AudioRecord.h revision fec2f93fae282ad10bbb5e3fcce9f60eff2cfb48
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 <binder/IMemory.h> 21#include <cutils/sched_policy.h> 22#include <media/AudioSystem.h> 23#include <media/AudioTimestamp.h> 24#include <media/Modulo.h> 25#include <utils/RefBase.h> 26#include <utils/threads.h> 27 28#include "android/media/IAudioRecord.h" 29 30namespace android { 31 32// ---------------------------------------------------------------------------- 33 34struct audio_track_cblk_t; 35class AudioRecordClientProxy; 36 37// ---------------------------------------------------------------------------- 38 39class AudioRecord : public AudioSystem::AudioDeviceCallback 40{ 41public: 42 43 /* Events used by AudioRecord callback function (callback_t). 44 * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*. 45 */ 46 enum event_type { 47 EVENT_MORE_DATA = 0, // Request to read available data from buffer. 48 // If this event is delivered but the callback handler 49 // does not want to read the available data, the handler must 50 // explicitly ignore the event by setting frameCount to zero. 51 EVENT_OVERRUN = 1, // Buffer overrun occurred. 52 EVENT_MARKER = 2, // Record head is at the specified marker position 53 // (See setMarkerPosition()). 54 EVENT_NEW_POS = 3, // Record head is at a new position 55 // (See setPositionUpdatePeriod()). 56 EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and 57 // voluntary invalidation by mediaserver, or mediaserver crash. 58 }; 59 60 /* Client should declare a Buffer and pass address to obtainBuffer() 61 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 62 */ 63 64 class Buffer 65 { 66 public: 67 // FIXME use m prefix 68 size_t frameCount; // number of sample frames corresponding to size; 69 // on input to obtainBuffer() it is the number of frames desired 70 // on output from obtainBuffer() it is the number of available 71 // frames to be read 72 // on input to releaseBuffer() it is currently ignored 73 74 size_t size; // input/output in bytes == frameCount * frameSize 75 // on input to obtainBuffer() it is ignored 76 // on output from obtainBuffer() it is the number of available 77 // bytes to be read, which is frameCount * frameSize 78 // on input to releaseBuffer() it is the number of bytes to 79 // release 80 // FIXME This is redundant with respect to frameCount. Consider 81 // removing size and making frameCount the primary field. 82 83 union { 84 void* raw; 85 short* i16; // signed 16-bit 86 int8_t* i8; // unsigned 8-bit, offset by 0x80 87 // input to obtainBuffer(): unused, output: pointer to buffer 88 }; 89 }; 90 91 /* As a convenience, if a callback is supplied, a handler thread 92 * is automatically created with the appropriate priority. This thread 93 * invokes the callback when a new buffer becomes available or various conditions occur. 94 * Parameters: 95 * 96 * event: type of event notified (see enum AudioRecord::event_type). 97 * user: Pointer to context for use by the callback receiver. 98 * info: Pointer to optional parameter according to event type: 99 * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read 100 * more bytes than indicated by 'size' field and update 'size' if 101 * fewer bytes are consumed. 102 * - EVENT_OVERRUN: unused. 103 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 104 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 105 * - EVENT_NEW_IAUDIORECORD: unused. 106 */ 107 108 typedef void (*callback_t)(int event, void* user, void *info); 109 110 /* Returns the minimum frame count required for the successful creation of 111 * an AudioRecord object. 112 * Returned status (from utils/Errors.h) can be: 113 * - NO_ERROR: successful operation 114 * - NO_INIT: audio server or audio hardware not initialized 115 * - BAD_VALUE: unsupported configuration 116 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 117 * and is undefined otherwise. 118 * FIXME This API assumes a route, and so should be deprecated. 119 */ 120 121 static status_t getMinFrameCount(size_t* frameCount, 122 uint32_t sampleRate, 123 audio_format_t format, 124 audio_channel_mask_t channelMask); 125 126 /* How data is transferred from AudioRecord 127 */ 128 enum transfer_type { 129 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 130 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 131 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 132 TRANSFER_SYNC, // synchronous read() 133 }; 134 135 /* Constructs an uninitialized AudioRecord. No connection with 136 * AudioFlinger takes place. Use set() after this. 137 * 138 * Parameters: 139 * 140 * opPackageName: The package name used for app ops. 141 */ 142 AudioRecord(const String16& opPackageName); 143 144 /* Creates an AudioRecord object and registers it with AudioFlinger. 145 * Once created, the track needs to be started before it can be used. 146 * Unspecified values are set to appropriate default values. 147 * 148 * Parameters: 149 * 150 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 151 * sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate. 152 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 153 * 16 bits per sample). 