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