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 <cutils/sched_policy.h> 21 #include <media/AudioSystem.h> 22 #include <media/AudioTimestamp.h> 23 #include <media/IAudioRecord.h> 24 #include <media/Modulo.h> 25 #include <utils/threads.h> 26 27 namespace android { 28 29 // ---------------------------------------------------------------------------- 30 31 struct audio_track_cblk_t; 32 class AudioRecordClientProxy; 33 34 // ---------------------------------------------------------------------------- 35 36 class AudioRecord : public AudioSystem::AudioDeviceCallback 37 { 38 public: 39 40 /* Events used by AudioRecord callback function (callback_t). 41 * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*. 42 */ 43 enum event_type { 44 EVENT_MORE_DATA = 0, // Request to read available data from buffer. 45 // If this event is delivered but the callback handler 46 // does not want to read the available data, the handler must 47 // explicitly ignore the event by setting frameCount to zero. 48 EVENT_OVERRUN = 1, // Buffer overrun occurred. 49 EVENT_MARKER = 2, // Record head is at the specified marker position 50 // (See setMarkerPosition()). 51 EVENT_NEW_POS = 3, // Record head is at a new position 52 // (See setPositionUpdatePeriod()). 53 EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and 54 // voluntary invalidation by mediaserver, or mediaserver crash. 55 }; 56 57 /* Client should declare a Buffer and pass address to obtainBuffer() 58 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 59 */ 60 61 class Buffer 62 { 63 public: 64 // FIXME use m prefix 65 size_t frameCount; // number of sample frames corresponding to size; 66 // on input to obtainBuffer() it is the number of frames desired 67 // on output from obtainBuffer() it is the number of available 68 // frames to be read 69 // on input to releaseBuffer() it is currently ignored 70 71 size_t size; // input/output in bytes == frameCount * frameSize 72 // on input to obtainBuffer() it is ignored 73 // on output from obtainBuffer() it is the number of available 74 // bytes to be read, which is frameCount * frameSize 75 // on input to releaseBuffer() it is the number of bytes to 76 // release 77 // FIXME This is redundant with respect to frameCount. Consider 78 // removing size and making frameCount the primary field. 79 80 union { 81 void* raw; 82 short* i16; // signed 16-bit 83 int8_t* i8; // unsigned 8-bit, offset by 0x80 84 // input to obtainBuffer(): unused, output: pointer to buffer 85 }; 86 }; 87 88 /* As a convenience, if a callback is supplied, a handler thread 89 * is automatically created with the appropriate priority. This thread 90 * invokes the callback when a new buffer becomes available or various conditions occur. 91 * Parameters: 92 * 93 * event: type of event notified (see enum AudioRecord::event_type). 94 * user: Pointer to context for use by the callback receiver. 95 * info: Pointer to optional parameter according to event type: 96 * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read 97 * more bytes than indicated by 'size' field and update 'size' if 98 * fewer bytes are consumed. 99 * - EVENT_OVERRUN: unused. 100 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 101 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 102 * - EVENT_NEW_IAUDIORECORD: unused. 103 */ 104 105 typedef void (*callback_t)(int event, void* user, void *info); 106 107 /* Returns the minimum frame count required for the successful creation of 108 * an AudioRecord object. 109 * Returned status (from utils/Errors.h) can be: 110 * - NO_ERROR: successful operation 111 * - NO_INIT: audio server or audio hardware not initialized 112 * - BAD_VALUE: unsupported configuration 113 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 114 * and is undefined otherwise. 115 * FIXME This API assumes a route, and so should be deprecated. 116 */ 117 118 static status_t getMinFrameCount(size_t* frameCount, 119 uint32_t sampleRate, 120 audio_format_t format, 121 audio_channel_mask_t channelMask); 122 123 /* How data is transferred from AudioRecord 124 */ 125 enum transfer_type { 126 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 127 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 128 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 129 TRANSFER_SYNC, // synchronous read() 130 }; 131 132 /* Constructs an uninitialized AudioRecord. No connection with 133 * AudioFlinger takes place. Use set() after this. 134 * 135 * Parameters: 136 * 137 * opPackageName: The package name used for app ops. 138 */ 139 AudioRecord(const String16& opPackageName); 140 141 /* Creates an AudioRecord object and registers it with AudioFlinger. 142 * Once created, the track needs to be started before it can be used. 143 * Unspecified values are set to appropriate default values. 144 * 145 * Parameters: 146 * 147 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 148 * sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate. 149 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 150 * 16 bits per sample). 151 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 152 * opPackageName: The package name used for app ops. 153 * frameCount: Minimum size of track PCM buffer in frames. This defines the 154 * application's contribution to the 155 * latency of the track. The actual size selected by the AudioRecord could 156 * be larger if the requested size is not compatible with current audio HAL 157 * latency. Zero means to use a default value. 