AudioTrack.cpp revision b1c0993b215c5c3eebd1c6bafc22bba23d57a70b
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 532status_t AudioTrack::setVolume(float volume) 533{ 534 return setVolume(volume, volume); 535} 536 537status_t AudioTrack::setAuxEffectSendLevel(float level) 538{ 539 ALOGV("setAuxEffectSendLevel(%f)", level); 540 if (level < 0.0f || level > 1.0f) { 541 return BAD_VALUE; 542 } 543 AutoMutex lock(mLock); 544 545 mSendLevel = level; 546 547 mCblk->setSendLevel(level); 548 549 return NO_ERROR; 550} 551 552void AudioTrack::getAuxEffectSendLevel(float* level) const 553{ 554 if (level != NULL) { 555 *level = mSendLevel; 556 } 557} 558 559status_t AudioTrack::setSampleRate(int rate) 560{ 561 int afSamplingRate; 562 563 if (mIsTimed) { 564 return INVALID_OPERATION; 565 } 566 567 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 568 return NO_INIT; 569 } 570 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 571 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 572 573 AutoMutex lock(mLock); 574 mCblk->sampleRate = rate; 575 return NO_ERROR; 576} 577 578uint32_t AudioTrack::getSampleRate() const 579{ 580 if (mIsTimed) { 581 return INVALID_OPERATION; 582 } 583 584 AutoMutex lock(mLock); 585 return mCblk->sampleRate; 586} 587 588status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 589{ 590 AutoMutex lock(mLock); 591 return setLoop_l(loopStart, loopEnd, loopCount); 592} 593 594// must be called with mLock held 595status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 596{ 597 audio_track_cblk_t* cblk = mCblk; 598 599 Mutex::Autolock _l(cblk->lock); 600 601 if (loopCount == 0) { 602 cblk->loopStart = UINT_MAX; 603 cblk->loopEnd = UINT_MAX; 604 cblk->loopCount = 0; 605 mLoopCount = 0; 606 return NO_ERROR; 607 } 608 609 if (mIsTimed) { 610 return INVALID_OPERATION; 611 } 612 613 if (loopStart >= loopEnd || 614 loopEnd - loopStart > cblk->frameCount || 615 cblk->server > loopStart) { 616 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, " 617 "user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 618 return BAD_VALUE; 619 } 620 621 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 622 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " 623 "framecount %d", 624 loopStart, loopEnd, cblk->frameCount); 625 return BAD_VALUE; 626 } 627 628 cblk->loopStart = loopStart; 629 cblk->loopEnd = loopEnd; 630 cblk->loopCount = loopCount; 631 mLoopCount = loopCount; 632 633 return NO_ERROR; 634} 635 636status_t AudioTrack::setMarkerPosition(uint32_t marker) 637{ 638 if (mCbf == NULL) return INVALID_OPERATION; 639 640 mMarkerPosition = marker; 641 mMarkerReached = false; 642 643 return NO_ERROR; 644} 645 646status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 647{ 648 if (marker == NULL) return BAD_VALUE; 649 650 *marker = mMarkerPosition; 651 652 return NO_ERROR; 653} 654 655status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 656{ 657 if (mCbf == NULL) return INVALID_OPERATION; 658 659 uint32_t curPosition; 660 getPosition(&curPosition); 661 mNewPosition = curPosition + updatePeriod; 662 mUpdatePeriod = updatePeriod; 663 664 return NO_ERROR; 665} 666 667status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 668{ 669 if (updatePeriod == NULL) return BAD_VALUE; 670 671 *updatePeriod = mUpdatePeriod; 672 673 return NO_ERROR; 674} 675 676status_t AudioTrack::setPosition(uint32_t position) 677{ 678 if (mIsTimed) return INVALID_OPERATION; 679 680 AutoMutex lock(mLock); 681 682 if (!stopped_l()) return INVALID_OPERATION; 683 684 Mutex::Autolock _l(mCblk->lock); 685 686 if (position > mCblk->user) return BAD_VALUE; 687 688 mCblk->server = position; 689 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 690 691 return NO_ERROR; 692} 693 694status_t AudioTrack::getPosition(uint32_t *position) 695{ 696 if (position == NULL) return BAD_VALUE; 697 AutoMutex lock(mLock); 698 *position = mFlushed ? 