AudioRecord.cpp revision 3151427b6b0adf99929433715bab6f1e505100c1
1/* 2** 3** Copyright 2008, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18//#define LOG_NDEBUG 0 19#define LOG_TAG "AudioRecord" 20 21#include <sys/resource.h> 22#include <binder/IPCThreadState.h> 23#include <media/AudioRecord.h> 24#include <utils/Log.h> 25#include <private/media/AudioTrackShared.h> 26#include <media/IAudioFlinger.h> 27 28#define WAIT_PERIOD_MS 10 29 30namespace android { 31// --------------------------------------------------------------------------- 32 33// static 34status_t AudioRecord::getMinFrameCount( 35 size_t* frameCount, 36 uint32_t sampleRate, 37 audio_format_t format, 38 audio_channel_mask_t channelMask) 39{ 40 if (frameCount == NULL) { 41 return BAD_VALUE; 42 } 43 44 // default to 0 in case of error 45 *frameCount = 0; 46 47 size_t size = 0; 48 status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size); 49 if (status != NO_ERROR) { 50 ALOGE("AudioSystem could not query the input buffer size; status %d", status); 51 return NO_INIT; 52 } 53 54 if (size == 0) { 55 ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x", 56 sampleRate, format, channelMask); 57 return BAD_VALUE; 58 } 59 60 // We double the size of input buffer for ping pong use of record buffer. 61 size <<= 1; 62 63 if (audio_is_linear_pcm(format)) { 64 uint32_t channelCount = popcount(channelMask); 65 size /= channelCount * audio_bytes_per_sample(format); 66 } 67 68 *frameCount = size; 69 return NO_ERROR; 70} 71 72// --------------------------------------------------------------------------- 73 74AudioRecord::AudioRecord() 75 : mStatus(NO_INIT), mSessionId(0), 76 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) 77{ 78} 79 80AudioRecord::AudioRecord( 81 audio_source_t inputSource, 82 uint32_t sampleRate, 83 audio_format_t format, 84 audio_channel_mask_t channelMask, 85 int frameCount, 86 callback_t cbf, 87 void* user, 88 int notificationFrames, 89 int sessionId, 90 transfer_type transferType, 91 audio_input_flags_t flags) 92 : mStatus(NO_INIT), mSessionId(0), 93 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 94 mPreviousSchedulingGroup(SP_DEFAULT), 95 mProxy(NULL) 96{ 97 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, 98 notificationFrames, false /*threadCanCallJava*/, sessionId, transferType); 99} 100 101AudioRecord::~AudioRecord() 102{ 103 if (mStatus == NO_ERROR) { 104 // Make sure that callback function exits in the case where 105 // it is looping on buffer empty condition in obtainBuffer(). 106 // Otherwise the callback thread will never exit. 107 stop(); 108 if (mAudioRecordThread != 0) { 109 mAudioRecordThread->requestExit(); // see comment in AudioRecord.h 110 mAudioRecordThread->requestExitAndWait(); 111 mAudioRecordThread.clear(); 112 } 113 if (mAudioRecord != 0) { 114 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 115 mAudioRecord.clear(); 116 } 117 IPCThreadState::self()->flushCommands(); 118 AudioSystem::releaseAudioSessionId(mSessionId); 119 } 120} 121 122status_t AudioRecord::set( 123 audio_source_t inputSource, 124 uint32_t sampleRate, 125 audio_format_t format, 126 audio_channel_mask_t channelMask, 127 int frameCountInt, 128 callback_t cbf, 129 void* user, 130 int notificationFrames, 131 bool threadCanCallJava, 132 int sessionId, 133 transfer_type transferType, 134 audio_input_flags_t flags) 135{ 136 switch (transferType) { 137 case TRANSFER_DEFAULT: 138 if (cbf == NULL || threadCanCallJava) { 139 transferType = TRANSFER_SYNC; 140 } else { 141 transferType = TRANSFER_CALLBACK; 142 } 143 break; 144 case TRANSFER_CALLBACK: 145 if (cbf == NULL) { 146 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL"); 147 return BAD_VALUE; 148 } 149 break; 150 case TRANSFER_OBTAIN: 151 case TRANSFER_SYNC: 152 break; 153 default: 154 ALOGE("Invalid transfer type %d", transferType); 155 return BAD_VALUE; 156 } 157 mTransfer = transferType; 158 159 // FIXME "int" here is legacy and will be replaced by size_t later 160 if (frameCountInt < 0) { 161 ALOGE("Invalid