154 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 155 * opPackageName: The package name used for app ops. 156 * frameCount: Minimum size of track PCM buffer in frames. This defines the 157 * application's contribution to the 158 * latency of the track. The actual size selected by the AudioRecord could 159 * be larger if the requested size is not compatible with current audio HAL 160 * latency. Zero means to use a default value. 161 * cbf: Callback function. If not null, this function is called periodically 162 * to consume new data in TRANSFER_CALLBACK mode 163 * and inform of marker, position updates, etc. 164 * user: Context for use by the callback receiver. 165 * notificationFrames: The callback function is called each time notificationFrames PCM 166 * frames are ready in record track output buffer. 167 * sessionId: Not yet supported. 168 * transferType: How data is transferred from AudioRecord. 169 * flags: See comments on audio_input_flags_t in <system/audio.h> 170 * pAttributes: If not NULL, supersedes inputSource for use case selection. 171 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 172 */ 173 174 AudioRecord(audio_source_t inputSource, 175 uint32_t sampleRate, 176 audio_format_t format, 177 audio_channel_mask_t channelMask, 178 const String16& opPackageName, 179 size_t frameCount = 0, 180 callback_t cbf = NULL, 181 void* user = NULL, 182 uint32_t notificationFrames = 0, 183 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 184 transfer_type transferType = TRANSFER_DEFAULT, 185 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 186 uid_t uid = AUDIO_UID_INVALID, 187 pid_t pid = -1, 188 const audio_attributes_t* pAttributes = NULL, 189 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 190 191 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 192 * Also destroys all resources associated with the AudioRecord. 193 */ 194protected: 195 virtual ~AudioRecord(); 196public: 197 198 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 199 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 200 * set() is not multi-thread safe. 201 * Returned status (from utils/Errors.h) can be: 202 * - NO_ERROR: successful intialization 203 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 204 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 205 * - NO_INIT: audio server or audio hardware not initialized 206 * - PERMISSION_DENIED: recording is not allowed for the requesting process 207 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 208 * 209 * Parameters not listed in the AudioRecord constructors above: 210 * 211 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 212 */ 213 status_t set(audio_source_t inputSource, 214 uint32_t sampleRate, 215 audio_format_t format, 216 audio_channel_mask_t channelMask, 217 size_t frameCount = 0, 218 callback_t cbf = NULL, 219 void* user = NULL, 220 uint32_t notificationFrames = 0, 221 bool threadCanCallJava = false, 222 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 223 transfer_type transferType = TRANSFER_DEFAULT, 224 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 225 uid_t uid = AUDIO_UID_INVALID, 226 pid_t pid = -1, 227 const audio_attributes_t* pAttributes = NULL, 228 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 229 230 /* Result of constructing the AudioRecord. This must be checked for successful initialization 231 * before using any AudioRecord API (except for set()), because using 232 * an uninitialized AudioRecord produces undefined results. 233 * See set() method above for possible return codes. 234 */ 235 status_t initCheck() const { return mStatus; } 236 237 /* Returns this track's estimated latency in milliseconds. 238 * This includes the latency due to AudioRecord buffer size, resampling if applicable, 239 * and audio hardware driver. 240 */ 241 uint32_t latency() const { return mLatency; } 242 243 /* getters, see constructor and set() */ 244 245 audio_format_t format() const { return mFormat; } 246 uint32_t channelCount() const { return mChannelCount; } 247 size_t frameCount() const { return mFrameCount; } 248 size_t frameSize() const { return mFrameSize; } 249 audio_source_t inputSource() const { return mAttributes.source; } 250 251 /* 252 * Return the period of the notification callback in frames. 253 * This value is set when the AudioRecord is constructed. 254 * It can be modified if the AudioRecord is rerouted. 255 */ 256 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 257 258 /* After it's created the track is not active. Call start() to 259 * make it active. If set, the callback will start being called. 260 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 261 * the specified event occurs on the specified trigger session. 262 */ 263 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 264 audio_session_t triggerSession = AUDIO_SESSION_NONE); 265 266 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 267 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 268 */ 269 void stop(); 270 bool stopped() const; 271 272 /* Return the sink sample rate for this record track in Hz. 273 * If specified as zero in constructor or set(), this will be the source sample rate. 274 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 275 */ 276 uint32_t getSampleRate() const { return mSampleRate; } 277 278 /* Sets marker position. When record reaches the number of frames specified, 279 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 280 * with marker == 0 cancels marker notification callback. 