AudioTrack.cpp revision a8190fc518b6769257896605f3aee091aeb60b50
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <stdint.h> 23#include <sys/types.h> 24#include <limits.h> 25 26#include <sched.h> 27#include <sys/resource.h> 28 29#include <private/media/AudioTrackShared.h> 30 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Log.h> 35#include <binder/Parcel.h> 36#include <binder/IPCThreadState.h> 37#include <utils/Timers.h> 38#include <utils/Atomic.h> 39 40#include <cutils/bitops.h> 41#include <cutils/compiler.h> 42 43#include <system/audio.h> 44#include <system/audio_policy.h> 45 46#include <audio_utils/primitives.h> 47 48namespace android { 49// --------------------------------------------------------------------------- 50 51// static 52status_t AudioTrack::getMinFrameCount( 53 size_t* frameCount, 54 audio_stream_type_t streamType, 55 uint32_t sampleRate) 56{ 57 if (frameCount == NULL) { 58 return BAD_VALUE; 59 } 60 61 // default to 0 in case of error 62 *frameCount = 0; 63 64 // FIXME merge with similar code in createTrack_l(), except we're missing 65 // some information here that is available in createTrack_l(): 66 // audio_io_handle_t output 67 // audio_format_t format 68 // audio_channel_mask_t channelMask 69 // audio_output_flags_t flags 70 uint32_t afSampleRate; 71 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 72 return NO_INIT; 73 } 74 size_t afFrameCount; 75 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 76 return NO_INIT; 77 } 78 uint32_t afLatency; 79 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 80 return NO_INIT; 81 } 82 83 // Ensure that buffer depth covers at least audio hardware latency 84 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 85 if (minBufCount < 2) minBufCount = 2; 86 87 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 88 afFrameCount * minBufCount * sampleRate / afSampleRate; 89 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 90 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 91 return NO_ERROR; 92} 93 94// --------------------------------------------------------------------------- 95 96AudioTrack::AudioTrack() 97 : mStatus(NO_INIT), 98 mIsTimed(false), 99 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 100 mPreviousSchedulingGroup(SP_DEFAULT) 101{ 102} 103 104AudioTrack::AudioTrack( 105 audio_stream_type_t streamType, 106 uint32_t sampleRate, 107 audio_format_t format, 108 audio_channel_mask_t channelMask, 109 int frameCount, 110 audio_output_flags_t flags, 111 callback_t cbf, 112 void* user, 113 int notificationFrames, 114 int sessionId) 115 : mStatus(NO_INIT), 116 mIsTimed(false), 117 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 118 mPreviousSchedulingGroup(SP_DEFAULT) 119{ 120 mStatus = set(streamType, sampleRate, format, channelMask, 121 frameCount, flags, cbf, user, notificationFrames, 122 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); 123} 124 125AudioTrack::AudioTrack( 126 audio_stream_type_t streamType, 127 uint32_t sampleRate, 128 audio_format_t format, 129 audio_channel_mask_t channelMask, 130 const sp<IMemory>& sharedBuffer, 131 audio_output_flags_t flags, 132 callback_t cbf, 133 void* user, 134 int notificationFrames, 135 int sessionId) 136 : mStatus(NO_INIT), 137 mIsTimed(false), 138 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 139 mPreviousSchedulingGroup(SP_DEFAULT) 140{ 141 if (sharedBuffer == 0) { 142 ALOGE("sharedBuffer must be non-0"); 143 mStatus = BAD_VALUE; 144 return; 145 } 146 mStatus = set(streamType, sampleRate, format, channelMask, 147 0 /*frameCount*/, flags, cbf, user, notificationFrames, 148 sharedBuffer, false /*threadCanCallJava*/, sessionId); 149} 150 151AudioTrack::~AudioTrack() 152{ 153 ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 154 155 if (mStatus == NO_ERROR) { 156 // Make sure that callback function exits in the case where 157 // it is looping on buffer full condition in obtainBuffer(). 158 // Otherwise the callback thread will never exit. 159 stop(); 160 if (mAudioTrackThread != 0) { 161 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 162 mAudioTrackThread->requestExitAndWait(); 163 mAudioTrackThread.clear(); 164 } 165 mAudioTrack.clear(); 166 IPCThreadState::self()->flushCommands(); 167 AudioSystem::releaseAudioSessionId(mSessionId); 168 } 169} 170 171status_t AudioTrack::set( 172 audio_stream_type_t streamType, 173 uint32_t sampleRate, 174 audio_format_t format, 175 audio_channel_mask_t channelMask, 176 int frameCountInt, 177 audio_output_flags_t flags, 178 callback_t cbf, 179 void* user, 180 int notificationFrames, 181 const sp<IMemory>& sharedBuffer, 182 bool threadCanCallJava, 183 int sessionId) 184{ 185 // FIXME "int" here is legacy and will be replaced by size_t later 186 if (frameCountInt < 0) { 187 ALOGE("Invalid frame count %d", frameCountInt); 188 return BAD_VALUE; 189 } 190 size_t frameCount = frameCountInt; 191 192 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 193 sharedBuffer->size()); 194 195 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 196 197 AutoMutex lock(mLock); 198 if (mAudioTrack != 0) { 199 ALOGE("Track already in use"); 200 return INVALID_OPERATION; 201 } 202 203 // handle default values first. 