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/MediaAnalyticsItem.h> 25#include <media/Modulo.h> 26#include <media/MicrophoneInfo.h> 27#include <utils/RefBase.h> 28#include <utils/threads.h> 29#include <vector> 30 31#include "android/media/IAudioRecord.h" 32 33namespace android { 34 35// ---------------------------------------------------------------------------- 36 37struct audio_track_cblk_t; 38class AudioRecordClientProxy; 39 40// ---------------------------------------------------------------------------- 41 42class AudioRecord : public AudioSystem::AudioDeviceCallback 43{ 44public: 45 46 /* Events used by AudioRecord callback function (callback_t). 47 * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*. 48 */ 49 enum event_type { 50 EVENT_MORE_DATA = 0, // Request to read available data from buffer. 51 // If this event is delivered but the callback handler 52 // does not want to read the available data, the handler must 53 // explicitly ignore the event by setting frameCount to zero. 54 EVENT_OVERRUN = 1, // Buffer overrun occurred. 55 EVENT_MARKER = 2, // Record head is at the specified marker position 56 // (See setMarkerPosition()). 57 EVENT_NEW_POS = 3, // Record head is at a new position 58 // (See setPositionUpdatePeriod()). 59 EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and 60 // voluntary invalidation by mediaserver, or mediaserver crash. 61 }; 62 63 /* Client should declare a Buffer and pass address to obtainBuffer() 64 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 65 */ 66 67 class Buffer 68 { 69 public: 70 // FIXME use m prefix 71 size_t frameCount; // number of sample frames corresponding to size; 72 // on input to obtainBuffer() it is the number of frames desired 73 // on output from obtainBuffer() it is the number of available 74 // frames to be read 75 // on input to releaseBuffer() it is currently ignored 76 77 size_t size; // input/output in bytes == frameCount * frameSize 78 // on input to obtainBuffer() it is ignored 79 // on output from obtainBuffer() it is the number of available 80 // bytes to be read, which is frameCount * frameSize 81 // on input to releaseBuffer() it is the number of bytes to 82 // release 83 // FIXME This is redundant with respect to frameCount. Consider 84 // removing size and making frameCount the primary field. 85 86 union { 87 void* raw; 88 short* i16; // signed 16-bit 89 int8_t* i8; // unsigned 8-bit, offset by 0x80 90 // input to obtainBuffer(): unused, output: pointer to buffer 91 }; 92 }; 93 94 /* As a convenience, if a callback is supplied, a handler thread 95 * is automatically created with the appropriate priority. This thread 96 * invokes the callback when a new buffer becomes available or various conditions occur. 97 * Parameters: 98 * 99 * event: type of event notified (see enum AudioRecord::event_type). 100 * user: Pointer to context for use by the callback receiver. 101 * info: Pointer to optional parameter according to event type: 102 * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read 103 * more bytes than indicated by 'size' field and update 'size' if 104 * fewer bytes are consumed. 105 * - EVENT_OVERRUN: unused. 106 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 107 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 108 * - EVENT_NEW_IAUDIORECORD: unused. 109 */ 110 111 typedef void (*callback_t)(int event, void* user, void *info); 112 113 /* Returns the minimum frame count required for the successful creation of 114 * an AudioRecord object. 115 * Returned status (from utils/Errors.h) can be: 116 * - NO_ERROR: successful operation 117 * - NO_INIT: audio server or audio hardware not initialized 118 * - BAD_VALUE: unsupported configuration 119 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 120 * and is undefined otherwise. 121 * FIXME This API assumes a route, and so should be deprecated. 122 */ 123 124 static status_t getMinFrameCount(size_t* frameCount, 125 uint32_t sampleRate, 126 audio_format_t format, 127 audio_channel_mask_t channelMask); 128 129 /* How data is transferred from AudioRecord 130 */ 131 enum transfer_type { 132 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 133 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 134 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 135 TRANSFER_SYNC, // synchronous read() 136 }; 137 138 /* Constructs an uninitialized AudioRecord. No connection with 139 * AudioFlinger takes place. Use set() after this. 140 * 141 * Parameters: 142 * 143 * opPackageName: The package name used for app ops. 144 */ 145 AudioRecord(const String16& opPackageName); 146 147 /* Creates an AudioRecord object and registers it with AudioFlinger. 