AudioRecord.h revision d198b85a163330b03e7507c9e8bfeb5f4d958a6c
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 46 // ignore the event by setting frameCount to zero. 47 EVENT_OVERRUN = 1, // Buffer overrun occurred. 48 EVENT_MARKER = 2, // Record head is at the specified marker position 49 // (See setMarkerPosition()). 50 EVENT_NEW_POS = 3, // Record head is at a new position 51 // (See setPositionUpdatePeriod()). 52 EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and 53 // voluntary invalidation by mediaserver, or mediaserver crash. 54 }; 55 56 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 57 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 58 */ 59 60 class Buffer 61 { 62 public: 63 // FIXME use m prefix 64 size_t frameCount; // number of sample frames corresponding to size; 65 // on input it is the number of frames available, 66 // on output is the number of frames actually drained 67 // (currently ignored but will make the primary field in future) 68 69 size_t size; // input/output in bytes == frameCount * frameSize 70 // on output is the number of bytes actually drained 71 // FIXME this is redundant with respect to frameCount, 72 // and TRANSFER_OBTAIN mode is broken for 8-bit data 73 // since we don't define the frame format 74 75 union { 76 void* raw; 77 short* i16; // signed 16-bit 78 int8_t* i8; // unsigned 8-bit, offset by 0x80 79 }; 80 }; 81 82 /* As a convenience, if a callback is supplied, a handler thread 83 * is automatically created with the appropriate priority. This thread 84 * invokes the callback when a new buffer becomes available or various conditions occur. 85 * Parameters: 86 * 87 * event: type of event notified (see enum AudioRecord::event_type). 88 * user: Pointer to context for use by the callback receiver. 89 * info: Pointer to optional parameter according to event type: 90 * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read 91 * more bytes than indicated by 'size' field and update 'size' if 92 * fewer bytes are consumed. 93 * - EVENT_OVERRUN: unused. 94 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 95 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 96 * - EVENT_NEW_IAUDIORECORD: unused. 97 */ 98 99 typedef void (*callback_t)(int event, void* user, void *info); 100 101 /* Returns the minimum frame count required for the successful creation of 102 * an AudioRecord object. 103 * Returned status (from utils/Errors.h) can be: 104 * - NO_ERROR: successful operation 105 * - NO_INIT: audio server or audio hardware not initialized 106 * - BAD_VALUE: unsupported configuration 107 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 108 * and is undefined otherwise. 109 * FIXME This API assumes a route, and so should be deprecated. 110 */ 111 112 static status_t getMinFrameCount(size_t* frameCount, 113 uint32_t sampleRate, 114 audio_format_t format, 115 audio_channel_mask_t channelMask); 116 117 /* How data is transferred from AudioRecord 118 */ 119 enum transfer_type { 120 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 121 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 122 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 123 TRANSFER_SYNC, // synchronous read() 124 }; 125 126 /* Constructs an uninitialized AudioRecord. No connection with 127 * AudioFlinger takes place. Use set() after this. 128 */ 129 AudioRecord(); 130 131 /* Creates an AudioRecord object and registers it with AudioFlinger. 132 * Once created, the track needs to be started before it can be used. 133 * Unspecified values are set to appropriate default values. 134 * 135 * Parameters: 136 * 137 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 138 * sampleRate: Data sink sampling rate in Hz. 139 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 140 * 16 bits per sample). 141 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 142 * frameCount: Minimum size of track PCM buffer in frames. This defines the 143 * application's contribution to the 144 * latency of the track. The actual size selected by the AudioRecord could 145 * be larger if the requested size is not compatible with current audio HAL 146 * latency. Zero means to use a default value. 147 * cbf: Callback function. If not null, this function is called periodically 148 * to consume new data and inform of marker, position updates, etc. 149 * user: Context for use by the callback receiver. 150 * notificationFrames: The callback function is called each time notificationFrames PCM 151 * frames are ready in record track output buffer. 152 * sessionId: Not yet supported. 153 * transferType: How data is transferred from AudioRecord. 154 * flags: See comments on audio_input_flags_t in <system/audio.h> 155 * pAttributes: If not NULL, supersedes inputSource for use case selection. 