1 /* 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 <binder/Parcel.h> 36 #include <binder/IPCThreadState.h> 37 #include <utils/Timers.h> 38 #include <utils/Atomic.h> 39 40 #include <cutils/bitops.h> 41 #include <cutils/compiler.h> 42 43 #include <system/audio.h> 44 #include <system/audio_policy.h> 45 46 #include <audio_utils/primitives.h> 47 48 namespace android { 49 // --------------------------------------------------------------------------- 50 51 // static 52 status_t AudioTrack::getMinFrameCount( 53 int* frameCount, 54 audio_stream_type_t streamType, 55 uint32_t sampleRate) 56 { 57 // FIXME merge with similar code in createTrack_l(), except we're missing 58 // some information here that is available in createTrack_l(): 59 // audio_io_handle_t output 60 // audio_format_t format 61 // audio_channel_mask_t channelMask 62 // audio_output_flags_t flags 63 int afSampleRate; 64 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 65 return NO_INIT; 66 } 67 int afFrameCount; 68 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 69 return NO_INIT; 70 } 71 uint32_t afLatency; 72 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 73 return NO_INIT; 74 } 75 76 // Ensure that buffer depth covers at least audio hardware latency 77 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 78 if (minBufCount < 2) minBufCount = 2; 79 80 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 81 afFrameCount * minBufCount * sampleRate / afSampleRate; 82 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 83 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 84 return NO_ERROR; 85 } 86 87 // --------------------------------------------------------------------------- 88 89 AudioTrack::AudioTrack() 90 : mStatus(NO_INIT), 91 mIsTimed(false), 92 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 93 mPreviousSchedulingGroup(SP_DEFAULT) 94 { 95 } 96 97 AudioTrack::AudioTrack( 98 audio_stream_type_t streamType, 99 uint32_t sampleRate, 100 audio_format_t format, 101 int channelMask, 102 int frameCount, 103 audio_output_flags_t flags, 104 callback_t cbf, 105 void* user, 106 int notificationFrames, 107 int sessionId) 108 : mStatus(NO_INIT), 109 mIsTimed(false), 110 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 111 mPreviousSchedulingGroup(SP_DEFAULT) 112 { 113 mStatus = set(streamType, sampleRate, format, channelMask, 114 frameCount, flags, cbf, user, notificationFrames, 115 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); 116 } 117 118 // DEPRECATED 119 AudioTrack::AudioTrack( 120 int streamType, 121 uint32_t sampleRate, 122 int format, 123 int channelMask, 124 int frameCount, 125 uint32_t flags, 126 callback_t cbf, 127 void* user, 128 int notificationFrames, 129 int sessionId) 130 : mStatus(NO_INIT), 131 mIsTimed(false), 132 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) 133 { 134 mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask, 135 frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames, 136 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); 137 } 138 139 AudioTrack::AudioTrack( 140 audio_stream_type_t streamType, 141 uint32_t sampleRate, 142 audio_format_t format, 143 int channelMask, 144 const sp<IMemory>& sharedBuffer, 145 audio_output_flags_t flags, 146 callback_t cbf, 147 void* user, 148 int notificationFrames, 149 int sessionId) 150 : mStatus(NO_INIT), 151 mIsTimed(false), 152 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 153 mPreviousSchedulingGroup(SP_DEFAULT) 154 { 155 mStatus = set(streamType, sampleRate, format, channelMask, 156 0 /*frameCount*/, flags, cbf, user, notificationFrames, 157 sharedBuffer, false /*threadCanCallJava*/, sessionId); 158 } 159 160 AudioTrack::~AudioTrack() 161 { 162 ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 163 164 if (mStatus == NO_ERROR) { 165 // Make sure that callback function exits in the case where 166 // it is looping on buffer full condition in obtainBuffer(). 167 // Otherwise the callback thread will never exit. 168 stop(); 169 if (mAudioTrackThread != 0) { 170 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 171 mAudioTrackThread->requestExitAndWait(); 172 mAudioTrackThread.clear(); 173 } 174 mAudioTrack.clear(); 175 IPCThreadState::self()->flushCommands(); 176 AudioSystem::releaseAudioSessionId(mSessionId); 177 } 178 } 179 180 status_t AudioTrack::set( 181 audio_stream_type_t streamType, 182 uint32_t sampleRate, 183 audio_format_t format, 184 int channelMask, 185 int frameCount, 186 audio_output_flags_t flags, 187 callback_t cbf, 188 void* user, 189 int notificationFrames, 190 const sp<IMemory>& sharedBuffer, 191 bool threadCanCallJava, 192 int sessionId) 193 { 194 195 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 196 197 ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags); 198 199 AutoMutex lock(mLock); 200 if (mAudioTrack != 0) { 201 ALOGE("Track already in use"); 202 return INVALID_OPERATION; 203 } 204 205 // handle default values first. 