AudioRecord.cpp revision 6d88aaf9cd810d96a4888dff8bd33d44cd01ccaa
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(AUDIO_SESSION_ALLOCATE), 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(AUDIO_SESSION_ALLOCATE), 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 mProxy->interrupt(); 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 mInputSource = inputSource; 180 181 if (sampleRate == 0) { 182 ALOGE("Invalid sample rate %u", sampleRate); 183 return BAD_VALUE; 184 } 185 mSampleRate = sampleRate; 186 187 // these below should probably come from the audioFlinger too... 188 if (format == AUDIO_FORMAT_DEFAULT) { 189 format = AUDIO_FORMAT_PCM_16_BIT; 190 } 191 192 // validate parameters 193 if (!audio_is_valid_format(format)) { 194 ALOGE("Invalid format %d", format); 195 return BAD_VALUE; 196 } 197 // Temporary restriction: AudioFlinger currently supports 16-bit PCM only 198 if (format != AUDIO_FORMAT_PCM_16_BIT) { 199 ALOGE("Format %d is not supported", format); 200 return BAD_VALUE; 201 } 202 mFormat = format; 203 204 if (!audio_is_input_channel(channelMask)) { 205 ALOGE("Invalid channel mask %#x", channelMask); 206 return BAD_VALUE; 207 } 208 mChannelMask = channelMask; 209 uint32_t channelCount = popcount(channelMask); 210 mChannelCount = channelCount; 211 212 // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t) 213 mFrameSize = channelCount * audio_bytes_per_sample(format); 214 215 // validate framecount 216 size_t minFrameCount = 0; 217 status_t status = AudioRecord::getMinFrameCount(&minFrameCount, 218 sampleRate, format, channelMask); 219 if (status != NO_ERROR) { 220 ALOGE("getMinFrameCount() failed; status %d", status); 221 return status; 222 } 223 ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount); 224 225 if (frameCount == 0) { 226 frameCount = minFrameCount; 227 } else if (frameCount < minFrameCount) { 228 ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount); 229 return BAD_VALUE; 230 } 231 mFrameCount = frameCount; 232 233 mNotificationFramesReq = notificationFrames; 234 mNotificationFramesAct = 0; 235 236 if (sessionId == AUDIO_SESSION_ALLOCATE) { 237 mSessionId = AudioSystem::newAudioSessionId(); 238 } else { 239 mSessionId = sessionId; 240 } 241 ALOGV("set(): mSessionId %d", mSessionId); 242 243 mFlags = flags; 244 245 // create the IAudioRecord 246 status = openRecord_l(0 /*epoch*/); 247 if (status) { 248 return status; 249 } 250 251 if (cbf != NULL) { 252 mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava); 253 mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO); 254 } 255 256 mStatus = NO_ERROR; 257 258 // Update buffer size in case it has been limited by AudioFlinger during track creation 259 mFrameCount = mCblk->frameCount_; 260 261 mActive = false; 262 mCbf = cbf; 263 mRefreshRemaining = true; 264 mUserData = user; 265 // TODO: add audio hardware input latency here 266 mLatency = (1000*mFrameCount) / sampleRate; 267 mMarkerPosition = 0; 268 mMarkerReached = false; 269 mNewPosition = 0; 270 mUpdatePeriod = 0; 271 AudioSystem::acquireAudioSessionId(mSessionId); 272 mSequence = 1; 273 mObservedSequence = mSequence; 274 mInOverrun = false; 275 276 return NO_ERROR; 277} 278 279// ------------------------------------------------------------------------- 280 281status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) 282{ 283 ALOGV("start, sync event %d trigger session %d", event, triggerSession); 284 285 AutoMutex lock(mLock); 286 if (mActive) { 287 return NO_ERROR; 288 } 289 290 // reset current position as seen by client to 0 291 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 292 // force refresh of remaining frames by processAudioBuffer() as last 293 // read before stop could be partial. 294 mRefreshRemaining = true; 295 296 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 297 int32_t flags = android_atomic_acquire_load(&mCblk->mFlags); 298 299 status_t status = NO_ERROR; 300 if (!