AudioRecord.h revision 7cd9cf70e36ad4b8eb12e24f9adbbe6fd69edebd
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 29class audio_track_cblk_t; 30class AudioRecordClientProxy; 31 32// ---------------------------------------------------------------------------- 33 34class AudioRecord : public RefBase 35{ 36public: 37 38 static const int DEFAULT_SAMPLE_RATE = 8000; 39 40 /* Events used by AudioRecord callback function (callback_t). 41 * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*. 42 */ 43 enum event_type { 44 EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer. 45 EVENT_OVERRUN = 1, // PCM buffer overrun occurred. 46 EVENT_MARKER = 2, // Record head is at the specified marker position 47 // (See setMarkerPosition()). 48 EVENT_NEW_POS = 3, // Record head is at a new position 49 // (See setPositionUpdatePeriod()). 50 EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and 51 // voluntary invalidation by mediaserver, or mediaserver crash. 52 }; 53 54 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 55 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 56 */ 57 58 class Buffer 59 { 60 public: 61 // FIXME use m prefix 62 size_t frameCount; // number of sample frames corresponding to size; 63 // on input it is the number of frames available, 64 // on output is the number of frames actually drained 65 66 size_t size; // input/output in bytes == frameCount * frameSize 67 // FIXME this is redundant with respect to frameCount, 68 // and TRANSFER_OBTAIN mode is broken for 8-bit data 69 // since we don't define the frame format 70 71 union { 72 void* raw; 73 short* i16; // signed 16-bit 74 int8_t* i8; // unsigned 8-bit, offset by 0x80 75 }; 76 }; 77 78 /* As a convenience, if a callback is supplied, a handler thread 79 * is automatically created with the appropriate priority. This thread 80 * invokes the callback when a new buffer becomes ready or various conditions occur. 81 * Parameters: 82 * 83 * event: type of event notified (see enum AudioRecord::event_type). 84 * user: Pointer to context for use by the callback receiver. 85 * info: Pointer to optional parameter according to event type: 86 * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read 87 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 88 * consumed. 89 * - EVENT_OVERRUN: unused. 90 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 91 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 92 * - EVENT_NEW_IAUDIORECORD: unused. 93 */ 94 95 typedef void (*callback_t)(int event, void* user, void *info); 96 97 /* Returns the minimum frame count required for the successful creation of 98 * an AudioRecord object. 99 * Returned status (from utils/Errors.h) can be: 100 * - NO_ERROR: successful operation 101 * - NO_INIT: audio server or audio hardware not initialized 102 * - BAD_VALUE: unsupported configuration 103 */ 104 105 static status_t getMinFrameCount(size_t* frameCount, 106 uint32_t sampleRate, 107 audio_format_t format, 108 audio_channel_mask_t channelMask); 109 110 /* How data is transferred from AudioRecord 111 */ 112 enum transfer_type { 113 TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters 114 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 115 TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() 116 TRANSFER_SYNC, // synchronous read() 117 }; 118 119 /* Constructs an uninitialized AudioRecord. No connection with 120 * AudioFlinger takes place. Use set() after this. 121 */ 122 AudioRecord(); 123 124 /* Creates an AudioRecord object and registers it with AudioFlinger. 125 * Once created, the track needs to be started before it can be used. 126 * Unspecified values are set to appropriate default values. 127 * 128 * Parameters: 129 * 130 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 131 * sampleRate: Data sink sampling rate in Hz. 132 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 133 * 16 bits per sample). 134 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 135 * frameCount: Minimum size of track PCM buffer in frames. This defines the 136 * application's contribution to the 137 * latency of the track. The actual size selected by the AudioRecord could 138 * be larger if the requested size is not compatible with current audio HAL 139 * latency. Zero means to use a default value. 140 * cbf: Callback function. If not null, this function is called periodically 141 * to consume new PCM data and inform of marker, position updates, etc. 142 * user: Context for use by the callback receiver. 143 * notificationFrames: The callback function is called each time notificationFrames PCM 144 * frames are ready in record track output buffer. 145 * sessionId: Not yet supported. 