0 : mCblk->server; 699 700 return NO_ERROR; 701} 702 703status_t AudioTrack::reload() 704{ 705 AutoMutex lock(mLock); 706 707 if (!stopped_l()) return INVALID_OPERATION; 708 709 flush_l(); 710 711 mCblk->stepUser(mCblk->frameCount); 712 713 return NO_ERROR; 714} 715 716audio_io_handle_t AudioTrack::getOutput() 717{ 718 AutoMutex lock(mLock); 719 return getOutput_l(); 720} 721 722// must be called with mLock held 723audio_io_handle_t AudioTrack::getOutput_l() 724{ 725 return AudioSystem::getOutput(mStreamType, 726 mCblk->sampleRate, mFormat, mChannelMask, mFlags); 727} 728 729int AudioTrack::getSessionId() const 730{ 731 return mSessionId; 732} 733 734status_t AudioTrack::attachAuxEffect(int effectId) 735{ 736 ALOGV("attachAuxEffect(%d)", effectId); 737 status_t status = mAudioTrack->attachAuxEffect(effectId); 738 if (status == NO_ERROR) { 739 mAuxEffectId = effectId; 740 } 741 return status; 742} 743 744// ------------------------------------------------------------------------- 745 746// must be called with mLock held 747status_t AudioTrack::createTrack_l( 748 audio_stream_type_t streamType, 749 uint32_t sampleRate, 750 audio_format_t format, 751 audio_channel_mask_t channelMask, 752 int frameCount, 753 audio_output_flags_t flags, 754 const sp<IMemory>& sharedBuffer, 755 audio_io_handle_t output) 756{ 757 status_t status; 758 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 759 if (audioFlinger == 0) { 760 ALOGE("Could not get audioflinger"); 761 return NO_INIT; 762 } 763 764 uint32_t afLatency; 765 if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { 766 return NO_INIT; 767 } 768 769 // Client decides whether the track is TIMED (see below), but can only express a preference 770 // for FAST. Server will perform additional tests. 771 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 772 // either of these use cases: 773 // use case 1: shared buffer 774 (sharedBuffer != 0) || 775 // use case 2: callback handler 776 (mCbf != NULL))) { 777 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 778 // once denied, do not request again if IAudioTrack is re-created 779 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 780 mFlags = flags; 781 } 782 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 783 784 mNotificationFramesAct = mNotificationFramesReq; 785 786 if (!audio_is_linear_pcm(format)) { 787 788 if (sharedBuffer != 0) { 789 // Same comment as below about ignoring frameCount parameter for set() 790 frameCount = sharedBuffer->size(); 791 } else if (frameCount == 0) { 792 int afFrameCount; 793 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 794 return NO_INIT; 795 } 796 frameCount = afFrameCount; 797 } 798 799 } else if (sharedBuffer != 0) { 800 801 // Ensure that buffer alignment matches channelCount 802 int channelCount = popcount(channelMask); 803 // 8-bit data in shared memory is not currently supported by AudioFlinger 804 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 805 if (channelCount > 1) { 806 // More than 2 channels does not require stronger alignment than stereo 807 alignment <<= 1; 808 } 809 if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 810 ALOGE("Invalid buffer alignment: address %p, channelCount %d", 811 sharedBuffer->pointer(), channelCount); 812 return BAD_VALUE; 813 } 814 815 // When initializing a shared buffer AudioTrack via constructors, 816 // there's no frameCount parameter. 817 // But when initializing a shared buffer AudioTrack via set(), 818 // there _is_ a frameCount parameter. We silently ignore it. 819 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 820 821 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 822 823 // FIXME move these calculations and associated checks to server 824 int afSampleRate; 825 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 826 return NO_INIT; 827 } 828 int afFrameCount; 829 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 830 return NO_INIT; 831 } 832 833 // Ensure that buffer depth covers at least audio hardware latency 834 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 835 if (minBufCount < 2) minBufCount = 2; 836 837 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 838 ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d" 839 ", afLatency=%d", 840 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 