158 * cbf: Callback function. If not null, this function is called periodically 159 * to consume new data in TRANSFER_CALLBACK mode 160 * and inform of marker, position updates, etc. 161 * user: Context for use by the callback receiver. 162 * notificationFrames: The callback function is called each time notificationFrames PCM 163 * frames are ready in record track output buffer. 164 * sessionId: Not yet supported. 165 * transferType: How data is transferred from AudioRecord. 166 * flags: See comments on audio_input_flags_t in <system/audio.h> 167 * pAttributes: If not NULL, supersedes inputSource for use case selection. 168 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 169 */ 170 171 AudioRecord(audio_source_t inputSource, 172 uint32_t sampleRate, 173 audio_format_t format, 174 audio_channel_mask_t channelMask, 175 const String16& opPackageName, 176 size_t frameCount = 0, 177 callback_t cbf = NULL, 178 void* user = NULL, 179 uint32_t notificationFrames = 0, 180 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 181 transfer_type transferType = TRANSFER_DEFAULT, 182 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 183 uid_t uid = AUDIO_UID_INVALID, 184 pid_t pid = -1, 185 const audio_attributes_t* pAttributes = NULL); 186 187 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 188 * Also destroys all resources associated with the AudioRecord. 189 */ 190 protected: 191 virtual ~AudioRecord(); 192 public: 193 194 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 195 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 196 * set() is not multi-thread safe. 197 * Returned status (from utils/Errors.h) can be: 198 * - NO_ERROR: successful intialization 199 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 200 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 201 * - NO_INIT: audio server or audio hardware not initialized 202 * - PERMISSION_DENIED: recording is not allowed for the requesting process 203 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 204 * 205 * Parameters not listed in the AudioRecord constructors above: 206 * 207 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 208 */ 209 status_t set(audio_source_t inputSource, 210 uint32_t sampleRate, 211 audio_format_t format, 212 audio_channel_mask_t channelMask, 213 size_t frameCount = 0, 214 callback_t cbf = NULL, 215 void* user = NULL, 216 uint32_t notificationFrames = 0, 217 bool threadCanCallJava = false, 218 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 219 transfer_type transferType = TRANSFER_DEFAULT, 220 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 221 uid_t uid = AUDIO_UID_INVALID, 222 pid_t pid = -1, 223 const audio_attributes_t* pAttributes = NULL); 224 225 /* Result of constructing the AudioRecord. This must be checked for successful initialization 226 * before using any AudioRecord API (except for set()), because using 227 * an uninitialized AudioRecord produces undefined results. 228 * See set() method above for possible return codes. 229 */ 230 status_t initCheck() const { return mStatus; } 231 232 /* Returns this track's estimated latency in milliseconds. 233 * This includes the latency due to AudioRecord buffer size, resampling if applicable, 234 * and audio hardware driver. 235 */ 236 uint32_t latency() const { return mLatency; } 237 238 /* getters, see constructor and set() */ 239 240 audio_format_t format() const { return mFormat; } 241 uint32_t channelCount() const { return mChannelCount; } 242 size_t frameCount() const { return mFrameCount; } 243 size_t frameSize() const { return mFrameSize; } 244 audio_source_t inputSource() const { return mAttributes.source; } 245 246 /* 247 * Return the period of the notification callback in frames. 248 * This value is set when the AudioRecord is constructed. 249 * It can be modified if the AudioRecord is rerouted. 250 */ 251 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 252 253 /* After it's created the track is not active. Call start() to 254 * make it active. If set, the callback will start being called. 255 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 256 * the specified event occurs on the specified trigger session. 257 */ 258 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 259 audio_session_t triggerSession = AUDIO_SESSION_NONE); 260 261 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 262 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 263 */ 264 void stop(); 265 bool stopped() const; 266 267 /* Return the sink sample rate for this record track in Hz. 268 * If specified as zero in constructor or set(), this will be the source sample rate. 269 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 270 */ 271 uint32_t getSampleRate() const { return mSampleRate; } 272 273 /* Sets marker position. When record reaches the number of frames specified, 274 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 275 * with marker == 0 cancels marker notification callback. 276 * To set a marker at a position which would compute as 0, 277 * a workaround is to set the marker at a nearby position such as ~0 or 1. 