206 if (streamType == AUDIO_STREAM_DEFAULT) { 207 streamType = AUDIO_STREAM_MUSIC; 208 } 209 210 if (sampleRate == 0) { 211 int afSampleRate; 212 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 213 return NO_INIT; 214 } 215 sampleRate = afSampleRate; 216 } 217 218 // these below should probably come from the audioFlinger too... 219 if (format == AUDIO_FORMAT_DEFAULT) { 220 format = AUDIO_FORMAT_PCM_16_BIT; 221 } 222 if (channelMask == 0) { 223 channelMask = AUDIO_CHANNEL_OUT_STEREO; 224 } 225 226 // validate parameters 227 if (!audio_is_valid_format(format)) { 228 ALOGE("Invalid format"); 229 return BAD_VALUE; 230 } 231 232 // AudioFlinger does not currently support 8-bit data in shared memory 233 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 234 ALOGE("8-bit data in shared memory is not supported"); 235 return BAD_VALUE; 236 } 237 238 // force direct flag if format is not linear PCM 239 if (!audio_is_linear_pcm(format)) { 240 flags = (audio_output_flags_t) 241 // FIXME why can't we allow direct AND fast? 242 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 243 } 244 // only allow deep buffering for music stream type 245 if (streamType != AUDIO_STREAM_MUSIC) { 246 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 247 } 248 249 if (!audio_is_output_channel(channelMask)) { 250 ALOGE("Invalid channel mask"); 251 return BAD_VALUE; 252 } 253 uint32_t channelCount = popcount(channelMask); 254 255 audio_io_handle_t output = AudioSystem::getOutput( 256 streamType, 257 sampleRate, format, channelMask, 258 flags); 259 260 if (output == 0) { 261 ALOGE("Could not get audio output for stream type %d", streamType); 262 return BAD_VALUE; 263 } 264 265 mVolume[LEFT] = 1.0f; 266 mVolume[RIGHT] = 1.0f; 267 mSendLevel = 0.0f; 268 mFrameCount = frameCount; 269 mNotificationFramesReq = notificationFrames; 270 mSessionId = sessionId; 271 mAuxEffectId = 0; 272 mFlags = flags; 273 mCbf = cbf; 274 275 if (cbf != NULL) { 276 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 277 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 278 } 279 280 // create the IAudioTrack 281 status_t status = createTrack_l(streamType, 282 sampleRate, 283 format, 284 (uint32_t)channelMask, 285 frameCount, 286 flags, 287 sharedBuffer, 288 output); 289 290 if (status != NO_ERROR) { 291 if (mAudioTrackThread != 0) { 292 mAudioTrackThread->requestExit(); 293 mAudioTrackThread.clear(); 294 } 295 return status; 296 } 297 298 mStatus = NO_ERROR; 299 300 mStreamType = streamType; 301 mFormat = format; 302 mChannelMask = (uint32_t)channelMask; 303 mChannelCount = channelCount; 304 mSharedBuffer = sharedBuffer; 305 mMuted = false; 306 mActive = false; 307 mUserData = user; 308 mLoopCount = 0; 309 mMarkerPosition = 0; 310 mMarkerReached = false; 311 mNewPosition = 0; 312 mUpdatePeriod = 0; 313 mFlushed = false; 314 AudioSystem::acquireAudioSessionId(mSessionId); 315 mRestoreStatus = NO_ERROR; 316 return NO_ERROR; 317 } 318 319 status_t AudioTrack::initCheck() const 320 { 321 return mStatus; 322 } 323 324 // ------------------------------------------------------------------------- 325 326 uint32_t AudioTrack::latency() const 327 { 328 return mLatency; 329 } 330 331 audio_stream_type_t AudioTrack::streamType() const 332 { 333 return mStreamType; 334 } 335 336 audio_format_t AudioTrack::format() const 337 { 338 return mFormat; 339 } 340 341 int AudioTrack::channelCount() const 342 { 343 return mChannelCount; 344 } 345 346 uint32_t AudioTrack::frameCount() const 347 { 348 return mCblk->frameCount; 349 } 350 351 size_t AudioTrack::frameSize() const 352 { 353 if (audio_is_linear_pcm(mFormat)) { 354 return channelCount()*audio_bytes_per_sample(mFormat); 355 } else { 356 return sizeof(uint8_t); 357 } 358 } 359 360 sp<IMemory>& AudioTrack::sharedBuffer() 361 { 362 return mSharedBuffer; 363 } 364 365 // ------------------------------------------------------------------------- 366 367 void AudioTrack::start() 368 { 369 sp<AudioTrackThread> t = mAudioTrackThread; 370 status_t status = NO_ERROR; 371 372 ALOGV("start %p", this); 373 374 AutoMutex lock(mLock); 375 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 376 // while we are accessing the cblk 377 sp<IAudioTrack> audioTrack = mAudioTrack; 378 sp<IMemory> iMem = mCblkMemory; 379 audio_track_cblk_t* cblk = mCblk; 380 381 if (!mActive) { 382 mFlushed = false; 383 mActive = true; 384 mNewPosition = cblk->server + mUpdatePeriod; 385 cblk->lock.lock(); 386 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 387 cblk->waitTimeMs = 0; 388 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 389 if (t != 0) { 390 t->resume(); 391 } else { 392 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 393 get_sched_policy(0, &mPreviousSchedulingGroup); 394 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 395 } 396 397 ALOGV("start %p before lock cblk %p", this, mCblk); 398 if (!(cblk->flags & CBLK_INVALID_MSK)) { 399 cblk->lock.unlock(); 400 ALOGV("mAudioTrack->start()"); 401 status = mAudioTrack->start(); 402 cblk->lock.lock(); 403 if (status == DEAD_OBJECT) { 404 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 405 } 406 } 407 if (cblk->flags & CBLK_INVALID_MSK) { 408 status = restoreTrack_l(cblk, true); 409 } 410 cblk->lock.unlock(); 411 if (status != NO_ERROR) { 412 ALOGV("start() failed"); 413 mActive = false; 414 if (t != 0) { 415 t->pause(); 416 } else { 417 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 418 set_sched_policy(0, mPreviousSchedulingGroup); 419 } 420 } 421 } 422 423 } 424 425 void AudioTrack::stop() 426 { 427 sp<AudioTrackThread> t = mAudioTrackThread; 428 429 ALOGV("stop %p", this); 430 431 AutoMutex lock(mLock); 432 if (mActive) { 433 mActive = false; 434 mCblk->cv.signal(); 435 mAudioTrack->stop(); 436 // Cancel loops (If we are in the middle of a loop, playback 437 // would not stop until loopCount reaches 0). 438 setLoop_l(0, 0, 0); 439 // the playback head position will reset to 0, so if a marker is set, we need 440 // to activate it again 441 mMarkerReached = false; 442 // Force flush if a shared buffer is used otherwise audioflinger 443 // will not stop before end of buffer is reached. 