841 842 if (frameCount == 0) { 843 frameCount = minFrameCount; 844 } 845 if (mNotificationFramesAct == 0) { 846 mNotificationFramesAct = frameCount/2; 847 } 848 // Make sure that application is notified with sufficient margin 849 // before underrun 850 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 851 mNotificationFramesAct = frameCount/2; 852 } 853 if (frameCount < minFrameCount) { 854 // not ALOGW because it happens all the time when playing key clicks over A2DP 855 ALOGV("Minimum buffer size corrected from %d to %d", 856 frameCount, minFrameCount); 857 frameCount = minFrameCount; 858 } 859 860 } else { 861 // For fast tracks, the frame count calculations and checks are done by server 862 } 863 864 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 865 if (mIsTimed) { 866 trackFlags |= IAudioFlinger::TRACK_TIMED; 867 } 868 869 pid_t tid = -1; 870 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 871 trackFlags |= IAudioFlinger::TRACK_FAST; 872 if (mAudioTrackThread != 0) { 873 tid = mAudioTrackThread->getTid(); 874 } 875 } 876 877 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 878 streamType, 879 sampleRate, 880 format, 881 channelMask, 882 frameCount, 883 trackFlags, 884 sharedBuffer, 885 output, 886 tid, 887 &mSessionId, 888 &status); 889 890 if (track == 0) { 891 ALOGE("AudioFlinger could not create track, status: %d", status); 892 return status; 893 } 894 sp<IMemory> cblk = track->getCblk(); 895 if (cblk == 0) { 896 ALOGE("Could not get control block"); 897 return NO_INIT; 898 } 899 mAudioTrack = track; 900 mCblkMemory = cblk; 901 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 902 // old has the previous value of mCblk->flags before the "or" operation 903 int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 904 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 905 if (old & CBLK_FAST) { 906 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount); 907 } else { 908 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount); 909 // once denied, do not request again if IAudioTrack is re-created 910 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 911 mFlags = flags; 912 } 913 if (sharedBuffer == 0) { 914 mNotificationFramesAct = mCblk->frameCount/2; 915 } 916 } 917 if (sharedBuffer == 0) { 918 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 919 } else { 920 mCblk->buffers = sharedBuffer->pointer(); 921 // Force buffer full condition as data is already present in shared memory 922 mCblk->stepUser(mCblk->frameCount); 923 } 924 925 mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 926 uint16_t(mVolume[LEFT] * 0x1000)); 927 mCblk->setSendLevel(mSendLevel); 928 mAudioTrack->attachAuxEffect(mAuxEffectId); 929 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 930 mCblk->waitTimeMs = 0; 931 mRemainingFrames = mNotificationFramesAct; 932 // FIXME don't believe this lie 933 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 934 // If IAudioTrack is re-created, don't let the requested frameCount 935 // decrease. This can confuse clients that cache frameCount(). 936 if (mCblk->frameCount > mFrameCount) { 937 mFrameCount = mCblk->frameCount; 938 } 939 return NO_ERROR; 940} 941 942status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 943{ 944 AutoMutex lock(mLock); 945 bool active; 946 status_t result = NO_ERROR; 947 audio_track_cblk_t* cblk = mCblk; 948 uint32_t framesReq = audioBuffer->frameCount; 949 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 950 951 audioBuffer->frameCount = 0; 952 audioBuffer->size = 0; 953 954 uint32_t framesAvail = cblk->framesAvailable(); 955 956 cblk->lock.lock(); 957 if (cblk->flags & CBLK_INVALID_MSK) { 958 goto create_new_track; 959 } 960 cblk->lock.unlock(); 961 962 if (framesAvail == 0) { 963 cblk->lock.lock(); 964 goto start_loop_here; 965 while (framesAvail == 0) { 966 active = mActive; 967 if (CC_UNLIKELY(!active)) { 968 ALOGV("Not active and NO_MORE_BUFFERS"); 969 cblk->lock.unlock(); 970 return NO_MORE_BUFFERS; 971 } 972 if (CC_UNLIKELY(!waitCount)) { 973 cblk->lock.unlock(); 974 return WOULD_BLOCK; 975 } 976 if (!