AudioTrack.cpp revision 28b76b334f92a15a2be3cc9e2f7d229a3275d1ac
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 if (frameCount == NULL) return BAD_VALUE; 58 59 // default to 0 in case of error 60 *frameCount = 0; 61 62 // FIXME merge with similar code in createTrack_l(), except we're missing 63 // some information here that is available in createTrack_l(): 64 // audio_io_handle_t output 65 // audio_format_t format 66 // audio_channel_mask_t channelMask 67 // audio_output_flags_t flags 68 int afSampleRate; 69 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 70 return NO_INIT; 71 } 72 int afFrameCount; 73 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 74 return NO_INIT; 75 } 76 uint32_t afLatency; 77 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 78 return NO_INIT; 79 } 80 81 // Ensure that buffer depth covers at least audio hardware latency 82 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 83 if (minBufCount < 2) minBufCount = 2; 84 85 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 86 afFrameCount * minBufCount * sampleRate / afSampleRate; 87 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 88 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 89 return NO_ERROR; 90} 91 92// --------------------------------------------------------------------------- 93 94AudioTrack::AudioTrack() 95 : mStatus(NO_INIT), 96 mIsTimed(false), 97 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 98 mPreviousSchedulingGroup(SP_DEFAULT) 99{ 100} 101 102AudioTrack::AudioTrack( 103 audio_stream_type_t streamType, 104 uint32_t sampleRate, 105 audio_format_t format, 106 audio_channel_mask_t channelMask, 107 int frameCount, 108 audio_output_flags_t flags, 109 callback_t cbf, 110 void* user, 111 int notificationFrames, 112 int sessionId) 113 : mStatus(NO_INIT), 114 mIsTimed(false), 115 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 116 mPreviousSchedulingGroup(SP_DEFAULT) 117{ 118 mStatus = set(streamType, sampleRate, format, channelMask, 119 frameCount, flags, cbf, user, notificationFrames, 120 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); 121} 122 123// DEPRECATED 124AudioTrack::AudioTrack( 125 int streamType, 126 uint32_t sampleRate, 127 int format, 128 int channelMask, 129 int frameCount, 130 uint32_t flags, 131 callback_t cbf, 132 void* user, 133 int notificationFrames, 134 int sessionId) 135 : mStatus(NO_INIT), 136 mIsTimed(false), 137 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) 138{ 139 mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, 140 (audio_channel_mask_t) channelMask, 141 frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames, 142 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); 143} 144 145AudioTrack::AudioTrack( 146 audio_stream_type_t streamType, 147 uint32_t sampleRate, 148 audio_format_t format, 149 audio_channel_mask_t channelMask, 150 const sp<IMemory>& sharedBuffer, 151 audio_output_flags_t flags, 152 callback_t cbf, 153 void* user, 154 int notificationFrames, 155 int sessionId) 156 : mStatus(NO_INIT), 157 mIsTimed(false), 158 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 159 mPreviousSchedulingGroup(SP_DEFAULT) 160{ 161 mStatus = set(streamType, sampleRate, format, channelMask, 162 0 /*frameCount*/, flags, cbf, user, notificationFrames, 163 sharedBuffer, false /*threadCanCallJava*/, sessionId); 164} 165 166AudioTrack::~AudioTrack() 167{ 168 ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 169 170 if (mStatus == NO_ERROR) { 171 // Make sure that callback function exits in the case where 172 // it is looping on buffer full condition in obtainBuffer(). 173 // Otherwise the callback thread will never exit. 174 stop(); 175 if (mAudioTrackThread != 0) { 176 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 177 mAudioTrackThread->requestExitAndWait(); 178 mAudioTrackThread.clear(); 179 } 180 mAudioTrack.clear(); 181 IPCThreadState::self()->flushCommands(); 182 AudioSystem::releaseAudioSessionId(mSessionId); 183 } 184} 185 186status_t AudioTrack::set( 187 audio_stream_type_t streamType, 188 uint32_t sampleRate, 189 audio_format_t format, 190 audio_channel_mask_t channelMask, 191 int frameCount, 192 audio_output_flags_t flags, 193 callback_t cbf, 194 void* user, 195 int notificationFrames, 196 const sp<IMemory>& sharedBuffer, 197 bool threadCanCallJava, 198 int sessionId) 199{ 200 201 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 202 203 ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags); 204 205 AutoMutex lock(mLock); 206 if (mAudioTrack != 0) { 207 ALOGE("Track already in use"); 208 return INVALID_OPERATION; 209 } 210 211 // handle default values first. 212 if (streamType == AUDIO_STREAM_DEFAULT) { 213 streamType = AUDIO_STREAM_MUSIC; 214 } 215 216 if (sampleRate == 0) { 217 int afSampleRate; 218 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 219 return NO_INIT; 220 } 221 sampleRate = afSampleRate; 222 } 223 224 // these below should probably come from the audioFlinger too... 225 if (format == AUDIO_FORMAT_DEFAULT) { 226 format = AUDIO_FORMAT_PCM_16_BIT; 227 } 228 if (channelMask == 0) { 229 channelMask = AUDIO_CHANNEL_OUT_STEREO; 230 } 231 232 // validate parameters 233 if (!audio_is_valid_format(format)) { 234 ALOGE("Invalid format"); 235 return BAD_VALUE; 236 } 237 238 // AudioFlinger does not currently support 8-bit data in shared memory 239 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 240 ALOGE("8-bit data in shared memory is not supported"); 241 return BAD_VALUE; 242 } 243 244 // force direct flag if format is not linear PCM 245 if (!audio_is_linear_pcm(format)) { 246 flags = (audio_output_flags_t) 247 // FIXME why can't we allow direct AND fast? 248 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 249 } 250 // only allow deep buffering for music stream type 251 if (streamType != AUDIO_STREAM_MUSIC) { 252 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 253 } 254 255 if (!audio_is_output_channel(channelMask)) { 256 ALOGE("Invalid channel mask %#x", channelMask); 257 return BAD_VALUE; 258 } 259 uint32_t channelCount = popcount(channelMask); 260 261 audio_io_handle_t output = AudioSystem::getOutput( 262 streamType, 263 sampleRate, format, channelMask, 264 flags); 265 266 if (output == 0) { 267 ALOGE("Could not get audio output for stream type %d", streamType); 268 return BAD_VALUE; 269 } 270 271 mVolume[LEFT] = 1.0f; 272 mVolume[RIGHT] = 1.0f; 273 mSendLevel = 0.0f; 274 mFrameCount = frameCount; 275 mNotificationFramesReq = notificationFrames; 276 mSessionId = sessionId; 277 mAuxEffectId = 0; 278 mFlags = flags; 279 mCbf = cbf; 280 281 if (cbf != NULL) { 282 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 283 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 284 } 285 286 // create the IAudioTrack 287 status_t status = createTrack_l(streamType, 288 sampleRate, 289 format, 290 channelMask, 291 frameCount, 292 flags, 293 sharedBuffer, 294 output); 295 296 if (status != NO_ERROR) { 297 if (mAudioTrackThread != 0) { 298 mAudioTrackThread->requestExit(); 299 mAudioTrackThread.clear(); 300 } 301 return status; 302 } 303 304 mStatus = NO_ERROR; 305 306 mStreamType = streamType; 307 mFormat = format; 308 mChannelMask = channelMask; 309 mChannelCount = channelCount; 310 mSharedBuffer = sharedBuffer; 311 mMuted = false; 312 mActive = false; 313 mUserData = user; 314 mLoopCount = 0; 315 mMarkerPosition = 0; 316 mMarkerReached = false; 317 mNewPosition = 0; 318 mUpdatePeriod = 0; 319 mFlushed = false; 320 AudioSystem::acquireAudioSessionId(mSessionId); 321 mRestoreStatus = NO_ERROR; 322 return NO_ERROR; 323} 324 325status_t AudioTrack::initCheck() const 326{ 327 return mStatus; 328} 329 330// ------------------------------------------------------------------------- 331 332uint32_t AudioTrack::latency() const 333{ 334 return mLatency; 335} 336 337audio_stream_type_t AudioTrack::streamType() const 338{ 339 return mStreamType; 340} 341 342audio_format_t AudioTrack::format() const 343{ 344 return mFormat; 345} 346 347int AudioTrack::channelCount() const 348{ 349 return mChannelCount; 350} 351 352uint32_t AudioTrack::frameCount() const 353{ 354 return mCblk->frameCount; 355} 356 357size_t AudioTrack::frameSize() const 358{ 359 if (audio_is_linear_pcm(mFormat)) { 360 return channelCount()*audio_bytes_per_sample(mFormat); 361 } else { 362 return sizeof(uint8_t); 363 } 364} 365 366sp<IMemory>& AudioTrack::sharedBuffer() 367{ 368 return mSharedBuffer; 369} 370 371// ------------------------------------------------------------------------- 372 373void AudioTrack::start() 374{ 375 sp<AudioTrackThread> t = mAudioTrackThread; 376 377 ALOGV("start %p", this); 378 379 AutoMutex lock(mLock); 380 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 381 // while we are accessing the cblk 382 sp<IAudioTrack> audioTrack = mAudioTrack; 383 sp<IMemory> iMem = mCblkMemory; 384 audio_track_cblk_t* cblk = mCblk; 385 386 if (!mActive) { 387 mFlushed = false; 388 mActive = true; 389 mNewPosition = cblk->server + mUpdatePeriod; 390 cblk->lock.lock(); 391 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 392 cblk->waitTimeMs = 0; 393 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 394 if (t != 0) { 395 t->resume(); 396 } else { 397 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 398 get_sched_policy(0, &mPreviousSchedulingGroup); 399 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 400 } 401 402 ALOGV("start %p before lock cblk %p", this, mCblk); 403 status_t status = NO_ERROR; 404 if (!(cblk->flags & CBLK_INVALID_MSK)) { 405 cblk->lock.unlock(); 406 ALOGV("mAudioTrack->start()"); 407 status = mAudioTrack->start(); 408 cblk->lock.lock(); 409 if (status == DEAD_OBJECT) { 410 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 411 } 412 } 413 if (cblk->flags & CBLK_INVALID_MSK) { 414 status = restoreTrack_l(cblk, true); 415 } 416 cblk->lock.unlock(); 417 if (status != NO_ERROR) { 418 ALOGV("start() failed"); 419 mActive = false; 420 if (t != 0) { 421 t->pause(); 422 } else { 423 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 424 set_sched_policy(0, mPreviousSchedulingGroup); 425 } 426 } 427 } 428 429} 430 431void AudioTrack::stop() 432{ 433 sp<AudioTrackThread> t = mAudioTrackThread; 434 435 ALOGV("stop %p", this); 436 437 AutoMutex lock(mLock); 438 if (mActive) { 439 mActive = false; 440 mCblk->cv.signal(); 441 mAudioTrack->stop(); 442 // Cancel loops (If we are in the middle of a loop, playback 443 // would not stop until loopCount reaches 0). 444 setLoop_l(0, 0, 0); 445 // the playback head position will reset to 0, so if a marker is set, we need 446 // to activate it again 447 mMarkerReached = false; 448 // Force flush if a shared buffer is used otherwise audioflinger 449 // will not stop before end of buffer is reached. 