278 * If the AudioRecord has been opened with no callback function associated, 279 * the operation will fail. 280 * 281 * Parameters: 282 * 283 * marker: marker position expressed in wrapping (overflow) frame units, 284 * like the return value of getPosition(). 285 * 286 * Returned status (from utils/Errors.h) can be: 287 * - NO_ERROR: successful operation 288 * - INVALID_OPERATION: the AudioRecord has no callback installed. 289 */ 290 status_t setMarkerPosition(uint32_t marker); 291 status_t getMarkerPosition(uint32_t *marker) const; 292 293 /* Sets position update period. Every time the number of frames specified has been recorded, 294 * a callback with event type EVENT_NEW_POS is called. 295 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 296 * callback. 297 * If the AudioRecord has been opened with no callback function associated, 298 * the operation will fail. 299 * Extremely small values may be rounded up to a value the implementation can support. 300 * 301 * Parameters: 302 * 303 * updatePeriod: position update notification period expressed in frames. 304 * 305 * Returned status (from utils/Errors.h) can be: 306 * - NO_ERROR: successful operation 307 * - INVALID_OPERATION: the AudioRecord has no callback installed. 308 */ 309 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 310 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 311 312 /* Return the total number of frames recorded since recording started. 313 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 314 * It is reset to zero by stop(). 315 * 316 * Parameters: 317 * 318 * position: Address where to return record head position. 319 * 320 * Returned status (from utils/Errors.h) can be: 321 * - NO_ERROR: successful operation 322 * - BAD_VALUE: position is NULL 323 */ 324 status_t getPosition(uint32_t *position) const; 325 326 /* Return the record timestamp. 327 * 328 * Parameters: 329 * timestamp: A pointer to the timestamp to be filled. 330 * 331 * Returned status (from utils/Errors.h) can be: 332 * - NO_ERROR: successful operation 333 * - BAD_VALUE: timestamp is NULL 334 */ 335 status_t getTimestamp(ExtendedTimestamp *timestamp); 336 337 /** 338 * @param transferType 339 * @return text string that matches the enum name 340 */ 341 static const char * convertTransferToText(transfer_type transferType); 342 343 /* Returns a handle on the audio input used by this AudioRecord. 344 * 345 * Parameters: 346 * none. 347 * 348 * Returned value: 349 * handle on audio hardware input 350 */ 351 // FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp 352 audio_io_handle_t getInput() const __attribute__((__deprecated__)) 353 { return getInputPrivate(); } 354 private: 355 audio_io_handle_t getInputPrivate() const; 356 public: 357 358 /* Returns the audio session ID associated with this AudioRecord. 359 * 360 * Parameters: 361 * none. 362 * 363 * Returned value: 364 * AudioRecord session ID. 365 * 366 * No lock needed because session ID doesn't change after first set(). 367 */ 368 audio_session_t getSessionId() const { return mSessionId; } 369 370 /* Public API for TRANSFER_OBTAIN mode. 371 * Obtains a buffer of up to "audioBuffer->frameCount" full frames. 372 * After draining these frames of data, the caller should release them with releaseBuffer(). 373 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 374 * full frames as are available immediately. 375 * 376 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 377 * additional non-contiguous frames that are predicted to be available immediately, 378 * if the client were to release the first frames and then call obtainBuffer() again. 379 * This value is only a prediction, and needs to be confirmed. 380 * It will be set to zero for an error return. 381 * 382 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 383 * regardless of the value of waitCount. 384 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 385 * maximum timeout based on waitCount; see chart below. 386 * Buffers will be returned until the pool 387 * is exhausted, at which point obtainBuffer() will either block 388 * or return WOULD_BLOCK depending on the value of the "waitCount" 389 * parameter. 390 * 391 * Interpretation of waitCount: 392 * +n limits wait time to n * WAIT_PERIOD_MS, 393 * -1 causes an (almost) infinite wait time, 394 * 0 non-blocking. 395 * 396 * Buffer fields 397 * On entry: 398 * frameCount number of frames requested 399 * size ignored 400 * raw ignored 401 * After error return: 402 * frameCount 0 403 * size 0 404 * raw undefined 405 * After successful return: 406 * frameCount actual number of frames available, <= number requested 407 * size actual number of bytes available 408 * raw pointer to the buffer 409 */ 410 411 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 412 size_t *nonContig = NULL); 413 414 // Explicit Routing 415 /** 416 * TODO Document this method. 417 */ 418 status_t setInputDevice(audio_port_handle_t deviceId); 419 420 /** 421 * TODO Document this method. 422 */ 423 audio_port_handle_t getInputDevice(); 424 425 /* Returns the ID of the audio device actually used by the input to which this AudioRecord 426 * is attached. 427 * The device ID is relevant only if the AudioRecord is active. 