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 25 namespace android { 26 27 // ---------------------------------------------------------------------------- 28 29 struct audio_track_cblk_t; 30 class AudioRecordClientProxy; 31 32 // ---------------------------------------------------------------------------- 33 34 class AudioRecord : public RefBase 35 { 36 public: 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 fewer bytes are 92 * 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 */ 110 111 static status_t getMinFrameCount(size_t* frameCount, 112 uint32_t sampleRate, 113 audio_format_t format, 114 audio_channel_mask_t channelMask); 115 116 /* How data is transferred from AudioRecord 117 */ 118 enum transfer_type { 119 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 120 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 121 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 122 TRANSFER_SYNC, // synchronous read() 123 }; 124 125 /* Constructs an uninitialized AudioRecord. No connection with 126 * AudioFlinger takes place. Use set() after this. 127 */ 128 AudioRecord(); 129 130 /* Creates an AudioRecord object and registers it with AudioFlinger. 131 * Once created, the track needs to be started before it can be used. 132 * Unspecified values are set to appropriate default values. 133 * 134 * Parameters: 135 * 136 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 137 * sampleRate: Data sink sampling rate in Hz. 138 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 139 * 16 bits per sample). 140 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 141 * frameCount: Minimum size of track PCM buffer in frames. This defines the 142 * application's contribution to the 143 * latency of the track. The actual size selected by the AudioRecord could 144 * be larger if the requested size is not compatible with current audio HAL 145 * latency. Zero means to use a default value. 146 * cbf: Callback function. If not null, this function is called periodically 147 * to consume new data and inform of marker, position updates, etc. 148 * user: Context for use by the callback receiver. 149 * notificationFrames: The callback function is called each time notificationFrames PCM 150 * frames are ready in record track output buffer. 151 * sessionId: Not yet supported. 152 * transferType: How data is transferred from AudioRecord. 153 * flags: See comments on audio_input_flags_t in <system/audio.h> 154 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 155 */ 156 157 AudioRecord(audio_source_t inputSource, 158 uint32_t sampleRate, 159 audio_format_t format, 160 audio_channel_mask_t channelMask, 161 size_t frameCount = 0, 162 callback_t cbf = NULL, 163 void* user = NULL, 164 uint32_t notificationFrames = 0, 165 int sessionId = AUDIO_SESSION_ALLOCATE, 166 transfer_type transferType = TRANSFER_DEFAULT, 167 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE); 168 169 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 170 * Also destroys all resources associated with the AudioRecord. 171 */ 172 protected: 173 virtual ~AudioRecord(); 174 public: 175 176 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 177 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 178 * Returned status (from utils/Errors.h) can be: 179 * - NO_ERROR: successful intialization 180 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 181 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 182 * - NO_INIT: audio server or audio hardware not initialized 183 * - PERMISSION_DENIED: recording is not allowed for the requesting process 184 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 185 * 186 * Parameters not listed in the AudioRecord constructors above: 187 * 188 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 189 */ 190 status_t set(audio_source_t inputSource, 191 uint32_t sampleRate, 192 audio_format_t format, 193 audio_channel_mask_t channelMask, 194 size_t frameCount = 0, 195 callback_t cbf = NULL, 196 void* user = NULL, 197 uint32_t notificationFrames = 0, 198 bool threadCanCallJava = false, 199 int sessionId = AUDIO_SESSION_ALLOCATE, 200 transfer_type transferType = TRANSFER_DEFAULT, 201 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE); 202 203 /* Result of constructing the AudioRecord. This must be checked for successful initialization 204 * before using any AudioRecord API (except for set()), because using 205 * an uninitialized AudioRecord produces undefined results. 206 * See set() method above for possible return codes. 207 */ initCheck()208 status_t initCheck() const { return mStatus; } 209 210 /* Returns this track's estimated latency in milliseconds. 211 * This includes the latency due to AudioRecord buffer size, 212 * and audio hardware driver. 213 */ latency()214 uint32_t latency() const { return mLatency; } 215 216 /* getters, see constructor and set() */ 217 format()218 audio_format_t format() const { return mFormat; } channelCount()219 uint32_t channelCount() const { return mChannelCount; } frameCount()220 size_t frameCount() const { return mFrameCount; } frameSize()221 size_t frameSize() const { return mFrameSize; } inputSource()222 audio_source_t inputSource() const { return mInputSource; } 223 224 /* After it's created the track is not active. Call start() to 225 * make it active. If set, the callback will start being called. 226 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 227 * the specified event occurs on the specified trigger session. 228 */ 229 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 230 int triggerSession = 0); 231 232 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 233 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 234 */ 235 void stop(); 236 bool stopped() const; 237 238 /* Return the sink sample rate for this record track in Hz. 239 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 240 */ getSampleRate()241 uint32_t getSampleRate() const { return mSampleRate; } 242 243 /* Return the notification frame count. 