler thread 131 * is automatically created with the appropriate priority. This thread 132 * invokes the callback when a new buffer becomes available or various conditions occur. 133 * Parameters: 134 * 135 * event: type of event notified (see enum AudioTrack::event_type). 136 * user: Pointer to context for use by the callback receiver. 137 * info: Pointer to optional parameter according to event type: 138 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 139 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 140 * written. 141 * - EVENT_UNDERRUN: unused. 142 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 143 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 144 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 145 * - EVENT_BUFFER_END: unused. 146 * - EVENT_NEW_IAUDIOTRACK: unused. 147 * - EVENT_STREAM_END: unused. 148 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 149 */ 150 151 typedef void (*legacy_callback_t)(int event, void* user, void* info); 152 class IAudioTrackCallback : public virtual RefBase { 153 friend AudioTrack; 154 protected: 155 /* Request to write more data to buffer. 156 * This event only occurs for TRANSFER_CALLBACK. 157 * If this event is delivered but the callback handler does not want to write more data, 158 * the handler must ignore the event by returning zero. 159 * This might occur, for example, if the application is waiting for source data or is at 160 * the end of stream. 161 * For data filling, it is preferred that the callback does not block and instead returns 162 * a short count of the amount of data actually delivered. 163 * Parameters: 164 * - buffer: Buffer to fill 165 * Returns: 166 * Amount of data actually written in bytes. 167 */ onMoreData(const AudioTrack::Buffer & buffer)168 virtual size_t onMoreData([[maybe_unused]] const AudioTrack::Buffer& buffer) { return 0; } 169 170 // Buffer underrun occurred. This will not occur for static tracks. onUnderrun()171 virtual void onUnderrun() {} 172 173 /* Sample loop end was reached; playback restarted from loop start if loop count was not 0 174 * for a static track. 175 * Parameters: 176 * - loopsRemaining: Number of loops remaining to be played. -1 if infinite looping. 177 */ onLoopEnd(int32_t loopsRemaining)178 virtual void onLoopEnd([[maybe_unused]] int32_t loopsRemaining) {} 179 180 /* Playback head is at the specified marker (See setMarkerPosition()). 181 * Parameters: 182 * - onMarker: Marker position in frames 183 */ onMarker(uint32_t markerPosition)184 virtual void onMarker([[maybe_unused]] uint32_t markerPosition) {} 185 186 /* Playback head is at a new position (See setPositionUpdatePeriod()). 187 * Parameters: 188 * - newPos: New position in frames 189 */ onNewPos(uint32_t newPos)190 virtual void onNewPos([[maybe_unused]] uint32_t newPos) {} 191 192 // Playback has completed for a static track. onBufferEnd()193 virtual void onBufferEnd() {} 194 195 // IAudioTrack was re-created, either due to re-routing and voluntary invalidation 196 // by mediaserver, or mediaserver crash. onNewIAudioTrack()197 virtual void onNewIAudioTrack() {} 198 199 // Sent after all the buffers queued in AF and HW are played back (after stop is called) 200 // for an offloaded track. onStreamEnd()201 virtual void onStreamEnd() {} 202 203 /* Delivered periodically and when there's a significant change 204 * in the mapping from frame position to presentation time. 205 * See AudioTimestamp for the information included with event. 206 * TODO not yet implemented. 207 * Parameters: 208 * - timestamp: New frame position and presentation time mapping. 209 */ onNewTimestamp(AudioTimestamp timestamp)210 virtual void onNewTimestamp([[maybe_unused]] AudioTimestamp timestamp) {} 211 212 /* Notification that more data can be given by write() 213 * This event only occurs for TRANSFER_SYNC_NOTIF_CALLBACK. 214 * Similar to onMoreData(), return the number of frames actually written 215 * Parameters: 216 * - buffer: Buffer to fill 217 * Returns: 218 * Amount of data actually written in bytes. 219 */ onCanWriteMoreData(const AudioTrack::Buffer & buffer)220 virtual size_t onCanWriteMoreData([[maybe_unused]] const AudioTrack::Buffer& buffer) { 221 return 0; 222 } 223 }; 224 225 /* Returns the minimum frame count required for the successful creation of 226 * an AudioTrack object. 227 * Returned status (from utils/Errors.h) can be: 228 * - NO_ERROR: successful operation 229 * - NO_INIT: audio server or audio hardware not initialized 230 * - BAD_VALUE: unsupported configuration 231 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 232 * and is undefined otherwise. 233 * FIXME This API assumes a route, and so should be deprecated. 234 */ 235 236 static status_t getMinFrameCount(size_t* frameCount, 237 audio_stream_type_t streamType, 238 uint32_t sampleRate); 239 240 /* Check if direct playback is possible for the given audio configuration and attributes. 241 * Return true if output is possible for the given parameters. Otherwise returns false. 242 */ 243 static bool isDirectOutputSupported(const audio_config_base_t& config, 244 const audio_attributes_t& attributes); 245 246 /* How data is transferred to AudioTrack 247 */ 248 enum transfer_type { 249 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 250 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 251 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 252 TRANSFER_SYNC, // synchronous write() 253 TRANSFER_SHARED, // shared memory 254 TRANSFER_SYNC_NOTIF_CALLBACK, // synchronous write(), notif EVENT_CAN_WRITE_MORE_DATA 255 }; 256 257 /* Constructs an uninitialized AudioTrack. No connection with 258 * AudioFlinger takes place. Use set() after this. 259 */ 260 AudioTrack(); 261 262 AudioTrack(const AttributionSourceState& attributionSourceState); 263 264 /* Creates an AudioTrack object and registers it with AudioFlinger. 265 * Once created, the track needs to be started before it can be used. 266 * Unspecified values are set to appropriate default values. 267 * 268 * Parameters: 269 * 270 * streamType: Select the type of audio stream this track is attached to 271 * (e.g. AUDIO_STREAM_MUSIC). 272 * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate. 273 * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set. 274 * 0 will not work with current policy implementation for direct output 275 * selection where an exact match is needed for sampling rate. 276 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 277 * For direct and offloaded tracks, the possible format(s) depends on the 278 * output sink. 279 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 280 * frameCount: Minimum size of track PCM buffer in frames. This defines the 281 * application's contribution to the 282 * latency of the track. The actual size selected by the AudioTrack could be 283 * larger if the requested size is not compatible with current audio HAL 284 * configuration. Zero means to use a default value. 285 * flags: See comments on audio_output_flags_t in <system/audio.h>. 286 * cbf: Callback function. If not null, this function is called periodically 287 * to provide new data in TRANSFER_CALLBACK mode 288 * and inform of marker, position updates, etc. 289 * user: Context for use by the callback receiver. 290 * notificationFrames: The callback function is called each time notificationFrames PCM 291 * frames have been consumed from track input buffer by server. 292 * Zero means to use a default value, which is typically: 293 * - fast tracks: HAL buffer size, even if track frameCount is larger 294 * - normal tracks: 1/2 of track frameCount 295 * A positive value means that many frames at initial source sample rate. 296 * A negative value for this parameter specifies the negative of the 297 * requested number of notifications (sub-buffers) in the entire buffer. 