AudioTrack.cpp revision 8af901cdea0af7e536579dee6d56e69987035a01
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(), 202 sharedBuffer->size()); 203 204 ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags); 205 206 AutoMutex lock(mLock); 207 if (mAudioTrack != 0) { 208 ALOGE("Track already in use"); 209 return INVALID_OPERATION; 210 } 211 212 // handle default values first. 213 if (streamType == AUDIO_STREAM_DEFAULT) { 214 streamType = AUDIO_STREAM_MUSIC; 215 } 216 217 if (sampleRate == 0) { 218 int afSampleRate; 219 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 220 return NO_INIT; 221 } 222 sampleRate = afSampleRate; 223 } 224 225 // these below should probably come from the audioFlinger too... 226 if (format == AUDIO_FORMAT_DEFAULT) { 227 format = AUDIO_FORMAT_PCM_16_BIT; 228 } 229 if (channelMask == 0) { 230 channelMask = AUDIO_CHANNEL_OUT_STEREO; 231 } 232 233 // validate parameters 234 if (!audio_is_valid_format(format)) { 235 ALOGE("Invalid format"); 236 return BAD_VALUE; 237 } 238 239 // AudioFlinger does not currently support 8-bit data in shared memory 240 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 241 ALOGE("8-bit data in shared memory is not supported"); 242 return BAD_VALUE; 243 } 244 245 // force direct flag if format is not linear PCM 246 if (!audio_is_linear_pcm(format)) { 247 flags = (audio_output_flags_t) 248 // FIXME why can't we allow direct AND fast? 249 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 250 } 251 // only allow deep buffering for music stream type 252 if (streamType != AUDIO_STREAM_MUSIC) { 253 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 254 } 255 256 if (!audio_is_output_channel(channelMask)) { 257 ALOGE("Invalid channel mask %#x", channelMask); 258 return BAD_VALUE; 259 } 260 uint32_t channelCount = popcount(channelMask); 261 262 audio_io_handle_t output = AudioSystem::getOutput( 263 streamType, 264 sampleRate, format, channelMask, 265 flags); 266 267 if (output == 0) { 268 ALOGE("Could not get audio output for stream type %d", streamType); 269 return BAD_VALUE; 270 } 271 272 mVolume[LEFT] = 1.0f; 273 mVolume[RIGHT] = 1.0f; 274 mSendLevel = 0.0f; 275 mFrameCount = frameCount; 276 mNotificationFramesReq = notificationFrames; 277 mSessionId = sessionId; 278 mAuxEffectId = 0; 279 mFlags = flags; 280 mCbf = cbf; 281 282 if (cbf != NULL) { 283 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 284 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 285 } 286 287 // create the IAudioTrack 288 status_t status = createTrack_l(streamType, 289 sampleRate, 290 format, 291 channelMask, 292 frameCount, 293 flags, 294 sharedBuffer, 295 output); 296 297 if (status != NO_ERROR) { 298 if (mAudioTrackThread != 0) { 299 mAudioTrackThread->requestExit(); 300 mAudioTrackThread.clear(); 301 } 302 return status; 303 } 304 305 mStatus = NO_ERROR; 306 307 mStreamType = streamType; 308 mFormat = format; 309 mChannelMask = channelMask; 310 mChannelCount = channelCount; 311 mSharedBuffer = sharedBuffer; 312 mMuted = false; 313 mActive = false; 314 mUserData = user; 315 mLoopCount = 0; 316 mMarkerPosition = 0; 317 mMarkerReached = false; 318 mNewPosition = 0; 319 mUpdatePeriod = 0; 320 mFlushed = false; 321 AudioSystem::acquireAudioSessionId(mSessionId); 322 mRestoreStatus = NO_ERROR; 323 return NO_ERROR; 324} 325 326status_t AudioTrack::initCheck() const 327{ 328 return mStatus; 329} 330 331// ------------------------------------------------------------------------- 332 333uint32_t AudioTrack::latency() const 334{ 335 return mLatency; 336} 337 338audio_stream_type_t AudioTrack::streamType() const 339{ 340 return mStreamType; 341} 342 343audio_format_t AudioTrack::format() const 344{ 345 return mFormat; 346} 347 348int AudioTrack::channelCount() const 349{ 350 return mChannelCount; 351} 352 353uint32_t AudioTrack::frameCount() const 354{ 355 return mCblk->frameCount; 356} 357 358size_t AudioTrack::frameSize() const 359{ 360 if (audio_is_linear_pcm(mFormat)) { 361 return channelCount()*audio_bytes_per_sample(mFormat); 362 } else { 363 return sizeof(uint8_t); 364 } 365} 366 367sp<IMemory>& AudioTrack::sharedBuffer() 368{ 369 return mSharedBuffer; 370} 371 372// ------------------------------------------------------------------------- 373 374void AudioTrack::start() 375{ 376 sp<AudioTrackThread> t = mAudioTrackThread; 377 378 ALOGV("start %p", this); 379 380 AutoMutex lock(mLock); 381 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 382 // while we are accessing the cblk 383 sp<IAudioTrack> audioTrack = mAudioTrack; 384 sp<IMemory> iMem = mCblkMemory; 385 audio_track_cblk_t* cblk = mCblk; 386 387 if (!mActive) { 388 mFlushed = false; 389 mActive = true; 390 mNewPosition = cblk->server + mUpdatePeriod; 391 cblk->lock.lock(); 392 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 393 cblk->waitTimeMs = 0; 394 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 395 if (t != 0) { 396 t->resume(); 397 } else { 398 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 399 get_sched_policy(0, &mPreviousSchedulingGroup); 400 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 401 } 402 403 ALOGV("start %p before lock cblk %p", this, mCblk); 404 status_t status = NO_ERROR; 405 if (!