204 if (streamType == AUDIO_STREAM_DEFAULT) { 205 streamType = AUDIO_STREAM_MUSIC; 206 } 207 208 if (sampleRate == 0) { 209 uint32_t afSampleRate; 210 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 211 return NO_INIT; 212 } 213 sampleRate = afSampleRate; 214 } 215 216 // these below should probably come from the audioFlinger too... 217 if (format == AUDIO_FORMAT_DEFAULT) { 218 format = AUDIO_FORMAT_PCM_16_BIT; 219 } 220 if (channelMask == 0) { 221 channelMask = AUDIO_CHANNEL_OUT_STEREO; 222 } 223 224 // validate parameters 225 if (!audio_is_valid_format(format)) { 226 ALOGE("Invalid format"); 227 return BAD_VALUE; 228 } 229 230 // AudioFlinger does not currently support 8-bit data in shared memory 231 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 232 ALOGE("8-bit data in shared memory is not supported"); 233 return BAD_VALUE; 234 } 235 236 // force direct flag if format is not linear PCM 237 if (!audio_is_linear_pcm(format)) { 238 flags = (audio_output_flags_t) 239 // FIXME why can't we allow direct AND fast? 240 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 241 } 242 // only allow deep buffering for music stream type 243 if (streamType != AUDIO_STREAM_MUSIC) { 244 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 245 } 246 247 if (!audio_is_output_channel(channelMask)) { 248 ALOGE("Invalid channel mask %#x", channelMask); 249 return BAD_VALUE; 250 } 251 mChannelMask = channelMask; 252 uint32_t channelCount = popcount(channelMask); 253 mChannelCount = channelCount; 254 255 audio_io_handle_t output = AudioSystem::getOutput( 256 streamType, 257 sampleRate, format, channelMask, 258 flags); 259 260 if (output == 0) { 261 ALOGE("Could not get audio output for stream type %d", streamType); 262 return BAD_VALUE; 263 } 264 265 mVolume[LEFT] = 1.0f; 266 mVolume[RIGHT] = 1.0f; 267 mSendLevel = 0.0f; 268 mFrameCount = frameCount; 269 mReqFrameCount = frameCount; 270 mNotificationFramesReq = notificationFrames; 271 mSessionId = sessionId; 272 mAuxEffectId = 0; 273 mFlags = flags; 274 mCbf = cbf; 275 276 if (cbf != NULL) { 277 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 278 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 279 } 280 281 // create the IAudioTrack 282 status_t status = createTrack_l(streamType, 283 sampleRate, 284 format, 285 frameCount, 286 flags, 287 sharedBuffer, 288 output); 289 290 if (status != NO_ERROR) { 291 if (mAudioTrackThread != 0) { 292 mAudioTrackThread->requestExit(); 293 mAudioTrackThread.clear(); 294 } 295 return status; 296 } 297 298 mStatus = NO_ERROR; 299 300 mStreamType = streamType; 301 mFormat = format; 302 303 if (audio_is_linear_pcm(format)) { 304 mFrameSize = channelCount * audio_bytes_per_sample(format); 305 mFrameSizeAF = channelCount * sizeof(int16_t); 306 } else { 307 mFrameSize = sizeof(uint8_t); 308 mFrameSizeAF = sizeof(uint8_t); 309 } 310 311 mSharedBuffer = sharedBuffer; 312 mActive = false; 313 mUserData = user; 314 mLoopCount = 0; 315 mMarkerPosition = 0; 316 mMarkerReached = false; 317 mNewPosition = 0; 318 mUpdatePeriod = 0; 319 mFlushed = false; 320 AudioSystem::acquireAudioSessionId(mSessionId); 321 return NO_ERROR; 322} 323 324// ------------------------------------------------------------------------- 325 326void AudioTrack::start() 327{ 328 sp<AudioTrackThread> t = mAudioTrackThread; 329 330 ALOGV("start %p", this); 331 332 AutoMutex lock(mLock); 333 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 334 // while we are accessing the cblk 335 sp<IAudioTrack> audioTrack = mAudioTrack; 336 sp<IMemory> iMem = mCblkMemory; 337 audio_track_cblk_t* cblk = mCblk; 338 339 if (!mActive) { 340 mFlushed = false; 341 mActive = true; 342 mNewPosition = cblk->server + mUpdatePeriod; 343 cblk->lock.lock(); 344 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 345 cblk->waitTimeMs = 0; 346 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 347 if (t != 0) { 348 t->resume(); 349 } else { 350 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 351 get_sched_policy(0, &mPreviousSchedulingGroup); 352 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 353 } 354 355 ALOGV("start %p before lock cblk %p", this, cblk); 356 status_t status = NO_ERROR; 357 if (!(cblk->flags & CBLK_INVALID)) { 358 cblk->lock.unlock(); 359 ALOGV("mAudioTrack->start()"); 360 status = mAudioTrack->start(); 361 cblk->lock.lock(); 362 if (status == DEAD_OBJECT) { 363 android_atomic_or(CBLK_INVALID, &cblk->flags); 364 } 365 } 366 if (cblk->flags & CBLK_INVALID) { 367 audio_track_cblk_t* temp = cblk; 368 status = restoreTrack_l(temp, true /*fromStart*/); 369 cblk = temp; 370 } 371 cblk->lock.unlock(); 372 if (status != NO_ERROR) { 373 ALOGV("start() failed"); 374 mActive = false; 375 if (t != 0) { 376 t->pause(); 377 } else { 378 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 379 set_sched_policy(0, mPreviousSchedulingGroup); 380 } 381 } 382 } 383 384} 385 386void AudioTrack::stop() 387{ 388 sp<AudioTrackThread> t = mAudioTrackThread; 389 390 ALOGV("stop %p", this); 391 392 AutoMutex lock(mLock); 393 if (mActive) { 394 mActive = false; 395 mCblk->cv.signal(); 396 mAudioTrack->stop(); 397 // Cancel loops (If we are in the middle of a loop, playback 398 // would not stop until loopCount reaches 0). 