450 if (mSharedBuffer != 0) { 451 flush_l(); 452 } 453 if (t != 0) { 454 t->pause(); 455 } else { 456 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 457 set_sched_policy(0, mPreviousSchedulingGroup); 458 } 459 } 460 461} 462 463bool AudioTrack::stopped() const 464{ 465 AutoMutex lock(mLock); 466 return stopped_l(); 467} 468 469void AudioTrack::flush() 470{ 471 AutoMutex lock(mLock); 472 flush_l(); 473} 474 475// must be called with mLock held 476void AudioTrack::flush_l() 477{ 478 ALOGV("flush"); 479 480 // clear playback marker and periodic update counter 481 mMarkerPosition = 0; 482 mMarkerReached = false; 483 mUpdatePeriod = 0; 484 485 if (!mActive) { 486 mFlushed = true; 487 mAudioTrack->flush(); 488 // Release AudioTrack callback thread in case it was waiting for new buffers 489 // in AudioTrack::obtainBuffer() 490 mCblk->cv.signal(); 491 } 492} 493 494void AudioTrack::pause() 495{ 496 ALOGV("pause"); 497 AutoMutex lock(mLock); 498 if (mActive) { 499 mActive = false; 500 mCblk->cv.signal(); 501 mAudioTrack->pause(); 502 } 503} 504 505void AudioTrack::mute(bool e) 506{ 507 mAudioTrack->mute(e); 508 mMuted = e; 509} 510 511bool AudioTrack::muted() const 512{ 513 return mMuted; 514} 515 516status_t AudioTrack::setVolume(float left, float right) 517{ 518 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 519 return BAD_VALUE; 520 } 521 522 AutoMutex lock(mLock); 523 mVolume[LEFT] = left; 524 mVolume[RIGHT] = right; 525 526 mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 527 528 return NO_ERROR; 529} 530 531void AudioTrack::getVolume(float* left, float* right) const 532{ 533 if (left != NULL) { 534 *left = mVolume[LEFT]; 535 } 536 if (right != NULL) { 537 *right = mVolume[RIGHT]; 538 } 539} 540 541status_t AudioTrack::setAuxEffectSendLevel(float level) 542{ 543 ALOGV("setAuxEffectSendLevel(%f)", level); 544 if (level < 0.0f || level > 1.0f) { 545 return BAD_VALUE; 546 } 547 AutoMutex lock(mLock); 548 549 mSendLevel = level; 550 551 mCblk->setSendLevel(level); 552 553 return NO_ERROR; 554} 555 556void AudioTrack::getAuxEffectSendLevel(float* level) const 557{ 558 if (level != NULL) { 559 *level = mSendLevel; 560 } 561} 562 563status_t AudioTrack::setSampleRate(int rate) 564{ 565 int afSamplingRate; 566 567 if (mIsTimed) { 568 return INVALID_OPERATION; 569 } 570 571 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 572 return NO_INIT; 573 } 574 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 575 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 576 577 AutoMutex lock(mLock); 578 mCblk->sampleRate = rate; 579 return NO_ERROR; 580} 581 582uint32_t AudioTrack::getSampleRate() const 583{ 584 if (mIsTimed) { 585 return INVALID_OPERATION; 586 } 587 588 AutoMutex lock(mLock); 589 return mCblk->sampleRate; 590} 591 592status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 593{ 594 AutoMutex lock(mLock); 595 return setLoop_l(loopStart, loopEnd, loopCount); 596} 597 598// must be called with mLock held 599status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 600{ 601 audio_track_cblk_t* cblk = mCblk; 602 603 Mutex::Autolock _l(cblk->lock); 604 605 if (loopCount == 0) { 606 cblk->loopStart = UINT_MAX; 607 cblk->loopEnd = UINT_MAX; 608 cblk->loopCount = 0; 609 mLoopCount = 0; 610 return NO_ERROR; 611 } 612 613 if (mIsTimed) { 614 return INVALID_OPERATION; 615 } 616 617 if (loopStart >= loopEnd || 618 loopEnd - loopStart > cblk->frameCount || 619 cblk->server > loopStart) { 620 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 621 return BAD_VALUE; 622 } 623 624 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 625 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", 626 loopStart, loopEnd, cblk->frameCount); 627 return BAD_VALUE; 628 } 629 630 cblk->loopStart = loopStart; 631 cblk->loopEnd = loopEnd; 632 cblk->loopCount = loopCount; 633 mLoopCount = loopCount; 634 635 return NO_ERROR; 636} 637 638status_t AudioTrack::setMarkerPosition(uint32_t marker) 639{ 640 if (mCbf == NULL) return INVALID_OPERATION; 641 642 mMarkerPosition = marker; 643 mMarkerReached = false; 644 645 return NO_ERROR; 646} 647 648status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 649{ 650 if (marker == NULL) return BAD_VALUE; 651 652 *marker = mMarkerPosition; 653 654 return NO_ERROR; 655} 656 657status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 658{ 659 if (mCbf == NULL) return INVALID_OPERATION; 660 661 uint32_t curPosition; 662 getPosition(&curPosition); 663 mNewPosition = curPosition + updatePeriod; 664 mUpdatePeriod = updatePeriod; 665 666 return NO_ERROR; 667} 668 669status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 670{ 671 if (updatePeriod == NULL) return BAD_VALUE; 672 673 *updatePeriod = mUpdatePeriod; 674 675 return NO_ERROR; 676} 677 678status_t AudioTrack::setPosition(uint32_t position) 679{ 680 if (mIsTimed) return INVALID_OPERATION; 681 682 AutoMutex lock(mLock); 683 684 if (!stopped_l()) return INVALID_OPERATION; 685 686 Mutex::Autolock _l(mCblk->lock); 687 688 if (position > mCblk->user) return BAD_VALUE; 689 690 mCblk->server = position; 691 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 692 693 return NO_ERROR; 694} 695 696status_t AudioTrack::getPosition(uint32_t *position) 697{ 698 if (position == NULL) return BAD_VALUE; 699 AutoMutex lock(mLock); 700 *position = mFlushed ? 