AudioRecord.cpp revision f0002d142e6d24c5438600b2c259679de710f8ac
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 mActive = false; 259 mCbf = cbf; 260 mRefreshRemaining = true; 261 mUserData = user; 262 // TODO: add audio hardware input latency here 263 mLatency = (1000*mFrameCount) / sampleRate; 264 mMarkerPosition = 0; 265 mMarkerReached = false; 266 mNewPosition = 0; 267 mUpdatePeriod = 0; 268 AudioSystem::acquireAudioSessionId(mSessionId); 269 mSequence = 1; 270 mObservedSequence = mSequence; 271 mInOverrun = false; 272 273 return NO_ERROR; 274} 275 276// ------------------------------------------------------------------------- 277 278status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) 279{ 280 ALOGV("start, sync event %d trigger session %d", event, triggerSession); 281 282 AutoMutex lock(mLock); 283 if (mActive) { 284 return NO_ERROR; 285 } 286 287 // reset current position as seen by client to 0 288 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 289 // force refresh of remaining frames by processAudioBuffer() as last 290 // read before stop could be partial. 291 mRefreshRemaining = true; 292 293 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 294 int32_t flags = android_atomic_acquire_load(&mCblk->mFlags); 295 296 status_t status = NO_ERROR; 297 if (!(flags & CBLK_INVALID)) { 298 ALOGV("mAudioRecord->start()"); 299 status = mAudioRecord->start(event, triggerSession); 300 if (status == DEAD_OBJECT) { 301 flags |= CBLK_INVALID; 302 } 303 } 304 if (flags & CBLK_INVALID) { 305 status = restoreRecord_l("start"); 306 } 307 308 if (status != NO_ERROR) { 309 ALOGE("start() status %d", status); 310 } else { 311 mActive = true; 312 sp<AudioRecordThread> t = mAudioRecordThread; 313 if (t != 0) { 314 t->resume(); 315 } else { 316 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 317 get_sched_policy(0, &mPreviousSchedulingGroup); 318 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 319 } 320 } 321 322 return status; 323} 324 325void AudioRecord::stop() 326{ 327 AutoMutex lock(mLock); 328 if (!mActive) { 329 return; 330 } 331 332 mActive = false; 333 mProxy->interrupt(); 334 mAudioRecord->stop(); 335 // the record head position will reset to 0, so if a marker is set, we need 336 // to activate it again 337 mMarkerReached = false; 338 sp<AudioRecordThread> t = mAudioRecordThread; 339 if (t != 0) { 340 t->pause(); 341 } else { 342 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 343 set_sched_policy(0, mPreviousSchedulingGroup); 344 } 345} 346 347bool AudioRecord::stopped() const 348{ 349 AutoMutex lock(mLock); 350 return !mActive; 351} 352 353status_t AudioRecord::setMarkerPosition(uint32_t marker) 354{ 355 // The only purpose of setting marker position is to get a callback 356 if (mCbf == NULL) { 357 return INVALID_OPERATION; 358 } 359 360 AutoMutex lock(mLock); 361 mMarkerPosition = marker; 362 mMarkerReached = false; 363 364 return NO_ERROR; 365} 366 367status_t AudioRecord::getMarkerPosition(uint32_t *marker) const 368{ 369 if (marker == NULL) { 370 return BAD_VALUE; 371 } 372 373 AutoMutex lock(mLock); 374 *marker = mMarkerPosition; 375 376 return NO_ERROR; 377} 378 379status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) 380{ 381 // The only purpose of setting position update period is to get a callback 382 if (mCbf == NULL) { 383 return INVALID_OPERATION; 384 } 385 386 AutoMutex lock(mLock); 387 mNewPosition = mProxy->getPosition() + updatePeriod; 388 mUpdatePeriod = updatePeriod; 389 390 return NO_ERROR; 391} 392 393status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const 394{ 395 if (updatePeriod == NULL) { 396 return BAD_VALUE; 397 } 398 399 AutoMutex lock(mLock); 400 *updatePeriod = mUpdatePeriod; 401 402 return NO_ERROR; 403} 404 405status_t AudioRecord::getPosition(uint32_t *position) const 406{ 407 if (position == NULL) { 408 return BAD_VALUE; 409 } 410 411 