399 setLoop_l(0, 0, 0); 400 // the playback head position will reset to 0, so if a marker is set, we need 401 // to activate it again 402 mMarkerReached = false; 403 // Force flush if a shared buffer is used otherwise audioflinger 404 // will not stop before end of buffer is reached. 405 // It may be needed to make sure that we stop playback, likely in case looping is on. 406 if (mSharedBuffer != 0) { 407 flush_l(); 408 } 409 if (t != 0) { 410 t->pause(); 411 } else { 412 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 413 set_sched_policy(0, mPreviousSchedulingGroup); 414 } 415 } 416 417} 418 419bool AudioTrack::stopped() const 420{ 421 AutoMutex lock(mLock); 422 return stopped_l(); 423} 424 425void AudioTrack::flush() 426{ 427 AutoMutex lock(mLock); 428 if (!mActive && mSharedBuffer == 0) { 429 flush_l(); 430 } 431} 432 433void AudioTrack::flush_l() 434{ 435 ALOGV("flush"); 436 ALOG_ASSERT(!mActive); 437 438 // clear playback marker and periodic update counter 439 mMarkerPosition = 0; 440 mMarkerReached = false; 441 mUpdatePeriod = 0; 442 443 mFlushed = true; 444 mAudioTrack->flush(); 445 // Release AudioTrack callback thread in case it was waiting for new buffers 446 // in AudioTrack::obtainBuffer() 447 mCblk->cv.signal(); 448} 449 450void AudioTrack::pause() 451{ 452 ALOGV("pause"); 453 AutoMutex lock(mLock); 454 if (mActive) { 455 mActive = false; 456 mCblk->cv.signal(); 457 mAudioTrack->pause(); 458 } 459} 460 461status_t AudioTrack::setVolume(float left, float right) 462{ 463 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 464 return BAD_VALUE; 465 } 466 467 AutoMutex lock(mLock); 468 mVolume[LEFT] = left; 469 mVolume[RIGHT] = right; 470 471 mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 472 473 return NO_ERROR; 474} 475 476status_t AudioTrack::setVolume(float volume) 477{ 478 return setVolume(volume, volume); 479} 480 481status_t AudioTrack::setAuxEffectSendLevel(float level) 482{ 483 ALOGV("setAuxEffectSendLevel(%f)", level); 484 if (level < 0.0f || level > 1.0f) { 485 return BAD_VALUE; 486 } 487 AutoMutex lock(mLock); 488 489 mSendLevel = level; 490 491 mCblk->setSendLevel(level); 492 493 return NO_ERROR; 494} 495 496void AudioTrack::getAuxEffectSendLevel(float* level) const 497{ 498 if (level != NULL) { 499 *level = mSendLevel; 500 } 501} 502 503status_t AudioTrack::setSampleRate(uint32_t rate) 504{ 505 uint32_t afSamplingRate; 506 507 if (mIsTimed) { 508 return INVALID_OPERATION; 509 } 510 511 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 512 return NO_INIT; 513 } 514 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 515 if (rate == 0 || rate > afSamplingRate*2 ) { 516 return BAD_VALUE; 517 } 518 519 AutoMutex lock(mLock); 520 mCblk->sampleRate = rate; 521 return NO_ERROR; 522} 523 524uint32_t AudioTrack::getSampleRate() const 525{ 526 if (mIsTimed) { 527 return 0; 528 } 529 530 AutoMutex lock(mLock); 531 return mCblk->sampleRate; 532} 533 534status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 535{ 536 AutoMutex lock(mLock); 537 return setLoop_l(loopStart, loopEnd, loopCount); 538} 539 540// must be called with mLock held 541status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 542{ 543 if (mSharedBuffer == 0 || mIsTimed) { 544 return INVALID_OPERATION; 545 } 546 547 audio_track_cblk_t* cblk = mCblk; 548 549 Mutex::Autolock _l(cblk->lock); 550 551 if (loopCount == 0) { 552 cblk->loopStart = UINT_MAX; 553 cblk->loopEnd = UINT_MAX; 554 cblk->loopCount = 0; 555 mLoopCount = 0; 556 return NO_ERROR; 557 } 558 559 if (loopStart >= loopEnd || 560 loopEnd - loopStart > mFrameCount || 561 cblk->server > loopStart) { 562 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, " 563 "user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user); 564 return BAD_VALUE; 565 } 566 567 if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) { 568 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " 569 "framecount %d", 570 loopStart, loopEnd, mFrameCount); 571 return BAD_VALUE; 572 } 573 574 cblk->loopStart = loopStart; 575 cblk->loopEnd = loopEnd; 576 cblk->loopCount = loopCount; 577 mLoopCount = loopCount; 578 579 return NO_ERROR; 580} 581 582status_t AudioTrack::setMarkerPosition(uint32_t marker) 583{ 584 if (mCbf == NULL) { 585 return INVALID_OPERATION; 586 } 587 588 mMarkerPosition = marker; 589 mMarkerReached = false; 590 591 return NO_ERROR; 592} 593 594status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 595{ 596 if (marker == NULL) { 597 return BAD_VALUE; 598 } 599 600 *marker = mMarkerPosition; 601 602 return NO_ERROR; 603} 604 605status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 606{ 607 if (mCbf == NULL) { 608 return INVALID_OPERATION; 609 } 610 611 uint32_t curPosition; 612 getPosition(&curPosition); 613 mNewPosition = curPosition + updatePeriod; 614 mUpdatePeriod = updatePeriod; 615 616 return NO_ERROR; 617} 618 619status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 620{ 621 if (updatePeriod == NULL) { 622 return BAD_VALUE; 623 } 624 625 *updatePeriod = mUpdatePeriod; 626 627 return NO_ERROR; 628} 629 630status_t AudioTrack::setPosition(uint32_t position) 631{ 632 if (mSharedBuffer == 0 || mIsTimed) { 633 return INVALID_OPERATION; 634 } 635 636 AutoMutex lock(mLock); 637 638 if (!stopped_l()) { 639 return INVALID_OPERATION; 640 } 641 642 audio_track_cblk_t* cblk = mCblk; 643 Mutex::Autolock _l(cblk->lock); 644 645 if (position > cblk->user) { 646 return BAD_VALUE; 647 } 648 649 cblk->server = position; 650 android_atomic_or(CBLK_FORCEREADY, &cblk->flags); 651 652 return NO_ERROR; 653} 654 655status_t AudioTrack::getPosition(uint32_t *position) 656{ 657 if (position == NULL) { 658 return BAD_VALUE; 659 } 660 AutoMutex lock(mLock); 661 *position = mFlushed ? 