0 : mCblk->server; 701 702 return NO_ERROR; 703} 704 705status_t AudioTrack::reload() 706{ 707 AutoMutex lock(mLock); 708 709 if (!stopped_l()) return INVALID_OPERATION; 710 711 flush_l(); 712 713 mCblk->stepUser(mCblk->frameCount); 714 715 return NO_ERROR; 716} 717 718audio_io_handle_t AudioTrack::getOutput() 719{ 720 AutoMutex lock(mLock); 721 return getOutput_l(); 722} 723 724// must be called with mLock held 725audio_io_handle_t AudioTrack::getOutput_l() 726{ 727 return AudioSystem::getOutput(mStreamType, 728 mCblk->sampleRate, mFormat, mChannelMask, mFlags); 729} 730 731int AudioTrack::getSessionId() const 732{ 733 return mSessionId; 734} 735 736status_t AudioTrack::attachAuxEffect(int effectId) 737{ 738 ALOGV("attachAuxEffect(%d)", effectId); 739 status_t status = mAudioTrack->attachAuxEffect(effectId); 740 if (status == NO_ERROR) { 741 mAuxEffectId = effectId; 742 } 743 return status; 744} 745 746// ------------------------------------------------------------------------- 747 748// must be called with mLock held 749status_t AudioTrack::createTrack_l( 750 audio_stream_type_t streamType, 751 uint32_t sampleRate, 752 audio_format_t format, 753 audio_channel_mask_t channelMask, 754 int frameCount, 755 audio_output_flags_t flags, 756 const sp<IMemory>& sharedBuffer, 757 audio_io_handle_t output) 758{ 759 status_t status; 760 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 761 if (audioFlinger == 0) { 762 ALOGE("Could not get audioflinger"); 763 return NO_INIT; 764 } 765 766 uint32_t afLatency; 767 if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { 768 return NO_INIT; 769 } 770 771 // Client decides whether the track is TIMED (see below), but can only express a preference 772 // for FAST. Server will perform additional tests. 773 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 774 // either of these use cases: 775 // use case 1: shared buffer 776 (sharedBuffer != 0) || 777 // use case 2: callback handler 778 (mCbf != NULL))) { 779 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 780 // once denied, do not request again if IAudioTrack is re-created 781 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 782 mFlags = flags; 783 } 784 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 785 786 mNotificationFramesAct = mNotificationFramesReq; 787 788 if (!audio_is_linear_pcm(format)) { 789 790 if (sharedBuffer != 0) { 791 // Same comment as below about ignoring frameCount parameter for set() 792 frameCount = sharedBuffer->size(); 793 } else if (frameCount == 0) { 794 int afFrameCount; 795 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 796 return NO_INIT; 797 } 798 frameCount = afFrameCount; 799 } 800 801 } else if (sharedBuffer != 0) { 802 803 // Ensure that buffer alignment matches channelCount 804 int channelCount = popcount(channelMask); 805 // 8-bit data in shared memory is not currently supported by AudioFlinger 806 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 807 if (channelCount > 1) { 808 // More than 2 channels does not require stronger alignment than stereo 809 alignment <<= 1; 810 } 811 if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 812 ALOGE("Invalid buffer alignment: address %p, channelCount %d", 813 sharedBuffer->pointer(), channelCount); 814 return BAD_VALUE; 815 } 816 817 // When initializing a shared buffer AudioTrack via constructors, 818 // there's no frameCount parameter. 819 // But when initializing a shared buffer AudioTrack via set(), 820 // there _is_ a frameCount parameter. We silently ignore it. 821 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 822 823 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 824 825 // FIXME move these calculations and associated checks to server 826 int afSampleRate; 827 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 828 return NO_INIT; 829 } 830 int afFrameCount; 831 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 832 return NO_INIT; 833 } 834 835 // Ensure that buffer depth covers at least audio hardware latency 836 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 837 if (minBufCount < 2) minBufCount = 2; 838 839 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 840 ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d" 841 ", afLatency=%d", 842 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 843 844 if (frameCount == 0) { 845 frameCount = minFrameCount; 846 } 847 if (mNotificationFramesAct == 0) { 848 mNotificationFramesAct = frameCount/2; 849 } 850 // Make sure that application is notified with sufficient margin 851 // before underrun 852 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 853 mNotificationFramesAct = frameCount/2; 854 } 855 if (frameCount < minFrameCount) { 856 // not ALOGW because it happens all the time when playing key clicks over A2DP 857 ALOGV("Minimum buffer size corrected from %d to %d", 858 frameCount, minFrameCount); 859 frameCount = minFrameCount; 860 } 861 862 } else { 863 // For fast tracks, the frame count calculations and checks are done by server 864 } 865 866 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 867 if (mIsTimed) { 868 trackFlags |= IAudioFlinger::TRACK_TIMED; 869 } 870 871 pid_t tid = -1; 872 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 873 trackFlags |= IAudioFlinger::TRACK_FAST; 874 if (mAudioTrackThread != 0) { 875 tid = mAudioTrackThread->getTid(); 876 } 877 } 878 879 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 880 streamType, 881 sampleRate, 882 format, 883 channelMask, 884 frameCount, 885 trackFlags, 886 sharedBuffer, 887 output, 888 tid, 889 &mSessionId, 890 &status); 891 892 if (track == 0) { 893 ALOGE("AudioFlinger could not create track, status: %d", status); 894 return status; 895 } 896 sp<IMemory> cblk = track->getCblk(); 897 if (cblk == 0) { 898 ALOGE("Could not get control block"); 899 return NO_INIT; 900 } 901 mAudioTrack = track; 902 mCblkMemory = cblk; 903 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 904 // old has the previous value of mCblk->flags before the "or" operation 905 int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 906 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 907 if (old & CBLK_FAST) { 