AudioTrack.cpp revision 093000f7d11839b920e8dfaa42ed1d09f48e24b8
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 48namespace android { 49// --------------------------------------------------------------------------- 50 51// static 52status_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 89AudioTrack::AudioTrack() 90 : mStatus(NO_INIT), 91 mIsTimed(false), 92 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 93 mPreviousSchedulingGroup(SP_DEFAULT) 94{ 95} 96 97AudioTrack::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 119AudioTrack::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 139AudioTrack::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 160AudioTrack::~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 180status_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 319status_t AudioTrack::initCheck() const 320{ 321 return mStatus; 322} 323 324// ------------------------------------------------------------------------- 325 326uint32_t AudioTrack::latency() const 327{ 328 return mLatency; 329} 330 331audio_stream_type_t AudioTrack::streamType() const 332{ 333 return mStreamType; 334} 335 336audio_format_t AudioTrack::format() const 337{ 338 return mFormat; 339} 340 341int AudioTrack::channelCount() const 342{ 343 return mChannelCount; 344} 345 346uint32_t AudioTrack::frameCount() const 347{ 348 return mCblk->frameCount; 349} 350 351size_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 360sp<IMemory>& AudioTrack::sharedBuffer() 361{ 362 return mSharedBuffer; 363} 364 365// ------------------------------------------------------------------------- 366 367void 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 425void 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 457bool AudioTrack::stopped() const 458{ 459 AutoMutex lock(mLock); 460 return stopped_l(); 461} 462 463void AudioTrack::flush() 464{ 465 AutoMutex lock(mLock); 466 flush_l(); 467} 468 469// must be called with mLock held 470void 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 488void AudioTrack::pause() 489{ 490 ALOGV("pause"); 491 AutoMutex lock(mLock); 492 if (mActive) { 493 mActive = false; 494 mAudioTrack->pause(); 495 } 496} 497 498void AudioTrack::mute(bool e) 499{ 500 mAudioTrack->mute(e); 501 mMuted = e; 502} 503 504bool AudioTrack::muted() const 505{ 506 return mMuted; 507} 508 509status_t AudioTrack::setVolume(float left, float right) 510{ 511 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 512 return BAD_VALUE; 513 } 514 515 AutoMutex lock(mLock); 516 mVolume[LEFT] = left; 517 mVolume[RIGHT] = right; 518 519 mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 520 521 return NO_ERROR; 522} 523 524void AudioTrack::getVolume(float* left, float* right) const 525{ 526 if (left != NULL) { 527 *left = mVolume[LEFT]; 528 } 529 if (right != NULL) { 530 *right = mVolume[RIGHT]; 531 } 532} 533 534status_t AudioTrack::setAuxEffectSendLevel(float level) 535{ 536 ALOGV("setAuxEffectSendLevel(%f)", level); 537 if (level < 0.0f || level > 1.0f) { 538 return BAD_VALUE; 539 } 540 AutoMutex lock(mLock); 541 542 mSendLevel = level; 543 544 mCblk->setSendLevel(level); 545 546 return NO_ERROR; 547} 548 549void AudioTrack::getAuxEffectSendLevel(float* level) const 550{ 551 if (level != NULL) { 552 *level = mSendLevel; 553 } 554} 555 556status_t AudioTrack::setSampleRate(int rate) 557{ 558 int afSamplingRate; 559 560 if (mIsTimed) { 561 return INVALID_OPERATION; 562 } 563 564 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 565 return NO_INIT; 566 } 567 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 568 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 569 570 AutoMutex lock(mLock); 571 mCblk->sampleRate = rate; 572 return NO_ERROR; 573} 574 575uint32_t AudioTrack::getSampleRate() const 576{ 577 if (mIsTimed) { 578 return INVALID_OPERATION; 579 } 580 581 AutoMutex lock(mLock); 582 return mCblk->sampleRate; 583} 584 585status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 586{ 587 AutoMutex lock(mLock); 588 return setLoop_l(loopStart, loopEnd, loopCount); 589} 590 591// must be called with mLock held 592status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 593{ 594 audio_track_cblk_t* cblk = mCblk; 595 596 Mutex::Autolock _l(cblk->lock); 597 598 if (loopCount == 0) { 599 cblk->loopStart = UINT_MAX; 600 cblk->loopEnd = UINT_MAX; 601 cblk->loopCount = 0; 602 mLoopCount = 0; 603 return NO_ERROR; 604 } 605 606 if (mIsTimed) { 607 return