AudioTrack.cpp revision ec7dcac79c121ef015ee237891a5c90e67a977ab
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 size_t* frameCount, 54 audio_stream_type_t streamType, 55 uint32_t sampleRate) 56{ 57 if (frameCount == NULL) { 58 return BAD_VALUE; 59 } 60 61 // default to 0 in case of error 62 *frameCount = 0; 63 64 // FIXME merge with similar code in createTrack_l(), except we're missing 65 // some information here that is available in createTrack_l(): 66 // audio_io_handle_t output 67 // audio_format_t format 68 // audio_channel_mask_t channelMask 69 // audio_output_flags_t flags 70 uint32_t afSampleRate; 71 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 72 return NO_INIT; 73 } 74 size_t afFrameCount; 75 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 76 return NO_INIT; 77 } 78 uint32_t afLatency; 79 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 80 return NO_INIT; 81 } 82 83 // Ensure that buffer depth covers at least audio hardware latency 84 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 85 if (minBufCount < 2) minBufCount = 2; 86 87 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 88 afFrameCount * minBufCount * sampleRate / afSampleRate; 89 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 90 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 91 return NO_ERROR; 92} 93 94// --------------------------------------------------------------------------- 95 96AudioTrack::AudioTrack() 97 : mStatus(NO_INIT), 98 mIsTimed(false), 99 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 100 mPreviousSchedulingGroup(SP_DEFAULT) 101{ 102} 103 104AudioTrack::AudioTrack( 105 audio_stream_type_t streamType, 106 uint32_t sampleRate, 107 audio_format_t format, 108 audio_channel_mask_t channelMask, 109 int frameCount, 110 audio_output_flags_t flags, 111 callback_t cbf, 112 void* user, 113 int notificationFrames, 114 int sessionId) 115 : mStatus(NO_INIT), 116 mIsTimed(false), 117 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 118 mPreviousSchedulingGroup(SP_DEFAULT) 119{ 120 mStatus = set(streamType, sampleRate, format, channelMask, 121 frameCount, flags, cbf, user, notificationFrames, 122 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); 123} 124 125AudioTrack::AudioTrack( 126 audio_stream_type_t streamType, 127 uint32_t sampleRate, 128 audio_format_t format, 129 audio_channel_mask_t channelMask, 130 const sp<IMemory>& sharedBuffer, 131 audio_output_flags_t flags, 132 callback_t cbf, 133 void* user, 134 int notificationFrames, 135 int sessionId) 136 : mStatus(NO_INIT), 137 mIsTimed(false), 138 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 139 mPreviousSchedulingGroup(SP_DEFAULT) 140{ 141 mStatus = set(streamType, sampleRate, format, channelMask, 142 0 /*frameCount*/, flags, cbf, user, notificationFrames, 143 sharedBuffer, false /*threadCanCallJava*/, sessionId); 144} 145 146AudioTrack::~AudioTrack() 147{ 148 ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 149 150 if (mStatus == NO_ERROR) { 151 // Make sure that callback function exits in the case where 152 // it is looping on buffer full condition in obtainBuffer(). 153 // Otherwise the callback thread will never exit. 154 stop(); 155 if (mAudioTrackThread != 0) { 156 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 157 mAudioTrackThread->requestExitAndWait(); 158 mAudioTrackThread.clear(); 159 } 160 mAudioTrack.clear(); 161 IPCThreadState::self()->flushCommands(); 162 AudioSystem::releaseAudioSessionId(mSessionId); 163 } 164} 165 166status_t AudioTrack::set( 167 audio_stream_type_t streamType, 168 uint32_t sampleRate, 169 audio_format_t format, 170 audio_channel_mask_t channelMask, 171 int frameCountInt, 172 audio_output_flags_t flags, 173 callback_t cbf, 174 void* user, 175 int notificationFrames, 176 const sp<IMemory>& sharedBuffer, 177 bool threadCanCallJava, 178 int sessionId) 179{ 180 // FIXME "int" here is legacy and will be replaced by size_t later 181 if (frameCountInt < 0) { 182 ALOGE("Invalid frame count %d", frameCountInt); 183 return BAD_VALUE; 184 } 185 size_t frameCount = frameCountInt; 186 187 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 188 sharedBuffer->size()); 189 190 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 191 192 AutoMutex lock(mLock); 193 if (mAudioTrack != 0) { 194 ALOGE("Track already in use"); 195 return INVALID_OPERATION; 196 } 197 198 // handle default values first. 199 if (streamType == AUDIO_STREAM_DEFAULT) { 200 streamType = AUDIO_STREAM_MUSIC; 201 } 202 203 if (sampleRate == 0) { 204 uint32_t afSampleRate; 205 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 206 return NO_INIT; 207 } 208 sampleRate = afSampleRate; 209 } 210 211 // these below should probably come from the audioFlinger too... 212 if (format == AUDIO_FORMAT_DEFAULT) { 213 format = AUDIO_FORMAT_PCM_16_BIT; 214 } 215 if (channelMask == 0) { 216 channelMask = AUDIO_CHANNEL_OUT_STEREO; 217 } 218 219 // validate parameters 220 if (!audio_is_valid_format(format)) { 221 ALOGE("Invalid format"); 222 return BAD_VALUE; 223 } 224 225 // AudioFlinger does not currently support 8-bit data in shared memory 226 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 227 ALOGE("8-bit data in shared memory is not supported"); 228 return BAD_VALUE; 229 } 230 231 // force direct flag if format is not linear PCM 232 if (!audio_is_linear_pcm(format)) { 233 flags = (audio_output_flags_t) 234 // FIXME why can't we allow direct AND fast? 235 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 236 } 237 // only allow deep buffering for music stream type 238 if (streamType != AUDIO_STREAM_MUSIC) { 239 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 240 } 241 242 