908 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount); 909 } else { 910 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount); 911 // once denied, do not request again if IAudioTrack is re-created 912 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 913 mFlags = flags; 914 } 915 if (sharedBuffer == 0) { 916 mNotificationFramesAct = mCblk->frameCount/2; 917 } 918 } 919 if (sharedBuffer == 0) { 920 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 921 } else { 922 mCblk->buffers = sharedBuffer->pointer(); 923 // Force buffer full condition as data is already present in shared memory 924 mCblk->stepUser(mCblk->frameCount); 925 } 926 927 mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); 928 mCblk->setSendLevel(mSendLevel); 929 mAudioTrack->attachAuxEffect(mAuxEffectId); 930 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 931 mCblk->waitTimeMs = 0; 932 mRemainingFrames = mNotificationFramesAct; 933 // FIXME don't believe this lie 934 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 935 // If IAudioTrack is re-created, don't let the requested frameCount 936 // decrease. This can confuse clients that cache frameCount(). 937 if (mCblk->frameCount > mFrameCount) { 938 mFrameCount = mCblk->frameCount; 939 } 940 return NO_ERROR; 941} 942 943status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 944{ 945 AutoMutex lock(mLock); 946 bool active; 947 status_t result = NO_ERROR; 948 audio_track_cblk_t* cblk = mCblk; 949 uint32_t framesReq = audioBuffer->frameCount; 950 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 951 952 audioBuffer->frameCount = 0; 953 audioBuffer->size = 0; 954 955 uint32_t framesAvail = cblk->framesAvailable(); 956 957 cblk->lock.lock(); 958 if (cblk->flags & CBLK_INVALID_MSK) { 959 goto create_new_track; 960 } 961 cblk->lock.unlock(); 962 963 if (framesAvail == 0) { 964 cblk->lock.lock(); 965 goto start_loop_here; 966 while (framesAvail == 0) { 967 active = mActive; 968 if (CC_UNLIKELY(!active)) { 969 ALOGV("Not active and NO_MORE_BUFFERS"); 970 cblk->lock.unlock(); 971 return NO_MORE_BUFFERS; 972 } 973 if (CC_UNLIKELY(!waitCount)) { 974 cblk->lock.unlock(); 975 return WOULD_BLOCK; 976 } 977 if (!(cblk->flags & CBLK_INVALID_MSK)) { 978 mLock.unlock(); 979 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 980 cblk->lock.unlock(); 981 mLock.lock(); 982 if (!mActive) { 983 return status_t(STOPPED); 984 } 985 cblk->lock.lock(); 986 } 987 988 if (cblk->flags & CBLK_INVALID_MSK) { 989 goto create_new_track; 990 } 991 if (CC_UNLIKELY(result != NO_ERROR)) { 992 cblk->waitTimeMs += waitTimeMs; 993 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 994 // timing out when a loop has been set and we have already written upto loop end 995 // is a normal condition: no need to wake AudioFlinger up. 996 if (cblk->user < cblk->loopEnd) { 997 ALOGW( "obtainBuffer timed out (is the CPU pegged?) %p name=%#x" 998 "user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server); 999 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 1000 cblk->lock.unlock(); 1001 result = mAudioTrack->start(); 1002 cblk->lock.lock(); 1003 if (result == DEAD_OBJECT) { 1004 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 1005create_new_track: 1006 result = restoreTrack_l(cblk, false); 1007 } 1008 if (result != NO_ERROR) { 1009 ALOGW("obtainBuffer create Track error %d", result); 1010 cblk->lock.unlock(); 1011 return result; 1012 } 1013 } 1014 cblk->waitTimeMs = 0; 1015 } 1016 1017 if (--waitCount == 0) { 1018 cblk->lock.unlock(); 1019 return TIMED_OUT; 1020 } 1021 } 1022 // read the server count again 1023 start_loop_here: 1024 framesAvail = cblk->framesAvailable_l(); 1025 } 1026 cblk->lock.unlock(); 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 if (audioBuffer->frameCount > 0) { 1061 // restart track if it was disabled by audioflinger due to previous underrun 1062 if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1063 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1064 ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName); 1065 mAudioTrack->start(); 1066 } 1067 } 1068} 1069 1070// ------------------------------------------------------------------------- 1071 1072ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1073{ 1074 1075 if (mSharedBuffer != 0) return INVALID_OPERATION; 1076 if (mIsTimed) return INVALID_OPERATION; 1077 1078 if (ssize_t(userSize) < 0) { 1079 // Sanity-check: user is most-likely passing an error code, and it would 1080 // make the return value ambiguous (actualSize vs error). 1081 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1082 buffer, userSize, userSize); 1083 return BAD_VALUE; 1084 } 1085 1086 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1087 1088 if (userSize == 0) { 1089 return 0; 1090 } 1091 1092 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1093 // while we are accessing the cblk 1094 mLock.lock(); 1095 sp<IAudioTrack> audioTrack = mAudioTrack; 1096 sp<IMemory> iMem = mCblkMemory; 1097 mLock.unlock(); 1098 1099 ssize_t written = 0; 1100 const int8_t *src = (const int8_t *)buffer; 1101 Buffer audioBuffer; 1102 size_t frameSz = frameSize(); 1103 1104 do { 1105 audioBuffer.frameCount = userSize/frameSz; 1106 1107 status_t err = obtainBuffer(&audioBuffer, -1); 1108 if (err < 0) { 1109 // out of buffers, return #bytes written 1110 if (err == status_t(NO_MORE_BUFFERS)) 1111 break; 1112 return ssize_t(err); 1113 } 1114 1115 size_t toWrite; 1116 1117 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1118 // Divide capacity by 2 to take expansion into