frame count %d", frameCountInt); 162 return BAD_VALUE; 163 } 164 size_t frameCount = frameCountInt; 165 166 ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask, 167 frameCount); 168 169 AutoMutex lock(mLock); 170 171 if (mAudioRecord != 0) { 172 ALOGE("Track already in use"); 173 return INVALID_OPERATION; 174 } 175 176 if (inputSource == AUDIO_SOURCE_DEFAULT) { 177 inputSource = AUDIO_SOURCE_MIC; 178 } 179 180 if (sampleRate == 0) { 181 sampleRate = DEFAULT_SAMPLE_RATE; 182 } 183 mSampleRate = sampleRate; 184 185 // these below should probably come from the audioFlinger too... 186 if (format == AUDIO_FORMAT_DEFAULT) { 187 format = AUDIO_FORMAT_PCM_16_BIT; 188 } 189 190 // validate parameters 191 if (!audio_is_valid_format(format)) { 192 ALOGE("Invalid format %d", format); 193 return BAD_VALUE; 194 } 195 // Temporary restriction: AudioFlinger currently supports 16-bit PCM only 196 if (format != AUDIO_FORMAT_PCM_16_BIT) { 197 ALOGE("Format %d is not supported", format); 198 return BAD_VALUE; 199 } 200 mFormat = format; 201 202 if (!audio_is_input_channel(channelMask)) { 203 ALOGE("Invalid channel mask %#x", channelMask); 204 return BAD_VALUE; 205 } 206 mChannelMask = channelMask; 207 uint32_t channelCount = popcount(channelMask); 208 mChannelCount = channelCount; 209 210 if (audio_is_linear_pcm(format)) { 211 mFrameSize = channelCount * audio_bytes_per_sample(format); 212 } else { 213 mFrameSize = sizeof(uint8_t); 214 } 215 216 if (sessionId == 0 ) { 217 mSessionId = AudioSystem::newAudioSessionId(); 218 } else { 219 mSessionId = sessionId; 220 } 221 ALOGV("set(): mSessionId %d", mSessionId); 222 223 mFlags = flags; 224 225 audio_io_handle_t input = AudioSystem::getInput(inputSource, 226 sampleRate, 227 format, 228 channelMask, 229 mSessionId); 230 if (input == 0) { 231 ALOGE("Could not get audio input for record source %d", inputSource); 232 return BAD_VALUE; 233 } 234 235 // validate framecount 236 size_t minFrameCount = 0; 237 status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask); 238 if (status != NO_ERROR) { 239 ALOGE("getMinFrameCount() failed; status %d", status); 240 return status; 241 } 242 ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount); 243 244 if (frameCount == 0) { 245 frameCount = minFrameCount; 246 } else if (frameCount < minFrameCount) { 247 ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount); 248 return BAD_VALUE; 249 } 250 251 if (notificationFrames == 0) { 252 notificationFrames = frameCount/2; 253 } 254 255 // create the IAudioRecord 256 status = openRecord_l(sampleRate, format, frameCount, mFlags, input, 0 /*epoch*/); 257 if (status != NO_ERROR) { 258 return status; 259 } 260 261 if (cbf != NULL) { 262 mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava); 263 mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO); 264 } 265 266 mStatus = NO_ERROR; 267 268 // Update buffer size in case it has been limited by AudioFlinger during track creation 269 mFrameCount = mCblk->frameCount_; 270 271 mActive = false; 272 mCbf = cbf; 273 mNotificationFrames = notificationFrames; 274 mRefreshRemaining = true; 275 mUserData = user; 276 // TODO: add audio hardware input latency here 277 mLatency = (1000*mFrameCount) / sampleRate; 278 mMarkerPosition = 0; 279 mMarkerReached = false; 280 mNewPosition = 0; 281 mUpdatePeriod = 0; 282 mInputSource = inputSource; 283 mInput = input; 284 AudioSystem::acquireAudioSessionId(mSessionId); 285 mSequence = 1; 286 mObservedSequence = mSequence; 287 mInOverrun = false; 288 289 return NO_ERROR; 290} 291 292// ------------------------------------------------------------------------- 293 294status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) 295{ 296 ALOGV("start, sync event %d trigger session %d", event, triggerSession); 297 298 AutoMutex lock(mLock); 299 if (mActive) { 300 return NO_ERROR; 301 } 302 303 // reset current position as seen by client to 0 304 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 305 306 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 307 int32_t flags = android_atomic_acquire_load(&mCblk->mFlags); 308 309 status_t status = NO_ERROR; 310 if (!