AudioRecord.cpp revision 2b2165c75790050810460c8de3f414876bce4c0e
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 // 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 417unsigned int 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 int originalSessionId = mSessionId; 468 sp<IAudioRecord> record = audioFlinger->openRecord(input, 469 mSampleRate, mFormat, 470 mChannelMask, 471 mFrameCount, 472 &trackFlags, 473 tid, 474 &mSessionId, 475 &status); 476 ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId, 477 "session ID changed from %d to %d", originalSessionId, mSessionId); 478 479 if (record == 0 || status != NO_ERROR) { 480 ALOGE("AudioFlinger could not create record track, status: %d", status); 481 AudioSystem::releaseInput(input); 482 return status; 483 } 484 sp<IMemory> iMem = record->getCblk(); 485 if (iMem == 0) { 486 ALOGE("Could not get control block"); 487 return NO_INIT; 488 } 489 void *iMemPointer = iMem->pointer(); 490 if (iMemPointer == NULL) { 491 ALOGE("Could not get control block pointer"); 492 return NO_INIT; 493 } 494 if (mAudioRecord != 0) { 495 mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); 496 mDeathNotifier.clear(); 497 } 498 mInput = input; 499 mAudioRecord = record; 500 mCblkMemory = iMem; 501 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 502 mCblk = cblk; 503 // FIXME missing fast track frameCount logic 504 mAwaitBoost = false; 505 if (mFlags & AUDIO_INPUT_FLAG_FAST) { 506 if (trackFlags & IAudioFlinger::TRACK_FAST) { 507 ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", mFrameCount); 508 mAwaitBoost = true; 509 // double-buffering is not required for fast tracks, due to tighter scheduling 510 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount) { 511 mNotificationFramesAct = mFrameCount; 512 } 513 } else { 514 ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", mFrameCount); 515 // once denied, do not request again if IAudioRecord is re-created 516 mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST); 517 if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) { 518 mNotificationFramesAct = mFrameCount/2; 519 } 520 } 521 } 522 523 // starting address of buffers in shared memory 524 void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); 525 526 // update proxy 527 mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize); 528 mProxy->setEpoch(epoch); 529 mProxy->setMinimum(mNotificationFramesAct); 530 531 mDeathNotifier = new DeathNotifier(this); 532 mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); 533 534 return NO_ERROR; 535} 536 537status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 538{ 539 if (audioBuffer == NULL) { 540 return BAD_VALUE; 541 } 542 if (mTransfer != TRANSFER_OBTAIN) { 543 audioBuffer->frameCount = 0; 544 audioBuffer->size = 0; 545 audioBuffer->raw = NULL; 546 return INVALID_OPERATION; 547 } 548 549 const struct timespec *requested; 550 if (waitCount == -1) { 551 requested = &ClientProxy::kForever; 552 } else if (waitCount == 0) { 553 requested = &ClientProxy::kNonBlocking; 554 } else if (waitCount > 0) { 555 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 556 struct timespec timeout; 557 timeout.tv_sec = ms / 1000; 558 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 559 requested = &timeout; 560 } else { 561 ALOGE("%s invalid waitCount %d", __func__, waitCount); 562 requested = NULL; 563 } 564 return obtainBuffer(audioBuffer, requested); 565} 566 567status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 568 struct timespec *elapsed, size_t *nonContig) 569{ 570 // previous and new IAudioRecord sequence numbers are used to detect track re-creation 571 uint32_t oldSequence = 0; 572 uint32_t newSequence; 573 574 Proxy::Buffer buffer; 575 status_t status = NO_ERROR; 576 577 static const int32_t kMaxTries = 5; 578 int32_t tryCounter = kMaxTries; 579 580 do { 581 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 582 // keep them from going away if another thread re-creates the track during obtainBuffer() 583 sp<AudioRecordClientProxy> proxy; 584 sp<IMemory> iMem; 585 { 586 // start of lock scope 587 AutoMutex lock(mLock); 588 589 newSequence = mSequence; 590 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 591 if (status == DEAD_OBJECT) { 592 // re-create track, unless someone else has already done so 593 if (newSequence == oldSequence) { 594 status = restoreRecord_l("obtainBuffer"); 595 if (status != NO_ERROR) { 596 break; 597 } 598 } 599 } 600 oldSequence = newSequence; 601 602 // Keep the extra references 603 proxy = mProxy; 604 iMem = mCblkMemory; 605 606 // Non-blocking if track is stopped 607 if (!mActive) { 608 requested = &ClientProxy::kNonBlocking; 609 } 610 611 } // end of lock scope 612 613 buffer.mFrameCount = audioBuffer->frameCount; 614 // FIXME starts the requested timeout and elapsed over from scratch 615 status = proxy->obtainBuffer(&buffer, requested, elapsed); 616 617 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 618 619 audioBuffer->frameCount = buffer.mFrameCount; 620 audioBuffer->size = buffer.mFrameCount * mFrameSize; 621 audioBuffer->raw = buffer.mRaw; 622 if (nonContig != NULL) { 623 *nonContig = buffer.mNonContig; 624 } 625 return status; 626} 627 628void AudioRecord::releaseBuffer(Buffer* audioBuffer) 629{ 630 // all TRANSFER_* are valid 631 632 size_t stepCount = audioBuffer->size / mFrameSize; 633 if (stepCount == 0) { 634 return; 635 } 636 637 Proxy::Buffer buffer; 638 buffer.mFrameCount = stepCount; 639 buffer.mRaw = audioBuffer->raw; 640 641 AutoMutex lock(mLock); 642 mInOverrun = false; 643 mProxy->releaseBuffer(&buffer); 644 645 // the server does not automatically disable recorder on overrun, so no need to restart 646} 647 648audio_io_handle_t AudioRecord::getInput() const 649{ 650 AutoMutex lock(mLock); 651 return mInput; 652} 653 654// ------------------------------------------------------------------------- 655 656ssize_t AudioRecord::read(void* buffer, size_t userSize) 657{ 658 if (mTransfer != TRANSFER_SYNC) { 659 return INVALID_OPERATION; 660 } 661 662 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 663 // sanity-check. user is most-likely passing an error code, and it would 664 // make the return value ambiguous (actualSize vs error). 665 ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 666 return BAD_VALUE; 667 } 668 669 ssize_t read = 0; 670 Buffer audioBuffer; 671 672 while (userSize >= mFrameSize) { 673 audioBuffer.frameCount = userSize / mFrameSize; 674 675 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 676 if (err < 0) { 677 if (read > 0) { 678 break; 679 } 680 return ssize_t(err); 681 } 682 683 size_t bytesRead = audioBuffer.size; 684 memcpy(buffer, audioBuffer.i8, bytesRead); 685 buffer = ((char *) buffer) + bytesRead; 686 userSize -= bytesRead; 687 read += bytesRead; 688 689 releaseBuffer(&audioBuffer); 690 } 691 692 return read; 693} 694 695// ------------------------------------------------------------------------- 696 697nsecs_t AudioRecord::processAudioBuffer() 698{ 699 mLock.lock(); 700 if (mAwaitBoost) { 701 mAwaitBoost = false; 702 mLock.unlock(); 703 static const int32_t kMaxTries = 5; 704 int32_t tryCounter = kMaxTries; 705 uint32_t pollUs = 10000; 706 do { 707 int policy = sched_getscheduler(0); 708 if (policy == SCHED_FIFO || policy == SCHED_RR) { 709 break; 710 } 711 usleep(pollUs); 712 pollUs <<= 1; 713 } while (tryCounter-- > 0); 714 if (tryCounter < 0) { 715 ALOGE("did not receive expected priority boost on time"); 716 } 717 // Run again immediately 718 return 0; 719 } 720 721 // Can only reference mCblk while locked 722 int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); 723 724 // Check for track invalidation 725 if (flags & CBLK_INVALID) { 726 (void) restoreRecord_l("processAudioBuffer"); 727 mLock.unlock(); 728 // Run again immediately, but with a new IAudioRecord 729 return 0; 730 } 731 732 bool active = mActive; 733 734 // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() 735 bool newOverrun = false; 736 if (flags & CBLK_OVERRUN) { 737 if (!mInOverrun) { 738 mInOverrun = true; 739 newOverrun = true; 740 } 741 } 742 743 // Get current position of server 744 size_t position = mProxy->getPosition(); 745 746 // Manage marker callback 747 bool markerReached = false; 748 size_t markerPosition = mMarkerPosition; 749 // FIXME fails for wraparound, need 64 bits 750 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 751 mMarkerReached = markerReached = true; 752 } 753 754 // Determine the number of new position callback(s) that will be needed, while locked 755 size_t newPosCount = 0; 756 size_t newPosition = mNewPosition; 757 uint32_t updatePeriod = mUpdatePeriod; 758 // FIXME fails for