account 1119 toWrite = audioBuffer.size>>1; 1120 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1121 } else { 1122 toWrite = audioBuffer.size; 1123 memcpy(audioBuffer.i8, src, toWrite); 1124 src += toWrite; 1125 } 1126 userSize -= toWrite; 1127 written += toWrite; 1128 1129 releaseBuffer(&audioBuffer); 1130 } while (userSize >= frameSz); 1131 1132 return written; 1133} 1134 1135// ------------------------------------------------------------------------- 1136 1137TimedAudioTrack::TimedAudioTrack() { 1138 mIsTimed = true; 1139} 1140 1141status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1142{ 1143 status_t result = UNKNOWN_ERROR; 1144 1145 // If the track is not invalid already, try to allocate a buffer. alloc 1146 // fails indicating that the server is dead, flag the track as invalid so 1147 // we can attempt to restore in in just a bit. 1148 if (!(mCblk->flags & CBLK_INVALID_MSK)) { 1149 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1150 if (result == DEAD_OBJECT) { 1151 android_atomic_or(CBLK_INVALID_ON, &mCblk->flags); 1152 } 1153 } 1154 1155 // If the track is invalid at this point, attempt to restore it. and try the 1156 // allocation one more time. 1157 if (mCblk->flags & CBLK_INVALID_MSK) { 1158 mCblk->lock.lock(); 1159 result = restoreTrack_l(mCblk, false); 1160 mCblk->lock.unlock(); 1161 1162 if (result == OK) 1163 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1164 } 1165 1166 return result; 1167} 1168 1169status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1170 int64_t pts) 1171{ 1172 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1173 { 1174 AutoMutex lock(mLock); 1175 // restart track if it was disabled by audioflinger due to previous underrun 1176 if (buffer->size() != 0 && status == NO_ERROR && 1177 mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1178 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1179 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1180 mAudioTrack->start(); 1181 } 1182 } 1183 return status; 1184} 1185 1186status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1187 TargetTimeline target) 1188{ 1189 return mAudioTrack->setMediaTimeTransform(xform, target); 1190} 1191 1192// ------------------------------------------------------------------------- 1193 1194bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1195{ 1196 Buffer audioBuffer; 1197 uint32_t frames; 1198 size_t writtenSize; 1199 1200 mLock.lock(); 1201 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1202 // while we are accessing the cblk 1203 sp<IAudioTrack> audioTrack = mAudioTrack; 1204 sp<IMemory> iMem = mCblkMemory; 1205 audio_track_cblk_t* cblk = mCblk; 1206 bool active = mActive; 1207 mLock.unlock(); 1208 1209 // Manage underrun callback 1210 if (active && (cblk->framesAvailable() == cblk->frameCount)) { 1211 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1212 if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { 1213 mCbf(EVENT_UNDERRUN, mUserData, 0); 1214 if (cblk->server == cblk->frameCount) { 1215 mCbf(EVENT_BUFFER_END, mUserData, 0); 1216 } 1217 if (mSharedBuffer != 0) return false; 1218 } 1219 } 1220 1221 // Manage loop end callback 1222 while (mLoopCount > cblk->loopCount) { 1223 int loopCount = -1; 1224 mLoopCount--; 1225 if (mLoopCount >= 0) loopCount = mLoopCount; 1226 1227 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1228 } 1229 1230 // Manage marker callback 1231 if (!mMarkerReached && (mMarkerPosition > 0)) { 1232 if (cblk->server >= mMarkerPosition) { 1233 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1234 mMarkerReached = true; 1235 } 1236 } 1237 1238 // Manage new position callback 1239 if (mUpdatePeriod > 0) { 1240 while (cblk->server >= mNewPosition) { 1241 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1242 mNewPosition += mUpdatePeriod; 1243 } 1244 } 1245 1246 // If Shared buffer is used, no data is requested from client. 1247 if (mSharedBuffer != 0) { 1248 frames = 0; 1249 } else { 1250 frames = mRemainingFrames; 1251 } 1252 1253 // See description of waitCount parameter at declaration of obtainBuffer(). 1254 // The logic below prevents us from being stuck below at obtainBuffer() 1255 // not being able to handle timed events (position, markers, loops). 1256 int32_t waitCount = -1; 1257 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1258 waitCount = 1; 1259 } 1260 1261 do { 1262 1263 audioBuffer.frameCount = frames; 1264 1265 status_t err = obtainBuffer(&audioBuffer, waitCount); 1266 if (err < NO_ERROR) { 1267 if (err != TIMED_OUT) { 1268 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); 1269 return false; 1270 } 1271 break; 1272 } 1273 if (err == status_t(STOPPED)) return false; 1274 1275 // Divide buffer size by 2 to take into account the expansion 1276 // due to 8 to 16 bit conversion: the callback must fill only half 1277 // of the destination buffer 1278 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1279 audioBuffer.size >>= 1; 1280 } 1281 1282 size_t reqSize = audioBuffer.size; 1283 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1284 writtenSize = audioBuffer.size; 1285 1286 // Sanity check on returned size 1287 if (ssize_t(writtenSize) <= 0) { 1288 // The callback is done filling buffers 1289 // Keep this thread going to handle timed events and 1290 // still try to get more data in intervals of WAIT_PERIOD_MS 1291 // but don't just loop and block the CPU, so wait 1292 usleep(WAIT_PERIOD_MS*1000); 1293 break; 1294 } 1295 1296 if (writtenSize > reqSize) writtenSize = reqSize; 1297 1298 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1299 // 8 to 16 bit conversion, note that source and destination are the same address 1300 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1301 writtenSize <<= 1; 1302 } 1303 1304 audioBuffer.size = writtenSize; 1305 // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for 1306 // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of 1307 // 16 bit. 1308 audioBuffer.frameCount = writtenSize/mCblk->frameSize; 1309 1310 frames -= audioBuffer.frameCount; 1311 1312 releaseBuffer(&audioBuffer); 1313 } 1314 while (frames); 1315 1316 if (frames == 0) { 1317 mRemainingFrames = mNotificationFramesAct; 1318 } else { 1319 mRemainingFrames = frames; 1320 } 1321 return true; 1322} 1323 1324// must be called with mLock and cblk.lock held. Callers must also hold strong references on 1325// the IAudioTrack and IMemory in case they are recreated here. 1326// If the IAudioTrack is successfully restored, the cblk pointer is updated 1327status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) 1328{ 1329 status_t result; 1330 1331 if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { 1332 ALOGW("dead IAudioTrack, creating a new one from %s TID %d", 1333 fromStart ? "start()" : "obtainBuffer()", gettid()); 1334 1335 // signal old cblk condition so that other threads waiting for available buffers stop 1336 // waiting now 1337 cblk->cv.broadcast(); 1338 cblk->lock.unlock(); 1339 1340 // refresh the audio configuration cache in this process to make sure we get new 1341 // output parameters in getOutput_l() and createTrack_l() 1342 AudioSystem::clearAudioConfigCache(); 1343 1344 // if the new IAudioTrack is created, createTrack_l() will modify the 1345 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1346 // It will also delete the strong references on previous IAudioTrack and IMemory 1347 result = createTrack_l(mStreamType, 1348 cblk->sampleRate, 1349 mFormat, 1350 mChannelMask, 1351 mFrameCount, 1352 mFlags, 1353 mSharedBuffer, 1354 getOutput_l()); 1355 1356 if (result == NO_ERROR) { 1357 uint32_t user = cblk->user; 1358 uint32_t server = cblk->server; 1359 // restore write index and set other indexes to reflect empty buffer status 1360 mCblk->user = user; 1361 mCblk->server = user; 1362 mCblk->userBase = user; 1363 mCblk->serverBase = user; 1364 // restore loop: this is not guaranteed to succeed if new frame count is not 1365 // compatible with loop length 1366 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1367 if (!fromStart) { 1368 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1369 // Make sure that a client relying on callback events indicating underrun or 1370 // the actual amount of audio frames played (e.g SoundPool) receives them. 1371 if (mSharedBuffer == 0) { 1372 uint32_t frames = 0; 1373 if (user > server) { 1374 frames = ((user - server) > mCblk->frameCount) ? 1375 mCblk->frameCount : (user - server); 1376 memset(mCblk->buffers, 0, frames * mCblk->frameSize); 1377 } 1378 // restart playback even if buffer is not completely filled. 1379 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 1380 // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to 1381 // the client 1382 mCblk->stepUser(frames); 1383 } 1384 } 1385 if (mSharedBuffer != 0) { 1386 mCblk->stepUser(mCblk->frameCount); 1387 } 1388 if (mActive) { 1389 result = mAudioTrack->start(); 1390 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1391 } 1392 if (fromStart && result == NO_ERROR) { 1393 mNewPosition = mCblk->server + mUpdatePeriod; 1394 } 1395 } 1396 if (result != NO_ERROR) { 1397 android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); 1398 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1399 } 1400 mRestoreStatus = result; 1401 // signal old cblk condition for other threads waiting for restore completion 1402 android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); 1403 cblk->cv.broadcast(); 1404 } else { 1405 if (!(cblk->flags & CBLK_RESTORED_MSK)) { 1406 ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); 1407 mLock.unlock(); 1408 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1409 if (result == NO_ERROR) { 1410 result = mRestoreStatus; 1411 } 1412 cblk->lock.unlock(); 1413 mLock.lock(); 1414 } else { 1415 ALOGW("dead IAudioTrack, already restored TID %d", gettid()); 1416 result = mRestoreStatus; 1417 cblk->lock.unlock(); 1418 } 1419 } 1420 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1421 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1422 1423 if (result == NO_ERROR) { 1424 // from now on we switch to the newly created cblk 1425 cblk = mCblk; 1426 } 1427 cblk->lock.lock(); 1428 1429 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1430 1431 return result; 1432} 1433 1434status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1435{ 1436 1437 const size_t SIZE = 256; 1438 char buffer[SIZE]; 1439 String8 result; 1440 1441 result.append(" AudioTrack::dump\n"); 1442 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); 1443 result.append(buffer); 1444 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount); 1445 result.append(buffer); 1446 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1447 result.append(buffer); 1448 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1449 result.append(buffer); 1450 ::write(fd, result.string(), result.size()); 1451 return NO_ERROR; 1452} 1453 1454// ========================================================================= 1455 1456AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1457 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1458{ 1459} 1460 