(flags & CBLK_INVALID)) { 311 ALOGV("mAudioRecord->start()"); 312 status = mAudioRecord->start(event, triggerSession); 313 if (status == DEAD_OBJECT) { 314 flags |= CBLK_INVALID; 315 } 316 } 317 if (flags & CBLK_INVALID) { 318 status = restoreRecord_l("start"); 319 } 320 321 if (status != NO_ERROR) { 322 ALOGE("start() status %d", status); 323 } else { 324 mActive = true; 325 sp<AudioRecordThread> t = mAudioRecordThread; 326 if (t != 0) { 327 t->resume(); 328 } else { 329 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 330 get_sched_policy(0, &mPreviousSchedulingGroup); 331 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 332 } 333 } 334 335 return status; 336} 337 338void AudioRecord::stop() 339{ 340 AutoMutex lock(mLock); 341 if (!mActive) { 342 return; 343 } 344 345 mActive = false; 346 mProxy->interrupt(); 347 mAudioRecord->stop(); 348 // the record head position will reset to 0, so if a marker is set, we need 349 // to activate it again 350 mMarkerReached = false; 351 sp<AudioRecordThread> t = mAudioRecordThread; 352 if (t != 0) { 353 t->pause(); 354 } else { 355 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 356 set_sched_policy(0, mPreviousSchedulingGroup); 357 } 358} 359 360bool AudioRecord::stopped() const 361{ 362 AutoMutex lock(mLock); 363 return !mActive; 364} 365 366status_t AudioRecord::setMarkerPosition(uint32_t marker) 367{ 368 if (mCbf == NULL) { 369 return INVALID_OPERATION; 370 } 371 372 AutoMutex lock(mLock); 373 mMarkerPosition = marker; 374 mMarkerReached = false; 375 376 return NO_ERROR; 377} 378 379status_t AudioRecord::getMarkerPosition(uint32_t *marker) const 380{ 381 if (marker == NULL) { 382 return BAD_VALUE; 383 } 384 385 AutoMutex lock(mLock); 386 *marker = mMarkerPosition; 387 388 return NO_ERROR; 389} 390 391status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) 392{ 393 if (mCbf == NULL) { 394 return INVALID_OPERATION; 395 } 396 397 AutoMutex lock(mLock); 398 mNewPosition = mProxy->getPosition() + updatePeriod; 399 mUpdatePeriod = updatePeriod; 400 401 return NO_ERROR; 402} 403 404status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const 405{ 406 if (updatePeriod == NULL) { 407 return BAD_VALUE; 408 } 409 410 AutoMutex lock(mLock); 411 *updatePeriod = mUpdatePeriod; 412 413 return NO_ERROR; 414} 415 416status_t AudioRecord::getPosition(uint32_t *position) const 417{ 418 if (position == NULL) { 419 return BAD_VALUE; 420 } 421 422 AutoMutex lock(mLock); 423 *position = mProxy->getPosition(); 424 425 return NO_ERROR; 426} 427 428unsigned int AudioRecord::getInputFramesLost() const 429{ 430 // no need to check mActive, because if inactive this will return 0, which is what we want 431 return AudioSystem::getInputFramesLost(getInput()); 432} 433 434// ------------------------------------------------------------------------- 435 436// must be called with mLock held 437status_t AudioRecord::openRecord_l( 438 uint32_t sampleRate, 439 audio_format_t format, 440 size_t frameCount, 441 audio_input_flags_t flags, 442 audio_io_handle_t input, 443 size_t epoch) 444{ 445 status_t status; 446 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 447 if (audioFlinger == 0) { 448 ALOGE("Could not get audioflinger"); 449 return NO_INIT; 450 } 451 452 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 453 pid_t tid = -1; 454 455 // Client can only express a preference for FAST. Server will perform additional tests. 