AudioRecord.cpp revision 3aa03e40668dd90390d9f1702f8c576e15b366c3
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 // Assumes audio_is_linear_pcm(format) 64 uint32_t channelCount = popcount(channelMask); 65 size /= channelCount * audio_bytes_per_sample(format); 66 67 *frameCount = size; 68 return NO_ERROR; 69} 70 71// --------------------------------------------------------------------------- 72 73AudioRecord::AudioRecord() 74 : mStatus(NO_INIT), mSessionId(0), 75 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) 76{ 77} 78 79AudioRecord::AudioRecord( 80 audio_source_t inputSource, 81 uint32_t sampleRate, 82 audio_format_t format, 83 audio_channel_mask_t channelMask, 84 int frameCount, 85 callback_t cbf, 86 void* user, 87 int notificationFrames, 88 int sessionId, 89 transfer_type transferType, 90 audio_input_flags_t flags) 91 : mStatus(NO_INIT), mSessionId(0), 92 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 93 mPreviousSchedulingGroup(SP_DEFAULT), 94 mProxy(NULL) 95{ 96 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, 97 notificationFrames, false /*threadCanCallJava*/, sessionId, transferType); 98} 99 100AudioRecord::~AudioRecord() 101{ 102 if (mStatus == NO_ERROR) { 103 // Make sure that callback function exits in the case where 104 // it is looping on buffer empty condition in obtainBuffer(). 105 // Otherwise the callback thread will never exit. 106 stop(); 107 if (mAudioRecordThread != 0) { 108 mAudioRecordThread->requestExit(); // see comment in AudioRecord.h 109 mAudioRecordThread->requestExitAndWait(); 110 mAudioRecordThread.clear(); 111 } 112 if (mAudioRecord != 0) { 113 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 114 mAudioRecord.clear(); 115 } 116 IPCThreadState::self()->flushCommands(); 117 AudioSystem::releaseAudioSessionId(mSessionId); 118 } 119} 120 121status_t AudioRecord::set( 122 audio_source_t inputSource, 123 uint32_t sampleRate, 124 audio_format_t format, 125 audio_channel_mask_t channelMask, 126 int frameCountInt, 127 callback_t cbf, 128 void* user, 129 int notificationFrames, 130 bool threadCanCallJava, 131 int sessionId, 132 transfer_type transferType, 133 audio_input_flags_t flags) 134{ 135 switch (transferType) { 136 case TRANSFER_DEFAULT: 137 if (cbf == NULL || threadCanCallJava) { 138 transferType = TRANSFER_SYNC; 139 } else { 140 transferType = TRANSFER_CALLBACK; 141 } 142 break; 143 case TRANSFER_CALLBACK: 144 if (cbf == NULL) { 145 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL"); 146 return BAD_VALUE; 147 } 148 break; 149 case TRANSFER_OBTAIN: 150 case TRANSFER_SYNC: 151 break; 152 default: 153 ALOGE("Invalid transfer type %d", transferType); 154 return BAD_VALUE; 155 } 156 mTransfer = transferType; 157 158 // FIXME "int" here is legacy and will be replaced by size_t later 159 if (frameCountInt < 0) { 160 ALOGE("Invalid frame count %d", frameCountInt); 161 return BAD_VALUE; 162 } 163 size_t frameCount = frameCountInt; 164 165 ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask, 166 frameCount); 167 168 AutoMutex lock(mLock); 169 170 if (mAudioRecord != 0) { 171 ALOGE("Track already in use"); 172 return INVALID_OPERATION; 173 } 174 175 if (inputSource == AUDIO_SOURCE_DEFAULT) { 176 inputSource = AUDIO_SOURCE_MIC; 177 } 178 179 if (sampleRate == 0) { 180 ALOGE("Invalid sample rate %u", sampleRate); 181 return BAD_VALUE; 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 // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t) 211 mFrameSize = channelCount * audio_bytes_per_sample(format); 212 213 if (sessionId == 0 ) { 214 mSessionId = AudioSystem::newAudioSessionId(); 215 } else { 216 mSessionId = sessionId; 217 } 218 ALOGV("set(): mSessionId %d", mSessionId); 219 220 mFlags = flags; 221 222 audio_io_handle_t input = AudioSystem::getInput(inputSource, 223 sampleRate, 224 format, 225 channelMask, 226 mSessionId); 227 if (input == 0) { 228 ALOGE("Could not get audio input for record source %d", inputSource); 229 return BAD_VALUE; 230 } 231 232 // validate framecount 233 size_t minFrameCount = 0; 234 status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask); 235 if (status != NO_ERROR) { 236 ALOGE("getMinFrameCount() failed; status %d", status); 237 return status; 238 } 239 ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount); 