AudioTrack.cpp revision 1179bc9b0e3d17c984e8f4ad38561c049dd102fa
1/* //device/extlibs/pv/android/AudioTrack.cpp 2** 3** Copyright 2007, 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 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <stdint.h> 23#include <sys/types.h> 24#include <limits.h> 25 26#include <sched.h> 27#include <sys/resource.h> 28 29#include <private/media/AudioTrackShared.h> 30 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Log.h> 35#include <utils/MemoryDealer.h> 36#include <utils/Parcel.h> 37#include <utils/IPCThreadState.h> 38#include <utils/Timers.h> 39#include <cutils/atomic.h> 40 41#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) 42#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) 43 44namespace android { 45 46// --------------------------------------------------------------------------- 47 48AudioTrack::AudioTrack() 49 : mStatus(NO_INIT) 50{ 51} 52 53AudioTrack::AudioTrack( 54 int streamType, 55 uint32_t sampleRate, 56 int format, 57 int channelCount, 58 int frameCount, 59 uint32_t flags, 60 callback_t cbf, 61 void* user, 62 int notificationFrames) 63 : mStatus(NO_INIT) 64{ 65 mStatus = set(streamType, sampleRate, format, channelCount, 66 frameCount, flags, cbf, user, notificationFrames, 0); 67} 68 69AudioTrack::AudioTrack( 70 int streamType, 71 uint32_t sampleRate, 72 int format, 73 int channelCount, 74 const sp<IMemory>& sharedBuffer, 75 uint32_t flags, 76 callback_t cbf, 77 void* user, 78 int notificationFrames) 79 : mStatus(NO_INIT) 80{ 81 mStatus = set(streamType, sampleRate, format, channelCount, 82 0, flags, cbf, user, notificationFrames, sharedBuffer); 83} 84 85AudioTrack::~AudioTrack() 86{ 87 LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 88 89 if (mStatus == NO_ERROR) { 90 // Make sure that callback function exits in the case where 91 // it is looping on buffer full condition in obtainBuffer(). 92 // Otherwise the callback thread will never exit. 93 stop(); 94 if (mAudioTrackThread != 0) { 95 mCblk->cv.signal(); 96 mAudioTrackThread->requestExitAndWait(); 97 mAudioTrackThread.clear(); 98 } 99 mAudioTrack.clear(); 100 IPCThreadState::self()->flushCommands(); 101 } 102} 103 104status_t AudioTrack::set( 105 int streamType, 106 uint32_t sampleRate, 107 int format, 108 int channelCount, 109 int frameCount, 110 uint32_t flags, 111 callback_t cbf, 112 void* user, 113 int notificationFrames, 114 const sp<IMemory>& sharedBuffer, 115 bool threadCanCallJava) 116{ 117 118 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 119 120 if (mAudioFlinger != 0) { 121 LOGE("Track already in use"); 122 return INVALID_OPERATION; 123 } 124 125 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 126 if (audioFlinger == 0) { 127 LOGE("Could not get audioflinger"); 128 return NO_INIT; 129 } 130 int afSampleRate; 131 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 132 return NO_INIT; 133 } 134 int afFrameCount; 135 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 136 return NO_INIT; 137 } 138 uint32_t afLatency; 139 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 140 return NO_INIT; 141 } 142 143 // handle default values first. 144 if (streamType == AudioSystem::DEFAULT) { 145 streamType = AudioSystem::MUSIC; 146 } 147 if (sampleRate == 0) { 148 sampleRate = afSampleRate; 149 } 150 // these below should probably come from the audioFlinger too... 151 if (format == 0) { 152 format = AudioSystem::PCM_16_BIT; 153 } 154 if (channelCount == 0) { 155 channelCount = 2; 156 } 157 158 // validate parameters 159 if (((format != AudioSystem::PCM_8_BIT) || sharedBuffer != 0) && 160 (format != AudioSystem::PCM_16_BIT)) { 161 LOGE("Invalid format"); 162 return BAD_VALUE; 163 } 164 if (channelCount != 1 && channelCount != 2) { 165 LOGE("Invalid channel number"); 166 return BAD_VALUE; 167 } 168 169 // Ensure that buffer depth covers at least audio hardware latency 170 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 171 if (minBufCount < 2) minBufCount = 2; 172 173 // When playing from shared buffer, playback will start even if last audioflinger 174 // block is partly filled. 