if (!audio_is_output_channel(channelMask)) { 243 ALOGE("Invalid channel mask %#x", channelMask); 244 return BAD_VALUE; 245 } 246 mChannelMask = channelMask; 247 uint32_t channelCount = popcount(channelMask); 248 mChannelCount = channelCount; 249 250 audio_io_handle_t output = AudioSystem::getOutput( 251 streamType, 252 sampleRate, format, channelMask, 253 flags); 254 255 if (output == 0) { 256 ALOGE("Could not get audio output for stream type %d", streamType); 257 return BAD_VALUE; 258 } 259 260 mVolume[LEFT] = 1.0f; 261 mVolume[RIGHT] = 1.0f; 262 mSendLevel = 0.0f; 263 mFrameCount = frameCount; 264 mReqFrameCount = frameCount; 265 mNotificationFramesReq = notificationFrames; 266 mSessionId = sessionId; 267 mAuxEffectId = 0; 268 mFlags = flags; 269 mCbf = cbf; 270 271 if (cbf != NULL) { 272 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 273 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 274 } 275 276 // create the IAudioTrack 277 status_t status = createTrack_l(streamType, 278 sampleRate, 279 format, 280 frameCount, 281 flags, 282 sharedBuffer, 283 output); 284 285 if (status != NO_ERROR) { 286 if (mAudioTrackThread != 0) { 287 mAudioTrackThread->requestExit(); 288 mAudioTrackThread.clear(); 289 } 290 return status; 291 } 292 293 mStatus = NO_ERROR; 294 295 mStreamType = streamType; 296 mFormat = format; 297 298 if (audio_is_linear_pcm(format)) { 299 mFrameSize = channelCount * audio_bytes_per_sample(format); 300 mFrameSizeAF = channelCount * sizeof(int16_t); 301 } else { 302 mFrameSize = sizeof(uint8_t); 303 mFrameSizeAF = sizeof(uint8_t); 304 } 305 306 mSharedBuffer = sharedBuffer; 307 mActive = false; 308 mUserData = user; 309 mLoopCount = 0; 310 mMarkerPosition = 0; 311 mMarkerReached = false; 312 mNewPosition = 0; 313 mUpdatePeriod = 0; 314 mFlushed = false; 315 AudioSystem::acquireAudioSessionId(mSessionId); 316 return NO_ERROR; 317} 318 319// ------------------------------------------------------------------------- 320 321void AudioTrack::start() 322{ 323 sp<AudioTrackThread> t = mAudioTrackThread; 324 325 ALOGV("start %p", this); 326 327 AutoMutex lock(mLock); 328 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 329 // while we are accessing the cblk 330 sp<IAudioTrack> audioTrack = mAudioTrack; 331 sp<IMemory> iMem = mCblkMemory; 332 audio_track_cblk_t* cblk = mCblk; 333 334 if (!mActive) { 335 mFlushed = false; 336 mActive = true; 337 mNewPosition = cblk->server + mUpdatePeriod; 338 cblk->lock.lock(); 339 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 340 cblk->waitTimeMs = 0; 341 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 342 if (t != 0) { 343 t->resume(); 344 } else { 345 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 346 get_sched_policy(0, &mPreviousSchedulingGroup); 347 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 348 } 349 350 ALOGV("start %p before lock cblk %p", this, cblk); 351 status_t status = NO_ERROR; 352 if (!(cblk->flags & CBLK_INVALID)) { 353 cblk->lock.unlock(); 354 ALOGV("mAudioTrack->start()"); 355 status = mAudioTrack->start(); 356 cblk->lock.lock(); 357 if (status == DEAD_OBJECT) { 358 android_atomic_or(CBLK_INVALID, &cblk->flags); 359 } 360 } 361 if (cblk->flags & CBLK_INVALID) { 362 audio_track_cblk_t* temp = cblk; 363 status = restoreTrack_l(temp, true /*fromStart*/); 364 cblk = temp; 365 } 366 cblk->lock.unlock(); 367 if (status != NO_ERROR) { 368 ALOGV("start() failed"); 369 mActive = false; 370 if (t != 0) { 371 t->pause(); 372 } else { 373 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 374 set_sched_policy(0, mPreviousSchedulingGroup); 375 } 376 } 377 } 378 379} 380 381void AudioTrack::stop() 382{ 383 sp<AudioTrackThread> t = mAudioTrackThread; 384 385 ALOGV("stop %p", this); 386 387 AutoMutex lock(mLock); 388 if (mActive) { 389 mActive = false; 390 mCblk->cv.signal(); 391 mAudioTrack->stop(); 392 // Cancel loops (If we are in the middle of a loop, playback 393 // would not stop until loopCount reaches 0). 394 setLoop_l(0, 0, 0); 395 // the playback head position will reset to 0, so if a marker is set, we need 396 // to activate it again 397 mMarkerReached = false; 398 // Force flush if a shared buffer is used otherwise audioflinger 399 // will not stop before end of buffer is reached. 400 // It may be needed to make sure that we stop playback, likely in case looping is on. 401 if (mSharedBuffer != 0) { 402 flush_l(); 403 } 404 if (t != 0) { 405 t->pause(); 406 } else { 407 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 408 set_sched_policy(0, mPreviousSchedulingGroup); 409 } 410 } 411 412} 413 414bool AudioTrack::stopped() const 415{ 416 AutoMutex lock(mLock); 417 return stopped_l(); 418} 419 420void AudioTrack::flush() 421{ 422 AutoMutex lock(mLock); 423 if (!mActive && mSharedBuffer == 0) { 424 flush_l(); 425 } 426} 427 428void AudioTrack::flush_l() 429{ 430 ALOGV("flush"); 431 ALOG_ASSERT(!mActive); 432 433 // clear playback marker and periodic update counter 434 mMarkerPosition = 0; 435 mMarkerReached = false; 436 mUpdatePeriod = 0; 437 438 mFlushed = true; 439 mAudioTrack->flush(); 440 // Release AudioTrack callback thread in case it was waiting for new buffers 441 // in AudioTrack::obtainBuffer() 442 mCblk->cv.signal(); 443} 444 445void AudioTrack::pause() 446{ 447 ALOGV("pause"); 448 AutoMutex lock(mLock); 449 if (mActive) { 450 mActive = false; 451 mCblk->cv.signal(); 452 mAudioTrack->pause(); 453 } 454} 455 456status_t AudioTrack::setVolume(float left, float right) 457{ 458 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 459 return BAD_VALUE; 460 } 461 462 AutoMutex lock(mLock); 463 mVolume[LEFT] = left; 464 mVolume[RIGHT] = right; 465 466 mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 467 468 return NO_ERROR; 469} 470 471status_t AudioTrack::setVolume(float volume) 472{ 473 return setVolume(volume, volume); 474} 475 476status_t AudioTrack::setAuxEffectSendLevel(float level) 477{ 478 ALOGV("setAuxEffectSendLevel(%f)", level); 479 if (level < 0.0f || level > 1.0f) { 480 return BAD_VALUE; 481 } 482 AutoMutex lock(mLock); 483 484 mSendLevel = level; 485 486 mCblk->setSendLevel(level); 487 488 return NO_ERROR; 489} 490 491void AudioTrack::getAuxEffectSendLevel(float* level) const 492{ 493 if (level != NULL) { 494 *level = mSendLevel; 495 } 496} 497 498status_t AudioTrack::setSampleRate(uint32_t rate) 499{ 500 uint32_t afSamplingRate; 501 502 if (mIsTimed) { 503 return INVALID_OPERATION; 504 } 505 506 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 507 return NO_INIT; 508 } 509 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 510 if (rate == 0 || rate > afSamplingRate*2 ) { 511 return BAD_VALUE; 512 } 513 514 AutoMutex lock(mLock); 515 mCblk->sampleRate = rate; 516 return NO_ERROR; 517} 518 519uint32_t AudioTrack::getSampleRate() const 520{ 521 if (mIsTimed) { 522 return 0; 523 } 524 525 AutoMutex lock(mLock); 526 return mCblk->sampleRate; 527} 528 529status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 530{ 531 AutoMutex lock(mLock); 532 return setLoop_l(loopStart, loopEnd, loopCount); 533} 534 535// must be called with mLock held 536status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 537{ 538 audio_track_cblk_t* cblk = mCblk; 539 540 Mutex::Autolock _l(cblk->lock); 541 542 if (loopCount == 0) { 543 cblk->loopStart = UINT_MAX; 544 cblk->loopEnd = UINT_MAX; 545 cblk->loopCount = 0; 546 mLoopCount = 0; 547 return NO_ERROR; 548 } 549 550 if (mIsTimed) { 551 return INVALID_OPERATION; 552 } 553 554 if (loopStart >= loopEnd || 555 loopEnd - loopStart > mFrameCount || 556 cblk->server > loopStart) { 557 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, " 558 "user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user); 559 return BAD_VALUE; 560 } 561 562 if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) { 563 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " 564 "framecount %d", 565 loopStart, loopEnd, mFrameCount); 566 return BAD_VALUE; 567 } 568 569 cblk->loopStart = loopStart; 570 cblk->loopEnd = loopEnd; 571 cblk->loopCount = loopCount; 572 mLoopCount = loopCount; 573 574 return NO_ERROR; 575} 576 577status_t AudioTrack::setMarkerPosition(uint32_t marker) 578{ 579 if (mCbf == NULL) { 580 return INVALID_OPERATION; 581 } 582 583 mMarkerPosition = marker; 584 mMarkerReached = false; 585 586 return NO_ERROR; 587} 588 589status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 590{ 591 if (marker == NULL) { 592 return BAD_VALUE; 593 } 594 595 *marker = mMarkerPosition; 596 597 return NO_ERROR; 598} 599 600status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 601{ 602 if (mCbf == NULL) { 603 return INVALID_OPERATION; 604 } 605 606 uint32_t curPosition; 607 getPosition(&curPosition); 608 mNewPosition = curPosition + updatePeriod; 609 mUpdatePeriod = updatePeriod; 610 611 return NO_ERROR; 612} 613 614status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 615{ 616 if (updatePeriod == NULL) { 617 return BAD_VALUE; 618 } 619 620 *updatePeriod = mUpdatePeriod; 621 622 return NO_ERROR; 623} 624 625status_t AudioTrack::setPosition(uint32_t position) 626{ 627 if (mIsTimed) { 628 return INVALID_OPERATION; 629 } 630 631 AutoMutex lock(mLock); 632 633 if (!stopped_l()) { 634 return INVALID_OPERATION; 635 } 636 637 audio_track_cblk_t* cblk = mCblk; 638 Mutex::Autolock _l(cblk->lock); 639 640 if (position > cblk->user) { 641 return BAD_VALUE; 642 } 643 644 cblk->server = position; 645 android_atomic_or(CBLK_FORCEREADY, &cblk->flags); 646 647 return NO_ERROR; 648} 649 650status_t AudioTrack::getPosition(uint32_t *position) 651{ 652 if (position == NULL) { 653 return BAD_VALUE; 654 } 655 AutoMutex lock(mLock); 656 *position = mFlushed ? 0 : mCblk->server; 657 658 return NO_ERROR; 659} 660 661status_t AudioTrack::reload() 662{ 663 AutoMutex lock(mLock); 664 665 if (!stopped_l()) { 666 return INVALID_OPERATION; 667 } 668 669 flush_l(); 670 671 audio_track_cblk_t* cblk = mCblk; 672 cblk->stepUserOut(mFrameCount, mFrameCount); 673 674 return NO_ERROR; 675} 676 677audio_io_handle_t AudioTrack::getOutput() 678{ 679 AutoMutex lock(mLock); 680 return getOutput_l(); 681} 682 683// must be called with mLock held 684audio_io_handle_t AudioTrack::getOutput_l() 685{ 686 return AudioSystem::getOutput(mStreamType, 687 mCblk->sampleRate, mFormat, mChannelMask, mFlags); 688} 689 690status_t AudioTrack::attachAuxEffect(int effectId) 691{ 692 ALOGV("attachAuxEffect(%d)", effectId); 693 status_t status = mAudioTrack->attachAuxEffect(effectId); 694 if (status == NO_ERROR) { 695 mAuxEffectId = effectId; 696 } 697 return status; 698} 699 700// ------------------------------------------------------------------------- 701 702// must be called with mLock held 703status_t AudioTrack::createTrack_l( 704 audio_stream_type_t streamType, 705 uint32_t sampleRate, 706 audio_format_t format, 707 size_t frameCount, 708 audio_output_flags_t flags, 709 const sp<IMemory>& sharedBuffer, 710 audio_io_handle_t output) 711{ 712 status_t status; 713 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 714 if (audioFlinger == 0) { 715 ALOGE("Could not get audioflinger"); 716 return NO_INIT; 717 } 718 719 uint32_t afLatency; 720 if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { 721 return NO_INIT; 722 } 723 724 // Client decides whether the track is TIMED (see below), but can only express a preference 725 // for FAST. Server will perform additional tests. 