(cblk->flags & CBLK_INVALID_MSK)) { 406 cblk->lock.unlock(); 407 ALOGV("mAudioTrack->start()"); 408 status = mAudioTrack->start(); 409 cblk->lock.lock(); 410 if (status == DEAD_OBJECT) { 411 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 412 } 413 } 414 if (cblk->flags & CBLK_INVALID_MSK) { 415 status = restoreTrack_l(cblk, true); 416 } 417 cblk->lock.unlock(); 418 if (status != NO_ERROR) { 419 ALOGV("start() failed"); 420 mActive = false; 421 if (t != 0) { 422 t->pause(); 423 } else { 424 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 425 set_sched_policy(0, mPreviousSchedulingGroup); 426 } 427 } 428 } 429 430} 431 432void AudioTrack::stop() 433{ 434 sp<AudioTrackThread> t = mAudioTrackThread; 435 436 ALOGV("stop %p", this); 437 438 AutoMutex lock(mLock); 439 if (mActive) { 440 mActive = false; 441 mCblk->cv.signal(); 442 mAudioTrack->stop(); 443 // Cancel loops (If we are in the middle of a loop, playback 444 // would not stop until loopCount reaches 0). 445 setLoop_l(0, 0, 0); 446 // the playback head position will reset to 0, so if a marker is set, we need 447 // to activate it again 448 mMarkerReached = false; 449 // Force flush if a shared buffer is used otherwise audioflinger 450 // will not stop before end of buffer is reached. 451 if (mSharedBuffer != 0) { 452 flush_l(); 453 } 454 if (t != 0) { 455 t->pause(); 456 } else { 457 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 458 set_sched_policy(0, mPreviousSchedulingGroup); 459 } 460 } 461 462} 463 464bool AudioTrack::stopped() const 465{ 466 AutoMutex lock(mLock); 467 return stopped_l(); 468} 469 470void AudioTrack::flush() 471{ 472 AutoMutex lock(mLock); 473 flush_l(); 474} 475 476// must be called with mLock held 477void AudioTrack::flush_l() 478{ 479 ALOGV("flush"); 480 481 // clear playback marker and periodic update counter 482 mMarkerPosition = 0; 483 mMarkerReached = false; 484 mUpdatePeriod = 0; 485 486 if (!mActive) { 487 mFlushed = true; 488 mAudioTrack->flush(); 489 // Release AudioTrack callback thread in case it was waiting for new buffers 490 // in AudioTrack::obtainBuffer() 491 mCblk->cv.signal(); 492 } 493} 494 495void AudioTrack::pause() 496{ 497 ALOGV("pause"); 498 AutoMutex lock(mLock); 499 if (mActive) { 500 mActive = false; 501 mCblk->cv.signal(); 502 mAudioTrack->pause(); 503 } 504} 505 506void AudioTrack::mute(bool e) 507{ 508 mAudioTrack->mute(e); 509 mMuted = e; 510} 511 512bool AudioTrack::muted() const 513{ 514 return mMuted; 515} 516 517status_t AudioTrack::setVolume(float left, float right) 518{ 519 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 520 return BAD_VALUE; 521 } 522 523 AutoMutex lock(mLock); 524 mVolume[LEFT] = left; 525 mVolume[RIGHT] = right; 526 527 mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 528 529 return NO_ERROR; 530} 531 532void AudioTrack::getVolume(float* left, float* right) const 533{ 534 if (left != NULL) { 535 *left = mVolume[LEFT]; 536 } 537 if (right != NULL) { 538 *right = mVolume[RIGHT]; 539 } 540} 541 542status_t AudioTrack::setAuxEffectSendLevel(float level) 543{ 544 ALOGV("setAuxEffectSendLevel(%f)", level); 545 if (level < 0.0f || level > 1.0f) { 546 return BAD_VALUE; 547 } 548 AutoMutex lock(mLock); 549 550 mSendLevel = level; 551 552 mCblk->setSendLevel(level); 553 554 return NO_ERROR; 555} 556 557void AudioTrack::getAuxEffectSendLevel(float* level) const 558{ 559 if (level != NULL) { 560 *level = mSendLevel; 561 } 562} 563 564status_t AudioTrack::setSampleRate(int rate) 565{ 566 int afSamplingRate; 567 568 if (mIsTimed) { 569 return INVALID_OPERATION; 570 } 571 572 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 573 return NO_INIT; 574 } 575 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 576 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 577 578 AutoMutex lock(mLock); 579 mCblk->sampleRate = rate; 580 return NO_ERROR; 581} 582 583uint32_t AudioTrack::getSampleRate() const 584{ 585 if (mIsTimed) { 586 return INVALID_OPERATION; 587 } 588 589 AutoMutex lock(mLock); 590 return mCblk->sampleRate; 591} 592 593status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 594{ 595 AutoMutex lock(mLock); 596 return setLoop_l(loopStart, loopEnd, loopCount); 597} 598 599// must be called with mLock held 600status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 601{ 602 audio_track_cblk_t* cblk = mCblk; 603 604 Mutex::Autolock _l(cblk->lock); 605 606 if (loopCount == 0) { 607 cblk->loopStart = UINT_MAX; 608 cblk->loopEnd = UINT_MAX; 609 cblk->loopCount = 0; 610 mLoopCount = 0; 611 return NO_ERROR; 612 } 613 614 if (mIsTimed) { 615 return INVALID_OPERATION; 616 } 617 618 if (loopStart >= loopEnd || 619 loopEnd - loopStart > cblk->frameCount || 620 cblk->server > loopStart) { 621 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, " 622 "user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 623 return BAD_VALUE; 624 } 625 626 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 627 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " 628 "framecount %d", 629 loopStart, loopEnd, cblk->frameCount); 630 return BAD_VALUE; 631 } 632 633 cblk->loopStart = loopStart; 