AudioRecord.cpp revision 33f3177c08d238285b296d137e527ec99e34228f
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 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 450 pid_t tid = -1; 451 452 // Client can only express a preference for FAST. Server will perform additional tests. 453 // The only supported use case for FAST is callback transfer mode. 454 if (flags & AUDIO_INPUT_FLAG_FAST) { 455 if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) { 456 ALOGW("AUDIO_INPUT_FLAG_FAST denied by client"); 457 // once denied, do not request again if IAudioRecord is re-created 458 mFlags = (audio_input_flags_t) (flags & ~AUDIO_INPUT_FLAG_FAST); 459 } else { 460 trackFlags |= IAudioFlinger::TRACK_FAST; 461 tid = mAudioRecordThread->getTid(); 462 } 463 } 464 465 int originalSessionId = mSessionId; 466 sp<IAudioRecord> record = audioFlinger->openRecord(input, 467 sampleRate, format, 468 mChannelMask, 469 frameCount, 470 &trackFlags, 471 tid, 472 &mSessionId, 473 &status); 474 ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId, 475 "session ID changed from %d to %d", originalSessionId, mSessionId); 476 477 if (record == 0) { 478 ALOGE("AudioFlinger could not create record track, status: %d", status); 479 return status; 480 } 481 sp<IMemory> iMem = record->getCblk(); 482 if (iMem == 0) { 483 ALOGE("Could not get control block"); 484 return NO_INIT; 485 } 486 if (mAudioRecord != 0) { 487 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 488 mDeathNotifier.clear(); 489 } 490 mAudioRecord = record; 491 mCblkMemory = iMem; 492 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 493 mCblk = cblk; 494 495 // starting address of buffers in shared memory 496 void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); 497 498 // update proxy 499 mProxy = new AudioRecordClientProxy(cblk, buffers, frameCount, mFrameSize); 500 mProxy->setEpoch(epoch); 501 mProxy->setMinimum(mNotificationFrames); 502 503 mDeathNotifier = new DeathNotifier(this); 504 mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); 505 506 return NO_ERROR; 507} 508 509status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 510{ 511 if (audioBuffer == NULL) { 512 return BAD_VALUE; 513 } 514 if (mTransfer != TRANSFER_OBTAIN) { 515 audioBuffer->frameCount = 0; 516 audioBuffer->size = 0; 517 audioBuffer->raw = NULL; 518 return INVALID_OPERATION; 519 } 520 521 const struct timespec *requested; 522 if (waitCount == -1) { 523 requested = &ClientProxy::kForever; 524 } else if (waitCount == 0) { 525 requested = &ClientProxy::kNonBlocking; 526 } else if (waitCount > 0) { 527 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 528 struct timespec timeout; 529 timeout.tv_sec = ms / 1000; 530 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 531 requested = &timeout; 532 } else { 533 ALOGE("%s invalid waitCount %d", __func__, waitCount); 534 requested = NULL; 535 } 536 return obtainBuffer(audioBuffer, requested); 537} 538 539status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 540 struct timespec *elapsed, size_t *nonContig) 541{ 542 // previous and new IAudioRecord sequence numbers are used to detect track re-creation 543 uint32_t oldSequence = 0; 544 uint32_t newSequence; 545 546 Proxy::Buffer buffer; 547 status_t status = NO_ERROR; 548 549 static const int32_t kMaxTries = 5; 550 int32_t tryCounter = kMaxTries; 551 552 do { 553 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 554 // keep them from going away if another thread re-creates the track during obtainBuffer() 555 sp<AudioRecordClientProxy> proxy; 556 sp<IMemory> iMem; 557 { 558 // start of lock scope 559 AutoMutex lock(mLock); 560 561 newSequence = mSequence; 562 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 563 if (status == DEAD_OBJECT) { 564 // re-create track, unless someone else has already done so 565 if (newSequence == oldSequence) { 566 status = restoreRecord_l("obtainBuffer"); 567 if (status != NO_ERROR) { 568 break; 569 } 570 } 571 } 572 oldSequence = newSequence; 573 574 // Keep the extra references 575 proxy = mProxy; 576 iMem = mCblkMemory; 577 578 // Non-blocking if track is stopped 579 if (!mActive) { 580 requested = &ClientProxy::kNonBlocking; 581 } 582 583 } // end of lock scope 