726 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 727 // either of these use cases: 728 // use case 1: shared buffer 729 (sharedBuffer != 0) || 730 // use case 2: callback handler 731 (mCbf != NULL))) { 732 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 733 // once denied, do not request again if IAudioTrack is re-created 734 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 735 mFlags = flags; 736 } 737 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 738 739 mNotificationFramesAct = mNotificationFramesReq; 740 741 if (!audio_is_linear_pcm(format)) { 742 743 if (sharedBuffer != 0) { 744 // Same comment as below about ignoring frameCount parameter for set() 745 frameCount = sharedBuffer->size(); 746 } else if (frameCount == 0) { 747 size_t afFrameCount; 748 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 749 return NO_INIT; 750 } 751 frameCount = afFrameCount; 752 } 753 754 } else if (sharedBuffer != 0) { 755 756 // Ensure that buffer alignment matches channel count 757 // 8-bit data in shared memory is not currently supported by AudioFlinger 758 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 759 if (mChannelCount > 1) { 760 // More than 2 channels does not require stronger alignment than stereo 761 alignment <<= 1; 762 } 763 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 764 ALOGE("Invalid buffer alignment: address %p, channel count %u", 765 sharedBuffer->pointer(), mChannelCount); 766 return BAD_VALUE; 767 } 768 769 // When initializing a shared buffer AudioTrack via constructors, 770 // there's no frameCount parameter. 771 // But when initializing a shared buffer AudioTrack via set(), 772 // there _is_ a frameCount parameter. We silently ignore it. 773 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 774 775 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 776 777 // FIXME move these calculations and associated checks to server 778 uint32_t afSampleRate; 779 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 780 return NO_INIT; 781 } 782 size_t afFrameCount; 783 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 784 return NO_INIT; 785 } 786 787 // Ensure that buffer depth covers at least audio hardware latency 788 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 789 if (minBufCount < 2) minBufCount = 2; 790 791 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 792 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 793 ", afLatency=%d", 794 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 795 796 if (frameCount == 0) { 797 frameCount = minFrameCount; 798 } 799 if (mNotificationFramesAct == 0) { 800 mNotificationFramesAct = frameCount/2; 801 } 802 // Make sure that application is notified with sufficient margin 803 // before underrun 804 if (mNotificationFramesAct > frameCount/2) { 805 mNotificationFramesAct = frameCount/2; 806 } 807 if (frameCount < minFrameCount) { 808 // not ALOGW because it happens all the time when playing key clicks over A2DP 809 ALOGV("Minimum buffer size corrected from %d to %d", 810 frameCount, minFrameCount); 811 frameCount = minFrameCount; 812 } 813 814 } else { 815 // For fast tracks, the frame count calculations and checks are done by server 816 } 817 818 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 819 if (mIsTimed) { 820 trackFlags |= IAudioFlinger::TRACK_TIMED; 821 } 822 823 pid_t tid = -1; 824 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 825 trackFlags |= IAudioFlinger::TRACK_FAST; 826 if (mAudioTrackThread != 0) { 827 tid = mAudioTrackThread->getTid(); 828 } 829 } 830 831 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 832 streamType, 833 sampleRate, 834 // AudioFlinger only sees 16-bit PCM 835 format == AUDIO_FORMAT_PCM_8_BIT ? 836 AUDIO_FORMAT_PCM_16_BIT : format, 837 mChannelMask, 838 frameCount, 839 &trackFlags, 840 sharedBuffer, 841 output, 842 tid, 843 &mSessionId, 844 &status); 845 846 if (track == 0) { 847 ALOGE("AudioFlinger could not create track, status: %d", status); 848 return status; 849 } 850 sp<IMemory> iMem = track->getCblk(); 851 if (iMem == 0) { 852 ALOGE("Could not get control block"); 853 return NO_INIT; 854 } 855 mAudioTrack = track; 856 mCblkMemory = iMem; 857 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 858 mCblk = cblk; 859 size_t temp = cblk->frameCount_; 860 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 861 // In current design, AudioTrack client checks and ensures frame count validity before 862 // passing it to AudioFlinger so AudioFlinger should not return a different value except 863 // for fast track as it uses a special method of assigning frame count. 864 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 865 } 866 frameCount = temp; 867 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 868 if (trackFlags & IAudioFlinger::TRACK_FAST) { 869 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 870 } else { 871 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 872 // once denied, do not request again if IAudioTrack is re-created 873 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 874 mFlags = flags; 875 } 876 if (sharedBuffer == 0) { 877 mNotificationFramesAct = frameCount/2; 878 } 879 } 880 if (sharedBuffer == 0) { 881 mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); 882 } else { 883 mBuffers = sharedBuffer->pointer(); 884 // Force buffer full condition as data is already present in shared memory 885 cblk->stepUserOut(frameCount, frameCount); 886 } 887 888 cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 889 uint16_t(mVolume[LEFT] * 0x1000)); 890 cblk->setSendLevel(mSendLevel); 891 mAudioTrack->attachAuxEffect(mAuxEffectId); 892 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 893 cblk->waitTimeMs = 0; 894 mRemainingFrames = mNotificationFramesAct; 895 // FIXME don't believe this lie 896 mLatency = afLatency + (1000*frameCount) / sampleRate; 897 mFrameCount = frameCount; 898 // If IAudioTrack is re-created, don't let the requested frameCount 899 // decrease. This can confuse clients that cache frameCount(). 