281 * To set a marker at a position which would compute as 0, 282 * a workaround is to set the marker at a nearby position such as ~0 or 1. 283 * If the AudioRecord has been opened with no callback function associated, 284 * the operation will fail. 285 * 286 * Parameters: 287 * 288 * marker: marker position expressed in wrapping (overflow) frame units, 289 * like the return value of getPosition(). 290 * 291 * Returned status (from utils/Errors.h) can be: 292 * - NO_ERROR: successful operation 293 * - INVALID_OPERATION: the AudioRecord has no callback installed. 294 */ 295 status_t setMarkerPosition(uint32_t marker); 296 status_t getMarkerPosition(uint32_t *marker) const; 297 298 /* Sets position update period. Every time the number of frames specified has been recorded, 299 * a callback with event type EVENT_NEW_POS is called. 300 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 301 * callback. 302 * If the AudioRecord has been opened with no callback function associated, 303 * the operation will fail. 304 * Extremely small values may be rounded up to a value the implementation can support. 305 * 306 * Parameters: 307 * 308 * updatePeriod: position update notification period expressed in frames. 309 * 310 * Returned status (from utils/Errors.h) can be: 311 * - NO_ERROR: successful operation 312 * - INVALID_OPERATION: the AudioRecord has no callback installed. 313 */ 314 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 315 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 316 317 /* Return the total number of frames recorded since recording started. 318 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 319 * It is reset to zero by stop(). 320 * 321 * Parameters: 322 * 323 * position: Address where to return record head position. 324 * 325 * Returned status (from utils/Errors.h) can be: 326 * - NO_ERROR: successful operation 327 * - BAD_VALUE: position is NULL 328 */ 329 status_t getPosition(uint32_t *position) const; 330 331 /* Return the record timestamp. 332 * 333 * Parameters: 334 * timestamp: A pointer to the timestamp to be filled. 335 * 336 * Returned status (from utils/Errors.h) can be: 337 * - NO_ERROR: successful operation 338 * - BAD_VALUE: timestamp is NULL 339 */ 340 status_t getTimestamp(ExtendedTimestamp *timestamp); 341 342 /** 343 * @param transferType 344 * @return text string that matches the enum name 345 */ 346 static const char * convertTransferToText(transfer_type transferType); 347 348 /* Returns a handle on the audio input used by this AudioRecord. 349 * 350 * Parameters: 351 * none. 352 * 353 * Returned value: 354 * handle on audio hardware input 355 */ 356// FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp 357 audio_io_handle_t getInput() const __attribute__((__deprecated__)) 358 { return getInputPrivate(); } 359private: 360 audio_io_handle_t getInputPrivate() const; 361public: 362 363 /* Returns the audio session ID associated with this AudioRecord. 364 * 365 * Parameters: 366 * none. 367 * 368 * Returned value: 369 * AudioRecord session ID. 370 * 371 * No lock needed because session ID doesn't change after first set(). 372 */ 373 audio_session_t getSessionId() const { return mSessionId; } 374 375 /* Public API for TRANSFER_OBTAIN mode. 376 * Obtains a buffer of up to "audioBuffer->frameCount" full frames. 377 * After draining these frames of data, the caller should release them with releaseBuffer(). 378 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 379 * full frames as are available immediately. 380 * 381 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 382 * additional non-contiguous frames that are predicted to be available immediately, 383 * if the client were to release the first frames and then call obtainBuffer() again. 384 * This value is only a prediction, and needs to be confirmed. 385 * It will be set to zero for an error return. 386 * 387 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 388 * regardless of the value of waitCount. 389 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 390 * maximum timeout based on waitCount; see chart below. 391 * Buffers will be returned until the pool 392 * is exhausted, at which point obtainBuffer() will either block 393 * or return WOULD_BLOCK depending on the value of the "waitCount" 394 * parameter. 395 * 396 * Interpretation of waitCount: 397 * +n limits wait time to n * WAIT_PERIOD_MS, 398 * -1 causes an (almost) infinite wait time, 399 * 0 non-blocking. 400 * 401 * Buffer fields 402 * On entry: 403 * frameCount number of frames requested 404 * size ignored 405 * raw ignored 406 * After error return: 407 * frameCount 0 408 * size 0 409 * raw undefined 410 * After successful return: 411 * frameCount actual number of frames available, <= number requested 412 * size actual number of bytes available 413 * raw pointer to the buffer 414 */ 415 416 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 417 size_t *nonContig = NULL); 418 419 // Explicit Routing 420 /** 421 * TODO Document this method. 422 */ 423 status_t setInputDevice(audio_port_handle_t deviceId); 424 425 /** 426 * TODO Document this method. 427 */ 428 audio_port_handle_t getInputDevice(); 429 430 /* Returns the ID of the audio device actually used by the input to which this AudioRecord 431 * is attached. 