584 585 buffer.mFrameCount = audioBuffer->frameCount; 586 // FIXME starts the requested timeout and elapsed over from scratch 587 status = proxy->obtainBuffer(&buffer, requested, elapsed); 588 589 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 590 591 audioBuffer->frameCount = buffer.mFrameCount; 592 audioBuffer->size = buffer.mFrameCount * mFrameSize; 593 audioBuffer->raw = buffer.mRaw; 594 if (nonContig != NULL) { 595 *nonContig = buffer.mNonContig; 596 } 597 return status; 598} 599 600void AudioRecord::releaseBuffer(Buffer* audioBuffer) 601{ 602 // all TRANSFER_* are valid 603 604 size_t stepCount = audioBuffer->size / mFrameSize; 605 if (stepCount == 0) { 606 return; 607 } 608 609 Proxy::Buffer buffer; 610 buffer.mFrameCount = stepCount; 611 buffer.mRaw = audioBuffer->raw; 612 613 AutoMutex lock(mLock); 614 mInOverrun = false; 615 mProxy->releaseBuffer(&buffer); 616 617 // the server does not automatically disable recorder on overrun, so no need to restart 618} 619 620audio_io_handle_t AudioRecord::getInput() const 621{ 622 AutoMutex lock(mLock); 623 return mInput; 624} 625 626// must be called with mLock held 627audio_io_handle_t AudioRecord::getInput_l() 628{ 629 mInput = AudioSystem::getInput(mInputSource, 630 mSampleRate, 631 mFormat, 632 mChannelMask, 633 mSessionId); 634 return mInput; 635} 636 637// ------------------------------------------------------------------------- 638 639ssize_t AudioRecord::read(void* buffer, size_t userSize) 640{ 641 if (mTransfer != TRANSFER_SYNC) { 642 return INVALID_OPERATION; 643 } 644 645 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 646 // sanity-check. user is most-likely passing an error code, and it would 647 // make the return value ambiguous (actualSize vs error). 648 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 649 return BAD_VALUE; 650 } 651 652 ssize_t read = 0; 653 Buffer audioBuffer; 654 655 while (userSize >= mFrameSize) { 656 audioBuffer.frameCount = userSize / mFrameSize; 657 658 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 659 if (err < 0) { 660 if (read > 0) { 661 break; 662 } 663 return ssize_t(err); 664 } 665 666 size_t bytesRead = audioBuffer.size; 667 memcpy(buffer, audioBuffer.i8, bytesRead); 668 buffer = ((char *) buffer) + bytesRead; 669 userSize -= bytesRead; 670 read += bytesRead; 671 672 releaseBuffer(&audioBuffer); 673 } 674 675 return read; 676} 677 678// ------------------------------------------------------------------------- 679 680nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) 681{ 682 mLock.lock(); 683 if (mAwaitBoost) { 684 mAwaitBoost = false; 685 mLock.unlock(); 686 static const int32_t kMaxTries = 5; 687 int32_t tryCounter = kMaxTries; 688 uint32_t pollUs = 10000; 689 do { 690 int policy = sched_getscheduler(0); 691 if (policy == SCHED_FIFO || policy == SCHED_RR) { 692 break; 693 } 694 usleep(pollUs); 695 pollUs <<= 1; 696 } while (tryCounter-- > 0); 697 if (tryCounter < 0) { 698 ALOGE("did not receive expected priority boost on time"); 699 } 700 // Run again immediately 701 return 0; 702 } 703 704 // Can only reference mCblk while locked 705 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 706 707 // Check for track invalidation 708 if (flags & CBLK_INVALID) { 709 (void) restoreRecord_l("processAudioBuffer"); 710 mLock.unlock(); 711 // Run again immediately, but with a new IAudioRecord 712 return 0; 713 } 714 715 bool active = mActive; 716 717 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 718 bool newOverrun = false; 719 if (flags & CBLK_OVERRUN) { 720 if (!mInOverrun) { 721 mInOverrun = true; 722 newOverrun = true; 723 } 724 } 725 726 // Get current position of server 727 size_t position = mProxy->getPosition(); 728 729 // Manage marker callback 730 bool markerReached = false; 731 size_t markerPosition = mMarkerPosition; 732 // FIXME fails for wraparound, need 64 bits 733 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 734 mMarkerReached = markerReached = true; 735 } 736 737 // Determine the number of new position callback(s) that will be needed, while locked 738 size_t newPosCount = 0; 739 size_t newPosition = mNewPosition; 740 uint32_t updatePeriod = mUpdatePeriod; 741 // FIXME fails for wraparound, need 64 