1 /* 2 * Copyright (C) 2008 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_AUDIORECORD_H 18 #define ANDROID_AUDIORECORD_H 19 20 #include <memory> 21 #include <vector> 22 23 #include <binder/IMemory.h> 24 #include <cutils/sched_policy.h> 25 #include <media/AudioSystem.h> 26 #include <media/AudioTimestamp.h> 27 #include <media/MediaMetricsItem.h> 28 #include <media/Modulo.h> 29 #include <media/RecordingActivityTracker.h> 30 #include <utils/RefBase.h> 31 #include <utils/threads.h> 32 33 #include "android/media/IAudioRecord.h" 34 #include <android/content/AttributionSourceState.h> 35 36 namespace android { 37 38 // ---------------------------------------------------------------------------- 39 40 struct audio_track_cblk_t; 41 class AudioRecordClientProxy; 42 // ---------------------------------------------------------------------------- 43 44 class AudioRecord : public AudioSystem::AudioDeviceCallback 45 { 46 public: 47 48 class Buffer 49 { 50 friend AudioRecord; 51 public: size()52 size_t size() const { return mSize; } getFrameCount()53 size_t getFrameCount() const { return frameCount; } data()54 uint8_t* data() const { return ui8; } 55 // Leaving public for now to assist refactoring. This class will 56 // be replaced. 57 size_t frameCount; // number of sample frames corresponding to size; 58 // on input to obtainBuffer() it is the number of frames desired 59 // on output from obtainBuffer() it is the number of available 60 // frames to be read 61 // on input to releaseBuffer() it is currently ignored 62 63 private: 64 size_t mSize; // input/output in bytes == frameCount * frameSize 65 // on input to obtainBuffer() it is ignored 66 // on output from obtainBuffer() it is the number of available 67 // bytes to be read, which is frameCount * frameSize 68 // on input to releaseBuffer() it is the number of bytes to 69 // release 70 // FIXME This is redundant with respect to frameCount. Consider 71 // removing size and making frameCount the primary field. 72 73 union { 74 void* raw; 75 int16_t* i16; // signed 16-bit 76 uint8_t* ui8; // unsigned 8-bit, offset by 0x80 77 // input to obtainBuffer(): unused, output: pointer to buffer 78 }; 79 80 uint32_t sequence; // IAudioRecord instance sequence number, as of obtainBuffer(). 81 // It is set by obtainBuffer() and confirmed by releaseBuffer(). 82 // Not "user-serviceable". 83 // TODO Consider sp<IMemory> instead, or in addition to this. 84 }; 85 86 /* As a convenience, if a callback is supplied, a handler thread 87 * is automatically created with the appropriate priority. This thread 88 * invokes the callback when a new buffer becomes available or various conditions occur. 89 * Parameters: 90 * 91 * event: type of event notified (see enum AudioRecord::event_type). 92 * user: Pointer to context for use by the callback receiver. 93 * info: Pointer to optional parameter according to event type: 94 * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read 95 * more bytes than indicated by 'size' field and update 'size' if 96 * fewer bytes are consumed. 97 * - EVENT_OVERRUN: unused. 98 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 99 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 100 * - EVENT_NEW_IAUDIORECORD: unused. 101 */ 102 103 104 class IAudioRecordCallback : public virtual RefBase { 105 friend AudioRecord; 106 protected: 107 // Request for client to read newly available data. 108 // Used for TRANSFER_CALLBACK mode. 109 // Parameters: 110 // - buffer : Buffer to read from 111 // Returns: 112 // - Number of bytes actually consumed. onMoreData(const AudioRecord::Buffer & buffer)113 virtual size_t onMoreData([[maybe_unused]] const AudioRecord::Buffer& buffer) { return 0; } 114 // A buffer overrun occurred. onOverrun()115 virtual void onOverrun() {} 116 // Record head is at the specified marker (see setMarkerPosition()). onMarker(uint32_t markerPosition)117 virtual void onMarker([[maybe_unused]] uint32_t markerPosition) {} 118 // Record head is at a new position (see setPositionUpdatePeriod()). onNewPos(uint32_t newPos)119 virtual void onNewPos([[maybe_unused]] uint32_t newPos) {} 120 // IAudioRecord was recreated due to re-routing, server invalidation or 121 // server crash. onNewIAudioRecord()122 virtual void onNewIAudioRecord() {} 123 }; 124 125 /* Returns the minimum frame count required for the successful creation of 126 * an AudioRecord object. 