AudioRecord.h revision 551b5355d34aa42890811fc3606d3b63429296cd
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 25namespace android { 26 27// ---------------------------------------------------------------------------- 28 29struct audio_track_cblk_t; 30class AudioRecordClientProxy; 31 32// ---------------------------------------------------------------------------- 33 34class AudioRecord : public RefBase 35{ 36public: 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 AudioRecord(); 134 135 /* Creates an AudioRecord object and registers it with AudioFlinger. 136 * Once created, the track needs to be started before it can be used. 137 * Unspecified values are set to appropriate default values. 138 * 139 * Parameters: 140 * 141 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 142 * sampleRate: Data sink sampling rate in Hz. 143 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 144 * 16 bits per sample). 145 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 146 * frameCount: Minimum size of track PCM buffer in frames. This defines the 147 * application's contribution to the 148 * latency of the track. The actual size selected by the AudioRecord could 149 * be larger if the requested size is not compatible with current audio HAL 150 * latency. Zero means to use a default value. 151 * cbf: Callback function. If not null, this function is called periodically 152 * to consume new data in TRANSFER_CALLBACK mode 153 * and inform of marker, position updates, etc. 154 * user: Context for use by the callback receiver. 155 * notificationFrames: The callback function is called each time notificationFrames PCM 156 * frames are ready in record track output buffer. 157 * sessionId: Not yet supported. 158 * transferType: How data is transferred from AudioRecord. 159 * flags: See comments on audio_input_flags_t in <system/audio.h> 160 * pAttributes: If not NULL, supersedes inputSource for use case selection. 161 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 162 */ 163 164 AudioRecord(audio_source_t inputSource, 165 uint32_t sampleRate, 166 audio_format_t format, 167 audio_channel_mask_t channelMask, 168 size_t frameCount = 0, 169 callback_t cbf = NULL, 170 void* user = NULL, 171 uint32_t notificationFrames = 0, 172 int sessionId = AUDIO_SESSION_ALLOCATE, 173 transfer_type transferType = TRANSFER_DEFAULT, 174 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 175 const audio_attributes_t* pAttributes = NULL); 176 177 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 178 * Also destroys all resources associated with the AudioRecord. 179 */ 180protected: 181 virtual ~AudioRecord(); 182public: 183 184 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 185 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 186 * set() is not multi-thread safe. 187 * Returned status (from utils/Errors.h) can be: 188 * - NO_ERROR: successful intialization 189 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 190 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 191 * - NO_INIT: audio server or audio hardware not initialized 192 * - PERMISSION_DENIED: recording is not allowed for the requesting process 193 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 194 * 195 * Parameters not listed in the AudioRecord constructors above: 196 * 197 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 198 */ 199 status_t set(audio_source_t inputSource, 200 uint32_t sampleRate, 201 audio_format_t format, 202 audio_channel_mask_t channelMask, 203 size_t frameCount = 0, 204 callback_t cbf = NULL, 205 void* user = NULL, 206 uint32_t notificationFrames = 0, 207 bool threadCanCallJava = false, 208 int sessionId = AUDIO_SESSION_ALLOCATE, 209 transfer_type transferType = TRANSFER_DEFAULT, 210 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 211 const audio_attributes_t* pAttributes = NULL); 212 213 /* Result of constructing the AudioRecord. This must be checked for successful initialization 214 * before using any AudioRecord API (except for set()), because using 215 * an uninitialized AudioRecord produces undefined results. 216 * See set() method above for possible return codes. 217 */ 218 status_t initCheck() const { return mStatus; } 219 220 /* Returns this track's estimated latency in milliseconds. 221 * This includes the latency due to AudioRecord buffer size, resampling if applicable, 222 * and audio hardware driver. 223 */ 224 uint32_t latency() const { return mLatency; } 225 226 /* getters, see constructor and set() */ 227 228 audio_format_t format() const { return mFormat; } 229 uint32_t channelCount() const { return mChannelCount; } 230 size_t frameCount() const { return mFrameCount; } 231 size_t frameSize() const { return mFrameSize; } 232 audio_source_t inputSource() const { return mAttributes.source; } 233 234 /* After it's created the track is not active. Call start() to 235 * make it active. If set, the callback will start being called. 236 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 237 * the specified event occurs on the specified trigger session. 238 */ 239 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 240 int triggerSession = 0); 241 242 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 243 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 244 */ 245 void stop(); 246 bool stopped() const; 247 248 /* Return the sink sample rate for this record track in Hz. 249 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 250 */ 251 uint32_t getSampleRate() const { return mSampleRate; } 252 253 /* Sets marker position. When record reaches the number of frames specified, 254 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 255 * with marker == 0 cancels marker notification callback. 256 * To set a marker at a position which would compute as 0, 257 * a workaround is to set the marker at a nearby position such as ~0 or 1. 258 * If the AudioRecord has been opened with no callback function associated, 259 * the operation will fail. 260 * 261 * Parameters: 262 * 263 * marker: marker position expressed in wrapping (overflow) frame units, 264 * like the return value of getPosition(). 