AudioRecord.cpp revision 743649fa70392b668377fb507d251b346c7b2769
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, mFlags, 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_input_flags_t flags, 439 audio_io_handle_t input, 440 size_t epoch) 441{ 442 status_t status; 443 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 444 if (audioFlinger == 0) { 445 ALOGE("Could not get audioflinger"); 446 return NO_INIT; 447 } 448 449 pid_t tid = -1; 450 // FIXME see similar logic at AudioTrack for tid 451 452 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 453 int originalSessionId = mSessionId; 454 sp<IAudioRecord> record = audioFlinger->openRecord(input, 455 sampleRate, format, 456 mChannelMask, 457 frameCount, 458 &trackFlags, 459 tid, 460 &mSessionId, 461 &status); 462 ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId, 463 "session ID changed from %d to %d", originalSessionId, mSessionId); 464 465 if (record == 0) { 466 ALOGE("AudioFlinger could not create record track, status: %d", status); 467 return status; 468 } 469 sp<IMemory> iMem = record->getCblk(); 470 if (iMem == 0) { 471 ALOGE("Could not get control block"); 472 return NO_INIT; 473 } 474 if (mAudioRecord != 0) { 475 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 476 mDeathNotifier.clear(); 477 } 478 mAudioRecord = record; 479 mCblkMemory = iMem; 480 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 481 mCblk = cblk; 482 483 // starting address of buffers in shared memory 484 void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); 485 486 // update proxy 487 mProxy = new AudioRecordClientProxy(cblk, buffers, frameCount, mFrameSize); 488 mProxy->setEpoch(epoch); 489 mProxy->setMinimum(mNotificationFrames); 490 491 mDeathNotifier = new DeathNotifier(this); 492 mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); 493 494 return NO_ERROR; 495} 496 497status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 498{ 499 if (audioBuffer == NULL) { 500 return BAD_VALUE; 501 } 502 if (mTransfer != TRANSFER_OBTAIN) { 503 audioBuffer->frameCount = 0; 504 audioBuffer->size = 0; 505 audioBuffer->raw = NULL; 506 return INVALID_OPERATION; 507 } 508 509 const struct timespec *requested; 510 if (waitCount == -1) { 511 requested = &ClientProxy::kForever; 512 } else if (waitCount == 0) { 513 requested = &ClientProxy::kNonBlocking; 514 } else if (waitCount > 0) { 515 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 516 struct timespec timeout; 517 timeout.tv_sec = ms / 1000; 518 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 519 requested = &timeout; 520 } else { 521 ALOGE("%s invalid waitCount %d", __func__, waitCount); 522 requested = NULL; 523 } 524 return obtainBuffer(audioBuffer, requested); 525} 526 527status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 528 struct timespec *elapsed, size_t *nonContig) 529{ 530 // previous and new IAudioRecord sequence numbers are used to detect track re-creation 531 uint32_t oldSequence = 0; 532 uint32_t newSequence; 533 534 Proxy::Buffer buffer; 535 status_t status = NO_ERROR; 536 537 static const int32_t kMaxTries = 5; 538 int32_t tryCounter = kMaxTries; 539 540 do { 541 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 542 // keep them from going away if another thread re-creates the track during obtainBuffer() 543 sp<AudioRecordClientProxy> proxy; 544 sp<IMemory> iMem; 545 { 546 // start of lock scope 547 AutoMutex lock(mLock); 548 549 newSequence = mSequence; 550 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 551 if (status == DEAD_OBJECT) { 552 // re-create track, unless someone else has already done so 553 if (newSequence == oldSequence) { 554 status = restoreRecord_l("obtainBuffer"); 555 if (status != NO_ERROR) { 556 break; 557 } 558 } 559 } 560 oldSequence = newSequence; 561 562 // Keep the extra references 563 proxy = mProxy; 564 iMem = mCblkMemory; 565 566 // Non-blocking if track is stopped 567 if (!mActive) { 568 requested = &ClientProxy::kNonBlocking; 569 } 570 571 } // end of lock scope 572 573 buffer.mFrameCount = audioBuffer->frameCount; 574 // FIXME starts the requested timeout and elapsed over from scratch 575 status = proxy->obtainBuffer(&buffer, requested, elapsed); 576 577 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 578 579 audioBuffer->frameCount = buffer.mFrameCount; 580 audioBuffer->size = buffer.mFrameCount * mFrameSize; 581 audioBuffer->raw = buffer.mRaw; 582 if (nonContig != NULL) { 583 *nonContig = buffer.mNonContig; 584 } 585 return status; 586} 587 588void AudioRecord::releaseBuffer(Buffer* audioBuffer) 589{ 590 // all TRANSFER_* are valid 591 592 size_t stepCount = audioBuffer->size / mFrameSize; 593 if (stepCount == 0) { 594 return; 595 } 596 597 Proxy::Buffer buffer; 598 buffer.mFrameCount = stepCount; 599 buffer.mRaw = audioBuffer->raw; 600 601 AutoMutex lock(mLock); 602 mInOverrun = false; 603 mProxy->releaseBuffer(&buffer); 604 605 // the server does not automatically disable recorder on overrun, so no need to restart 606} 607 608audio_io_handle_t AudioRecord::getInput() const 609{ 610 AutoMutex lock(mLock); 611 return mInput; 612} 613 614// must be called with mLock held 615audio_io_handle_t AudioRecord::getInput_l() 616{ 617 mInput = AudioSystem::getInput(mInputSource, 618 mSampleRate, 619 mFormat, 620 mChannelMask, 621 mSessionId); 622 return mInput; 623} 624 625// ------------------------------------------------------------------------- 626 627ssize_t AudioRecord::read(void* buffer, size_t userSize) 628{ 629 if (mTransfer != TRANSFER_SYNC) { 630 return INVALID_OPERATION; 631 } 632 633 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 634 // sanity-check. user is most-likely passing an error code, and it would 635 // make the return value ambiguous (actualSize vs error). 636 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 637 return BAD_VALUE; 638 } 639 640 ssize_t read = 0; 641 Buffer audioBuffer; 642 643 while (userSize >= mFrameSize) { 644 audioBuffer.frameCount = userSize / mFrameSize; 645 646 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 647 if (err < 0) { 648 if (read > 0) { 649 break; 650 } 651 return ssize_t(err); 652 } 653 654 size_t bytesRead = audioBuffer.size; 655 memcpy(buffer, audioBuffer.i8, bytesRead); 656 buffer = ((char *) buffer) + bytesRead; 657 userSize -= bytesRead; 658 read += bytesRead; 659 660 releaseBuffer(&audioBuffer); 661 } 662 663 return read; 664} 665 666// ------------------------------------------------------------------------- 667 668nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) 669{ 670 mLock.lock(); 671 if (mAwaitBoost) { 672 mAwaitBoost = false; 673 mLock.unlock(); 674 static const int32_t kMaxTries = 5; 675 int32_t tryCounter = kMaxTries; 676 uint32_t pollUs = 10000; 677 do { 678 int policy = sched_getscheduler(0); 679 if (policy == SCHED_FIFO || policy == SCHED_RR) { 680 break; 681 } 682 usleep(pollUs); 683 pollUs <<= 1; 684 } while (tryCounter-- > 0); 685 if (tryCounter < 0) { 686 ALOGE("did not receive expected priority boost on time"); 687 } 688 // Run again immediately 689 return 0; 690 } 691 692 // Can only reference mCblk while locked 693 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 694 695 // Check for track invalidation 696 if (flags & CBLK_INVALID) { 697 (void) restoreRecord_l("processAudioBuffer"); 698 mLock.unlock(); 699 // Run again immediately, but with a new IAudioRecord 700 return 0; 701 } 702 703 bool active = mActive; 704 705 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 706 bool newOverrun = false; 707 if (flags & CBLK_OVERRUN) { 708 if (!mInOverrun) { 709 mInOverrun = true; 710 newOverrun = true; 711 } 712 } 713 714 // Get current position of server 715 size_t position = mProxy->getPosition(); 716 717 // Manage marker callback 718 bool markerReached = false; 719 size_t markerPosition = mMarkerPosition; 720 // FIXME fails for wraparound, need 64 bits 721 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 722 mMarkerReached = markerReached = true; 723 } 724 725 // Determine the number of new position callback(s) that will be needed, while locked 726 size_t newPosCount = 0; 727 size_t newPosition = mNewPosition; 728 uint32_t updatePeriod = mUpdatePeriod; 729 // FIXME fails for wraparound, need 64 bits 730 if (updatePeriod > 0 && position >= newPosition) { 731 newPosCount = ((position - newPosition) / updatePeriod) + 1; 732 mNewPosition += updatePeriod * newPosCount; 733 } 734 735 // Cache other fields that will be needed soon 736 size_t notificationFrames = mNotificationFrames; 737 if (mRefreshRemaining) { 738 mRefreshRemaining = false; 739 mRemainingFrames = notificationFrames; 740 mRetryOnPartialBuffer = false; 741 } 742 size_t misalignment = mProxy->getMisalignment(); 743 int32_t sequence = mSequence; 744 745 // These fields don't need to be cached, because they are assigned only by set(): 746 // mTransfer, mCbf, mUserData, mSampleRate 747 748 mLock.unlock(); 749 750 // perform callbacks while unlocked 751 if (newOverrun) { 752 mCbf(EVENT_OVERRUN, mUserData, NULL); 753 } 754 if (markerReached) { 755 mCbf(EVENT_MARKER, mUserData, &markerPosition); 756 } 757 while (newPosCount > 0) { 758 size_t temp = newPosition; 759 mCbf(EVENT_NEW_POS, mUserData, &temp); 760 newPosition += updatePeriod; 761 newPosCount--; 762 } 763 if (mObservedSequence != sequence) { 764 mObservedSequence = sequence; 765 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 766 } 767 768 // if inactive, then don't run me again until re-started 769 if (!active) { 770 return NS_INACTIVE; 771 } 772 773 // Compute the estimated time until the next timed event (position, markers) 774 uint32_t minFrames = ~0; 775 if (!markerReached && position < markerPosition) { 776 minFrames = markerPosition - position; 777 } 778 if (updatePeriod > 0 && updatePeriod < minFrames) { 779 minFrames = updatePeriod; 780 } 781 782 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 783 static const uint32_t kPoll = 0; 784 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 785 minFrames = kPoll * notificationFrames; 786 } 787 788 // Convert frame units to time units 789 nsecs_t ns = NS_WHENEVER; 790 if (minFrames != (uint32_t) ~0) { 791 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 792 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 793 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 794 } 795 796 // If not supplying data by EVENT_MORE_DATA, then we're done 797 if (mTransfer != TRANSFER_CALLBACK) { 798 return ns; 799 } 800 801 struct timespec timeout; 802 const struct timespec *requested = &ClientProxy::kForever; 803 if (ns != NS_WHENEVER) { 804 timeout.tv_sec = ns / 1000000000LL; 805 timeout.tv_nsec = ns % 1000000000LL; 806 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 807 requested = &timeout; 808 } 809 810 while (mRemainingFrames > 0) { 811 812 Buffer audioBuffer; 813 audioBuffer.frameCount = mRemainingFrames; 814 size_t nonContig; 815 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 816 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 817 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 818 requested = &ClientProxy::kNonBlocking; 819 size_t avail = audioBuffer.frameCount + nonContig; 820 ALOGV("obtainBuffer(%u) returned %u = %u + %u", 821 mRemainingFrames, avail, audioBuffer.frameCount, nonContig); 822 if (err != NO_ERROR) { 823 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 824 break; 825 } 826 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 827 return NS_NEVER; 828 } 829 830 if (mRetryOnPartialBuffer) { 831 mRetryOnPartialBuffer = false; 832 if (avail < mRemainingFrames) { 833 int64_t myns = ((mRemainingFrames - avail) * 834 1100000000LL) / mSampleRate; 835 if (ns < 0 || myns < ns) { 836 ns = myns; 837 } 838 return ns; 839 } 840 } 841 842 size_t reqSize = audioBuffer.size; 843 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 844 size_t readSize = audioBuffer.size; 845 846 // Sanity check on returned size 847 if (ssize_t(readSize) < 0 || readSize > reqSize) { 848 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 849 reqSize, (int) readSize); 850 return NS_NEVER; 851 } 852 853 if (readSize == 0) { 854 // The callback is done consuming buffers 855 // Keep this thread going to handle timed events and 856 // still try to provide more data in intervals of WAIT_PERIOD_MS 857 // but don't just loop and block the CPU, so wait 858 return WAIT_PERIOD_MS * 1000000LL; 859 } 860 861 size_t releasedFrames = readSize / mFrameSize; 862 audioBuffer.frameCount = releasedFrames; 863 