AudioRecord.cpp revision 3a0b6bd22aa32daa729b05c33896400807027eee
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 buffer.mFrameCount = 0; 600 buffer.mRaw = NULL; 601 buffer.mNonContig = 0; 602 break; 603 } 604 } 605 } 606 oldSequence = newSequence; 607 608 // Keep the extra references 609 proxy = mProxy; 610 iMem = mCblkMemory; 611 612 // Non-blocking if track is stopped 613 if (!mActive) { 614 requested = &ClientProxy::kNonBlocking; 615 } 616 617 } // end of lock scope 618 619 buffer.mFrameCount = audioBuffer->frameCount; 620 // FIXME starts the requested timeout and elapsed over from scratch 621 status = proxy->obtainBuffer(&buffer, requested, elapsed); 622 623 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 624 625 audioBuffer->frameCount = buffer.mFrameCount; 626 audioBuffer->size = buffer.mFrameCount * mFrameSize; 627 audioBuffer->raw = buffer.mRaw; 628 if (nonContig != NULL) { 629 *nonContig = buffer.mNonContig; 630 } 631 return status; 632} 633 634void AudioRecord::releaseBuffer(Buffer* audioBuffer) 635{ 636 // all TRANSFER_* are valid 637 638 size_t stepCount = audioBuffer->size / mFrameSize; 639 if (stepCount == 0) { 640 return; 641 } 642 643 Proxy::Buffer buffer; 644 buffer.mFrameCount = stepCount; 645 buffer.mRaw = audioBuffer->raw; 646 647 AutoMutex lock(mLock); 648 mInOverrun = false; 649 mProxy->releaseBuffer(&buffer); 650 651 // the server does not automatically disable recorder on overrun, so no need to restart 652} 653 654audio_io_handle_t AudioRecord::getInput() const 655{ 656 AutoMutex lock(mLock); 657 return mInput; 658} 659 660// ------------------------------------------------------------------------- 661 662ssize_t AudioRecord::read(void* buffer, size_t userSize) 663{ 664 if (mTransfer != TRANSFER_SYNC) { 665 return INVALID_OPERATION; 666 } 667 668 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 669 // sanity-check. user is most-likely passing an error code, and it would 670 // make the return value ambiguous (actualSize vs error). 671 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 672 return BAD_VALUE; 673 } 674 675 ssize_t read = 0; 676 Buffer audioBuffer; 677 678 while (userSize >= mFrameSize) { 679 audioBuffer.frameCount = userSize / mFrameSize; 680 681 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 682 if (err < 0) { 683 if (read > 0) { 684 break; 685 } 686 return ssize_t(err); 687 } 688 689 size_t bytesRead = audioBuffer.size; 690 memcpy(buffer, audioBuffer.i8, bytesRead); 691 buffer = ((char *) buffer) + bytesRead; 692 userSize -= bytesRead; 693 read += bytesRead; 694 695 releaseBuffer(&audioBuffer); 696 } 697 698 return read; 699} 700 701// ------------------------------------------------------------------------- 702 703nsecs_t AudioRecord::processAudioBuffer() 704{ 705 mLock.lock(); 706 if (mAwaitBoost) { 707 mAwaitBoost = false; 708 mLock.unlock(); 709 static const int32_t kMaxTries = 5; 710 int32_t tryCounter = kMaxTries; 711 uint32_t pollUs = 10000; 712 do { 713 int policy = sched_getscheduler(0); 714 if (policy == SCHED_FIFO || policy == SCHED_RR) { 715 break; 716 } 717 usleep(pollUs); 718 pollUs <<= 1; 719 } while (tryCounter-- > 0); 720 if (tryCounter < 0) { 721 ALOGE("did not receive expected priority boost on time"); 722 } 723 // Run again immediately 724 return 0; 725 } 726 727 // Can only reference mCblk while locked 728 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 729 730 // Check for track invalidation 731 if (flags & CBLK_INVALID) { 732 (void) restoreRecord_l("processAudioBuffer"); 733 mLock.unlock(); 734 // Run again immediately, but with a new IAudioRecord 735 return 0; 736 } 737 738 bool active = mActive; 739 740 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 741 bool newOverrun = false; 742 if (flags & CBLK_OVERRUN) { 743 if (!mInOverrun) { 744 mInOverrun = true; 745 newOverrun = true; 746 } 747 } 748 749 // Get current position of server 750 size_t position = mProxy->getPosition(); 751 752 // Manage marker callback 753 bool markerReached = false; 754 size_t markerPosition = mMarkerPosition; 755 // FIXME fails for wraparound, need 64 bits 756 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 757 mMarkerReached = markerReached = true; 758 } 759 760 // Determine the number of new position callback(s) that will be needed, while locked 761 size_t newPosCount = 0; 762 size_t newPosition = mNewPosition; 