156 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 157 */ 158 159 AudioRecord(audio_source_t inputSource, 160 uint32_t sampleRate, 161 audio_format_t format, 162 audio_channel_mask_t channelMask, 163 size_t frameCount = 0, 164 callback_t cbf = NULL, 165 void* user = NULL, 166 uint32_t notificationFrames = 0, 167 int sessionId = AUDIO_SESSION_ALLOCATE, 168 transfer_type transferType = TRANSFER_DEFAULT, 169 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 170 const audio_attributes_t* pAttributes = NULL); 171 172 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 173 * Also destroys all resources associated with the AudioRecord. 174 */ 175protected: 176 virtual ~AudioRecord(); 177public: 178 179 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 180 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 181 * Returned status (from utils/Errors.h) can be: 182 * - NO_ERROR: successful intialization 183 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 184 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 185 * - NO_INIT: audio server or audio hardware not initialized 186 * - PERMISSION_DENIED: recording is not allowed for the requesting process 187 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 188 * 189 * Parameters not listed in the AudioRecord constructors above: 190 * 191 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 192 */ 193 status_t set(audio_source_t inputSource, 194 uint32_t sampleRate, 195 audio_format_t format, 196 audio_channel_mask_t channelMask, 197 size_t frameCount = 0, 198 callback_t cbf = NULL, 199 void* user = NULL, 200 uint32_t notificationFrames = 0, 201 bool threadCanCallJava = false, 202 int sessionId = AUDIO_SESSION_ALLOCATE, 203 transfer_type transferType = TRANSFER_DEFAULT, 204 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 205 const audio_attributes_t* pAttributes = NULL); 206 207 /* Result of constructing the AudioRecord. This must be checked for successful initialization 208 * before using any AudioRecord API (except for set()), because using 209 * an uninitialized AudioRecord produces undefined results. 210 * See set() method above for possible return codes. 211 */ 212 status_t initCheck() const { return mStatus; } 213 214 /* Returns this track's estimated latency in milliseconds. 215 * This includes the latency due to AudioRecord buffer size, 216 * and audio hardware driver. 217 */ 218 uint32_t latency() const { return mLatency; } 219 220 /* getters, see constructor and set() */ 221 222 audio_format_t format() const { return mFormat; } 223 uint32_t channelCount() const { return mChannelCount; } 224 size_t frameCount() const { return mFrameCount; } 225 size_t frameSize() const { return mFrameSize; } 226 audio_source_t inputSource() const { return mAttributes.source; } 227 228 /* After it's created the track is not active. Call start() to 229 * make it active. If set, the callback will start being called. 230 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 231 * the specified event occurs on the specified trigger session. 232 */ 233 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 234 int triggerSession = 0); 235 236 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 237 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 238 */ 239 void stop(); 240 bool stopped() const; 241 242 /* Return the sink sample rate for this record track in Hz. 243 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 244 */ 245 uint32_t getSampleRate() const { return mSampleRate; } 246 247 /* Return the notification frame count. 248 * This is approximately how often the callback is invoked, for transfer type TRANSFER_CALLBACK. 249 */ 250 size_t notificationFrames() const { return mNotificationFramesAct; } 251 252 /* Sets marker position. When record reaches the number of frames specified, 253 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 254 * with marker == 0 cancels marker notification callback. 255 * To set a marker at a position which would compute as 0, 256 * a workaround is to set the marker at a nearby position such as ~0 or 1. 257 * If the AudioRecord has been opened with no callback function associated, 258 * the operation will fail. 259 * 260 * Parameters: 261 * 262 * marker: marker position expressed in wrapping (overflow) frame units, 263 * like the return value of getPosition(). 264 * 265 * Returned status (from utils/Errors.h) can be: 266 * - NO_ERROR: successful operation 267 * - INVALID_OPERATION: the AudioRecord has no callback installed. 268 */ 269 status_t setMarkerPosition(uint32_t marker); 270 status_t getMarkerPosition(uint32_t *marker) const; 271 272 /* Sets position update period. Every time the number of frames specified has been recorded, 273 * a callback with event type EVENT_NEW_POS is called. 274 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 275 * callback. 276 * If the AudioRecord has been opened with no callback function associated, 277 * the operation will fail. 278 * Extremely small values may be rounded up to a value the implementation can support. 279 * 280 * Parameters: 281 * 282 * updatePeriod: position update notification period expressed in frames. 