INVALID_OPERATION; 608 } 609 610 if (loopStart >= loopEnd || 611 loopEnd - loopStart > cblk->frameCount || 612 cblk->server > loopStart) { 613 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 614 return BAD_VALUE; 615 } 616 617 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 618 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", 619 loopStart, loopEnd, cblk->frameCount); 620 return BAD_VALUE; 621 } 622 623 cblk->loopStart = loopStart; 624 cblk->loopEnd = loopEnd; 625 cblk->loopCount = loopCount; 626 mLoopCount = loopCount; 627 628 return NO_ERROR; 629} 630 631status_t AudioTrack::setMarkerPosition(uint32_t marker) 632{ 633 if (mCbf == NULL) return INVALID_OPERATION; 634 635 mMarkerPosition = marker; 636 mMarkerReached = false; 637 638 return NO_ERROR; 639} 640 641status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 642{ 643 if (marker == NULL) return BAD_VALUE; 644 645 *marker = mMarkerPosition; 646 647 return NO_ERROR; 648} 649 650status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 651{ 652 if (mCbf == NULL) return INVALID_OPERATION; 653 654 uint32_t curPosition; 655 getPosition(&curPosition); 656 mNewPosition = curPosition + updatePeriod; 657 mUpdatePeriod = updatePeriod; 658 659 return NO_ERROR; 660} 661 662status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 663{ 664 if (updatePeriod == NULL) return BAD_VALUE; 665 666 *updatePeriod = mUpdatePeriod; 667 668 return NO_ERROR; 669} 670 671status_t AudioTrack::setPosition(uint32_t position) 672{ 673 if (mIsTimed) return INVALID_OPERATION; 674 675 AutoMutex lock(mLock); 676 677 if (!stopped_l()) return INVALID_OPERATION; 678 679 Mutex::Autolock _l(mCblk->lock); 680 681 if (position > mCblk->user) return BAD_VALUE; 682 683 mCblk->server = position; 684 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 685 686 return NO_ERROR; 687} 688 689status_t AudioTrack::getPosition(uint32_t *position) 690{ 691 if (position == NULL) return BAD_VALUE; 692 AutoMutex lock(mLock); 693 *position = mFlushed ? 0 : mCblk->server; 694 695 return NO_ERROR; 696} 697 698status_t AudioTrack::reload() 699{ 700 AutoMutex lock(mLock); 701 702 if (!stopped_l()) return INVALID_OPERATION; 703 704 flush_l(); 705 706 mCblk->stepUser(mCblk->frameCount); 707 708 return NO_ERROR; 709} 710 711audio_io_handle_t AudioTrack::getOutput() 712{ 713 AutoMutex lock(mLock); 714 return getOutput_l(); 715} 716 717// must be called with mLock held 718audio_io_handle_t AudioTrack::getOutput_l() 719{ 720 return AudioSystem::getOutput(mStreamType, 721 mCblk->sampleRate, mFormat, mChannelMask, mFlags); 722} 723 724int AudioTrack::getSessionId() const 725{ 726 return mSessionId; 727} 728 729status_t AudioTrack::attachAuxEffect(int effectId) 730{ 731 ALOGV("attachAuxEffect(%d)", effectId); 732 status_t status = mAudioTrack->attachAuxEffect(effectId); 733 if (status == NO_ERROR) { 734 mAuxEffectId = effectId; 735 } 736 return status; 737} 738 739// ------------------------------------------------------------------------- 740 741// must be called with mLock held 742status_t AudioTrack::createTrack_l( 743 audio_stream_type_t streamType, 744 uint32_t sampleRate, 745 audio_format_t format, 746 uint32_t channelMask, 747 int frameCount, 748 audio_output_flags_t flags, 749 const sp<IMemory>& sharedBuffer, 750 audio_io_handle_t output) 751{ 752 status_t status; 753 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 754 if (audioFlinger == 0) { 755 ALOGE("Could not get audioflinger"); 756 return NO_INIT; 757 } 758 759 uint32_t afLatency; 760 if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { 761 return NO_INIT; 762 } 763 764 // Client decides whether the track is TIMED (see below), but can only express a preference 765 // for FAST. Server will perform additional tests. 766 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 767 // either of these use cases: 768 // use case 1: shared buffer 769 (sharedBuffer != 0) || 770 // use case 2: callback handler 771 (mCbf != NULL))) { 772 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 773 // once denied, do not request again if IAudioTrack is re-created 774 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 775 mFlags = flags; 776 } 777 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 778 779 mNotificationFramesAct = mNotificationFramesReq; 780 781 if (!audio_is_linear_pcm(format)) { 782 783 if (sharedBuffer != 0) { 784 // Same comment as below about ignoring frameCount parameter for set() 