AutoMutex lock(mLock); 412 *position = mProxy->getPosition(); 413 414 return NO_ERROR; 415} 416 417uint32_t AudioRecord::getInputFramesLost() const 418{ 419 // no need to check mActive, because if inactive this will return 0, which is what we want 420 return AudioSystem::getInputFramesLost(getInput()); 421} 422 423// ------------------------------------------------------------------------- 424 425// must be called with mLock held 426status_t AudioRecord::openRecord_l(size_t epoch) 427{ 428 status_t status; 429 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 430 if (audioFlinger == 0) { 431 ALOGE("Could not get audioflinger"); 432 return NO_INIT; 433 } 434 435 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 436 pid_t tid = -1; 437 438 // Client can only express a preference for FAST. Server will perform additional tests. 439 // The only supported use case for FAST is callback transfer mode. 440 if (mFlags & AUDIO_INPUT_FLAG_FAST) { 441 if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) { 442 ALOGW("AUDIO_INPUT_FLAG_FAST denied by client"); 443 // once denied, do not request again if IAudioRecord is re-created 444 mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); 445 } else { 446 trackFlags |= IAudioFlinger::TRACK_FAST; 447 tid = mAudioRecordThread->getTid(); 448 } 449 } 450 451 mNotificationFramesAct = mNotificationFramesReq; 452 453 if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) { 454 // Make sure that application is notified with sufficient margin before overrun 455 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) { 456 mNotificationFramesAct = mFrameCount/2; 457 } 458 } 459 460 audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat, 461 mChannelMask, mSessionId); 462 if (input == 0) { 463 ALOGE("Could not get audio input for record source %d", mInputSource); 464 return BAD_VALUE; 465 } 466 467 size_t temp = mFrameCount; // temp may be replaced by a revised value of frameCount, 468 // but we will still need the original value also 469 int originalSessionId = mSessionId; 470 sp<IAudioRecord> record = audioFlinger->openRecord(input, 471 mSampleRate, mFormat, 472 mChannelMask, 473 &temp, 474 &trackFlags, 475 tid, 476 &mSessionId, 477 &status); 478 ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, 479 "session ID changed from %d to %d", originalSessionId, mSessionId); 480 481 if (record == 0 || status != NO_ERROR) { 482 ALOGE("AudioFlinger could not create record track, status: %d", status); 483 AudioSystem::releaseInput(input); 484 return status; 485 } 486 sp<IMemory> iMem = record->getCblk(); 487 if (iMem == 0) { 488 ALOGE("Could not get control block"); 489 return NO_INIT; 490 } 491 void *iMemPointer = iMem->pointer(); 492 if (iMemPointer == NULL) { 493 ALOGE("Could not get control block pointer"); 494 return NO_INIT; 495 } 496 if (mAudioRecord != 0) { 497 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 498 mDeathNotifier.clear(); 499 } 500 mInput = input; 501 mAudioRecord = record; 502 mCblkMemory = iMem; 503 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 504 mCblk = cblk; 505 // note that temp is the (possibly revised) value of mFrameCount 506 if (temp < mFrameCount || (mFrameCount == 0 && temp == 0)) { 507 ALOGW("Requested frameCount %u but received frameCount %u", mFrameCount, temp); 508 } 509 mFrameCount = temp; 510 511 // FIXME missing fast track frameCount logic 512 mAwaitBoost = false; 513 if (mFlags & AUDIO_INPUT_FLAG_FAST) { 514 if (trackFlags & IAudioFlinger::TRACK_FAST) { 515 ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", mFrameCount); 516 mAwaitBoost = true; 517 // double-buffering is not required for fast tracks, due to tighter scheduling 518 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount) { 519 mNotificationFramesAct = mFrameCount; 520 } 521 } else { 522 ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", mFrameCount); 