283 * 284 * Returned status (from utils/Errors.h) can be: 285 * - NO_ERROR: successful operation 286 * - INVALID_OPERATION: the AudioRecord has no callback installed. 287 */ 288 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 289 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 290 291 /* Return the total number of frames recorded since recording started. 292 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 293 * It is reset to zero by stop(). 294 * 295 * Parameters: 296 * 297 * position: Address where to return record head position. 298 * 299 * Returned status (from utils/Errors.h) can be: 300 * - NO_ERROR: successful operation 301 * - BAD_VALUE: position is NULL 302 */ 303 status_t getPosition(uint32_t *position) const; 304 305 /* Returns a handle on the audio input used by this AudioRecord. 306 * 307 * Parameters: 308 * none. 309 * 310 * Returned value: 311 * handle on audio hardware input 312 */ 313 audio_io_handle_t getInput() const; 314 315 /* Returns the audio session ID associated with this AudioRecord. 316 * 317 * Parameters: 318 * none. 319 * 320 * Returned value: 321 * AudioRecord session ID. 322 * 323 * No lock needed because session ID doesn't change after first set(). 324 */ 325 int getSessionId() const { return mSessionId; } 326 327 /* Obtains a buffer of up to "audioBuffer->frameCount" full frames. 328 * After draining these frames of data, the caller should release them with releaseBuffer(). 329 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 330 * full frames as are available immediately. 331 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 332 * regardless of the value of waitCount. 333 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 334 * maximum timeout based on waitCount; see chart below. 335 * Buffers will be returned until the pool 336 * is exhausted, at which point obtainBuffer() will either block 337 * or return WOULD_BLOCK depending on the value of the "waitCount" 338 * parameter. 339 * 340 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 341 * which should use read() or callback EVENT_MORE_DATA instead. 342 * 343 * Interpretation of waitCount: 344 * +n limits wait time to n * WAIT_PERIOD_MS, 345 * -1 causes an (almost) infinite wait time, 346 * 0 non-blocking. 347 * 348 * Buffer fields 349 * On entry: 350 * frameCount number of frames requested 351 * After error return: 352 * frameCount 0 353 * size 0 354 * raw undefined 355 * After successful return: 356 * frameCount actual number of frames available, <= number requested 357 * size actual number of bytes available 358 * raw pointer to the buffer 359 */ 360 361 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 362 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 363 __attribute__((__deprecated__)); 364 365private: 366 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 367 * additional non-contiguous frames that are available immediately. 368 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 369 * in case the requested amount of frames is in two or more non-contiguous regions. 370 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 371 */ 372 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 373 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 374public: 375 376 /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */ 377 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 378 void releaseBuffer(Buffer* audioBuffer); 379 380 /* As a convenience we provide a read() interface to the audio buffer. 381 * Input parameter 'size' is in byte units. 382 * This is implemented on top of obtainBuffer/releaseBuffer. For best 383 * performance use callbacks. Returns actual number of bytes read >= 0, 384 * or one of the following negative status codes: 385 * INVALID_OPERATION AudioRecord is configured for streaming mode 386 * BAD_VALUE size is invalid 387 * WOULD_BLOCK when obtainBuffer() returns same, or 388 * AudioRecord was stopped during the read 389 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 390 */ 391 ssize_t read(void* buffer, size_t size); 392 393 /* Return the number of input frames lost in the audio driver since the last call of this 394 * function. Audio driver is expected to reset the value to 0 and restart counting upon 395 * returning the current value by this function call. Such loss typically occurs when the 396 * user space process is blocked longer than the capacity of audio driver buffers. 397 * Units: the number of input audio frames. 398 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 399 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 400 */ 401 uint32_t getInputFramesLost() const; 402 403private: 404 /* copying audio record objects is not allowed */ 405 AudioRecord(const AudioRecord& other); 406 AudioRecord& operator = (const AudioRecord& other); 407 408 /* a small internal class to handle the callback */ 409 class AudioRecordThread : public Thread 410 { 411 public: 412 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 413 414 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 415 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 416 virtual void requestExit(); 417 418 void pause(); // suspend thread from execution at next loop boundary 419 void resume(); // allow thread to execute, if not requested to exit 420 void wake(); // wake to handle changed notification conditions. 