(cblk->flags & CBLK_INVALID_MSK)) { 977 mLock.unlock(); 978 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 979 cblk->lock.unlock(); 980 mLock.lock(); 981 if (!mActive) { 982 return status_t(STOPPED); 983 } 984 cblk->lock.lock(); 985 } 986 987 if (cblk->flags & CBLK_INVALID_MSK) { 988 goto create_new_track; 989 } 990 if (CC_UNLIKELY(result != NO_ERROR)) { 991 cblk->waitTimeMs += waitTimeMs; 992 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 993 // timing out when a loop has been set and we have already written upto loop end 994 // is a normal condition: no need to wake AudioFlinger up. 995 if (cblk->user < cblk->loopEnd) { 996 ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " 997 "server=%08x", this, cblk->mName, cblk->user, cblk->server); 998 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 999 cblk->lock.unlock(); 1000 result = mAudioTrack->start(); 1001 cblk->lock.lock(); 1002 if (result == DEAD_OBJECT) { 1003 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 1004create_new_track: 1005 result = restoreTrack_l(cblk, false); 1006 } 1007 if (result != NO_ERROR) { 1008 ALOGW("obtainBuffer create Track error %d", result); 1009 cblk->lock.unlock(); 1010 return result; 1011 } 1012 } 1013 cblk->waitTimeMs = 0; 1014 } 1015 1016 if (--waitCount == 0) { 1017 cblk->lock.unlock(); 1018 return TIMED_OUT; 1019 } 1020 } 1021 // read the server count again 1022 start_loop_here: 1023 framesAvail = cblk->framesAvailable_l(); 1024 } 1025 cblk->lock.unlock(); 1026 } 1027 1028 cblk->waitTimeMs = 0; 1029 1030 if (framesReq > framesAvail) { 1031 framesReq = framesAvail; 1032 } 1033 1034 uint32_t u = cblk->user; 1035 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 1036 1037 if (framesReq > bufferEnd - u) { 1038 framesReq = bufferEnd - u; 1039 } 1040 1041 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 1042 audioBuffer->channelCount = mChannelCount; 1043 audioBuffer->frameCount = framesReq; 1044 audioBuffer->size = framesReq * cblk->frameSize; 1045 if (audio_is_linear_pcm(mFormat)) { 1046 audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; 1047 } else { 1048 audioBuffer->format = mFormat; 1049 } 1050 audioBuffer->raw = (int8_t *)cblk->buffer(u); 1051 active = mActive; 1052 return active ? status_t(NO_ERROR) : status_t(STOPPED); 1053} 1054 1055void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1056{ 1057 AutoMutex lock(mLock); 1058 mCblk->stepUser(audioBuffer->frameCount); 1059 if (audioBuffer->frameCount > 0) { 1060 // restart track if it was disabled by audioflinger due to previous underrun 1061 if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1062 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1063 ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName); 1064 mAudioTrack->start(); 1065 } 1066 } 1067} 1068 1069// ------------------------------------------------------------------------- 1070 1071ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1072{ 1073 1074 if (mSharedBuffer != 0) return INVALID_OPERATION; 1075 if (mIsTimed) return INVALID_OPERATION; 1076 1077 if (ssize_t(userSize) < 0) { 1078 // Sanity-check: user is most-likely passing an error code, and it would 1079 // make the return value ambiguous (actualSize vs error). 1080 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1081 buffer, userSize, userSize); 1082 return BAD_VALUE; 1083 } 1084 1085 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1086 1087 if (userSize == 0) { 1088 return 0; 1089 } 1090 1091 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1092 // while we are accessing the cblk 1093 mLock.lock(); 1094 sp<IAudioTrack> audioTrack = mAudioTrack; 1095 sp<IMemory> iMem = mCblkMemory; 1096 mLock.unlock(); 1097 1098 ssize_t written = 0; 1099 const int8_t *src = (const int8_t *)buffer; 1100 Buffer audioBuffer; 1101 size_t frameSz = frameSize(); 1102 1103 do { 1104 audioBuffer.frameCount = userSize/frameSz; 1105 1106 status_t err = obtainBuffer(&audioBuffer, -1); 1107 if (err < 0) { 1108 // out of buffers, return #bytes written 1109 if (err == status_t(NO_MORE_BUFFERS)) 1110 break; 1111 