175 if (sharedBuffer != 0 && minBufCount > 1) { 176 minBufCount--; 177 } 178 179 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 180 181 if (sharedBuffer == 0) { 182 if (frameCount == 0) { 183 frameCount = minFrameCount; 184 } 185 if (notificationFrames == 0) { 186 notificationFrames = frameCount/2; 187 } 188 // Make sure that application is notified with sufficient margin 189 // before underrun 190 if (notificationFrames > frameCount/2) { 191 notificationFrames = frameCount/2; 192 } 193 } else { 194 // Ensure that buffer alignment matches channelcount 195 if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) { 196 LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount); 197 return BAD_VALUE; 198 } 199 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 200 } 201 202 if (frameCount < minFrameCount) { 203 LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount); 204 return BAD_VALUE; 205 } 206 207 // create the track 208 status_t status; 209 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 210 streamType, sampleRate, format, channelCount, frameCount, flags, sharedBuffer, &status); 211 212 if (track == 0) { 213 LOGE("AudioFlinger could not create track, status: %d", status); 214 return status; 215 } 216 sp<IMemory> cblk = track->getCblk(); 217 if (cblk == 0) { 218 LOGE("Could not get control block"); 219 return NO_INIT; 220 } 221 if (cbf != 0) { 222 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 223 if (mAudioTrackThread == 0) { 224 LOGE("Could not create callback thread"); 225 return NO_INIT; 226 } 227 } 228 229 mStatus = NO_ERROR; 230 231 mAudioFlinger = audioFlinger; 232 mAudioTrack = track; 233 mCblkMemory = cblk; 234 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 235 mCblk->out = 1; 236 // Update buffer size in case it has been limited by AudioFlinger during track creation 237 mFrameCount = mCblk->frameCount; 238 if (sharedBuffer == 0) { 239 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 240 } else { 241 mCblk->buffers = sharedBuffer->pointer(); 242 // Force buffer full condition as data is already present in shared memory 243 mCblk->stepUser(mFrameCount); 244 } 245 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 246 mVolume[LEFT] = 1.0f; 247 mVolume[RIGHT] = 1.0f; 248 mSampleRate = sampleRate; 249 mStreamType = streamType; 250 mFormat = format; 251 mChannelCount = channelCount; 252 mSharedBuffer = sharedBuffer; 253 mMuted = false; 254 mActive = 0; 255 mCbf = cbf; 256 mNotificationFrames = notificationFrames; 257 mRemainingFrames = notificationFrames; 258 mUserData = user; 259 mLatency = afLatency + (1000*mFrameCount) / mSampleRate; 260 mLoopCount = 0; 261 mMarkerPosition = 0; 262 mNewPosition = 0; 263 mUpdatePeriod = 0; 264 265 return NO_ERROR; 266} 267 268status_t AudioTrack::initCheck() const 269{ 270 return mStatus; 271} 272 273// ------------------------------------------------------------------------- 274 275uint32_t AudioTrack::latency() const 276{ 277 return mLatency; 278} 279 280int AudioTrack::streamType() const 281{ 282 return mStreamType; 283} 284 285uint32_t AudioTrack::sampleRate() const 286{ 287 return mSampleRate; 288} 289 290int AudioTrack::format() const 291{ 292 return mFormat; 293} 294 295int AudioTrack::channelCount() const 296{ 297 return mChannelCount; 298} 299 300uint32_t