AudioRecord.cpp revision 5a6cd224d07c05b496b6aca050ce5ecf96f125af
1/* 2** 3** Copyright 2008, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18//#define LOG_NDEBUG 0 19#define LOG_TAG "AudioRecord" 20 21#include <sys/resource.h> 22#include <binder/IPCThreadState.h> 23#include <media/AudioRecord.h> 24#include <utils/Log.h> 25#include <private/media/AudioTrackShared.h> 26#include <media/IAudioFlinger.h> 27 28#define WAIT_PERIOD_MS 10 29 30namespace android { 31// --------------------------------------------------------------------------- 32 33// static 34status_t AudioRecord::getMinFrameCount( 35 size_t* frameCount, 36 uint32_t sampleRate, 37 audio_format_t format, 38 audio_channel_mask_t channelMask) 39{ 40 if (frameCount == NULL) { 41 return BAD_VALUE; 42 } 43 44 // default to 0 in case of error 45 *frameCount = 0; 46 47 size_t size = 0; 48 status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size); 49 if (status != NO_ERROR) { 50 ALOGE("AudioSystem could not query the input buffer size; status %d", status); 51 return NO_INIT; 52 } 53 54 if (size == 0) { 55 ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x", 56 sampleRate, format, channelMask); 57 return BAD_VALUE; 58 } 59 60 // We double the size of input buffer for ping pong use of record buffer. 61 size <<= 1; 62 63 // Assumes audio_is_linear_pcm(format) 64 uint32_t channelCount = popcount(channelMask); 65 size /= channelCount * audio_bytes_per_sample(format); 66 67 *frameCount = size; 68 return NO_ERROR; 69} 70 71// --------------------------------------------------------------------------- 72 73AudioRecord::AudioRecord() 74 : mStatus(NO_INIT), mSessionId(0), 75 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) 76{ 77} 78 79AudioRecord::AudioRecord( 80 audio_source_t inputSource, 81 uint32_t sampleRate, 82 audio_format_t format, 83 audio_channel_mask_t channelMask, 84 int frameCount, 85 callback_t cbf, 86 void* user, 87 int notificationFrames, 88 int sessionId, 89 transfer_type transferType, 90 audio_input_flags_t flags) 91 : mStatus(NO_INIT), mSessionId(0), 92 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 93 mPreviousSchedulingGroup(SP_DEFAULT), 94 mProxy(NULL) 95{ 96 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, 97 notificationFrames, false /*threadCanCallJava*/, sessionId, transferType); 98} 99 100AudioRecord::~AudioRecord() 101{ 102 if (mStatus == NO_ERROR) { 103 // Make sure that callback function exits in the case where 104 // it is looping on buffer empty condition in obtainBuffer(). 105 // Otherwise the callback thread will never exit. 106 stop(); 107 if (mAudioRecordThread != 0) { 108 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 == 0 ) { 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()); 292 293 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 if (mCbf == NULL) { 356 return INVALID_OPERATION; 357 } 358 359 AutoMutex lock(mLock); 360 mMarkerPosition = marker; 361 mMarkerReached = false; 362 363 return NO_ERROR; 364} 365 366status_t AudioRecord::getMarkerPosition(uint32_t *marker) const 367{ 368 if (marker == NULL) { 369 return BAD_VALUE; 370 } 371 372 AutoMutex lock(mLock); 373 *marker = mMarkerPosition; 374 375 return NO_ERROR; 376} 377 378status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) 379{ 380 if (mCbf == NULL) { 381 return INVALID_OPERATION; 382 } 383 384 AutoMutex lock(mLock); 385 mNewPosition = mProxy->getPosition() + updatePeriod; 386 mUpdatePeriod = updatePeriod; 387 388 return NO_ERROR; 389} 390 391status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const 392{ 393 if (updatePeriod == NULL) { 394 return BAD_VALUE; 395 } 396 397 AutoMutex lock(mLock); 398 *updatePeriod = mUpdatePeriod; 399 400 return NO_ERROR; 401} 402 403status_t AudioRecord::getPosition(uint32_t *position) const 404{ 405 if (position == NULL) { 406 return BAD_VALUE; 407 } 408 409 AutoMutex lock(mLock); 410 *position = mProxy->getPosition(); 411 412 return NO_ERROR; 413} 414 415unsigned int AudioRecord::getInputFramesLost() const 416{ 417 // no need to check mActive, because if inactive this will return 0, which is what we want 418 return AudioSystem::getInputFramesLost(getInput()); 419} 420 421// ------------------------------------------------------------------------- 422 423// must be called with mLock held 424status_t AudioRecord::openRecord_l(size_t epoch) 425{ 426 status_t status; 427 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 428 if (audioFlinger == 0) { 429 ALOGE("Could not get audioflinger"); 430 return NO_INIT; 431 } 432 433 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 434 pid_t tid = -1; 435 436 // Client can only express a preference for FAST. Server will perform additional tests. 