146 * transferType: How data is transferred from AudioRecord. 147 * flags: See comments on audio_input_flags_t in <system/audio.h> 148 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 149 */ 150 151 AudioRecord(audio_source_t inputSource, 152 uint32_t sampleRate, 153 audio_format_t format, 154 audio_channel_mask_t channelMask, 155 int frameCount = 0, 156 callback_t cbf = NULL, 157 void* user = NULL, 158 int notificationFrames = 0, 159 int sessionId = 0, 160 transfer_type transferType = TRANSFER_DEFAULT, 161 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE); 162 163 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 164 * Also destroys all resources associated with the AudioRecord. 165 */ 166protected: 167 virtual ~AudioRecord(); 168public: 169 170 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 171 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 172 * Returned status (from utils/Errors.h) can be: 173 * - NO_ERROR: successful intialization 174 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 175 * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) 176 * - NO_INIT: audio server or audio hardware not initialized 177 * - PERMISSION_DENIED: recording is not allowed for the requesting process 178 * 179 * Parameters not listed in the AudioRecord constructors above: 180 * 181 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 182 */ 183 status_t set(audio_source_t inputSource, 184 uint32_t sampleRate, 185 audio_format_t format, 186 audio_channel_mask_t channelMask, 187 int frameCount = 0, 188 callback_t cbf = NULL, 189 void* user = NULL, 190 int notificationFrames = 0, 191 bool threadCanCallJava = false, 192 int sessionId = 0, 193 transfer_type transferType = TRANSFER_DEFAULT, 194 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE); 195 196 /* Result of constructing the AudioRecord. This must be checked 197 * before using any AudioRecord API (except for set()), because using 198 * an uninitialized AudioRecord produces undefined results. 199 * See set() method above for possible return codes. 200 */ 201 status_t initCheck() const { return mStatus; } 202 203 /* Returns this track's estimated latency in milliseconds. 204 * This includes the latency due to AudioRecord buffer size, 205 * and audio hardware driver. 206 */ 207 uint32_t latency() const { return mLatency; } 208 209 /* getters, see constructor and set() */ 210 211 audio_format_t format() const { return mFormat; } 212 uint32_t channelCount() const { return mChannelCount; } 213 size_t frameCount() const { return mFrameCount; } 214 size_t frameSize() const { return mFrameSize; } 215 audio_source_t inputSource() const { return mInputSource; } 216 217 /* After it's created the track is not active. Call start() to 218 * make it active. If set, the callback will start being called. 219 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 220 * the specified event occurs on the specified trigger session. 221 */ 222 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 223 int triggerSession = 0); 224 225 /* Stop a track. If set, the callback will cease being called. Note that obtainBuffer() still 226 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 227 */ 228 void stop(); 229 bool stopped() const; 230 231 /* Return the sink sample rate for this record track in Hz. 232 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 233 */ 234 uint32_t getSampleRate() const { return mSampleRate; } 235 236 /* Sets marker position. When record reaches the number of frames specified, 237 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 238 * with marker == 0 cancels marker notification callback. 239 * To set a marker at a position which would compute as 0, 240 * a workaround is to the set the marker at a nearby position such as ~0 or 1. 241 * If the AudioRecord has been opened with no callback function associated, 242 * the operation will fail. 243 * 244 * Parameters: 245 * 246 * marker: marker position expressed in wrapping (overflow) frame units, 247 * like the return value of getPosition(). 248 * 249 * Returned status (from utils/Errors.h) can be: 250 * - NO_ERROR: successful operation 251 * - INVALID_OPERATION: the AudioRecord has no callback installed. 252 */ 253 status_t setMarkerPosition(uint32_t marker); 254 status_t getMarkerPosition(uint32_t *marker) const; 255 256 /* Sets position update period. Every time the number of frames specified has been recorded, 257 * a callback with event type EVENT_NEW_POS is called. 258 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 259 * callback. 