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