148 * Once created, the track needs to be started before it can be used. 149 * Unspecified values are set to appropriate default values. 150 * 151 * Parameters: 152 * 153 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 154 * sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate. 155 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 156 * 16 bits per sample). 157 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 158 * opPackageName: The package name used for app ops. 159 * frameCount: Minimum size of track PCM buffer in frames. This defines the 160 * application's contribution to the 161 * latency of the track. The actual size selected by the AudioRecord could 162 * be larger if the requested size is not compatible with current audio HAL 163 * latency. Zero means to use a default value. 164 * cbf: Callback function. If not null, this function is called periodically 165 * to consume new data in TRANSFER_CALLBACK mode 166 * and inform of marker, position updates, etc. 167 * user: Context for use by the callback receiver. 168 * notificationFrames: The callback function is called each time notificationFrames PCM 169 * frames are ready in record track output buffer. 170 * sessionId: Not yet supported. 171 * transferType: How data is transferred from AudioRecord. 172 * flags: See comments on audio_input_flags_t in <system/audio.h> 173 * pAttributes: If not NULL, supersedes inputSource for use case selection. 174 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 175 */ 176 177 AudioRecord(audio_source_t inputSource, 178 uint32_t sampleRate, 179 audio_format_t format, 180 audio_channel_mask_t channelMask, 181 const String16& opPackageName, 182 size_t frameCount = 0, 183 callback_t cbf = NULL, 184 void* user = NULL, 185 uint32_t notificationFrames = 0, 186 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 187 transfer_type transferType = TRANSFER_DEFAULT, 188 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 189 uid_t uid = AUDIO_UID_INVALID, 190 pid_t pid = -1, 191 const audio_attributes_t* pAttributes = NULL, 192 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 193 194 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 195 * Also destroys all resources associated with the AudioRecord. 196 */ 197protected: 198 virtual ~AudioRecord(); 199public: 200 201 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 202 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 203 * set() is not multi-thread safe. 204 * Returned status (from utils/Errors.h) can be: 205 * - NO_ERROR: successful intialization 206 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 207 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 208 * - NO_INIT: audio server or audio hardware not initialized 209 * - PERMISSION_DENIED: recording is not allowed for the requesting process 210 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 211 * 212 * Parameters not listed in the AudioRecord constructors above: 213 * 214 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 215 */ 216 status_t set(audio_source_t inputSource, 217 uint32_t sampleRate, 218 audio_format_t format, 219 audio_channel_mask_t channelMask, 220 size_t frameCount = 0, 221 callback_t cbf = NULL, 222 void* user = NULL, 223 uint32_t notificationFrames = 0, 224 bool threadCanCallJava = false, 225 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 226 transfer_type transferType = TRANSFER_DEFAULT, 227 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 228 uid_t uid = AUDIO_UID_INVALID, 229 pid_t pid = -1, 230 const audio_attributes_t* pAttributes = NULL, 231 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 232 233 /* Result of constructing the AudioRecord. This must be checked for successful initialization 234 * before using any AudioRecord API (except for set()), because using 235 * an uninitialized AudioRecord produces undefined results. 236 * See set() method above for possible return codes. 237 */ 238 status_t initCheck() const { return mStatus; } 239 240 /* Returns this track's estimated latency in milliseconds. 241 * This includes the latency due to AudioRecord buffer size, resampling if applicable, 242 * and audio hardware driver. 243 */ 244 uint32_t latency() const { return mLatency; } 245 246 /* getters, see constructor and set() */ 247 248 audio_format_t format() const { return mFormat; } 249 uint32_t channelCount() const { return mChannelCount; } 250 size_t frameCount() const { return mFrameCount; } 251 size_t frameSize() const { return mFrameSize; } 252 audio_source_t inputSource() const { return mAttributes.source; } 253 254 /* 255 * Return the period of the notification callback in frames. 