260 * If the AudioRecord has been opened with no callback function associated, 261 * the operation will fail. 262 * Extremely small values may be rounded up to a value the implementation can support. 263 * 264 * Parameters: 265 * 266 * updatePeriod: position update notification period expressed in frames. 267 * 268 * Returned status (from utils/Errors.h) can be: 269 * - NO_ERROR: successful operation 270 * - INVALID_OPERATION: the AudioRecord has no callback installed. 271 */ 272 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 273 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 274 275 /* Return the total number of frames recorded since recording started. 276 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 277 * It is reset to zero by stop(). 278 * 279 * Parameters: 280 * 281 * position: Address where to return record head position. 282 * 283 * Returned status (from utils/Errors.h) can be: 284 * - NO_ERROR: successful operation 285 * - BAD_VALUE: position is NULL 286 */ 287 status_t getPosition(uint32_t *position) const; 288 289 /* Returns a handle on the audio input used by this AudioRecord. 290 * 291 * Parameters: 292 * none. 293 * 294 * Returned value: 295 * handle on audio hardware input 296 */ 297 audio_io_handle_t getInput() const; 298 299 /* Returns the audio session ID associated with this AudioRecord. 300 * 301 * Parameters: 302 * none. 303 * 304 * Returned value: 305 * AudioRecord session ID. 306 * 307 * No lock needed because session ID doesn't change after first set(). 308 */ 309 int getSessionId() const { return mSessionId; } 310 311 /* Obtains a buffer of up to "audioBuffer->frameCount" full frames. 312 * After draining these frames of data, the caller should release them with releaseBuffer(). 313 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 314 * full frames as are available immediately. 315 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 316 * regardless of the value of waitCount. 317 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 318 * maximum timeout based on waitCount; see chart below. 319 * Buffers will be returned until the pool 320 * is exhausted, at which point obtainBuffer() will either block 321 * or return WOULD_BLOCK depending on the value of the "waitCount" 322 * parameter. 323 * 324 * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, 325 * which should use read() or callback EVENT_MORE_DATA instead. 326 * 327 * Interpretation of waitCount: 328 * +n limits wait time to n * WAIT_PERIOD_MS, 329 * -1 causes an (almost) infinite wait time, 330 * 0 non-blocking. 331 * 332 * Buffer fields 333 * On entry: 334 * frameCount number of frames requested 335 * After error return: 336 * frameCount 0 337 * size 0 338 * raw undefined 339 * After successful return: 340 * frameCount actual number of frames available, <= number requested 341 * size actual number of bytes available 342 * raw pointer to the buffer 343 */ 344 345 /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ 346 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 347 __attribute__((__deprecated__)); 348 349private: 350 /* New internal API. 351 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 352 * additional non-contiguous frames that are available immediately. 353 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 354 * in case the requested amount of frames is in two or more non-contiguous regions. 355 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 356 */ 357 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 358 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 359public: 360 361 /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */ 362 // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed 363 void releaseBuffer(Buffer* audioBuffer); 364 365 /* As a convenience we provide a read() interface to the audio buffer. 366 * Input parameter 'size' is in byte units. 367 * This is implemented on top of obtainBuffer/releaseBuffer. For best 368 * performance use callbacks. Returns actual number of bytes read >= 0, 369 * or a negative status code. 370 */ 371 ssize_t read(void* buffer, size_t size); 372 373 /* Return the number of input frames lost in the audio driver since the last call of this 374 * function. Audio driver is expected to reset the value to 0 and restart counting upon 375 * returning the current value by this function call. Such loss typically occurs when the 376 * user space process is blocked longer than the capacity of audio driver buffers. 377 * Units: the number of input audio frames. 