634 cblk->loopEnd = loopEnd; 635 cblk->loopCount = loopCount; 636 mLoopCount = loopCount; 637 638 return NO_ERROR; 639} 640 641status_t AudioTrack::setMarkerPosition(uint32_t marker) 642{ 643 if (mCbf == NULL) return INVALID_OPERATION; 644 645 mMarkerPosition = marker; 646 mMarkerReached = false; 647 648 return NO_ERROR; 649} 650 651status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 652{ 653 if (marker == NULL) return BAD_VALUE; 654 655 *marker = mMarkerPosition; 656 657 return NO_ERROR; 658} 659 660status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 661{ 662 if (mCbf == NULL) return INVALID_OPERATION; 663 664 uint32_t curPosition; 665 getPosition(&curPosition); 666 mNewPosition = curPosition + updatePeriod; 667 mUpdatePeriod = updatePeriod; 668 669 return NO_ERROR; 670} 671 672status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 673{ 674 if (updatePeriod == NULL) return BAD_VALUE; 675 676 *updatePeriod = mUpdatePeriod; 677 678 return NO_ERROR; 679} 680 681status_t AudioTrack::setPosition(uint32_t position) 682{ 683 if (mIsTimed) return INVALID_OPERATION; 684 685 AutoMutex lock(mLock); 686 687 if (!stopped_l()) return INVALID_OPERATION; 688 689 Mutex::Autolock _l(mCblk->lock); 690 691 if (position > mCblk->user) return BAD_VALUE; 692 693 mCblk->server = position; 694 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 695 696 return NO_ERROR; 697} 698 699status_t AudioTrack::getPosition(uint32_t *position) 700{ 701 if (position == NULL) return BAD_VALUE; 702 AutoMutex lock(mLock); 703 *position = mFlushed ? 0 : mCblk->server; 704 705 return NO_ERROR; 706} 707 708status_t AudioTrack::reload() 709{ 710 AutoMutex lock(mLock); 711 712 if (!stopped_l()) return INVALID_OPERATION; 713 714 flush_l(); 715 716 mCblk->stepUser(mCblk->frameCount); 717 718 return NO_ERROR; 719} 720 721audio_io_handle_t AudioTrack::getOutput() 722{ 723 AutoMutex lock(mLock); 724 return getOutput_l(); 725} 726 727// must be called with mLock held 728audio_io_handle_t AudioTrack::getOutput_l() 729{ 730 return AudioSystem::getOutput(mStreamType, 731 mCblk->sampleRate, mFormat, mChannelMask, mFlags); 732} 733 734int AudioTrack::getSessionId() const 735{ 736 return mSessionId; 737} 738 739status_t AudioTrack::attachAuxEffect(int effectId) 740{ 741 ALOGV("attachAuxEffect(%d)", effectId); 742 status_t status = mAudioTrack->attachAuxEffect(effectId); 743 if (status == NO_ERROR) { 744 mAuxEffectId = effectId; 745 } 746 return status; 747} 748 749// ------------------------------------------------------------------------- 750 751// must be called with mLock held 752status_t AudioTrack::createTrack_l( 753 audio_stream_type_t streamType, 754 uint32_t sampleRate, 755 audio_format_t format, 756 audio_channel_mask_t channelMask, 757 int frameCount, 758 audio_output_flags_t flags, 759 const sp<IMemory>& sharedBuffer, 760 audio_io_handle_t output) 761{ 762 status_t status; 763 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 764 if (audioFlinger == 0) { 765 ALOGE("Could not get audioflinger"); 766 return NO_INIT; 767 } 768 769 uint32_t afLatency; 770 if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { 771 return NO_INIT; 772 } 773 774 // Client decides whether the track is TIMED (see below), but can only express a preference 775 // for FAST. Server will perform additional tests. 776 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 777 // either of these use cases: 778 // use case 1: shared buffer 779 (sharedBuffer != 0) || 780 // use case 2: callback handler 781 (mCbf != NULL))) { 782 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 783 // once denied, do not request again if IAudioTrack is re-created 784 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 785 mFlags = flags; 786 } 787 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 788 789 mNotificationFramesAct = mNotificationFramesReq; 790 791 if (!audio_is_linear_pcm(format)) { 792 793 if (sharedBuffer != 0) { 794 // Same comment as below about ignoring frameCount parameter for set() 795 frameCount = sharedBuffer->size(); 796 } else if (frameCount == 0) { 797 int afFrameCount; 798 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 799 return NO_INIT; 800 } 801 frameCount = afFrameCount; 802 } 803 804 } else if (sharedBuffer != 0) { 805 806 // Ensure that buffer alignment matches channelCount 807 int channelCount = popcount(channelMask); 808 // 8-bit data in shared memory is not currently supported by AudioFlinger 809 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 810 if (channelCount > 1) { 811 // More than 2 channels does not require stronger alignment than stereo 812 alignment <<= 1; 813 } 814 if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 815 ALOGE("Invalid buffer alignment: address %p, channelCount %d", 816 sharedBuffer->pointer(), channelCount); 817 return BAD_VALUE; 818 } 819 820 // When initializing a shared buffer AudioTrack via constructors, 821 // there's no frameCount parameter. 