(flags & CBLK_INVALID)) { 301 ALOGV("mAudioRecord->start()"); 302 status = mAudioRecord->start(event, triggerSession); 303 if (status == DEAD_OBJECT) { 304 flags |= CBLK_INVALID; 305 } 306 } 307 if (flags & CBLK_INVALID) { 308 status = restoreRecord_l("start"); 309 } 310 311 if (status != NO_ERROR) { 312 ALOGE("start() status %d", status); 313 } else { 314 mActive = true; 315 sp<AudioRecordThread> t = mAudioRecordThread; 316 if (t != 0) { 317 t->resume(); 318 } else { 319 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 320 get_sched_policy(0, &mPreviousSchedulingGroup); 321 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 322 } 323 } 324 325 return status; 326} 327 328void AudioRecord::stop() 329{ 330 AutoMutex lock(mLock); 331 if (!mActive) { 332 return; 333 } 334 335 mActive = false; 336 mProxy->interrupt(); 337 mAudioRecord->stop(); 338 // the record head position will reset to 0, so if a marker is set, we need 339 // to activate it again 340 mMarkerReached = false; 341 sp<AudioRecordThread> t = mAudioRecordThread; 342 if (t != 0) { 343 t->pause(); 344 } else { 345 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 346 set_sched_policy(0, mPreviousSchedulingGroup); 347 } 348} 349 350bool AudioRecord::stopped() const 351{ 352 AutoMutex lock(mLock); 353 return !mActive; 354} 355 356status_t AudioRecord::setMarkerPosition(uint32_t marker) 357{ 358 // The only purpose of setting marker position is to get a callback 359 if (mCbf == NULL) { 360 return INVALID_OPERATION; 361 } 362 363 AutoMutex lock(mLock); 364 mMarkerPosition = marker; 365 mMarkerReached = false; 366 367 return NO_ERROR; 368} 369 370status_t AudioRecord::getMarkerPosition(uint32_t *marker) const 371{ 372 if (marker == NULL) { 373 return BAD_VALUE; 374 } 375 376 AutoMutex lock(mLock); 377 *marker = mMarkerPosition; 378 379 return NO_ERROR; 380} 381 382status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) 383{ 384 // The only purpose of setting position update period is to get a callback 385 if (mCbf == NULL) { 386 return INVALID_OPERATION; 387 } 388 389 AutoMutex lock(mLock); 390 mNewPosition = mProxy->getPosition() + updatePeriod; 391 mUpdatePeriod = updatePeriod; 392 393 return NO_ERROR; 394} 395 396status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const 397{ 398 if (updatePeriod == NULL) { 399 return BAD_VALUE; 400 } 401 402 AutoMutex lock(mLock); 403 *updatePeriod = mUpdatePeriod; 404 405 return NO_ERROR; 406} 407 408status_t AudioRecord::getPosition(uint32_t *position) const 409{ 410 if (position == NULL) { 411 return BAD_VALUE; 412 } 413 414 AutoMutex lock(mLock); 415 *position = mProxy->getPosition(); 416 417 return NO_ERROR; 418} 419 420uint32_t AudioRecord::getInputFramesLost() const 421{ 422 // no need to check mActive, because if inactive this will return 0, which is what we want 423 return AudioSystem::getInputFramesLost(getInput()); 424} 425 426// ------------------------------------------------------------------------- 427 428// must be called with mLock held 429status_t AudioRecord::openRecord_l(size_t epoch) 430{ 431 status_t status; 432 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 433 if (audioFlinger == 0) { 434 ALOGE("Could not get audioflinger"); 435 return NO_INIT; 436 } 437 438 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 439 pid_t tid = -1; 440 441 // Client can only express a preference for FAST. Server will perform additional tests. 442 // The only supported use case for FAST is callback transfer mode. 443 if (mFlags & AUDIO_INPUT_FLAG_FAST) { 444 if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) { 445 ALOGW("AUDIO_INPUT_FLAG_FAST denied by client"); 446 // once denied, do not request again if IAudioRecord is re-created 447 mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); 448 } else { 449 trackFlags |= IAudioFlinger::TRACK_FAST; 450 tid = mAudioRecordThread->getTid(); 451 } 452 } 453 454 mNotificationFramesAct = mNotificationFramesReq; 455 456 if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) { 457 // Make sure that application is notified with sufficient margin before overrun 458 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) { 459 mNotificationFramesAct = mFrameCount/2; 460 } 461 } 462 463 audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat, 464 mChannelMask, mSessionId); 465 if (input == 0) { 466 ALOGE("Could not get audio input for record source %d", mInputSource); 467 return BAD_VALUE; 468 } 469 470 int originalSessionId = mSessionId; 471 sp<IAudioRecord> record = audioFlinger->openRecord(input, 472 