1461AudioTrack::AudioTrackThread::~AudioTrackThread() 1462{ 1463} 1464 1465bool AudioTrack::AudioTrackThread::threadLoop() 1466{ 1467 { 1468 AutoMutex _l(mMyLock); 1469 if (mPaused) { 1470 mMyCond.wait(mMyLock); 1471 // caller will check for exitPending() 1472 return true; 1473 } 1474 } 1475 if (!mReceiver.processAudioBuffer(this)) { 1476 pause(); 1477 } 1478 return true; 1479} 1480 1481void AudioTrack::AudioTrackThread::requestExit() 1482{ 1483 // must be in this order to avoid a race condition 1484 Thread::requestExit(); 1485 resume(); 1486} 1487 1488void AudioTrack::AudioTrackThread::pause() 1489{ 1490 AutoMutex _l(mMyLock); 1491 mPaused = true; 1492} 1493 1494void AudioTrack::AudioTrackThread::resume() 1495{ 1496 AutoMutex _l(mMyLock); 1497 if (mPaused) { 1498 mPaused = false; 1499 mMyCond.signal(); 1500 } 1501} 1502 1503// ========================================================================= 1504 1505 1506audio_track_cblk_t::audio_track_cblk_t() 1507 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1508 userBase(0), serverBase(0), buffers(NULL), frameCount(0), 1509 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), 1510 mSendLevel(0), flags(0) 1511{ 1512} 1513 1514uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1515{ 1516 ALOGV("stepuser %08x %08x %d", user, server, frameCount); 1517 1518 uint32_t u = user; 1519 u += frameCount; 1520 // Ensure that user is never ahead of server for AudioRecord 1521 if (flags & CBLK_DIRECTION_MSK) { 1522 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1523 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1524 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1525 } 1526 } else if (u > server) { 1527 ALOGW("stepUser occurred after track reset"); 1528 u = server; 1529 } 1530 1531 uint32_t fc = this->frameCount; 1532 if (u >= fc) { 1533 // common case, user didn't just wrap 1534 if (u - fc >= userBase ) { 1535 userBase += fc; 1536 } 1537 } else if (u >= userBase + fc) { 1538 // user just wrapped 1539 userBase += fc; 1540 } 1541 1542 user = u; 1543 1544 // Clear flow control error condition as new data has been written/read to/from buffer. 1545 if (flags & CBLK_UNDERRUN_MSK) { 1546 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1547 } 1548 1549 return u; 1550} 1551 1552bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1553{ 1554 ALOGV("stepserver %08x %08x %d", user, server, frameCount); 1555 1556 if (!tryLock()) { 1557 ALOGW("stepServer() could not lock cblk"); 1558 return false; 1559 } 1560 1561 uint32_t s = server; 1562 bool flushed = (s == user); 1563 1564 s += frameCount; 1565 if (flags & CBLK_DIRECTION_MSK) { 1566 // Mark that we have read the first buffer so that next time stepUser() is called 1567 // we switch to normal obtainBuffer() timeout period 1568 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1569 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1570 } 1571 // It is possible that we receive a flush() 1572 // while the mixer is processing a block: in this case, 1573 // stepServer() is called After the flush() has reset u & s and 1574 // we have s > u 1575 if (flushed) { 1576 ALOGW("stepServer occurred after track reset"); 1577 s = user; 1578 } 1579 } 1580 1581 if (s >= loopEnd) { 1582 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1583 s = loopStart; 1584 if (--loopCount == 0) { 1585 loopEnd = UINT_MAX; 1586 loopStart = UINT_MAX; 1587 } 1588 } 1589 1590 uint32_t fc = this->frameCount; 1591 if (s >= fc) { 1592 // common case, server didn't just wrap 1593 if (s - fc >= serverBase ) { 1594 serverBase += fc; 1595 } 1596 } else if (s >= serverBase + fc) { 1597 // server just wrapped 1598 serverBase += fc; 1599 } 1600 1601 server = s; 1602 1603 if (!(flags & CBLK_INVALID_MSK)) { 1604 cv.signal(); 1605 } 1606 lock.unlock(); 1607 return true; 1608} 1609 1610void* audio_track_cblk_t::buffer(uint32_t offset) const 1611{ 1612 return (int8_t *)buffers + (offset - userBase) * frameSize; 1613} 1614 1615uint32_t audio_track_cblk_t::framesAvailable() 1616{ 1617 Mutex::Autolock _l(lock); 1618 return framesAvailable_l(); 1619} 1620 1621uint32_t audio_track_cblk_t::framesAvailable_l() 1622{ 1623 uint32_t u = user; 1624 uint32_t s = server; 1625 1626 if (flags & CBLK_DIRECTION_MSK) { 1627 uint32_t limit = (s < loopStart) ? s : loopStart; 1628 return limit + frameCount - u; 1629 } else { 1630 return frameCount + u - s; 1631 } 1632} 1633 1634uint32_t audio_track_cblk_t::framesReady() 1635{ 1636 uint32_t u = user; 1637 uint32_t s = server; 1638 1639 if (flags & CBLK_DIRECTION_MSK) { 1640 if (u < loopEnd) { 1641 return u - s; 1642 } else { 1643 // do not block on mutex shared with client on AudioFlinger side 1644 if (!tryLock()) { 1645 ALOGW("framesReady() could not lock cblk"); 1646 return 0; 1647 } 1648 uint32_t frames = UINT_MAX; 1649 if (loopCount >= 0) { 1650 frames = (loopEnd - loopStart)*loopCount + u - s; 1651 } 1652 lock.unlock(); 1653 return frames; 1654 } 1655 } else { 1656 return s - u; 1657 } 1658} 1659 1660bool audio_track_cblk_t::tryLock() 1661{ 1662 // the code below simulates lock-with-timeout 1663 // we MUST do this to protect the AudioFlinger server 1664 // as this lock is shared with the client. 1665 status_t err; 1666 1667 err = lock.tryLock(); 1668 if (err == -EBUSY) { // just wait a bit 1669 usleep(1000); 1670 err = lock.tryLock(); 1671 } 1672 if (err != NO_ERROR) { 1673 // probably, the client just died. 1674 return false; 1675 } 1676 return true; 1677} 1678 1679// ------------------------------------------------------------------------- 1680 1681}; // namespace android 1682