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 RefBase 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 int uid = -1, 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 int uid = -1, 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 /* After it's created the track is not active. Call start() to 247 * make it active. If set, the callback will start being called. 248 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 249 * the specified event occurs on the specified trigger session. 250 */ 251 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 252 audio_session_t triggerSession = AUDIO_SESSION_NONE); 253 254 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 255 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 256 */ 257 void stop(); 258 bool stopped() const; 259 260 /* Return the sink sample rate for this record track in Hz. 261 * If specified as zero in constructor or set(), this will be the source sample rate. 262 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 263 */ 264 uint32_t getSampleRate() const { return mSampleRate; } 265 266 /* Sets marker position. When record reaches the number of frames specified, 267 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 268 * with marker == 0 cancels marker notification callback. 269 * To set a marker at a position which would compute as 0, 270 * a workaround is to set the marker at a nearby position such as ~0 or 1. 271 * If the AudioRecord has been opened with no callback function associated, 272 * the operation will fail. 273 * 274 * Parameters: 275 * 276 * marker: marker position expressed in wrapping (overflow) frame units, 277 * like the return value of getPosition(). 278 * 279 * Returned status (from utils/Errors.h) can be: 280 * - NO_ERROR: successful operation 281 * - INVALID_OPERATION: the AudioRecord has no callback installed. 282 */ 283 status_t setMarkerPosition(uint32_t marker); 284 status_t getMarkerPosition(uint32_t *marker) const; 285 286 /* Sets position update period. Every time the number of frames specified has been recorded, 287 * a callback with event type EVENT_NEW_POS is called. 288 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 289 * callback. 290 * If the AudioRecord has been opened with no callback function associated, 291 * the operation will fail. 292 * Extremely small values may be rounded up to a value the implementation can support. 293 * 294 * Parameters: 295 * 296 * updatePeriod: position update notification period expressed in frames. 297 * 298 * Returned status (from utils/Errors.h) can be: 299 * - NO_ERROR: successful operation 300 * - INVALID_OPERATION: the AudioRecord has no callback installed. 301 */ 302 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 303 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 304 305 /* Return the total number of frames recorded since recording started. 306 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 307 * It is reset to zero by stop(). 308 * 309 * Parameters: 310 * 311 * position: Address where to return record head position. 312 * 313 * Returned status (from utils/Errors.h) can be: 314 * - NO_ERROR: successful operation 315 * - BAD_VALUE: position is NULL 316 */ 317 status_t getPosition(uint32_t *position) const; 318 319 /* Return the record timestamp. 320 * 321 * Parameters: 322 * timestamp: A pointer to the timestamp to be filled. 323 * 324 * Returned status (from utils/Errors.h) can be: 325 * - NO_ERROR: successful operation 326 * - BAD_VALUE: timestamp is NULL 327 */ 328 status_t getTimestamp(ExtendedTimestamp *timestamp); 329 330 /* Returns a handle on the audio input used by this AudioRecord. 331 * 332 * Parameters: 333 * none. 334 * 335 * Returned value: 336 * handle on audio hardware input 337 */ 338 // FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp 339 audio_io_handle_t getInput() const __attribute__((__deprecated__)) 340 { return getInputPrivate(); } 341 private: 342 audio_io_handle_t getInputPrivate() const; 343 public: 344 345 /* Returns the audio session ID associated with this AudioRecord. 346 * 347 * Parameters: 348 * none. 349 * 350 * Returned value: 351 * AudioRecord session ID. 352 * 353 * No lock needed because session ID doesn't change after first set(). 354 */ 355 audio_session_t getSessionId() const { return mSessionId; } 356 357 /* Public API for TRANSFER_OBTAIN mode. 358 * Obtains a buffer of up to "audioBuffer->frameCount" full frames. 359 * After draining these frames of data, the caller should release them with releaseBuffer(). 360 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 361 * full frames as are available immediately. 362 * 363 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 364 * additional non-contiguous frames that are predicted to be available immediately, 365 * if the client were to release the first frames and then call obtainBuffer() again. 366 * This value is only a prediction, and needs to be confirmed. 367 * It will be set to zero for an error return. 368 * 369 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 370 * regardless of the value of waitCount. 371 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 372 * maximum timeout based on waitCount; see chart below. 373 * Buffers will be returned until the pool 374 * is exhausted, at which point obtainBuffer() will either block 375 * or return WOULD_BLOCK depending on the value of the "waitCount" 376 * parameter. 