mSampleRate, mFormat, 473 mChannelMask, 474 mFrameCount, 475 &trackFlags, 476 tid, 477 &mSessionId, 478 &status); 479 ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, 480 "session ID changed from %d to %d", originalSessionId, mSessionId); 481 482 if (record == 0 || status != NO_ERROR) { 483 ALOGE("AudioFlinger could not create record track, status: %d", status); 484 AudioSystem::releaseInput(input); 485 return status; 486 } 487 sp<IMemory> iMem = record->getCblk(); 488 if (iMem == 0) { 489 ALOGE("Could not get control block"); 490 return NO_INIT; 491 } 492 void *iMemPointer = iMem->pointer(); 493 if (iMemPointer == NULL) { 494 ALOGE("Could not get control block pointer"); 495 return NO_INIT; 496 } 497 if (mAudioRecord != 0) { 498 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 499 mDeathNotifier.clear(); 500 } 501 mInput = input; 502 mAudioRecord = record; 503 mCblkMemory = iMem; 504 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 505 mCblk = cblk; 506 // FIXME missing fast track frameCount logic 507 mAwaitBoost = false; 508 if (mFlags & AUDIO_INPUT_FLAG_FAST) { 509 if (trackFlags & IAudioFlinger::TRACK_FAST) { 510 ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", mFrameCount); 511 mAwaitBoost = true; 512 // double-buffering is not required for fast tracks, due to tighter scheduling 513 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount) { 514 mNotificationFramesAct = mFrameCount; 515 } 516 } else { 517 ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", mFrameCount); 518 // once denied, do not request again if IAudioRecord is re-created 519 mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); 520 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) { 521 mNotificationFramesAct = mFrameCount/2; 522 } 523 } 524 } 525 526 // starting address of buffers in shared memory 527 void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); 528 529 // update proxy 530 mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize); 531 mProxy->setEpoch(epoch); 532 mProxy->setMinimum(mNotificationFramesAct); 533 534 mDeathNotifier = new DeathNotifier(this); 535 mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); 536 537 return NO_ERROR; 538} 539 540status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 541{ 542 if (audioBuffer == NULL) { 543 return BAD_VALUE; 544 } 545 if (mTransfer != TRANSFER_OBTAIN) { 546 audioBuffer->frameCount = 0; 547 audioBuffer->size = 0; 548 audioBuffer->raw = NULL; 549 return INVALID_OPERATION; 550 } 551 552 const struct timespec *requested; 553 if (waitCount == -1) { 554 requested = &ClientProxy::kForever; 555 } else if (waitCount == 0) { 556 requested = &ClientProxy::kNonBlocking; 557 } else if (waitCount > 0) { 558 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 559 struct timespec timeout; 560 timeout.tv_sec = ms / 1000; 561 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 562 requested = &timeout; 563 } else { 564 ALOGE("%s invalid waitCount %d", __func__, waitCount); 565 requested = NULL; 566 } 567 return obtainBuffer(audioBuffer, requested); 568} 569 570status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 571 struct timespec *elapsed, size_t *nonContig) 572{ 573 // previous and new IAudioRecord sequence numbers are used to detect track re-creation 574 uint32_t oldSequence = 0; 575 uint32_t newSequence; 576 577 Proxy::Buffer buffer; 578 status_t status = NO_ERROR; 579 580 static const int32_t kMaxTries = 5; 581 int32_t tryCounter = kMaxTries; 582 583 do { 584 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 585 // keep them from going away if another thread re-creates the track during obtainBuffer() 586 sp<AudioRecordClientProxy> proxy; 587 sp<IMemory> iMem; 588 { 589 // start of lock scope 590 AutoMutex lock(mLock); 591 592 newSequence = mSequence; 593 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 594 if (status == DEAD_OBJECT) { 595 // re-create track, unless someone else has already done so 596 if (newSequence == oldSequence) { 597 status = restoreRecord_l("obtainBuffer"); 598 if (status != NO_ERROR) { 599 break; 600 } 601 } 602 } 603 oldSequence = newSequence; 604 605 // Keep the extra references 606 proxy = mProxy; 607 iMem = mCblkMemory; 608 609 // Non-blocking if track