298 * For fast tracks, the FastMixer will process one sub-buffer at a time. 299 * The size of each sub-buffer is determined by the HAL. 300 * To get "double buffering", for example, one should pass -2. 301 * The minimum number of sub-buffers is 1 (expressed as -1), 302 * and the maximum number of sub-buffers is 8 (expressed as -8). 303 * Negative is only permitted for fast tracks, and if frameCount is zero. 304 * TODO It is ugly to overload a parameter in this way depending on 305 * whether it is positive, negative, or zero. Consider splitting apart. 306 * sessionId: Specific session ID, or zero to use default. 307 * transferType: How data is transferred to AudioTrack. 308 * offloadInfo: If not NULL, provides offload parameters for 309 * AudioSystem::getOutputForAttr(). 310 * attributionSource: The attribution source of the app which initially requested this 311 * AudioTrack. 312 * Includes the UID and PID for power management tracking, or -1 for 313 * current user/process ID, plus the package name. 314 * pAttributes: If not NULL, supersedes streamType for use case selection. 315 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 316 binder to AudioFlinger. 317 It will return an error instead. The application will recreate 318 the track based on offloading or different channel configuration, etc. 319 * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow 320 * maxRequiredSpeed playback. Values less than 1.0f and greater than 321 * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks 322 * and direct or offloaded tracks, this parameter is ignored. 323 * selectedDeviceId: Selected device id of the app which initially requested the AudioTrack 324 * to open with a specific device. 325 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 326 */ 327 328 AudioTrack( audio_stream_type_t streamType, 329 uint32_t sampleRate, 330 audio_format_t format, 331 audio_channel_mask_t channelMask, 332 size_t frameCount = 0, 333 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 334 const wp<IAudioTrackCallback>& callback = nullptr, 335 int32_t notificationFrames = 0, 336 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 337 transfer_type transferType = TRANSFER_DEFAULT, 338 const audio_offload_info_t *offloadInfo = nullptr, 339 const AttributionSourceState& attributionSource = 340 AttributionSourceState(), 341 const audio_attributes_t* pAttributes = nullptr, 342 bool doNotReconnect = false, 343 float maxRequiredSpeed = 1.0f, 344 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 345 346 347 AudioTrack( audio_stream_type_t streamType, 348 uint32_t sampleRate, 349 audio_format_t format, 350 audio_channel_mask_t channelMask, 351 size_t frameCount, 352 audio_output_flags_t flags, 353 legacy_callback_t cbf, 354 void* user = nullptr, 355 int32_t notificationFrames = 0, 356 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 357 transfer_type transferType = TRANSFER_DEFAULT, 358 const audio_offload_info_t *offloadInfo = nullptr, 359 const AttributionSourceState& attributionSource = 360 AttributionSourceState(), 361 const audio_attributes_t* pAttributes = nullptr, 362 bool doNotReconnect = false, 363 float maxRequiredSpeed = 1.0f, 364 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 365 366 /* Creates an audio track and registers it with AudioFlinger. 367 * With this constructor, the track is configured for static buffer mode. 368 * Data to be rendered is passed in a shared memory buffer 369 * identified by the argument sharedBuffer, which should be non-0. 370 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 371 * but without the ability to specify a non-zero value for the frameCount parameter. 372 * The memory should be initialized to the desired data before calling start(). 373 * The write() method is not supported in this case. 374 * It is recommended to pass a callback function to be notified of playback end by an 375 * EVENT_UNDERRUN event. 376 */ 377 AudioTrack( audio_stream_type_t streamType, 378 uint32_t sampleRate, 379 audio_format_t format, 380 audio_channel_mask_t channelMask, 381 const sp<IMemory>& sharedBuffer, 382 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 383 const wp<IAudioTrackCallback>& callback = nullptr, 384 int32_t notificationFrames = 0, 385 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 386 transfer_type transferType = TRANSFER_DEFAULT, 387 const audio_offload_info_t *offloadInfo = nullptr, 388 const AttributionSourceState& attributionSource = 389 AttributionSourceState(), 390 const audio_attributes_t* pAttributes = nullptr, 391 bool doNotReconnect = false, 392 float maxRequiredSpeed = 1.0f); 393 394 395 AudioTrack( audio_stream_type_t streamType, 396 uint32_t sampleRate, 397 audio_format_t format, 398 audio_channel_mask_t channelMask, 399 const sp<IMemory>& sharedBuffer, 400 audio_output_flags_t flags, 401 legacy_callback_t cbf, 402 void* user = nullptr, 403 int32_t notificationFrames = 0, 404 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 405 transfer_type transferType = TRANSFER_DEFAULT, 406 const audio_offload_info_t *offloadInfo = nullptr, 407 const AttributionSourceState& attributionSource = 408 AttributionSourceState(), 409 const audio_attributes_t* pAttributes = nullptr, 410 bool doNotReconnect = false, 411 float maxRequiredSpeed = 1.0f); 412 413 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 414 * Also destroys all resources associated with the AudioTrack. 415 */ 416 protected: 417 virtual ~AudioTrack(); 418 public: 419 420 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 421 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 422 * set() is not multi-thread safe. 423 * Returned status (from utils/Errors.h) can be: 424 * - NO_ERROR: successful initialization 425 * - INVALID_OPERATION: AudioTrack is already initialized 426 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 427 * - NO_INIT: audio server or audio hardware not initialized 428 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 429 * If sharedBuffer is non-0, the frameCount parameter is ignored and 430 * replaced by the shared buffer's total allocated size in frame units. 431 * 432 * Parameters not listed in the AudioTrack constructors above: 433 * 434 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 435 * Only set to true when AudioTrack object is used for a java android.media.AudioTrack 436 * in its JNI code. 437 * 438 * Internal state post condition: 439 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 440 */ 441 status_t set(audio_stream_type_t streamType, 442 uint32_t sampleRate, 443 audio_format_t format, 444 audio_channel_mask_t channelMask, 445 size_t frameCount = 0, 446 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 447 const wp<IAudioTrackCallback>& callback = nullptr, 448 int32_t notificationFrames = 0, 449 const sp<IMemory>& sharedBuffer = 0, 450 bool threadCanCallJava = false, 451 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 452 transfer_type transferType = TRANSFER_DEFAULT, 453 const audio_offload_info_t *offloadInfo = nullptr, 454 const AttributionSourceState& attributionSource = 455 AttributionSourceState(), 456 const audio_attributes_t* pAttributes = nullptr, 457 bool doNotReconnect = false, 458 float maxRequiredSpeed = 1.0f, 459 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 460 461 struct SetParams { 462 audio_stream_type_t streamType; 463 uint32_t sampleRate; 464 audio_format_t format; 465 audio_channel_mask_t