785 frameCount = sharedBuffer->size(); 786 } else if (frameCount == 0) { 787 int afFrameCount; 788 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 789 return NO_INIT; 790 } 791 frameCount = afFrameCount; 792 } 793 794 } else if (sharedBuffer != 0) { 795 796 // Ensure that buffer alignment matches channelCount 797 int channelCount = popcount(channelMask); 798 // 8-bit data in shared memory is not currently supported by AudioFlinger 799 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 800 if (channelCount > 1) { 801 // More than 2 channels does not require stronger alignment than stereo 802 alignment <<= 1; 803 } 804 if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 805 ALOGE("Invalid buffer alignment: address %p, channelCount %d", 806 sharedBuffer->pointer(), channelCount); 807 return BAD_VALUE; 808 } 809 810 // When initializing a shared buffer AudioTrack via constructors, 811 // there's no frameCount parameter. 812 // But when initializing a shared buffer AudioTrack via set(), 813 // there _is_ a frameCount parameter. We silently ignore it. 814 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 815 816 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 817 818 // FIXME move these calculations and associated checks to server 819 int afSampleRate; 820 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 821 return NO_INIT; 822 } 823 int afFrameCount; 824 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 825 return NO_INIT; 826 } 827 828 // Ensure that buffer depth covers at least audio hardware latency 829 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 830 if (minBufCount < 2) minBufCount = 2; 831 832 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 833 ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d" 834 ", afLatency=%d", 835 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 836 837 if (frameCount == 0) { 838 frameCount = minFrameCount; 839 } 840 if (mNotificationFramesAct == 0) { 841 mNotificationFramesAct = frameCount/2; 842 } 843 // Make sure that application is notified with sufficient margin 844 // before underrun 845 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 846 mNotificationFramesAct = frameCount/2; 847 } 848 if (frameCount < minFrameCount) { 849 // not ALOGW because it happens all the time when playing key clicks over A2DP 850 ALOGV("Minimum buffer size corrected from %d to %d", 851 frameCount, minFrameCount); 852 frameCount = minFrameCount; 853 } 854 855 } else { 856 // For fast tracks, the frame count calculations and checks are done by server 857 } 858 859 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 860 if (mIsTimed) { 861 trackFlags |= IAudioFlinger::TRACK_TIMED; 862 } 863 864 pid_t tid = -1; 865 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 866 trackFlags |= IAudioFlinger::TRACK_FAST; 867 if (mAudioTrackThread != 0) { 868 tid = mAudioTrackThread->getTid(); 869 } 870 } 871 872 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 873 streamType, 874 sampleRate, 875 format, 876 channelMask, 877 frameCount, 878 trackFlags, 879 sharedBuffer, 880 output, 881 tid, 882 &mSessionId, 883 &status); 884 885 if (track == 0) { 886 ALOGE("AudioFlinger could not create track, status: %d", status); 887 return status; 888 } 889 sp<IMemory> cblk = track->getCblk(); 890 if (cblk == 0) { 891 ALOGE("Could not get control block"); 892 return NO_INIT; 893 } 894 mAudioTrack = track; 895 mCblkMemory = cblk; 896 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 897 // old has the previous value of mCblk->flags before the "or" operation 898 int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 899 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 900 if (old & CBLK_FAST) { 901 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount); 902 } else { 903 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount); 904 // once denied, do not request again if IAudioTrack is re-created 905 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 906 mFlags = flags; 907 } 908 if (sharedBuffer == 0) { 909 mNotificationFramesAct = mCblk->frameCount/2; 910 } 911 } 912 if (sharedBuffer == 0) { 913 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 914 } else { 915 mCblk->buffers = sharedBuffer->pointer(); 916 // Force buffer full condition as data is already present in shared memory 917 mCblk->stepUser(mCblk->frameCount); 918 } 919 920 mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); 921 mCblk->setSendLevel(mSendLevel); 922 mAudioTrack->attachAuxEffect(mAuxEffectId); 923 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 924 mCblk->waitTimeMs = 0; 925 mRemainingFrames = mNotificationFramesAct; 926 // FIXME don't believe this lie 927 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 928 // If IAudioTrack is re-created, don't let the requested frameCount 929 // decrease. This can confuse clients that cache frameCount(). 