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 27namespace android { 28 29// ---------------------------------------------------------------------------- 30 31struct audio_track_cblk_t; 32class AudioRecordClientProxy; 33 34// ---------------------------------------------------------------------------- 35 36class AudioRecord : public RefBase 37{ 38public: 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 */ 190protected: 191 virtual ~AudioRecord(); 192public: 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 /* Returns a handle on the audio input used by this AudioRecord. 338 * 339 * Parameters: 340 * none. 341 * 342 * Returned value: 343 * handle on audio hardware input 344 */ 345// FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp 346 audio_io_handle_t getInput() const __attribute__((__deprecated__)) 347 { return getInputPrivate(); } 348private: 349 audio_io_handle_t getInputPrivate() const; 350public: 351 352 /* Returns the audio session ID associated with this AudioRecord. 353 * 354 * Parameters: 355 * none. 356 * 357 * Returned value: 358 * AudioRecord session ID. 359 * 360 * No lock needed because session ID doesn't change after first set(). 361 */ 362 audio_session_t getSessionId() const { return mSessionId; } 363 364 /* Public API for TRANSFER_OBTAIN mode. 365 * Obtains a buffer of up to "audioBuffer->frameCount" full frames. 366 * After draining these frames of data, the caller should release them with releaseBuffer(). 367 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 368 * full frames as are available immediately. 369 * 370 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 371 * additional non-contiguous frames that are predicted to be available immediately, 372 * if the client were to release the first frames and then call obtainBuffer() again. 373 * This value is only a prediction, and needs to be confirmed. 374 * It will be set to zero for an error return. 375 * 376 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 377 * regardless of the value of waitCount. 378 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 379 * maximum timeout based on waitCount; see chart below. 380 * Buffers will be returned until the pool 381 * is exhausted, at which point obtainBuffer() will either block 382 * or return WOULD_BLOCK depending on the value of the "waitCount" 383 * parameter. 384 * 385 * Interpretation of waitCount: 386 * +n limits wait time to n * WAIT_PERIOD_MS, 387 * -1 causes an (almost) infinite wait time, 388 * 0 non-blocking. 389 * 390 * Buffer fields 391 * On entry: 392 * frameCount number of frames requested 393 * size ignored 394 * raw ignored 395 * After error return: 396 * frameCount 0 397 * size 0 398 * raw undefined 399 * After successful return: 400 * frameCount actual number of frames available, <= number requested 401 * size actual number of bytes available 402 * raw pointer to the buffer 403 */ 404 405 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 406 size_t *nonContig = NULL); 407 408 // Explicit Routing 409 /** 410 * TODO Document this method. 411 */ 412 status_t setInputDevice(audio_port_handle_t deviceId); 413 414 /** 415 * TODO Document this method. 416 */ 417 audio_port_handle_t getInputDevice(); 418 419 /* Returns the ID of the audio device actually used by the input to which this AudioRecord 420 * is attached. 421 * A value of AUDIO_PORT_HANDLE_NONE indicates the AudioRecord is not attached to any input. 422 * 423 * Parameters: 424 * none. 425 */ 426 audio_port_handle_t getRoutedDeviceId(); 427 428 /* Add an AudioDeviceCallback. The caller will be notified when the audio device 429 * to which this AudioRecord is routed is updated. 430 * Replaces any previously installed callback. 431 * Parameters: 432 * callback: The callback interface 433 * Returns NO_ERROR if successful. 434 * INVALID_OPERATION if the same callback is already installed. 435 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 436 * BAD_VALUE if the callback is NULL 437 */ 438 status_t addAudioDeviceCallback( 439 const sp<AudioSystem::AudioDeviceCallback>& callback); 440 441 /* remove an AudioDeviceCallback. 442 * Parameters: 443 * callback: The callback interface 444 * Returns NO_ERROR if successful. 445 * INVALID_OPERATION if the callback is not installed 446 * BAD_VALUE if the callback is NULL 447 */ 448 status_t removeAudioDeviceCallback( 449 const sp<AudioSystem::AudioDeviceCallback>& callback); 450 451private: 452 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 453 * additional non-contiguous frames that are predicted to be available immediately, 454 * if the client were to release the first frames and then call obtainBuffer() again. 455 * This value is only a prediction, and needs to be confirmed. 456 * It will be set to zero for an error return. 457 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 458 * in case the requested amount of frames is in two or more non-contiguous regions. 459 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 460 */ 461 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 462 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 463public: 464 465 /* Public API for TRANSFER_OBTAIN mode. 466 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. 467 * 468 * Buffer fields: 469 * frameCount currently ignored but recommend to set to actual number of frames consumed 470 * size actual number of bytes consumed, must be multiple of frameSize 471 * raw ignored 472 */ 473 void releaseBuffer(const Buffer* audioBuffer); 474 475 /* As a convenience we provide a read() interface to the audio buffer. 476 * Input parameter 'size' is in byte units. 477 * This is implemented on top of obtainBuffer/releaseBuffer. For best 478 * performance use callbacks. Returns actual number of bytes read >= 0, 479 * or one of the following negative status codes: 480 * INVALID_OPERATION AudioRecord is configured for streaming mode 481 * BAD_VALUE size is invalid 482 * WOULD_BLOCK when obtainBuffer() returns same, or 483 * AudioRecord was stopped during the read 484 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 485 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 486 * false for the method to return immediately without waiting to try multiple times to read 487 * the full content of the buffer. 488 */ 489 ssize_t read(void* buffer, size_t size, bool blocking = true); 490 491 /* Return the number of input frames lost in the audio driver since the last call of this 492 * function. Audio driver is expected to reset the value to 0 and restart counting upon 493 * returning the current value by this function call. Such loss typically occurs when the 494 * user space process is blocked longer than the capacity of audio driver buffers. 