channelMask; 466 size_t frameCount; 467 audio_output_flags_t flags; 468 wp<IAudioTrackCallback> callback; 469 int32_t notificationFrames; 470 sp<IMemory> sharedBuffer; 471 bool threadCanCallJava; 472 audio_session_t sessionId; 473 transfer_type transferType; 474 // TODO don't take pointers here 475 const audio_offload_info_t *offloadInfo; 476 AttributionSourceState attributionSource; 477 const audio_attributes_t* pAttributes; 478 bool doNotReconnect; 479 float maxRequiredSpeed; 480 audio_port_handle_t selectedDeviceId; 481 }; 482 private: 483 // Note: Consumes parameters set(SetParams & s)484 void set(SetParams& s) { 485 (void)set(s.streamType, s.sampleRate, s.format, s.channelMask, s.frameCount, 486 s.flags, std::move(s.callback), s.notificationFrames, 487 std::move(s.sharedBuffer), s.threadCanCallJava, s.sessionId, 488 s.transferType, s.offloadInfo, std::move(s.attributionSource), 489 s.pAttributes, s.doNotReconnect, s.maxRequiredSpeed, s.selectedDeviceId); 490 } 491 void onFirstRef() override; 492 public: 493 status_t set(audio_stream_type_t streamType, 494 uint32_t sampleRate, 495 audio_format_t format, 496 audio_channel_mask_t channelMask, 497 size_t frameCount, 498 audio_output_flags_t flags, 499 legacy_callback_t callback, 500 void * user = nullptr, 501 int32_t notificationFrames = 0, 502 const sp<IMemory>& sharedBuffer = 0, 503 bool threadCanCallJava = false, 504 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 505 transfer_type transferType = TRANSFER_DEFAULT, 506 const audio_offload_info_t *offloadInfo = nullptr, 507 const AttributionSourceState& attributionSource = 508 AttributionSourceState(), 509 const audio_attributes_t* pAttributes = nullptr, 510 bool doNotReconnect = false, 511 float maxRequiredSpeed = 1.0f, 512 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 513 514 // FIXME(b/169889714): Vendor code depends on the old method signature at link time 515 status_t set(audio_stream_type_t streamType, 516 uint32_t sampleRate, 517 audio_format_t format, 518 uint32_t channelMask, 519 size_t frameCount = 0, 520 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 521 legacy_callback_t cbf = nullptr, 522 void* user = nullptr, 523 int32_t notificationFrames = 0, 524 const sp<IMemory>& sharedBuffer = 0, 525 bool threadCanCallJava = false, 526 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 527 transfer_type transferType = TRANSFER_DEFAULT, 528 const audio_offload_info_t *offloadInfo = nullptr, 529 uid_t uid = AUDIO_UID_INVALID, 530 pid_t pid = -1, 531 const audio_attributes_t* pAttributes = nullptr, 532 bool doNotReconnect = false, 533 float maxRequiredSpeed = 1.0f, 534 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 535 536 /* Result of constructing the AudioTrack. This must be checked for successful initialization 537 * before using any AudioTrack API (except for set()), because using 538 * an uninitialized AudioTrack produces undefined results. 539 * See set() method above for possible return codes. 540 */ initCheck()541 status_t initCheck() const { return mStatus; } 542 543 /* Returns this track's estimated latency in milliseconds. 544 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 545 * and audio hardware driver. 546 */ 547 uint32_t latency(); 548 549 /* Returns the number of application-level buffer underruns 550 * since the AudioTrack was created. 551 */ 552 uint32_t getUnderrunCount() const; 553 554 /* getters, see constructors and set() */ 555 556 audio_stream_type_t streamType() const; format()557 audio_format_t format() const { return mFormat; } 558 559 /* Return frame size in bytes, which for linear PCM is 560 * channelCount * (bit depth per channel / 8). 561 * channelCount is determined from channelMask, and bit depth comes from format. 562 * For non-linear formats, the frame size is typically 1 byte. 563 */ frameSize()564 size_t frameSize() const { return mFrameSize; } 565 channelCount()566 uint32_t channelCount() const { return mChannelCount; } frameCount()567 size_t frameCount() const { return mFrameCount; } channelMask()568 audio_channel_mask_t channelMask() const { return mChannelMask; } 569 570 /* 571 * Return the period of the notification callback in frames. 572 * This value is set when the AudioTrack is constructed. 573 * It can be modified if the AudioTrack is rerouted. 574 */ getNotificationPeriodInFrames()575 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 576 577 /* Return effective size of audio buffer that an application writes to 578 * or a negative error if the track is uninitialized. 579 */ 580 ssize_t getBufferSizeInFrames(); 581 582 /* Returns the buffer duration in microseconds at current playback rate. 583 */ 584 status_t getBufferDurationInUs(int64_t *duration); 585 586 /* Set the effective size of audio buffer that an application writes to. 587 * This is used to determine the amount of available room in the buffer, 588 * which determines when a write will block. 589 * This allows an application to raise and lower the audio latency. 590 * The requested size may be adjusted so that it is 591 * greater or equal to the absolute minimum and 592 * less than or equal to the getBufferCapacityInFrames(). 593 * It may also be adjusted slightly for internal reasons. 594 * 595 * Return the final size or a negative value (NO_INIT) if the track is uninitialized. 596 */ 597 ssize_t setBufferSizeInFrames(size_t size); 598 599 /* Returns the start threshold on the buffer for audio streaming 600 * or a negative value if the AudioTrack is not initialized. 601 */ 602 ssize_t getStartThresholdInFrames() const; 603 604 /* Sets the start threshold in frames on the buffer for audio streaming. 605 * 606 * May be clamped internally. Returns the actual value set, or a negative 607 * value if the AudioTrack is not initialized or if the input 608 * is zero or greater than INT_MAX. 609 */ 610 ssize_t setStartThresholdInFrames(size_t startThresholdInFrames); 611 612 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ sharedBuffer()613 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 614 615 /* 616 * return metrics information for the current track. 617 */ 618 status_t getMetrics(mediametrics::Item * &item); 619 620 /* 621 * Set name of API that is using this object. 622 * For example "aaudio" or "opensles". 623 * This may be logged or reported as part of MediaMetrics. 624 */ setCallerName(const std::string & name)625 void setCallerName(const std::string &name) { 626 mCallerName = name; 627 } 628 getCallerName()629 std::string getCallerName() const { 630 return mCallerName; 631 }; 632 633 /* After it's created the track is not active. Call start() to 634 * make it active. If set, the callback will start being called. 635 * If the track was previously paused, volume is ramped up over the first mix buffer. 636 */ 637 status_t start(); 638 639 /* Stop a track. 640 * In static buffer mode, the track is stopped immediately. 641 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 642 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 643 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 644 * is first drained, mixed, and output, and only then is the track marked as stopped. 645 */ 646 void stop(); 647 bool stopped() const; 648 649 /* Call stop() and then wait for all of the callbacks to return. 650 * It is safe to call this if stop() or pause() has already been called. 651 * 652 * This function is called from the destructor. But since AudioTrack 653 * is ref counted, the destructor may be called later than desired. 654 * This can be called explicitly as part of closing an AudioTrack 655 * if you want to be certain that callbacks have completely finished. 