428 * When the AudioRecord is inactive, the device ID returned can be either: 429 * - AUDIO_PORT_HANDLE_NONE if the AudioRecord is not attached to any output. 430 * - The device ID used before paused or stopped. 431 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioRecord 432 * has not been started yet. 433 * 434 * Parameters: 435 * none. 436 */ 437 audio_port_handle_t getRoutedDeviceId(); 438 439 /* Add an AudioDeviceCallback. The caller will be notified when the audio device 440 * to which this AudioRecord is routed is updated. 441 * Replaces any previously installed callback. 442 * Parameters: 443 * callback: The callback interface 444 * Returns NO_ERROR if successful. 445 * INVALID_OPERATION if the same callback is already installed. 446 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 447 * BAD_VALUE if the callback is NULL 448 */ 449 status_t addAudioDeviceCallback( 450 const sp<AudioSystem::AudioDeviceCallback>& callback); 451 452 /* remove an AudioDeviceCallback. 453 * Parameters: 454 * callback: The callback interface 455 * Returns NO_ERROR if successful. 456 * INVALID_OPERATION if the callback is not installed 457 * BAD_VALUE if the callback is NULL 458 */ 459 status_t removeAudioDeviceCallback( 460 const sp<AudioSystem::AudioDeviceCallback>& callback); 461 462 // AudioSystem::AudioDeviceCallback> virtuals 463 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 464 audio_port_handle_t deviceId); 465 466 private: 467 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 468 * additional non-contiguous frames that are predicted to be available immediately, 469 * if the client were to release the first frames and then call obtainBuffer() again. 470 * This value is only a prediction, and needs to be confirmed. 471 * It will be set to zero for an error return. 472 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 473 * in case the requested amount of frames is in two or more non-contiguous regions. 474 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 475 */ 476 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 477 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 478 public: 479 480 /* Public API for TRANSFER_OBTAIN mode. 481 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. 482 * 483 * Buffer fields: 484 * frameCount currently ignored but recommend to set to actual number of frames consumed 485 * size actual number of bytes consumed, must be multiple of frameSize 486 * raw ignored 487 */ 488 void releaseBuffer(const Buffer* audioBuffer); 489 490 /* As a convenience we provide a read() interface to the audio buffer. 491 * Input parameter 'size' is in byte units. 492 * This is implemented on top of obtainBuffer/releaseBuffer. For best 493 * performance use callbacks. Returns actual number of bytes read >= 0, 494 * or one of the following negative status codes: 495 * INVALID_OPERATION AudioRecord is configured for streaming mode 496 * BAD_VALUE size is invalid 497 * WOULD_BLOCK when obtainBuffer() returns same, or 498 * AudioRecord was stopped during the read 499 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 500 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 501 * false for the method to return immediately without waiting to try multiple times to read 502 * the full content of the buffer. 503 */ 504 ssize_t read(void* buffer, size_t size, bool blocking = true); 505 506 /* Return the number of input frames lost in the audio driver since the last call of this 507 * function. Audio driver is expected to reset the value to 0 and restart counting upon 508 * returning the current value by this function call. Such loss typically occurs when the 509 * user space process is blocked longer than the capacity of audio driver buffers. 510 * Units: the number of input audio frames. 511 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 512 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 513 */ 514 uint32_t getInputFramesLost() const; 515 516 /* Get the flags */ 517 audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 518 519 private: 520 /* copying audio record objects is not allowed */ 521 AudioRecord(const AudioRecord& other); 522 AudioRecord& operator = (const AudioRecord& other); 523 524 /* a small internal class to handle the callback */ 525 class AudioRecordThread : public Thread 526 { 527 public: 528 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 529 530 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 531 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 532 virtual void requestExit(); 533 534 void pause(); // suspend thread from execution at next loop boundary 535 void resume(); // allow thread to execute, if not requested to exit 536 void wake(); // wake to handle changed notification conditions. 537 538 private: 539 void pauseInternal(nsecs_t ns = 0LL); 540 // like pause(), but only used internally within thread 541 542 friend class AudioRecord; 543 virtual bool threadLoop(); 544 AudioRecord& mReceiver; 545 virtual ~AudioRecordThread(); 546 Mutex mMyLock; // Thread::mLock is private 547 Condition mMyCond; // Thread::mThreadExitedCondition is private 548 bool mPaused; // whether thread is requested to pause at next loop entry 549 bool mPausedInt; // whether thread internally requests pause 550 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 551 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 552 // to processAudioBuffer() as state may have changed 553 // since pause time calculated. 