244 * This is approximately how often the callback is invoked, for transfer type TRANSFER_CALLBACK. 245 */ notificationFrames()246 size_t notificationFrames() const { return mNotificationFramesAct; } 247 248 /* Sets marker position. When record reaches the number of frames specified, 249 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 250 * with marker == 0 cancels marker notification callback. 251 * To set a marker at a position which would compute as 0, 252 * a workaround is to set the marker at a nearby position such as ~0 or 1. 253 * If the AudioRecord has been opened with no callback function associated, 254 * the operation will fail. 255 * 256 * Parameters: 257 * 258 * marker: marker position expressed in wrapping (overflow) frame units, 259 * like the return value of getPosition(). 260 * 261 * Returned status (from utils/Errors.h) can be: 262 * - NO_ERROR: successful operation 263 * - INVALID_OPERATION: the AudioRecord has no callback installed. 264 */ 265 status_t setMarkerPosition(uint32_t marker); 266 status_t getMarkerPosition(uint32_t *marker) const; 267 268 /* Sets position update period. Every time the number of frames specified has been recorded, 269 * a callback with event type EVENT_NEW_POS is called. 270 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 271 * callback. 272 * If the AudioRecord has been opened with no callback function associated, 273 * the operation will fail. 274 * Extremely small values may be rounded up to a value the implementation can support. 275 * 276 * Parameters: 277 * 278 * updatePeriod: position update notification period expressed in frames. 279 * 280 * Returned status (from utils/Errors.h) can be: 281 * - NO_ERROR: successful operation 282 * - INVALID_OPERATION: the AudioRecord has no callback installed. 283 */ 284 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 285 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 286 287 /* Return the total number of frames recorded since recording started. 288 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 289 * It is reset to zero by stop(). 290 * 291 * Parameters: 292 * 293 * position: Address where to return record head position. 294 * 295 * Returned status (from utils/Errors.h) can be: 296 * - NO_ERROR: successful operation 297 * - BAD_VALUE: position is NULL 298 */ 299 status_t getPosition(uint32_t *position) const; 300 301 /* Returns a handle on the audio input used by this AudioRecord. 302 * 303 * Parameters: 304 * none. 305 * 306 * Returned value: 307 * handle on audio hardware input 308 */ 309 audio_io_handle_t getInput() const; 310 311 /* Returns the audio session ID associated with this AudioRecord. 312 * 313 * Parameters: 314 * none. 315 * 316 * Returned value: 317 * AudioRecord session ID. 318 * 319 * No lock needed because session ID doesn't change after first set(). 320 */ getSessionId()321 int getSessionId() const { return mSessionId; } 322 323 /* Obtains a buffer of up to "audioBuffer->frameCount" full frames. 324 * After draining these frames of data, the caller should release them with releaseBuffer(). 325 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 326 * full frames as are available immediately. 327 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 328 * regardless of the value of waitCount. 329 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 330 * maximum timeout based on waitCount; see chart below. 331 * Buffers will be returned until the pool 332 * is exhausted, at which point obtainBuffer() will either block 333 * or return WOULD_BLOCK depending on the value of the "waitCount" 334 * parameter. 335 * 336 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 337 * which should use read() or callback EVENT_MORE_DATA instead. 338 * 339 * Interpretation of waitCount: 340 * +n limits wait time to n * WAIT_PERIOD_MS, 341 * -1 causes an (almost) infinite wait time, 342 * 0 non-blocking. 343 * 344 * Buffer fields 345 * On entry: 346 * frameCount number of frames requested 347 * After error return: 348 * frameCount 0 349 * size 0 350 * raw undefined 351 * After successful return: 352 * frameCount actual number of frames available, <= number requested 353 * size actual number of bytes available 354 * raw pointer to the buffer 355 */ 356 357 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 358 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 359 __attribute__((__deprecated__)); 360 361 private: 362 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 363 * additional non-contiguous frames that are available immediately. 364 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 365 * in case the requested amount of frames is in two or more non-contiguous regions. 366 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 367 */ 368 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 369 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 370 public: 371 372 /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */ 373 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 374 void releaseBuffer(Buffer* audioBuffer); 375 376 /* As a convenience we provide a read() interface to the audio buffer. 377 * Input parameter 'size' is in byte units. 378 * This is implemented on top of obtainBuffer/releaseBuffer. For best 379 * performance use callbacks. Returns actual number of bytes read >= 0, 380 * or one of the following negative status codes: 381 * INVALID_OPERATION AudioRecord is configured for streaming mode 382 * BAD_VALUE size is invalid 383 * WOULD_BLOCK when obtainBuffer() returns same, or 384 * AudioRecord was stopped during the read 385 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 386 */ 387 ssize_t read(void* buffer, size_t size); 388 389 /* Return the number of input frames lost in the audio driver since the last call of this 390 * function. Audio driver is expected to reset the value to 0 and restart counting upon 391 * returning the current value by this function call. Such loss typically occurs when the 392 * user space process is blocked longer than the capacity of audio driver buffers. 