523 // once denied, do not request again if IAudioRecord is re-created 524 mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); 525 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) { 526 mNotificationFramesAct = mFrameCount/2; 527 } 528 } 529 } 530 531 // starting address of buffers in shared memory 532 void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); 533 534 // update proxy 535 mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize); 536 mProxy->setEpoch(epoch); 537 mProxy->setMinimum(mNotificationFramesAct); 538 539 mDeathNotifier = new DeathNotifier(this); 540 mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); 541 542 return NO_ERROR; 543} 544 545status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 546{ 547 if (audioBuffer == NULL) { 548 return BAD_VALUE; 549 } 550 if (mTransfer != TRANSFER_OBTAIN) { 551 audioBuffer->frameCount = 0; 552 audioBuffer->size = 0; 553 audioBuffer->raw = NULL; 554 return INVALID_OPERATION; 555 } 556 557 const struct timespec *requested; 558 if (waitCount == -1) { 559 requested = &ClientProxy::kForever; 560 } else if (waitCount == 0) { 561 requested = &ClientProxy::kNonBlocking; 562 } else if (waitCount > 0) { 563 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 564 struct timespec timeout; 565 timeout.tv_sec = ms / 1000; 566 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 567 requested = &timeout; 568 } else { 569 ALOGE("%s invalid waitCount %d", __func__, waitCount); 570 requested = NULL; 571 } 572 return obtainBuffer(audioBuffer, requested); 573} 574 575status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 576 struct timespec *elapsed, size_t *nonContig) 577{ 578 // previous and new IAudioRecord sequence numbers are used to detect track re-creation 579 uint32_t oldSequence = 0; 580 uint32_t newSequence; 581 582 Proxy::Buffer buffer; 583 status_t status = NO_ERROR; 584 585 static const int32_t kMaxTries = 5; 586 int32_t tryCounter = kMaxTries; 587 588 do { 589 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 590 // keep them from going away if another thread re-creates the track during obtainBuffer() 591 sp<AudioRecordClientProxy> proxy; 592 sp<IMemory> iMem; 593 { 594 // start of lock scope 595 AutoMutex lock(mLock); 596 597 newSequence = mSequence; 598 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 599 if (status == DEAD_OBJECT) { 600 // re-create track, unless someone else has already done so 601 if (newSequence == oldSequence) { 602 status = restoreRecord_l("obtainBuffer"); 603 if (status != NO_ERROR) { 604 buffer.mFrameCount = 0; 605 buffer.mRaw = NULL; 606 buffer.mNonContig = 0; 607 break; 608 } 609 } 610 } 611 oldSequence = newSequence; 612 613 // Keep the extra references 614 proxy = mProxy; 615 iMem = mCblkMemory; 616 617 // Non-blocking if track is stopped 618 if (!mActive) { 619 requested = &ClientProxy::kNonBlocking; 620 } 621 622 } // end of lock scope 623 624 buffer.mFrameCount = audioBuffer->frameCount; 625 // FIXME starts the requested timeout and elapsed over from scratch 626 status = proxy->obtainBuffer(&buffer, requested, elapsed); 627 628 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 629 630 audioBuffer->frameCount = buffer.mFrameCount; 631 audioBuffer->size = buffer.mFrameCount * mFrameSize; 632 audioBuffer->raw = buffer.mRaw; 633 if (nonContig != NULL) { 634 *nonContig = buffer.mNonContig; 635 } 636 return status; 637} 638 639void AudioRecord::releaseBuffer(Buffer* audioBuffer) 640{ 641 // all TRANSFER_* are valid 642 643 size_t stepCount = audioBuffer->size / mFrameSize; 644 if (stepCount == 0) { 645 return; 646 } 647 648 Proxy::Buffer buffer; 649 buffer.mFrameCount = stepCount; 650 buffer.mRaw = audioBuffer->raw; 651 652 AutoMutex lock(mLock); 653 mInOverrun = false; 654 mProxy->releaseBuffer(&buffer); 655 656 // the server does not automatically disable recorder on overrun, so no need to restart 657} 658 659audio_io_handle_t AudioRecord::getInput() const 660{ 661 AutoMutex lock(mLock); 662 return mInput; 663} 664 665// ------------------------------------------------------------------------- 666 667ssize_t AudioRecord::read(void* buffer, size_t userSize) 668{ 669 if (mTransfer != TRANSFER_SYNC) { 670 return INVALID_OPERATION; 671 } 672 673 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 674 // sanity-check. user is most-likely passing an error code, and it would 675 // make the return value ambiguous (actualSize vs error). 