656 * 657 * This is not thread safe and should only be called from one thread, 658 * ideally as the AudioTrack is being closed. 659 */ 660 void stopAndJoinCallbacks(); 661 662 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 663 * This has the effect of draining the buffers without mixing or output. 664 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 665 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 666 */ 667 void flush(); 668 669 /* Pause a track. After pause, the callback will cease being called and 670 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 671 * and will fill up buffers until the pool is exhausted. 672 * Volume is ramped down over the next mix buffer following the pause request, 673 * and then the track is marked as paused. It can be resumed with ramp up by start(). 674 */ 675 void pause(); 676 677 /* Pause and wait (with timeout) for the audio track to ramp to silence. 678 * 679 * \param timeout is the time limit to wait before returning. 680 * A negative number is treated as 0. 681 * \return true if the track is ramped to silence, false if the timeout occurred. 682 */ 683 bool pauseAndWait(const std::chrono::milliseconds& timeout); 684 685 /* Set volume for this track, mostly used for games' sound effects 686 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 687 * This is the older API. New applications should use setVolume(float) when possible. 688 */ 689 status_t setVolume(float left, float right); 690 691 /* Set volume for all channels. This is the preferred API for new applications, 692 * especially for multi-channel content. 693 */ 694 status_t setVolume(float volume); 695 696 /* Set the send level for this track. An auxiliary effect should be attached 697 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 698 */ 699 status_t setAuxEffectSendLevel(float level); 700 void getAuxEffectSendLevel(float* level) const; 701 702 /* Set source sample rate for this track in Hz, mostly used for games' sound effects. 703 * Zero is not permitted. 704 */ 705 status_t setSampleRate(uint32_t sampleRate); 706 707 /* Return current source sample rate in Hz. 708 * If specified as zero in constructor or set(), this will be the sink sample rate. 709 */ 710 uint32_t getSampleRate() const; 711 712 /* Return the original source sample rate in Hz. This corresponds to the sample rate 713 * if playback rate had normal speed and pitch. 714 */ 715 uint32_t getOriginalSampleRate() const; 716 717 /* Sets the Dual Mono mode presentation on the output device. */ 718 status_t setDualMonoMode(audio_dual_mono_mode_t mode); 719 720 /* Returns the Dual Mono mode presentation setting. */ 721 status_t getDualMonoMode(audio_dual_mono_mode_t* mode) const; 722 723 /* Sets the Audio Description Mix level in dB. */ 724 status_t setAudioDescriptionMixLevel(float leveldB); 725 726 /* Returns the Audio Description Mix level in dB. */ 727 status_t getAudioDescriptionMixLevel(float* leveldB) const; 728 729 /* Set source playback rate for timestretch 730 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 731 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 732 * 733 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 734 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 735 * 736 * Speed increases the playback rate of media, but does not alter pitch. 737 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 738 */ 739 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 740 741 /* Return current playback rate */ 742 const AudioPlaybackRate& getPlaybackRate(); 743 744 /* Enables looping and sets the start and end points of looping. 745 * Only supported for static buffer mode. 746 * 747 * Parameters: 748 * 749 * loopStart: loop start in frames relative to start of buffer. 750 * loopEnd: loop end in frames relative to start of buffer. 751 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 752 * pending or active loop. loopCount == -1 means infinite looping. 753 * 754 * For proper operation the following condition must be respected: 755 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 756 * 757 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 758 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 759 * 760 */ 761 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 762 763 /* Sets marker position. When playback reaches the number of frames specified, a callback with 764 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 765 * notification callback. To set a marker at a position which would compute as 0, 766 * a workaround is to set the marker at a nearby position such as ~0 or 1. 767 * If the AudioTrack has been opened with no callback function associated, the operation will 768 * fail. 769 * 770 * Parameters: 771 * 772 * marker: marker position expressed in wrapping (overflow) frame units, 773 * like the return value of getPosition(). 774 * 775 * Returned status (from utils/Errors.h) can be: 776 * - NO_ERROR: successful operation 777 * - INVALID_OPERATION: the AudioTrack has no callback installed. 778 */ 779 status_t setMarkerPosition(uint32_t marker); 780 status_t getMarkerPosition(uint32_t *marker) const; 781 782 /* Sets position update period. Every time the number of frames specified has been played, 783 * a callback with event type EVENT_NEW_POS is called. 784 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 785 * callback. 786 * If the AudioTrack has been opened with no callback function associated, the operation will 787 * fail. 788 * Extremely small values may be rounded up to a value the implementation can support. 789 * 790 * Parameters: 791 * 792 * updatePeriod: position update notification period expressed in frames. 793 * 794 * Returned status (from utils/Errors.h) can be: 795 * - NO_ERROR: successful operation 796 * - INVALID_OPERATION: the AudioTrack has no callback installed. 797 */ 798 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 799 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 800 801 /* Sets playback head position. 802 * Only supported for static buffer mode. 803 * 804 * Parameters: 805 * 806 * position: New playback head position in frames relative to start of buffer. 807 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 808 * but will result in an immediate underrun if started. 809 * 810 * Returned status (from utils/Errors.h) can be: 811 * - NO_ERROR: successful operation 812 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 813 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 814 * buffer 815 */ 816 status_t setPosition(uint32_t position); 817 818 /* Return the total number of frames played since playback start. 819 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 820 * It is reset to zero by flush(), reload(), and stop(). 821 * 822 * Parameters: 823 * 824 * position: Address where to return play head position. 825 * 826 * Returned status (from utils/Errors.h) can be: 827 * - NO_ERROR: successful operation 828 * - BAD_VALUE: position is NULL 829 */ 830 status_t getPosition(uint32_t *position); 831 832 /* For static buffer mode only, this returns the current playback position in frames 833 * relative to start of buffer. It is analogous to the position units used by 834 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 835 */ 836 status_t getBufferPosition(uint32_t *position); 837 838 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 839 * rewriting the buffer before restarting playback after a stop. 840 * This method must be called with the AudioTrack in paused or stopped state. 