900 if (frameCount > mReqFrameCount) { 901 mReqFrameCount = frameCount; 902 } 903 return NO_ERROR; 904} 905 906status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 907{ 908 AutoMutex lock(mLock); 909 bool active; 910 status_t result = NO_ERROR; 911 audio_track_cblk_t* cblk = mCblk; 912 uint32_t framesReq = audioBuffer->frameCount; 913 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 914 915 audioBuffer->frameCount = 0; 916 audioBuffer->size = 0; 917 918 uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); 919 920 cblk->lock.lock(); 921 if (cblk->flags & CBLK_INVALID) { 922 goto create_new_track; 923 } 924 cblk->lock.unlock(); 925 926 if (framesAvail == 0) { 927 cblk->lock.lock(); 928 goto start_loop_here; 929 while (framesAvail == 0) { 930 active = mActive; 931 if (CC_UNLIKELY(!active)) { 932 ALOGV("Not active and NO_MORE_BUFFERS"); 933 cblk->lock.unlock(); 934 return NO_MORE_BUFFERS; 935 } 936 if (CC_UNLIKELY(!waitCount)) { 937 cblk->lock.unlock(); 938 return WOULD_BLOCK; 939 } 940 if (!(cblk->flags & CBLK_INVALID)) { 941 mLock.unlock(); 942 // this condition is in shared memory, so if IAudioTrack and control block 943 // are replaced due to mediaserver death or IAudioTrack invalidation then 944 // cv won't be signalled, but fortunately the timeout will limit the wait 945 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 946 cblk->lock.unlock(); 947 mLock.lock(); 948 if (!mActive) { 949 return status_t(STOPPED); 950 } 951 // IAudioTrack may have been re-created while mLock was unlocked 952 cblk = mCblk; 953 cblk->lock.lock(); 954 } 955 956 if (cblk->flags & CBLK_INVALID) { 957 goto create_new_track; 958 } 959 if (CC_UNLIKELY(result != NO_ERROR)) { 960 cblk->waitTimeMs += waitTimeMs; 961 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 962 // timing out when a loop has been set and we have already written upto loop end 963 // is a normal condition: no need to wake AudioFlinger up. 964 if (cblk->user < cblk->loopEnd) { 965 ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " 966 "server=%08x", this, cblk->mName, cblk->user, cblk->server); 967 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 968 cblk->lock.unlock(); 969 result = mAudioTrack->start(); 970 cblk->lock.lock(); 971 if (result == DEAD_OBJECT) { 972 android_atomic_or(CBLK_INVALID, &cblk->flags); 973create_new_track: 974 audio_track_cblk_t* temp = cblk; 975 result = restoreTrack_l(temp, false /*fromStart*/); 976 cblk = temp; 977 } 978 if (result != NO_ERROR) { 979 ALOGW("obtainBuffer create Track error %d", result); 980 cblk->lock.unlock(); 981 return result; 982 } 983 } 984 cblk->waitTimeMs = 0; 985 } 986 987 if (--waitCount == 0) { 988 cblk->lock.unlock(); 989 return TIMED_OUT; 990 } 991 } 992 // read the server count again 993 start_loop_here: 994 framesAvail = cblk->framesAvailableOut_l(mFrameCount); 995 } 996 cblk->lock.unlock(); 997 } 998 999 cblk->waitTimeMs = 0; 1000 1001 if (framesReq > framesAvail) { 1002 framesReq = framesAvail; 1003 } 1004 1005 uint32_t u = cblk->user; 1006 uint32_t bufferEnd = cblk->userBase + mFrameCount; 1007 1008 if (framesReq > bufferEnd - u) { 1009 framesReq = bufferEnd - u; 1010 } 1011 1012 audioBuffer->frameCount = framesReq; 1013 audioBuffer->size = framesReq * mFrameSizeAF; 1014 audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u); 1015 active = mActive; 1016 return active ? status_t(NO_ERROR) : status_t(STOPPED); 1017} 1018 1019void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1020{ 1021 AutoMutex lock(mLock); 1022 audio_track_cblk_t* cblk = mCblk; 1023 cblk->stepUserOut(audioBuffer->frameCount, mFrameCount); 1024 if (audioBuffer->frameCount > 0) { 1025 // restart track if it was disabled by audioflinger due to previous underrun 1026 if (mActive && (cblk->flags & CBLK_DISABLED)) { 1027 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1028 ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName); 1029 mAudioTrack->start(); 1030 } 1031 } 1032} 1033 1034// ------------------------------------------------------------------------- 1035 1036ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1037{ 1038 1039 if (mSharedBuffer != 0) { 1040 return INVALID_OPERATION; 1041 } 1042 if (mIsTimed) { 1043 return INVALID_OPERATION; 1044 } 1045 1046 if (ssize_t(userSize) < 0) { 1047 // Sanity-check: user is most-likely passing an error code, and it would 1048 // make the return value ambiguous (actualSize vs error). 1049 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1050 buffer, userSize, userSize); 1051 return BAD_VALUE; 1052 } 1053 1054 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1055 1056 if (userSize == 0) { 1057 return 0; 1058 } 1059 1060 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1061 // while we are accessing the cblk 1062 mLock.lock(); 1063 sp<IAudioTrack> audioTrack = mAudioTrack; 1064 sp<IMemory> iMem = mCblkMemory; 1065 mLock.unlock(); 1066 1067 // since mLock is unlocked the IAudioTrack and shared memory may be re-created, 1068 // so all cblk references might still refer to old shared memory, but that should be benign 1069 1070 ssize_t written = 0; 1071 const int8_t *src = (const int8_t *)buffer; 1072 Buffer audioBuffer; 1073 size_t frameSz = frameSize(); 1074 1075 do { 1076 audioBuffer.frameCount = userSize/frameSz; 1077 1078 status_t err = obtainBuffer(&audioBuffer, -1); 1079 if (err < 0) { 1080 // out of buffers, return #bytes written 1081 if (err == status_t(NO_MORE_BUFFERS)) { 1082 break; 1083 } 1084 return ssize_t(err); 1085 } 1086 1087 size_t toWrite; 1088 1089 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1090 // Divide capacity by 2 to take expansion into account 1091 toWrite = audioBuffer.size>>1; 1092 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1093 } else { 1094 toWrite = audioBuffer.size; 1095 memcpy(audioBuffer.i8, src, toWrite); 1096 } 1097 src += toWrite; 1098 userSize -= toWrite; 1099 written += toWrite; 1100 1101 releaseBuffer(&audioBuffer); 1102 } while (userSize >= frameSz); 1103 1104 return written; 1105} 1106 1107// ------------------------------------------------------------------------- 1108 1109TimedAudioTrack::TimedAudioTrack() { 1110 mIsTimed = true; 1111} 1112 1113status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1114{ 1115 AutoMutex lock(mLock); 1116 status_t result = UNKNOWN_ERROR; 1117 1118 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1119 // while we are accessing the cblk 1120 sp<IAudioTrack> audioTrack = mAudioTrack; 1121 sp<IMemory> iMem = mCblkMemory; 1122 1123 // If the track is not invalid already, try to allocate a buffer. alloc 1124 // fails indicating that the server is dead, flag the track as invalid so 1125 // we can attempt to restore in just a bit. 1126 audio_track_cblk_t* cblk = mCblk; 1127 if (!(cblk->flags & CBLK_INVALID)) { 1128 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1129 if (result == DEAD_OBJECT) { 1130 android_atomic_or(CBLK_INVALID, &cblk->flags); 1131 } 1132 } 1133 1134 // If the track is invalid at this point, attempt to restore it. and try the 1135 // allocation one more time. 1136 if (cblk->flags & CBLK_INVALID) { 1137 cblk->lock.lock(); 1138 audio_track_cblk_t* temp = cblk; 1139 result = restoreTrack_l(temp, false /*fromStart*/); 1140 cblk = temp; 1141 cblk->lock.unlock(); 1142 1143 if (result == OK) { 1144 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1145 } 1146 } 1147 1148 return result; 1149} 1150 1151status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1152 int64_t pts) 1153{ 1154 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1155 { 1156 AutoMutex lock(mLock); 1157 audio_track_cblk_t* cblk = mCblk; 1158 // restart track if it was disabled by audioflinger due to previous underrun 1159 if (buffer->size() != 0 && status == NO_ERROR && 1160 mActive && (cblk->flags & CBLK_DISABLED)) { 1161 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1162 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1163 mAudioTrack->start(); 1164 } 1165 } 1166 return status; 1167} 1168 1169status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1170 TargetTimeline target) 1171{ 1172 return mAudioTrack->setMediaTimeTransform(xform, target); 1173} 1174 1175// ------------------------------------------------------------------------- 1176 1177bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1178{ 1179 Buffer audioBuffer; 1180 uint32_t frames; 1181 size_t writtenSize; 1182 1183 mLock.lock(); 1184 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1185 // while we are accessing the cblk 1186 sp<IAudioTrack> audioTrack = mAudioTrack; 1187 sp<IMemory> iMem = mCblkMemory; 1188 audio_track_cblk_t* cblk = mCblk; 1189 bool active = mActive; 1190 mLock.unlock(); 1191 1192 // since mLock is unlocked the IAudioTrack and shared memory may be re-created, 1193 // so all cblk references might still refer to old shared memory, but that should be benign 1194 1195 // Manage underrun callback 1196 if (active && (cblk->framesAvailableOut(mFrameCount) == mFrameCount)) { 1197 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1198 if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { 1199 mCbf(EVENT_UNDERRUN, mUserData, 0); 1200 if (cblk->server == mFrameCount) { 1201 mCbf(EVENT_BUFFER_END, mUserData, 0); 1202 } 1203 if (mSharedBuffer != 0) { 1204 return false; 1205 } 1206 } 1207 } 1208 1209 // Manage loop end callback 1210 while (mLoopCount > cblk->loopCount) { 1211 int loopCount = -1; 1212 mLoopCount--; 1213 if (mLoopCount >= 0) loopCount = mLoopCount; 1214 1215 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1216 } 1217 1218 // Manage marker callback 1219 if (!mMarkerReached && (mMarkerPosition > 0)) { 1220 if (cblk->server >= mMarkerPosition) { 1221 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1222 mMarkerReached = true; 1223 } 1224 } 1225 1226 // Manage new position callback 1227 if (mUpdatePeriod > 0) { 1228 while (cblk->server >= mNewPosition) { 1229 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1230 mNewPosition += mUpdatePeriod; 1231 } 1232 } 1233 1234 // If Shared buffer is used, no data is requested from client. 1235 if (mSharedBuffer != 0) { 1236 frames = 0; 1237 } else { 1238 frames = mRemainingFrames; 1239 } 1240 1241 // See description of waitCount parameter at declaration of obtainBuffer(). 1242 // The logic below prevents us from being stuck below at obtainBuffer() 1243 // not being able to handle timed events (position, markers, loops). 