AudioRecord.h revision caf7f48a0ef558689d39aafd187c1571ff4128b4
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 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 * pAttributes: if not NULL, supersedes inputSource for use case selection 156 */ 157 158 AudioRecord(audio_source_t inputSource, 159 uint32_t sampleRate, 160 audio_format_t format, 161 audio_channel_mask_t channelMask, 162 size_t frameCount = 0, 163 callback_t cbf = NULL, 164 void* user = NULL, 165 uint32_t notificationFrames = 0, 166 int sessionId = AUDIO_SESSION_ALLOCATE, 167 transfer_type transferType = TRANSFER_DEFAULT, 168 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 169 const audio_attributes_t* pAttributes = NULL); 170 171 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 172 * Also destroys all resources associated with the AudioRecord. 173 */ 174protected: 175 virtual ~AudioRecord(); 176public: 177 178 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 179 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 180 * Returned status (from utils/Errors.h) can be: 181 * - NO_ERROR: successful intialization 182 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 183 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 184 * - NO_INIT: audio server or audio hardware not initialized 185 * - PERMISSION_DENIED: recording is not allowed for the requesting process 186 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 187 * 188 * Parameters not listed in the AudioRecord constructors above: 189 * 190 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 191 */ 192 status_t set(audio_source_t inputSource, 193 uint32_t sampleRate, 194 audio_format_t format, 195 audio_channel_mask_t channelMask, 196 size_t frameCount = 0, 197 callback_t cbf = NULL, 198 void* user = NULL, 199 uint32_t notificationFrames = 0, 200 bool threadCanCallJava = false, 201 int sessionId = AUDIO_SESSION_ALLOCATE, 202 transfer_type transferType = TRANSFER_DEFAULT, 203 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 204 const audio_attributes_t* pAttributes = NULL); 205 206 /* Result of constructing the AudioRecord. This must be checked for successful initialization 207 * before using any AudioRecord API (except for set()), because using 208 * an uninitialized AudioRecord produces undefined results. 209 * See set() method above for possible return codes. 210 */ 211 status_t initCheck() const { return mStatus; } 212 213 /* Returns this track's estimated latency in milliseconds. 214 * This includes the latency due to AudioRecord buffer size, 215 * and audio hardware driver. 216 */ 217 uint32_t latency() const { return mLatency; } 218 219 /* getters, see constructor and set() */ 220 221 audio_format_t format() const { return mFormat; } 222 uint32_t channelCount() const { return mChannelCount; } 223 size_t frameCount() const { return mFrameCount; } 224 size_t frameSize() const { return mFrameSize; } 225 audio_source_t inputSource() const { return mAttributes.source; } 226 227 /* After it's created the track is not active. Call start() to 228 * make it active. If set, the callback will start being called. 229 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 230 * the specified event occurs on the specified trigger session. 231 */ 232 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 233 int triggerSession = 0); 234 235 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 236 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 237 */ 238 void stop(); 239 bool stopped() const; 240 241 /* Return the sink sample rate for this record track in Hz. 242 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 243 */ 244 uint32_t getSampleRate() const { return mSampleRate; } 245 246 /* Return the notification frame count. 247 * This is approximately how often the callback is invoked, for transfer type TRANSFER_CALLBACK. 248 */ 249 size_t notificationFrames() const { return mNotificationFramesAct; } 250 251 /* Sets marker position. When record reaches the number of frames specified, 252 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 253 * with marker == 0 cancels marker notification callback. 254 * To set a marker at a position which would compute as 0, 255 * a workaround is to set the marker at a nearby position such as ~0 or 1. 256 * If the AudioRecord has been opened with no callback function associated, 257 * the operation will fail. 258 * 259 * Parameters: 260 * 261 * marker: marker position expressed in wrapping (overflow) frame units, 262 * like the return value of getPosition(). 263 * 264 * Returned status (from utils/Errors.h) can be: 265 * - NO_ERROR: successful operation 266 * - INVALID_OPERATION: the AudioRecord has no callback installed. 267 */ 268 status_t setMarkerPosition(uint32_t marker); 269 status_t getMarkerPosition(uint32_t *marker) const; 270 271 /* Sets position update period. Every time the number of frames specified has been recorded, 272 * a callback with event type EVENT_NEW_POS is called. 273 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 274 * callback. 