127 * Returned status (from utils/Errors.h) can be: 128 * - NO_ERROR: successful operation 129 * - NO_INIT: audio server or audio hardware not initialized 130 * - BAD_VALUE: unsupported configuration 131 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 132 * and is undefined otherwise. 133 * FIXME This API assumes a route, and so should be deprecated. 134 */ 135 136 static status_t getMinFrameCount(size_t* frameCount, 137 uint32_t sampleRate, 138 audio_format_t format, 139 audio_channel_mask_t channelMask); 140 141 /* How data is transferred from AudioRecord 142 */ 143 enum transfer_type { 144 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 145 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 146 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 147 TRANSFER_SYNC, // synchronous read() 148 }; 149 150 /* Constructs an uninitialized AudioRecord. No connection with 151 * AudioFlinger takes place. Use set() after this. 152 * 153 * Parameters: 154 * 155 * client: The attribution source of the owner of the record 156 */ 157 AudioRecord(const android::content::AttributionSourceState& client); 158 159 /* Creates an AudioRecord object and registers it with AudioFlinger. 160 * Once created, the track needs to be started before it can be used. 161 * Unspecified values are set to appropriate default values. 162 * 163 * Parameters: 164 * 165 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 166 * sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate. 167 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 168 * 16 bits per sample). 169 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 170 * client: The attribution source of the owner of the record 171 * frameCount: Minimum size of track PCM buffer in frames. This defines the 172 * application's contribution to the 173 * latency of the track. The actual size selected by the AudioRecord could 174 * be larger if the requested size is not compatible with current audio HAL 175 * latency. Zero means to use a default value. 176 * cbf: Callback function. If not null, this function is called periodically 177 * to consume new data in TRANSFER_CALLBACK mode 178 * and inform of marker, position updates, etc. 179 * user: Context for use by the callback receiver. 180 * notificationFrames: The callback function is called each time notificationFrames PCM 181 * frames are ready in record track output buffer. 182 * sessionId: Not yet supported. 183 * transferType: How data is transferred from AudioRecord. 184 * flags: See comments on audio_input_flags_t in <system/audio.h> 185 * pAttributes: If not NULL, supersedes inputSource for use case selection. 186 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 187 */ 188 AudioRecord(audio_source_t inputSource, 189 uint32_t sampleRate, 190 audio_format_t format, 191 audio_channel_mask_t channelMask, 192 const android::content::AttributionSourceState& client, 193 size_t frameCount = 0, 194 const wp<IAudioRecordCallback> &callback = nullptr, 195 uint32_t notificationFrames = 0, 196 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 197 transfer_type transferType = TRANSFER_DEFAULT, 198 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 199 const audio_attributes_t* pAttributes = nullptr, 200 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, 201 audio_microphone_direction_t 202 selectedMicDirection = MIC_DIRECTION_UNSPECIFIED, 203 float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT); 204 205 206 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 207 * Also destroys all resources associated with the AudioRecord. 