mRemainingFrames -= releasedFrames; 864 if (misalignment >= releasedFrames) { 865 misalignment -= releasedFrames; 866 } else { 867 misalignment = 0; 868 } 869 870 releaseBuffer(&audioBuffer); 871 872 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 873 // if callback doesn't like to accept the full chunk 874 if (readSize < reqSize) { 875 continue; 876 } 877 878 // There could be enough non-contiguous frames available to satisfy the remaining request 879 if (mRemainingFrames <= nonContig) { 880 continue; 881 } 882 883#if 0 884 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 885 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 886 // that total to a sum == notificationFrames. 887 if (0 < misalignment && misalignment <= mRemainingFrames) { 888 mRemainingFrames = misalignment; 889 return (mRemainingFrames * 1100000000LL) / mSampleRate; 890 } 891#endif 892 893 } 894 mRemainingFrames = notificationFrames; 895 mRetryOnPartialBuffer = true; 896 897 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 898 return 0; 899} 900 901status_t AudioRecord::restoreRecord_l(const char *from) 902{ 903 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 904 ++mSequence; 905 status_t result; 906 907 // if the new IAudioRecord is created, openRecord_l() will modify the 908 // following member variables: mAudioRecord, mCblkMemory and mCblk. 909 // It will also delete the strong references on previous IAudioRecord and IMemory 910 size_t position = mProxy->getPosition(); 911 mNewPosition = position + mUpdatePeriod; 912 result = openRecord_l(mSampleRate, mFormat, mFrameCount, mFlags, getInput_l(), position); 913 if (result == NO_ERROR) { 914 if (mActive) { 915 // callback thread or sync event hasn't changed 916 // FIXME this fails if we have a new AudioFlinger instance 917 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 918 } 919 } 920 if (result != NO_ERROR) { 921 ALOGW("restoreRecord_l() failed status %d", result); 922 mActive = false; 923 } 924 925 return result; 926} 927 928// ========================================================================= 929 930void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who) 931{ 932 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 933 if (audioRecord != 0) { 934 AutoMutex lock(audioRecord->mLock); 935 audioRecord->mProxy->binderDied(); 936 } 937} 938 939// ========================================================================= 940 941AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 942 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 943{ 944} 945 946AudioRecord::AudioRecordThread::~AudioRecordThread() 947{ 948} 949 950bool AudioRecord::AudioRecordThread::threadLoop() 951{ 952 { 953 AutoMutex _l(mMyLock); 954 if (mPaused) { 955 mMyCond.wait(mMyLock); 956 // caller will check for exitPending() 957 return true; 958 } 959 } 960 nsecs_t ns = mReceiver.processAudioBuffer(this); 961 switch (ns) { 962 case 0: 963 return true; 964 case NS_WHENEVER: 965 sleep(1); 966 return true; 967 case NS_INACTIVE: 968 pauseConditional(); 969 return true; 970 case NS_NEVER: 971 return false; 972 default: 973 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 974 struct timespec req; 975 req.tv_sec = ns / 1000000000LL; 976 req.tv_nsec = ns % 1000000000LL; 977 nanosleep(&req, NULL /*rem*/); 978 return true; 979 } 980} 981 982void AudioRecord::AudioRecordThread::requestExit() 983{ 984 // must be in this order to avoid a race condition 985 Thread::requestExit(); 986 resume(); 987} 988 989void AudioRecord::AudioRecordThread::pause() 990{ 991 AutoMutex _l(mMyLock); 992 mPaused = true; 993 mResumeLatch = false; 994} 995 996void AudioRecord::AudioRecordThread::pauseConditional() 997{ 998 AutoMutex _l(mMyLock); 999 if (mResumeLatch) { 1000 mResumeLatch = false; 1001 } else { 1002 mPaused = true; 1003 } 1004} 1005 1006void AudioRecord::AudioRecordThread::resume() 1007{ 1008 AutoMutex _l(mMyLock); 1009 if (mPaused) { 1010 mPaused = false; 1011 mResumeLatch = false; 1012 mMyCond.signal(); 1013 } else { 1014 mResumeLatch = true; 1015 } 1016} 1017 1018// ------------------------------------------------------------------------- 1019 1020}; // namespace android 1021