bits 742 if (updatePeriod > 0 && position >= newPosition) { 743 newPosCount = ((position - newPosition) / updatePeriod) + 1; 744 mNewPosition += updatePeriod * newPosCount; 745 } 746 747 // Cache other fields that will be needed soon 748 size_t notificationFrames = mNotificationFrames; 749 if (mRefreshRemaining) { 750 mRefreshRemaining = false; 751 mRemainingFrames = notificationFrames; 752 mRetryOnPartialBuffer = false; 753 } 754 size_t misalignment = mProxy->getMisalignment(); 755 int32_t sequence = mSequence; 756 757 // These fields don't need to be cached, because they are assigned only by set(): 758 // mTransfer, mCbf, mUserData, mSampleRate 759 760 mLock.unlock(); 761 762 // perform callbacks while unlocked 763 if (newOverrun) { 764 mCbf(EVENT_OVERRUN, mUserData, NULL); 765 } 766 if (markerReached) { 767 mCbf(EVENT_MARKER, mUserData, &markerPosition); 768 } 769 while (newPosCount > 0) { 770 size_t temp = newPosition; 771 mCbf(EVENT_NEW_POS, mUserData, &temp); 772 newPosition += updatePeriod; 773 newPosCount--; 774 } 775 if (mObservedSequence != sequence) { 776 mObservedSequence = sequence; 777 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 778 } 779 780 // if inactive, then don't run me again until re-started 781 if (!active) { 782 return NS_INACTIVE; 783 } 784 785 // Compute the estimated time until the next timed event (position, markers) 786 uint32_t minFrames = ~0; 787 if (!markerReached && position < markerPosition) { 788 minFrames = markerPosition - position; 789 } 790 if (updatePeriod > 0 && updatePeriod < minFrames) { 791 minFrames = updatePeriod; 792 } 793 794 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 795 static const uint32_t kPoll = 0; 796 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 797 minFrames = kPoll * notificationFrames; 798 } 799 800 // Convert frame units to time units 801 nsecs_t ns = NS_WHENEVER; 802 if (minFrames != (uint32_t) ~0) { 803 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 804 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 805 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 806 } 807 808 // If not supplying data by EVENT_MORE_DATA, then we're done 809 if (mTransfer != TRANSFER_CALLBACK) { 810 return ns; 811 } 812 813 struct timespec timeout; 814 const struct timespec *requested = &ClientProxy::kForever; 815 if (ns != NS_WHENEVER) { 816 timeout.tv_sec = ns / 1000000000LL; 817 timeout.tv_nsec = ns % 1000000000LL; 818 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 819 requested = &timeout; 820 } 821 822 while (mRemainingFrames > 0) { 823 824 Buffer audioBuffer; 825 audioBuffer.frameCount = mRemainingFrames; 826 size_t nonContig; 827 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 828 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 829 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 830 requested = &ClientProxy::kNonBlocking; 831 size_t avail = audioBuffer.frameCount + nonContig; 832 ALOGV("obtainBuffer(%u) returned %u = %u + %u", 833 mRemainingFrames, avail, audioBuffer.frameCount, nonContig); 834 if (err != NO_ERROR) { 835 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 836 break; 837 } 838 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 839 return NS_NEVER; 840 } 841 842 if (mRetryOnPartialBuffer) { 843 mRetryOnPartialBuffer = false; 844 if (avail < mRemainingFrames) { 845 int64_t myns = ((mRemainingFrames - avail) * 846 1100000000LL) / mSampleRate; 847 if (ns < 0 || myns < ns) { 848 ns = myns; 849 } 850 return ns; 851 } 852 } 853 854 size_t reqSize = audioBuffer.size; 855 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 856 size_t readSize = audioBuffer.size; 857 858 // Sanity check on returned size 859 if (ssize_t(readSize) < 0 || readSize > reqSize) { 860 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 861 reqSize, (int) readSize); 862 return NS_NEVER; 863 } 864 865 if (readSize == 0) { 866 // The callback is done consuming buffers 867 // Keep this thread going to handle timed events and 868 // still try to provide more data in intervals of WAIT_PERIOD_MS 869 // but don't just loop and block the CPU, so wait 870 return WAIT_PERIOD_MS * 1000000LL; 871 } 872 873 size_t releasedFrames = readSize / mFrameSize; 874 