456 // The only supported use case for FAST is callback transfer mode. 457 if (flags & AUDIO_INPUT_FLAG_FAST) { 458 if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) { 459 ALOGW("AUDIO_INPUT_FLAG_FAST denied by client"); 460 // once denied, do not request again if IAudioRecord is re-created 461 mFlags = (audio_input_flags_t) (flags & ~AUDIO_INPUT_FLAG_FAST); 462 } else { 463 trackFlags |= IAudioFlinger::TRACK_FAST; 464 tid = mAudioRecordThread->getTid(); 465 } 466 } 467 468 int originalSessionId = mSessionId; 469 sp<IAudioRecord> record = audioFlinger->openRecord(input, 470 sampleRate, format, 471 mChannelMask, 472 frameCount, 473 &trackFlags, 474 tid, 475 &mSessionId, 476 &status); 477 ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId, 478 "session ID changed from %d to %d", originalSessionId, mSessionId); 479 480 if (record == 0) { 481 ALOGE("AudioFlinger could not create record track, status: %d", status); 482 return status; 483 } 484 sp<IMemory> iMem = record->getCblk(); 485 if (iMem == 0) { 486 ALOGE("Could not get control block"); 487 return NO_INIT; 488 } 489 if (mAudioRecord != 0) { 490 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 491 mDeathNotifier.clear(); 492 } 493 mAudioRecord = record; 494 mCblkMemory = iMem; 495 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 496 mCblk = cblk; 497 498 // starting address of buffers in shared memory 499 void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); 500 501 // update proxy 502 mProxy = new AudioRecordClientProxy(cblk, buffers, frameCount, mFrameSize); 503 mProxy->setEpoch(epoch); 504 mProxy->setMinimum(mNotificationFrames); 505 506 mDeathNotifier = new DeathNotifier(this); 507 mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); 508 509 return NO_ERROR; 510} 511 512status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 513{ 514 if (audioBuffer == NULL) { 515 return BAD_VALUE; 516 } 517 if (mTransfer != TRANSFER_OBTAIN) { 518 audioBuffer->frameCount = 0; 519 audioBuffer->size = 0; 520 audioBuffer->raw = NULL; 521 return INVALID_OPERATION; 522 } 523 524 const struct timespec *requested; 525 if (waitCount == -1) { 526 requested = &ClientProxy::kForever; 527 } else if (waitCount == 0) { 528 requested = &ClientProxy::kNonBlocking; 529 } else if (waitCount > 0) { 530 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 531 struct timespec timeout; 532 timeout.tv_sec = ms / 1000; 533 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 534 requested = &timeout; 535 } else { 536 ALOGE("%s invalid waitCount %d", __func__, waitCount); 537 requested = NULL; 538 } 539 return obtainBuffer(audioBuffer, requested); 540} 541 542status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 543 struct timespec *elapsed, size_t *nonContig) 544{ 545 // previous and new IAudioRecord sequence numbers are used to detect track re-creation 546 uint32_t oldSequence = 0; 547 uint32_t newSequence; 548 549 Proxy::Buffer buffer; 550 status_t status = NO_ERROR; 551 552 static const int32_t kMaxTries = 5; 553 int32_t tryCounter = kMaxTries; 554 555 do { 556 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 557 // keep them from going away if another thread re-creates the track during obtainBuffer() 558 sp<AudioRecordClientProxy> proxy; 559 sp<IMemory> iMem; 560 { 561 // start of lock scope 562 AutoMutex lock(mLock); 563 564 newSequence = mSequence; 565 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 566 if (status == DEAD_OBJECT) { 567 // re-create track, unless someone else has already done so 568 if (newSequence == oldSequence) { 569 status = restoreRecord_l("obtainBuffer"); 570 if (status != NO_ERROR) { 571 break; 572 } 573 } 574 } 575 oldSequence = newSequence; 576 577 // Keep the extra references 578 proxy = mProxy; 579 iMem = mCblkMemory; 580 581 // Non-blocking if track is stopped 582 if (!mActive) { 583 requested = &ClientProxy::kNonBlocking; 584 } 585 586 } // end of lock scope 587 588 buffer.mFrameCount = audioBuffer->frameCount; 589 // FIXME starts the requested timeout and elapsed over from scratch 590 status = proxy->obtainBuffer(&buffer, requested, elapsed); 591 592 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 593 594 audioBuffer->frameCount = buffer.mFrameCount; 595 audioBuffer->size = buffer.mFrameCount * mFrameSize; 596 audioBuffer->raw = buffer.mRaw; 597 if (nonContig != NULL) { 598 *nonContig = buffer.mNonContig; 599 } 600 return status; 601} 602 603void AudioRecord::releaseBuffer(Buffer* audioBuffer) 604{ 605 // all TRANSFER_* are valid 606 