240 241 if (frameCount == 0) { 242 frameCount = minFrameCount; 243 } else if (frameCount < minFrameCount) { 244 ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount); 245 return BAD_VALUE; 246 } 247 248 if (notificationFrames == 0) { 249 notificationFrames = frameCount/2; 250 } 251 252 // create the IAudioRecord 253 status = openRecord_l(sampleRate, format, frameCount, input, 0 /*epoch*/); 254 if (status != NO_ERROR) { 255 return status; 256 } 257 258 if (cbf != NULL) { 259 mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava); 260 mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO); 261 } 262 263 mStatus = NO_ERROR; 264 265 // Update buffer size in case it has been limited by AudioFlinger during track creation 266 mFrameCount = mCblk->frameCount_; 267 268 mActive = false; 269 mCbf = cbf; 270 mNotificationFrames = notificationFrames; 271 mRefreshRemaining = true; 272 mUserData = user; 273 // TODO: add audio hardware input latency here 274 mLatency = (1000*mFrameCount) / sampleRate; 275 mMarkerPosition = 0; 276 mMarkerReached = false; 277 mNewPosition = 0; 278 mUpdatePeriod = 0; 279 mInputSource = inputSource; 280 mInput = input; 281 AudioSystem::acquireAudioSessionId(mSessionId); 282 mSequence = 1; 283 mObservedSequence = mSequence; 284 mInOverrun = false; 285 286 return NO_ERROR; 287} 288 289// ------------------------------------------------------------------------- 290 291status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) 292{ 293 ALOGV("start, sync event %d trigger session %d", event, triggerSession); 294 295 AutoMutex lock(mLock); 296 if (mActive) { 297 return NO_ERROR; 298 } 299 300 // reset current position as seen by client to 0 301 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 302 303 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 304 int32_t flags = android_atomic_acquire_load(&mCblk->mFlags); 305 306 status_t status = NO_ERROR; 307 if (!(flags & CBLK_INVALID)) { 308 ALOGV("mAudioRecord->start()"); 309 status = mAudioRecord->start(event, triggerSession); 310 if (status == DEAD_OBJECT) { 311 flags |= CBLK_INVALID; 312 } 313 } 314 if (flags & CBLK_INVALID) { 315 status = restoreRecord_l("start"); 316 } 317 318 if (status != NO_ERROR) { 319 ALOGE("start() status %d", status); 320 } else { 321 mActive = true; 322 sp<AudioRecordThread> t = mAudioRecordThread; 323 if (t != 0) { 324 t->resume(); 325 } else { 326 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 327 get_sched_policy(0, &mPreviousSchedulingGroup); 328 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 329 } 330 } 331 332 return status; 333} 334 335void AudioRecord::stop() 336{ 337 AutoMutex lock(mLock); 338 if (!mActive) { 339 return; 340 } 341 342 mActive = false; 343 mProxy->interrupt(); 344 mAudioRecord->stop(); 345 // the record head position will reset to 0, so if a marker is set, we need 346 // to activate it again 347 mMarkerReached = false; 348 sp<AudioRecordThread> t = mAudioRecordThread; 349 if (t != 0) { 350 t->pause(); 351 } else { 352 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 353 set_sched_policy(0, mPreviousSchedulingGroup); 354 } 355} 356 357bool AudioRecord::stopped() const 358{ 359 AutoMutex lock(mLock); 360 return !mActive; 361} 362 363status_t AudioRecord::setMarkerPosition(uint32_t marker) 364{ 365 if (mCbf == NULL) { 366 return INVALID_OPERATION; 367 } 368 369 AutoMutex lock(mLock); 370 mMarkerPosition = marker; 371 mMarkerReached = false; 372 373 return NO_ERROR; 374} 375 376status_t AudioRecord::getMarkerPosition(uint32_t *marker) const 377{ 378 if (marker == NULL) { 379 return BAD_VALUE; 380 } 381 382 AutoMutex lock(mLock); 383 *marker = mMarkerPosition; 384 385 return NO_ERROR; 386} 387 388status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) 389{ 390 if (mCbf == NULL) { 391 return INVALID_OPERATION; 392 } 393 394 AutoMutex lock(mLock); 395 mNewPosition = mProxy->getPosition() + updatePeriod; 396 mUpdatePeriod = updatePeriod; 397 398 return NO_ERROR; 399} 400 401status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const 402{ 403 if (updatePeriod == NULL) { 404 return BAD_VALUE; 405 } 406 407 AutoMutex lock(mLock); 408 *updatePeriod = mUpdatePeriod; 409 410 return NO_ERROR; 411} 412 413status_t AudioRecord::getPosition(uint32_t *position) const 414{ 415 if (position == NULL) { 416 return BAD_VALUE; 417 } 418 419 AutoMutex lock(mLock); 420 *position = mProxy->getPosition(); 421 422 return NO_ERROR; 423} 424 425unsigned int AudioRecord::getInputFramesLost() const 426{ 427 // no need to check mActive, because if inactive this will return 0, which is what we want 428 return AudioSystem::getInputFramesLost(getInput()); 429} 430 431// ------------------------------------------------------------------------- 432 433// must be called with mLock held 434status_t AudioRecord::openRecord_l( 435 uint32_t sampleRate, 436 audio_format_t format, 437 size_t frameCount, 438 audio_io_handle_t input, 439 size_t epoch) 440{ 441 status_t status; 442 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 443 if (audioFlinger == 0) { 444 ALOGE("Could not get audioflinger"); 445 return NO_INIT; 446 } 447 448 pid_t tid = -1; 449 // FIXME see similar logic at AudioTrack for tid 450 451 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 452 int originalSessionId = mSessionId; 453 sp<IAudioRecord> record = audioFlinger->openRecord(input, 454 sampleRate, format, 455 mChannelMask, 456 frameCount, 457 &trackFlags, 458 tid, 459 &mSessionId, 460 &status); 461 ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId, 462 "session ID changed from %d to %d", originalSessionId, mSessionId); 463 464 if (record == 0) { 465 ALOGE("AudioFlinger could not create record track, status: %d", status); 466 return status; 467 } 468 sp<IMemory> iMem = record->getCblk(); 469 if (iMem == 0) { 470 ALOGE("Could not get control block"); 471 return NO_INIT; 472 } 473 if (mAudioRecord != 0) { 474 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 475 mDeathNotifier.clear(); 476 } 477 mAudioRecord = record; 478 mCblkMemory = iMem; 479 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 480 mCblk = cblk; 481 482 // starting address of buffers in shared memory 483 void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); 484 485 // update proxy 486 mProxy = new AudioRecordClientProxy(cblk, buffers, frameCount, mFrameSize); 487 mProxy->setEpoch(epoch); 488 mProxy->setMinimum(mNotificationFrames); 489 490 mDeathNotifier = new DeathNotifier(this); 491 mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); 492 493 return NO_ERROR; 494} 495 496status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 497{ 498 if (audioBuffer == NULL) { 499 return BAD_VALUE; 500 } 501 if (mTransfer != TRANSFER_OBTAIN) { 502 audioBuffer->frameCount = 0; 503 audioBuffer->size = 0; 504 audioBuffer->raw = NULL; 505 return INVALID_OPERATION; 506 } 507 508 const struct timespec *requested; 509 if (waitCount == -1) { 510 requested = &ClientProxy::kForever; 511 } else if (waitCount == 0) { 512 requested = &ClientProxy::kNonBlocking; 513 } else if (waitCount > 0) { 514 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 515 struct timespec timeout; 516 timeout.tv_sec = ms / 1000; 517 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 518 requested = &timeout; 519 } else { 520 ALOGE("%s invalid waitCount %d", __func__, waitCount); 521 requested = NULL; 522 } 523 return obtainBuffer(audioBuffer, requested); 524} 525 526status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 527 struct timespec *elapsed, size_t *nonContig) 528{ 529 // previous and new IAudioRecord sequence numbers are used to detect track re-creation 530 uint32_t oldSequence = 0; 531 uint32_t newSequence; 532 533 Proxy::Buffer buffer; 534 status_t status = NO_ERROR; 535 536 static const int32_t kMaxTries = 5; 537 int32_t tryCounter = kMaxTries; 538 539 do { 540 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 541 // keep them from going away if another thread re-creates the track during obtainBuffer() 542 sp<AudioRecordClientProxy> proxy; 543 sp<IMemory> iMem; 544 { 545 // start of lock scope 546 AutoMutex lock(mLock); 547 548 newSequence = mSequence; 549 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 550 if (status == DEAD_OBJECT) { 551 // re-create track, unless someone else has already done so 552 if (newSequence == oldSequence) { 553 status = restoreRecord_l("obtainBuffer"); 554 if (status != NO_ERROR) { 555 break; 556 } 557 } 558 } 559 oldSequence = newSequence; 560 561 // Keep the extra references 562 proxy = mProxy; 563 iMem = mCblkMemory; 564 565 // Non-blocking if track is stopped 566 if (!mActive) { 567 requested = &ClientProxy::kNonBlocking; 568 } 569 570 } // end of lock scope 571 572 buffer.mFrameCount = audioBuffer->frameCount; 573 // FIXME starts the requested timeout and elapsed over from scratch 574 status = proxy->obtainBuffer(&buffer, requested, elapsed); 575 576 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 577 578 audioBuffer->frameCount = buffer.mFrameCount; 579 audioBuffer->size = buffer.mFrameCount * mFrameSize; 580 audioBuffer->raw = buffer.mRaw; 581 if (nonContig != NULL) { 582 *nonContig = buffer.mNonContig; 583 } 584 return status; 585} 586 587void AudioRecord::releaseBuffer(Buffer* audioBuffer) 588{ 589 // all TRANSFER_* are valid 590 591 size_t stepCount = audioBuffer->size / mFrameSize; 592 if (stepCount == 0) { 593 return; 594 } 595 596 Proxy::Buffer buffer; 597 buffer.mFrameCount = stepCount; 598 buffer.mRaw = audioBuffer->raw; 599 600 AutoMutex lock(mLock); 601 mInOverrun = false; 602 mProxy->releaseBuffer(&buffer); 603 604 // the server does not automatically disable recorder on overrun, so no need to restart 605} 606 607audio_io_handle_t AudioRecord::getInput() const 608{ 609 AutoMutex lock(mLock); 610 return mInput; 611} 612 613// must be called with mLock held 614audio_io_handle_t AudioRecord::getInput_l() 615{ 616 mInput = AudioSystem::getInput(mInputSource, 617 mSampleRate, 618 mFormat, 619 mChannelMask, 620 mSessionId); 621 return mInput; 622} 623 624// ------------------------------------------------------------------------- 625 626ssize_t AudioRecord::read(void* buffer, size_t userSize) 627{ 628 if (mTransfer != TRANSFER_SYNC) { 629 return INVALID_OPERATION; 630 } 631 632 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 633 // sanity-check. user is most-likely passing an error code, and it would 634 // make the return value ambiguous (actualSize vs error). 635 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 636 return BAD_VALUE; 637 } 638 639 ssize_t read = 0; 640 Buffer audioBuffer; 641 642 while (userSize >= mFrameSize) { 643 audioBuffer.frameCount = userSize / mFrameSize; 644 645 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 646 if (err < 0) { 647 if (read > 0) { 648 break; 649 } 650 return ssize_t(err); 651 } 652 653 size_t bytesRead = audioBuffer.size; 654 memcpy(buffer, audioBuffer.i8, bytesRead); 655 buffer = ((char *) buffer) + bytesRead; 656 userSize -= bytesRead; 657 read += bytesRead; 658 659 releaseBuffer(&audioBuffer); 660 } 661 662 return read; 663} 664 665// ------------------------------------------------------------------------- 666 667nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) 668{ 669 mLock.lock(); 670 if (mAwaitBoost) { 671 mAwaitBoost = false; 672 mLock.unlock(); 673 static const int32_t kMaxTries = 5; 674 int32_t tryCounter = kMaxTries; 675 uint32_t pollUs = 10000; 676 do { 677 int policy = sched_getscheduler(0); 678 if (policy == SCHED_FIFO || policy == SCHED_RR) { 679 break; 680 } 681 usleep(pollUs); 682 pollUs <<= 1; 683 } while (tryCounter-- > 0); 684 if (tryCounter < 0) { 685 ALOGE("did not receive expected priority boost on time"); 686 } 687 // Run again immediately 688 return 0; 689 } 690 691 // Can only reference mCblk while locked 692 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 693 694 // Check for track invalidation 695 if (flags & CBLK_INVALID) { 696 (void) restoreRecord_l("processAudioBuffer"); 697 mLock.unlock(); 698 // Run again immediately, but with a new IAudioRecord 699 return 0; 700 } 701 702 bool active = mActive; 703 704 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 705 bool newOverrun = false; 706 if (flags & CBLK_OVERRUN) { 707 if (!mInOverrun) { 708 mInOverrun = true; 709 newOverrun = true; 710 } 711 } 712 713 // Get current position of server 714 size_t position = mProxy->getPosition(); 715 716 // Manage marker callback 717 bool markerReached = false; 718 size_t markerPosition = mMarkerPosition; 719 // FIXME fails for wraparound, need 64 bits 720 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 721 mMarkerReached = markerReached = true; 722 } 723 724 // Determine the number of new position callback(s) that will be needed, while locked 725 size_t newPosCount = 0; 726 size_t newPosition = mNewPosition; 727 uint32_t updatePeriod = mUpdatePeriod; 728 // FIXME fails for wraparound, need 64 bits 729 if (updatePeriod > 0 && position >= newPosition) { 730 newPosCount = ((position - newPosition) / updatePeriod) + 1; 731 mNewPosition += updatePeriod * newPosCount; 732 } 733 734 // Cache other fields that will be needed soon 735 size_t notificationFrames = mNotificationFrames; 736 if (mRefreshRemaining) { 737 mRefreshRemaining = false; 738 mRemainingFrames = notificationFrames; 739 mRetryOnPartialBuffer = false; 740 } 741 size_t misalignment = mProxy->getMisalignment(); 742 int32_t sequence = mSequence; 743 744 // These fields don't need to be cached, because they are assigned only by set(): 745 // mTransfer, mCbf, mUserData, mSampleRate 746 747 mLock.unlock(); 748 749 // perform callbacks while unlocked 750 if (newOverrun) { 751 mCbf(EVENT_OVERRUN, mUserData, NULL); 752 } 753 if (markerReached) { 754 mCbf(EVENT_MARKER, mUserData, &markerPosition); 755 } 756 while (newPosCount > 0) { 757 size_t temp = newPosition; 758 mCbf(EVENT_NEW_POS, mUserData, &temp); 759 newPosition += updatePeriod; 760 newPosCount--; 761 } 762 if (mObservedSequence != sequence) { 763 mObservedSequence = sequence; 764 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 765 } 766 767 // if inactive, then don't run me again until re-started 768 if (!active) { 769 return NS_INACTIVE; 770 } 771 772 // Compute the estimated time until the next timed event (position, markers) 773 uint32_t minFrames = ~0; 774 if (!markerReached && position < markerPosition) { 775 minFrames = markerPosition - position; 776 } 777 if (updatePeriod > 0 && updatePeriod < minFrames) { 778 minFrames = updatePeriod; 779 } 780 781 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 782 static const uint32_t kPoll = 0; 783 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 784 minFrames = kPoll * notificationFrames; 785 } 786 787 // Convert frame units to time units 788 nsecs_t ns = NS_WHENEVER; 789 if (minFrames != (uint32_t) ~0) { 790 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 791 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 792 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 793 } 794 795 // If not supplying data by EVENT_MORE_DATA, then we're done 796 if (mTransfer != TRANSFER_CALLBACK) { 797 return ns; 798 } 799 800 struct timespec timeout; 801 const struct timespec *requested = &ClientProxy::kForever; 802 if (ns != NS_WHENEVER) { 803 timeout.tv_sec = ns / 1000000000LL; 804 timeout.tv_nsec = ns % 1000000000LL; 805 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 806 requested = &timeout; 807 } 808 809 while (mRemainingFrames > 0) { 810 811 Buffer audioBuffer; 812 audioBuffer.frameCount = mRemainingFrames; 813 size_t nonContig; 814 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 815 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 816 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 817 requested = &ClientProxy::kNonBlocking; 818 size_t avail = audioBuffer.frameCount + nonContig; 819 ALOGV("obtainBuffer(%u) returned %u = %u + %u", 820 mRemainingFrames, avail, audioBuffer.frameCount, nonContig); 821 if (err != NO_ERROR) { 822 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 823 break; 824 } 825 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 826 return NS_NEVER; 827 } 828 829 if (mRetryOnPartialBuffer) { 830 mRetryOnPartialBuffer = false; 831 if (avail < mRemainingFrames) { 832 int64_t myns = ((mRemainingFrames - avail) * 833 1100000000LL) / mSampleRate; 834 if (ns < 0 || myns < ns) { 835 ns = myns; 836 } 837 return ns; 838 } 839 } 840 841 size_t reqSize = audioBuffer.size; 842 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 843 size_t readSize = audioBuffer.size; 844 845 // Sanity check on returned size 846 if (ssize_t(readSize) < 0 || readSize > reqSize) { 847 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 848 reqSize, (int) readSize); 849 return NS_NEVER; 850 } 851 852 if (readSize == 0) { 853 // The callback is done consuming buffers 854 // Keep this thread going to handle timed events and 855 // still try to provide more data in intervals of WAIT_PERIOD_MS 856 // but don't just loop and block the CPU, so wait 857 return WAIT_PERIOD_MS * 