437 // The only supported use case for FAST is callback transfer mode. 438 if (mFlags & AUDIO_INPUT_FLAG_FAST) { 439 if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) { 440 ALOGW("AUDIO_INPUT_FLAG_FAST denied by client"); 441 // once denied, do not request again if IAudioRecord is re-created 442 mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); 443 } else { 444 trackFlags |= IAudioFlinger::TRACK_FAST; 445 tid = mAudioRecordThread->getTid(); 446 } 447 } 448 449 mNotificationFramesAct = mNotificationFramesReq; 450 451 if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) { 452 // Make sure that application is notified with sufficient margin before overrun 453 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) { 454 mNotificationFramesAct = mFrameCount/2; 455 } 456 } 457 458 audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat, 459 mChannelMask, mSessionId); 460 if (input == 0) { 461 ALOGE("Could not get audio input for record source %d", mInputSource); 462 return BAD_VALUE; 463 } 464 465 int originalSessionId = mSessionId; 466 sp<IAudioRecord> record = audioFlinger->openRecord(input, 467 mSampleRate, mFormat, 468 mChannelMask, 469 mFrameCount, 470 &trackFlags, 471 tid, 472 &mSessionId, 473 &status); 474 ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId, 475 "session ID changed from %d to %d", originalSessionId, mSessionId); 476 477 if (record == 0) { 478 ALOGE("AudioFlinger could not create record track, status: %d", status); 479 AudioSystem::releaseInput(input); 480 return status; 481 } 482 sp<IMemory> iMem = record->getCblk(); 483 if (iMem == 0) { 484 ALOGE("Could not get control block"); 485 return NO_INIT; 486 } 487 if (mAudioRecord != 0) { 488 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 489 mDeathNotifier.clear(); 490 } 491 mInput = input; 492 mAudioRecord = record; 493 mCblkMemory = iMem; 494 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 495 mCblk = cblk; 496 // FIXME missing fast track frameCount logic 497 mAwaitBoost = false; 498 if (mFlags & AUDIO_INPUT_FLAG_FAST) { 499 if (trackFlags & IAudioFlinger::TRACK_FAST) { 500 ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", mFrameCount); 501 mAwaitBoost = true; 502 // double-buffering is not required for fast tracks, due to tighter scheduling 503 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount) { 504 mNotificationFramesAct = mFrameCount; 505 } 506 } else { 507 ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", mFrameCount); 508 // once denied, do not request again if IAudioRecord is re-created 509 mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); 510 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) { 511 mNotificationFramesAct = mFrameCount/2; 512 } 513 } 514 } 515 516 // starting address of buffers in shared memory 517 void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); 518 519 // update proxy 520 mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize); 521 mProxy->setEpoch(epoch); 522 mProxy->setMinimum(mNotificationFramesAct); 523 524 mDeathNotifier = new DeathNotifier(this); 525 mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); 526 527 return NO_ERROR; 528} 529 530status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 531{ 532 if (audioBuffer == NULL) { 533 return BAD_VALUE; 534 } 535 if (mTransfer != TRANSFER_OBTAIN) { 536 audioBuffer->frameCount = 0; 537 audioBuffer->size = 0; 538 audioBuffer->raw = NULL; 539 return INVALID_OPERATION; 540 } 541 542 const struct timespec *requested; 543 if (waitCount == -1) { 544 requested = &ClientProxy::kForever; 545 } else if (waitCount == 0) { 546 requested = &ClientProxy::kNonBlocking; 547 } else if (waitCount > 0) { 548 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 549 struct timespec timeout; 550 timeout.tv_sec = ms / 1000; 551 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 552 requested = &timeout; 553 } else { 554 ALOGE("%s invalid waitCount %d", __func__, waitCount); 555 requested = NULL; 556 } 557 return obtainBuffer(audioBuffer, requested); 558} 559 560status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 561 struct timespec *elapsed, size_t *nonContig) 562{ 563 // previous and new IAudioRecord sequence numbers are used to detect track re-creation 564 uint32_t oldSequence = 0; 565 uint32_t newSequence; 566 567 