841 * Not allowed in streaming mode. 842 * 843 * Returned status (from utils/Errors.h) can be: 844 * - NO_ERROR: successful operation 845 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 846 */ 847 status_t reload(); 848 849 /** 850 * @param transferType 851 * @return text string that matches the enum name 852 */ 853 static const char * convertTransferToText(transfer_type transferType); 854 855 /* Returns a handle on the audio output used by this AudioTrack. 856 * 857 * Parameters: 858 * none. 859 * 860 * Returned value: 861 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 862 * track needed to be re-created but that failed 863 */ 864 audio_io_handle_t getOutput() const; 865 866 /* Selects the audio device to use for output of this AudioTrack. A value of 867 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 868 * 869 * Parameters: 870 * The device ID of the selected device (as returned by the AudioDevicesManager API). 871 * 872 * Returned value: 873 * - NO_ERROR: successful operation 874 * TODO: what else can happen here? 875 */ 876 status_t setOutputDevice(audio_port_handle_t deviceId); 877 878 /* Returns the ID of the audio device selected for this AudioTrack. 879 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 880 * 881 * Parameters: 882 * none. 883 */ 884 audio_port_handle_t getOutputDevice(); 885 886 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 887 * attached. 888 * When the AudioTrack is inactive, the device ID returned can be either: 889 * - AUDIO_PORT_HANDLE_NONE if the AudioTrack is not attached to any output. 890 * - The device ID used before paused or stopped. 891 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioTrack 892 * has not been started yet. 893 * 894 * Parameters: 895 * none. 896 */ 897 audio_port_handle_t getRoutedDeviceId(); 898 899 /* Returns the unique session ID associated with this track. 900 * 901 * Parameters: 902 * none. 903 * 904 * Returned value: 905 * AudioTrack session ID. 906 */ getSessionId()907 audio_session_t getSessionId() const { return mSessionId; } 908 909 /* Attach track auxiliary output to specified effect. Use effectId = 0 910 * to detach track from effect. 911 * 912 * Parameters: 913 * 914 * effectId: effectId obtained from AudioEffect::id(). 915 * 916 * Returned status (from utils/Errors.h) can be: 917 * - NO_ERROR: successful operation 918 * - INVALID_OPERATION: the effect is not an auxiliary effect. 919 * - BAD_VALUE: The specified effect ID is invalid 920 */ 921 status_t attachAuxEffect(int effectId); 922 923 /* Public API for TRANSFER_OBTAIN mode. 924 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 925 * After filling these slots with data, the caller should release them with releaseBuffer(). 926 * If the track buffer is not full, obtainBuffer() returns as many contiguous 927 * [empty slots for] frames as are available immediately. 928 * 929 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 930 * additional non-contiguous frames that are predicted to be available immediately, 931 * if the client were to release the first frames and then call obtainBuffer() again. 932 * This value is only a prediction, and needs to be confirmed. 933 * It will be set to zero for an error return. 934 * 935 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 936 * regardless of the value of waitCount. 937 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 938 * maximum timeout based on waitCount; see chart below. 939 * Buffers will be returned until the pool 940 * is exhausted, at which point obtainBuffer() will either block 941 * or return WOULD_BLOCK depending on the value of the "waitCount" 942 * parameter. 943 * 944 * Interpretation of waitCount: 945 * +n limits wait time to n * WAIT_PERIOD_MS, 946 * -1 causes an (almost) infinite wait time, 947 * 0 non-blocking. 948 * 949 * Buffer fields 950 * On entry: 951 * frameCount number of [empty slots for] frames requested 952 * size ignored 953 * raw ignored 954 * sequence ignored 955 * After error return: 956 * frameCount 0 957 * size 0 958 * raw undefined 959 * sequence undefined 960 * After successful return: 961 * frameCount actual number of [empty slots for] frames available, <= number requested 962 * size actual number of bytes available 963 * raw pointer to the buffer 964 * sequence IAudioTrack instance sequence number, as of obtainBuffer() 965 */ 966 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 967 size_t *nonContig = NULL); 968 969 private: 970 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 971 * additional non-contiguous frames that are predicted to be available immediately, 972 * if the client were to release the first frames and then call obtainBuffer() again. 973 * This value is only a prediction, and needs to be confirmed. 974 * It will be set to zero for an error return. 975 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 976 * in case the requested amount of frames is in two or more non-contiguous regions. 977 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 978 */ 979 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 980 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 981 public: 982 983 /* Public API for TRANSFER_OBTAIN mode. 984 * Release a filled buffer of frames for AudioFlinger to process. 985 * 986 * Buffer fields: 987 * frameCount currently ignored but recommend to set to actual number of frames filled 988 * size actual number of bytes filled, must be multiple of frameSize 989 * raw ignored 990 */ 991 void releaseBuffer(const Buffer* audioBuffer); 992 993 /* As a convenience we provide a write() interface to the audio buffer. 994 * Input parameter 'size' is in byte units. 995 * This is implemented on top of obtainBuffer/releaseBuffer. For best 996 * performance use callbacks. Returns actual number of bytes written >= 0, 997 * or one of the following negative status codes: 998 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 999 * BAD_VALUE size is invalid 1000 * WOULD_BLOCK when obtainBuffer() returns same, or 1001 * AudioTrack was stopped during the write 1002 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 1003 * the track cannot be automatically restored. 1004 * The application needs to recreate the AudioTrack 1005 * because the audio device changed or AudioFlinger died. 1006 * This typically occurs for direct or offload tracks 1007 * or if mDoNotReconnect is true. 1008 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 1009 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 1010 * false for the method to return immediately without waiting to try multiple times to write 1011 * the full content of the buffer. 1012 */ 1013 ssize_t write(const void* buffer, size_t size, bool blocking = true); 1014 1015 /* 1016 * Dumps the state of an audio track. 1017 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 1018 */ 1019 status_t dump(int fd, const Vector<String16>& args) const; 1020 1021 /* 1022 * Return the total number of frames which AudioFlinger desired but were unavailable, 1023 * and thus which resulted in an underrun. Reset to zero by stop(). 1024 */ 1025 uint32_t getUnderrunFrames() const; 1026 1027 /* Get the flags */ getFlags()1028 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 1029 1030 /* Set parameters - only possible when using direct output */ 1031 status_t setParameters(const String8& keyValuePairs); 1032 1033 /* Sets the volume shaper object */ 1034 media::VolumeShaper::Status applyVolumeShaper( 1035 const sp<media::VolumeShaper::Configuration>& configuration, 1036 const sp<media::VolumeShaper::Operation>& operation); 1037 1038 /* Gets the volume shaper state */ 1039 sp<media::VolumeShaper::State> getVolumeShaperState(int id); 1040 1041 /* Selects the presentation (if available) */ 1042 status_t selectPresentation(int presentationId, int programId); 1043 1044 /* Get parameters */ 1045 String8 getParameters(const String8& keys); 1046 1047 /* Poll for a timestamp on demand. 