1244 int32_t waitCount = -1; 1245 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1246 waitCount = 1; 1247 } 1248 1249 do { 1250 1251 audioBuffer.frameCount = frames; 1252 1253 status_t err = obtainBuffer(&audioBuffer, waitCount); 1254 if (err < NO_ERROR) { 1255 if (err != TIMED_OUT) { 1256 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), 1257 "Error obtaining an audio buffer, giving up."); 1258 return false; 1259 } 1260 break; 1261 } 1262 if (err == status_t(STOPPED)) { 1263 return false; 1264 } 1265 1266 // Divide buffer size by 2 to take into account the expansion 1267 // due to 8 to 16 bit conversion: the callback must fill only half 1268 // of the destination buffer 1269 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1270 audioBuffer.size >>= 1; 1271 } 1272 1273 size_t reqSize = audioBuffer.size; 1274 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1275 writtenSize = audioBuffer.size; 1276 1277 // Sanity check on returned size 1278 if (ssize_t(writtenSize) <= 0) { 1279 // The callback is done filling buffers 1280 // Keep this thread going to handle timed events and 1281 // still try to get more data in intervals of WAIT_PERIOD_MS 1282 // but don't just loop and block the CPU, so wait 1283 usleep(WAIT_PERIOD_MS*1000); 1284 break; 1285 } 1286 1287 if (writtenSize > reqSize) { 1288 writtenSize = reqSize; 1289 } 1290 1291 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1292 // 8 to 16 bit conversion, note that source and destination are the same address 1293 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1294 writtenSize <<= 1; 1295 } 1296 1297 audioBuffer.size = writtenSize; 1298 // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for 1299 // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of 1300 // 16 bit. 1301 audioBuffer.frameCount = writtenSize / mFrameSizeAF; 1302 1303 frames -= audioBuffer.frameCount; 1304 1305 releaseBuffer(&audioBuffer); 1306 } 1307 while (frames); 1308 1309 if (frames == 0) { 1310 mRemainingFrames = mNotificationFramesAct; 1311 } else { 1312 mRemainingFrames = frames; 1313 } 1314 return true; 1315} 1316 1317// must be called with mLock and refCblk.lock held. Callers must also hold strong references on 1318// the IAudioTrack and IMemory in case they are recreated here. 1319// If the IAudioTrack is successfully restored, the refCblk pointer is updated 1320// FIXME Don't depend on caller to hold strong references. 1321status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart) 1322{ 1323 status_t result; 1324 1325 audio_track_cblk_t* cblk = refCblk; 1326 audio_track_cblk_t* newCblk = cblk; 1327 ALOGW("dead IAudioTrack, creating a new one from %s", 1328 fromStart ? "start()" : "obtainBuffer()"); 1329 1330 // signal old cblk condition so that other threads waiting for available buffers stop 1331 // waiting now 1332 cblk->cv.broadcast(); 1333 cblk->lock.unlock(); 1334 1335 // refresh the audio configuration cache in this process to make sure we get new 1336 // output parameters in getOutput_l() and createTrack_l() 1337 AudioSystem::clearAudioConfigCache(); 1338 1339 // if the new IAudioTrack is created, createTrack_l() will modify the 1340 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1341 // It will also delete the strong references on previous IAudioTrack and IMemory 1342 result = createTrack_l(mStreamType, 1343 cblk->sampleRate, 1344 mFormat, 1345 mReqFrameCount, // so that frame count never goes down 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 newCblk = mCblk; 1355 newCblk->user = user; 1356 newCblk->server = user; 1357 newCblk->userBase = user; 1358 newCblk->serverBase = user; 1359 // restore loop: this is not guaranteed to succeed if new frame count is not 1360 // compatible with loop length 1361 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1362 if (!fromStart) { 1363 newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1364 // Make sure that a client relying on callback events indicating underrun or 1365 // the actual amount of audio frames played (e.g SoundPool) receives them. 1366 if (mSharedBuffer == 0) { 1367 uint32_t frames = 0; 1368 if (user > server) { 1369 frames = ((user - server) > mFrameCount) ? 1370 mFrameCount : (user - server); 1371 memset(mBuffers, 0, frames * mFrameSizeAF); 1372 } 1373 // restart playback even if buffer is not completely filled. 1374 android_atomic_or(CBLK_FORCEREADY, &newCblk->flags); 1375 // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to 1376 // the client 1377 newCblk->stepUserOut(frames, mFrameCount); 1378 } 1379 } 1380 if (mSharedBuffer != 0) { 1381 newCblk->stepUserOut(mFrameCount, mFrameCount); 1382 } 1383 if (mActive) { 1384 result = mAudioTrack->start(); 1385 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1386 } 1387 if (fromStart && result == NO_ERROR) { 1388 mNewPosition = newCblk->server + mUpdatePeriod; 1389 } 1390 } 1391 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1392 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1393 result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); 1394 1395 if (result == NO_ERROR) { 1396 // from now on we switch to the newly created cblk 1397 refCblk = newCblk; 1398 } 1399 newCblk->lock.lock(); 1400 1401 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result); 1402 1403 return result; 1404} 1405 1406status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1407{ 1408 1409 const size_t SIZE = 256; 1410 char buffer[SIZE]; 1411 String8 result; 1412 1413 audio_track_cblk_t* cblk = mCblk; 1414 result.append(" AudioTrack::dump\n"); 1415 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1416 mVolume[0], mVolume[1]); 1417 result.append(buffer); 1418 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1419 mChannelCount, mFrameCount); 1420 result.append(buffer); 1421 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", 1422 (cblk == 0) ? 