audioBuffer.frameCount = releasedFrames; 875 mRemainingFrames -= releasedFrames; 876 if (misalignment >= releasedFrames) { 877 misalignment -= releasedFrames; 878 } else { 879 misalignment = 0; 880 } 881 882 releaseBuffer(&audioBuffer); 883 884 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 885 // if callback doesn't like to accept the full chunk 886 if (readSize < reqSize) { 887 continue; 888 } 889 890 // There could be enough non-contiguous frames available to satisfy the remaining request 891 if (mRemainingFrames <= nonContig) { 892 continue; 893 } 894 895#if 0 896 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 897 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 898 // that total to a sum == notificationFrames. 899 if (0 < misalignment && misalignment <= mRemainingFrames) { 900 mRemainingFrames = misalignment; 901 return (mRemainingFrames * 1100000000LL) / mSampleRate; 902 } 903#endif 904 905 } 906 mRemainingFrames = notificationFrames; 907 mRetryOnPartialBuffer = true; 908 909 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 910 return 0; 911} 912 913status_t AudioRecord::restoreRecord_l(const char *from) 914{ 915 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 916 ++mSequence; 917 status_t result; 918 919 // if the new IAudioRecord is created, openRecord_l() will modify the 920 // following member variables: mAudioRecord, mCblkMemory and mCblk. 921 // It will also delete the strong references on previous IAudioRecord and IMemory 922 size_t position = mProxy->getPosition(); 923 mNewPosition = position + mUpdatePeriod; 924 result = openRecord_l(mSampleRate, mFormat, mFrameCount, mFlags, getInput_l(), position); 925 if (result == NO_ERROR) { 926 if (mActive) { 927 // callback thread or sync event hasn't changed 928 // FIXME this fails if we have a new AudioFlinger instance 929 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 930 } 931 } 932 if (result != NO_ERROR) { 933 ALOGW("restoreRecord_l() failed status %d", result); 934 mActive = false; 935 } 936 937 return result; 938} 939 940// ========================================================================= 941 942void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who) 943{ 944 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 945 if (audioRecord != 0) { 946 AutoMutex lock(audioRecord->mLock); 947 audioRecord->mProxy->binderDied(); 948 } 949} 950 951// ========================================================================= 952 953AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 954 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 955{ 956} 957 958AudioRecord::AudioRecordThread::~AudioRecordThread() 959{ 960} 961 962bool AudioRecord::AudioRecordThread::threadLoop() 963{ 964 { 965 AutoMutex _l(mMyLock); 966 if (mPaused) { 967 mMyCond.wait(mMyLock); 968 // caller will check for exitPending() 969 return true; 970 } 971 } 972 nsecs_t ns = mReceiver.processAudioBuffer(this); 973 switch (ns) { 974 case 0: 975 return true; 976 case NS_WHENEVER: 977 sleep(1); 978 return true; 979 case NS_INACTIVE: 980 pauseConditional(); 981 return true; 982 case NS_NEVER: 983 return false; 984 default: 985 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 986 struct timespec req; 987 req.tv_sec = ns / 1000000000LL; 988 req.tv_nsec = ns % 1000000000LL; 989 nanosleep(&req, NULL /*rem*/); 990 return true; 991 } 992} 993 994void AudioRecord::AudioRecordThread::requestExit() 995{ 996 // must be in this order to avoid a race condition 997 Thread::requestExit(); 998 resume(); 999} 1000 1001void AudioRecord::AudioRecordThread::pause() 1002{ 1003 AutoMutex _l(mMyLock); 1004 mPaused = true; 1005 mResumeLatch = false; 1006} 1007 1008void AudioRecord::AudioRecordThread::pauseConditional() 1009{ 1010 AutoMutex _l(mMyLock); 1011 if (mResumeLatch) { 1012 mResumeLatch = false; 1013 } else { 1014 mPaused = true; 1015 } 1016} 1017 1018void AudioRecord::AudioRecordThread::resume() 1019{ 1020 AutoMutex _l(mMyLock); 1021 if (mPaused) { 1022 mPaused = false; 1023 mResumeLatch = false; 1024 mMyCond.signal(); 1025 } else { 1026 mResumeLatch = true; 1027 } 1028} 1029 1030// ------------------------------------------------------------------------- 1031 1032}; // namespace android 1033