256 * This value is set when the AudioRecord is constructed. 257 * It can be modified if the AudioRecord is rerouted. 258 */ 259 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 260 261 /* 262 * return metrics information for the current instance. 263 */ 264 status_t getMetrics(MediaAnalyticsItem * &item); 265 266 /* After it's created the track is not active. Call start() to 267 * make it active. If set, the callback will start being called. 268 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 269 * the specified event occurs on the specified trigger session. 270 */ 271 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 272 audio_session_t triggerSession = AUDIO_SESSION_NONE); 273 274 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 275 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 276 */ 277 void stop(); 278 bool stopped() const; 279 280 /* Return the sink sample rate for this record track in Hz. 281 * If specified as zero in constructor or set(), this will be the source sample rate. 282 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 283 */ 284 uint32_t getSampleRate() const { return mSampleRate; } 285 286 /* Sets marker position. When record reaches the number of frames specified, 287 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 288 * with marker == 0 cancels marker notification callback. 289 * To set a marker at a position which would compute as 0, 290 * a workaround is to set the marker at a nearby position such as ~0 or 1. 291 * If the AudioRecord has been opened with no callback function associated, 292 * the operation will fail. 293 * 294 * Parameters: 295 * 296 * marker: marker position expressed in wrapping (overflow) frame units, 297 * like the return value of getPosition(). 298 * 299 * Returned status (from utils/Errors.h) can be: 300 * - NO_ERROR: successful operation 301 * - INVALID_OPERATION: the AudioRecord has no callback installed. 302 */ 303 status_t setMarkerPosition(uint32_t marker); 304 status_t getMarkerPosition(uint32_t *marker) const; 305 306 /* Sets position update period. Every time the number of frames specified has been recorded, 307 * a callback with event type EVENT_NEW_POS is called. 308 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 309 * callback. 310 * If the AudioRecord has been opened with no callback function associated, 311 * the operation will fail. 312 * Extremely small values may be rounded up to a value the implementation can support. 313 * 314 * Parameters: 315 * 316 * updatePeriod: position update notification period expressed in frames. 317 * 318 * Returned status (from utils/Errors.h) can be: 319 * - NO_ERROR: successful operation 320 * - INVALID_OPERATION: the AudioRecord has no callback installed. 321 */ 322 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 323 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 324 325 /* Return the total number of frames recorded since recording started. 326 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 327 * It is reset to zero by stop(). 328 * 329 * Parameters: 330 * 331 * position: Address where to return record head position. 332 * 333 * Returned status (from utils/Errors.h) can be: 334 * - NO_ERROR: successful operation 335 * - BAD_VALUE: position is NULL 336 */ 337 status_t getPosition(uint32_t *position) const; 338 339 /* Return the record timestamp. 340 * 341 * Parameters: 342 * timestamp: A pointer to the timestamp to be filled. 343 * 344 * Returned status (from utils/Errors.h) can be: 345 * - NO_ERROR: successful operation 346 * - BAD_VALUE: timestamp is NULL 347 */ 348 status_t getTimestamp(ExtendedTimestamp *timestamp); 349 350 /** 351 * @param transferType 352 * @return text string that matches the enum name 353 */ 354 static const char * convertTransferToText(transfer_type transferType); 355 356 /* Returns a handle on the audio input used by this AudioRecord. 357 * 358 * Parameters: 359 * none. 360 * 361 * Returned value: 362 * handle on audio hardware input 363 */ 364// FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp 365 audio_io_handle_t getInput() const __attribute__((__deprecated__)) 366 { return getInputPrivate(); } 367private: 368 audio_io_handle_t getInputPrivate() const; 369public: 370 371 /* Returns the audio session ID associated with this AudioRecord. 372 * 373 * Parameters: 374 * none. 375 * 376 * Returned value: 377 * AudioRecord session ID. 378 * 379 * No lock needed because session ID doesn't change after first set(). 380 */ 381 audio_session_t getSessionId() const { return mSessionId; } 382 383 /* Public API for TRANSFER_OBTAIN mode. 384 * Obtains a buffer of up to "audioBuffer->frameCount" full frames. 