return ssize_t(err); 1112 } 1113 1114 size_t toWrite; 1115 1116 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1117 // Divide capacity by 2 to take expansion into account 1118 toWrite = audioBuffer.size>>1; 1119 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1120 } else { 1121 toWrite = audioBuffer.size; 1122 memcpy(audioBuffer.i8, src, toWrite); 1123 src += toWrite; 1124 } 1125 userSize -= toWrite; 1126 written += toWrite; 1127 1128 releaseBuffer(&audioBuffer); 1129 } while (userSize >= frameSz); 1130 1131 return written; 1132} 1133 1134// ------------------------------------------------------------------------- 1135 1136TimedAudioTrack::TimedAudioTrack() { 1137 mIsTimed = true; 1138} 1139 1140status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1141{ 1142 status_t result = UNKNOWN_ERROR; 1143 1144 // If the track is not invalid already, try to allocate a buffer. alloc 1145 // fails indicating that the server is dead, flag the track as invalid so 1146 // we can attempt to restore in just a bit. 1147 if (!(mCblk->flags & CBLK_INVALID_MSK)) { 1148 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1149 if (result == DEAD_OBJECT) { 1150 android_atomic_or(CBLK_INVALID_ON, &mCblk->flags); 1151 } 1152 } 1153 1154 // If the track is invalid at this point, attempt to restore it. and try the 1155 // allocation one more time. 1156 if (mCblk->flags & CBLK_INVALID_MSK) { 1157 mCblk->lock.lock(); 1158 result = restoreTrack_l(mCblk, false); 1159 mCblk->lock.unlock(); 1160 1161 if (result == OK) 1162 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1163 } 1164 1165 return result; 1166} 1167 1168status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1169 int64_t pts) 1170{ 1171 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1172 { 1173 AutoMutex lock(mLock); 1174 // restart track if it was disabled by audioflinger due to previous underrun 1175 if (buffer->size() != 0 && status == NO_ERROR && 1176 mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { 1177 android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); 1178 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1179 mAudioTrack->start(); 1180 } 1181 } 1182 return status; 1183} 1184 1185status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1186 TargetTimeline target) 1187{ 1188 return mAudioTrack->setMediaTimeTransform(xform, target); 1189} 1190 1191// ------------------------------------------------------------------------- 1192 1193bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1194{ 1195 Buffer audioBuffer; 1196 uint32_t frames; 1197 size_t writtenSize; 1198 1199 mLock.lock(); 1200 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1201 // while we are accessing the cblk 1202 sp<IAudioTrack> audioTrack = mAudioTrack; 1203 sp<IMemory> iMem = mCblkMemory; 1204 audio_track_cblk_t* cblk = mCblk; 1205 bool active = mActive; 1206 mLock.unlock(); 1207 1208 // Manage underrun callback 1209 if (active && (cblk->framesAvailable() == cblk->frameCount)) { 1210 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1211 if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { 1212 mCbf(EVENT_UNDERRUN, mUserData, 0); 1213 if (cblk->server == cblk->frameCount) { 1214 mCbf(EVENT_BUFFER_END, mUserData, 0); 1215 } 1216 if (mSharedBuffer != 0) return false; 1217 } 1218 } 1219 1220 // Manage loop end callback 1221 while (mLoopCount > cblk->loopCount) { 1222 int loopCount = -1; 1223 mLoopCount--; 1224 if (mLoopCount >= 0) loopCount = mLoopCount; 1225 1226 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1227 } 1228 1229 // Manage marker callback 1230 if (!mMarkerReached && (mMarkerPosition > 0)) { 1231 if (cblk->server >= mMarkerPosition) { 1232 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1233 mMarkerReached = true; 1234 } 1235 } 1236 1237 // Manage new position callback 1238 if (mUpdatePeriod > 0) { 1239 while (cblk->server >= mNewPosition) { 1240 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1241 mNewPosition += mUpdatePeriod; 1242 } 1243 } 1244 1245 // If Shared buffer is used, no data is requested from client. 