930 if (mCblk->frameCount > mFrameCount) { 931 mFrameCount = mCblk->frameCount; 932 } 933 return NO_ERROR; 934} 935 936status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 937{ 938 AutoMutex lock(mLock); 939 bool active; 940 status_t result = NO_ERROR; 941 audio_track_cblk_t* cblk = mCblk; 942 uint32_t framesReq = audioBuffer->frameCount; 943 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 944 945 audioBuffer->frameCount = 0; 946 audioBuffer->size = 0; 947 948 uint32_t framesAvail = cblk->framesAvailable(); 949 950 cblk->lock.lock(); 951 if (cblk->flags & CBLK_INVALID_MSK) { 952 goto create_new_track; 953 } 954 cblk->lock.unlock(); 955 956 if (framesAvail == 0) { 957 cblk->lock.lock(); 958 goto start_loop_here; 959 while (framesAvail == 0) { 960 active = mActive; 961 if (CC_UNLIKELY(!active)) { 962 ALOGV("Not active and NO_MORE_BUFFERS"); 963 cblk->lock.unlock(); 964 return NO_MORE_BUFFERS; 965 } 966 if (CC_UNLIKELY(!waitCount)) { 967 cblk->lock.unlock(); 968 return WOULD_BLOCK; 969 } 970 if (!(cblk->flags & CBLK_INVALID_MSK)) { 971 mLock.unlock(); 972 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 973 cblk->lock.unlock(); 974 mLock.lock(); 975 if (!mActive) { 976 return status_t(STOPPED); 977 } 978 cblk->lock.lock(); 979 } 980 981 if (cblk->flags & CBLK_INVALID_MSK) { 982 goto create_new_track; 983 } 984 if (CC_UNLIKELY(result != NO_ERROR)) { 985 cblk->waitTimeMs += waitTimeMs; 986 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 987 // timing out when a loop has been set and we have already written upto loop end 988 // is a normal condition: no need to wake AudioFlinger up. 989 if (cblk->user < cblk->loopEnd) { 990 ALOGW( "obtainBuffer timed out (is the CPU pegged?) %p " 991 "user=%08x, server=%08x", this, cblk->user, cblk->server); 992 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 993 cblk->lock.unlock(); 994 result = mAudioTrack->start(); 995 cblk->lock.lock(); 996 if (result == DEAD_OBJECT) { 997 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 998create_new_track: 999 result = restoreTrack_l(cblk, false); 1000 } 1001 if (result != NO_ERROR) { 1002 ALOGW("obtainBuffer create Track error %d", result); 1003 cblk->lock.unlock(); 1004 return result; 1005 } 1006 } 1007 cblk->waitTimeMs = 0; 1008 } 1009 1010 if (--waitCount == 0) { 1011 cblk->lock.unlock(); 1012 return TIMED_OUT; 1013 } 1014 } 1015 // read the server count again 1016 start_loop_here: 1017 framesAvail = cblk->framesAvailable_l(); 1018 } 1019 cblk->lock.unlock(); 1020 } 1021 1022 // restart track if it was disabled by audioflinger due to previous underrun 1023 if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) { 1024 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 1025 ALOGW("obtainBuffer() track %p disabled, restarting", this); 1026 mAudioTrack->start(); 1027 } 1028 1029 cblk->waitTimeMs = 0; 1030 1031 if (framesReq > framesAvail) { 1032 framesReq = framesAvail; 1033 } 1034 1035 uint32_t u = cblk->user; 1036 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 1037 1038 if (framesReq > bufferEnd - u) { 1039 framesReq = bufferEnd - u; 1040 } 1041 1042 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 1043 audioBuffer->channelCount = mChannelCount; 1044 audioBuffer->frameCount = framesReq; 1045 audioBuffer->size = framesReq * cblk->frameSize; 1046 if (audio_is_linear_pcm(mFormat)) { 1047 audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; 1048 } else { 1049 audioBuffer->format = mFormat; 1050 } 1051 audioBuffer->raw = (int8_t *)cblk->buffer(u); 1052 active = mActive; 1053 return active ? status_t(NO_ERROR) : status_t(STOPPED); 1054} 1055 1056void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1057{ 1058 AutoMutex lock(mLock); 1059 mCblk->stepUser(audioBuffer->frameCount); 1060} 1061 1062// ------------------------------------------------------------------------- 1063 1064ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1065{ 1066 1067 if (mSharedBuffer != 0) return INVALID_OPERATION; 1068 if (mIsTimed) return INVALID_OPERATION; 1069 1070 if (ssize_t(userSize) < 0) { 1071 // Sanity-check: user is most-likely passing an error code, and it would 1072 // make the return value ambiguous (actualSize vs error). 