0 : mCblk->server; 662 663 return NO_ERROR; 664} 665 666status_t AudioTrack::reload() 667{ 668 if (mSharedBuffer == 0 || mIsTimed) { 669 return INVALID_OPERATION; 670 } 671 672 AutoMutex lock(mLock); 673 674 if (!stopped_l()) { 675 return INVALID_OPERATION; 676 } 677 678 flush_l(); 679 680 audio_track_cblk_t* cblk = mCblk; 681 cblk->stepUserOut(mFrameCount, mFrameCount); 682 683 return NO_ERROR; 684} 685 686audio_io_handle_t AudioTrack::getOutput() 687{ 688 AutoMutex lock(mLock); 689 return getOutput_l(); 690} 691 692// must be called with mLock held 693audio_io_handle_t AudioTrack::getOutput_l() 694{ 695 return AudioSystem::getOutput(mStreamType, 696 mCblk->sampleRate, mFormat, mChannelMask, mFlags); 697} 698 699status_t AudioTrack::attachAuxEffect(int effectId) 700{ 701 ALOGV("attachAuxEffect(%d)", effectId); 702 status_t status = mAudioTrack->attachAuxEffect(effectId); 703 if (status == NO_ERROR) { 704 mAuxEffectId = effectId; 705 } 706 return status; 707} 708 709// ------------------------------------------------------------------------- 710 711// must be called with mLock held 712status_t AudioTrack::createTrack_l( 713 audio_stream_type_t streamType, 714 uint32_t sampleRate, 715 audio_format_t format, 716 size_t frameCount, 717 audio_output_flags_t flags, 718 const sp<IMemory>& sharedBuffer, 719 audio_io_handle_t output) 720{ 721 status_t status; 722 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 723 if (audioFlinger == 0) { 724 ALOGE("Could not get audioflinger"); 725 return NO_INIT; 726 } 727 728 uint32_t afLatency; 729 if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { 730 return NO_INIT; 731 } 732 733 // Client decides whether the track is TIMED (see below), but can only express a preference 734 // for FAST. Server will perform additional tests. 735 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 736 // either of these use cases: 737 // use case 1: shared buffer 738 (sharedBuffer != 0) || 739 // use case 2: callback handler 740 (mCbf != NULL))) { 741 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 742 // once denied, do not request again if IAudioTrack is re-created 743 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 744 mFlags = flags; 745 } 746 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 747 748 mNotificationFramesAct = mNotificationFramesReq; 749 750 if (!audio_is_linear_pcm(format)) { 751 752 if (sharedBuffer != 0) { 753 // Same comment as below about ignoring frameCount parameter for set() 754 frameCount = sharedBuffer->size(); 755 } else if (frameCount == 0) { 756 size_t afFrameCount; 757 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 758 return NO_INIT; 759 } 760 frameCount = afFrameCount; 761 } 762 763 } else if (sharedBuffer != 0) { 764 765 // Ensure that buffer alignment matches channel count 766 // 8-bit data in shared memory is not currently supported by AudioFlinger 767 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 768 if (mChannelCount > 1) { 769 // More than 2 channels does not require stronger alignment than stereo 770 alignment <<= 1; 771 } 772 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 773 ALOGE("Invalid buffer alignment: address %p, channel count %u", 774 sharedBuffer->pointer(), mChannelCount); 775 return BAD_VALUE; 776 } 777 778 // When initializing a shared buffer AudioTrack via constructors, 779 // there's no frameCount parameter. 780 // But when initializing a shared buffer AudioTrack via set(), 781 // there _is_ a frameCount parameter. We silently ignore it. 782 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 783 784 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 785 786 // FIXME move these calculations and associated checks to server 787 uint32_t afSampleRate; 788 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 789 return NO_INIT; 790 } 791 size_t afFrameCount; 792 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 793 return NO_INIT; 794 } 795 796 // Ensure that buffer depth covers at least audio hardware latency 797 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 798 if (minBufCount < 2) minBufCount = 2; 799 800 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 801 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 802 ", afLatency=%d", 803 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 804 805 if (frameCount == 0) { 806 frameCount = minFrameCount; 807 } 808 if (mNotificationFramesAct == 0) { 809 mNotificationFramesAct = frameCount/2; 810 } 811 // Make sure that application is notified with sufficient margin 812 // before underrun 813 if (mNotificationFramesAct > frameCount/2) { 