763 uint32_t updatePeriod = mUpdatePeriod; 764 // FIXME fails for wraparound, need 64 bits 765 if (updatePeriod > 0 && position >= newPosition) { 766 newPosCount = ((position - newPosition) / updatePeriod) + 1; 767 mNewPosition += updatePeriod * newPosCount; 768 } 769 770 // Cache other fields that will be needed soon 771 size_t notificationFrames = mNotificationFramesAct; 772 if (mRefreshRemaining) { 773 mRefreshRemaining = false; 774 mRemainingFrames = notificationFrames; 775 mRetryOnPartialBuffer = false; 776 } 777 size_t misalignment = mProxy->getMisalignment(); 778 uint32_t sequence = mSequence; 779 780 // These fields don't need to be cached, because they are assigned only by set(): 781 // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize 782 783 mLock.unlock(); 784 785 // perform callbacks while unlocked 786 if (newOverrun) { 787 mCbf(EVENT_OVERRUN, mUserData, NULL); 788 } 789 if (markerReached) { 790 mCbf(EVENT_MARKER, mUserData, &markerPosition); 791 } 792 while (newPosCount > 0) { 793 size_t temp = newPosition; 794 mCbf(EVENT_NEW_POS, mUserData, &temp); 795 newPosition += updatePeriod; 796 newPosCount--; 797 } 798 if (mObservedSequence != sequence) { 799 mObservedSequence = sequence; 800 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 801 } 802 803 // if inactive, then don't run me again until re-started 804 if (!active) { 805 return NS_INACTIVE; 806 } 807 808 // Compute the estimated time until the next timed event (position, markers) 809 uint32_t minFrames = ~0; 810 if (!markerReached && position < markerPosition) { 811 minFrames = markerPosition - position; 812 } 813 if (updatePeriod > 0 && updatePeriod < minFrames) { 814 minFrames = updatePeriod; 815 } 816 817 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 818 static const uint32_t kPoll = 0; 819 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 820 minFrames = kPoll * notificationFrames; 821 } 822 823 // Convert frame units to time units 824 nsecs_t ns = NS_WHENEVER; 825 if (minFrames != (uint32_t) ~0) { 826 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 827 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 828 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 829 } 830 831 // If not supplying data by EVENT_MORE_DATA, then we're done 832 if (mTransfer != TRANSFER_CALLBACK) { 833 return ns; 834 } 835 836 struct timespec timeout; 837 const struct timespec *requested = &ClientProxy::kForever; 838 if (ns != NS_WHENEVER) { 839 timeout.tv_sec = ns / 1000000000LL; 840 timeout.tv_nsec = ns % 1000000000LL; 841 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 842 requested = &timeout; 843 } 844 845 while (mRemainingFrames > 0) { 846 847 Buffer audioBuffer; 848 audioBuffer.frameCount = mRemainingFrames; 849 size_t nonContig; 850 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 851 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 852 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 853 requested = &ClientProxy::kNonBlocking; 854 size_t avail = audioBuffer.frameCount + nonContig; 855 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 856 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 857 if (err != NO_ERROR) { 858 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 859 break; 860 } 861 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 862 return NS_NEVER; 863 } 864 865 if (mRetryOnPartialBuffer) { 866 mRetryOnPartialBuffer = false; 867 if (avail < mRemainingFrames) { 868 int64_t myns = ((mRemainingFrames - avail) * 869 1100000000LL) / mSampleRate; 870 if (ns < 0 || myns < ns) { 871 ns = myns; 872 } 873 return ns; 874 } 875 } 876 877 size_t reqSize = audioBuffer.size; 878 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 879 size_t readSize = audioBuffer.size; 880 881 // Sanity check on returned size 882 if (ssize_t(readSize) < 0 || readSize > reqSize) { 883 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 884 reqSize, (int) readSize); 885 return NS_NEVER; 886 } 887 888 if (readSize == 0) { 889 // The callback is done consuming buffers 890 // Keep this thread going to handle timed events and 891 // still try to provide more data in intervals of WAIT_PERIOD_MS 892 // but don't just loop and block the CPU, so wait 893 return WAIT_PERIOD_MS * 1000000LL; 894 } 895 896 size_t releasedFrames = readSize / mFrameSize; 897 audioBuffer.frameCount = releasedFrames; 