208 */ 209 protected: 210 virtual ~AudioRecord(); 211 public: 212 213 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 214 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 215 * set() is not multi-thread safe. 216 * Returned status (from utils/Errors.h) can be: 217 * - NO_ERROR: successful intialization 218 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 219 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 220 * - NO_INIT: audio server or audio hardware not initialized 221 * - PERMISSION_DENIED: recording is not allowed for the requesting process 222 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 223 * 224 * Parameters not listed in the AudioRecord constructors above: 225 * 226 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 227 */ 228 status_t set(audio_source_t inputSource, 229 uint32_t sampleRate, 230 audio_format_t format, 231 audio_channel_mask_t channelMask, 232 size_t frameCount = 0, 233 const wp<IAudioRecordCallback> &callback = nullptr, 234 uint32_t notificationFrames = 0, 235 bool threadCanCallJava = false, 236 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 237 transfer_type transferType = TRANSFER_DEFAULT, 238 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 239 uid_t uid = AUDIO_UID_INVALID, 240 pid_t pid = -1, 241 const audio_attributes_t* pAttributes = nullptr, 242 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, 243 audio_microphone_direction_t 244 selectedMicDirection = MIC_DIRECTION_UNSPECIFIED, 245 float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT, 246 int32_t maxSharedAudioHistoryMs = 0); 247 248 /* Result of constructing the AudioRecord. This must be checked for successful initialization 249 * before using any AudioRecord API (except for set()), because using 250 * an uninitialized AudioRecord produces undefined results. 251 * See set() method above for possible return codes. 252 */ initCheck()253 status_t initCheck() const { return mStatus; } 254 255 /* Returns this track's estimated latency in milliseconds. 256 * This includes the latency due to AudioRecord buffer size, resampling if applicable, 257 * and audio hardware driver. 258 */ latency()259 uint32_t latency() const { return mLatency; } 260 261 /* getters, see constructor and set() */ 262 format()263 audio_format_t format() const { return mFormat; } channelCount()264 uint32_t channelCount() const { return mChannelCount; } frameCount()265 size_t frameCount() const { return mFrameCount; } frameSize()266 size_t frameSize() const { return mFrameSize; } inputSource()267 audio_source_t inputSource() const { return mAttributes.source; } channelMask()268 audio_channel_mask_t channelMask() const { return mChannelMask; } 269 270 /* 271 * Return the period of the notification callback in frames. 272 * This value is set when the AudioRecord is constructed. 273 * It can be modified if the AudioRecord is rerouted. 274 */ getNotificationPeriodInFrames()275 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 276 277 /* 278 * return metrics information for the current instance. 279 */ 280 status_t getMetrics(mediametrics::Item * &item); 281 282 /* 283 * Set name of API that is using this object. 284 * For example "aaudio" or "opensles". 285 * This may be logged or reported as part of MediaMetrics. 286 */ setCallerName(const std::string & name)287 void setCallerName(const std::string &name) { 288 mCallerName = name; 289 } 290 getCallerName()291 std::string getCallerName() const { 292 return mCallerName; 293 }; 294 295 /* After it's created the track is not active. Call start() to 296 * make it active. If set, the callback will start being called. 297 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 298 * the specified event occurs on the specified trigger session. 299 */ 300 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 301 audio_session_t triggerSession = AUDIO_SESSION_NONE); 302 303 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 304 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 305 */ 306 void stop(); 307 bool stopped() const; 308 309 /* Calls stop() and then wait for all of the callbacks to return. 310 * It is safe to call this if stop() or pause() has already been called. 311 * 312 * This function is called from the destructor. But since AudioRecord 313 * is ref counted, the destructor may be called later than desired. 314 * This can be called explicitly as part of closing an AudioRecord 315 * if you want to be certain that callbacks have completely finished. 316 * 317 * This is not thread safe and should only be called from one thread, 318 * ideally as the AudioRecord is being closed. 