Proxy::Buffer buffer; 568 status_t status = NO_ERROR; 569 570 static const int32_t kMaxTries = 5; 571 int32_t tryCounter = kMaxTries; 572 573 do { 574 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 575 // keep them from going away if another thread re-creates the track during obtainBuffer() 576 sp<AudioRecordClientProxy> proxy; 577 sp<IMemory> iMem; 578 { 579 // start of lock scope 580 AutoMutex lock(mLock); 581 582 newSequence = mSequence; 583 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 584 if (status == DEAD_OBJECT) { 585 // re-create track, unless someone else has already done so 586 if (newSequence == oldSequence) { 587 status = restoreRecord_l("obtainBuffer"); 588 if (status != NO_ERROR) { 589 break; 590 } 591 } 592 } 593 oldSequence = newSequence; 594 595 // Keep the extra references 596 proxy = mProxy; 597 iMem = mCblkMemory; 598 599 // Non-blocking if track is stopped 600 if (!mActive) { 601 requested = &ClientProxy::kNonBlocking; 602 } 603 604 } // end of lock scope 605 606 buffer.mFrameCount = audioBuffer->frameCount; 607 // FIXME starts the requested timeout and elapsed over from scratch 608 status = proxy->obtainBuffer(&buffer, requested, elapsed); 609 610 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 611 612 audioBuffer->frameCount = buffer.mFrameCount; 613 audioBuffer->size = buffer.mFrameCount * mFrameSize; 614 audioBuffer->raw = buffer.mRaw; 615 if (nonContig != NULL) { 616 *nonContig = buffer.mNonContig; 617 } 618 return status; 619} 620 621void AudioRecord::releaseBuffer(Buffer* audioBuffer) 622{ 623 // all TRANSFER_* are valid 624 625 size_t stepCount = audioBuffer->size / mFrameSize; 626 if (stepCount == 0) { 627 return; 628 } 629 630 Proxy::Buffer buffer; 631 buffer.mFrameCount = stepCount; 632 buffer.mRaw = audioBuffer->raw; 633 634 AutoMutex lock(mLock); 635 mInOverrun = false; 636 mProxy->releaseBuffer(&buffer); 637 638 // the server does not automatically disable recorder on overrun, so no need to restart 639} 640 641audio_io_handle_t AudioRecord::getInput() const 642{ 643 AutoMutex lock(mLock); 644 return mInput; 645} 646 647// ------------------------------------------------------------------------- 648 649ssize_t AudioRecord::read(void* buffer, size_t userSize) 650{ 651 if (mTransfer != TRANSFER_SYNC) { 652 return INVALID_OPERATION; 653 } 654 655 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 656 // sanity-check. user is most-likely passing an error code, and it would 657 // make the return value ambiguous (actualSize vs error). 658 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 659 return BAD_VALUE; 660 } 661 662 ssize_t read = 0; 663 Buffer audioBuffer; 664 665 while (userSize >= mFrameSize) { 666 audioBuffer.frameCount = userSize / mFrameSize; 667 668 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 669 if (err < 0) { 670 if (read > 0) { 671 break; 672 } 673 return ssize_t(err); 674 } 675 676 size_t bytesRead = audioBuffer.size; 677 memcpy(buffer, audioBuffer.i8, bytesRead); 678 buffer = ((char *) buffer) + bytesRead; 679 userSize -= bytesRead; 680 read += bytesRead; 681 682 releaseBuffer(&audioBuffer); 683 } 684 685 return read; 686} 687 688// ------------------------------------------------------------------------- 689 690nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) 691{ 692 mLock.lock(); 693 if (mAwaitBoost) { 694 mAwaitBoost = false; 695 mLock.unlock(); 696 static const int32_t kMaxTries = 5; 697 int32_t tryCounter = kMaxTries; 698 uint32_t pollUs = 10000; 699 do { 700 int policy = sched_getscheduler(0); 701 if (policy == SCHED_FIFO || policy == SCHED_RR) { 702 break; 703 } 704 usleep(pollUs); 705 pollUs <<= 1; 706 } while (tryCounter-- > 0); 707 if (tryCounter < 0) { 708 ALOGE("did not receive expected priority boost on time"); 709 } 710 // Run again immediately 711 return 0; 712 } 713 714 // Can only reference mCblk while locked 715 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 716 717 // Check for track invalidation 718 if (flags & CBLK_INVALID) { 719 (void) restoreRecord_l("processAudioBuffer"); 720 mLock.unlock(); 721 // Run again immediately, but with a new IAudioRecord 722 return 0; 723 } 724 725 bool active = mActive; 726 727 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 728 bool newOverrun = false; 729 if (flags & CBLK_OVERRUN) { 730 if (!mInOverrun) { 731 mInOverrun = true; 732 newOverrun = true; 733 } 734 } 735 736 // Get current position of server 737 size_t position = mProxy->getPosition(); 738 739 // Manage marker callback 