265 * 266 * Returned status (from utils/Errors.h) can be: 267 * - NO_ERROR: successful operation 268 * - INVALID_OPERATION: the AudioRecord has no callback installed. 269 */ 270 status_t setMarkerPosition(uint32_t marker); 271 status_t getMarkerPosition(uint32_t *marker) const; 272 273 /* Sets position update period. Every time the number of frames specified has been recorded, 274 * a callback with event type EVENT_NEW_POS is called. 275 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 276 * callback. 277 * If the AudioRecord has been opened with no callback function associated, 278 * the operation will fail. 279 * Extremely small values may be rounded up to a value the implementation can support. 280 * 281 * Parameters: 282 * 283 * updatePeriod: position update notification period expressed in frames. 284 * 285 * Returned status (from utils/Errors.h) can be: 286 * - NO_ERROR: successful operation 287 * - INVALID_OPERATION: the AudioRecord has no callback installed. 288 */ 289 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 290 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 291 292 /* Return the total number of frames recorded since recording started. 293 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 294 * It is reset to zero by stop(). 295 * 296 * Parameters: 297 * 298 * position: Address where to return record head position. 299 * 300 * Returned status (from utils/Errors.h) can be: 301 * - NO_ERROR: successful operation 302 * - BAD_VALUE: position is NULL 303 */ 304 status_t getPosition(uint32_t *position) const; 305 306 /* Returns a handle on the audio input used by this AudioRecord. 307 * 308 * Parameters: 309 * none. 310 * 311 * Returned value: 312 * handle on audio hardware input 313 */ 314// FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp 315 audio_io_handle_t getInput() const __attribute__((__deprecated__)) 316 { return getInputPrivate(); } 317private: 318 audio_io_handle_t getInputPrivate() const; 319public: 320 321 /* Returns the audio session ID associated with this AudioRecord. 322 * 323 * Parameters: 324 * none. 325 * 326 * Returned value: 327 * AudioRecord session ID. 328 * 329 * No lock needed because session ID doesn't change after first set(). 330 */ 331 int getSessionId() const { return mSessionId; } 332 333 /* Public API for TRANSFER_OBTAIN mode. 334 * Obtains a buffer of up to "audioBuffer->frameCount" full frames. 335 * After draining these frames of data, the caller should release them with releaseBuffer(). 336 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 337 * full frames as are available immediately. 338 * 339 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 340 * additional non-contiguous frames that are predicted to be available immediately, 341 * if the client were to release the first frames and then call obtainBuffer() again. 342 * This value is only a prediction, and needs to be confirmed. 343 * It will be set to zero for an error return. 344 * 345 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 346 * regardless of the value of waitCount. 347 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 348 * maximum timeout based on waitCount; see chart below. 349 * Buffers will be returned until the pool 350 * is exhausted, at which point obtainBuffer() will either block 351 * or return WOULD_BLOCK depending on the value of the "waitCount" 352 * parameter. 353 * 354 * Interpretation of waitCount: 355 * +n limits wait time to n * WAIT_PERIOD_MS, 356 * -1 causes an (almost) infinite wait time, 357 * 0 non-blocking. 358 * 359 * Buffer fields 360 * On entry: 361 * frameCount number of frames requested 362 * size ignored 363 * raw ignored 364 * After error return: 365 * frameCount 0 366 * size 0 367 * raw undefined 368 * After successful return: 369 * frameCount actual number of frames available, <= number requested 370 * size actual number of bytes available 371 * raw pointer to the buffer 372 */ 373 374 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 375 size_t *nonContig = NULL); 376 377private: 378 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 379 * additional non-contiguous frames that are predicted to be available immediately, 380 * if the client were to release the first frames and then call obtainBuffer() again. 381 * This value is only a prediction, and needs to be confirmed. 382 * It will be set to zero for an error return. 383 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 384 * in case the requested amount of frames is in two or more non-contiguous regions. 385 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 386 */ 387 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 388 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 389public: 390 391 /* Public API for TRANSFER_OBTAIN mode. 392 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. 393 * 394 * Buffer fields: 395 * frameCount currently ignored but recommend to set to actual number of frames consumed 396 * size actual number of bytes consumed, must be multiple of frameSize 397 * raw ignored 398 */ 399 void releaseBuffer(const Buffer* audioBuffer); 400 401 /* As a convenience we provide a read() interface to the audio buffer. 402 * Input parameter 'size' is in byte units. 403 * This is implemented on top of obtainBuffer/releaseBuffer. For best 404 * performance use callbacks. Returns actual number of bytes read >= 0, 405 * or one of the following negative status codes: 406 * INVALID_OPERATION AudioRecord is configured for streaming mode 407 * BAD_VALUE size is invalid 408 * WOULD_BLOCK when obtainBuffer() returns same, or 409 * AudioRecord was stopped during the read 410 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 411 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 412 * false for the method to return immediately without waiting to try multiple times to read 413 * the full content of the buffer. 