898 mRemainingFrames -= releasedFrames; 899 if (misalignment >= releasedFrames) { 900 misalignment -= releasedFrames; 901 } else { 902 misalignment = 0; 903 } 904 905 releaseBuffer(&audioBuffer); 906 907 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 908 // if callback doesn't like to accept the full chunk 909 if (readSize < reqSize) { 910 continue; 911 } 912 913 // There could be enough non-contiguous frames available to satisfy the remaining request 914 if (mRemainingFrames <= nonContig) { 915 continue; 916 } 917 918#if 0 919 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 920 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 921 // that total to a sum == notificationFrames. 922 if (0 < misalignment && misalignment <= mRemainingFrames) { 923 mRemainingFrames = misalignment; 924 return (mRemainingFrames * 1100000000LL) / mSampleRate; 925 } 926#endif 927 928 } 929 mRemainingFrames = notificationFrames; 930 mRetryOnPartialBuffer = true; 931 932 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 933 return 0; 934} 935 936status_t AudioRecord::restoreRecord_l(const char *from) 937{ 938 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 939 ++mSequence; 940 status_t result; 941 942 // if the new IAudioRecord is created, openRecord_l() will modify the 943 // following member variables: mAudioRecord, mCblkMemory and mCblk. 944 // It will also delete the strong references on previous IAudioRecord and IMemory 945 size_t position = mProxy->getPosition(); 946 mNewPosition = position + mUpdatePeriod; 947 result = openRecord_l(position); 948 if (result == NO_ERROR) { 949 if (mActive) { 950 // callback thread or sync event hasn't changed 951 // FIXME this fails if we have a new AudioFlinger instance 952 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 953 } 954 } 955 if (result != NO_ERROR) { 956 ALOGW("restoreRecord_l() failed status %d", result); 957 mActive = false; 958 } 959 960 return result; 961} 962 963// ========================================================================= 964 965void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 966{ 967 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 968 if (audioRecord != 0) { 969 AutoMutex lock(audioRecord->mLock); 970 audioRecord->mProxy->binderDied(); 971 } 972} 973 974// ========================================================================= 975 976AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 977 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 978 mIgnoreNextPausedInt(false) 979{ 980} 981 982AudioRecord::AudioRecordThread::~AudioRecordThread() 983{ 984} 985 986bool AudioRecord::AudioRecordThread::threadLoop() 987{ 988 { 989 AutoMutex _l(mMyLock); 990 if (mPaused) { 991 mMyCond.wait(mMyLock); 992 // caller will check for exitPending() 993 return true; 994 } 995 if (mIgnoreNextPausedInt) { 996 mIgnoreNextPausedInt = false; 997 mPausedInt = false; 998 } 999 if (mPausedInt) { 1000 if (mPausedNs > 0) { 1001 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1002 } else { 1003 mMyCond.wait(mMyLock); 1004 } 1005 mPausedInt = false; 1006 return true; 1007 } 1008 } 1009 nsecs_t ns = mReceiver.processAudioBuffer(); 1010 switch (ns) { 1011 case 0: 1012 return true; 1013 case NS_INACTIVE: 1014 pauseInternal(); 1015 return true; 1016 case NS_NEVER: 1017 return false; 1018 case NS_WHENEVER: 1019 // FIXME increase poll interval, or make event-driven 1020 ns = 1000000000LL; 1021 // fall through 1022 default: 1023 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1024 pauseInternal(ns); 1025 return true; 1026 } 1027} 1028 1029void AudioRecord::AudioRecordThread::requestExit() 1030{ 1031 // must be in this order to avoid a race condition 1032 Thread::requestExit(); 1033 resume(); 1034} 1035 1036void AudioRecord::AudioRecordThread::pause() 1037{ 1038 AutoMutex _l(mMyLock); 1039 mPaused = true; 1040} 1041 1042void AudioRecord::AudioRecordThread::resume() 1043{ 1044 AutoMutex _l(mMyLock); 1045 mIgnoreNextPausedInt = true; 1046 if (mPaused || mPausedInt) { 1047 mPaused = false; 1048 mPausedInt = false; 1049 mMyCond.signal(); 1050 } 1051} 1052 1053void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns) 1054{ 1055 AutoMutex _l(mMyLock); 1056 mPausedInt = true; 1057 mPausedNs = ns; 1058} 1059 1060// ------------------------------------------------------------------------- 1061 1062}; // namespace android 1063