444 if (mSharedBuffer != 0) { 445 flush_l(); 446 } 447 if (t != 0) { 448 t->pause(); 449 } else { 450 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 451 set_sched_policy(0, mPreviousSchedulingGroup); 452 } 453 } 454 455 } 456 457 bool AudioTrack::stopped() const 458 { 459 AutoMutex lock(mLock); 460 return stopped_l(); 461 } 462 463 void AudioTrack::flush() 464 { 465 AutoMutex lock(mLock); 466 flush_l(); 467 } 468 469 // must be called with mLock held 470 void AudioTrack::flush_l() 471 { 472 ALOGV("flush"); 473 474 // clear playback marker and periodic update counter 475 mMarkerPosition = 0; 476 mMarkerReached = false; 477 mUpdatePeriod = 0; 478 479 if (!mActive) { 480 mFlushed = true; 481 mAudioTrack->flush(); 482 // Release AudioTrack callback thread in case it was waiting for new buffers 483 // in AudioTrack::obtainBuffer() 484 mCblk->cv.signal(); 485 } 486 } 487 488 void AudioTrack::pause() 489 { 490 ALOGV("pause"); 491 AutoMutex lock(mLock); 492 if (mActive) { 493 mActive = false; 494 mCblk->cv.signal(); 495 mAudioTrack->pause(); 496 } 497 } 498 499 void AudioTrack::mute(bool e) 500 { 501 mAudioTrack->mute(e); 502 mMuted = e; 503 } 504 505 bool AudioTrack::muted() const 506 { 507 return mMuted; 508 } 509 510 status_t AudioTrack::setVolume(float left, float right) 511 { 512 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 513 return BAD_VALUE; 514 } 515 516 AutoMutex lock(mLock); 517 mVolume[LEFT] = left; 518 mVolume[RIGHT] = right; 519 520 mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 521 522 return NO_ERROR; 523 } 524 525 void AudioTrack::getVolume(float* left, float* right) const 526 { 527 if (left != NULL) { 528 *left = mVolume[LEFT]; 529 } 530 if (right != NULL) { 531 *right = mVolume[RIGHT]; 532 } 533 } 534 535 status_t AudioTrack::setAuxEffectSendLevel(float level) 536 { 537 ALOGV("setAuxEffectSendLevel(%f)", level); 538 if (level < 0.0f || level > 1.0f) { 539 return BAD_VALUE; 540 } 541 AutoMutex lock(mLock); 542 543 mSendLevel = level; 544 545 mCblk->setSendLevel(level); 546 547 return NO_ERROR; 548 } 549 550 void AudioTrack::getAuxEffectSendLevel(float* level) const 551 { 552 if (level != NULL) { 553 *level = mSendLevel; 554 } 555 } 556 557 status_t AudioTrack::setSampleRate(int rate) 558 { 559 int afSamplingRate; 560 561 if (mIsTimed) { 562 return INVALID_OPERATION; 563 } 564 565 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 566 return NO_INIT; 567 } 568 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 569 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 570 571 AutoMutex lock(mLock); 572 mCblk->sampleRate = rate; 573 return NO_ERROR; 574 } 575 576 uint32_t AudioTrack::getSampleRate() const 577 { 578 if (mIsTimed) { 579 return INVALID_OPERATION; 580 } 581 582 AutoMutex lock(mLock); 583 return mCblk->sampleRate; 584 } 585 586 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 587 { 588 AutoMutex lock(mLock); 589 return setLoop_l(loopStart, loopEnd, loopCount); 590 } 591 592 // must be called with mLock held 593 status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 594 { 595 audio_track_cblk_t* cblk = mCblk; 596 597 Mutex::Autolock _l(cblk->lock); 598 599 if (loopCount == 0) { 600 cblk->loopStart = UINT_MAX; 601 cblk->loopEnd = UINT_MAX; 602 cblk->loopCount = 0; 603 mLoopCount = 0; 604 return NO_ERROR; 605 } 606 607 if (mIsTimed) { 608 return INVALID_OPERATION; 609 } 610 611 if (loopStart >= loopEnd || 612 loopEnd - loopStart > cblk->frameCount || 613 cblk->server > loopStart) { 614 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 615 return BAD_VALUE; 616 } 617 618 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 619 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", 620 loopStart, loopEnd, cblk->frameCount); 621 return BAD_VALUE; 622 } 623 624 cblk->loopStart = loopStart; 625 cblk->loopEnd = loopEnd; 626 cblk->loopCount = loopCount; 627 mLoopCount = loopCount; 628 629 return NO_ERROR; 630 } 631 632 status_t AudioTrack::setMarkerPosition(uint32_t marker) 633 { 634 if (mCbf == NULL) return INVALID_OPERATION; 635 636 mMarkerPosition = marker; 637 mMarkerReached = false; 638 639 return NO_ERROR; 640 } 641 642 status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 643 { 644 if (marker == NULL) return BAD_VALUE; 645 646 *marker = mMarkerPosition; 647 648 return NO_ERROR; 649 } 650 651 status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 652 { 653 if (mCbf == NULL) return INVALID_OPERATION; 654 655 uint32_t curPosition; 656 getPosition(&curPosition); 657 mNewPosition = curPosition + updatePeriod; 658 mUpdatePeriod = updatePeriod; 659 660 return NO_ERROR; 661 } 662 663 status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 664 { 665 if (updatePeriod == NULL) return BAD_VALUE; 666 667 *updatePeriod = mUpdatePeriod; 668 669 return NO_ERROR; 670 } 671 672 status_t AudioTrack::setPosition(uint32_t position) 673 { 674 if (mIsTimed) return INVALID_OPERATION; 675 676 AutoMutex lock(mLock); 677 678 if (!stopped_l()) return INVALID_OPERATION; 679 680 Mutex::Autolock _l(mCblk->lock); 681 682 if (position > mCblk->user) return BAD_VALUE; 683 684 mCblk->server = position; 685 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 686 687 return NO_ERROR; 688 } 689 690 status_t AudioTrack::getPosition(uint32_t *position) 691 { 692 if (position == NULL) return BAD_VALUE; 693 AutoMutex lock(mLock); 694 *position = mFlushed ? 