814 mNotificationFramesAct = frameCount/2; 815 } 816 if (frameCount < minFrameCount) { 817 // not ALOGW because it happens all the time when playing key clicks over A2DP 818 ALOGV("Minimum buffer size corrected from %d to %d", 819 frameCount, minFrameCount); 820 frameCount = minFrameCount; 821 } 822 823 } else { 824 // For fast tracks, the frame count calculations and checks are done by server 825 } 826 827 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 828 if (mIsTimed) { 829 trackFlags |= IAudioFlinger::TRACK_TIMED; 830 } 831 832 pid_t tid = -1; 833 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 834 trackFlags |= IAudioFlinger::TRACK_FAST; 835 if (mAudioTrackThread != 0) { 836 tid = mAudioTrackThread->getTid(); 837 } 838 } 839 840 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 841 streamType, 842 sampleRate, 843 // AudioFlinger only sees 16-bit PCM 844 format == AUDIO_FORMAT_PCM_8_BIT ? 845 AUDIO_FORMAT_PCM_16_BIT : format, 846 mChannelMask, 847 frameCount, 848 &trackFlags, 849 sharedBuffer, 850 output, 851 tid, 852 &mSessionId, 853 &status); 854 855 if (track == 0) { 856 ALOGE("AudioFlinger could not create track, status: %d", status); 857 return status; 858 } 859 sp<IMemory> iMem = track->getCblk(); 860 if (iMem == 0) { 861 ALOGE("Could not get control block"); 862 return NO_INIT; 863 } 864 mAudioTrack = track; 865 mCblkMemory = iMem; 866 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 867 mCblk = cblk; 868 size_t temp = cblk->frameCount_; 869 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 870 // In current design, AudioTrack client checks and ensures frame count validity before 871 // passing it to AudioFlinger so AudioFlinger should not return a different value except 872 // for fast track as it uses a special method of assigning frame count. 873 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 874 } 875 frameCount = temp; 876 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 877 if (trackFlags & IAudioFlinger::TRACK_FAST) { 878 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 879 } else { 880 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 881 // once denied, do not request again if IAudioTrack is re-created 882 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 883 mFlags = flags; 884 } 885 if (sharedBuffer == 0) { 886 mNotificationFramesAct = frameCount/2; 887 } 888 } 889 if (sharedBuffer == 0) { 890 mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); 891 } else { 892 mBuffers = sharedBuffer->pointer(); 893 // Force buffer full condition as data is already present in shared memory 894 cblk->stepUserOut(frameCount, frameCount); 895 } 896 897 cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 898 uint16_t(mVolume[LEFT] * 0x1000)); 899 cblk->setSendLevel(mSendLevel); 900 mAudioTrack->attachAuxEffect(mAuxEffectId); 901 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 902 cblk->waitTimeMs = 0; 903 mRemainingFrames = mNotificationFramesAct; 904 // FIXME don't believe this lie 905 mLatency = afLatency + (1000*frameCount) / sampleRate; 906 mFrameCount = frameCount; 907 // If IAudioTrack is re-created, don't let the requested frameCount 908 // decrease. This can confuse clients that cache frameCount(). 909 if (frameCount > mReqFrameCount) { 910 mReqFrameCount = frameCount; 911 } 912 return NO_ERROR; 913} 914 915status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 916{ 917 AutoMutex lock(mLock); 918 bool active; 919 status_t result = NO_ERROR; 920 audio_track_cblk_t* cblk = mCblk; 921 uint32_t framesReq = audioBuffer->frameCount; 922 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 923 924 audioBuffer->frameCount = 0; 925 audioBuffer->size = 0; 926 927 uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); 928 929 cblk->lock.lock(); 930 if (cblk->flags & CBLK_INVALID) { 931 goto create_new_track; 932 } 933 cblk->lock.unlock(); 934 935 if (framesAvail == 0) { 936 cblk->lock.lock(); 937 goto start_loop_here; 938 while (framesAvail == 0) { 939 active = mActive; 940 if (CC_UNLIKELY(!active)) { 941 ALOGV("Not active and NO_MORE_BUFFERS"); 942 cblk->lock.unlock(); 943 return NO_MORE_BUFFERS; 944 } 945 if (CC_UNLIKELY(!waitCount)) { 946 cblk->lock.unlock(); 947 return WOULD_BLOCK; 948 } 949 if (!(cblk->flags & CBLK_INVALID)) { 950 mLock.unlock(); 951 // this condition is in shared memory, so if IAudioTrack and control block 952 // are replaced due to mediaserver death or IAudioTrack invalidation then 953 // cv won't be signalled, but fortunately the timeout will limit the wait 954 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 955 cblk->lock.unlock(); 956 mLock.lock(); 957 if (!mActive) { 958 return status_t(STOPPED); 959 } 960 // IAudioTrack may have been re-created while mLock was unlocked 961 cblk = mCblk; 962 cblk->lock.lock(); 963 } 964 965 if (cblk->flags & CBLK_INVALID) { 966 goto create_new_track; 967 } 968 if (CC_UNLIKELY(result != NO_ERROR)) { 969 cblk->waitTimeMs += waitTimeMs; 970 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 971 // timing out when a loop has been set and we have already written upto loop end 972 // is a normal condition: no need to wake AudioFlinger up. 973 if (cblk->user < cblk->loopEnd) { 