385 * After draining these frames of data, the caller should release them with releaseBuffer(). 386 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 387 * full frames as are available immediately. 388 * 389 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 390 * additional non-contiguous frames that are predicted to be available immediately, 391 * if the client were to release the first frames and then call obtainBuffer() again. 392 * This value is only a prediction, and needs to be confirmed. 393 * It will be set to zero for an error return. 394 * 395 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 396 * regardless of the value of waitCount. 397 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 398 * maximum timeout based on waitCount; see chart below. 399 * Buffers will be returned until the pool 400 * is exhausted, at which point obtainBuffer() will either block 401 * or return WOULD_BLOCK depending on the value of the "waitCount" 402 * parameter. 403 * 404 * Interpretation of waitCount: 405 * +n limits wait time to n * WAIT_PERIOD_MS, 406 * -1 causes an (almost) infinite wait time, 407 * 0 non-blocking. 408 * 409 * Buffer fields 410 * On entry: 411 * frameCount number of frames requested 412 * size ignored 413 * raw ignored 414 * After error return: 415 * frameCount 0 416 * size 0 417 * raw undefined 418 * After successful return: 419 * frameCount actual number of frames available, <= number requested 420 * size actual number of bytes available 421 * raw pointer to the buffer 422 */ 423 424 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 425 size_t *nonContig = NULL); 426 427 // Explicit Routing 428 /** 429 * TODO Document this method. 430 */ 431 status_t setInputDevice(audio_port_handle_t deviceId); 432 433 /** 434 * TODO Document this method. 435 */ 436 audio_port_handle_t getInputDevice(); 437 438 /* Returns the ID of the audio device actually used by the input to which this AudioRecord 439 * is attached. 440 * The device ID is relevant only if the AudioRecord is active. 441 * When the AudioRecord is inactive, the device ID returned can be either: 442 * - AUDIO_PORT_HANDLE_NONE if the AudioRecord is not attached to any output. 443 * - The device ID used before paused or stopped. 444 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioRecord 445 * has not been started yet. 446 * 447 * Parameters: 448 * none. 449 */ 450 audio_port_handle_t getRoutedDeviceId(); 451 452 /* Add an AudioDeviceCallback. The caller will be notified when the audio device 453 * to which this AudioRecord is routed is updated. 454 * Replaces any previously installed callback. 455 * Parameters: 456 * callback: The callback interface 457 * Returns NO_ERROR if successful. 458 * INVALID_OPERATION if the same callback is already installed. 459 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 460 * BAD_VALUE if the callback is NULL 461 */ 462 status_t addAudioDeviceCallback( 463 const sp<AudioSystem::AudioDeviceCallback>& callback); 464 465 /* remove an AudioDeviceCallback. 466 * Parameters: 467 * callback: The callback interface 468 * Returns NO_ERROR if successful. 469 * INVALID_OPERATION if the callback is not installed 470 * BAD_VALUE if the callback is NULL 471 */ 472 status_t removeAudioDeviceCallback( 473 const sp<AudioSystem::AudioDeviceCallback>& callback); 474 475 // AudioSystem::AudioDeviceCallback> virtuals 476 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 477 audio_port_handle_t deviceId); 478 479private: 480 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 481 * additional non-contiguous frames that are predicted to be available immediately, 482 * if the client were to release the first frames and then call obtainBuffer() again. 483 * This value is only a prediction, and needs to be confirmed. 484 * It will be set to zero for an error return. 485 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 486 * in case the requested amount of frames is in two or more non-contiguous regions. 487 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 488 */ 489 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 490 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 491public: 492 493 /* Public API for TRANSFER_OBTAIN mode. 494 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. 495 * 496 * Buffer fields: 497 * frameCount currently ignored but recommend to set to actual number of frames consumed 498 * size actual number of bytes consumed, must be multiple of frameSize 499 * raw ignored 500 */ 501 void releaseBuffer(const Buffer* audioBuffer); 502 503 /* As a convenience we provide a read() interface to the audio buffer. 504 * Input parameter 'size' is in byte units. 