0 : mCblk->server; 695 696 return NO_ERROR; 697 } 698 699 status_t AudioTrack::reload() 700 { 701 AutoMutex lock(mLock); 702 703 if (!stopped_l()) return INVALID_OPERATION; 704 705 flush_l(); 706 707 mCblk->stepUser(mCblk->frameCount); 708 709 return NO_ERROR; 710 } 711 712 audio_io_handle_t AudioTrack::getOutput() 713 { 714 AutoMutex lock(mLock); 715 return getOutput_l(); 716 } 717 718 // must be called with mLock held 719 audio_io_handle_t AudioTrack::getOutput_l() 720 { 721 return AudioSystem::getOutput(mStreamType, 722 mCblk->sampleRate, mFormat, mChannelMask, mFlags); 723 } 724 725 int AudioTrack::getSessionId() const 726 { 727 return mSessionId; 728 } 729 730 status_t AudioTrack::attachAuxEffect(int effectId) 731 { 732 ALOGV("attachAuxEffect(%d)", effectId); 733 status_t status = mAudioTrack->attachAuxEffect(effectId); 734 if (status == NO_ERROR) { 735 mAuxEffectId = effectId; 736 } 737 return status; 738 } 739 740 // ------------------------------------------------------------------------- 741 742 // must be called with mLock held 743 status_t AudioTrack::createTrack_l( 744 audio_stream_type_t streamType, 745 uint32_t sampleRate, 746 audio_format_t format, 747 uint32_t channelMask, 748 int frameCount, 749 audio_output_flags_t flags, 750 const sp<IMemory>& sharedBuffer, 751 audio_io_handle_t output) 752 { 753 status_t status; 754 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 755 if (audioFlinger == 0) { 756 ALOGE("Could not get audioflinger"); 757 return NO_INIT; 758 } 759 760 uint32_t afLatency; 761 if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { 762 return NO_INIT; 763 } 764 765 // Client decides whether the track is TIMED (see below), but can only express a preference 766 // for FAST. Server will perform additional tests. 767 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 768 // either of these use cases: 769 // use case 1: shared buffer 770 (sharedBuffer != 0) || 771 // use case 2: callback handler 772 (mCbf != NULL))) { 773 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 774 // once denied, do not request again if IAudioTrack is re-created 775 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 776 mFlags = flags; 777 } 778 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 779 780 mNotificationFramesAct = mNotificationFramesReq; 781 782 if (!audio_is_linear_pcm(format)) { 783 784 if (sharedBuffer != 0) { 785 // Same comment as below about ignoring frameCount parameter for set() 786 frameCount = sharedBuffer->size(); 787 } else if (frameCount == 0) { 788 int afFrameCount; 789 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 790 return NO_INIT; 791 } 792 frameCount = afFrameCount; 793 } 794 795 } else if (sharedBuffer != 0) { 796 797 // Ensure that buffer alignment matches channelCount 798 int channelCount = popcount(channelMask); 799 // 8-bit data in shared memory is not currently supported by AudioFlinger 800 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 801 if (channelCount > 1) { 802 // More than 2 channels does not require stronger alignment than stereo 803 alignment <<= 1; 804 } 805 if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 806 ALOGE("Invalid buffer alignment: address %p, channelCount %d", 807 sharedBuffer->pointer(), channelCount); 808 return BAD_VALUE; 809 } 810 811 // When initializing a shared buffer AudioTrack via constructors, 812 // there's no frameCount parameter. 813 // But when initializing a shared buffer AudioTrack via set(), 814 // there _is_ a frameCount parameter. We silently ignore it. 815 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 816 817 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 818 819 // FIXME move these calculations and associated checks to server 820 int afSampleRate; 821 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 822 return NO_INIT; 823 } 824 int afFrameCount; 825 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 826 return NO_INIT; 827 } 828 829 // Ensure that buffer depth covers at least audio hardware latency 830 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 831 if (minBufCount < 2) minBufCount = 2; 832 833 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 834 ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d" 835 ", afLatency=%d", 836 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 837 838 if (frameCount == 0) { 839 frameCount = minFrameCount; 840 } 841 if (mNotificationFramesAct == 0) { 842 mNotificationFramesAct = frameCount/2; 843 } 844 // Make sure that application is notified with sufficient margin 845 // before underrun 846 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 847 mNotificationFramesAct = frameCount/2; 848 } 849 if (frameCount < minFrameCount) { 850 // not ALOGW because it happens all the time when playing key clicks over A2DP 851 ALOGV("Minimum buffer size corrected from %d to %d", 852 frameCount, minFrameCount); 853 frameCount = minFrameCount; 854 } 855 856 } else { 857 // For fast tracks, the frame count calculations and checks are done by server 858 } 859 860 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 861 if (mIsTimed) { 862 trackFlags |= IAudioFlinger::TRACK_TIMED; 863 } 864 865 pid_t tid = -1; 866 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 867 trackFlags |= IAudioFlinger::TRACK_FAST; 868 if (mAudioTrackThread != 0) { 869 tid = mAudioTrackThread->getTid(); 870 } 871 } 872 873 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 874 streamType, 875 sampleRate, 876 format, 877 channelMask, 878 frameCount, 879 trackFlags, 880 sharedBuffer, 881 output, 882 tid, 883 &mSessionId, 884 &status); 885 886 if (track == 0) { 887 ALOGE("AudioFlinger could not create track, status: %d", status); 888 return status; 889 } 890 sp<IMemory> cblk = track->getCblk(); 891 if (cblk == 0) { 892 ALOGE("Could not get control block"); 893 return NO_INIT; 894 } 895 mAudioTrack = track; 896 mCblkMemory = cblk; 897 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 898 // old has the previous value of mCblk->flags before the "or" operation 899 int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 900 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 901 if (old & CBLK_FAST) { 902 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount); 903 } else { 904 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount); 905 // once denied, do not request again if IAudioTrack is re-created 906 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 907 mFlags = flags; 908 } 909 if (sharedBuffer == 0) { 910 mNotificationFramesAct = mCblk->frameCount/2; 911 } 912 } 913 if (sharedBuffer == 0) { 914 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 915 } else { 916 mCblk->buffers = sharedBuffer->pointer(); 917 // Force buffer full condition as data is already present in shared memory 918 mCblk->stepUser(mCblk->frameCount); 919 } 920 921 mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); 922 mCblk->setSendLevel(mSendLevel); 923 mAudioTrack->attachAuxEffect(mAuxEffectId); 924 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 925 mCblk->waitTimeMs = 0; 926 mRemainingFrames = mNotificationFramesAct; 927 // FIXME don't believe this lie 928 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 929 // If IAudioTrack is re-created, don't let the requested frameCount 930 // decrease. This can confuse clients that cache frameCount(). 