wraparound, need 64 bits 759 if (updatePeriod > 0 && position >= newPosition) { 760 newPosCount = ((position - newPosition) / updatePeriod) + 1; 761 mNewPosition += updatePeriod * newPosCount; 762 } 763 764 // Cache other fields that will be needed soon 765 size_t notificationFrames = mNotificationFramesAct; 766 if (mRefreshRemaining) { 767 mRefreshRemaining = false; 768 mRemainingFrames = notificationFrames; 769 mRetryOnPartialBuffer = false; 770 } 771 size_t misalignment = mProxy->getMisalignment(); 772 int32_t sequence = mSequence; 773 774 // These fields don't need to be cached, because they are assigned only by set(): 775 // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize 776 777 mLock.unlock(); 778 779 // perform callbacks while unlocked 780 if (newOverrun) { 781 mCbf(EVENT_OVERRUN, mUserData, NULL); 782 } 783 if (markerReached) { 784 mCbf(EVENT_MARKER, mUserData, &markerPosition); 785 } 786 while (newPosCount > 0) { 787 size_t temp = newPosition; 788 mCbf(EVENT_NEW_POS, mUserData, &temp); 789 newPosition += updatePeriod; 790 newPosCount--; 791 } 792 if (mObservedSequence != sequence) { 793 mObservedSequence = sequence; 794 mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); 795 } 796 797 // if inactive, then don't run me again until re-started 798 if (!active) { 799 return NS_INACTIVE; 800 } 801 802 // Compute the estimated time until the next timed event (position, markers) 803 uint32_t minFrames = ~0; 804 if (!markerReached && position < markerPosition) { 805 minFrames = markerPosition - position; 806 } 807 if (updatePeriod > 0 && updatePeriod < minFrames) { 808 minFrames = updatePeriod; 809 } 810 811 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 812 static const uint32_t kPoll = 0; 813 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 814 minFrames = kPoll * notificationFrames; 815 } 816 817 // Convert frame units to time units 818 nsecs_t ns = NS_WHENEVER; 819 if (minFrames != (uint32_t) ~0) { 820 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 821 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 822 ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; 823 } 824 825 // If not supplying data by EVENT_MORE_DATA, then we're done 826 if (mTransfer != TRANSFER_CALLBACK) { 827 return ns; 828 } 829 830 struct timespec timeout; 831 const struct timespec *requested = &ClientProxy::kForever; 832 if (ns != NS_WHENEVER) { 833 timeout.tv_sec = ns / 1000000000LL; 834 timeout.tv_nsec = ns % 1000000000LL; 835 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 836 requested = &timeout; 837 } 838 839 while (mRemainingFrames > 0) { 840 841 Buffer audioBuffer; 842 audioBuffer.frameCount = mRemainingFrames; 843 size_t nonContig; 844 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 845 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 846 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 847 requested = &ClientProxy::kNonBlocking; 848 size_t avail = audioBuffer.frameCount + nonContig; 849 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 850 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 851 if (err != NO_ERROR) { 852 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 853 break; 854 } 855 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 856 return NS_NEVER; 857 } 858 859 if (mRetryOnPartialBuffer) { 860 mRetryOnPartialBuffer = false; 861 if (avail < mRemainingFrames) { 862 int64_t myns = ((mRemainingFrames - avail) * 863 1100000000LL) / mSampleRate; 864 if (ns < 0 || myns < ns) { 865 ns = myns; 866 } 867 return ns; 868 } 869 } 870 871 size_t reqSize = audioBuffer.size; 872 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 873 size_t readSize = audioBuffer.size; 874 875 // Sanity check on returned size 876 if (ssize_t(readSize) < 0 || readSize > reqSize) { 877 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 878 reqSize, (int) readSize); 879 return NS_NEVER; 880 } 881 882 if (readSize == 0) { 883 // The callback is done consuming buffers 884 // Keep this thread going to handle timed events and 885 // still try to provide more data in intervals of WAIT_PERIOD_MS 886 // but don't just loop and block the CPU, so wait 887 return WAIT_PERIOD_MS * 1000000LL; 888 } 889 890 size_t releasedFrames = readSize / mFrameSize; 891 audioBuffer.frameCount = releasedFrames; 892 mRemainingFrames -= releasedFrames; 