607 size_t stepCount = audioBuffer->size / mFrameSize; 608 if (stepCount == 0) { 609 return; 610 } 611 612 Proxy::Buffer buffer; 613 buffer.mFrameCount = stepCount; 614 buffer.mRaw = audioBuffer->raw; 615 616 AutoMutex lock(mLock); 617 mInOverrun = false; 618 mProxy->releaseBuffer(&buffer); 619 620 // the server does not automatically disable recorder on overrun, so no need to restart 621} 622 623audio_io_handle_t AudioRecord::getInput() const 624{ 625 AutoMutex lock(mLock); 626 return mInput; 627} 628 629// must be called with mLock held 630audio_io_handle_t AudioRecord::getInput_l() 631{ 632 mInput = AudioSystem::getInput(mInputSource, 633 mSampleRate, 634 mFormat, 635 mChannelMask, 636 mSessionId); 637 return mInput; 638} 639 640// ------------------------------------------------------------------------- 641 642ssize_t AudioRecord::read(void* buffer, size_t userSize) 643{ 644 if (mTransfer != TRANSFER_SYNC) { 645 return INVALID_OPERATION; 646 } 647 648 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 649 // sanity-check. user is most-likely passing an error code, and it would 650 // make the return value ambiguous (actualSize vs error). 651 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 652 return BAD_VALUE; 653 } 654 655 ssize_t read = 0; 656 Buffer audioBuffer; 657 658 while (userSize >= mFrameSize) { 659 audioBuffer.frameCount = userSize / mFrameSize; 660 661 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 662 if (err < 0) { 663 if (read > 0) { 664 break; 665 } 666 return ssize_t(err); 667 } 668 669 size_t bytesRead = audioBuffer.size; 670 memcpy(buffer, audioBuffer.i8, bytesRead); 671 buffer = ((char *) buffer) + bytesRead; 672 userSize -= bytesRead; 673 read += bytesRead; 674 675 releaseBuffer(&audioBuffer); 676 } 677 678 return read; 679} 680 681// ------------------------------------------------------------------------- 682 683nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) 684{ 685 mLock.lock(); 686 if (mAwaitBoost) { 687 mAwaitBoost = false; 688 mLock.unlock(); 689 static const int32_t kMaxTries = 5; 690 int32_t tryCounter = kMaxTries; 691 uint32_t pollUs = 10000; 692 do { 693 int policy = sched_getscheduler(0); 694 if (policy == SCHED_FIFO || policy == SCHED_RR) { 695 break; 696 } 697 usleep(pollUs); 698 pollUs <<= 1; 699 } while (tryCounter-- > 0); 700 if (tryCounter < 0) { 701 ALOGE("did not receive expected priority boost on time"); 702 } 703 // Run again immediately 704 return 0; 705 } 706 707 // Can only reference mCblk while locked 708 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 709 710 // Check for track invalidation 711 if (flags & CBLK_INVALID) { 712 (void) restoreRecord_l("processAudioBuffer"); 713 mLock.unlock(); 714 // Run again immediately, but with a new IAudioRecord 715 return 0; 716 } 717 718 bool active = mActive; 719 720 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 721 bool newOverrun = false; 722 if (flags & CBLK_OVERRUN) { 723 if (!mInOverrun) { 724 mInOverrun = true; 725 newOverrun = true; 726 } 727 } 728 729 // Get current position of server 730 size_t position = mProxy->getPosition(); 731 732 // Manage marker callback 733 bool markerReached = false; 734 size_t markerPosition = mMarkerPosition; 735 // FIXME fails for wraparound, need 64 bits 736 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 737 mMarkerReached = markerReached = true; 738 } 739 740 // Determine the number of new position callback(s) that will be needed, while locked 741 size_t newPosCount = 0; 742 size_t newPosition = mNewPosition; 743 uint32_t updatePeriod = mUpdatePeriod; 744 // FIXME fails for wraparound, need 64 bits 745 if (updatePeriod > 0 && position >= newPosition) { 746 newPosCount = ((position - newPosition) / updatePeriod) + 1; 747 mNewPosition += updatePeriod * newPosCount; 748 } 749 750 // Cache other fields that will be needed soon 751 size_t notificationFrames = mNotificationFrames; 752 if (mRefreshRemaining) { 753 mRefreshRemaining = false; 754 mRemainingFrames = notificationFrames; 755 