822 // But when initializing a shared buffer AudioTrack via set(), 823 // there _is_ a frameCount parameter. We silently ignore it. 824 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 825 826 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 827 828 // FIXME move these calculations and associated checks to server 829 int afSampleRate; 830 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 831 return NO_INIT; 832 } 833 int afFrameCount; 834 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 835 return NO_INIT; 836 } 837 838 // Ensure that buffer depth covers at least audio hardware latency 839 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 840 if (minBufCount < 2) minBufCount = 2; 841 842 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 843 ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d" 844 ", afLatency=%d", 845 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 846 847 if (frameCount == 0) { 848 frameCount = minFrameCount; 849 } 850 if (mNotificationFramesAct == 0) { 851 mNotificationFramesAct = frameCount/2; 852 } 853 // Make sure that application is notified with sufficient margin 854 // before underrun 855 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 856 mNotificationFramesAct = frameCount/2; 857 } 858 if (frameCount < minFrameCount) { 859 // not ALOGW because it happens all the time when playing key clicks over A2DP 860 ALOGV("Minimum buffer size corrected from %d to %d", 861 frameCount, minFrameCount); 862 frameCount = minFrameCount; 863 } 864 865 } else { 866 // For fast tracks, the frame count calculations and checks are done by server 867 } 868 869 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 870 if (mIsTimed) { 871 trackFlags |= IAudioFlinger::TRACK_TIMED; 872 } 873 874 pid_t tid = -1; 875 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 876 trackFlags |= IAudioFlinger::TRACK_FAST; 877 if (mAudioTrackThread != 0) { 878 tid = mAudioTrackThread->getTid(); 879 } 880 } 881 882 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 883 streamType, 884 sampleRate, 885 format, 886 channelMask, 887 frameCount, 888 trackFlags, 889 sharedBuffer, 890 output, 891 tid, 892 &mSessionId, 893 &status); 894 895 if (track == 0) { 896 ALOGE("AudioFlinger could not create track, status: %d", status); 897 return status; 898 } 899 sp<IMemory> cblk = track->getCblk(); 900 if (cblk == 0) { 901 ALOGE("Could not get control block"); 902 return NO_INIT; 903 } 904 mAudioTrack = track; 905 mCblkMemory = cblk; 906 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 907 // old has the previous value of mCblk->flags before the "or" operation 908 int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 909 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 910 if (old & CBLK_FAST) { 911 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount); 912 } else { 913 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount); 914 // once denied, do not request again if IAudioTrack is re-created 915 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 916 mFlags = flags; 917 } 918 if (sharedBuffer == 0) { 919 mNotificationFramesAct = mCblk->frameCount/2; 920 } 921 } 922 if (sharedBuffer == 0) { 923 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 924 } else { 925 mCblk->buffers = sharedBuffer->pointer(); 926 // Force buffer full condition as data is already present in shared memory 927 mCblk->stepUser(mCblk->frameCount); 928 } 929 930 mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 931 uint16_t(mVolume[LEFT] * 0x1000)); 932 mCblk->setSendLevel(mSendLevel); 933 mAudioTrack->attachAuxEffect(mAuxEffectId); 934 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 935 mCblk->waitTimeMs = 0; 936 mRemainingFrames = mNotificationFramesAct; 937 // FIXME don't believe this lie 938 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 939 // If IAudioTrack is re-created, don't let the requested frameCount 940 // decrease. This can confuse clients that cache frameCount(). 941 if (mCblk->frameCount > mFrameCount) { 942 mFrameCount = mCblk->frameCount; 943 } 944 return NO_ERROR; 945} 946 947status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 948{ 949 AutoMutex lock(mLock); 950 bool active; 951 status_t result = NO_ERROR; 952 audio_track_cblk_t* cblk = mCblk; 953 uint32_t framesReq = audioBuffer->frameCount; 954 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 955 956 audioBuffer->frameCount = 0; 957 audioBuffer->size = 0; 958 959 uint32_t framesAvail = cblk->framesAvailable(); 960 961 cblk->lock.lock(); 962 if (cblk->flags & CBLK_INVALID_MSK) { 963 goto create_new_track; 964 } 965 cblk->lock.unlock(); 966 967 if (framesAvail == 0) { 968 cblk->lock.lock(); 969 goto start_loop_here; 970 while (framesAvail == 0) { 971 active = mActive; 972 if (CC_UNLIKELY(!active)) { 973 ALOGV("Not active and NO_MORE_BUFFERS"); 974 cblk->lock.unlock(); 975 return NO_MORE_BUFFERS; 976 } 977 if (CC_UNLIKELY(!waitCount)) { 978 cblk->lock.unlock(); 979 return WOULD_BLOCK; 980 } 981 if (!