740 bool markerReached = false; 741 size_t markerPosition = mMarkerPosition; 742 // FIXME fails for wraparound, need 64 bits 743 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 744 mMarkerReached = markerReached = true; 745 } 746 747 // Determine the number of new position callback(s) that will be needed, while locked 748 size_t newPosCount = 0; 749 size_t newPosition = mNewPosition; 750 uint32_t updatePeriod = mUpdatePeriod; 751 // FIXME fails for wraparound, need 64 bits 752 if (updatePeriod > 0 && position >= newPosition) { 753 newPosCount = ((position - newPosition) / updatePeriod) + 1; 754 mNewPosition += updatePeriod * newPosCount; 755 } 756 757 // Cache other fields that will be needed soon 758 size_t notificationFrames = mNotificationFramesAct; 759 if (mRefreshRemaining) { 760 mRefreshRemaining = false; 761 mRemainingFrames = notificationFrames; 762 mRetryOnPartialBuffer = false; 763 } 764 size_t misalignment = mProxy->getMisalignment(); 765 int32_t sequence = mSequence; 766 767 // These fields don't need to be cached, because they are assigned only by set(): 768 // mTransfer, mCbf, mUserData, mSampleRate 769 770 mLock.unlock(); 771 772 // perform callbacks while unlocked 773 if (newOverrun) { 774 mCbf(EVENT_OVERRUN, mUserData, NULL); 775 } 776 if (markerReached) { 777 mCbf(EVENT_MARKER, mUserData, &markerPosition); 778 } 779 while (newPosCount > 0) { 780 size_t temp = newPosition; 781 mCbf(EVENT_NEW_POS, mUserData, &temp); 782 newPosition += updatePeriod; 783 newPosCount--; 784 } 785 if (mObservedSequence != sequence) { 786 mObservedSequence = sequence; 787 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 788 } 789 790 // if inactive, then don't run me again until re-started 791 if (!active) { 792 return NS_INACTIVE; 793 } 794 795 // Compute the estimated time until the next timed event (position, markers) 796 uint32_t minFrames = ~0; 797 if (!markerReached && position < markerPosition) { 798 minFrames = markerPosition - position; 799 } 800 if (updatePeriod > 0 && updatePeriod < minFrames) { 801 minFrames = updatePeriod; 802 } 803 804 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 805 static const uint32_t kPoll = 0; 806 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 807 minFrames = kPoll * notificationFrames; 808 } 809 810 // Convert frame units to time units 811 nsecs_t ns = NS_WHENEVER; 812 if (minFrames != (uint32_t) ~0) { 813 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 814 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 815 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 816 } 817 818 // If not supplying data by EVENT_MORE_DATA, then we're done 819 if (mTransfer != TRANSFER_CALLBACK) { 820 return ns; 821 } 822 823 struct timespec timeout; 824 const struct timespec *requested = &ClientProxy::kForever; 825 if (ns != NS_WHENEVER) { 826 timeout.tv_sec = ns / 1000000000LL; 827 timeout.tv_nsec = ns % 1000000000LL; 828 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 829 requested = &timeout; 830 } 831 832 while (mRemainingFrames > 0) { 833 834 Buffer audioBuffer; 835 audioBuffer.frameCount = mRemainingFrames; 836 size_t nonContig; 837 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 838 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 839 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 840 requested = &ClientProxy::kNonBlocking; 841 size_t avail = audioBuffer.frameCount + nonContig; 842 ALOGV("obtainBuffer(%u) returned %u = %u + %u", 843 mRemainingFrames, avail, audioBuffer.frameCount, nonContig); 844 if (err != NO_ERROR) { 845 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 846 break; 847 } 848 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 849 return NS_NEVER; 850 } 851 852 if (mRetryOnPartialBuffer) { 853 mRetryOnPartialBuffer = false; 854 if (avail < mRemainingFrames) { 855 int64_t myns = ((mRemainingFrames - avail) * 856 1100000000LL) / mSampleRate; 857 if (ns < 0 || myns < ns) { 858 ns = myns; 859 } 860 return ns; 861 } 862 } 863 864 size_t reqSize = audioBuffer.size; 865 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 866 size_t readSize = audioBuffer.size; 867 868 // Sanity check on returned size 869 if (ssize_t(readSize) < 0 || readSize > reqSize) { 870 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 871 reqSize, (int) readSize); 872 return NS_NEVER; 873 } 874 875 if (readSize == 0) { 876 // The callback is done consuming buffers 877 // Keep this thread going to handle timed events and 878 // still try to provide more data