1246 if (mSharedBuffer != 0) { 1247 frames = 0; 1248 } else { 1249 frames = mRemainingFrames; 1250 } 1251 1252 // See description of waitCount parameter at declaration of obtainBuffer(). 1253 // The logic below prevents us from being stuck below at obtainBuffer() 1254 // not being able to handle timed events (position, markers, loops). 1255 int32_t waitCount = -1; 1256 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1257 waitCount = 1; 1258 } 1259 1260 do { 1261 1262 audioBuffer.frameCount = frames; 1263 1264 status_t err = obtainBuffer(&audioBuffer, waitCount); 1265 if (err < NO_ERROR) { 1266 if (err != TIMED_OUT) { 1267 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), 1268 "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 bool haveLogged = false; 1406 for (;;) { 1407 if (cblk->flags & CBLK_RESTORED_MSK) { 1408 ALOGW("dead IAudioTrack restored"); 1409 result = mRestoreStatus; 1410 cblk->lock.unlock(); 1411 break; 1412 } 1413 if (!haveLogged) { 1414 ALOGW("dead IAudioTrack, waiting for a new one"); 1415 haveLogged = true; 1416 } 1417 mLock.unlock(); 1418 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1419 cblk->lock.unlock(); 1420 mLock.lock(); 1421 if (result != NO_ERROR) { 1422 ALOGW("timed out"); 1423 break; 1424 } 1425 cblk->lock.lock(); 1426 } 1427 } 1428 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1429 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1430 1431 if (result == NO_ERROR) { 1432 // from now on we switch to the newly created cblk 1433 cblk = mCblk; 1434 } 1435 cblk->lock.lock(); 1436 1437 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1438 1439 return result; 1440} 1441 1442status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1443{ 1444 1445 const size_t SIZE = 256; 1446 char buffer[SIZE]; 1447 String8 result; 1448 1449 result.append(" AudioTrack::dump\n"); 1450 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1451 mVolume[0], mVolume[1]); 1452 result.append(buffer); 1453 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1454 mChannelCount, mCblk->frameCount); 1455 result.append(buffer); 1456 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", 1457 (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1458 result.append(buffer); 1459 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1460 result.append(buffer); 1461 ::write(fd, result.string(), result.size()); 1462 return NO_ERROR; 1463} 1464 1465// ========================================================================= 1466 1467AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1468 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1469{ 1470} 1471 1472AudioTrack::AudioTrackThread::~AudioTrackThread() 1473{ 1474} 1475 1476bool AudioTrack::AudioTrackThread::threadLoop() 1477{ 1478 { 1479 AutoMutex _l(mMyLock); 1480 if (mPaused) { 1481 mMyCond.wait(mMyLock); 1482 // caller will check for exitPending() 1483 return true; 1484 } 1485 } 1486 if (!mReceiver.processAudioBuffer(this)) { 1487 pause(); 1488 } 1489 return true; 1490} 1491 1492void AudioTrack::AudioTrackThread::requestExit() 1493{ 1494 // must be in this order to avoid a race condition 1495 Thread::requestExit(); 1496 resume(); 1497} 1498 1499void AudioTrack::AudioTrackThread::pause() 1500{ 1501 AutoMutex _l(mMyLock); 1502 mPaused = true; 1503} 1504 1505void AudioTrack::AudioTrackThread::resume() 1506{ 1507 AutoMutex _l(mMyLock); 1508 if (mPaused) { 1509 mPaused = false; 1510 mMyCond.signal(); 1511 } 1512} 1513 1514// ========================================================================= 1515 1516 1517audio_track_cblk_t::audio_track_cblk_t() 1518 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1519 userBase(0), serverBase(0), buffers(NULL), frameCount(0), 1520 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), 1521 mSendLevel(0), flags(0) 1522{ 1523} 1524 1525uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1526{ 1527 ALOGV("stepuser %08x %08x %d", user, server, frameCount); 1528 1529 uint32_t u = user; 1530 u += frameCount; 1531 // Ensure that user is never ahead of server for AudioRecord 1532 if (flags & CBLK_DIRECTION_MSK) { 1533 