319 */ 320 void stopAndJoinCallbacks(); 321 322 /* Return the sink sample rate for this record track in Hz. 323 * If specified as zero in constructor or set(), this will be the source sample rate. 324 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 325 */ getSampleRate()326 uint32_t getSampleRate() const { return mSampleRate; } 327 328 /* Return the sample rate from the AudioFlinger input thread. */ 329 uint32_t getHalSampleRate() const; 330 331 /* Return the channel count from the AudioFlinger input thread. */ 332 uint32_t getHalChannelCount() const; 333 334 /* Return the HAL format from the AudioFlinger input thread. */ 335 audio_format_t getHalFormat() const; 336 337 /* Sets marker position. When record reaches the number of frames specified, 338 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 339 * with marker == 0 cancels marker notification callback. 340 * To set a marker at a position which would compute as 0, 341 * a workaround is to set the marker at a nearby position such as ~0 or 1. 342 * If the AudioRecord has been opened with no callback function associated, 343 * the operation will fail. 344 * 345 * Parameters: 346 * 347 * marker: marker position expressed in wrapping (overflow) frame units, 348 * like the return value of getPosition(). 349 * 350 * Returned status (from utils/Errors.h) can be: 351 * - NO_ERROR: successful operation 352 * - INVALID_OPERATION: the AudioRecord has no callback installed. 353 */ 354 status_t setMarkerPosition(uint32_t marker); 355 status_t getMarkerPosition(uint32_t *marker) const; 356 357 /* Sets position update period. Every time the number of frames specified has been recorded, 358 * a callback with event type EVENT_NEW_POS is called. 359 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 360 * callback. 361 * If the AudioRecord has been opened with no callback function associated, 362 * the operation will fail. 363 * Extremely small values may be rounded up to a value the implementation can support. 364 * 365 * Parameters: 366 * 367 * updatePeriod: position update notification period expressed in frames. 368 * 369 * Returned status (from utils/Errors.h) can be: 370 * - NO_ERROR: successful operation 371 * - INVALID_OPERATION: the AudioRecord has no callback installed. 372 */ 373 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 374 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 375 376 /* Return the total number of frames recorded since recording started. 377 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 378 * It is reset to zero by stop(). 379 * 380 * Parameters: 381 * 382 * position: Address where to return record head position. 383 * 384 * Returned status (from utils/Errors.h) can be: 385 * - NO_ERROR: successful operation 386 * - BAD_VALUE: position is NULL 387 */ 388 status_t getPosition(uint32_t *position) const; 389 390 /* Return the record timestamp. 391 * 392 * Parameters: 393 * timestamp: A pointer to the timestamp to be filled. 394 * 395 * Returned status (from utils/Errors.h) can be: 396 * - NO_ERROR: successful operation 397 * - BAD_VALUE: timestamp is NULL 398 */ 399 status_t getTimestamp(ExtendedTimestamp *timestamp); 400 401 /** 402 * @param transferType 403 * @return text string that matches the enum name 404 */ 405 static const char * convertTransferToText(transfer_type transferType); 406 407 /* Returns a handle on the audio input used by this AudioRecord. 408 * 409 * Parameters: 410 * none. 411 * 412 * Returned value: 413 * handle on audio hardware input 414 */ 415 // FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp getInput()416 audio_io_handle_t getInput() const __attribute__((__deprecated__)) 417 { return getInputPrivate(); } 418 private: 419 audio_io_handle_t getInputPrivate() const; 420 public: 421 422 /* Returns the audio session ID associated with this AudioRecord. 423 * 424 * Parameters: 425 * none. 426 * 427 * Returned value: 428 * AudioRecord session ID. 429 * 430 * No lock needed because session ID doesn't change after first set(). 431 */ getSessionId()432 audio_session_t getSessionId() const { return mSessionId; } 433 434 /* Public API for TRANSFER_OBTAIN mode. 435 * Obtains a buffer of up to "audioBuffer->frameCount" full frames. 436 * After draining these frames of data, the caller should release them with releaseBuffer(). 437 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 438 * full frames as are available immediately. 