is stopped 610 if (!mActive) { 611 requested = &ClientProxy::kNonBlocking; 612 } 613 614 } // end of lock scope 615 616 buffer.mFrameCount = audioBuffer->frameCount; 617 // FIXME starts the requested timeout and elapsed over from scratch 618 status = proxy->obtainBuffer(&buffer, requested, elapsed); 619 620 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 621 622 audioBuffer->frameCount = buffer.mFrameCount; 623 audioBuffer->size = buffer.mFrameCount * mFrameSize; 624 audioBuffer->raw = buffer.mRaw; 625 if (nonContig != NULL) { 626 *nonContig = buffer.mNonContig; 627 } 628 return status; 629} 630 631void AudioRecord::releaseBuffer(Buffer* audioBuffer) 632{ 633 // all TRANSFER_* are valid 634 635 size_t stepCount = audioBuffer->size / mFrameSize; 636 if (stepCount == 0) { 637 return; 638 } 639 640 Proxy::Buffer buffer; 641 buffer.mFrameCount = stepCount; 642 buffer.mRaw = audioBuffer->raw; 643 644 AutoMutex lock(mLock); 645 mInOverrun = false; 646 mProxy->releaseBuffer(&buffer); 647 648 // the server does not automatically disable recorder on overrun, so no need to restart 649} 650 651audio_io_handle_t AudioRecord::getInput() const 652{ 653 AutoMutex lock(mLock); 654 return mInput; 655} 656 657// ------------------------------------------------------------------------- 658 659ssize_t AudioRecord::read(void* buffer, size_t userSize) 660{ 661 if (mTransfer != TRANSFER_SYNC) { 662 return INVALID_OPERATION; 663 } 664 665 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 666 // sanity-check. user is most-likely passing an error code, and it would 667 // make the return value ambiguous (actualSize vs error). 668 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 669 return BAD_VALUE; 670 } 671 672 ssize_t read = 0; 673 Buffer audioBuffer; 674 675 while (userSize >= mFrameSize) { 676 audioBuffer.frameCount = userSize / mFrameSize; 677 678 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 679 if (err < 0) { 680 if (read > 0) { 681 break; 682 } 683 return ssize_t(err); 684 } 685 686 size_t bytesRead = audioBuffer.size; 687 memcpy(buffer, audioBuffer.i8, bytesRead); 688 buffer = ((char *) buffer) + bytesRead; 689 userSize -= bytesRead; 690 read += bytesRead; 691 692 releaseBuffer(&audioBuffer); 693 } 694 695 return read; 696} 697 698// ------------------------------------------------------------------------- 699 700nsecs_t AudioRecord::processAudioBuffer() 701{ 702 mLock.lock(); 703 if (mAwaitBoost) { 704 mAwaitBoost = false; 705 mLock.unlock(); 706 static const int32_t kMaxTries = 5; 707 int32_t tryCounter = kMaxTries; 708 uint32_t pollUs = 10000; 709 do { 710 int policy = sched_getscheduler(0); 711 if (policy == SCHED_FIFO || policy == SCHED_RR) { 712 break; 713 } 714 usleep(pollUs); 715 pollUs <<= 1; 716 } while (tryCounter-- > 0); 717 if (tryCounter < 0) { 718 ALOGE("did not receive expected priority boost on time"); 719 } 720 // Run again immediately 721 return 0; 722 } 723 724 // Can only reference mCblk while locked 725 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 726 727 // Check for track invalidation 728 if (flags & CBLK_INVALID) { 729 (void) restoreRecord_l("processAudioBuffer"); 730 mLock.unlock(); 731 // Run again immediately, but with a new IAudioRecord 732 return 0; 733 } 734 735 bool active = mActive; 736 737 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 738 bool newOverrun = false; 739 if (flags & CBLK_OVERRUN) { 740 if (!mInOverrun) { 741 mInOverrun = true; 742 newOverrun = true; 743 } 744 } 745 746 // Get current position of server 747 size_t position = mProxy->getPosition(); 748 749 // Manage marker callback 750 bool markerReached = false; 751 size_t markerPosition = mMarkerPosition; 752 // FIXME fails for wraparound, need 64 bits 753 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 754 mMarkerReached = markerReached = true; 755 } 756 757 // Determine the number of new position callback(s) that will be needed, while locked 758 size_t newPosCount = 0; 759 size_t newPosition = mNewPosition; 760 uint32_t updatePeriod = mUpdatePeriod; 761 // FIXME fails for wraparound, need 64 bits 762 if (updatePeriod > 0 && position >= newPosition) { 763 newPosCount = ((position - newPosition) / updatePeriod) + 1; 764 mNewPosition += updatePeriod * newPosCount; 765 } 766 767 // Cache other fields that will be needed soon 768 size_t notificationFrames = mNotificationFramesAct; 769 if (mRefreshRemaining) { 770 mRefreshRemaining = false; 771 mRemainingFrames = notificationFrames; 772 mRetryOnPartialBuffer = false; 773 } 774 size_t misalignment = mProxy->getMisalignment(); 775 int32_t sequence = mSequence; 776 777 // These fields don't need to be cached, because they are assigned only by set(): 778 // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize 779 780 mLock.unlock(); 781 782 // perform callbacks while unlocked 783 if (newOverrun) { 784 mCbf(EVENT_OVERRUN, mUserData, NULL); 785 } 786 if (markerReached) { 787 mCbf(EVENT_MARKER, mUserData, &markerPosition); 788 } 789 while (newPosCount > 0) { 790 size_t temp = newPosition; 791 mCbf(EVENT_NEW_POS, mUserData, &temp); 792 newPosition += updatePeriod; 793 newPosCount--; 794 } 795 if (mObservedSequence != sequence) { 796 mObservedSequence = sequence; 797 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 798 } 799 800 // if inactive, then don't run me again until re-started 801 if (!active) { 802 return NS_INACTIVE; 803 } 804 805 // Compute the estimated time until the next timed event (position, markers) 806 uint32_t minFrames = ~0; 807 if (!markerReached && position < markerPosition) { 808 minFrames = markerPosition - position; 809 } 810 if (updatePeriod > 0 && updatePeriod < minFrames) { 811 minFrames = updatePeriod; 812 } 813 814 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 815 static const uint32_t kPoll = 0; 816 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 817 minFrames = kPoll * notificationFrames; 818 } 819 820 // Convert frame units to time units 821 nsecs_t ns = NS_WHENEVER; 822 if (minFrames != (uint32_t) ~0) { 823 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 824 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 825 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 826 } 827 828 // If not supplying data by EVENT_MORE_DATA, then we're done 829 if (mTransfer != TRANSFER_CALLBACK) { 830 return ns; 831 } 832 833 struct timespec timeout; 834 const struct timespec *requested = &ClientProxy::kForever; 835 if (ns != NS_WHENEVER) { 836 timeout.tv_sec = ns / 1000000000LL; 837 timeout.tv_nsec = ns % 1000000000LL; 838 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 839 requested = &timeout; 840 } 841 842 while (mRemainingFrames > 0) { 843 844 Buffer audioBuffer; 845 audioBuffer.frameCount = mRemainingFrames; 846 size_t nonContig; 847 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 848 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 849 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 850 requested = &ClientProxy::kNonBlocking; 851 size_t avail = audioBuffer.frameCount + nonContig; 852 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 853 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 854 if (err != NO_ERROR) { 855 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 856 break; 857 } 858 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 859 return NS_NEVER; 860 } 861 862 if (mRetryOnPartialBuffer) { 863 mRetryOnPartialBuffer = false; 864 if (avail < mRemainingFrames) { 865 int64_t myns = ((mRemainingFrames - avail) * 866 1100000000LL) / mSampleRate; 867 if (ns < 0 || myns < ns) { 868 ns = myns; 869 } 870 return ns; 871 } 872 } 873 874 size_t reqSize = audioBuffer.size; 875 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 876 size_t readSize = audioBuffer.size; 877 878 // Sanity check on returned size 879 if (ssize_t(readSize) < 0 || readSize > reqSize) { 880 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 881 reqSize, (int) readSize); 882 return NS_NEVER; 883 } 884 885 if (readSize == 0) { 886 // The callback is done consuming buffers 887 // Keep this thread going to handle timed events and 888 // still try to provide more data in intervals of WAIT_PERIOD_MS 889 // but don't just loop and block the CPU, so wait 890 return WAIT_PERIOD_MS * 1000000LL; 891 } 892 893 size_t releasedFrames = readSize / mFrameSize; 894 audioBuffer.frameCount = releasedFrames; 895 mRemainingFrames -= releasedFrames; 896 if (misalignment >= releasedFrames) { 897 misalignment -= releasedFrames; 898 } else { 899 misalignment = 0; 900 } 901 902 releaseBuffer(&audioBuffer); 903 904 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 905 // if callback doesn't like to accept the full chunk 906 if (readSize < reqSize) { 907 continue; 908 } 909 910 // There could be enough non-contiguous frames available to satisfy the remaining request 911 if (mRemainingFrames <= nonContig) { 912 continue; 913 } 914 915#if 0 916 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 917 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 918 // that total to a sum == notificationFrames. 