1073 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1074 buffer, userSize, userSize); 1075 return BAD_VALUE; 1076 } 1077 1078 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1079 1080 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1081 // while we are accessing the cblk 1082 mLock.lock(); 1083 sp<IAudioTrack> audioTrack = mAudioTrack; 1084 sp<IMemory> iMem = mCblkMemory; 1085 mLock.unlock(); 1086 1087 ssize_t written = 0; 1088 const int8_t *src = (const int8_t *)buffer; 1089 Buffer audioBuffer; 1090 size_t frameSz = frameSize(); 1091 1092 do { 1093 audioBuffer.frameCount = userSize/frameSz; 1094 1095 status_t err = obtainBuffer(&audioBuffer, -1); 1096 if (err < 0) { 1097 // out of buffers, return #bytes written 1098 if (err == status_t(NO_MORE_BUFFERS)) 1099 break; 1100 return ssize_t(err); 1101 } 1102 1103 size_t toWrite; 1104 1105 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1106 // Divide capacity by 2 to take expansion into account 1107 toWrite = audioBuffer.size>>1; 1108 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1109 } else { 1110 toWrite = audioBuffer.size; 1111 memcpy(audioBuffer.i8, src, toWrite); 1112 src += toWrite; 1113 } 1114 userSize -= toWrite; 1115 written += toWrite; 1116 1117 releaseBuffer(&audioBuffer); 1118 } while (userSize >= frameSz); 1119 1120 return written; 1121} 1122 1123// ------------------------------------------------------------------------- 1124 1125TimedAudioTrack::TimedAudioTrack() { 1126 mIsTimed = true; 1127} 1128 1129status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1130{ 1131 status_t result = UNKNOWN_ERROR; 1132 1133 // If the track is not invalid already, try to allocate a buffer. alloc 1134 // fails indicating that the server is dead, flag the track as invalid so 1135 // we can attempt to restore in in just a bit. 1136 if (!(mCblk->flags & CBLK_INVALID_MSK)) { 1137 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1138 if (result == DEAD_OBJECT) { 1139 android_atomic_or(CBLK_INVALID_ON, &mCblk->flags); 1140 } 1141 } 1142 1143 // If the track is invalid at this point, attempt to restore it. and try the 1144 // allocation one more time. 1145 if (mCblk->flags & CBLK_INVALID_MSK) { 1146 mCblk->lock.lock(); 1147 result = restoreTrack_l(mCblk, false); 1148 mCblk->lock.unlock(); 1149 1150 if (result == OK) 1151 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1152 } 1153 1154 return result; 1155} 1156 1157status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1158 int64_t pts) 1159{ 1160 // restart track if it was disabled by audioflinger due to previous underrun 1161 if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1162 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1163 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1164 mAudioTrack->start(); 1165 } 1166 1167 return mAudioTrack->queueTimedBuffer(buffer, pts); 1168} 1169 1170status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1171 TargetTimeline target) 1172{ 1173 return mAudioTrack->setMediaTimeTransform(xform, target); 1174} 1175 1176// ------------------------------------------------------------------------- 1177 1178bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1179{ 1180 Buffer audioBuffer; 1181 uint32_t frames; 1182 size_t writtenSize; 1183 1184 mLock.lock(); 1185 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1186 // while we are accessing the cblk 1187 sp<IAudioTrack> audioTrack = mAudioTrack; 1188 sp<IMemory> iMem = mCblkMemory; 1189 audio_track_cblk_t* cblk = mCblk; 1190 bool active = mActive; 1191 mLock.unlock(); 1192 1193 // Manage underrun callback 1194 if (active && (cblk->framesAvailable() == cblk->frameCount)) { 1195 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1196 if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { 1197 mCbf(EVENT_UNDERRUN, mUserData, 0); 1198 if (cblk->server == cblk->frameCount) { 1199 mCbf(EVENT_BUFFER_END, mUserData, 0); 1200 } 1201 if (mSharedBuffer != 0) return false; 1202 } 1203 } 1204 1205 // Manage loop end callback 1206 while (mLoopCount > cblk->loopCount) { 1207 int loopCount = -1; 1208 mLoopCount--; 1209 if (mLoopCount >= 0) loopCount = mLoopCount; 1210 1211 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1212 } 1213 1214 // Manage marker callback 1215 if (!mMarkerReached && (mMarkerPosition > 0)) { 1216 if (cblk->server >= mMarkerPosition) { 1217 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1218 mMarkerReached = true; 1219 } 1220 } 1221 1222 // Manage new position callback 1223 if (mUpdatePeriod > 0) { 1224 while (cblk->server >= mNewPosition) { 1225 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1226 mNewPosition += mUpdatePeriod; 1227 } 1228 } 1229 1230 // If Shared buffer is used, no data is requested from client. 