432 * The device ID is relevant only if the AudioRecord is active. 433 * When the AudioRecord is inactive, the device ID returned can be either: 434 * - AUDIO_PORT_HANDLE_NONE if the AudioRecord is not attached to any output. 435 * - The device ID used before paused or stopped. 436 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioRecord 437 * has not been started yet. 438 * 439 * Parameters: 440 * none. 441 */ 442 audio_port_handle_t getRoutedDeviceId(); 443 444 /* Add an AudioDeviceCallback. The caller will be notified when the audio device 445 * to which this AudioRecord is routed is updated. 446 * Replaces any previously installed callback. 447 * Parameters: 448 * callback: The callback interface 449 * Returns NO_ERROR if successful. 450 * INVALID_OPERATION if the same callback is already installed. 451 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 452 * BAD_VALUE if the callback is NULL 453 */ 454 status_t addAudioDeviceCallback( 455 const sp<AudioSystem::AudioDeviceCallback>& callback); 456 457 /* remove an AudioDeviceCallback. 458 * Parameters: 459 * callback: The callback interface 460 * Returns NO_ERROR if successful. 461 * INVALID_OPERATION if the callback is not installed 462 * BAD_VALUE if the callback is NULL 463 */ 464 status_t removeAudioDeviceCallback( 465 const sp<AudioSystem::AudioDeviceCallback>& callback); 466 467 // AudioSystem::AudioDeviceCallback> virtuals 468 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 469 audio_port_handle_t deviceId); 470 471private: 472 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 473 * additional non-contiguous frames that are predicted to be available immediately, 474 * if the client were to release the first frames and then call obtainBuffer() again. 475 * This value is only a prediction, and needs to be confirmed. 476 * It will be set to zero for an error return. 477 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 478 * in case the requested amount of frames is in two or more non-contiguous regions. 479 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 480 */ 481 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 482 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 483public: 484 485 /* Public API for TRANSFER_OBTAIN mode. 486 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. 487 * 488 * Buffer fields: 489 * frameCount currently ignored but recommend to set to actual number of frames consumed 490 * size actual number of bytes consumed, must be multiple of frameSize 491 * raw ignored 492 */ 493 void releaseBuffer(const Buffer* audioBuffer); 494 495 /* As a convenience we provide a read() interface to the audio buffer. 496 * Input parameter 'size' is in byte units. 497 * This is implemented on top of obtainBuffer/releaseBuffer. For best 498 * performance use callbacks. Returns actual number of bytes read >= 0, 499 * or one of the following negative status codes: 500 * INVALID_OPERATION AudioRecord is configured for streaming mode 501 * BAD_VALUE size is invalid 502 * WOULD_BLOCK when obtainBuffer() returns same, or 503 * AudioRecord was stopped during the read 504 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 505 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 506 * false for the method to return immediately without waiting to try multiple times to read 507 * the full content of the buffer. 508 */ 509 ssize_t read(void* buffer, size_t size, bool blocking = true); 510 511 /* Return the number of input frames lost in the audio driver since the last call of this 512 * function. Audio driver is expected to reset the value to 0 and restart counting upon 513 * returning the current value by this function call. Such loss typically occurs when the 514 * user space process is blocked longer than the capacity of audio driver buffers. 515 * Units: the number of input audio frames. 516 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 517 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 518 */ 519 uint32_t getInputFramesLost() const; 520 521 /* Get the flags */ 522 audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 523 524private: 525 /* copying audio record objects is not allowed */ 526 AudioRecord(const AudioRecord& other); 527 AudioRecord& operator = (const AudioRecord& other); 528 529 /* a small internal class to handle the callback */ 530 class AudioRecordThread : public Thread 531 { 532 public: 533 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 534 535 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 536 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 537 virtual void requestExit(); 538 539 void pause(); // suspend thread from execution at next loop boundary 540 void resume(); // allow thread to execute, if not requested to exit 541 void wake(); // wake to handle changed notification conditions. 542 543 private: 544 void pauseInternal(nsecs_t ns = 0LL); 545 // like pause(), but only used internally within thread 546 547 friend class AudioRecord; 548 virtual bool threadLoop(); 549 AudioRecord& mReceiver; 550 virtual ~AudioRecordThread(); 551 Mutex mMyLock; // Thread::mLock is private 552 Condition mMyCond; // Thread::mThreadExitedCondition is private 553 bool mPaused; // whether thread is requested to pause at next loop entry 554 bool mPausedInt; // whether thread internally requests pause 555 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 556 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 557 // to processAudioBuffer() as state may have changed 558 // since pause time calculated. 