(cblk->flags & CBLK_INVALID_MSK)) { 982 mLock.unlock(); 983 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 984 cblk->lock.unlock(); 985 mLock.lock(); 986 if (!mActive) { 987 return status_t(STOPPED); 988 } 989 cblk->lock.lock(); 990 } 991 992 if (cblk->flags & CBLK_INVALID_MSK) { 993 goto create_new_track; 994 } 995 if (CC_UNLIKELY(result != NO_ERROR)) { 996 cblk->waitTimeMs += waitTimeMs; 997 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 998 // timing out when a loop has been set and we have already written upto loop end 999 // is a normal condition: no need to wake AudioFlinger up. 1000 if (cblk->user < cblk->loopEnd) { 1001 ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " 1002 "server=%08x", this, cblk->mName, cblk->user, cblk->server); 1003 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 1004 cblk->lock.unlock(); 1005 result = mAudioTrack->start(); 1006 cblk->lock.lock(); 1007 if (result == DEAD_OBJECT) { 1008 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 1009create_new_track: 1010 result = restoreTrack_l(cblk, false); 1011 } 1012 if (result != NO_ERROR) { 1013 ALOGW("obtainBuffer create Track error %d", result); 1014 cblk->lock.unlock(); 1015 return result; 1016 } 1017 } 1018 cblk->waitTimeMs = 0; 1019 } 1020 1021 if (--waitCount == 0) { 1022 cblk->lock.unlock(); 1023 return TIMED_OUT; 1024 } 1025 } 1026 // read the server count again 1027 start_loop_here: 1028 framesAvail = cblk->framesAvailable_l(); 1029 } 1030 cblk->lock.unlock(); 1031 } 1032 1033 cblk->waitTimeMs = 0; 1034 1035 if (framesReq > framesAvail) { 1036 framesReq = framesAvail; 1037 } 1038 1039 uint32_t u = cblk->user; 1040 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 1041 1042 if (framesReq > bufferEnd - u) { 1043 framesReq = bufferEnd - u; 1044 } 1045 1046 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 1047 audioBuffer->channelCount = mChannelCount; 1048 audioBuffer->frameCount = framesReq; 1049 audioBuffer->size = framesReq * cblk->frameSize; 1050 if (audio_is_linear_pcm(mFormat)) { 1051 audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; 1052 } else { 1053 audioBuffer->format = mFormat; 1054 } 1055 audioBuffer->raw = (int8_t *)cblk->buffer(u); 1056 active = mActive; 1057 return active ? status_t(NO_ERROR) : status_t(STOPPED); 1058} 1059 1060void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1061{ 1062 AutoMutex lock(mLock); 1063 mCblk->stepUser(audioBuffer->frameCount); 1064 if (audioBuffer->frameCount > 0) { 1065 // restart track if it was disabled by audioflinger due to previous underrun 1066 if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1067 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1068 ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName); 1069 mAudioTrack->start(); 1070 } 1071 } 1072} 1073 1074// ------------------------------------------------------------------------- 1075 1076ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1077{ 1078 1079 if (mSharedBuffer != 0) return INVALID_OPERATION; 1080 if (mIsTimed) return INVALID_OPERATION; 1081 1082 if (ssize_t(userSize) < 0) { 1083 // Sanity-check: user is most-likely passing an error code, and it would 1084 // make the return value ambiguous (actualSize vs error). 1085 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1086 buffer, userSize, userSize); 1087 return BAD_VALUE; 1088 } 1089 1090 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1091 1092 if (userSize == 0) { 1093 return 0; 1094 } 1095 1096 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1097 // while we are accessing the cblk 1098 mLock.lock(); 1099 sp<IAudioTrack> audioTrack = mAudioTrack; 1100 sp<IMemory> iMem = mCblkMemory; 1101 mLock.unlock(); 1102 1103 ssize_t written = 0; 1104 const int8_t *src = (const int8_t *)buffer; 1105 Buffer audioBuffer; 1106 size_t frameSz = frameSize(); 1107 1108 do { 1109 audioBuffer.frameCount = userSize/frameSz; 1110 1111 status_t err = obtainBuffer(&audioBuffer, -1); 1112 if (err < 0) { 1113 // out of buffers, return #bytes written 1114 if (err == status_t(NO_MORE_BUFFERS)) 1115 break; 1116 return ssize_t(err); 1117 } 1118 1119 size_t toWrite; 1120 1121 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1122 // Divide capacity by 2 to take expansion into account 1123 toWrite = audioBuffer.size>>1; 1124 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1125 } else { 1126 toWrite = audioBuffer.size; 1127 memcpy(audioBuffer.i8, src, toWrite); 1128 src += toWrite; 1129 } 1130 userSize -= toWrite; 1131 written += toWrite; 1132 1133 releaseBuffer(&audioBuffer); 1134 } while (userSize >= frameSz); 1135 1136 return written; 1137} 1138 1139// ------------------------------------------------------------------------- 1140 1141TimedAudioTrack::TimedAudioTrack() { 1142 mIsTimed = true; 1143} 1144 1145status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1146{ 1147 status_t result = UNKNOWN_ERROR; 1148 1149 // If the track is not invalid already, try to allocate a buffer. alloc 1150 // fails indicating that the server is dead, flag the track as invalid so 1151 // we can attempt to restore in just a bit. 1152 if (!