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