377 * 378 * Interpretation of waitCount: 379 * +n limits wait time to n * WAIT_PERIOD_MS, 380 * -1 causes an (almost) infinite wait time, 381 * 0 non-blocking. 382 * 383 * Buffer fields 384 * On entry: 385 * frameCount number of frames requested 386 * size ignored 387 * raw ignored 388 * After error return: 389 * frameCount 0 390 * size 0 391 * raw undefined 392 * After successful return: 393 * frameCount actual number of frames available, <= number requested 394 * size actual number of bytes available 395 * raw pointer to the buffer 396 */ 397 398 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 399 size_t *nonContig = NULL); 400 401 // Explicit Routing 402 /** 403 * TODO Document this method. 404 */ 405 status_t setInputDevice(audio_port_handle_t deviceId); 406 407 /** 408 * TODO Document this method. 409 */ 410 audio_port_handle_t getInputDevice(); 411 412 /* Returns the ID of the audio device actually used by the input to which this AudioRecord 413 * is attached. 414 * A value of AUDIO_PORT_HANDLE_NONE indicates the AudioRecord is not attached to any input. 415 * 416 * Parameters: 417 * none. 418 */ 419 audio_port_handle_t getRoutedDeviceId(); 420 421 /* Add an AudioDeviceCallback. The caller will be notified when the audio device 422 * to which this AudioRecord is routed is updated. 423 * Replaces any previously installed callback. 424 * Parameters: 425 * callback: The callback interface 426 * Returns NO_ERROR if successful. 427 * INVALID_OPERATION if the same callback is already installed. 428 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 429 * BAD_VALUE if the callback is NULL 430 */ 431 status_t addAudioDeviceCallback( 432 const sp<AudioSystem::AudioDeviceCallback>& callback); 433 434 /* remove an AudioDeviceCallback. 435 * Parameters: 436 * callback: The callback interface 437 * Returns NO_ERROR if successful. 438 * INVALID_OPERATION if the callback is not installed 439 * BAD_VALUE if the callback is NULL 440 */ 441 status_t removeAudioDeviceCallback( 442 const sp<AudioSystem::AudioDeviceCallback>& callback); 443 444 private: 445 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 446 * additional non-contiguous frames that are predicted to be available immediately, 447 * if the client were to release the first frames and then call obtainBuffer() again. 448 * This value is only a prediction, and needs to be confirmed. 449 * It will be set to zero for an error return. 450 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 451 * in case the requested amount of frames is in two or more non-contiguous regions. 452 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 453 */ 454 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 455 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 456 public: 457 458 /* Public API for TRANSFER_OBTAIN mode. 459 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. 460 * 461 * Buffer fields: 462 * frameCount currently ignored but recommend to set to actual number of frames consumed 463 * size actual number of bytes consumed, must be multiple of frameSize 464 * raw ignored 465 */ 466 void releaseBuffer(const Buffer* audioBuffer); 467 468 /* As a convenience we provide a read() interface to the audio buffer. 469 * Input parameter 'size' is in byte units. 470 * This is implemented on top of obtainBuffer/releaseBuffer. For best 471 * performance use callbacks. Returns actual number of bytes read >= 0, 472 * or one of the following negative status codes: 473 * INVALID_OPERATION AudioRecord is configured for streaming mode 474 * BAD_VALUE size is invalid 475 * WOULD_BLOCK when obtainBuffer() returns same, or 476 * AudioRecord was stopped during the read 477 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 478 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 479 * false for the method to return immediately without waiting to try multiple times to read 480 * the full content of the buffer. 481 */ 482 ssize_t read(void* buffer, size_t size, bool blocking = true); 483 484 /* Return the number of input frames lost in the audio driver since the last call of this 485 * function. Audio driver is expected to reset the value to 0 and restart counting upon 486 * returning the current value by this function call. Such loss typically occurs when the 487 * user space process is blocked longer than the capacity of audio driver buffers. 488 * Units: the number of input audio frames. 489 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 490 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 491 */ 492 uint32_t getInputFramesLost() const; 493 494 /* Get the flags */ 495 audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 496 497 private: 498 /* copying audio record objects is not allowed */ 499 AudioRecord(const AudioRecord& other); 500 AudioRecord& operator = (const AudioRecord& other); 501 502 /* a small internal class to handle the callback */ 503 class AudioRecordThread : public Thread 504 { 505 public: 506 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 507 508 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 509 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 510 virtual void requestExit(); 511 512 void pause(); // suspend thread from execution at next loop boundary 513 void resume(); // allow thread to execute, if not requested to exit 514 void wake(); // wake to handle changed notification conditions. 