(mCblk->flags & CBLK_INVALID_MSK)) { 1153 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1154 if (result == DEAD_OBJECT) { 1155 android_atomic_or(CBLK_INVALID_ON, &mCblk->flags); 1156 } 1157 } 1158 1159 // If the track is invalid at this point, attempt to restore it. and try the 1160 // allocation one more time. 1161 if (mCblk->flags & CBLK_INVALID_MSK) { 1162 mCblk->lock.lock(); 1163 result = restoreTrack_l(mCblk, false); 1164 mCblk->lock.unlock(); 1165 1166 if (result == OK) 1167 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1168 } 1169 1170 return result; 1171} 1172 1173status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1174 int64_t pts) 1175{ 1176 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1177 { 1178 AutoMutex lock(mLock); 1179 // restart track if it was disabled by audioflinger due to previous underrun 1180 if (buffer->size() != 0 && status == NO_ERROR && 1181 mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1182 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1183 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1184 mAudioTrack->start(); 1185 } 1186 } 1187 return status; 1188} 1189 1190status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1191 TargetTimeline target) 1192{ 1193 return mAudioTrack->setMediaTimeTransform(xform, target); 1194} 1195 1196// ------------------------------------------------------------------------- 1197 1198bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1199{ 1200 Buffer audioBuffer; 1201 uint32_t frames; 1202 size_t writtenSize; 1203 1204 mLock.lock(); 1205 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1206 // while we are accessing the cblk 1207 sp<IAudioTrack> audioTrack = mAudioTrack; 1208 sp<IMemory> iMem = mCblkMemory; 1209 audio_track_cblk_t* cblk = mCblk; 1210 bool active = mActive; 1211 mLock.unlock(); 1212 1213 // Manage underrun callback 1214 if (active && (cblk->framesAvailable() == cblk->frameCount)) { 1215 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1216 if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { 1217 mCbf(EVENT_UNDERRUN, mUserData, 0); 1218 if (cblk->server == cblk->frameCount) { 1219 mCbf(EVENT_BUFFER_END, mUserData, 0); 1220 } 1221 if (mSharedBuffer != 0) return false; 1222 } 1223 } 1224 1225 // Manage loop end callback 1226 while (mLoopCount > cblk->loopCount) { 1227 int loopCount = -1; 1228 mLoopCount--; 1229 if (mLoopCount >= 0) loopCount = mLoopCount; 1230 1231 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1232 } 1233 1234 // Manage marker callback 1235 if (!mMarkerReached && (mMarkerPosition > 0)) { 1236 if (cblk->server >= mMarkerPosition) { 1237 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1238 mMarkerReached = true; 1239 } 1240 } 1241 1242 // Manage new position callback 1243 if (mUpdatePeriod > 0) { 1244 while (cblk->server >= mNewPosition) { 1245 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1246 mNewPosition += mUpdatePeriod; 1247 } 1248 } 1249 1250 // If Shared buffer is used, no data is requested from client. 1251 if (mSharedBuffer != 0) { 1252 frames = 0; 1253 } else { 1254 frames = mRemainingFrames; 1255 } 1256 1257 // See description of waitCount parameter at declaration of obtainBuffer(). 1258 // The logic below prevents us from being stuck below at obtainBuffer() 1259 // not being able to handle timed events (position, markers, loops). 1260 int32_t waitCount = -1; 1261 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1262 waitCount = 1; 1263 } 1264 1265 do { 1266 1267 audioBuffer.frameCount = frames; 1268 1269 status_t err = obtainBuffer(&audioBuffer, waitCount); 1270 if (err < NO_ERROR) { 1271 if (err != TIMED_OUT) { 1272 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), 1273 "Error obtaining an audio buffer, giving up."); 1274 return false; 1275 } 1276 break; 1277 } 1278 if (err == status_t(STOPPED)) return false; 1279 1280 // Divide buffer size by 2 to take into account the expansion 1281 // due to 8 to 16 bit conversion: the callback must fill only half 1282 // of the destination buffer 1283 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1284 audioBuffer.size >>= 1; 1285 } 1286 1287 size_t reqSize = audioBuffer.size; 1288 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1289 writtenSize = audioBuffer.size; 1290 1291 // Sanity check on returned size 1292 if (ssize_t(writtenSize) <= 0) { 1293 // The callback is done filling buffers 1294 // Keep this thread going to handle timed events and 1295 // still try to get more data in intervals of WAIT_PERIOD_MS 1296 // but don't just loop and block the CPU, so wait 1297 usleep(WAIT_PERIOD_MS*1000); 1298 break; 1299 } 1300 1301 if (writtenSize > reqSize) writtenSize = reqSize; 1302 1303 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1304 // 8 to 16 bit conversion, note that source and destination are the same address 1305 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1306 writtenSize <<= 1; 1307 } 1308 1309 audioBuffer.size = writtenSize; 1310 // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for 1311 // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of 1312 // 16 bit. 1313 audioBuffer.frameCount = writtenSize/mCblk->frameSize; 1314 1315 frames -= audioBuffer.frameCount; 1316 1317 releaseBuffer(&audioBuffer); 1318 } 1319 while (frames); 1320 1321 if (frames == 0) { 1322 mRemainingFrames = mNotificationFramesAct; 1323 } else { 1324 mRemainingFrames = frames; 1325 } 1326 return true; 1327} 1328 1329// must be called with mLock and cblk.lock held. Callers must also hold strong references on 1330// the IAudioTrack and IMemory in case they are recreated here. 1331// If the IAudioTrack