505 * This is implemented on top of obtainBuffer/releaseBuffer. For best 506 * performance use callbacks. Returns actual number of bytes read >= 0, 507 * or one of the following negative status codes: 508 * INVALID_OPERATION AudioRecord is configured for streaming mode 509 * BAD_VALUE size is invalid 510 * WOULD_BLOCK when obtainBuffer() returns same, or 511 * AudioRecord was stopped during the read 512 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 513 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 514 * false for the method to return immediately without waiting to try multiple times to read 515 * the full content of the buffer. 516 */ 517 ssize_t read(void* buffer, size_t size, bool blocking = true); 518 519 /* Return the number of input frames lost in the audio driver since the last call of this 520 * function. Audio driver is expected to reset the value to 0 and restart counting upon 521 * returning the current value by this function call. Such loss typically occurs when the 522 * user space process is blocked longer than the capacity of audio driver buffers. 523 * Units: the number of input audio frames. 524 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 525 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 526 */ 527 uint32_t getInputFramesLost() const; 528 529 /* Get the flags */ 530 audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 531 532 /* Get active microphones. A empty vector of MicrophoneInfo will be passed as a parameter, 533 * the data will be filled when querying the hal. 534 */ 535 status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones); 536 537 /* 538 * Dumps the state of an audio record. 539 */ 540 status_t dump(int fd, const Vector<String16>& args) const; 541 542private: 543 /* copying audio record objects is not allowed */ 544 AudioRecord(const AudioRecord& other); 545 AudioRecord& operator = (const AudioRecord& other); 546 547 /* a small internal class to handle the callback */ 548 class AudioRecordThread : public Thread 549 { 550 public: 551 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 552 553 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 554 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 555 virtual void requestExit(); 556 557 void pause(); // suspend thread from execution at next loop boundary 558 void resume(); // allow thread to execute, if not requested to exit 559 void wake(); // wake to handle changed notification conditions. 560 561 private: 562 void pauseInternal(nsecs_t ns = 0LL); 563 // like pause(), but only used internally within thread 564 565 friend class AudioRecord; 566 virtual bool threadLoop(); 567 AudioRecord& mReceiver; 568 virtual ~AudioRecordThread(); 569 Mutex mMyLock; // Thread::mLock is private 570 Condition mMyCond; // Thread::mThreadExitedCondition is private 571 bool mPaused; // whether thread is requested to pause at next loop entry 572 bool mPausedInt; // whether thread internally requests pause 573 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 574 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 575 // to processAudioBuffer() as state may have changed 576 // since pause time calculated. 577 }; 578 579 // body of AudioRecordThread::threadLoop() 580 // returns the maximum amount of time before we would like to run again, where: 581 // 0 immediately 582 // > 0 no later than this many nanoseconds from now 583 // NS_WHENEVER still active but no particular deadline 584 // NS_INACTIVE inactive so don't run again until re-started 585 // NS_NEVER never again 586 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 587 nsecs_t processAudioBuffer(); 588 589 // caller must hold lock on mLock for all _l methods 590 591 status_t createRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName); 592 593 // FIXME enum is faster than strcmp() for parameter 'from' 594 status_t restoreRecord_l(const char *from); 595 596 void updateRoutedDeviceId_l(); 597 598 sp<AudioRecordThread> mAudioRecordThread; 599 mutable Mutex mLock; 600 601 // Current client state: false = stopped, true = active. Protected by mLock. If more states 602 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 603 bool mActive; 604 605 // for client callback handler 606 callback_t mCbf; // callback handler for events, or NULL 607 void* mUserData; 608 609 // for notification APIs 610 uint32_t mNotificationFramesReq; // requested number of frames between each 611 // notification callback 612 // as specified in constructor or set() 613 uint32_t mNotificationFramesAct; // actual number of frames between each 614 // notification callback 615 bool mRefreshRemaining; // processAudioBuffer() should refresh 616 // mRemainingFrames and