974 ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " 975 "server=%08x", this, cblk->mName, cblk->user, cblk->server); 976 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 977 cblk->lock.unlock(); 978 result = mAudioTrack->start(); 979 cblk->lock.lock(); 980 if (result == DEAD_OBJECT) { 981 android_atomic_or(CBLK_INVALID, &cblk->flags); 982create_new_track: 983 audio_track_cblk_t* temp = cblk; 984 result = restoreTrack_l(temp, false /*fromStart*/); 985 cblk = temp; 986 } 987 if (result != NO_ERROR) { 988 ALOGW("obtainBuffer create Track error %d", result); 989 cblk->lock.unlock(); 990 return result; 991 } 992 } 993 cblk->waitTimeMs = 0; 994 } 995 996 if (--waitCount == 0) { 997 cblk->lock.unlock(); 998 return TIMED_OUT; 999 } 1000 } 1001 // read the server count again 1002 start_loop_here: 1003 framesAvail = cblk->framesAvailableOut_l(mFrameCount); 1004 } 1005 cblk->lock.unlock(); 1006 } 1007 1008 cblk->waitTimeMs = 0; 1009 1010 if (framesReq > framesAvail) { 1011 framesReq = framesAvail; 1012 } 1013 1014 uint32_t u = cblk->user; 1015 uint32_t bufferEnd = cblk->userBase + mFrameCount; 1016 1017 if (framesReq > bufferEnd - u) { 1018 framesReq = bufferEnd - u; 1019 } 1020 1021 audioBuffer->frameCount = framesReq; 1022 audioBuffer->size = framesReq * mFrameSizeAF; 1023 audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u); 1024 active = mActive; 1025 return active ? status_t(NO_ERROR) : status_t(STOPPED); 1026} 1027 1028void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1029{ 1030 AutoMutex lock(mLock); 1031 audio_track_cblk_t* cblk = mCblk; 1032 cblk->stepUserOut(audioBuffer->frameCount, mFrameCount); 1033 if (audioBuffer->frameCount > 0) { 1034 // restart track if it was disabled by audioflinger due to previous underrun 1035 if (mActive && (cblk->flags & CBLK_DISABLED)) { 1036 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1037 ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName); 1038 mAudioTrack->start(); 1039 } 1040 } 1041} 1042 1043// ------------------------------------------------------------------------- 1044 1045ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1046{ 1047 1048 if (mSharedBuffer != 0 || mIsTimed) { 1049 return INVALID_OPERATION; 1050 } 1051 1052 if (ssize_t(userSize) < 0) { 1053 // Sanity-check: user is most-likely passing an error code, and it would 1054 // make the return value ambiguous (actualSize vs error). 1055 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1056 buffer, userSize, userSize); 1057 return BAD_VALUE; 1058 } 1059 1060 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1061 1062 if (userSize == 0) { 1063 return 0; 1064 } 1065 1066 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1067 // while we are accessing the cblk 1068 mLock.lock(); 1069 sp<IAudioTrack> audioTrack = mAudioTrack; 1070 sp<IMemory> iMem = mCblkMemory; 1071 mLock.unlock(); 1072 1073 // since mLock is unlocked the IAudioTrack and shared memory may be re-created, 1074 // so all cblk references might still refer to old shared memory, but that should be benign 1075 1076 ssize_t written = 0; 1077 const int8_t *src = (const int8_t *)buffer; 1078 Buffer audioBuffer; 1079 size_t frameSz = frameSize(); 1080 1081 do { 1082 audioBuffer.frameCount = userSize/frameSz; 1083 1084 status_t err = obtainBuffer(&audioBuffer, -1); 1085 if (err < 0) { 1086 // out of buffers, return #bytes written 1087 if (err == status_t(NO_MORE_BUFFERS)) { 1088 break; 1089 } 1090 return ssize_t(err); 1091 } 1092 1093 size_t toWrite; 1094 1095 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1096 // Divide capacity by 2 to take expansion into account 1097 toWrite = audioBuffer.size>>1; 1098 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1099 } else { 1100 toWrite = audioBuffer.size; 1101 memcpy(audioBuffer.i8, src, toWrite); 1102 } 1103 src += toWrite; 1104 userSize -= toWrite; 1105 written += toWrite; 1106 1107 releaseBuffer(&audioBuffer); 1108 } while (userSize >= frameSz); 1109 1110 return written; 1111} 1112 1113// ------------------------------------------------------------------------- 1114 1115TimedAudioTrack::TimedAudioTrack() { 1116 mIsTimed = true; 1117} 1118 1119status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1120{ 1121 AutoMutex lock(mLock); 1122 status_t result = UNKNOWN_ERROR; 1123 1124 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1125 // while we are accessing the cblk 1126 sp<IAudioTrack> audioTrack = mAudioTrack; 1127 sp<IMemory> iMem = mCblkMemory; 1128 1129 // If the track is not invalid already, try to allocate a buffer. alloc 1130 // fails indicating that the server is dead, flag the track as invalid so 1131 // we can attempt to restore in just a bit. 1132 audio_track_cblk_t* cblk = mCblk; 1133 if (!