is successfully restored, the cblk pointer is updated 1332status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) 1333{ 1334 status_t result; 1335 1336 if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { 1337 ALOGW("dead IAudioTrack, creating a new one from %s TID %d", 1338 fromStart ? "start()" : "obtainBuffer()", gettid()); 1339 1340 // signal old cblk condition so that other threads waiting for available buffers stop 1341 // waiting now 1342 cblk->cv.broadcast(); 1343 cblk->lock.unlock(); 1344 1345 // refresh the audio configuration cache in this process to make sure we get new 1346 // output parameters in getOutput_l() and createTrack_l() 1347 AudioSystem::clearAudioConfigCache(); 1348 1349 // if the new IAudioTrack is created, createTrack_l() will modify the 1350 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1351 // It will also delete the strong references on previous IAudioTrack and IMemory 1352 result = createTrack_l(mStreamType, 1353 cblk->sampleRate, 1354 mFormat, 1355 mChannelMask, 1356 mFrameCount, 1357 mFlags, 1358 mSharedBuffer, 1359 getOutput_l()); 1360 1361 if (result == NO_ERROR) { 1362 uint32_t user = cblk->user; 1363 uint32_t server = cblk->server; 1364 // restore write index and set other indexes to reflect empty buffer status 1365 mCblk->user = user; 1366 mCblk->server = user; 1367 mCblk->userBase = user; 1368 mCblk->serverBase = user; 1369 // restore loop: this is not guaranteed to succeed if new frame count is not 1370 // compatible with loop length 1371 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1372 if (!fromStart) { 1373 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1374 // Make sure that a client relying on callback events indicating underrun or 1375 // the actual amount of audio frames played (e.g SoundPool) receives them. 1376 if (mSharedBuffer == 0) { 1377 uint32_t frames = 0; 1378 if (user > server) { 1379 frames = ((user - server) > mCblk->frameCount) ? 1380 mCblk->frameCount : (user - server); 1381 memset(mCblk->buffers, 0, frames * mCblk->frameSize); 1382 } 1383 // restart playback even if buffer is not completely filled. 1384 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 1385 // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to 1386 // the client 1387 mCblk->stepUser(frames); 1388 } 1389 } 1390 if (mSharedBuffer != 0) { 1391 mCblk->stepUser(mCblk->frameCount); 1392 } 1393 if (mActive) { 1394 result = mAudioTrack->start(); 1395 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1396 } 1397 if (fromStart && result == NO_ERROR) { 1398 mNewPosition = mCblk->server + mUpdatePeriod; 1399 } 1400 } 1401 if (result != NO_ERROR) { 1402 android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); 1403 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1404 } 1405 mRestoreStatus = result; 1406 // signal old cblk condition for other threads waiting for restore completion 1407 android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); 1408 cblk->cv.broadcast(); 1409 } else { 1410 if (!(cblk->flags & CBLK_RESTORED_MSK)) { 1411 ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); 1412 mLock.unlock(); 1413 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1414 if (result == NO_ERROR) { 1415 result = mRestoreStatus; 1416 } 1417 cblk->lock.unlock(); 1418 mLock.lock(); 1419 } else { 1420 ALOGW("dead IAudioTrack, already restored TID %d", gettid()); 1421 result = mRestoreStatus; 1422 cblk->lock.unlock(); 1423 } 1424 } 1425 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1426 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1427 1428 if (result == NO_ERROR) { 1429 // from now on we switch to the newly created cblk 1430 cblk = mCblk; 1431 } 1432 cblk->lock.lock(); 1433 1434 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1435 1436 return result; 1437} 1438 1439status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1440{ 1441 1442 const size_t SIZE = 256; 1443 char buffer[SIZE]; 1444 String8 result; 1445 1446 result.append(" AudioTrack::dump\n"); 1447 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1448 mVolume[0], mVolume[1]); 1449 result.append(buffer); 1450 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1451 mChannelCount, mCblk->frameCount); 1452 result.append(buffer); 1453 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", 1454 (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1455 result.append(buffer); 1456 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1457 result.append(buffer); 1458 ::write(fd, result.string(), result.size()); 1459 return NO_ERROR; 1460} 1461 1462// ========================================================================= 1463 1464AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1465 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1466{ 1467} 1468 1469AudioTrack::AudioTrackThread::~AudioTrackThread() 1470{ 1471} 1472 1473bool AudioTrack::AudioTrackThread::threadLoop() 1474{ 1475 { 1476 AutoMutex _l(mMyLock); 1477 if (mPaused) { 1478 mMyCond.wait(mMyLock); 1479 // caller will check for exitPending() 1480 return true; 1481 } 1482 } 1483 if (!mReceiver.processAudioBuffer(this)) { 1484 pause(); 1485 } 1486 return true; 1487} 1488 1489void