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1534 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1535 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1536 } 1537 } else if (u > server) { 1538 ALOGW("stepUser occurred after track reset"); 1539 u = server; 1540 } 1541 1542 uint32_t fc = this->frameCount; 1543 if (u >= fc) { 1544 // common case, user didn't just wrap 1545 if (u - fc >= userBase ) { 1546 userBase += fc; 1547 } 1548 } else if (u >= userBase + fc) { 1549 // user just wrapped 1550 userBase += fc; 1551 } 1552 1553 user = u; 1554 1555 // Clear flow control error condition as new data has been written/read to/from buffer. 1556 if (flags & CBLK_UNDERRUN_MSK) { 1557 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1558 } 1559 1560 return u; 1561} 1562 1563bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1564{ 1565 ALOGV("stepserver %08x %08x %d", user, server, frameCount); 1566 1567 if (!tryLock()) { 1568 ALOGW("stepServer() could not lock cblk"); 1569 return false; 1570 } 1571 1572 uint32_t s = server; 1573 bool flushed = (s == user); 1574 1575 s += frameCount; 1576 if (flags & CBLK_DIRECTION_MSK) { 1577 // Mark that we have read the first buffer so that next time stepUser() is called 1578 // we switch to normal obtainBuffer() timeout period 1579 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1580 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1581 } 1582 // It is possible that we receive a flush() 1583 // while the mixer is processing a block: in this case, 1584 // stepServer() is called After the flush() has reset u & s and 1585 // we have s > u 1586 if (flushed) { 1587 ALOGW("stepServer occurred after track reset"); 1588 s = user; 1589 } 1590 } 1591 1592 if (s >= loopEnd) { 1593 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1594 s = loopStart; 1595 if (--loopCount == 0) { 1596 loopEnd = UINT_MAX; 1597 loopStart = UINT_MAX; 1598 } 1599 } 1600 1601 uint32_t fc = this->frameCount; 1602 if (s >= fc) { 1603 // common case, server didn't just wrap 1604 if (s - fc >= serverBase ) { 1605 serverBase += fc; 1606 } 1607 } else if (s >= serverBase + fc) { 1608 // server just wrapped 1609 serverBase += fc; 1610 } 1611 1612 server = s; 1613 1614 if (!(flags & CBLK_INVALID_MSK)) { 1615 cv.signal(); 1616 } 1617 lock.unlock(); 1618 return true; 1619} 1620 1621void* audio_track_cblk_t::buffer(uint32_t offset) const 1622{ 1623 return (int8_t *)buffers + (offset - userBase) * frameSize; 1624} 1625 1626uint32_t audio_track_cblk_t::framesAvailable() 1627{ 1628 Mutex::Autolock _l(lock); 1629 return framesAvailable_l(); 1630} 1631 1632uint32_t audio_track_cblk_t::framesAvailable_l() 1633{ 1634 uint32_t u = user; 1635 uint32_t s = server; 1636 1637 if (flags & CBLK_DIRECTION_MSK) { 1638 uint32_t limit = (s < loopStart) ? s : loopStart; 1639 return limit + frameCount - u; 1640 } else { 1641 return frameCount + u - s; 1642 } 1643} 1644 1645uint32_t audio_track_cblk_t::framesReady() 1646{ 1647 uint32_t u = user; 1648 uint32_t s = server; 1649 1650 if (flags & CBLK_DIRECTION_MSK) { 1651 if (u < loopEnd) { 1652 return u - s; 1653 } else { 1654 // do not block on mutex shared with client on AudioFlinger side 1655 if (!tryLock()) { 1656 ALOGW("framesReady() could not lock cblk"); 1657 return 0; 1658 } 1659 uint32_t frames = UINT_MAX; 1660 if (loopCount >= 0) { 1661 frames = (loopEnd - loopStart)*loopCount + u - s; 1662 } 1663 lock.unlock(); 1664 return frames; 1665 } 1666 } else { 1667 return s - u; 1668 } 1669} 1670 1671bool audio_track_cblk_t::tryLock() 1672{ 1673 // the code below simulates lock-with-timeout 1674 // we MUST do this to protect the AudioFlinger server 1675 // as this lock is shared with the client. 1676 status_t err; 1677 1678 err = lock.tryLock(); 1679 if (err == -EBUSY) { // just wait a bit 1680 usleep(1000); 1681 err = lock.tryLock(); 1682 } 1683 if (err != NO_ERROR) { 1684 // probably, the client just died. 1685 return false; 1686 } 1687 return true; 1688} 1689 1690// ------------------------------------------------------------------------- 1691 1692}; // namespace android 1693