0 : cblk->sampleRate, mStatus); 1423 result.append(buffer); 1424 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1425 result.append(buffer); 1426 ::write(fd, result.string(), result.size()); 1427 return NO_ERROR; 1428} 1429 1430// ========================================================================= 1431 1432AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1433 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1434{ 1435} 1436 1437AudioTrack::AudioTrackThread::~AudioTrackThread() 1438{ 1439} 1440 1441bool AudioTrack::AudioTrackThread::threadLoop() 1442{ 1443 { 1444 AutoMutex _l(mMyLock); 1445 if (mPaused) { 1446 mMyCond.wait(mMyLock); 1447 // caller will check for exitPending() 1448 return true; 1449 } 1450 } 1451 if (!mReceiver.processAudioBuffer(this)) { 1452 pause(); 1453 } 1454 return true; 1455} 1456 1457void AudioTrack::AudioTrackThread::requestExit() 1458{ 1459 // must be in this order to avoid a race condition 1460 Thread::requestExit(); 1461 resume(); 1462} 1463 1464void AudioTrack::AudioTrackThread::pause() 1465{ 1466 AutoMutex _l(mMyLock); 1467 mPaused = true; 1468} 1469 1470void AudioTrack::AudioTrackThread::resume() 1471{ 1472 AutoMutex _l(mMyLock); 1473 if (mPaused) { 1474 mPaused = false; 1475 mMyCond.signal(); 1476 } 1477} 1478 1479// ========================================================================= 1480 1481 1482audio_track_cblk_t::audio_track_cblk_t() 1483 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1484 userBase(0), serverBase(0), frameCount_(0), 1485 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), 1486 mSendLevel(0), flags(0) 1487{ 1488} 1489 1490uint32_t audio_track_cblk_t::stepUser(size_t stepCount, size_t frameCount, bool isOut) 1491{ 1492 ALOGV("stepuser %08x %08x %d", user, server, stepCount); 1493 1494 uint32_t u = user; 1495 u += stepCount; 1496 // Ensure that user is never ahead of server for AudioRecord 1497 if (isOut) { 1498 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1499 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1500 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1501 } 1502 } else if (u > server) { 1503 ALOGW("stepUser occurred after track reset"); 1504 u = server; 1505 } 1506 1507 if (u >= frameCount) { 1508 // common case, user didn't just wrap 1509 if (u - frameCount >= userBase ) { 1510 userBase += frameCount; 1511 } 1512 } else if (u >= userBase + frameCount) { 1513 // user just wrapped 1514 userBase += frameCount; 1515 } 1516 1517 user = u; 1518 1519 // Clear flow control error condition as new data has been written/read to/from buffer. 1520 if (flags & CBLK_UNDERRUN) { 1521 android_atomic_and(~CBLK_UNDERRUN, &flags); 1522 } 1523 1524 return u; 1525} 1526 1527bool audio_track_cblk_t::stepServer(size_t stepCount, size_t frameCount, bool isOut) 1528{ 1529 ALOGV("stepserver %08x %08x %d", user, server, stepCount); 1530 1531 if (!tryLock()) { 1532 ALOGW("stepServer() could not lock cblk"); 1533 return false; 1534 } 1535 1536 uint32_t s = server; 1537 bool flushed = (s == user); 1538 1539 s += stepCount; 1540 if (isOut) { 1541 // Mark that we have read the first buffer so that next time stepUser() is called 1542 // we switch to normal obtainBuffer() timeout period 1543 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1544 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1545 } 1546 // It is possible that we receive a flush() 1547 // while the mixer is processing a block: in this case, 1548 // stepServer() is called After the flush() has reset u & s and 1549 // we have s > u 1550 if (flushed) { 1551 ALOGW("stepServer occurred after track reset"); 1552 s = user; 1553 } 1554 } 1555 1556 if (s >= loopEnd) { 1557 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1558 s = loopStart; 1559 if (--loopCount == 0) { 1560 loopEnd = UINT_MAX; 1561 loopStart = UINT_MAX; 1562 } 1563 } 1564 1565 if (s >= frameCount) { 1566 // common case, server didn't just wrap 1567 if (s - frameCount >= serverBase ) { 1568 serverBase += frameCount; 1569 } 1570 } else if (s >= serverBase + frameCount) { 1571 // server just wrapped 1572 serverBase += frameCount; 1573 } 1574 1575 server = s; 1576 1577 if (!(flags & CBLK_INVALID)) { 1578 cv.signal(); 1579 } 1580 lock.unlock(); 1581 return true; 1582} 1583 1584void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const 1585{ 1586 return (int8_t *)buffers + (offset - userBase) * frameSize; 1587} 1588 1589uint32_t audio_track_cblk_t::framesAvailable(size_t frameCount, bool isOut) 1590{ 1591 Mutex::Autolock _l(lock); 1592 return framesAvailable_l(frameCount, isOut); 1593} 1594 1595uint32_t audio_track_cblk_t::framesAvailable_l(size_t frameCount, bool isOut) 1596{ 1597 uint32_t u = user; 1598 uint32_t s = server; 1599 1600 if (isOut) { 1601 uint32_t limit = (s < loopStart) ? s : loopStart; 1602 return limit + frameCount - u; 1603 } else { 1604 return frameCount + u - s; 1605 } 1606} 1607 1608uint32_t audio_track_cblk_t::framesReady(bool isOut) 1609{ 1610 uint32_t u = user; 1611 uint32_t s = server; 1612 1613 if (isOut) { 1614 if (u < loopEnd) { 1615 return u - s; 1616 } else { 1617 // do not block on mutex shared with client on AudioFlinger side 1618 if (!tryLock()) { 1619 ALOGW("framesReady() could not lock cblk"); 1620 return 0; 1621 } 1622 uint32_t frames = UINT_MAX; 1623 if (loopCount >= 0) { 1624 frames = (loopEnd - loopStart)*loopCount + u - s; 1625 } 1626 lock.unlock(); 1627 return frames; 1628 } 1629 } else { 1630 return s - u; 1631 } 1632} 1633 1634bool audio_track_cblk_t::tryLock() 1635{ 1636 // the code below simulates lock-with-timeout 1637 // we MUST do this to protect the AudioFlinger server 1638 // as this lock is shared with the client. 1639 status_t err; 1640 1641 err = lock.tryLock(); 1642 if (err == -EBUSY) { // just wait a bit 1643 usleep(1000); 1644 err = lock.tryLock(); 1645 } 1646 if (err != NO_ERROR) { 1647 // probably, the client just died. 1648 return false; 1649 } 1650 return true; 1651} 1652 1653// ------------------------------------------------------------------------- 1654 1655}; // namespace android 1656