439 * 440 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 441 * additional non-contiguous frames that are predicted to be available immediately, 442 * if the client were to release the first frames and then call obtainBuffer() again. 443 * This value is only a prediction, and needs to be confirmed. 444 * It will be set to zero for an error return. 445 * 446 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 447 * regardless of the value of waitCount. 448 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 449 * maximum timeout based on waitCount; see chart below. 450 * Buffers will be returned until the pool 451 * is exhausted, at which point obtainBuffer() will either block 452 * or return WOULD_BLOCK depending on the value of the "waitCount" 453 * parameter. 454 * 455 * Interpretation of waitCount: 456 * +n limits wait time to n * WAIT_PERIOD_MS, 457 * -1 causes an (almost) infinite wait time, 458 * 0 non-blocking. 459 * 460 * Buffer fields 461 * On entry: 462 * frameCount number of frames requested 463 * size ignored 464 * raw ignored 465 * sequence ignored 466 * After error return: 467 * frameCount 0 468 * size 0 469 * raw undefined 470 * sequence undefined 471 * After successful return: 472 * frameCount actual number of frames available, <= number requested 473 * size actual number of bytes available 474 * raw pointer to the buffer 475 * sequence IAudioRecord instance sequence number, as of obtainBuffer() 476 */ 477 478 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 479 size_t *nonContig = NULL); 480 481 // Explicit Routing 482 /** 483 * TODO Document this method. 484 */ 485 status_t setInputDevice(audio_port_handle_t deviceId); 486 487 /** 488 * TODO Document this method. 489 */ 490 audio_port_handle_t getInputDevice(); 491 492 /* Returns the ID of the audio device actually used by the input to which this AudioRecord 493 * is attached. 494 * The device ID is relevant only if the AudioRecord is active. 495 * When the AudioRecord is inactive, the device ID returned can be either: 496 * - AUDIO_PORT_HANDLE_NONE if the AudioRecord is not attached to any output. 497 * - The device ID used before paused or stopped. 498 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioRecord 499 * has not been started yet. 500 * 501 * Parameters: 502 * none. 503 */ 504 audio_port_handle_t getRoutedDeviceId(); 505 506 /* Add an AudioDeviceCallback. The caller will be notified when the audio device 507 * to which this AudioRecord is routed is updated. 508 * Replaces any previously installed callback. 509 * Parameters: 510 * callback: The callback interface 511 * Returns NO_ERROR if successful. 512 * INVALID_OPERATION if the same callback is already installed. 513 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 514 * BAD_VALUE if the callback is NULL 515 */ 516 status_t addAudioDeviceCallback( 517 const sp<AudioSystem::AudioDeviceCallback>& callback); 518 519 /* remove an AudioDeviceCallback. 520 * Parameters: 521 * callback: The callback interface 522 * Returns NO_ERROR if successful. 523 * INVALID_OPERATION if the callback is not installed 524 * BAD_VALUE if the callback is NULL 525 */ 526 status_t removeAudioDeviceCallback( 527 const sp<AudioSystem::AudioDeviceCallback>& callback); 528 529 // AudioSystem::AudioDeviceCallback> virtuals 530 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 531 audio_port_handle_t deviceId); 532 533 private: 534 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 535 * additional non-contiguous frames that are predicted to be available immediately, 536 * if the client were to release the first frames and then call obtainBuffer() again. 537 * This value is only a prediction, and needs to be confirmed. 538 * It will be set to zero for an error return. 539 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 540 * in case the requested amount of frames is in two or more non-contiguous regions. 541 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 542 */ 543 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 544 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 545 public: 546 547 /* Public API for TRANSFER_OBTAIN mode. 548 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. 