495 * Units: the number of input audio frames. 496 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 497 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 498 */ 499 uint32_t getInputFramesLost() const; 500 501 /* Get the flags */ 502 audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 503 504private: 505 /* copying audio record objects is not allowed */ 506 AudioRecord(const AudioRecord& other); 507 AudioRecord& operator = (const AudioRecord& other); 508 509 /* a small internal class to handle the callback */ 510 class AudioRecordThread : public Thread 511 { 512 public: 513 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 514 515 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 516 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 517 virtual void requestExit(); 518 519 void pause(); // suspend thread from execution at next loop boundary 520 void resume(); // allow thread to execute, if not requested to exit 521 void wake(); // wake to handle changed notification conditions. 522 523 private: 524 void pauseInternal(nsecs_t ns = 0LL); 525 // like pause(), but only used internally within thread 526 527 friend class AudioRecord; 528 virtual bool threadLoop(); 529 AudioRecord& mReceiver; 530 virtual ~AudioRecordThread(); 531 Mutex mMyLock; // Thread::mLock is private 532 Condition mMyCond; // Thread::mThreadExitedCondition is private 533 bool mPaused; // whether thread is requested to pause at next loop entry 534 bool mPausedInt; // whether thread internally requests pause 535 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 536 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 537 // to processAudioBuffer() as state may have changed 538 // since pause time calculated. 539 }; 540 541 // body of AudioRecordThread::threadLoop() 542 // returns the maximum amount of time before we would like to run again, where: 543 // 0 immediately 544 // > 0 no later than this many nanoseconds from now 545 // NS_WHENEVER still active but no particular deadline 546 // NS_INACTIVE inactive so don't run again until re-started 547 // NS_NEVER never again 548 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 549 nsecs_t processAudioBuffer(); 550 551 // caller must hold lock on mLock for all _l methods 552 553 status_t openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName); 554 555 // FIXME enum is faster than strcmp() for parameter 'from' 556 status_t restoreRecord_l(const char *from); 557 558 sp<AudioRecordThread> mAudioRecordThread; 559 mutable Mutex mLock; 560 561 // Current client state: false = stopped, true = active. Protected by mLock. If more states 562 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 563 bool mActive; 564 565 // for client callback handler 566 callback_t mCbf; // callback handler for events, or NULL 567 void* mUserData; 568 569 // for notification APIs 570 uint32_t mNotificationFramesReq; // requested number of frames between each 571 // notification callback 572 // as specified in constructor or set() 573 uint32_t mNotificationFramesAct; // actual number of frames between each 574 // notification callback 575 bool mRefreshRemaining; // processAudioBuffer() should refresh 576 // mRemainingFrames and mRetryOnPartialBuffer 577 578 // These are private to processAudioBuffer(), and are not protected by a lock 579 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 580 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 581 uint32_t mObservedSequence; // last observed value of mSequence 582 583 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 584 bool mMarkerReached; 585 Modulo<uint32_t> mNewPosition; // in frames 586 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 587 588 status_t mStatus; 589 590 String16 mOpPackageName; // The package name used for app ops. 591 592 size_t mFrameCount; // corresponds to current IAudioRecord, value is 593 // reported back by AudioFlinger to the client 594 size_t mReqFrameCount; // frame count to request the first or next time 595 // a new IAudioRecord is needed, non-decreasing 596 597 int64_t mFramesRead; // total frames read. reset to zero after 598 // the start() following stop(). It is not 599 // changed after restoring the track. 600 int64_t mFramesReadServerOffset; // An offset to server frames read due to 601 // restoring AudioRecord, or stop/start. 602 // constant after constructor or set() 603 uint32_t mSampleRate; 604 audio_format_t mFormat; 605 uint32_t mChannelCount; 606 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 607 uint32_t mLatency; // in ms 608 audio_channel_mask_t mChannelMask; 609 610 audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may 611 // be denied by client or server, such as 612 // AUDIO_INPUT_FLAG_FAST. mLock must be 613 // held to read or write those bits reliably. 614 audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const 615 616 audio_session_t mSessionId; 617 transfer_type mTransfer; 618 619 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 620 // provided the initial set() was successful 621 sp<IAudioRecord> mAudioRecord; 622 sp<IMemory> mCblkMemory; 623 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 624 sp<IMemory> mBufferMemory; 625 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 626 627 int mPreviousPriority; // before start() 628 SchedPolicy mPreviousSchedulingGroup; 629 bool mAwaitBoost; // thread should wait for priority boost before running 630 631 // The proxy should only be referenced while a lock is held because the proxy isn't 632 // multi-thread safe. 633 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 634 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 635 // them around in case they are replaced during the obtainBuffer(). 636 sp<AudioRecordClientProxy> mProxy; 637 638 bool mInOverrun; // whether recorder is currently in overrun state 639 640private: 641 class DeathNotifier : public IBinder::DeathRecipient { 642 public: 643 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 644 protected: 645 virtual void binderDied(const wp<IBinder>& who); 646 private: 647 const wp<AudioRecord> mAudioRecord; 648 }; 649 650 sp<DeathNotifier> mDeathNotifier; 651 uint32_t mSequence; // incremented for each new IAudioRecord attempt 652 uid_t mClientUid; 653 pid_t mClientPid; 654 audio_attributes_t mAttributes; 655 656 // For Device Selection API 657 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 658 audio_port_handle_t mSelectedDeviceId; 659 sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 660 audio_port_handle_t mPortId; // unique ID allocated by audio policy 661 662}; 663 664}; // namespace android 665 666#endif // ANDROID_AUDIORECORD_H 667