393 * Units: the number of input audio frames. 394 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 395 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 396 */ 397 uint32_t getInputFramesLost() const; 398 399 private: 400 /* copying audio record objects is not allowed */ 401 AudioRecord(const AudioRecord& other); 402 AudioRecord& operator = (const AudioRecord& other); 403 404 /* a small internal class to handle the callback */ 405 class AudioRecordThread : public Thread 406 { 407 public: 408 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 409 410 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 411 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 412 virtual void requestExit(); 413 414 void pause(); // suspend thread from execution at next loop boundary 415 void resume(); // allow thread to execute, if not requested to exit 416 417 private: 418 void pauseInternal(nsecs_t ns = 0LL); 419 // like pause(), but only used internally within thread 420 421 friend class AudioRecord; 422 virtual bool threadLoop(); 423 AudioRecord& mReceiver; 424 virtual ~AudioRecordThread(); 425 Mutex mMyLock; // Thread::mLock is private 426 Condition mMyCond; // Thread::mThreadExitedCondition is private 427 bool mPaused; // whether thread is requested to pause at next loop entry 428 bool mPausedInt; // whether thread internally requests pause 429 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 430 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 431 }; 432 433 // body of AudioRecordThread::threadLoop() 434 // returns the maximum amount of time before we would like to run again, where: 435 // 0 immediately 436 // > 0 no later than this many nanoseconds from now 437 // NS_WHENEVER still active but no particular deadline 438 // NS_INACTIVE inactive so don't run again until re-started 439 // NS_NEVER never again 440 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 441 nsecs_t processAudioBuffer(); 442 443 // caller must hold lock on mLock for all _l methods 444 445 status_t openRecord_l(size_t epoch); 446 447 // FIXME enum is faster than strcmp() for parameter 'from' 448 status_t restoreRecord_l(const char *from); 449 450 sp<AudioRecordThread> mAudioRecordThread; 451 mutable Mutex mLock; 452 453 // Current client state: false = stopped, true = active. Protected by mLock. If more states 454 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 455 bool mActive; 456 457 // for client callback handler 458 callback_t mCbf; // callback handler for events, or NULL 459 void* mUserData; 460 461 // for notification APIs 462 uint32_t mNotificationFramesReq; // requested number of frames between each 463 // notification callback 464 // as specified in constructor or set() 465 uint32_t mNotificationFramesAct; // actual number of frames between each 466 // notification callback 467 bool mRefreshRemaining; // processAudioBuffer() should refresh 468 // mRemainingFrames and mRetryOnPartialBuffer 469 470 // These are private to processAudioBuffer(), and are not protected by a lock 471 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 472 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 473 uint32_t mObservedSequence; // last observed value of mSequence 474 475 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 476 bool mMarkerReached; 477 uint32_t mNewPosition; // in frames 478 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 479 480 status_t mStatus; 481 482 size_t mFrameCount; // corresponds to current IAudioRecord, value is 483 // reported back by AudioFlinger to the client 484 size_t mReqFrameCount; // frame count to request the first or next time 485 // a new IAudioRecord is needed, non-decreasing 486 487 // constant after constructor or set() 488 uint32_t mSampleRate; 489 audio_format_t mFormat; 490 uint32_t mChannelCount; 491 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 492 audio_source_t mInputSource; 493 uint32_t mLatency; // in ms 494 audio_channel_mask_t mChannelMask; 495 audio_input_flags_t mFlags; 496 int mSessionId; 497 transfer_type mTransfer; 498 499 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 500 // provided the initial set() was successful 501 sp<IAudioRecord> mAudioRecord; 502 sp<IMemory> mCblkMemory; 503 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 504 sp<IMemory> mBufferMemory; 505 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 506 507 int mPreviousPriority; // before start() 508 SchedPolicy mPreviousSchedulingGroup; 509 bool mAwaitBoost; // thread should wait for priority boost before running 510 511 // The proxy should only be referenced while a lock is held because the proxy isn't 512 // multi-thread safe. 513 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 514 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 515 // them around in case they are replaced during the obtainBuffer(). 516 sp<AudioRecordClientProxy> mProxy; 517 518 bool mInOverrun; // whether recorder is currently in overrun state 519 520 private: 521 class DeathNotifier : public IBinder::DeathRecipient { 522 public: DeathNotifier(AudioRecord * audioRecord)523 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 524 protected: 525 virtual void binderDied(const wp<IBinder>& who); 526 private: 527 const wp<AudioRecord> mAudioRecord; 528 }; 529 530 sp<DeathNotifier> mDeathNotifier; 531 uint32_t mSequence; // incremented for each new IAudioRecord attempt 532 }; 533 534 }; // namespace android 535 536 #endif // ANDROID_AUDIORECORD_H 537