378 */ 379 unsigned int getInputFramesLost() const; 380 381private: 382 /* copying audio record objects is not allowed */ 383 AudioRecord(const AudioRecord& other); 384 AudioRecord& operator = (const AudioRecord& other); 385 386 /* a small internal class to handle the callback */ 387 class AudioRecordThread : public Thread 388 { 389 public: 390 AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false); 391 392 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 393 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 394 virtual void requestExit(); 395 396 void pause(); // suspend thread from execution at next loop boundary 397 void resume(); // allow thread to execute, if not requested to exit 398 void pauseConditional(); 399 // like pause(), but only if prior resume() wasn't latched 400 401 private: 402 friend class AudioRecord; 403 virtual bool threadLoop(); 404 AudioRecord& mReceiver; 405 virtual ~AudioRecordThread(); 406 Mutex mMyLock; // Thread::mLock is private 407 Condition mMyCond; // Thread::mThreadExitedCondition is private 408 bool mPaused; // whether thread is currently paused 409 bool mResumeLatch; // whether next pauseConditional() will be a nop 410 }; 411 412 // body of AudioRecordThread::threadLoop() 413 // returns the maximum amount of time before we would like to run again, where: 414 // 0 immediately 415 // > 0 no later than this many nanoseconds from now 416 // NS_WHENEVER still active but no particular deadline 417 // NS_INACTIVE inactive so don't run again until re-started 418 // NS_NEVER never again 419 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 420 nsecs_t processAudioBuffer(const sp<AudioRecordThread>& thread); 421 422 // caller must hold lock on mLock for all _l methods 423 status_t openRecord_l(uint32_t sampleRate, 424 audio_format_t format, 425 size_t frameCount, 426 audio_input_flags_t flags, 427 audio_io_handle_t input, 428 size_t epoch); 429 430 audio_io_handle_t getInput_l(); 431 432 // FIXME enum is faster than strcmp() for parameter 'from' 433 status_t restoreRecord_l(const char *from); 434 435 sp<AudioRecordThread> mAudioRecordThread; 436 mutable Mutex mLock; 437 438 // Current client state: false = stopped, true = active. Protected by mLock. If more states 439 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 440 bool mActive; 441 442 // for client callback handler 443 callback_t mCbf; // callback handler for events, or NULL 444 void* mUserData; // for client callback handler 445 446 // for notification APIs 447 uint32_t mNotificationFramesReq; // requested number of frames between each 448 // notification callback 449 uint32_t mNotificationFramesAct; // actual number of frames between each 450 // notification callback 451 bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 452 453 // These are private to processAudioBuffer(), and are not protected by a lock 454 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 455 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 456 int mObservedSequence; // last observed value of mSequence 457 458 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 459 bool mMarkerReached; 460 uint32_t mNewPosition; // in frames 461 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 462 463 status_t mStatus; 464 465 // constant after constructor or set() 466 uint32_t mSampleRate; 467 size_t mFrameCount; 468 audio_format_t mFormat; 469 uint32_t mChannelCount; 470 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 471 audio_source_t mInputSource; 472 uint32_t mLatency; // in ms 473 audio_channel_mask_t mChannelMask; 474 audio_input_flags_t mFlags; 475 int mSessionId; 476 transfer_type mTransfer; 477 478 audio_io_handle_t mInput; // returned by AudioSystem::getInput() 479 480 // may be changed if IAudioRecord object is re-created 481 sp<IAudioRecord> mAudioRecord; 482 sp<IMemory> mCblkMemory; 483 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 484 485 int mPreviousPriority; // before start() 486 SchedPolicy mPreviousSchedulingGroup; 487 bool mAwaitBoost; // thread should wait for priority boost before running 488 489 // The proxy should only be referenced while a lock is held because the proxy isn't 490 // multi-thread safe. 491 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 492 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 493 sp<AudioRecordClientProxy> mProxy; 494 495 bool mInOverrun; // whether recorder is currently in overrun state 496 497private: 498 class DeathNotifier : public IBinder::DeathRecipient { 499 public: 500 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 501 protected: 502 virtual void binderDied(const wp<IBinder>& who); 503 private: 504 const wp<AudioRecord> mAudioRecord; 505 }; 506 507 sp<DeathNotifier> mDeathNotifier; 508 uint32_t mSequence; // incremented for each new IAudioRecord attempt 509}; 510 511}; // namespace android 512 513#endif // ANDROID_AUDIORECORD_H 514