AudioTrack::frameCount() const 301{ 302 return mFrameCount; 303} 304 305int AudioTrack::frameSize() const 306{ 307 return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t)); 308} 309 310sp<IMemory>& AudioTrack::sharedBuffer() 311{ 312 return mSharedBuffer; 313} 314 315// ------------------------------------------------------------------------- 316 317void AudioTrack::start() 318{ 319 sp<AudioTrackThread> t = mAudioTrackThread; 320 321 LOGV("start %p", this); 322 if (t != 0) { 323 if (t->exitPending()) { 324 if (t->requestExitAndWait() == WOULD_BLOCK) { 325 LOGE("AudioTrack::start called from thread"); 326 return; 327 } 328 } 329 t->mLock.lock(); 330 } 331 332 if (android_atomic_or(1, &mActive) == 0) { 333 mNewPosition = mCblk->server + mUpdatePeriod; 334 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 335 mCblk->waitTimeMs = 0; 336 if (t != 0) { 337 t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT); 338 } else { 339 setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT); 340 } 341 mAudioTrack->start(); 342 } 343 344 if (t != 0) { 345 t->mLock.unlock(); 346 } 347} 348 349void AudioTrack::stop() 350{ 351 sp<AudioTrackThread> t = mAudioTrackThread; 352 353 LOGV("stop %p", this); 354 if (t != 0) { 355 t->mLock.lock(); 356 } 357 358 if (android_atomic_and(~1, &mActive) == 1) { 359 mAudioTrack->stop(); 360 // Cancel loops (If we are in the middle of a loop, playback 361 // would not stop until loopCount reaches 0). 362 setLoop(0, 0, 0); 363 // Force flush if a shared buffer is used otherwise audioflinger 364 // will not stop before end of buffer is reached. 365 if (mSharedBuffer != 0) { 366 flush(); 367 } 368 if (t != 0) { 369 t->requestExit(); 370 } else { 371 setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); 372 } 373 } 374 375 if (t != 0) { 376 t->mLock.unlock(); 377 } 378} 379 380bool AudioTrack::stopped() const 381{ 382 return !mActive; 383} 384 385void AudioTrack::flush() 386{ 387 LOGV("flush"); 388 389 if (!mActive) { 390 mAudioTrack->flush(); 391 // Release AudioTrack callback thread in case it was waiting for new buffers 392 // in AudioTrack::obtainBuffer() 393 mCblk->cv.signal(); 394 } 395} 396 397void AudioTrack::pause() 398{ 399 LOGV("pause"); 400 if (android_atomic_and(~1, &mActive) == 1) { 401 mActive = 0; 402 mAudioTrack->pause(); 403 } 404} 405 406void AudioTrack::mute(bool e) 407{ 408 mAudioTrack->mute(e); 409 mMuted = e; 410} 411 412bool AudioTrack::muted() const 413{ 414 return mMuted; 415} 416 417void AudioTrack::setVolume(float left, float right) 418{ 419 mVolume[LEFT] = left; 420 mVolume[RIGHT] = right; 421 422 // write must be atomic 423 mCblk->volumeLR = (int32_t(int16_t(left * 0x1000)) << 16) | int16_t(right * 0x1000); 424} 425 426void AudioTrack::getVolume(float* left, float* right) 427{ 428 *left = mVolume[LEFT]; 429 *right = mVolume[RIGHT]; 430} 431 432void AudioTrack::setSampleRate(int rate) 433{ 434 int afSamplingRate; 435 436 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 437 return; 438 } 439 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 440 if (rate <= 0) rate = 1; 441 if (rate > afSamplingRate*2) rate = afSamplingRate*2; 442 if (rate > MAX_SAMPLE_RATE) rate = MAX_SAMPLE_RATE; 443 444 mCblk->sampleRate = (uint16_t)rate; 445} 446 447uint32_t AudioTrack::getSampleRate() 448{ 449 return uint32_t(mCblk->sampleRate); 450} 451 452status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 453{ 454 audio_track_cblk_t* cblk = mCblk; 455 456 457 Mutex::Autolock _l(cblk->lock); 458 459 if (loopCount == 0) { 460 cblk->loopStart = UINT_MAX; 461 cblk->loopEnd = UINT_MAX; 462 cblk->loopCount = 0; 463 mLoopCount = 0; 464 return NO_ERROR; 465 } 466 467 if (loopStart >= loopEnd || 468 loopEnd - loopStart > mFrameCount) { 469 LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user); 470 return BAD_VALUE; 471 } 472 473 if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) { 474 LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", 475 loopStart, loopEnd, mFrameCount); 476 return BAD_VALUE; 477 } 478 479 cblk->loopStart = loopStart; 480 cblk->loopEnd = loopEnd; 481 cblk->loopCount = loopCount; 482 mLoopCount = loopCount; 483 484 return NO_ERROR; 485} 486 487status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) 488{ 489 if (loopStart != 0) { 490 *loopStart = mCblk->loopStart; 491 } 492 if (loopEnd != 0) { 493 *loopEnd = mCblk->loopEnd; 494 } 495 if (loopCount != 0) { 496 if (mCblk->loopCount < 0) { 497 *loopCount = -1; 498 } else { 499 *loopCount = mCblk->loopCount; 500 } 501 } 502 503 return NO_ERROR; 504} 505 506status_t AudioTrack::setMarkerPosition(uint32_t marker) 507{ 508 if (mCbf == 0) return INVALID_OPERATION; 509 510 mMarkerPosition = marker; 511 512 return NO_ERROR; 513} 514 515status_t AudioTrack::getMarkerPosition(uint32_t *marker) 516{ 517 if (marker == 0) return BAD_VALUE; 518 519 *marker = mMarkerPosition; 520 521 return NO_ERROR; 522} 523 524status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 525{ 526 if (mCbf == 0) return INVALID_OPERATION; 527 528 uint32_t curPosition; 529 getPosition(&curPosition); 530 mNewPosition = curPosition + updatePeriod; 531 mUpdatePeriod = updatePeriod; 532 533 return NO_ERROR; 534} 535 536status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) 537{ 538 if (updatePeriod == 0) return BAD_VALUE; 539 540 *updatePeriod = mUpdatePeriod; 541 542 return NO_ERROR; 543} 544 545status_t AudioTrack::setPosition(uint32_t position) 546{ 547 Mutex::Autolock _l(mCblk->lock); 548 549 if (!stopped()) return INVALID_OPERATION; 550 551 if (position > mCblk->user) return BAD_VALUE; 552 553 mCblk->server = position; 554 mCblk->forceReady = 1; 555 556 return NO_ERROR; 557} 558 559status_t AudioTrack::getPosition(uint32_t *position) 560{ 561 if (position == 0) return BAD_VALUE; 562 563 *position = mCblk->server; 564 565 return NO_ERROR; 566} 567 568status_t AudioTrack::reload() 569{ 570 if (!stopped()) return INVALID_OPERATION; 571 572 flush(); 573 574 mCblk->stepUser(mFrameCount); 575 576 return NO_ERROR; 577} 578 579// ------------------------------------------------------------------------- 580 581status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 582{ 583 int active; 584 int timeout = 0; 585 status_t result; 586 audio_track_cblk_t* cblk = mCblk; 587 uint32_t framesReq = audioBuffer->frameCount; 588 589 audioBuffer->frameCount = 0; 590 audioBuffer->size = 0; 591 592 uint32_t framesAvail = cblk->framesAvailable(); 593 594 if (framesAvail == 0) { 595 Mutex::Autolock _l(cblk->lock); 596 goto start_loop_here; 597 while (framesAvail == 0) { 598 active = mActive; 599 if (UNLIKELY(!active)) { 600 LOGV("Not active and NO_MORE_BUFFERS"); 601 return NO_MORE_BUFFERS; 602 } 603 if (UNLIKELY(!waitCount)) 604 return WOULD_BLOCK; 605 timeout = 0; 606 result = cblk->cv.waitRelative(cblk->lock, milliseconds(WAIT_PERIOD_MS)); 607 if (__builtin_expect(result!