676 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 677 return BAD_VALUE; 678 } 679 680 ssize_t read = 0; 681 Buffer audioBuffer; 682 683 while (userSize >= mFrameSize) { 684 audioBuffer.frameCount = userSize / mFrameSize; 685 686 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 687 if (err < 0) { 688 if (read > 0) { 689 break; 690 } 691 return ssize_t(err); 692 } 693 694 size_t bytesRead = audioBuffer.size; 695 memcpy(buffer, audioBuffer.i8, bytesRead); 696 buffer = ((char *) buffer) + bytesRead; 697 userSize -= bytesRead; 698 read += bytesRead; 699 700 releaseBuffer(&audioBuffer); 701 } 702 703 return read; 704} 705 706// ------------------------------------------------------------------------- 707 708nsecs_t AudioRecord::processAudioBuffer() 709{ 710 mLock.lock(); 711 if (mAwaitBoost) { 712 mAwaitBoost = false; 713 mLock.unlock(); 714 static const int32_t kMaxTries = 5; 715 int32_t tryCounter = kMaxTries; 716 uint32_t pollUs = 10000; 717 do { 718 int policy = sched_getscheduler(0); 719 if (policy == SCHED_FIFO || policy == SCHED_RR) { 720 break; 721 } 722 usleep(pollUs); 723 pollUs <<= 1; 724 } while (tryCounter-- > 0); 725 if (tryCounter < 0) { 726 ALOGE("did not receive expected priority boost on time"); 727 } 728 // Run again immediately 729 return 0; 730 } 731 732 // Can only reference mCblk while locked 733 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 734 735 // Check for track invalidation 736 if (flags & CBLK_INVALID) { 737 (void) restoreRecord_l("processAudioBuffer"); 738 mLock.unlock(); 739 // Run again immediately, but with a new IAudioRecord 740 return 0; 741 } 742 743 bool active = mActive; 744 745 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 746 bool newOverrun = false; 747 if (flags & CBLK_OVERRUN) { 748 if (!mInOverrun) { 749 mInOverrun = true; 750 newOverrun = true; 751 } 752 } 753 754 // Get current position of server 755 size_t position = mProxy->getPosition(); 756 757 // Manage marker callback 758 bool markerReached = false; 759 size_t markerPosition = mMarkerPosition; 760 // FIXME fails for wraparound, need 64 bits 761 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 762 mMarkerReached = markerReached = true; 763 } 764 765 // Determine the number of new position callback(s) that will be needed, while locked 766 size_t newPosCount = 0; 767 size_t newPosition = mNewPosition; 768 uint32_t updatePeriod = mUpdatePeriod; 769 // FIXME fails for wraparound, need 64 bits 770 if (updatePeriod > 0 && position >= newPosition) { 771 newPosCount = ((position - newPosition) / updatePeriod) + 1; 772 mNewPosition += updatePeriod * newPosCount; 773 } 774 775 // Cache other fields that will be needed soon 776 size_t notificationFrames = mNotificationFramesAct; 777 if (mRefreshRemaining) { 778 mRefreshRemaining = false; 779 mRemainingFrames = notificationFrames; 780 mRetryOnPartialBuffer = false; 781 } 782 size_t misalignment = mProxy->getMisalignment(); 783 uint32_t sequence = mSequence; 784 785 // These fields don't need to be cached, because they are assigned only by set(): 786 // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize 787 788 mLock.unlock(); 789 790 // perform callbacks while unlocked 791 if (newOverrun) { 792 mCbf(EVENT_OVERRUN, mUserData, NULL); 793 } 794 if (markerReached) { 795 mCbf(EVENT_MARKER, mUserData, &markerPosition); 