(cblk->flags & CBLK_INVALID)) { 1134 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1135 if (result == DEAD_OBJECT) { 1136 android_atomic_or(CBLK_INVALID, &cblk->flags); 1137 } 1138 } 1139 1140 // If the track is invalid at this point, attempt to restore it. and try the 1141 // allocation one more time. 1142 if (cblk->flags & CBLK_INVALID) { 1143 cblk->lock.lock(); 1144 audio_track_cblk_t* temp = cblk; 1145 result = restoreTrack_l(temp, false /*fromStart*/); 1146 cblk = temp; 1147 cblk->lock.unlock(); 1148 1149 if (result == OK) { 1150 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1151 } 1152 } 1153 1154 return result; 1155} 1156 1157status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1158 int64_t pts) 1159{ 1160 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1161 { 1162 AutoMutex lock(mLock); 1163 audio_track_cblk_t* cblk = mCblk; 1164 // restart track if it was disabled by audioflinger due to previous underrun 1165 if (buffer->size() != 0 && status == NO_ERROR && 1166 mActive && (cblk->flags & CBLK_DISABLED)) { 1167 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1168 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1169 mAudioTrack->start(); 1170 } 1171 } 1172 return status; 1173} 1174 1175status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1176 TargetTimeline target) 1177{ 1178 return mAudioTrack->setMediaTimeTransform(xform, target); 1179} 1180 1181// ------------------------------------------------------------------------- 1182 1183bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1184{ 1185 Buffer audioBuffer; 1186 uint32_t frames; 1187 size_t writtenSize; 1188 1189 mLock.lock(); 1190 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1191 // while we are accessing the cblk 1192 sp<IAudioTrack> audioTrack = mAudioTrack; 1193 sp<IMemory> iMem = mCblkMemory; 1194 audio_track_cblk_t* cblk = mCblk; 1195 bool active = mActive; 1196 mLock.unlock(); 1197 1198 // since mLock is unlocked the IAudioTrack and shared memory may be re-created, 1199 // so all cblk references might still refer to old shared memory, but that should be benign 1200 1201 // Manage underrun callback 1202 if (active && (cblk->framesAvailableOut(mFrameCount) == mFrameCount)) { 1203 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1204 if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { 1205 mCbf(EVENT_UNDERRUN, mUserData, 0); 1206 if (cblk->server == mFrameCount) { 1207 mCbf(EVENT_BUFFER_END, mUserData, 0); 1208 } 1209 if (mSharedBuffer != 0) { 1210 return false; 1211 } 1212 } 1213 } 1214 1215 // Manage loop end callback 1216 while (mLoopCount > cblk->loopCount) { 1217 int loopCount = -1; 1218 mLoopCount--; 1219 if (mLoopCount >= 0) loopCount = mLoopCount; 1220 1221 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1222 } 1223 1224 // Manage marker callback 1225 if (!mMarkerReached && (mMarkerPosition > 0)) { 1226 if (cblk->server >= mMarkerPosition) { 1227 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1228 mMarkerReached = true; 1229 } 1230 } 1231 1232 // Manage new position callback 1233 if (mUpdatePeriod > 0) { 1234 while (cblk->server >= mNewPosition) { 1235 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1236 mNewPosition += mUpdatePeriod; 1237 } 1238 } 1239 1240 // If Shared buffer is used, no data is requested from client. 1241 if (mSharedBuffer != 0) { 1242 frames = 0; 1243 } else { 1244 frames = mRemainingFrames; 1245 } 1246 1247 // See description of waitCount parameter at declaration of obtainBuffer(). 1248 // The logic below prevents us from being stuck below at obtainBuffer() 1249 // not being able to handle timed events (position, markers, loops). 1250 int32_t waitCount = -1; 1251 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1252 waitCount = 1; 1253 } 1254 1255 do { 1256 1257 audioBuffer.frameCount = frames; 1258 1259 status_t err = obtainBuffer(&audioBuffer, waitCount); 1260 if (err < NO_ERROR) { 1261 if (err != TIMED_OUT) { 1262 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), 1263 "Error obtaining an audio buffer, giving up."); 1264 return false; 1265 } 1266 break; 1267 } 1268 if (err == status_t(STOPPED)) { 1269 return false; 1270 } 1271 1272 // Divide buffer size by 2 to take into account the expansion 1273 // due to 8 to 16 bit conversion: the callback must fill only half 1274 // of the destination buffer 1275 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1276 audioBuffer.size >>= 1; 1277 } 1278 1279 size_t reqSize = audioBuffer.size; 1280 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1281 writtenSize = audioBuffer.size; 1282 1283 // Sanity check on returned size 1284 if (ssize_t(writtenSize) <= 0) { 1285 // The callback is done filling buffers 1286 // Keep this thread going to handle timed events and 1287 // still try to get more data in intervals of WAIT_PERIOD_MS 1288 // but don't just loop and block the CPU, so wait 1289 usleep(WAIT_PERIOD_MS*1000); 1290 break; 1291 } 1292 1293 if (writtenSize > reqSize) { 1294 writtenSize = reqSize; 1295 } 1296 1297 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1298 // 8 to 16 bit conversion, note that source and destination are the same address 1299 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1300 writtenSize <<= 1; 1301 } 1302 1303 audioBuffer.size = writtenSize; 1304 // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for 1305 // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of 1306 // 16 bit. 1307 audioBuffer.frameCount = writtenSize / mFrameSizeAF; 1308 1309 frames -= audioBuffer.frameCount; 1310 1311 releaseBuffer(&audioBuffer); 1312 } 1313 while (frames); 1314 1315 if (frames == 0) { 1316 mRemainingFrames = mNotificationFramesAct; 1317 } else { 1318 mRemainingFrames = frames; 1319 } 1320 return true; 1321} 1322 1323// must be called with mLock and refCblk.lock held. Callers must also hold strong references on 1324// the IAudioTrack and IMemory in case they are recreated here. 1325// If the IAudioTrack is successfully restored, the refCblk pointer is updated 1326// FIXME Don't depend on caller to hold strong references. 