414 */ 415 ssize_t read(void* buffer, size_t size, bool blocking = true); 416 417 /* Return the number of input frames lost in the audio driver since the last call of this 418 * function. Audio driver is expected to reset the value to 0 and restart counting upon 419 * returning the current value by this function call. Such loss typically occurs when the 420 * user space process is blocked longer than the capacity of audio driver buffers. 421 * Units: the number of input audio frames. 422 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 423 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 424 */ 425 uint32_t getInputFramesLost() const; 426 427private: 428 /* copying audio record objects is not allowed */ 429 AudioRecord(const AudioRecord& other); 430 AudioRecord& operator = (const AudioRecord& other); 431 432 /* a small internal class to handle the callback */ 433 class AudioRecordThread : public Thread 434 { 435 public: 436 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 437 438 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 439 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 440 virtual void requestExit(); 441 442 void pause(); // suspend thread from execution at next loop boundary 443 void resume(); // allow thread to execute, if not requested to exit 444 void wake(); // wake to handle changed notification conditions. 445 446 private: 447 void pauseInternal(nsecs_t ns = 0LL); 448 // like pause(), but only used internally within thread 449 450 friend class AudioRecord; 451 virtual bool threadLoop(); 452 AudioRecord& mReceiver; 453 virtual ~AudioRecordThread(); 454 Mutex mMyLock; // Thread::mLock is private 455 Condition mMyCond; // Thread::mThreadExitedCondition is private 456 bool mPaused; // whether thread is requested to pause at next loop entry 457 bool mPausedInt; // whether thread internally requests pause 458 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 459 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 460 // to processAudioBuffer() as state may have changed 461 // since pause time calculated. 462 }; 463 464 // body of AudioRecordThread::threadLoop() 465 // returns the maximum amount of time before we would like to run again, where: 466 // 0 immediately 467 // > 0 no later than this many nanoseconds from now 468 // NS_WHENEVER still active but no particular deadline 469 // NS_INACTIVE inactive so don't run again until re-started 470 // NS_NEVER never again 471 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 472 nsecs_t processAudioBuffer(); 473 474 // caller must hold lock on mLock for all _l methods 475 476 status_t openRecord_l(size_t epoch); 477 478 // FIXME enum is faster than strcmp() for parameter 'from' 479 status_t restoreRecord_l(const char *from); 480 481 sp<AudioRecordThread> mAudioRecordThread; 482 mutable Mutex mLock; 483 484 // Current client state: false = stopped, true = active. Protected by mLock. If more states 485 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 486 bool mActive; 487 488 // for client callback handler 489 callback_t mCbf; // callback handler for events, or NULL 490 void* mUserData; 491 492 // for notification APIs 493 uint32_t mNotificationFramesReq; // requested number of frames between each 494 // notification callback 495 // as specified in constructor or set() 496 uint32_t mNotificationFramesAct; // actual number of frames between each 497 // notification callback 498 bool mRefreshRemaining; // processAudioBuffer() should refresh 499 // mRemainingFrames and mRetryOnPartialBuffer 500 501 // These are private to processAudioBuffer(), and are not protected by a lock 502 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 503 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 504 uint32_t mObservedSequence; // last observed value of mSequence 505 506 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 507 bool mMarkerReached; 508 uint32_t mNewPosition; // in frames 509 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 510 511 status_t mStatus; 512 513 size_t mFrameCount; // corresponds to current IAudioRecord, value is 514 // reported back by AudioFlinger to the client 515 size_t mReqFrameCount; // frame count to request the first or next time 516 // a new IAudioRecord is needed, non-decreasing 517 518 // constant after constructor or set() 519 uint32_t mSampleRate; 520 audio_format_t mFormat; 521 uint32_t mChannelCount; 522 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 523 uint32_t mLatency; // in ms 524 audio_channel_mask_t mChannelMask; 525 audio_input_flags_t mFlags; 526 int mSessionId; 527 transfer_type mTransfer; 528 529 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 530 // provided the initial set() was successful 531 sp<IAudioRecord> mAudioRecord; 532 sp<IMemory> mCblkMemory; 533 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 534 sp<IMemory> mBufferMemory; 535 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 536 537 int mPreviousPriority; // before start() 538 SchedPolicy mPreviousSchedulingGroup; 539 bool mAwaitBoost; // thread should wait for priority boost before running 540 541 // The proxy should only be referenced while a lock is held because the proxy isn't 542 // multi-thread safe. 543 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 544 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 545 // them around in case they are replaced during the obtainBuffer(). 546 sp<AudioRecordClientProxy> mProxy; 547 548 bool mInOverrun; // whether recorder is currently in overrun state 549 550private: 551 class DeathNotifier : public IBinder::DeathRecipient { 552 public: 553 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 554 protected: 555 virtual void binderDied(const wp<IBinder>& who); 556 private: 557 const wp<AudioRecord> mAudioRecord; 558 }; 559 560 sp<DeathNotifier> mDeathNotifier; 561 uint32_t mSequence; // incremented for each new IAudioRecord attempt 562 audio_attributes_t mAttributes; 563}; 564 565}; // namespace android 566 567#endif // ANDROID_AUDIORECORD_H 568