549 * 550 * Buffer fields: 551 * frameCount currently ignored but recommend to set to actual number of frames consumed 552 * size actual number of bytes consumed, must be multiple of frameSize 553 * raw ignored 554 */ 555 void releaseBuffer(const Buffer* audioBuffer); 556 557 /* As a convenience we provide a read() interface to the audio buffer. 558 * Input parameter 'size' is in byte units. 559 * This is implemented on top of obtainBuffer/releaseBuffer. For best 560 * performance use callbacks. Returns actual number of bytes read >= 0, 561 * or one of the following negative status codes: 562 * INVALID_OPERATION AudioRecord is configured for streaming mode 563 * BAD_VALUE size is invalid 564 * WOULD_BLOCK when obtainBuffer() returns same, or 565 * AudioRecord was stopped during the read 566 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 567 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 568 * false for the method to return immediately without waiting to try multiple times to read 569 * the full content of the buffer. 570 */ 571 ssize_t read(void* buffer, size_t size, bool blocking = true); 572 573 /* Return the number of input frames lost in the audio driver since the last call of this 574 * function. Audio driver is expected to reset the value to 0 and restart counting upon 575 * returning the current value by this function call. Such loss typically occurs when the 576 * user space process is blocked longer than the capacity of audio driver buffers. 577 * Units: the number of input audio frames. 578 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 579 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 580 */ 581 uint32_t getInputFramesLost() const; 582 583 /* Get the flags */ getFlags()584 audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 585 586 /* Get active microphones. A empty vector of MicrophoneInfoFw will be passed as a parameter, 587 * the data will be filled when querying the hal. 588 */ 589 status_t getActiveMicrophones( 590 std::vector<media::MicrophoneInfoFw>* activeMicrophones); 591 592 /* Set the Microphone direction (for processing purposes). 593 */ 594 status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction); 595 596 /* Set the Microphone zoom factor (for processing purposes). 597 */ 598 status_t setPreferredMicrophoneFieldDimension(float zoom); 599 600 /* Get the unique port ID assigned to this AudioRecord instance by audio policy manager. 601 * The ID is unique across all audioserver clients and can change during the life cycle 602 * of a given AudioRecord instance if the connection to audioserver is restored. 603 */ getPortId()604 audio_port_handle_t getPortId() const { return mPortId; }; 605 606 /* Sets the LogSessionId field which is used for metrics association of 607 * this object with other objects. A nullptr or empty string clears 608 * the logSessionId. 609 */ 610 void setLogSessionId(const char *logSessionId); 611 612 613 status_t shareAudioHistory(const std::string& sharedPackageName, 614 int64_t sharedStartMs); 615 616 /* 617 * Dumps the state of an audio record. 618 */ 619 status_t dump(int fd, const Vector<String16>& args) const; 620 621 private: 622 /* copying audio record objects is not allowed */ 623 AudioRecord(const AudioRecord& other); 624 AudioRecord& operator = (const AudioRecord& other); 625 626 /* a small internal class to handle the callback */ 627 class AudioRecordThread : public Thread 628 { 629 public: 630 AudioRecordThread(AudioRecord& receiver); 631 632 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 633 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 634 virtual void requestExit(); 635 636 void pause(); // suspend thread from execution at next loop boundary 637 void resume(); // allow thread to execute, if not requested to exit 638 void wake(); // wake to handle changed notification conditions. 639 640 private: 641 void pauseInternal(nsecs_t ns = 0LL); 642 // like pause(), but only used internally within thread 643 644 friend class AudioRecord; 645 virtual bool threadLoop(); 646 AudioRecord& mReceiver; 647 virtual ~AudioRecordThread(); 648 Mutex mMyLock; // Thread::mLock is private 649 Condition mMyCond; // Thread::mThreadExitedCondition is private 650 bool mPaused; // whether thread is requested to pause at next loop entry 651 bool mPausedInt; // whether thread internally requests pause 652 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 653 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 654 // to processAudioBuffer() as state may have changed 655 // since pause time calculated. 