931 if (mCblk->frameCount > mFrameCount) { 932 mFrameCount = mCblk->frameCount; 933 } 934 return NO_ERROR; 935 } 936 937 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 938 { 939 AutoMutex lock(mLock); 940 bool active; 941 status_t result = NO_ERROR; 942 audio_track_cblk_t* cblk = mCblk; 943 uint32_t framesReq = audioBuffer->frameCount; 944 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 945 946 audioBuffer->frameCount = 0; 947 audioBuffer->size = 0; 948 949 uint32_t framesAvail = cblk->framesAvailable(); 950 951 cblk->lock.lock(); 952 if (cblk->flags & CBLK_INVALID_MSK) { 953 goto create_new_track; 954 } 955 cblk->lock.unlock(); 956 957 if (framesAvail == 0) { 958 cblk->lock.lock(); 959 goto start_loop_here; 960 while (framesAvail == 0) { 961 active = mActive; 962 if (CC_UNLIKELY(!active)) { 963 ALOGV("Not active and NO_MORE_BUFFERS"); 964 cblk->lock.unlock(); 965 return NO_MORE_BUFFERS; 966 } 967 if (CC_UNLIKELY(!waitCount)) { 968 cblk->lock.unlock(); 969 return WOULD_BLOCK; 970 } 971 if (!(cblk->flags & CBLK_INVALID_MSK)) { 972 mLock.unlock(); 973 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 974 cblk->lock.unlock(); 975 mLock.lock(); 976 if (!mActive) { 977 return status_t(STOPPED); 978 } 979 cblk->lock.lock(); 980 } 981 982 if (cblk->flags & CBLK_INVALID_MSK) { 983 goto create_new_track; 984 } 985 if (CC_UNLIKELY(result != NO_ERROR)) { 986 cblk->waitTimeMs += waitTimeMs; 987 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 988 // timing out when a loop has been set and we have already written upto loop end 989 // is a normal condition: no need to wake AudioFlinger up. 990 if (cblk->user < cblk->loopEnd) { 991 ALOGW( "obtainBuffer timed out (is the CPU pegged?) %p name=%#x" 992 "user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server); 993 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 994 cblk->lock.unlock(); 995 result = mAudioTrack->start(); 996 cblk->lock.lock(); 997 if (result == DEAD_OBJECT) { 998 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 999 create_new_track: 1000 result = restoreTrack_l(cblk, false); 1001 } 1002 if (result != NO_ERROR) { 1003 ALOGW("obtainBuffer create Track error %d", result); 1004 cblk->lock.unlock(); 1005 return result; 1006 } 1007 } 1008 cblk->waitTimeMs = 0; 1009 } 1010 1011 if (--waitCount == 0) { 1012 cblk->lock.unlock(); 1013 return TIMED_OUT; 1014 } 1015 } 1016 // read the server count again 1017 start_loop_here: 1018 framesAvail = cblk->framesAvailable_l(); 1019 } 1020 cblk->lock.unlock(); 1021 } 1022 1023 cblk->waitTimeMs = 0; 1024 1025 if (framesReq > framesAvail) { 1026 framesReq = framesAvail; 1027 } 1028 1029 uint32_t u = cblk->user; 1030 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 1031 1032 if (framesReq > bufferEnd - u) { 1033 framesReq = bufferEnd - u; 1034 } 1035 1036 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 1037 audioBuffer->channelCount = mChannelCount; 1038 audioBuffer->frameCount = framesReq; 1039 audioBuffer->size = framesReq * cblk->frameSize; 1040 if (audio_is_linear_pcm(mFormat)) { 1041 audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; 1042 } else { 1043 audioBuffer->format = mFormat; 1044 } 1045 audioBuffer->raw = (int8_t *)cblk->buffer(u); 1046 active = mActive; 1047 return active ? status_t(NO_ERROR) : status_t(STOPPED); 1048 } 1049 1050 void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1051 { 1052 AutoMutex lock(mLock); 1053 mCblk->stepUser(audioBuffer->frameCount); 1054 if (audioBuffer->frameCount > 0) { 1055 // restart track if it was disabled by audioflinger due to previous underrun 1056 if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1057 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1058 ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName); 1059 mAudioTrack->start(); 1060 } 1061 } 1062 } 1063 1064 // ------------------------------------------------------------------------- 1065 1066 ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1067 { 1068 1069 if (mSharedBuffer != 0) return INVALID_OPERATION; 1070 if (mIsTimed) return INVALID_OPERATION; 1071 1072 if (ssize_t(userSize) < 0) { 1073 // Sanity-check: user is most-likely passing an error code, and it would 1074 // make the return value ambiguous (actualSize vs error). 