1000000LL; 858 } 859 860 size_t releasedFrames = readSize / mFrameSize; 861 audioBuffer.frameCount = releasedFrames; 862 mRemainingFrames -= releasedFrames; 863 if (misalignment >= releasedFrames) { 864 misalignment -= releasedFrames; 865 } else { 866 misalignment = 0; 867 } 868 869 releaseBuffer(&audioBuffer); 870 871 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 872 // if callback doesn't like to accept the full chunk 873 if (readSize < reqSize) { 874 continue; 875 } 876 877 // There could be enough non-contiguous frames available to satisfy the remaining request 878 if (mRemainingFrames <= nonContig) { 879 continue; 880 } 881 882#if 0 883 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 884 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 885 // that total to a sum == notificationFrames. 886 if (0 < misalignment && misalignment <= mRemainingFrames) { 887 mRemainingFrames = misalignment; 888 return (mRemainingFrames * 1100000000LL) / mSampleRate; 889 } 890#endif 891 892 } 893 mRemainingFrames = notificationFrames; 894 mRetryOnPartialBuffer = true; 895 896 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 897 return 0; 898} 899 900status_t AudioRecord::restoreRecord_l(const char *from) 901{ 902 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 903 ++mSequence; 904 status_t result; 905 906 // if the new IAudioRecord is created, openRecord_l() will modify the 907 // following member variables: mAudioRecord, mCblkMemory and mCblk. 908 // It will also delete the strong references on previous IAudioRecord and IMemory 909 size_t position = mProxy->getPosition(); 910 mNewPosition = position + mUpdatePeriod; 911 result = openRecord_l(mSampleRate, mFormat, mFrameCount, getInput_l(), position); 912 if (result == NO_ERROR) { 913 if (mActive) { 914 // callback thread or sync event hasn't changed 915 // FIXME this fails if we have a new AudioFlinger instance 916 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 917 } 918 } 919 if (result != NO_ERROR) { 920 ALOGW("restoreRecord_l() failed status %d", result); 921 mActive = false; 922 } 923 924 return result; 925} 926 927// ========================================================================= 928 929void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who) 930{ 931 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 932 if (audioRecord != 0) { 933 AutoMutex lock(audioRecord->mLock); 934 audioRecord->mProxy->binderDied(); 935 } 936} 937 938// ========================================================================= 939 940AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 941 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 942{ 943} 944 945AudioRecord::AudioRecordThread::~AudioRecordThread() 946{ 947} 948 949bool AudioRecord::AudioRecordThread::threadLoop() 950{ 951 { 952 AutoMutex _l(mMyLock); 953 if (mPaused) { 954 mMyCond.wait(mMyLock); 955 // caller will check for exitPending() 956 return true; 957 } 958 } 959 nsecs_t ns = mReceiver.processAudioBuffer(this); 960 switch (ns) { 961 case 0: 962 return true; 963 case NS_WHENEVER: 964 sleep(1); 965 return true; 966 case NS_INACTIVE: 967 pauseConditional(); 968 return true; 969 case NS_NEVER: 970 return false; 971 default: 972 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 973 struct timespec req; 974 req.tv_sec = ns / 1000000000LL; 975 req.tv_nsec = ns % 1000000000LL; 976 nanosleep(&req, NULL /*rem*/); 977 return true; 978 } 979} 980 981void AudioRecord::AudioRecordThread::requestExit() 982{ 983 // must be in this order to avoid a race condition 984 Thread::requestExit(); 985 resume(); 986} 987 988void AudioRecord::AudioRecordThread::pause() 989{ 990 AutoMutex _l(mMyLock); 991 mPaused = true; 992 mResumeLatch = false; 993} 994 995void AudioRecord::AudioRecordThread::pauseConditional() 996{ 997 AutoMutex _l(mMyLock); 998 if (mResumeLatch) { 999 mResumeLatch = false; 1000 } else { 1001 mPaused = true; 1002 } 1003} 1004 1005void AudioRecord::AudioRecordThread::resume() 1006{ 1007 AutoMutex _l(mMyLock); 1008 if (mPaused) { 1009 mPaused = false; 1010 mResumeLatch = false; 1011 mMyCond.signal(); 1012 } else { 1013 mResumeLatch = true; 1014 } 1015} 1016 1017// ------------------------------------------------------------------------- 1018 1019}; // namespace android 1020