1231 if (mSharedBuffer != 0) { 1232 frames = 0; 1233 } else { 1234 frames = mRemainingFrames; 1235 } 1236 1237 // See description of waitCount parameter at declaration of obtainBuffer(). 1238 // The logic below prevents us from being stuck below at obtainBuffer() 1239 // not being able to handle timed events (position, markers, loops). 1240 int32_t waitCount = -1; 1241 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1242 waitCount = 1; 1243 } 1244 1245 do { 1246 1247 audioBuffer.frameCount = frames; 1248 1249 status_t err = obtainBuffer(&audioBuffer, waitCount); 1250 if (err < NO_ERROR) { 1251 if (err != TIMED_OUT) { 1252 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); 1253 return false; 1254 } 1255 break; 1256 } 1257 if (err == status_t(STOPPED)) return false; 1258 1259 // Divide buffer size by 2 to take into account the expansion 1260 // due to 8 to 16 bit conversion: the callback must fill only half 1261 // of the destination buffer 1262 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1263 audioBuffer.size >>= 1; 1264 } 1265 1266 size_t reqSize = audioBuffer.size; 1267 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1268 writtenSize = audioBuffer.size; 1269 1270 // Sanity check on returned size 1271 if (ssize_t(writtenSize) <= 0) { 1272 // The callback is done filling buffers 1273 // Keep this thread going to handle timed events and 1274 // still try to get more data in intervals of WAIT_PERIOD_MS 1275 // but don't just loop and block the CPU, so wait 1276 usleep(WAIT_PERIOD_MS*1000); 1277 break; 1278 } 1279 if (writtenSize > reqSize) writtenSize = reqSize; 1280 1281 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1282 // 8 to 16 bit conversion, note that source and destination are the same address 1283 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1284 writtenSize <<= 1; 1285 } 1286 1287 audioBuffer.size = writtenSize; 1288 // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for 1289 // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of 1290 // 16 bit. 1291 audioBuffer.frameCount = writtenSize/mCblk->frameSize; 1292 1293 frames -= audioBuffer.frameCount; 1294 1295 releaseBuffer(&audioBuffer); 1296 } 1297 while (frames); 1298 1299 if (frames == 0) { 1300 mRemainingFrames = mNotificationFramesAct; 1301 } else { 1302 mRemainingFrames = frames; 1303 } 1304 return true; 1305} 1306 1307// must be called with mLock and cblk.lock held. Callers must also hold strong references on 1308// the IAudioTrack and IMemory in case they are recreated here. 1309// If the IAudioTrack is successfully restored, the cblk pointer is updated 1310status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) 1311{ 1312 status_t result; 1313 1314 if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { 1315 ALOGW("dead IAudioTrack, creating a new one from %s TID %d", 1316 fromStart ? "start()" : "obtainBuffer()", gettid()); 1317 1318 // signal old cblk condition so that other threads waiting for available buffers stop 1319 // waiting now 1320 cblk->cv.broadcast(); 1321 cblk->lock.unlock(); 1322 1323 // refresh the audio configuration cache in this process to make sure we get new 1324 // output parameters in getOutput_l() and createTrack_l() 1325 AudioSystem::clearAudioConfigCache(); 1326 1327 // if the new IAudioTrack is created, createTrack_l() will modify the 1328 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1329 // It will also delete the strong references on previous IAudioTrack and IMemory 1330 result = createTrack_l(mStreamType, 1331 cblk->sampleRate, 1332 mFormat, 1333 mChannelMask, 1334 mFrameCount, 1335 mFlags, 1336 mSharedBuffer, 1337 getOutput_l()); 1338 1339 if (result == NO_ERROR) { 1340 uint32_t user = cblk->user; 1341 uint32_t server = cblk->server; 1342 // restore write index and set other indexes to reflect empty buffer status 1343 mCblk->user = user; 1344 mCblk->server = user; 1345 mCblk->userBase = user; 1346 mCblk->serverBase = user; 1347 // restore loop: this is not guaranteed to succeed if new frame count is not 1348 // compatible with loop length 1349 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1350 if (!fromStart) { 1351 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1352 // Make sure that a client relying on callback events indicating underrun or 1353 // the actual amount of audio frames played (e.g SoundPool) receives them. 1354 if (mSharedBuffer == 0) { 1355 uint32_t frames = 0; 1356 if (user > server) { 1357 frames = ((user - server) > mCblk->frameCount) ? 