=NO_ERROR, false)) { 608 cblk->waitTimeMs += WAIT_PERIOD_MS; 609 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 610 // timing out when a loop has been set and we have already written upto loop end 611 // is a normal condition: no need to wake AudioFlinger up. 612 if (cblk->user < cblk->loopEnd) { 613 LOGW( "obtainBuffer timed out (is the CPU pegged?) %p " 614 "user=%08x, server=%08x", this, cblk->user, cblk->server); 615 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 616 cblk->lock.unlock(); 617 mAudioTrack->start(); 618 cblk->lock.lock(); 619 timeout = 1; 620 } 621 cblk->waitTimeMs = 0; 622 } 623 624 if (--waitCount == 0) { 625 return TIMED_OUT; 626 } 627 } 628 // read the server count again 629 start_loop_here: 630 framesAvail = cblk->framesAvailable_l(); 631 } 632 } 633 634 cblk->waitTimeMs = 0; 635 636 if (framesReq > framesAvail) { 637 framesReq = framesAvail; 638 } 639 640 uint32_t u = cblk->user; 641 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 642 643 if (u + framesReq > bufferEnd) { 644 framesReq = bufferEnd - u; 645 } 646 647 LOGW_IF(timeout, 648 "*** SERIOUS WARNING *** obtainBuffer() timed out " 649 "but didn't need to be locked. We recovered, but " 650 "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server); 651 652 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 653 audioBuffer->channelCount= mChannelCount; 654 audioBuffer->format = AudioSystem::PCM_16_BIT; 655 audioBuffer->frameCount = framesReq; 656 audioBuffer->size = framesReq*mChannelCount*sizeof(int16_t); 657 audioBuffer->raw = (int8_t *)cblk->buffer(u); 658 active = mActive; 659 return active ? status_t(NO_ERROR) : status_t(STOPPED); 660} 661 662void AudioTrack::releaseBuffer(Buffer* audioBuffer) 663{ 664 audio_track_cblk_t* cblk = mCblk; 665 cblk->stepUser(audioBuffer->frameCount); 666} 667 668// ------------------------------------------------------------------------- 669 670ssize_t AudioTrack::write(const void* buffer, size_t userSize) 671{ 672 673 if (mSharedBuffer != 0) return INVALID_OPERATION; 674 675 if (ssize_t(userSize) < 0) { 676 // sanity-check. user is most-likely passing an error code. 677 LOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 678 buffer, userSize, userSize); 679 return BAD_VALUE; 680 } 681 682 LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 683 684 ssize_t written = 0; 685 const int8_t *src = (const int8_t *)buffer; 686 Buffer audioBuffer; 687 688 do { 689 audioBuffer.frameCount = userSize/mChannelCount; 690 if (mFormat == AudioSystem::PCM_16_BIT) { 691 audioBuffer.frameCount >>= 1; 692 } 693 // Calling obtainBuffer() with a negative wait count causes 694 // an (almost) infinite wait time. 695 status_t err = obtainBuffer(&audioBuffer, -1); 696 if (err < 0) { 697 // out of buffers, return #bytes written 698 if (err == status_t(NO_MORE_BUFFERS)) 699 break; 700 return ssize_t(err); 701 } 702 703 size_t toWrite; 704 if (mFormat == AudioSystem::PCM_8_BIT) { 705 // Divide capacity by 2 to take expansion into account 706 toWrite = audioBuffer.size>>1; 707 // 8 to 16 bit conversion 708 int count = toWrite; 709 int16_t *dst = (int16_t *)(audioBuffer.i8); 710 while(count--) { 711 *dst++ = (int16_t)(*src++^0x80) << 8; 712 } 713 }else { 714 toWrite = audioBuffer.size; 715 memcpy(audioBuffer.i8, src, toWrite); 716 src += toWrite; 717 } 718 userSize -= toWrite; 719 written += toWrite; 720 721 releaseBuffer(&audioBuffer); 722 } while (userSize); 723 724 return written; 725} 726 727// ------------------------------------------------------------------------- 728 729bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 730{ 731 Buffer audioBuffer; 732 uint32_t frames; 733 size_t writtenSize; 734 735 // Manage underrun callback 736 if (mActive && (mCblk->framesReady() == 0)) { 737 LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag); 738 if (mCblk->flowControlFlag == 0) { 739 mCbf(EVENT_UNDERRUN, mUserData, 0); 740 if (mCblk->server == mCblk->frameCount) { 741 mCbf(EVENT_BUFFER_END, mUserData, 0); 742 } 743 mCblk->flowControlFlag = 1; 744 if (mSharedBuffer != 0) return false; 745 } 746 } 747 748 // Manage loop end callback 749 while (mLoopCount > mCblk->loopCount) { 750 int loopCount = -1; 751 mLoopCount--; 752 if (mLoopCount >= 0) loopCount = mLoopCount; 753 754 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 755 } 756 757 // Manage marker callback 758 if(mMarkerPosition > 0) { 759 if (mCblk->server >= mMarkerPosition) { 760 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 761 mMarkerPosition = 0; 762 } 763 } 764 765 // Manage new position callback 766 if(mUpdatePeriod > 0) { 767 while (mCblk->server >= mNewPosition) { 768 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 769 mNewPosition += mUpdatePeriod; 770 } 771 } 772 773 // If Shared buffer is used, no data is requested from client. 774 if (mSharedBuffer != 0) { 775 frames = 0; 776 } else { 777 frames = mRemainingFrames; 778 } 779 780 do { 781 782 audioBuffer.frameCount = frames; 783 784 // Calling obtainBuffer() with a wait count of 1 785 // limits wait time to WAIT_PERIOD_MS. This prevents from being 786 // stuck here not being able to handle timed events (position, markers, loops). 787 status_t err = obtainBuffer(&audioBuffer, 1); 788 if (err < NO_ERROR) { 789 if (err != TIMED_OUT) { 790 LOGE("Error obtaining an audio buffer, giving up."); 791 return false; 792 } 793 break; 794 } 795 if (err == status_t(STOPPED)) return false; 796 797 // Divide buffer size by 2 to take into account the expansion 798 // due to 8 to 16 bit conversion: the callback must fill only half 799 // of the destination buffer 800 if (mFormat == AudioSystem::PCM_8_BIT) { 801 audioBuffer.size >>= 1; 802 } 803 804 size_t reqSize = audioBuffer.size; 805 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 806 writtenSize = audioBuffer.size; 807 808 // Sanity check on returned size 809 if (ssize_t(writtenSize) <= 0) { 810 // The callback is done filling buffers 811 // Keep this thread going to handle timed events and 812 // still try to get more data in intervals of WAIT_PERIOD_MS 813 // but don't just loop and block the CPU, so wait 814 usleep(WAIT_PERIOD_MS*1000); 815 break; 816 } 817 if (writtenSize > reqSize) writtenSize = reqSize; 818 819 if (mFormat == AudioSystem::PCM_8_BIT) { 820 // 8 to 16 bit conversion 821 const int8_t *src = audioBuffer.i8 + writtenSize-1; 822 int count = writtenSize; 823 int16_t *dst = audioBuffer.i16 + writtenSize-1; 824 while(count--) { 825 *dst-- = (int16_t)(*src--^0x80) << 8; 826 } 827 writtenSize <<= 1; 828 } 829 830 audioBuffer.size = writtenSize; 831 audioBuffer.frameCount = writtenSize/mChannelCount/sizeof(int16_t); 832 frames -= audioBuffer.frameCount; 833 834 releaseBuffer(&audioBuffer); 835 } 836 while (frames); 837 838 if (frames == 0) { 839 mRemainingFrames = mNotificationFrames; 840 } else { 841 mRemainingFrames = frames; 842 } 843 return true; 844} 845 846status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 847{ 848 849 const size_t SIZE = 256; 850 char buffer[SIZE]; 851 String8 result; 852 853 result.append(" AudioTrack::dump\n"); 854 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); 855 result.append(buffer); 856 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount); 857 result.append(buffer); 858 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", mSampleRate, mStatus, mMuted); 859 result.append(buffer); 860 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 861 result.append(buffer); 862 ::write(fd, result.string(), result.size()); 863 return NO_ERROR; 864} 865 866// ========================================================================= 867 868AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 869 : Thread(bCanCallJava), mReceiver(receiver) 870{ 871} 872 873bool AudioTrack::AudioTrackThread::threadLoop() 874{ 875 return mReceiver.processAudioBuffer(this); 876} 877 878status_t AudioTrack::AudioTrackThread::readyToRun() 879{ 880 