275 * If the AudioRecord has been opened with no callback function associated, 276 * the operation will fail. 277 * Extremely small values may be rounded up to a value the implementation can support. 278 * 279 * Parameters: 280 * 281 * updatePeriod: position update notification period expressed in frames. 282 * 283 * Returned status (from utils/Errors.h) can be: 284 * - NO_ERROR: successful operation 285 * - INVALID_OPERATION: the AudioRecord has no callback installed. 286 */ 287 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 288 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 289 290 /* Return the total number of frames recorded since recording started. 291 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 292 * It is reset to zero by stop(). 293 * 294 * Parameters: 295 * 296 * position: Address where to return record head position. 297 * 298 * Returned status (from utils/Errors.h) can be: 299 * - NO_ERROR: successful operation 300 * - BAD_VALUE: position is NULL 301 */ 302 status_t getPosition(uint32_t *position) const; 303 304 /* Returns a handle on the audio input used by this AudioRecord. 305 * 306 * Parameters: 307 * none. 308 * 309 * Returned value: 310 * handle on audio hardware input 311 */ 312 audio_io_handle_t getInput() const; 313 314 /* Returns the audio session ID associated with this AudioRecord. 315 * 316 * Parameters: 317 * none. 318 * 319 * Returned value: 320 * AudioRecord session ID. 321 * 322 * No lock needed because session ID doesn't change after first set(). 323 */ 324 int getSessionId() const { return mSessionId; } 325 326 /* Obtains a buffer of up to "audioBuffer->frameCount" full frames. 327 * After draining these frames of data, the caller should release them with releaseBuffer(). 328 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 329 * full frames as are available immediately. 330 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 331 * regardless of the value of waitCount. 332 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 333 * maximum timeout based on waitCount; see chart below. 334 * Buffers will be returned until the pool 335 * is exhausted, at which point obtainBuffer() will either block 336 * or return WOULD_BLOCK depending on the value of the "waitCount" 337 * parameter. 338 * 339 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 340 * which should use read() or callback EVENT_MORE_DATA instead. 341 * 342 * Interpretation of waitCount: 343 * +n limits wait time to n * WAIT_PERIOD_MS, 344 * -1 causes an (almost) infinite wait time, 345 * 0 non-blocking. 346 * 347 * Buffer fields 348 * On entry: 349 * frameCount number of frames requested 350 * After error return: 351 * frameCount 0 352 * size 0 353 * raw undefined 354 * After successful return: 355 * frameCount actual number of frames available, <= number requested 356 * size actual number of bytes available 357 * raw pointer to the buffer 358 */ 359 360 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 361 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 362 __attribute__((__deprecated__)); 363 364private: 365 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 366 * additional non-contiguous frames that are available immediately. 367 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 368 * in case the requested amount of frames is in two or more non-contiguous regions. 369 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 370 */ 371 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 372 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 373public: 374 375 /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */ 376 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 377 void releaseBuffer(Buffer* audioBuffer); 378 379 /* As a convenience we provide a read() interface to the audio buffer. 380 * Input parameter 'size' is in byte units. 381 * This is implemented on top of obtainBuffer/releaseBuffer. For best 382 * performance use callbacks. Returns actual number of bytes read >= 0, 383 * or one of the following negative status codes: 384 * INVALID_OPERATION AudioRecord is configured for streaming mode 385 * BAD_VALUE size is invalid 386 * WOULD_BLOCK when obtainBuffer() returns same, or 387 * AudioRecord was stopped during the read 388 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 389 */ 390 ssize_t read(void* buffer, size_t size); 391 392 /* Return the number of input frames lost in the audio driver since the last call of this 393 * function. Audio driver is expected to reset the value to 0 and restart counting upon 394 * returning the current value by this function call. Such loss typically occurs when the 395 * user space process is blocked longer than the capacity of audio driver buffers. 396 * Units: the number of input audio frames. 