554 }; 555 556 // body of AudioRecordThread::threadLoop() 557 // returns the maximum amount of time before we would like to run again, where: 558 // 0 immediately 559 // > 0 no later than this many nanoseconds from now 560 // NS_WHENEVER still active but no particular deadline 561 // NS_INACTIVE inactive so don't run again until re-started 562 // NS_NEVER never again 563 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 564 nsecs_t processAudioBuffer(); 565 566 // caller must hold lock on mLock for all _l methods 567 568 status_t openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName); 569 570 // FIXME enum is faster than strcmp() for parameter 'from' 571 status_t restoreRecord_l(const char *from); 572 573 void updateRoutedDeviceId_l(); 574 575 sp<AudioRecordThread> mAudioRecordThread; 576 mutable Mutex mLock; 577 578 // Current client state: false = stopped, true = active. Protected by mLock. If more states 579 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 580 bool mActive; 581 582 // for client callback handler 583 callback_t mCbf; // callback handler for events, or NULL 584 void* mUserData; 585 586 // for notification APIs 587 uint32_t mNotificationFramesReq; // requested number of frames between each 588 // notification callback 589 // as specified in constructor or set() 590 uint32_t mNotificationFramesAct; // actual number of frames between each 591 // notification callback 592 bool mRefreshRemaining; // processAudioBuffer() should refresh 593 // mRemainingFrames and mRetryOnPartialBuffer 594 595 // These are private to processAudioBuffer(), and are not protected by a lock 596 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 597 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 598 uint32_t mObservedSequence; // last observed value of mSequence 599 600 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 601 bool mMarkerReached; 602 Modulo<uint32_t> mNewPosition; // in frames 603 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 604 605 status_t mStatus; 606 607 String16 mOpPackageName; // The package name used for app ops. 608 609 size_t mFrameCount; // corresponds to current IAudioRecord, value is 610 // reported back by AudioFlinger to the client 611 size_t mReqFrameCount; // frame count to request the first or next time 612 // a new IAudioRecord is needed, non-decreasing 613 614 int64_t mFramesRead; // total frames read. reset to zero after 615 // the start() following stop(). It is not 616 // changed after restoring the track. 617 int64_t mFramesReadServerOffset; // An offset to server frames read due to 618 // restoring AudioRecord, or stop/start. 619 // constant after constructor or set() 620 uint32_t mSampleRate; 621 audio_format_t mFormat; 622 uint32_t mChannelCount; 623 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 624 uint32_t mLatency; // in ms 625 audio_channel_mask_t mChannelMask; 626 627 audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may 628 // be denied by client or server, such as 629 // AUDIO_INPUT_FLAG_FAST. mLock must be 630 // held to read or write those bits reliably. 631 audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const 632 633 audio_session_t mSessionId; 634 transfer_type mTransfer; 635 636 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 637 // provided the initial set() was successful 638 sp<IAudioRecord> mAudioRecord; 639 sp<IMemory> mCblkMemory; 640 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 641 sp<IMemory> mBufferMemory; 642 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 643 644 int mPreviousPriority; // before start() 645 SchedPolicy mPreviousSchedulingGroup; 646 bool mAwaitBoost; // thread should wait for priority boost before running 647 648 // The proxy should only be referenced while a lock is held because the proxy isn't 649 // multi-thread safe. 650 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 651 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 652 // them around in case they are replaced during the obtainBuffer(). 653 sp<AudioRecordClientProxy> mProxy; 654 655 bool mInOverrun; // whether recorder is currently in overrun state 656 657 private: 658 class DeathNotifier : public IBinder::DeathRecipient { 659 public: 660 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 661 protected: 662 virtual void binderDied(const wp<IBinder>& who); 663 private: 664 const wp<AudioRecord> mAudioRecord; 665 }; 666 667 sp<DeathNotifier> mDeathNotifier; 668 uint32_t mSequence; // incremented for each new IAudioRecord attempt 669 uid_t mClientUid; 670 pid_t mClientPid; 671 audio_attributes_t mAttributes; 672 673 // For Device Selection API 674 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 675 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 676 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 677 // May not match the app selection depending on other 678 // activity and connected devices 679 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 680 audio_port_handle_t mPortId; // unique ID allocated by audio policy 681 682 }; 683 684 }; // namespace android 685 686 #endif // ANDROID_AUDIORECORD_H 687