1075 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1076 buffer, userSize, userSize); 1077 return BAD_VALUE; 1078 } 1079 1080 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1081 1082 if (userSize == 0) { 1083 return 0; 1084 } 1085 1086 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1087 // while we are accessing the cblk 1088 mLock.lock(); 1089 sp<IAudioTrack> audioTrack = mAudioTrack; 1090 sp<IMemory> iMem = mCblkMemory; 1091 mLock.unlock(); 1092 1093 ssize_t written = 0; 1094 const int8_t *src = (const int8_t *)buffer; 1095 Buffer audioBuffer; 1096 size_t frameSz = frameSize(); 1097 1098 do { 1099 audioBuffer.frameCount = userSize/frameSz; 1100 1101 status_t err = obtainBuffer(&audioBuffer, -1); 1102 if (err < 0) { 1103 // out of buffers, return #bytes written 1104 if (err == status_t(NO_MORE_BUFFERS)) 1105 break; 1106 return ssize_t(err); 1107 } 1108 1109 size_t toWrite; 1110 1111 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1112 // Divide capacity by 2 to take expansion into account 1113 toWrite = audioBuffer.size>>1; 1114 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1115 } else { 1116 toWrite = audioBuffer.size; 1117 memcpy(audioBuffer.i8, src, toWrite); 1118 src += toWrite; 1119 } 1120 userSize -= toWrite; 1121 written += toWrite; 1122 1123 releaseBuffer(&audioBuffer); 1124 } while (userSize >= frameSz); 1125 1126 return written; 1127 } 1128 1129 // ------------------------------------------------------------------------- 1130 1131 TimedAudioTrack::TimedAudioTrack() { 1132 mIsTimed = true; 1133 } 1134 1135 status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1136 { 1137 status_t result = UNKNOWN_ERROR; 1138 1139 // If the track is not invalid already, try to allocate a buffer. alloc 1140 // fails indicating that the server is dead, flag the track as invalid so 1141 // we can attempt to restore in in just a bit. 1142 if (!(mCblk->flags & CBLK_INVALID_MSK)) { 1143 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1144 if (result == DEAD_OBJECT) { 1145 android_atomic_or(CBLK_INVALID_ON, &mCblk->flags); 1146 } 1147 } 1148 1149 // If the track is invalid at this point, attempt to restore it. and try the 1150 // allocation one more time. 1151 if (mCblk->flags & CBLK_INVALID_MSK) { 1152 mCblk->lock.lock(); 1153 result = restoreTrack_l(mCblk, false); 1154 mCblk->lock.unlock(); 1155 1156 if (result == OK) 1157 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1158 } 1159 1160 return result; 1161 } 1162 1163 status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1164 int64_t pts) 1165 { 1166 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1167 { 1168 AutoMutex lock(mLock); 1169 // restart track if it was disabled by audioflinger due to previous underrun 1170 if (buffer->size() != 0 && status == NO_ERROR && 1171 mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1172 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1173 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1174 mAudioTrack->start(); 1175 } 1176 } 1177 return status; 1178 } 1179 1180 status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1181 TargetTimeline target) 1182 { 1183 return mAudioTrack->setMediaTimeTransform(xform, target); 1184 } 1185 1186 // ------------------------------------------------------------------------- 1187 1188 bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1189 { 1190 Buffer audioBuffer; 1191 uint32_t frames; 1192 size_t writtenSize; 1193 1194 mLock.lock(); 1195 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1196 // while we are accessing the cblk 1197 sp<IAudioTrack> audioTrack = mAudioTrack; 1198 sp<IMemory> iMem = mCblkMemory; 1199 audio_track_cblk_t* cblk = mCblk; 1200 bool active = mActive; 1201 mLock.unlock(); 1202 1203 // Manage underrun callback 1204 if (active && (cblk->framesAvailable() == cblk->frameCount)) { 1205 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1206 if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { 1207 mCbf(EVENT_UNDERRUN, mUserData, 0); 1208 if (cblk->server == cblk->frameCount) { 1209 mCbf(EVENT_BUFFER_END, mUserData, 0); 1210 } 1211 if (mSharedBuffer != 0) return false; 1212 } 1213 } 1214 1215 // Manage loop end callback 1216 while (mLoopCount > cblk->loopCount) { 1217 int loopCount = -1; 1218 mLoopCount--; 1219 if (mLoopCount >= 0) loopCount = mLoopCount; 1220 1221 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1222 } 1223 1224 // Manage marker callback 1225 if (!mMarkerReached && (mMarkerPosition > 0)) { 1226 if (cblk->server >= mMarkerPosition) { 1227 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1228 mMarkerReached = true; 1229 } 1230 } 1231 1232 // Manage new position callback 1233 if (mUpdatePeriod > 0) { 1234 while (cblk->server >= mNewPosition) { 1235 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1236 mNewPosition += mUpdatePeriod; 1237 } 1238 } 1239 1240 // If Shared buffer is used, no data is requested from client. 1241 if (mSharedBuffer != 0) { 1242 frames = 0; 1243 } else { 1244 frames = mRemainingFrames; 1245 } 1246 1247 // See description of waitCount parameter at declaration of obtainBuffer(). 1248 // The logic below prevents us from being stuck below at obtainBuffer() 1249 // not being able to handle timed events (position, markers, loops). 1250 int32_t waitCount = -1; 1251 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1252 waitCount = 1; 1253 } 1254 1255 do { 1256 1257 audioBuffer.frameCount = frames; 1258 1259 status_t err = obtainBuffer(&audioBuffer, waitCount); 1260 if (err < NO_ERROR) { 1261 if (err != TIMED_OUT) { 1262 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); 1263 return false; 1264 } 1265 break; 1266 } 1267 if (err == status_t(STOPPED)) return false; 1268 1269 // Divide buffer size by 2 to take into account the expansion 1270 // due to 8 to 16 bit conversion: the callback must fill only half 1271 // of the destination buffer 1272 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1273 audioBuffer.size >>= 1; 1274 } 1275 1276 size_t reqSize = audioBuffer.size; 1277 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1278 writtenSize = audioBuffer.size; 1279 1280 // Sanity check on returned size 1281 if (ssize_t(writtenSize) <= 0) { 1282 // The callback is done filling buffers 1283 // Keep this thread going to handle timed events and 1284 // still try to get more data in intervals of WAIT_PERIOD_MS 1285 // but don't just loop and block the CPU, so wait 1286 