397 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 398 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 399 */ 400 uint32_t getInputFramesLost() const; 401 402private: 403 /* copying audio record objects is not allowed */ 404 AudioRecord(const AudioRecord& other); 405 AudioRecord& operator = (const AudioRecord& other); 406 407 /* a small internal class to handle the callback */ 408 class AudioRecordThread : public Thread 409 { 410 public: 411 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 412 413 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 414 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 415 virtual void requestExit(); 416 417 void pause(); // suspend thread from execution at next loop boundary 418 void resume(); // allow thread to execute, if not requested to exit 419 420 private: 421 void pauseInternal(nsecs_t ns = 0LL); 422 // like pause(), but only used internally within thread 423 424 friend class AudioRecord; 425 virtual bool threadLoop(); 426 AudioRecord& mReceiver; 427 virtual ~AudioRecordThread(); 428 Mutex mMyLock; // Thread::mLock is private 429 Condition mMyCond; // Thread::mThreadExitedCondition is private 430 bool mPaused; // whether thread is requested to pause at next loop entry 431 bool mPausedInt; // whether thread internally requests pause 432 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 433 bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request 434 }; 435 436 // body of AudioRecordThread::threadLoop() 437 // returns the maximum amount of time before we would like to run again, where: 438 // 0 immediately 439 // > 0 no later than this many nanoseconds from now 440 // NS_WHENEVER still active but no particular deadline 441 // NS_INACTIVE inactive so don't run again until re-started 442 // NS_NEVER never again 443 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 444 nsecs_t processAudioBuffer(); 445 446 // caller must hold lock on mLock for all _l methods 447 448 status_t openRecord_l(size_t epoch); 449 450 // FIXME enum is faster than strcmp() for parameter 'from' 451 status_t restoreRecord_l(const char *from); 452 453 sp<AudioRecordThread> mAudioRecordThread; 454 mutable Mutex mLock; 455 456 // Current client state: false = stopped, true = active. Protected by mLock. If more states 457 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 458 bool mActive; 459 460 // for client callback handler 461 callback_t mCbf; // callback handler for events, or NULL 462 void* mUserData; 463 464 // for notification APIs 465 uint32_t mNotificationFramesReq; // requested number of frames between each 466 // notification callback 467 // as specified in constructor or set() 468 uint32_t mNotificationFramesAct; // actual number of frames between each 469 // notification callback 470 bool mRefreshRemaining; // processAudioBuffer() should refresh 471 // mRemainingFrames and mRetryOnPartialBuffer 472 473 // These are private to processAudioBuffer(), and are not protected by a lock 474 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 475 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 476 uint32_t mObservedSequence; // last observed value of mSequence 477 478 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 479 bool mMarkerReached; 480 uint32_t mNewPosition; // in frames 481 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 482 483 status_t mStatus; 484 485 size_t mFrameCount; // corresponds to current IAudioRecord, value is 486 // reported back by AudioFlinger to the client 487 size_t mReqFrameCount; // frame count to request the first or next time 488 // a new IAudioRecord is needed, non-decreasing 489 490 // constant after constructor or set() 491 uint32_t mSampleRate; 492 audio_format_t mFormat; 493 uint32_t mChannelCount; 494 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 495 uint32_t mLatency; // in ms 496 audio_channel_mask_t mChannelMask; 497 audio_input_flags_t mFlags; 498 int mSessionId; 499 transfer_type mTransfer; 500 501 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 502 // provided the initial set() was successful 503 sp<IAudioRecord> mAudioRecord; 504 sp<IMemory> mCblkMemory; 505 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 506 sp<IMemory> mBufferMemory; 507 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 508 509 int mPreviousPriority; // before start() 510 SchedPolicy mPreviousSchedulingGroup; 511 bool mAwaitBoost; // thread should wait for priority boost before running 512 513 // The proxy should only be referenced while a lock is held because the proxy isn't 514 // multi-thread safe. 515 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 516 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 517 // them around in case they are replaced during the obtainBuffer(). 518 sp<AudioRecordClientProxy> mProxy; 519 520 bool mInOverrun; // whether recorder is currently in overrun state 521 522private: 523 class DeathNotifier : public IBinder::DeathRecipient { 524 public: 525 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 526 protected: 527 virtual void binderDied(const wp<IBinder>& who); 528 private: 529 const wp<AudioRecord> mAudioRecord; 530 }; 531 532 sp<DeathNotifier> mDeathNotifier; 533 uint32_t mSequence; // incremented for each new IAudioRecord attempt 534 audio_attributes_t mAttributes; 535}; 536 537}; // namespace android 538 539#endif // ANDROID_AUDIORECORD_H 540