1048 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 1049 * or if you need to get the most recent timestamp outside of the event callback handler. 1050 * Caution: calling this method too often may be inefficient; 1051 * if you need a high resolution mapping between frame position and presentation time, 1052 * consider implementing that at application level, based on the low resolution timestamps. 1053 * Returns NO_ERROR if timestamp is valid. 1054 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 1055 * start/ACTIVE, when the number of frames consumed is less than the 1056 * overall hardware latency to physical output. In WOULD_BLOCK cases, 1057 * one might poll again, or use getPosition(), or use 0 position and 1058 * current time for the timestamp. 1059 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 1060 * the track cannot be automatically restored. 1061 * The application needs to recreate the AudioTrack 1062 * because the audio device changed or AudioFlinger died. 1063 * This typically occurs for direct or offload tracks 1064 * or if mDoNotReconnect is true. 1065 * INVALID_OPERATION wrong state, or some other error. 1066 * 1067 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 1068 */ 1069 status_t getTimestamp(AudioTimestamp& timestamp); 1070 private: 1071 status_t getTimestamp_l(AudioTimestamp& timestamp); 1072 public: 1073 1074 /* Return the extended timestamp, with additional timebase info and improved drain behavior. 1075 * 1076 * This is similar to the AudioTrack.java API: 1077 * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) 1078 * 1079 * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method 1080 * 1081 * 1. stop() by itself does not reset the frame position. 1082 * A following start() resets the frame position to 0. 1083 * 2. flush() by itself does not reset the frame position. 1084 * The frame position advances by the number of frames flushed, 1085 * when the first frame after flush reaches the audio sink. 1086 * 3. BOOTTIME clock offsets are provided to help synchronize with 1087 * non-audio streams, e.g. sensor data. 1088 * 4. Position is returned with 64 bits of resolution. 1089 * 1090 * Parameters: 1091 * timestamp: A pointer to the caller allocated ExtendedTimestamp. 1092 * 1093 * Returns NO_ERROR on success; timestamp is filled with valid data. 1094 * BAD_VALUE if timestamp is NULL. 1095 * WOULD_BLOCK if called immediately after start() when the number 1096 * of frames consumed is less than the 1097 * overall hardware latency to physical output. In WOULD_BLOCK cases, 1098 * one might poll again, or use getPosition(), or use 0 position and 1099 * current time for the timestamp. 1100 * If WOULD_BLOCK is returned, the timestamp is still 1101 * modified with the LOCATION_CLIENT portion filled. 1102 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 1103 * the track cannot be automatically restored. 1104 * The application needs to recreate the AudioTrack 1105 * because the audio device changed or AudioFlinger died. 1106 * This typically occurs for direct or offloaded tracks 1107 * or if mDoNotReconnect is true. 1108 * INVALID_OPERATION if called on a offloaded or direct track. 1109 * Use getTimestamp(AudioTimestamp& timestamp) instead. 1110 */ 1111 status_t getTimestamp(ExtendedTimestamp *timestamp); 1112 private: 1113 status_t getTimestamp_l(ExtendedTimestamp *timestamp); 1114 public: 1115 1116 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 1117 * AudioTrack is routed is updated. 1118 * Replaces any previously installed callback. 1119 * Parameters: 1120 * callback: The callback interface 1121 * Returns NO_ERROR if successful. 1122 * INVALID_OPERATION if the same callback is already installed. 1123 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 1124 * BAD_VALUE if the callback is NULL 1125 */ 1126 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 1127 1128 /* remove an AudioDeviceCallback. 1129 * Parameters: 1130 * callback: The callback interface 1131 * Returns NO_ERROR if successful. 1132 * INVALID_OPERATION if the callback is not installed 1133 * BAD_VALUE if the callback is NULL 1134 */ 1135 status_t removeAudioDeviceCallback( 1136 const sp<AudioSystem::AudioDeviceCallback>& callback); 1137 1138 // AudioSystem::AudioDeviceCallback> virtuals 1139 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 1140 audio_port_handle_t deviceId); 1141 1142 /* Obtain the pending duration in milliseconds for playback of pure PCM 1143 * (mixable without embedded timing) data remaining in AudioTrack. 1144 * 1145 * This is used to estimate the drain time for the client-server buffer 1146 * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. 1147 * One may optionally request to find the duration to play through the HAL 1148 * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, 1149 * INVALID_OPERATION may be returned if the kernel location is unavailable. 1150 * 1151 * Returns NO_ERROR if successful. 1152 * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained 1153 * or the AudioTrack does not contain pure PCM data. 1154 * BAD_VALUE if msec is nullptr or location is invalid. 1155 */ 1156 status_t pendingDuration(int32_t *msec, 1157 ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); 1158 1159 /* hasStarted() is used to determine if audio is now audible at the device after 1160 * a start() command. The underlying implementation checks a nonzero timestamp position 1161 * or increment for the audible assumption. 1162 * 1163 * hasStarted() returns true if the track has been started() and audio is audible 1164 * and no subsequent pause() or flush() has been called. Immediately after pause() or 1165 * flush() hasStarted() will return false. 1166 * 1167 * If stop() has been called, hasStarted() will return true if audio is still being 1168 * delivered or has finished delivery (even if no audio was written) for both offloaded 1169 * and normal tracks. This property removes a race condition in checking hasStarted() 1170 * for very short clips, where stop() must be called to finish drain. 1171 * 1172 * In all cases, hasStarted() may turn false briefly after a subsequent start() is called 1173 * until audio becomes audible again. 1174 */ 1175 bool hasStarted(); // not const 1176 isPlaying()1177 bool isPlaying() { 1178 AutoMutex lock(mLock); 1179 return isPlaying_l(); 1180 } isPlaying_l()1181 bool isPlaying_l() { 1182 return mState == STATE_ACTIVE || mState == STATE_STOPPING; 1183 } 1184 1185 /* Get the unique port ID assigned to this AudioTrack instance by audio policy manager. 1186 * The ID is unique across all audioserver clients and can change during the life cycle 1187 * of a given AudioTrack instance if the connection to audioserver is restored. 1188 */ getPortId()1189 audio_port_handle_t getPortId() const { return mPortId; }; 1190 1191 /* Sets the LogSessionId field which is used for metrics association of 1192 * this object with other objects. A nullptr or empty string clears 1193 * the logSessionId. 1194 */ 1195 void setLogSessionId(const char *logSessionId); 1196 1197 /* Sets the playerIId field to associate the AudioTrack with an interface managed by 1198 * AudioService. 1199 * 1200 * If this value is not set, then the playerIId is reported as -1 1201 * (not associated with an AudioService player interface). 