796 } 797 while (newPosCount > 0) { 798 size_t temp = newPosition; 799 mCbf(EVENT_NEW_POS, mUserData, &temp); 800 newPosition += updatePeriod; 801 newPosCount--; 802 } 803 if (mObservedSequence != sequence) { 804 mObservedSequence = sequence; 805 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 806 } 807 808 // if inactive, then don't run me again until re-started 809 if (!active) { 810 return NS_INACTIVE; 811 } 812 813 // Compute the estimated time until the next timed event (position, markers) 814 uint32_t minFrames = ~0; 815 if (!markerReached && position < markerPosition) { 816 minFrames = markerPosition - position; 817 } 818 if (updatePeriod > 0 && updatePeriod < minFrames) { 819 minFrames = updatePeriod; 820 } 821 822 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 823 static const uint32_t kPoll = 0; 824 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 825 minFrames = kPoll * notificationFrames; 826 } 827 828 // Convert frame units to time units 829 nsecs_t ns = NS_WHENEVER; 830 if (minFrames != (uint32_t) ~0) { 831 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 832 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 833 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 834 } 835 836 // If not supplying data by EVENT_MORE_DATA, then we're done 837 if (mTransfer != TRANSFER_CALLBACK) { 838 return ns; 839 } 840 841 struct timespec timeout; 842 const struct timespec *requested = &ClientProxy::kForever; 843 if (ns != NS_WHENEVER) { 844 timeout.tv_sec = ns / 1000000000LL; 845 timeout.tv_nsec = ns % 1000000000LL; 846 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 847 requested = &timeout; 848 } 849 850 while (mRemainingFrames > 0) { 851 852 Buffer audioBuffer; 853 audioBuffer.frameCount = mRemainingFrames; 854 size_t nonContig; 855 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 856 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 857 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 858 requested = &ClientProxy::kNonBlocking; 859 size_t avail = audioBuffer.frameCount + nonContig; 860 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 861 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 862 if (err != NO_ERROR) { 863 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 864 break; 865 } 866 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 867 return NS_NEVER; 868 } 869 870 if (mRetryOnPartialBuffer) { 871 mRetryOnPartialBuffer = false; 872 if (avail < mRemainingFrames) { 873 int64_t myns = ((mRemainingFrames - avail) * 874 1100000000LL) / mSampleRate; 875 if (ns < 0 || myns < ns) { 876 ns = myns; 877 } 878 return ns; 879 } 880 } 881 882 size_t reqSize = audioBuffer.size; 883 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 884 size_t readSize = audioBuffer.size; 885 886 // Sanity check on returned size 887 if (ssize_t(readSize) < 0 || readSize > reqSize) { 888 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 889 reqSize, (int) readSize); 890 return NS_NEVER; 891 } 892 893 if (readSize == 0) { 894 // The callback is done consuming buffers 895 // Keep this thread going to handle timed events and 896 // still try to provide more data in intervals of WAIT_PERIOD_MS 897 // but don't just loop and block the CPU, so wait 898 return WAIT_PERIOD_MS * 1000000LL; 899 } 900 901 size_t releasedFrames = readSize / mFrameSize; 902 audioBuffer.frameCount = releasedFrames; 903 mRemainingFrames -= releasedFrames; 904 if (misalignment >= releasedFrames) { 905 misalignment -= releasedFrames; 906 } else { 907 misalignment = 0; 908 } 909 910 releaseBuffer(&audioBuffer); 911 912 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 913 // if callback doesn't like to accept the full chunk 914 if (readSize < reqSize) { 915 continue; 916 } 917 918 // There could be enough non-contiguous frames available to satisfy the remaining request 919 if (mRemainingFrames <= nonContig) { 920 continue; 921 } 922 923#if 0 924 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 925 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 926 // that total to a sum == notificationFrames. 