656 }; 657 658 // body of AudioRecordThread::threadLoop() 659 // returns the maximum amount of time before we would like to run again, where: 660 // 0 immediately 661 // > 0 no later than this many nanoseconds from now 662 // NS_WHENEVER still active but no particular deadline 663 // NS_INACTIVE inactive so don't run again until re-started 664 // NS_NEVER never again 665 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 666 nsecs_t processAudioBuffer(); 667 668 // caller must hold lock on mLock for all _l methods 669 670 status_t createRecord_l(const Modulo<uint32_t> &epoch); 671 672 // FIXME enum is faster than strcmp() for parameter 'from' 673 status_t restoreRecord_l(const char *from); 674 675 void updateRoutedDeviceId_l(); 676 677 sp<AudioRecordThread> mAudioRecordThread; 678 mutable Mutex mLock; 679 680 std::unique_ptr<RecordingActivityTracker> mTracker; 681 682 // Current client state: false = stopped, true = active. Protected by mLock. If more states 683 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 684 bool mActive = false; 685 686 // for client callback handler 687 688 wp<IAudioRecordCallback> mCallback; 689 sp<IAudioRecordCallback> mLegacyCallbackWrapper; 690 691 bool mInitialized = false; // Protect against double set 692 // for notification APIs 693 uint32_t mNotificationFramesReq; // requested number of frames between each 694 // notification callback 695 // as specified in constructor or set() 696 uint32_t mNotificationFramesAct; // actual number of frames between each 697 // notification callback 698 bool mRefreshRemaining; // processAudioBuffer() should refresh 699 // mRemainingFrames and mRetryOnPartialBuffer 700 701 // These are private to processAudioBuffer(), and are not protected by a lock 702 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 703 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 704 uint32_t mObservedSequence; // last observed value of mSequence 705 706 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 707 bool mMarkerReached; 708 Modulo<uint32_t> mNewPosition; // in frames 709 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 710 711 status_t mStatus = NO_INIT; 712 713 android::content::AttributionSourceState mClientAttributionSource; // Owner's attribution source 714 715 size_t mFrameCount; // corresponds to current IAudioRecord, value is 716 // reported back by AudioFlinger to the client 717 size_t mReqFrameCount; // frame count to request the first or next time 718 // a new IAudioRecord is needed, non-decreasing 719 720 int64_t mFramesRead; // total frames read. reset to zero after 721 // the start() following stop(). It is not 722 // changed after restoring the track. 723 int64_t mFramesReadServerOffset; // An offset to server frames read due to 724 // restoring AudioRecord, or stop/start. 725 // constant after constructor or set() 726 uint32_t mSampleRate; 727 audio_format_t mFormat; 728 uint32_t mChannelCount; 729 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 730 uint32_t mLatency; // in ms 731 audio_channel_mask_t mChannelMask; 732 733 audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may 734 // be denied by client or server, such as 735 // AUDIO_INPUT_FLAG_FAST. mLock must be 736 // held to read or write those bits reliably. 737 audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const 738 739 audio_session_t mSessionId = AUDIO_SESSION_ALLOCATE; 740 audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE; 741 742 /** 743 * mLogSessionId is a string identifying this AudioRecord for the metrics service. 