usleep(WAIT_PERIOD_MS*1000); 1287 break; 1288 } 1289 1290 if (writtenSize > reqSize) writtenSize = reqSize; 1291 1292 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1293 // 8 to 16 bit conversion, note that source and destination are the same address 1294 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1295 writtenSize <<= 1; 1296 } 1297 1298 audioBuffer.size = writtenSize; 1299 // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for 1300 // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of 1301 // 16 bit. 1302 audioBuffer.frameCount = writtenSize/mCblk->frameSize; 1303 1304 frames -= audioBuffer.frameCount; 1305 1306 releaseBuffer(&audioBuffer); 1307 } 1308 while (frames); 1309 1310 if (frames == 0) { 1311 mRemainingFrames = mNotificationFramesAct; 1312 } else { 1313 mRemainingFrames = frames; 1314 } 1315 return true; 1316 } 1317 1318 // must be called with mLock and cblk.lock held. Callers must also hold strong references on 1319 // the IAudioTrack and IMemory in case they are recreated here. 1320 // If the IAudioTrack is successfully restored, the cblk pointer is updated 1321 status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) 1322 { 1323 status_t result; 1324 1325 if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { 1326 ALOGW("dead IAudioTrack, creating a new one from %s TID %d", 1327 fromStart ? "start()" : "obtainBuffer()", gettid()); 1328 1329 // signal old cblk condition so that other threads waiting for available buffers stop 1330 // waiting now 1331 cblk->cv.broadcast(); 1332 cblk->lock.unlock(); 1333 1334 // refresh the audio configuration cache in this process to make sure we get new 1335 // output parameters in getOutput_l() and createTrack_l() 1336 AudioSystem::clearAudioConfigCache(); 1337 1338 // if the new IAudioTrack is created, createTrack_l() will modify the 1339 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1340 // It will also delete the strong references on previous IAudioTrack and IMemory 1341 result = createTrack_l(mStreamType, 1342 cblk->sampleRate, 1343 mFormat, 1344 mChannelMask, 1345 mFrameCount, 1346 mFlags, 1347 mSharedBuffer, 1348 getOutput_l()); 1349 1350 if (result == NO_ERROR) { 1351 uint32_t user = cblk->user; 1352 uint32_t server = cblk->server; 1353 // restore write index and set other indexes to reflect empty buffer status 1354 mCblk->user = user; 1355 mCblk->server = user; 1356 mCblk->userBase = user; 1357 mCblk->serverBase = user; 1358 // restore loop: this is not guaranteed to succeed if new frame count is not 1359 // compatible with loop length 1360 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1361 if (!fromStart) { 1362 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1363 // Make sure that a client relying on callback events indicating underrun or 1364 // the actual amount of audio frames played (e.g SoundPool) receives them. 1365 if (mSharedBuffer == 0) { 1366 uint32_t frames = 0; 1367 if (user > server) { 1368 frames = ((user - server) > mCblk->frameCount) ? 1369 mCblk->frameCount : (user - server); 1370 memset(mCblk->buffers, 0, frames * mCblk->frameSize); 1371 } 1372 // restart playback even if buffer is not completely filled. 1373 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 1374 // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to 1375 // the client 1376 mCblk->stepUser(frames); 1377 } 1378 } 1379 if (mSharedBuffer != 0) { 1380 mCblk->stepUser(mCblk->frameCount); 1381 } 1382 if (mActive) { 1383 result = mAudioTrack->start(); 1384 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1385 } 1386 if (fromStart && result == NO_ERROR) { 1387 mNewPosition = mCblk->server + mUpdatePeriod; 1388 } 1389 } 1390 if (result != NO_ERROR) { 1391 android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); 1392 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1393 } 1394 mRestoreStatus = result; 1395 // signal old cblk condition for other threads waiting for restore completion 1396 android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); 1397 cblk->cv.broadcast(); 1398 } else { 1399 if (!(cblk->flags & CBLK_RESTORED_MSK)) { 1400 ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); 1401 mLock.unlock(); 1402 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1403 if (result == NO_ERROR) { 1404 result = mRestoreStatus; 1405 } 1406 cblk->lock.unlock(); 1407 mLock.lock(); 1408 } else { 1409 ALOGW("dead IAudioTrack, already restored TID %d", gettid()); 1410 result = mRestoreStatus; 1411 cblk->lock.unlock(); 1412 } 1413 } 1414 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1415 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1416 1417 if (result == NO_ERROR) { 1418 // from now on we switch to the newly created cblk 1419 cblk = mCblk; 1420 } 1421 cblk->lock.lock(); 1422 1423 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1424 1425 return result; 1426 } 1427 1428 status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1429 { 1430 1431 const size_t SIZE = 256; 1432 char buffer[SIZE]; 1433 String8 result; 1434 1435 result.append(" AudioTrack::dump\n"); 1436 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); 1437 result.append(buffer); 1438 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount); 1439 result.append(buffer); 1440 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1441 result.append(buffer); 1442 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1443 result.append(buffer); 1444 ::write(fd, result.string(), result.size()); 1445 return NO_ERROR; 1446 } 1447 1448 // ========================================================================= 1449 1450 AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1451 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1452 { 1453 } 1454 1455 AudioTrack::AudioTrackThread::~AudioTrackThread() 