1202 * 1203 * For metrics purposes, we keep the playerIId association in the native 1204 * client AudioTrack to improve the robustness under track restoration. 1205 */ 1206 void setPlayerIId(int playerIId); 1207 setAudioTrackCallback(const sp<media::IAudioTrackCallback> & callback)1208 void setAudioTrackCallback(const sp<media::IAudioTrackCallback>& callback) { 1209 mAudioTrackCallback->setAudioTrackCallback(callback); 1210 } 1211 1212 protected: 1213 /* copying audio tracks is not allowed */ 1214 AudioTrack(const AudioTrack& other); 1215 AudioTrack& operator = (const AudioTrack& other); 1216 1217 /* a small internal class to handle the callback */ 1218 class AudioTrackThread : public Thread 1219 { 1220 public: 1221 explicit AudioTrackThread(AudioTrack& receiver); 1222 1223 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 1224 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 1225 virtual void requestExit(); 1226 1227 void pause(); // suspend thread from execution at next loop boundary 1228 void resume(); // allow thread to execute, if not requested to exit 1229 void wake(); // wake to handle changed notification conditions. 1230 1231 private: 1232 void pauseInternal(nsecs_t ns = 0LL); 1233 // like pause(), but only used internally within thread 1234 1235 friend class AudioTrack; 1236 virtual bool threadLoop(); 1237 AudioTrack& mReceiver; 1238 virtual ~AudioTrackThread(); 1239 Mutex mMyLock; // Thread::mLock is private 1240 Condition mMyCond; // Thread::mThreadExitedCondition is private 1241 bool mPaused; // whether thread is requested to pause at next loop entry 1242 bool mPausedInt; // whether thread internally requests pause 1243 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 1244 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 1245 // to processAudioBuffer() as state may have changed 1246 // since pause time calculated. 1247 }; 1248 1249 // body of AudioTrackThread::threadLoop() 1250 // returns the maximum amount of time before we would like to run again, where: 1251 // 0 immediately 1252 // > 0 no later than this many nanoseconds from now 1253 // NS_WHENEVER still active but no particular deadline 1254 // NS_INACTIVE inactive so don't run again until re-started 1255 // NS_NEVER never again 1256 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 1257 nsecs_t processAudioBuffer(); 1258 1259 // caller must hold lock on mLock for all _l methods 1260 1261 void updateLatency_l(); // updates mAfLatency and mLatency from AudioSystem cache 1262 1263 status_t createTrack_l(); 1264 1265 // can only be called when mState != STATE_ACTIVE 1266 void flush_l(); 1267 1268 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 1269 1270 // FIXME enum is faster than strcmp() for parameter 'from' 1271 status_t restoreTrack_l(const char *from); 1272 1273 uint32_t getUnderrunCount_l() const; 1274 1275 bool isOffloaded() const; 1276 bool isDirect() const; 1277 bool isOffloadedOrDirect() const; 1278 isOffloaded_l()1279 bool isOffloaded_l() const 1280 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 1281 isOffloadedOrDirect_l()1282 bool isOffloadedOrDirect_l() const 1283 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 1284 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 1285 isDirect_l()1286 bool isDirect_l() const 1287 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 1288 1289 // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) isPurePcmData_l()1290 bool isPurePcmData_l() const 1291 { return audio_is_linear_pcm(mFormat) 1292 && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } 1293 1294 // increment mPosition by the delta of mServer, and return new value of mPosition 1295 Modulo<uint32_t> updateAndGetPosition_l(); 1296 1297 // check sample rate and speed is compatible with AudioTrack 1298 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed); 1299 1300 void restartIfDisabled(); 1301 1302 void updateRoutedDeviceId_l(); 1303 1304 /* Sets the Dual Mono mode presentation on the output device. */ 1305 status_t setDualMonoMode_l(audio_dual_mono_mode_t mode); 1306 1307 /* Sets the Audio Description Mix level in dB. */ 1308 status_t setAudioDescriptionMixLevel_l(float leveldB); 1309 1310 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 1311 sp<media::IAudioTrack> mAudioTrack; 1312 sp<IMemory> mCblkMemory; 1313 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 1314 audio_io_handle_t mOutput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getOutputForAttr() 1315 1316 sp<AudioTrackThread> mAudioTrackThread; 1317 bool mThreadCanCallJava; 1318 1319 float mVolume[2]; 1320 float mSendLevel; 1321 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 1322 uint32_t mOriginalSampleRate; 1323 AudioPlaybackRate mPlaybackRate; 1324 float mMaxRequiredSpeed; // use PCM buffer size to allow this speed 1325 1326 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 1327 // This allocated buffer size is maintained by the proxy. 1328 size_t mFrameCount; // maximum size of buffer 1329 1330 size_t mReqFrameCount; // frame count to request the first or next time 1331 // a new IAudioTrack is needed, non-decreasing 1332 1333 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 1334 // These values can be used for informational purposes until the track is invalidated, 1335 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 1336 uint32_t mAfLatency; // AudioFlinger latency in ms 1337 size_t mAfFrameCount; // AudioFlinger frame count 1338 uint32_t mAfSampleRate; // AudioFlinger sample rate 1339 1340 // constant after constructor or set() 1341 audio_format_t mFormat; // as requested by client, not forced to 16-bit 1342 // mOriginalStreamType == AUDIO_STREAM_DEFAULT implies this AudioTrack has valid attributes 1343 audio_stream_type_t mOriginalStreamType = AUDIO_STREAM_DEFAULT; 1344 audio_stream_type_t mStreamType = AUDIO_STREAM_DEFAULT; 1345 uint32_t mChannelCount; 1346 audio_channel_mask_t mChannelMask; 1347 sp<IMemory> mSharedBuffer; 1348 transfer_type mTransfer; 1349 audio_offload_info_t mOffloadInfoCopy; 1350 audio_attributes_t mAttributes; 1351 1352 size_t mFrameSize; // frame size in bytes 1353 1354 status_t mStatus; 1355 1356 // can change dynamically when IAudioTrack invalidated 1357 uint32_t mLatency; // in ms 1358 1359 // Indicates the current track state. Protected by mLock. 1360 enum State { 1361 STATE_ACTIVE, 1362 STATE_STOPPED, 1363 STATE_PAUSED, 1364 STATE_PAUSED_STOPPING, 1365 STATE_FLUSHED, 1366 STATE_STOPPING, 1367 } mState; 1368 stateToString(State state)1369 static constexpr const char *stateToString(State state) 1370 { 1371 switch (state) { 1372 case STATE_ACTIVE: return "STATE_ACTIVE"; 1373 case STATE_STOPPED: return "STATE_STOPPED"; 1374 case STATE_PAUSED: return "STATE_PAUSED"; 1375 case STATE_PAUSED_STOPPING: return "STATE_PAUSED_STOPPING"; 1376 case STATE_FLUSHED: return "STATE_FLUSHED"; 1377 case STATE_STOPPING: return "STATE_STOPPING"; 1378 default: return "UNKNOWN"; 1379 } 1380 } 1381 1382 // for client callback handler 1383 wp<IAudioTrackCallback> mCallback; // callback handler for events, or NULL 1384 sp<IAudioTrackCallback> mLegacyCallbackWrapper; // wrapper for legacy callback interface 1385 // for notification APIs 1386 std::unique_ptr<SetParams> mSetParams; // Temporary copy of ctor params to allow for 1387 // deferred set after first reference. 