mRetryOnPartialBuffer 617 618 // These are private to processAudioBuffer(), and are not protected by a lock 619 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 620 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 621 uint32_t mObservedSequence; // last observed value of mSequence 622 623 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 624 bool mMarkerReached; 625 Modulo<uint32_t> mNewPosition; // in frames 626 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 627 628 status_t mStatus; 629 630 String16 mOpPackageName; // The package name used for app ops. 631 632 size_t mFrameCount; // corresponds to current IAudioRecord, value is 633 // reported back by AudioFlinger to the client 634 size_t mReqFrameCount; // frame count to request the first or next time 635 // a new IAudioRecord is needed, non-decreasing 636 637 int64_t mFramesRead; // total frames read. reset to zero after 638 // the start() following stop(). It is not 639 // changed after restoring the track. 640 int64_t mFramesReadServerOffset; // An offset to server frames read due to 641 // restoring AudioRecord, or stop/start. 642 // constant after constructor or set() 643 uint32_t mSampleRate; 644 audio_format_t mFormat; 645 uint32_t mChannelCount; 646 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 647 uint32_t mLatency; // in ms 648 audio_channel_mask_t mChannelMask; 649 650 audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may 651 // be denied by client or server, such as 652 // AUDIO_INPUT_FLAG_FAST. mLock must be 653 // held to read or write those bits reliably. 654 audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const 655 656 audio_session_t mSessionId; 657 transfer_type mTransfer; 658 659 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 660 // provided the initial set() was successful 661 sp<media::IAudioRecord> mAudioRecord; 662 sp<IMemory> mCblkMemory; 663 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 664 sp<IMemory> mBufferMemory; 665 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 666 667 int mPreviousPriority; // before start() 668 SchedPolicy mPreviousSchedulingGroup; 669 bool mAwaitBoost; // thread should wait for priority boost before running 670 671 // The proxy should only be referenced while a lock is held because the proxy isn't 672 // multi-thread safe. 673 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 674 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 675 // them around in case they are replaced during the obtainBuffer(). 676 sp<AudioRecordClientProxy> mProxy; 677 678 bool mInOverrun; // whether recorder is currently in overrun state 679 680private: 681 class DeathNotifier : public IBinder::DeathRecipient { 682 public: 683 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 684 protected: 685 virtual void binderDied(const wp<IBinder>& who); 686 private: 687 const wp<AudioRecord> mAudioRecord; 688 }; 689 690 sp<DeathNotifier> mDeathNotifier; 691 uint32_t mSequence; // incremented for each new IAudioRecord attempt 692 uid_t mClientUid; 693 pid_t mClientPid; 694 audio_attributes_t mAttributes; 695 696 // For Device Selection API 697 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 698 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 699 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 700 // May not match the app selection depending on other 701 // activity and connected devices 702 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 703 704private: 705 class MediaMetrics { 706 public: 707 MediaMetrics() : mAnalyticsItem(new MediaAnalyticsItem("audiorecord")), 708 mCreatedNs(systemTime(SYSTEM_TIME_REALTIME)), 709 mStartedNs(0), mDurationNs(0), mCount(0), 710 mLastError(NO_ERROR) { 711 } 712 ~MediaMetrics() { 713 // mAnalyticsItem alloc failure will be flagged in the constructor 714 // don't log empty records 715 if (mAnalyticsItem->count() > 0) { 716 mAnalyticsItem->selfrecord(); 717 } 718 } 719 void gather(const AudioRecord *record); 720 MediaAnalyticsItem *dup() { return mAnalyticsItem->dup(); } 721 722 void logStart(nsecs_t when) { mStartedNs = when; mCount++; } 723 void logStop(nsecs_t when) { mDurationNs += (when-mStartedNs); mStartedNs = 0;} 724 void markError(status_t errcode, const char *func) 725 { mLastError = errcode; mLastErrorFunc = func;} 726 private: 727 std::unique_ptr<MediaAnalyticsItem> mAnalyticsItem; 728 nsecs_t mCreatedNs; // XXX: perhaps not worth it in production 729 nsecs_t mStartedNs; 730 nsecs_t mDurationNs; 731 int32_t mCount; 732 733 status_t mLastError; 734 std::string mLastErrorFunc; 735 }; 736 MediaMetrics mMediaMetrics; 737}; 738 739}; // namespace android 740 741#endif // ANDROID_AUDIORECORD_H 742