mRetryOnPartialBuffer = false; 756 } 757 size_t misalignment = mProxy->getMisalignment(); 758 int32_t sequence = mSequence; 759 760 // These fields don't need to be cached, because they are assigned only by set(): 761 // mTransfer, mCbf, mUserData, mSampleRate 762 763 mLock.unlock(); 764 765 // perform callbacks while unlocked 766 if (newOverrun) { 767 mCbf(EVENT_OVERRUN, mUserData, NULL); 768 } 769 if (markerReached) { 770 mCbf(EVENT_MARKER, mUserData, &markerPosition); 771 } 772 while (newPosCount > 0) { 773 size_t temp = newPosition; 774 mCbf(EVENT_NEW_POS, mUserData, &temp); 775 newPosition += updatePeriod; 776 newPosCount--; 777 } 778 if (mObservedSequence != sequence) { 779 mObservedSequence = sequence; 780 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 781 } 782 783 // if inactive, then don't run me again until re-started 784 if (!active) { 785 return NS_INACTIVE; 786 } 787 788 // Compute the estimated time until the next timed event (position, markers) 789 uint32_t minFrames = ~0; 790 if (!markerReached && position < markerPosition) { 791 minFrames = markerPosition - position; 792 } 793 if (updatePeriod > 0 && updatePeriod < minFrames) { 794 minFrames = updatePeriod; 795 } 796 797 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 798 static const uint32_t kPoll = 0; 799 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 800 minFrames = kPoll * notificationFrames; 801 } 802 803 // Convert frame units to time units 804 nsecs_t ns = NS_WHENEVER; 805 if (minFrames != (uint32_t) ~0) { 806 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 807 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 808 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 809 } 810 811 // If not supplying data by EVENT_MORE_DATA, then we're done 812 if (mTransfer != TRANSFER_CALLBACK) { 813 return ns; 814 } 815 816 struct timespec timeout; 817 const struct timespec *requested = &ClientProxy::kForever; 818 if (ns != NS_WHENEVER) { 819 timeout.tv_sec = ns / 1000000000LL; 820 timeout.tv_nsec = ns % 1000000000LL; 821 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 822 requested = &timeout; 823 } 824 825 while (mRemainingFrames > 0) { 826 827 Buffer audioBuffer; 828 audioBuffer.frameCount = mRemainingFrames; 829 size_t nonContig; 830 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 831 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 832 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 833 requested = &ClientProxy::kNonBlocking; 834 size_t avail = audioBuffer.frameCount + nonContig; 835 ALOGV("obtainBuffer(%u) returned %u = %u + %u", 836 mRemainingFrames, avail, audioBuffer.frameCount, nonContig); 837 if (err != NO_ERROR) { 838 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 839 break; 840 } 841 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 842 return NS_NEVER; 843 } 844 845 if (mRetryOnPartialBuffer) { 846 mRetryOnPartialBuffer = false; 847 if (avail < mRemainingFrames) { 848 int64_t myns = ((mRemainingFrames - avail) * 849 1100000000LL) / mSampleRate; 850 if (ns < 0 || myns < ns) { 851 ns = myns; 852 } 853 return ns; 854 } 855 } 856 857 size_t reqSize = audioBuffer.size; 858 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 859 size_t readSize = audioBuffer.size; 860 861 // Sanity check on returned size 862 if (ssize_t(readSize) < 0 || readSize > reqSize) { 863 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 864 reqSize, (int) readSize); 865 return NS_NEVER; 866 } 867 868 if (readSize == 0) { 869 // The callback is done consuming buffers 870 // Keep this thread going to handle timed events and 871 // still try to provide more data in intervals of WAIT_PERIOD_MS 872 // but don't just loop and block the CPU, so wait 873 return WAIT_PERIOD_MS * 1000000LL; 874 } 875 876 size_t releasedFrames = readSize / mFrameSize; 877 audioBuffer.frameCount = releasedFrames; 878 mRemainingFrames -= releasedFrames; 879 if (misalignment >= releasedFrames) { 880 misalignment -= releasedFrames; 881 } else { 882 misalignment = 0; 883 } 884 885 releaseBuffer(&audioBuffer); 886 887 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 888 // if callback doesn't like to accept the full chunk 889 if (readSize < reqSize) { 890 continue; 891 } 892 893 // There could be enough non-contiguous frames available to satisfy the remaining request 894 if (mRemainingFrames <= nonContig) { 895 continue; 896 } 897 898#if 0 899 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 900 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 901 // that total to a sum == notificationFrames. 