893 if (misalignment >= releasedFrames) { 894 misalignment -= releasedFrames; 895 } else { 896 misalignment = 0; 897 } 898 899 releaseBuffer(&audioBuffer); 900 901 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 902 // if callback doesn't like to accept the full chunk 903 if (readSize < reqSize) { 904 continue; 905 } 906 907 // There could be enough non-contiguous frames available to satisfy the remaining request 908 if (mRemainingFrames <= nonContig) { 909 continue; 910 } 911 912#if 0 913 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 914 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 915 // that total to a sum == notificationFrames. 916 if (0 < misalignment && misalignment <= mRemainingFrames) { 917 mRemainingFrames = misalignment; 918 return (mRemainingFrames * 1100000000LL) / mSampleRate; 919 } 920#endif 921 922 } 923 mRemainingFrames = notificationFrames; 924 mRetryOnPartialBuffer = true; 925 926 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 927 return 0; 928} 929 930status_t AudioRecord::restoreRecord_l(const char *from) 931{ 932 ALOGW("dead IAudioRecord, creating a new one from %s()", from); 933 ++mSequence; 934 status_t result; 935 936 // if the new IAudioRecord is created, openRecord_l() will modify the 937 // following member variables: mAudioRecord, mCblkMemory and mCblk. 938 // It will also delete the strong references on previous IAudioRecord and IMemory 939 size_t position = mProxy->getPosition(); 940 mNewPosition = position + mUpdatePeriod; 941 result = openRecord_l(position); 942 if (result == NO_ERROR) { 943 if (mActive) { 944 // callback thread or sync event hasn't changed 945 // FIXME this fails if we have a new AudioFlinger instance 946 result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); 947 } 948 } 949 if (result != NO_ERROR) { 950 ALOGW("restoreRecord_l() failed status %d", result); 951 mActive = false; 952 } 953 954 return result; 955} 956 957// ========================================================================= 958 959void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 960{ 961 sp<AudioRecord> audioRecord = mAudioRecord.promote(); 962 if (audioRecord != 0) { 963 AutoMutex lock(audioRecord->mLock); 964 audioRecord->mProxy->binderDied(); 965 } 966} 967 968// ========================================================================= 969 970AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) 971 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 972 mIgnoreNextPausedInt(false) 973{ 974} 975 976AudioRecord::AudioRecordThread::~AudioRecordThread() 977{ 978} 979 980bool AudioRecord::AudioRecordThread::threadLoop() 981{ 982 { 983 AutoMutex _l(mMyLock); 984 if (mPaused) { 985 mMyCond.wait(mMyLock); 986 // caller will check for exitPending() 987 return true; 988 } 989 if (mIgnoreNextPausedInt) { 990 mIgnoreNextPausedInt = false; 991 mPausedInt = false; 992 } 993 if (mPausedInt) { 994 if (mPausedNs > 0) { 995 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 996 } else { 997 mMyCond.wait(mMyLock); 998 } 999 mPausedInt = false; 1000 return true; 1001 } 1002 } 1003 nsecs_t ns = mReceiver.processAudioBuffer(); 1004 switch (ns) { 1005 case 0: 1006 return true; 1007 case NS_INACTIVE: 1008 pauseInternal(); 1009 return true; 1010 case NS_NEVER: 1011 return false; 1012 case NS_WHENEVER: 1013 // FIXME increase poll interval, or make event-driven 1014 ns = 1000000000LL; 1015 // fall through 1016 default: 1017 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1018 pauseInternal(ns); 1019 return true; 1020 } 1021} 1022 1023void AudioRecord::AudioRecordThread::requestExit() 1024{ 1025 // must be in this order to avoid a race condition 1026 Thread::requestExit(); 1027 resume(); 1028} 1029 1030void AudioRecord::AudioRecordThread::pause() 1031{ 1032 AutoMutex _l(mMyLock); 1033 mPaused = true; 1034} 1035 1036void AudioRecord::AudioRecordThread::resume() 1037{ 1038 AutoMutex _l(mMyLock); 1039 mIgnoreNextPausedInt = true; 1040 if (mPaused || mPausedInt) { 1041 mPaused = false; 1042 mPausedInt = false; 1043 mMyCond.signal(); 1044 } 1045} 1046 1047void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns) 1048{ 1049 AutoMutex _l(mMyLock); 1050 mPausedInt = true; 1051 mPausedNs = ns; 1052} 1053 1054// ------------------------------------------------------------------------- 1055 1056}; // namespace android 1057