515 516 private: 517 void pauseInternal(nsecs_t ns = 0LL); 518 // like pause(), but only used internally within thread 519 520 friend class AudioRecord; 521 virtual bool threadLoop(); 522 AudioRecord& mReceiver; 523 virtual ~AudioRecordThread(); 524 Mutex mMyLock; // Thread::mLock is private 525 Condition mMyCond; // Thread::mThreadExitedCondition is private 526 bool mPaused; // whether thread is requested to pause at next loop entry 527 bool mPausedInt; // whether thread internally requests pause 528 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 529 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 530 // to processAudioBuffer() as state may have changed 531 // since pause time calculated. 532 }; 533 534 // body of AudioRecordThread::threadLoop() 535 // returns the maximum amount of time before we would like to run again, where: 536 // 0 immediately 537 // > 0 no later than this many nanoseconds from now 538 // NS_WHENEVER still active but no particular deadline 539 // NS_INACTIVE inactive so don't run again until re-started 540 // NS_NEVER never again 541 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 542 nsecs_t processAudioBuffer(); 543 544 // caller must hold lock on mLock for all _l methods 545 546 status_t openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName); 547 548 // FIXME enum is faster than strcmp() for parameter 'from' 549 status_t restoreRecord_l(const char *from); 550 551 sp<AudioRecordThread> mAudioRecordThread; 552 mutable Mutex mLock; 553 554 // Current client state: false = stopped, true = active. Protected by mLock. If more states 555 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 556 bool mActive; 557 558 // for client callback handler 559 callback_t mCbf; // callback handler for events, or NULL 560 void* mUserData; 561 562 // for notification APIs 563 uint32_t mNotificationFramesReq; // requested number of frames between each 564 // notification callback 565 // as specified in constructor or set() 566 uint32_t mNotificationFramesAct; // actual number of frames between each 567 // notification callback 568 bool mRefreshRemaining; // processAudioBuffer() should refresh 569 // mRemainingFrames and mRetryOnPartialBuffer 570 571 // These are private to processAudioBuffer(), and are not protected by a lock 572 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 573 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 574 uint32_t mObservedSequence; // last observed value of mSequence 575 576 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 577 bool mMarkerReached; 578 Modulo<uint32_t> mNewPosition; // in frames 579 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 580 581 status_t mStatus; 582 583 String16 mOpPackageName; // The package name used for app ops. 584 585 size_t mFrameCount; // corresponds to current IAudioRecord, value is 586 // reported back by AudioFlinger to the client 587 size_t mReqFrameCount; // frame count to request the first or next time 588 // a new IAudioRecord is needed, non-decreasing 589 590 int64_t mFramesRead; // total frames read. reset to zero after 591 // the start() following stop(). It is not 592 // changed after restoring the track. 593 int64_t mFramesReadServerOffset; // An offset to server frames read due to 594 // restoring AudioRecord, or stop/start. 595 // constant after constructor or set() 596 uint32_t mSampleRate; 597 audio_format_t mFormat; 598 uint32_t mChannelCount; 599 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 600 uint32_t mLatency; // in ms 601 audio_channel_mask_t mChannelMask; 602 603 audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may 604 // be denied by client or server, such as 605 // AUDIO_INPUT_FLAG_FAST. mLock must be 606 // held to read or write those bits reliably. 607 audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const 608 609 audio_session_t mSessionId; 610 transfer_type mTransfer; 611 612 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 613 // provided the initial set() was successful 614 sp<IAudioRecord> mAudioRecord; 615 sp<IMemory> mCblkMemory; 616 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 617 sp<IMemory> mBufferMemory; 618 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 619 620 int mPreviousPriority; // before start() 621 SchedPolicy mPreviousSchedulingGroup; 622 bool mAwaitBoost; // thread should wait for priority boost before running 623 624 // The proxy should only be referenced while a lock is held because the proxy isn't 625 // multi-thread safe. 626 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 627 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 628 // them around in case they are replaced during the obtainBuffer(). 629 sp<AudioRecordClientProxy> mProxy; 630 631 bool mInOverrun; // whether recorder is currently in overrun state 632 633 private: 634 class DeathNotifier : public IBinder::DeathRecipient { 635 public: 636 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 637 protected: 638 virtual void binderDied(const wp<IBinder>& who); 639 private: 640 const wp<AudioRecord> mAudioRecord; 641 }; 642 643 sp<DeathNotifier> mDeathNotifier; 644 uint32_t mSequence; // incremented for each new IAudioRecord attempt 645 int mClientUid; 646 pid_t mClientPid; 647 audio_attributes_t mAttributes; 648 649 // For Device Selection API 650 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 651 audio_port_handle_t mSelectedDeviceId; 652 sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 653 }; 654 655 }; // namespace android 656 657 #endif // ANDROID_AUDIORECORD_H 658