1388 1389 bool mInitialized = false; // Set after track is initialized 1390 // next 2 fields are const after constructor or set() 1391 uint32_t mNotificationFramesReq; // requested number of frames between each 1392 // notification callback, 1393 // at initial source sample rate 1394 uint32_t mNotificationsPerBufferReq; 1395 // requested number of notifications per buffer, 1396 // currently only used for fast tracks with 1397 // default track buffer size 1398 1399 uint32_t mNotificationFramesAct; // actual number of frames between each 1400 // notification callback, 1401 // at initial source sample rate 1402 bool mRefreshRemaining; // processAudioBuffer() should refresh 1403 // mRemainingFrames and mRetryOnPartialBuffer 1404 1405 // used for static track cbf and restoration 1406 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 1407 uint32_t mLoopStart; // last setLoop loopStart 1408 uint32_t mLoopEnd; // last setLoop loopEnd 1409 int32_t mLoopCountNotified; // the last loopCount notified by callback. 1410 // mLoopCountNotified counts down, matching 1411 // the remaining loop count for static track 1412 // playback. 1413 1414 // These are private to processAudioBuffer(), and are not protected by a lock 1415 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 1416 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 1417 uint32_t mObservedSequence; // last observed value of mSequence 1418 1419 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 1420 bool mMarkerReached; 1421 Modulo<uint32_t> mNewPosition; // in frames 1422 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 1423 1424 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 1425 // which is count of frames consumed by server, 1426 // reset by new IAudioTrack, 1427 // whether it is reset by stop() is TBD 1428 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 1429 // monotonically after new IAudioTrack, 1430 // and could be easily widened to uint64_t 1431 Modulo<uint32_t> mReleased; // count of frames released to server 1432 // but not necessarily consumed by server, 1433 // reset by stop() but continues monotonically 1434 // after new IAudioTrack to restore mPosition, 1435 // and could be easily widened to uint64_t 1436 int64_t mStartFromZeroUs; // the start time after flush or stop, 1437 // when position should be 0. 1438 // only used for offloaded and direct tracks. 1439 int64_t mStartNs; // the time when start() is called. 1440 ExtendedTimestamp mStartEts; // Extended timestamp at start for normal 1441 // AudioTracks. 1442 AudioTimestamp mStartTs; // Timestamp at start for offloaded or direct 1443 // AudioTracks. 1444 1445 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 1446 bool mTimestampStartupGlitchReported; // reduce log spam 1447 bool mTimestampRetrogradePositionReported; // reduce log spam 1448 bool mTimestampRetrogradeTimeReported; // reduce log spam 1449 bool mTimestampStallReported; // reduce log spam 1450 bool mTimestampStaleTimeReported; // reduce log spam 1451 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 1452 ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp 1453 1454 uint32_t mUnderrunCountOffset; // updated when restoring tracks 1455 1456 int64_t mFramesWritten; // total frames written. reset to zero after 1457 // the start() following stop(). It is not 1458 // changed after restoring the track or 1459 // after flush. 1460 int64_t mFramesWrittenServerOffset; // An offset to server frames due to 1461 // restoring AudioTrack, or stop/start. 1462 // This offset is also used for static tracks. 1463 int64_t mFramesWrittenAtRestore; // Frames written at restore point (or frames 1464 // delivered for static tracks). 1465 // -1 indicates no previous restore point. 1466 1467 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 1468 // be denied by client or server, such as 1469 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 1470 // held to read or write those bits reliably. 1471 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 1472 1473 bool mDoNotReconnect; 1474 1475 audio_session_t mSessionId; 1476 int mAuxEffectId; 1477 audio_port_handle_t mPortId; // Id from Audio Policy Manager 1478 1479 /** 1480 * mPlayerIId is the player id of the AudioTrack used by AudioManager. 1481 * For an AudioTrack created by the Java interface, this is generally set once. 1482 */ 1483 int mPlayerIId = -1; // AudioManager.h PLAYER_PIID_INVALID 1484 1485 /** 1486 * mLogSessionId is a string identifying this AudioTrack for the metrics service. 1487 * It may be unique or shared with other objects. An empty string means the 1488 * logSessionId is not set. 1489 */ 1490 std::string mLogSessionId{}; 1491 1492 mutable Mutex mLock; 1493 1494 int mPreviousPriority; // before start() 1495 SchedPolicy mPreviousSchedulingGroup; 1496 bool mAwaitBoost; // thread should wait for priority boost before running 1497 1498 // The proxy should only be referenced while a lock is held because the proxy isn't 1499 // multi-thread safe, especially the SingleStateQueue part of the proxy. 1500 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 1501 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 1502 // them around in case they are replaced during the obtainBuffer(). 1503 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 1504 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 1505 1506 bool mInUnderrun; // whether track is currently in underrun state 1507 uint32_t mPausedPosition; 1508 1509 // For Device Selection API 1510 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 1511 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 1512 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 1513 // May not match the app selection depending on other 1514 // activity and connected devices. 1515 1516 sp<media::VolumeHandler> mVolumeHandler; 1517 1518 private: 1519 class DeathNotifier : public IBinder::DeathRecipient { 1520 public: DeathNotifier(AudioTrack * audioTrack)1521 explicit DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 1522 protected: 1523 virtual void binderDied(const wp<IBinder>& who); 1524 private: 1525 const wp<AudioTrack> mAudioTrack; 1526 }; 1527 1528 sp<DeathNotifier> mDeathNotifier; 1529 uint32_t mSequence; // incremented for each new IAudioTrack attempt 1530 AttributionSourceState mClientAttributionSource; 1531 1532 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1533 1534 // Cached values to restore along with the AudioTrack. 1535 audio_dual_mono_mode_t mDualMonoMode = AUDIO_DUAL_MONO_MODE_OFF; 1536 float mAudioDescriptionMixLeveldB = -std::numeric_limits<float>::infinity(); 1537 1538 private: 1539 class MediaMetrics { 1540 public: MediaMetrics()1541 MediaMetrics() : mMetricsItem(mediametrics::Item::create("audiotrack")) { 1542 } ~MediaMetrics()1543 ~MediaMetrics() { 1544 // mMetricsItem alloc failure will be flagged in the constructor 1545 // don't log empty records 1546 if (mMetricsItem->count() > 0) { 1547 mMetricsItem->selfrecord(); 1548 } 1549 } 1550 void gather(const AudioTrack *track); dup()1551 mediametrics::Item *dup() { return mMetricsItem->dup(); } 1552 private: 1553 std::unique_ptr<mediametrics::Item> mMetricsItem; 1554 }; 1555 MediaMetrics mMediaMetrics; 1556 std::string mMetricsId; // GUARDED_BY(mLock), could change in createTrack_l(). 1557 std::string mCallerName; // for example "aaudio" 1558 1559 // report error to mediametrics. 1560 void reportError(status_t status, const char *event, const char *message) const; 1561 1562 private: 1563 class AudioTrackCallback : public media::BnAudioTrackCallback { 1564 public: 1565 binder::Status onCodecFormatChanged(const std::vector<uint8_t>& audioMetadata) override; 1566 1567 void setAudioTrackCallback(const sp<media::IAudioTrackCallback>& callback); 1568 private: 1569 Mutex mAudioTrackCbLock; 1570 wp<media::IAudioTrackCallback> mCallback; 1571 }; 1572 sp<AudioTrackCallback> mAudioTrackCallback; 1573 }; 1574 1575 }; // namespace android 1576 1577 #endif // ANDROID_AUDIOTRACK_H 1578