919 if (0 < misalignment && misalignment <= mRemainingFrames) { 920 mRemainingFrames = misalignment; 921 return (mRemainingFrames * 1100000000LL) / mSampleRate; 922 } 923#endif 924 925 } 926 mRemainingFrames = notificationFrames; 927 mRetryOnPartialBuffer = true; 928 929 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 930 return 0; 931} 932 933status_t AudioRecord::restoreRecord_l(const char *from) 934{ 935 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 936 ++mSequence; 937 status_t result; 938 939 // if the new IAudioRecord is created, openRecord_l() will modify the 940 // following member variables: mAudioRecord, mCblkMemory and mCblk. 941 // It will also delete the strong references on previous IAudioRecord and IMemory 942 size_t position = mProxy->getPosition(); 943 mNewPosition = position + mUpdatePeriod; 944 result = openRecord_l(position); 945 if (result == NO_ERROR) { 946 if (mActive) { 947 // callback thread or sync event hasn't changed 948 // FIXME this fails if we have a new AudioFlinger instance 949 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 950 } 951 } 952 if (result != NO_ERROR) { 953 ALOGW("restoreRecord_l() failed status %d", result); 954 mActive = false; 955 } 956 957 return result; 958} 959 960// ========================================================================= 961 962void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 963{ 964 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 965 if (audioRecord != 0) { 966 AutoMutex lock(audioRecord->mLock); 967 audioRecord->mProxy->binderDied(); 968 } 969} 970 971// ========================================================================= 972 973AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 974 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 975 mIgnoreNextPausedInt(false) 976{ 977} 978 979AudioRecord::AudioRecordThread::~AudioRecordThread() 980{ 981} 982 983bool AudioRecord::AudioRecordThread::threadLoop() 984{ 985 { 986 AutoMutex _l(mMyLock); 987 if (mPaused) { 988 mMyCond.wait(mMyLock); 989 // caller will check for exitPending() 990 return true; 991 } 992 if (mIgnoreNextPausedInt) { 993 mIgnoreNextPausedInt = false; 994 mPausedInt = false; 995 } 996 if (mPausedInt) { 997 if (mPausedNs > 0) { 998 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 999 } else { 1000 mMyCond.wait(mMyLock); 1001 } 1002 mPausedInt = false; 1003 return true; 1004 } 1005 } 1006 nsecs_t ns = mReceiver.processAudioBuffer(); 1007 switch (ns) { 1008 case 0: 1009 return true; 1010 case NS_INACTIVE: 1011 pauseInternal(); 1012 return true; 1013 case NS_NEVER: 1014 return false; 1015 case NS_WHENEVER: 1016 // FIXME increase poll interval, or make event-driven 1017 ns = 1000000000LL; 1018 // fall through 1019 default: 1020 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1021 pauseInternal(ns); 1022 return true; 1023 } 1024} 1025 1026void AudioRecord::AudioRecordThread::requestExit() 1027{ 1028 // must be in this order to avoid a race condition 1029 Thread::requestExit(); 1030 resume(); 1031} 1032 1033void AudioRecord::AudioRecordThread::pause() 1034{ 1035 AutoMutex _l(mMyLock); 1036 mPaused = true; 1037} 1038 1039void AudioRecord::AudioRecordThread::resume() 1040{ 1041 AutoMutex _l(mMyLock); 1042 mIgnoreNextPausedInt = true; 1043 if (mPaused || mPausedInt) { 1044 mPaused = false; 1045 mPausedInt = false; 1046 mMyCond.signal(); 1047 } 1048} 1049 1050void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns) 1051{ 1052 AutoMutex _l(mMyLock); 1053 mPausedInt = true; 1054 mPausedNs = ns; 1055} 1056 1057// ------------------------------------------------------------------------- 1058 1059}; // namespace android 1060