902 if (0 < misalignment && misalignment <= mRemainingFrames) { 903 mRemainingFrames = misalignment; 904 return (mRemainingFrames * 1100000000LL) / mSampleRate; 905 } 906#endif 907 908 } 909 mRemainingFrames = notificationFrames; 910 mRetryOnPartialBuffer = true; 911 912 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 913 return 0; 914} 915 916status_t AudioRecord::restoreRecord_l(const char *from) 917{ 918 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 919 ++mSequence; 920 status_t result; 921 922 // if the new IAudioRecord is created, openRecord_l() will modify the 923 // following member variables: mAudioRecord, mCblkMemory and mCblk. 924 // It will also delete the strong references on previous IAudioRecord and IMemory 925 size_t position = mProxy->getPosition(); 926 mNewPosition = position + mUpdatePeriod; 927 result = openRecord_l(mSampleRate, mFormat, mFrameCount, mFlags, getInput_l(), position); 928 if (result == NO_ERROR) { 929 if (mActive) { 930 // callback thread or sync event hasn't changed 931 // FIXME this fails if we have a new AudioFlinger instance 932 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 933 } 934 } 935 if (result != NO_ERROR) { 936 ALOGW("restoreRecord_l() failed status %d", result); 937 mActive = false; 938 } 939 940 return result; 941} 942 943// ========================================================================= 944 945void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who) 946{ 947 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 948 if (audioRecord != 0) { 949 AutoMutex lock(audioRecord->mLock); 950 audioRecord->mProxy->binderDied(); 951 } 952} 953 954// ========================================================================= 955 956AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 957 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 958{ 959} 960 961AudioRecord::AudioRecordThread::~AudioRecordThread() 962{ 963} 964 965bool AudioRecord::AudioRecordThread::threadLoop() 966{ 967 { 968 AutoMutex _l(mMyLock); 969 if (mPaused) { 970 mMyCond.wait(mMyLock); 971 // caller will check for exitPending() 972 return true; 973 } 974 } 975 nsecs_t ns = mReceiver.processAudioBuffer(this); 976 switch (ns) { 977 case 0: 978 return true; 979 case NS_WHENEVER: 980 sleep(1); 981 return true; 982 case NS_INACTIVE: 983 pauseConditional(); 984 return true; 985 case NS_NEVER: 986 return false; 987 default: 988 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 989 struct timespec req; 990 req.tv_sec = ns / 1000000000LL; 991 req.tv_nsec = ns % 1000000000LL; 992 nanosleep(&req, NULL /*rem*/); 993 return true; 994 } 995} 996 997void AudioRecord::AudioRecordThread::requestExit() 998{ 999 // must be in this order to avoid a race condition 1000 Thread::requestExit(); 1001 resume(); 1002} 1003 1004void AudioRecord::AudioRecordThread::pause() 1005{ 1006 AutoMutex _l(mMyLock); 1007 mPaused = true; 1008 mResumeLatch = false; 1009} 1010 1011void AudioRecord::AudioRecordThread::pauseConditional() 1012{ 1013 AutoMutex _l(mMyLock); 1014 if (mResumeLatch) { 1015 mResumeLatch = false; 1016 } else { 1017 mPaused = true; 1018 } 1019} 1020 1021void AudioRecord::AudioRecordThread::resume() 1022{ 1023 AutoMutex _l(mMyLock); 1024 if (mPaused) { 1025 mPaused = false; 1026 mResumeLatch = false; 1027 mMyCond.signal(); 1028 } else { 1029 mResumeLatch = true; 1030 } 1031} 1032 1033// ------------------------------------------------------------------------- 1034 1035}; // namespace android 1036