421 422 private: 423 void pauseInternal(nsecs_t ns = 0LL); 424 // like pause(), but only used internally within thread 425 426 friend class AudioRecord; 427 virtual bool threadLoop(); 428 AudioRecord& mReceiver; 429 virtual ~AudioRecordThread(); 430 Mutex mMyLock; // Thread::mLock is private 431 Condition mMyCond; // Thread::mThreadExitedCondition is private 432 bool mPaused; // whether thread is requested to pause at next loop entry 433 bool mPausedInt; // whether thread internally requests pause 434 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 435 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 436 // to processAudioBuffer() as state may have changed 437 // since pause time calculated. 438 }; 439 440 // body of AudioRecordThread::threadLoop() 441 // returns the maximum amount of time before we would like to run again, where: 442 // 0 immediately 443 // > 0 no later than this many nanoseconds from now 444 // NS_WHENEVER still active but no particular deadline 445 // NS_INACTIVE inactive so don't run again until re-started 446 // NS_NEVER never again 447 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 448 nsecs_t processAudioBuffer(); 449 450 // caller must hold lock on mLock for all _l methods 451 452 status_t openRecord_l(size_t epoch); 453 454 // FIXME enum is faster than strcmp() for parameter 'from' 455 status_t restoreRecord_l(const char *from); 456 457 sp<AudioRecordThread> mAudioRecordThread; 458 mutable Mutex mLock; 459 460 // Current client state: false = stopped, true = active. Protected by mLock. If more states 461 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 462 bool mActive; 463 464 // for client callback handler 465 callback_t mCbf; // callback handler for events, or NULL 466 void* mUserData; 467 468 // for notification APIs 469 uint32_t mNotificationFramesReq; // requested number of frames between each 470 // notification callback 471 // as specified in constructor or set() 472 uint32_t mNotificationFramesAct; // actual number of frames between each 473 // notification callback 474 bool mRefreshRemaining; // processAudioBuffer() should refresh 475 // mRemainingFrames and mRetryOnPartialBuffer 476 477 // These are private to processAudioBuffer(), and are not protected by a lock 478 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 479 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 480 uint32_t mObservedSequence; // last observed value of mSequence 481 482 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 483 bool mMarkerReached; 484 uint32_t mNewPosition; // in frames 485 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 486 487 status_t mStatus; 488 489 size_t mFrameCount; // corresponds to current IAudioRecord, value is 490 // reported back by AudioFlinger to the client 491 size_t mReqFrameCount; // frame count to request the first or next time 492 // a new IAudioRecord is needed, non-decreasing 493 494 // constant after constructor or set() 495 uint32_t mSampleRate; 496 audio_format_t mFormat; 497 uint32_t mChannelCount; 498 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 499 uint32_t mLatency; // in ms 500 audio_channel_mask_t mChannelMask; 501 audio_input_flags_t mFlags; 502 int mSessionId; 503 transfer_type mTransfer; 504 505 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 506 // provided the initial set() was successful 507 sp<IAudioRecord> mAudioRecord; 508 sp<IMemory> mCblkMemory; 509 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 510 sp<IMemory> mBufferMemory; 511 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 512 513 int mPreviousPriority; // before start() 514 SchedPolicy mPreviousSchedulingGroup; 515 bool mAwaitBoost; // thread should wait for priority boost before running 516 517 // The proxy should only be referenced while a lock is held because the proxy isn't 518 // multi-thread safe. 519 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 520 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 521 // them around in case they are replaced during the obtainBuffer(). 522 sp<AudioRecordClientProxy> mProxy; 523 524 bool mInOverrun; // whether recorder is currently in overrun state 525 526private: 527 class DeathNotifier : public IBinder::DeathRecipient { 528 public: 529 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 530 protected: 531 virtual void binderDied(const wp<IBinder>& who); 532 private: 533 const wp<AudioRecord> mAudioRecord; 534 }; 535 536 sp<DeathNotifier> mDeathNotifier; 537 uint32_t mSequence; // incremented for each new IAudioRecord attempt 538 audio_attributes_t mAttributes; 539}; 540 541}; // namespace android 542 543#endif // ANDROID_AUDIORECORD_H 544