744 * It may be unique or shared with other objects. An empty string means the 745 * logSessionId is not set. 746 */ 747 std::string mLogSessionId{}; 748 749 transfer_type mTransfer; 750 751 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 752 // provided the initial set() was successful 753 sp<media::IAudioRecord> mAudioRecord; 754 sp<IMemory> mCblkMemory; 755 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 756 sp<IMemory> mBufferMemory; 757 audio_io_handle_t mInput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getInputforAttr() 758 759 int mPreviousPriority = ANDROID_PRIORITY_NORMAL; // before start() 760 SchedPolicy mPreviousSchedulingGroup = SP_DEFAULT; 761 bool mAwaitBoost = false; // thread should wait for priority boost before running 762 763 // The proxy should only be referenced while a lock is held because the proxy isn't 764 // multi-thread safe. 765 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 766 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 767 // them around in case they are replaced during the obtainBuffer(). 768 sp<AudioRecordClientProxy> mProxy; 769 770 bool mInOverrun; // whether recorder is currently in overrun state 771 772 ExtendedTimestamp mPreviousTimestamp{}; // used to detect retrograde motion 773 bool mTimestampRetrogradePositionReported = false; // reduce log spam 774 bool mTimestampRetrogradeTimeReported = false; // reduce log spam 775 776 // Format conversion. Maybe needed for adding fast tracks whose format is different from server. 777 audio_config_base_t mServerConfig; 778 size_t mServerFrameSize; 779 size_t mServerSampleSize; 780 std::unique_ptr<uint8_t[]> mFormatConversionBufRaw; 781 Buffer mFormatConversionBuffer; 782 uint32_t mHalSampleRate; // AudioFlinger thread sample rate 783 uint32_t mHalChannelCount; // AudioFlinger thread channel count 784 audio_format_t mHalFormat; // AudioFlinger thread format 785 786 private: 787 class DeathNotifier : public IBinder::DeathRecipient { 788 public: DeathNotifier(AudioRecord * audioRecord)789 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 790 protected: 791 virtual void binderDied(const wp<IBinder>& who); 792 private: 793 const wp<AudioRecord> mAudioRecord; 794 }; 795 796 sp<DeathNotifier> mDeathNotifier; 797 uint32_t mSequence; // incremented for each new IAudioRecord attempt 798 audio_attributes_t mAttributes; 799 800 // For Device Selection API 801 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 802 803 // Device requested by the application. 804 audio_port_handle_t mSelectedDeviceId = AUDIO_PORT_HANDLE_NONE; 805 // Device actually selected by AudioPolicyManager: This may not match the app 806 // selection depending on other activity and connected devices 807 audio_port_handle_t mRoutedDeviceId = AUDIO_PORT_HANDLE_NONE; 808 809 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 810 811 audio_microphone_direction_t mSelectedMicDirection = MIC_DIRECTION_UNSPECIFIED; 812 float mSelectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT; 813 814 int32_t mMaxSharedAudioHistoryMs = 0; 815 std::string mSharedAudioPackageName = {}; 816 int64_t mSharedAudioStartMs = 0; 817 818 private: 819 class MediaMetrics { 820 public: MediaMetrics()821 MediaMetrics() : mMetricsItem(mediametrics::Item::create("audiorecord")), 822 mCreatedNs(systemTime(SYSTEM_TIME_REALTIME)), 823 mStartedNs(0), mDurationNs(0), mCount(0), 824 mLastError(NO_ERROR) { 825 } ~MediaMetrics()826 ~MediaMetrics() { 827 // mMetricsItem alloc failure will be flagged in the constructor 828 // don't log empty records 829 if (mMetricsItem->count() > 0) { 830 mMetricsItem->selfrecord(); 831 } 832 } 833 void gather(const AudioRecord *record); dup()834 mediametrics::Item *dup() { return mMetricsItem->dup(); } 835 logStart(nsecs_t when)836 void logStart(nsecs_t when) { mStartedNs = when; mCount++; } logStop(nsecs_t when)837 void logStop(nsecs_t when) { mDurationNs += (when-mStartedNs); mStartedNs = 0;} markError(status_t errcode,const char * func)838 void markError(status_t errcode, const char *func) 839 { mLastError = errcode; mLastErrorFunc = func;} 840 private: 841 std::unique_ptr<mediametrics::Item> mMetricsItem; 842 nsecs_t mCreatedNs; // XXX: perhaps not worth it in production 843 nsecs_t mStartedNs; 844 nsecs_t mDurationNs; 845 int32_t mCount; 846 847 status_t mLastError; 848 std::string mLastErrorFunc; 849 }; 850 MediaMetrics mMediaMetrics; 851 std::string mMetricsId; // GUARDED_BY(mLock), could change in createRecord_l(). 852 std::string mCallerName; // for example "aaudio" 853 854 void reportError(status_t status, const char *event, const char *message) const; 855 }; 856 857 }; // namespace android 858 859 #endif // ANDROID_AUDIORECORD_H 860