in intervals of WAIT_PERIOD_MS 879 // but don't just loop and block the CPU, so wait 880 return WAIT_PERIOD_MS * 1000000LL; 881 } 882 883 size_t releasedFrames = readSize / mFrameSize; 884 audioBuffer.frameCount = releasedFrames; 885 mRemainingFrames -= releasedFrames; 886 if (misalignment >= releasedFrames) { 887 misalignment -= releasedFrames; 888 } else { 889 misalignment = 0; 890 } 891 892 releaseBuffer(&audioBuffer); 893 894 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 895 // if callback doesn't like to accept the full chunk 896 if (readSize < reqSize) { 897 continue; 898 } 899 900 // There could be enough non-contiguous frames available to satisfy the remaining request 901 if (mRemainingFrames <= nonContig) { 902 continue; 903 } 904 905#if 0 906 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 907 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 908 // that total to a sum == notificationFrames. 909 if (0 < misalignment && misalignment <= mRemainingFrames) { 910 mRemainingFrames = misalignment; 911 return (mRemainingFrames * 1100000000LL) / mSampleRate; 912 } 913#endif 914 915 } 916 mRemainingFrames = notificationFrames; 917 mRetryOnPartialBuffer = true; 918 919 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 920 return 0; 921} 922 923status_t AudioRecord::restoreRecord_l(const char *from) 924{ 925 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 926 ++mSequence; 927 status_t result; 928 929 // if the new IAudioRecord is created, openRecord_l() will modify the 930 // following member variables: mAudioRecord, mCblkMemory and mCblk. 931 // It will also delete the strong references on previous IAudioRecord and IMemory 932 size_t position = mProxy->getPosition(); 933 mNewPosition = position + mUpdatePeriod; 934 result = openRecord_l(position); 935 if (result == NO_ERROR) { 936 if (mActive) { 937 // callback thread or sync event hasn't changed 938 // FIXME this fails if we have a new AudioFlinger instance 939 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 940 } 941 } 942 if (result != NO_ERROR) { 943 ALOGW("restoreRecord_l() failed status %d", result); 944 mActive = false; 945 } 946 947 return result; 948} 949 950// ========================================================================= 951 952void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who) 953{ 954 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 955 if (audioRecord != 0) { 956 AutoMutex lock(audioRecord->mLock); 957 audioRecord->mProxy->binderDied(); 958 } 959} 960 961// ========================================================================= 962 963AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 964 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL) 965{ 966} 967 968AudioRecord::AudioRecordThread::~AudioRecordThread() 969{ 970} 971 972bool AudioRecord::AudioRecordThread::threadLoop() 973{ 974 { 975 AutoMutex _l(mMyLock); 976 if (mPaused) { 977 mMyCond.wait(mMyLock); 978 // caller will check for exitPending() 979 return true; 980 } 981 if (mPausedInt) { 982 mPausedInt = false; 983 if (mPausedNs > 0) { 984 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 985 } else { 986 mMyCond.wait(mMyLock); 987 } 988 return true; 989 } 990 } 991 nsecs_t ns = mReceiver.processAudioBuffer(this); 992 switch (ns) { 993 case 0: 994 return true; 995 case NS_INACTIVE: 996 pauseInternal(); 997 return true; 998 case NS_NEVER: 999 return false; 1000 case NS_WHENEVER: 1001 // FIXME increase poll interval, or make event-driven 1002 ns = 1000000000LL; 1003 // fall through 1004 default: 1005 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1006 pauseInternal(ns); 1007 return true; 1008 } 1009} 1010 1011void AudioRecord::AudioRecordThread::requestExit() 1012{ 1013 // must be in this order to avoid a race condition 1014 Thread::requestExit(); 1015 AutoMutex _l(mMyLock); 1016 if (mPaused || mPausedInt) { 1017 mPaused = false; 1018 mPausedInt = false; 1019 mMyCond.signal(); 1020 } 1021} 1022 1023void AudioRecord::AudioRecordThread::pause() 1024{ 1025 AutoMutex _l(mMyLock); 1026 mPaused = true; 1027} 1028 1029void AudioRecord::AudioRecordThread::resume() 1030{ 1031 AutoMutex _l(mMyLock); 1032 if (mPaused) { 1033 mPaused = false; 1034 mMyCond.signal(); 1035 } 1036} 1037 1038void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns) 1039{ 1040 AutoMutex _l(mMyLock); 1041 mPausedInt = true; 1042 mPausedNs = ns; 1043} 1044 1045// ------------------------------------------------------------------------- 1046 1047}; // namespace android 1048