1358 mCblk->frameCount : (user - server); 1359 memset(mCblk->buffers, 0, frames * mCblk->frameSize); 1360 } 1361 // restart playback even if buffer is not completely filled. 1362 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 1363 // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to 1364 // the client 1365 mCblk->stepUser(frames); 1366 } 1367 } 1368 if (mActive) { 1369 result = mAudioTrack->start(); 1370 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1371 } 1372 if (fromStart && result == NO_ERROR) { 1373 mNewPosition = mCblk->server + mUpdatePeriod; 1374 } 1375 } 1376 if (result != NO_ERROR) { 1377 android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); 1378 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1379 } 1380 mRestoreStatus = result; 1381 // signal old cblk condition for other threads waiting for restore completion 1382 android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); 1383 cblk->cv.broadcast(); 1384 } else { 1385 if (!(cblk->flags & CBLK_RESTORED_MSK)) { 1386 ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); 1387 mLock.unlock(); 1388 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1389 if (result == NO_ERROR) { 1390 result = mRestoreStatus; 1391 } 1392 cblk->lock.unlock(); 1393 mLock.lock(); 1394 } else { 1395 ALOGW("dead IAudioTrack, already restored TID %d", gettid()); 1396 result = mRestoreStatus; 1397 cblk->lock.unlock(); 1398 } 1399 } 1400 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1401 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1402 1403 if (result == NO_ERROR) { 1404 // from now on we switch to the newly created cblk 1405 cblk = mCblk; 1406 } 1407 cblk->lock.lock(); 1408 1409 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1410 1411 return result; 1412} 1413 1414status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1415{ 1416 1417 const size_t SIZE = 256; 1418 char buffer[SIZE]; 1419 String8 result; 1420 1421 result.append(" AudioTrack::dump\n"); 1422 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); 1423 result.append(buffer); 1424 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount); 1425 result.append(buffer); 1426 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1427 result.append(buffer); 1428 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1429 result.append(buffer); 1430 ::write(fd, result.string(), result.size()); 1431 return NO_ERROR; 1432} 1433 1434// ========================================================================= 1435 1436AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1437 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1438{ 1439} 1440 1441AudioTrack::AudioTrackThread::~AudioTrackThread() 1442{ 1443} 1444 1445bool AudioTrack::AudioTrackThread::threadLoop() 1446{ 1447 { 1448 AutoMutex _l(mMyLock); 1449 if (mPaused) { 1450 mMyCond.wait(mMyLock); 1451 // caller will check for exitPending() 1452 return true; 1453 } 1454 } 1455 if (!mReceiver.processAudioBuffer(this)) { 1456 pause(); 1457 } 1458 return true; 1459} 1460 1461status_t AudioTrack::AudioTrackThread::readyToRun() 1462{ 1463 return NO_ERROR; 1464} 1465 1466void AudioTrack::AudioTrackThread::onFirstRef() 1467{ 1468} 1469 1470void AudioTrack::AudioTrackThread::requestExit() 1471{ 1472 // must be in this order to avoid a race condition 1473 Thread::requestExit(); 1474 resume(); 1475} 1476 1477void AudioTrack::AudioTrackThread::pause() 1478{ 1479 AutoMutex _l(mMyLock); 1480 mPaused = true; 1481} 1482 1483void AudioTrack::AudioTrackThread::resume() 1484{ 1485 AutoMutex _l(mMyLock); 1486 if (mPaused) { 1487 mPaused = false; 1488 mMyCond.signal(); 1489 } 1490} 1491 1492// ========================================================================= 1493 1494 1495audio_track_cblk_t::audio_track_cblk_t() 1496 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1497 userBase(0), serverBase(0), buffers(NULL), frameCount(0), 1498 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), 