return NO_ERROR; 881} 882 883void AudioTrack::AudioTrackThread::onFirstRef() 884{ 885} 886 887// ========================================================================= 888 889audio_track_cblk_t::audio_track_cblk_t() 890 : user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0), 891 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), flowControlFlag(1), forceReady(0) 892{ 893} 894 895uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 896{ 897 uint32_t u = this->user; 898 899 u += frameCount; 900 // Ensure that user is never ahead of server for AudioRecord 901 if (out) { 902 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 903 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 904 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 905 } 906 } else if (u > this->server) { 907 LOGW("stepServer occured after track reset"); 908 u = this->server; 909 } 910 911 if (u >= userBase + this->frameCount) { 912 userBase += this->frameCount; 913 } 914 915 this->user = u; 916 917 // Clear flow control error condition as new data has been written/read to/from buffer. 918 flowControlFlag = 0; 919 920 return u; 921} 922 923bool audio_track_cblk_t::stepServer(uint32_t frameCount) 924{ 925 // the code below simulates lock-with-timeout 926 // we MUST do this to protect the AudioFlinger server 927 // as this lock is shared with the client. 928 status_t err; 929 930 err = lock.tryLock(); 931 if (err == -EBUSY) { // just wait a bit 932 usleep(1000); 933 err = lock.tryLock(); 934 } 935 if (err != NO_ERROR) { 936 // probably, the client just died. 937 return false; 938 } 939 940 uint32_t s = this->server; 941 942 s += frameCount; 943 if (out) { 944 // Mark that we have read the first buffer so that next time stepUser() is called 945 // we switch to normal obtainBuffer() timeout period 946 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 947 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1; 948 } 949 // It is possible that we receive a flush() 950 // while the mixer is processing a block: in this case, 951 // stepServer() is called After the flush() has reset u & s and 952 // we have s > u 953 if (s > this->user) { 954 LOGW("stepServer occured after track reset"); 955 s = this->user; 956 } 957 } 958 959 if (s >= loopEnd) { 960 LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 961 s = loopStart; 962 if (--loopCount == 0) { 963 loopEnd = UINT_MAX; 964 loopStart = UINT_MAX; 965 } 966 } 967 if (s >= serverBase + this->frameCount) { 968 serverBase += this->frameCount; 969 } 970 971 this->server = s; 972 973 cv.signal(); 974 lock.unlock(); 975 return true; 976} 977 978void* audio_track_cblk_t::buffer(uint32_t offset) const 979{ 980 return (int16_t *)this->buffers + (offset-userBase)*this->channels; 981} 982 983uint32_t audio_track_cblk_t::framesAvailable() 984{ 985 Mutex::Autolock _l(lock); 986 return framesAvailable_l(); 987} 988 989uint32_t audio_track_cblk_t::framesAvailable_l() 990{ 991 uint32_t u = this->user; 992 uint32_t s = this->server; 993 994 if (out) { 995 uint32_t limit = (s < loopStart) ? s : loopStart; 996 return limit + frameCount - u; 997 } else { 998 return frameCount + u - s; 999 } 1000} 1001 1002uint32_t audio_track_cblk_t::framesReady() 1003{ 1004 uint32_t u = this->user; 1005 uint32_t s = this->server; 1006 1007 if (out) { 1008 if (u < loopEnd) { 1009 return u - s; 1010 } else { 1011 Mutex::Autolock _l(lock); 1012 if (loopCount >= 0) { 1013 return (loopEnd - loopStart)*loopCount + u - s; 1014 } else { 1015 return UINT_MAX; 1016 } 1017 } 1018 } else { 1019 return s - u; 1020 } 1021} 1022 1023// ------------------------------------------------------------------------- 1024 1025}; // namespace android 1026 1027