1456 { 1457 } 1458 1459 bool AudioTrack::AudioTrackThread::threadLoop() 1460 { 1461 { 1462 AutoMutex _l(mMyLock); 1463 if (mPaused) { 1464 mMyCond.wait(mMyLock); 1465 // caller will check for exitPending() 1466 return true; 1467 } 1468 } 1469 if (!mReceiver.processAudioBuffer(this)) { 1470 pause(); 1471 } 1472 return true; 1473 } 1474 1475 status_t AudioTrack::AudioTrackThread::readyToRun() 1476 { 1477 return NO_ERROR; 1478 } 1479 1480 void AudioTrack::AudioTrackThread::onFirstRef() 1481 { 1482 } 1483 1484 void AudioTrack::AudioTrackThread::requestExit() 1485 { 1486 // must be in this order to avoid a race condition 1487 Thread::requestExit(); 1488 resume(); 1489 } 1490 1491 void AudioTrack::AudioTrackThread::pause() 1492 { 1493 AutoMutex _l(mMyLock); 1494 mPaused = true; 1495 } 1496 1497 void AudioTrack::AudioTrackThread::resume() 1498 { 1499 AutoMutex _l(mMyLock); 1500 if (mPaused) { 1501 mPaused = false; 1502 mMyCond.signal(); 1503 } 1504 } 1505 1506 // ========================================================================= 1507 1508 1509 audio_track_cblk_t::audio_track_cblk_t() 1510 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1511 userBase(0), serverBase(0), buffers(NULL), frameCount(0), 1512 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), 1513 mSendLevel(0), flags(0) 1514 { 1515 } 1516 1517 uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1518 { 1519 ALOGV("stepuser %08x %08x %d", user, server, frameCount); 1520 1521 uint32_t u = user; 1522 u += frameCount; 1523 // Ensure that user is never ahead of server for AudioRecord 1524 if (flags & CBLK_DIRECTION_MSK) { 1525 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1526 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1527 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1528 } 1529 } else if (u > server) { 1530 ALOGW("stepUser occurred after track reset"); 1531 u = server; 1532 } 1533 1534 uint32_t fc = this->frameCount; 1535 if (u >= fc) { 1536 // common case, user didn't just wrap 1537 if (u - fc >= userBase ) { 1538 userBase += fc; 1539 } 1540 } else if (u >= userBase + fc) { 1541 // user just wrapped 1542 userBase += fc; 1543 } 1544 1545 user = u; 1546 1547 // Clear flow control error condition as new data has been written/read to/from buffer. 1548 if (flags & CBLK_UNDERRUN_MSK) { 1549 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1550 } 1551 1552 return u; 1553 } 1554 1555 bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1556 { 1557 ALOGV("stepserver %08x %08x %d", user, server, frameCount); 1558 1559 if (!tryLock()) { 1560 ALOGW("stepServer() could not lock cblk"); 1561 return false; 1562 } 1563 1564 uint32_t s = server; 1565 bool flushed = (s == user); 1566 1567 s += frameCount; 1568 if (flags & CBLK_DIRECTION_MSK) { 1569 // Mark that we have read the first buffer so that next time stepUser() is called 1570 // we switch to normal obtainBuffer() timeout period 1571 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1572 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1573 } 1574 // It is possible that we receive a flush() 1575 // while the mixer is processing a block: in this case, 1576 // stepServer() is called After the flush() has reset u & s and 1577 // we have s > u 1578 if (flushed) { 1579 ALOGW("stepServer occurred after track reset"); 1580 s = user; 1581 } 1582 } 1583 1584 if (s >= loopEnd) { 1585 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1586 s = loopStart; 1587 if (--loopCount == 0) { 1588 loopEnd = UINT_MAX; 1589 loopStart = UINT_MAX; 1590 } 1591 } 1592 1593 uint32_t fc = this->frameCount; 1594 if (s >= fc) { 1595 // common case, server didn't just wrap 1596 if (s - fc >= serverBase ) { 1597 serverBase += fc; 1598 } 1599 } else if (s >= serverBase + fc) { 1600 // server just wrapped 1601 serverBase += fc; 1602 } 1603 1604 server = s; 1605 1606 if (!(flags & CBLK_INVALID_MSK)) { 1607 cv.signal(); 1608 } 1609 lock.unlock(); 1610 return true; 1611 } 1612 1613 void* audio_track_cblk_t::buffer(uint32_t offset) const 1614 { 1615 return (int8_t *)buffers + (offset - userBase) * frameSize; 1616 } 1617 1618 uint32_t audio_track_cblk_t::framesAvailable() 1619 { 1620 Mutex::Autolock _l(lock); 1621 return framesAvailable_l(); 1622 } 1623 1624 uint32_t audio_track_cblk_t::framesAvailable_l() 1625 { 1626 uint32_t u = user; 1627 uint32_t s = server; 1628 1629 if (flags & CBLK_DIRECTION_MSK) { 1630 uint32_t limit = (s < loopStart) ? s : loopStart; 1631 return limit + frameCount - u; 1632 } else { 1633 return frameCount + u - s; 1634 } 1635 } 1636 1637 uint32_t audio_track_cblk_t::framesReady() 1638 { 1639 uint32_t u = user; 1640 uint32_t s = server; 1641 1642 if (flags & CBLK_DIRECTION_MSK) { 1643 if (u < loopEnd) { 1644 return u - s; 1645 } else { 1646 // do not block on mutex shared with client on AudioFlinger side 1647 if (!tryLock()) { 1648 ALOGW("framesReady() could not lock cblk"); 1649 return 0; 1650 } 1651 uint32_t frames = UINT_MAX; 1652 if (loopCount >= 0) { 1653 frames = (loopEnd - loopStart)*loopCount + u - s; 1654 } 1655 lock.unlock(); 1656 return frames; 1657 } 1658 } else { 1659 return s - u; 1660 } 1661 } 1662 1663 bool audio_track_cblk_t::tryLock() 1664 { 1665 // the code below simulates lock-with-timeout 1666 // we MUST do this to protect the AudioFlinger server 1667 // as this lock is shared with the client. 1668 status_t err; 1669 1670 err = lock.tryLock(); 1671 if (err == -EBUSY) { // just wait a bit 1672 usleep(1000); 1673 err = lock.tryLock(); 1674 } 1675 if (err != NO_ERROR) { 1676 // probably, the client just died. 1677 return false; 1678 } 1679 return true; 1680 } 1681 1682 // ------------------------------------------------------------------------- 1683 1684 }; // namespace android 1685