1327status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart) 1328{ 1329 status_t result; 1330 1331 audio_track_cblk_t* cblk = refCblk; 1332 audio_track_cblk_t* newCblk = cblk; 1333 ALOGW("dead IAudioTrack, creating a new one from %s", 1334 fromStart ? "start()" : "obtainBuffer()"); 1335 1336 // signal old cblk condition so that other threads waiting for available buffers stop 1337 // waiting now 1338 cblk->cv.broadcast(); 1339 cblk->lock.unlock(); 1340 1341 // refresh the audio configuration cache in this process to make sure we get new 1342 // output parameters in getOutput_l() and createTrack_l() 1343 AudioSystem::clearAudioConfigCache(); 1344 1345 // if the new IAudioTrack is created, createTrack_l() will modify the 1346 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1347 // It will also delete the strong references on previous IAudioTrack and IMemory 1348 result = createTrack_l(mStreamType, 1349 cblk->sampleRate, 1350 mFormat, 1351 mReqFrameCount, // so that frame count never goes down 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 newCblk = mCblk; 1361 newCblk->user = user; 1362 newCblk->server = user; 1363 newCblk->userBase = user; 1364 newCblk->serverBase = user; 1365 // restore loop: this is not guaranteed to succeed if new frame count is not 1366 // compatible with loop length 1367 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1368 if (!fromStart) { 1369 newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1370 // Make sure that a client relying on callback events indicating underrun or 1371 // the actual amount of audio frames played (e.g SoundPool) receives them. 1372 if (mSharedBuffer == 0) { 1373 uint32_t frames = 0; 1374 if (user > server) { 1375 frames = ((user - server) > mFrameCount) ? 1376 mFrameCount : (user - server); 1377 memset(mBuffers, 0, frames * mFrameSizeAF); 1378 } 1379 // restart playback even if buffer is not completely filled. 1380 android_atomic_or(CBLK_FORCEREADY, &newCblk->flags); 1381 // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to 1382 // the client 1383 newCblk->stepUserOut(frames, mFrameCount); 1384 } 1385 } 1386 if (mSharedBuffer != 0) { 1387 newCblk->stepUserOut(mFrameCount, mFrameCount); 1388 } 1389 if (mActive) { 1390 result = mAudioTrack->start(); 1391 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1392 } 1393 if (fromStart && result == NO_ERROR) { 1394 mNewPosition = newCblk->server + mUpdatePeriod; 1395 } 1396 } 1397 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1398 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1399 result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); 1400 1401 if (result == NO_ERROR) { 1402 // from now on we switch to the newly created cblk 1403 refCblk = newCblk; 1404 } 1405 newCblk->lock.lock(); 1406 1407 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result); 1408 1409 return result; 1410} 1411 1412status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1413{ 1414 1415 const size_t SIZE = 256; 1416 char buffer[SIZE]; 1417 String8 result; 1418 1419 audio_track_cblk_t* cblk = mCblk; 1420 result.append(" AudioTrack::dump\n"); 1421 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1422 mVolume[0], mVolume[1]); 1423 result.append(buffer); 1424 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1425 mChannelCount, mFrameCount); 1426 result.append(buffer); 1427 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", 1428 (cblk == 0) ? 0 : cblk->sampleRate, mStatus); 1429 result.append(buffer); 1430 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1431 result.append(buffer); 1432 ::write(fd, result.string(), result.size()); 1433 return NO_ERROR; 1434} 1435 1436// ========================================================================= 1437 1438AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1439 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1440{ 1441} 1442 1443AudioTrack::AudioTrackThread::~AudioTrackThread() 1444{ 1445} 1446 1447bool AudioTrack::AudioTrackThread::threadLoop() 1448{ 1449 { 1450 AutoMutex _l(mMyLock); 1451 if (mPaused) { 1452 mMyCond.wait(mMyLock); 1453 // caller will check for exitPending() 1454 return true; 1455 } 1456 } 1457 if (!mReceiver.processAudioBuffer(this)) { 1458 pause(); 1459 } 1460 return true; 1461} 1462 1463void AudioTrack::AudioTrackThread::requestExit() 1464{ 1465 // must be in this order to avoid a race condition 1466 Thread::requestExit(); 1467 resume(); 1468} 1469 1470void AudioTrack::AudioTrackThread::pause() 1471{ 1472 AutoMutex _l(mMyLock); 1473 mPaused = true; 1474} 1475 1476void AudioTrack::AudioTrackThread::resume() 1477{ 1478 AutoMutex _l(mMyLock); 1479 if (mPaused) { 1480 mPaused = false; 1481 mMyCond.signal(); 1482 } 1483} 1484 1485}; // namespace android 1486