AudioTrack::AudioTrackThread::requestExit() 1490{ 1491 // must be in this order to avoid a race condition 1492 Thread::requestExit(); 1493 resume(); 1494} 1495 1496void AudioTrack::AudioTrackThread::pause() 1497{ 1498 AutoMutex _l(mMyLock); 1499 mPaused = true; 1500} 1501 1502void AudioTrack::AudioTrackThread::resume() 1503{ 1504 AutoMutex _l(mMyLock); 1505 if (mPaused) { 1506 mPaused = false; 1507 mMyCond.signal(); 1508 } 1509} 1510 1511// ========================================================================= 1512 1513 1514audio_track_cblk_t::audio_track_cblk_t() 1515 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1516 userBase(0), serverBase(0), buffers(NULL), frameCount(0), 1517 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), 1518 mSendLevel(0), flags(0) 1519{ 1520} 1521 1522uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1523{ 1524 ALOGV("stepuser %08x %08x %d", user, server, frameCount); 1525 1526 uint32_t u = user; 1527 u += frameCount; 1528 // Ensure that user is never ahead of server for AudioRecord 1529 if (flags & CBLK_DIRECTION_MSK) { 1530 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1531 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1532 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1533 } 1534 } else if (u > server) { 1535 ALOGW("stepUser occurred after track reset"); 1536 u = server; 1537 } 1538 1539 uint32_t fc = this->frameCount; 1540 if (u >= fc) { 1541 // common case, user didn't just wrap 1542 if (u - fc >= userBase ) { 1543 userBase += fc; 1544 } 1545 } else if (u >= userBase + fc) { 1546 // user just wrapped 1547 userBase += fc; 1548 } 1549 1550 user = u; 1551 1552 // Clear flow control error condition as new data has been written/read to/from buffer. 1553 if (flags & CBLK_UNDERRUN_MSK) { 1554 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1555 } 1556 1557 return u; 1558} 1559 1560bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1561{ 1562 ALOGV("stepserver %08x %08x %d", user, server, frameCount); 1563 1564 if (!tryLock()) { 1565 ALOGW("stepServer() could not lock cblk"); 1566 return false; 1567 } 1568 1569 uint32_t s = server; 1570 bool flushed = (s == user); 1571 1572 s += frameCount; 1573 if (flags & CBLK_DIRECTION_MSK) { 1574 // Mark that we have read the first buffer so that next time stepUser() is called 1575 // we switch to normal obtainBuffer() timeout period 1576 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1577 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1578 } 1579 // It is possible that we receive a flush() 1580 // while the mixer is processing a block: in this case, 1581 // stepServer() is called After the flush() has reset u & s and 1582 // we have s > u 1583 if (flushed) { 1584 ALOGW("stepServer occurred after track reset"); 1585 s = user; 1586 } 1587 } 1588 1589 if (s >= loopEnd) { 1590 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1591 s = loopStart; 1592 if (--loopCount == 0) { 1593 loopEnd = UINT_MAX; 1594 loopStart = UINT_MAX; 1595 } 1596 } 1597 1598 uint32_t fc = this->frameCount; 1599 if (s >= fc) { 1600 // common case, server didn't just wrap 1601 if (s - fc >= serverBase ) { 1602 serverBase += fc; 1603 } 1604 } else if (s >= serverBase + fc) { 1605 // server just wrapped 1606 serverBase += fc; 1607 } 1608 1609 server = s; 1610 1611 if (!(flags & CBLK_INVALID_MSK)) { 1612 cv.signal(); 1613 } 1614 lock.unlock(); 1615 return true; 1616} 1617 1618void* audio_track_cblk_t::buffer(uint32_t offset) const 1619{ 1620 return (int8_t *)buffers + (offset - userBase) * frameSize; 1621} 1622 1623uint32_t audio_track_cblk_t::framesAvailable() 1624{ 1625 Mutex::Autolock _l(lock); 1626 return framesAvailable_l(); 1627} 1628 1629uint32_t audio_track_cblk_t::framesAvailable_l() 1630{ 1631 uint32_t u = user; 1632 uint32_t s = server; 1633 1634 if (flags & CBLK_DIRECTION_MSK) { 1635 uint32_t limit = (s < loopStart) ? s : loopStart; 1636 return limit + frameCount - u; 1637 } else { 1638 return frameCount + u - s; 1639 } 1640} 1641 1642uint32_t audio_track_cblk_t::framesReady() 1643{ 1644 uint32_t u = user; 1645 uint32_t s = server; 1646 1647 if (flags & CBLK_DIRECTION_MSK) { 1648 if (u < loopEnd) { 1649 return u - s; 1650 } else { 1651 // do not block on mutex shared with client on AudioFlinger side 1652 if (!tryLock()) { 1653 ALOGW("framesReady() could not lock cblk"); 1654 return 0; 1655 } 1656 uint32_t frames = UINT_MAX; 1657 if (loopCount >= 0) { 1658 frames = (loopEnd - loopStart)*loopCount + u - s; 1659 } 1660 lock.unlock(); 1661 return frames; 1662 } 1663 } else { 1664 return s - u; 1665 } 1666} 1667 1668bool audio_track_cblk_t::tryLock() 1669{ 1670 // the code below simulates lock-with-timeout 1671 // we MUST do this to protect the AudioFlinger server 1672 // as this lock is shared with the client. 1673 status_t err; 1674 1675 err = lock.tryLock(); 1676 if (err == -EBUSY) { // just wait a bit 1677 usleep(1000); 1678 err = lock.tryLock(); 1679 } 1680 if (err != NO_ERROR) { 1681 // probably, the client just died. 1682 return false; 1683 } 1684 return true; 1685} 1686 1687// ------------------------------------------------------------------------- 1688 1689}; // namespace android 1690