559 }; 560 561 // body of AudioRecordThread::threadLoop() 562 // returns the maximum amount of time before we would like to run again, where: 563 // 0 immediately 564 // > 0 no later than this many nanoseconds from now 565 // NS_WHENEVER still active but no particular deadline 566 // NS_INACTIVE inactive so don't run again until re-started 567 // NS_NEVER never again 568 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 569 nsecs_t processAudioBuffer(); 570 571 // caller must hold lock on mLock for all _l methods 572 573 status_t openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName); 574 575 // FIXME enum is faster than strcmp() for parameter 'from' 576 status_t restoreRecord_l(const char *from); 577 578 void updateRoutedDeviceId_l(); 579 580 sp<AudioRecordThread> mAudioRecordThread; 581 mutable Mutex mLock; 582 583 // Current client state: false = stopped, true = active. Protected by mLock. If more states 584 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 585 bool mActive; 586 587 // for client callback handler 588 callback_t mCbf; // callback handler for events, or NULL 589 void* mUserData; 590 591 // for notification APIs 592 uint32_t mNotificationFramesReq; // requested number of frames between each 593 // notification callback 594 // as specified in constructor or set() 595 uint32_t mNotificationFramesAct; // actual number of frames between each 596 // notification callback 597 bool mRefreshRemaining; // processAudioBuffer() should refresh 598 // mRemainingFrames and mRetryOnPartialBuffer 599 600 // These are private to processAudioBuffer(), and are not protected by a lock 601 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 602 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 603 uint32_t mObservedSequence; // last observed value of mSequence 604 605 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 606 bool mMarkerReached; 607 Modulo<uint32_t> mNewPosition; // in frames 608 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 609 610 status_t mStatus; 611 612 String16 mOpPackageName; // The package name used for app ops. 613 614 size_t mFrameCount; // corresponds to current IAudioRecord, value is 615 // reported back by AudioFlinger to the client 616 size_t mReqFrameCount; // frame count to request the first or next time 617 // a new IAudioRecord is needed, non-decreasing 618 619 int64_t mFramesRead; // total frames read. reset to zero after 620 // the start() following stop(). It is not 621 // changed after restoring the track. 622 int64_t mFramesReadServerOffset; // An offset to server frames read due to 623 // restoring AudioRecord, or stop/start. 624 // constant after constructor or set() 625 uint32_t mSampleRate; 626 audio_format_t mFormat; 627 uint32_t mChannelCount; 628 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 629 uint32_t mLatency; // in ms 630 audio_channel_mask_t mChannelMask; 631 632 audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may 633 // be denied by client or server, such as 634 // AUDIO_INPUT_FLAG_FAST. mLock must be 635 // held to read or write those bits reliably. 636 audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const 637 638 audio_session_t mSessionId; 639 transfer_type mTransfer; 640 641 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 642 // provided the initial set() was successful 643 sp<media::IAudioRecord> mAudioRecord; 644 sp<IMemory> mCblkMemory; 645 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 646 sp<IMemory> mBufferMemory; 647 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 648 649 int mPreviousPriority; // before start() 650 SchedPolicy mPreviousSchedulingGroup; 651 bool mAwaitBoost; // thread should wait for priority boost before running 652 653 // The proxy should only be referenced while a lock is held because the proxy isn't 654 // multi-thread safe. 655 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 656 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 657 // them around in case they are replaced during the obtainBuffer(). 658 sp<AudioRecordClientProxy> mProxy; 659 660 bool mInOverrun; // whether recorder is currently in overrun state 661 662private: 663 class DeathNotifier : public IBinder::DeathRecipient { 664 public: 665 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 666 protected: 667 virtual void binderDied(const wp<IBinder>& who); 668 private: 669 const wp<AudioRecord> mAudioRecord; 670 }; 671 672 sp<DeathNotifier> mDeathNotifier; 673 uint32_t mSequence; // incremented for each new IAudioRecord attempt 674 uid_t mClientUid; 675 pid_t mClientPid; 676 audio_attributes_t mAttributes; 677 678 // For Device Selection API 679 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 680 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 681 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 682 // May not match the app selection depending on other 683 // activity and connected devices 684 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 685 audio_port_handle_t mPortId; // unique ID allocated by audio policy 686 687}; 688 689}; // namespace android 690 691#endif // ANDROID_AUDIORECORD_H 692