927 if (0 < misalignment && misalignment <= mRemainingFrames) { 928 mRemainingFrames = misalignment; 929 return (mRemainingFrames * 1100000000LL) / mSampleRate; 930 } 931#endif 932 933 } 934 mRemainingFrames = notificationFrames; 935 mRetryOnPartialBuffer = true; 936 937 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 938 return 0; 939} 940 941status_t AudioRecord::restoreRecord_l(const char *from) 942{ 943 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 944 ++mSequence; 945 status_t result; 946 947 // if the new IAudioRecord is created, openRecord_l() will modify the 948 // following member variables: mAudioRecord, mCblkMemory and mCblk. 949 // It will also delete the strong references on previous IAudioRecord and IMemory 950 size_t position = mProxy->getPosition(); 951 mNewPosition = position + mUpdatePeriod; 952 result = openRecord_l(position); 953 if (result == NO_ERROR) { 954 if (mActive) { 955 // callback thread or sync event hasn't changed 956 // FIXME this fails if we have a new AudioFlinger instance 957 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 958 } 959 } 960 if (result != NO_ERROR) { 961 ALOGW("restoreRecord_l() failed status %d", result); 962 mActive = false; 963 } 964 965 return result; 966} 967 968// ========================================================================= 969 970void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 971{ 972 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 973 if (audioRecord != 0) { 974 AutoMutex lock(audioRecord->mLock); 975 audioRecord->mProxy->binderDied(); 976 } 977} 978 979// ========================================================================= 980 981AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 982 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 983 mIgnoreNextPausedInt(false) 984{ 985} 986 987AudioRecord::AudioRecordThread::~AudioRecordThread() 988{ 989} 990 991bool AudioRecord::AudioRecordThread::threadLoop() 992{ 993 { 994 AutoMutex _l(mMyLock); 995 if (mPaused) { 996 mMyCond.wait(mMyLock); 997 // caller will check for exitPending() 998 return true; 999 } 1000 if (mIgnoreNextPausedInt) { 1001 mIgnoreNextPausedInt = false; 1002 mPausedInt = false; 1003 } 1004 if (mPausedInt) { 1005 if (mPausedNs > 0) { 1006 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1007 } else { 1008 mMyCond.wait(mMyLock); 1009 } 1010 mPausedInt = false; 1011 return true; 1012 } 1013 } 1014 nsecs_t ns = mReceiver.processAudioBuffer(); 1015 switch (ns) { 1016 case 0: 1017 return true; 1018 case NS_INACTIVE: 1019 pauseInternal(); 1020 return true; 1021 case NS_NEVER: 1022 return false; 1023 case NS_WHENEVER: 1024 // FIXME increase poll interval, or make event-driven 1025 ns = 1000000000LL; 1026 // fall through 1027 default: 1028 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1029 pauseInternal(ns); 1030 return true; 1031 } 1032} 1033 1034void AudioRecord::AudioRecordThread::requestExit() 1035{ 1036 // must be in this order to avoid a race condition 1037 Thread::requestExit(); 1038 resume(); 1039} 1040 1041void AudioRecord::AudioRecordThread::pause() 1042{ 1043 AutoMutex _l(mMyLock); 1044 mPaused = true; 1045} 1046 1047void AudioRecord::AudioRecordThread::resume() 1048{ 1049 AutoMutex _l(mMyLock); 1050 mIgnoreNextPausedInt = true; 1051 if (mPaused || mPausedInt) { 1052 mPaused = false; 1053 mPausedInt = false; 1054 mMyCond.signal(); 1055 } 1056} 1057 1058void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns) 1059{ 1060 AutoMutex _l(mMyLock); 1061 mPausedInt = true; 1062 mPausedNs = ns; 1063} 1064 1065// ------------------------------------------------------------------------- 1066 1067}; // namespace android 1068