1499 mSendLevel(0), flags(0) 1500{ 1501} 1502 1503uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1504{ 1505 ALOGV("stepuser %08x %08x %d", user, server, frameCount); 1506 1507 uint32_t u = user; 1508 u += frameCount; 1509 // Ensure that user is never ahead of server for AudioRecord 1510 if (flags & CBLK_DIRECTION_MSK) { 1511 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1512 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1513 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1514 } 1515 } else if (u > server) { 1516 ALOGW("stepUser occurred after track reset"); 1517 u = server; 1518 } 1519 1520 uint32_t fc = this->frameCount; 1521 if (u >= fc) { 1522 // common case, user didn't just wrap 1523 if (u - fc >= userBase ) { 1524 userBase += fc; 1525 } 1526 } else if (u >= userBase + fc) { 1527 // user just wrapped 1528 userBase += fc; 1529 } 1530 1531 user = u; 1532 1533 // Clear flow control error condition as new data has been written/read to/from buffer. 1534 if (flags & CBLK_UNDERRUN_MSK) { 1535 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1536 } 1537 1538 return u; 1539} 1540 1541bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1542{ 1543 ALOGV("stepserver %08x %08x %d", user, server, frameCount); 1544 1545 if (!tryLock()) { 1546 ALOGW("stepServer() could not lock cblk"); 1547 return false; 1548 } 1549 1550 uint32_t s = server; 1551 bool flushed = (s == user); 1552 1553 s += frameCount; 1554 if (flags & CBLK_DIRECTION_MSK) { 1555 // Mark that we have read the first buffer so that next time stepUser() is called 1556 // we switch to normal obtainBuffer() timeout period 1557 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1558 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1559 } 1560 // It is possible that we receive a flush() 1561 // while the mixer is processing a block: in this case, 1562 // stepServer() is called After the flush() has reset u & s and 1563 // we have s > u 1564 if (flushed) { 1565 ALOGW("stepServer occurred after track reset"); 1566 s = user; 1567 } 1568 } 1569 1570 if (s >= loopEnd) { 1571 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1572 s = loopStart; 1573 if (--loopCount == 0) { 1574 loopEnd = UINT_MAX; 1575 loopStart = UINT_MAX; 1576 } 1577 } 1578 1579 uint32_t fc = this->frameCount; 1580 if (s >= fc) { 1581 // common case, server didn't just wrap 1582 if (s - fc >= serverBase ) { 1583 serverBase += fc; 1584 } 1585 } else if (s >= serverBase + fc) { 1586 // server just wrapped 1587 serverBase += fc; 1588 } 1589 1590 server = s; 1591 1592 if (!(flags & CBLK_INVALID_MSK)) { 1593 cv.signal(); 1594 } 1595 lock.unlock(); 1596 return true; 1597} 1598 1599void* audio_track_cblk_t::buffer(uint32_t offset) const 1600{ 1601 return (int8_t *)buffers + (offset - userBase) * frameSize; 1602} 1603 1604uint32_t audio_track_cblk_t::framesAvailable() 1605{ 1606 Mutex::Autolock _l(lock); 1607 return framesAvailable_l(); 1608} 1609 1610uint32_t audio_track_cblk_t::framesAvailable_l() 1611{ 1612 uint32_t u = user; 1613 uint32_t s = server; 1614 1615 if (flags & CBLK_DIRECTION_MSK) { 1616 uint32_t limit = (s < loopStart) ? s : loopStart; 1617 return limit + frameCount - u; 1618 } else { 1619 return frameCount + u - s; 1620 } 1621} 1622 1623uint32_t audio_track_cblk_t::framesReady() 1624{ 1625 uint32_t u = user; 1626 uint32_t s = server; 1627 1628 if (flags & CBLK_DIRECTION_MSK) { 1629 if (u < loopEnd) { 1630 return u - s; 1631 } else { 1632 // do not block on mutex shared with client on AudioFlinger side 1633 if (!tryLock()) { 1634 ALOGW("framesReady() could not lock cblk"); 1635 return 0; 1636 } 1637 uint32_t frames = UINT_MAX; 1638 if (loopCount >= 0) { 1639 frames = (loopEnd - loopStart)*loopCount + u - s; 1640 } 1641 lock.unlock(); 1642 return frames; 1643 } 1644 } else { 1645 return s - u; 1646 } 1647} 1648 1649bool audio_track_cblk_t::tryLock() 1650{ 1651 // the code below simulates lock-with-timeout 1652 // we MUST do this to protect the AudioFlinger server 1653 // as this lock is shared with the client. 1654 status_t err; 1655 1656 err = lock.tryLock(); 1657 if (err == -EBUSY) { // just wait a bit 1658 usleep(1000); 1659 err = lock.tryLock(); 1660 } 1661 if (err != NO_ERROR) { 1662 // probably, the client just died. 1663 return false; 1664 } 1665 return true; 1666} 1667 1668// ------------------------------------------------------------------------- 1669 1670}; // namespace android 1671