AudioTrack.cpp revision e33054eb968cbf8ccaee1b0ff0301403902deed6
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) 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 uint32_t afSampleRate; 69 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 70 return NO_INIT; 71 } 72 size_t 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 123AudioTrack::AudioTrack( 124 audio_stream_type_t streamType, 125 uint32_t sampleRate, 126 audio_format_t format, 127 audio_channel_mask_t channelMask, 128 const sp<IMemory>& sharedBuffer, 129 audio_output_flags_t flags, 130 callback_t cbf, 131 void* user, 132 int notificationFrames, 133 int sessionId) 134 : mStatus(NO_INIT), 135 mIsTimed(false), 136 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 137 mPreviousSchedulingGroup(SP_DEFAULT) 138{ 139 mStatus = set(streamType, sampleRate, format, channelMask, 140 0 /*frameCount*/, flags, cbf, user, notificationFrames, 141 sharedBuffer, false /*threadCanCallJava*/, sessionId); 142} 143 144AudioTrack::~AudioTrack() 145{ 146 ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 147 148 if (mStatus == NO_ERROR) { 149 // Make sure that callback function exits in the case where 150 // it is looping on buffer full condition in obtainBuffer(). 151 // Otherwise the callback thread will never exit. 152 stop(); 153 if (mAudioTrackThread != 0) { 154 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 155 mAudioTrackThread->requestExitAndWait(); 156 mAudioTrackThread.clear(); 157 } 158 mAudioTrack.clear(); 159 IPCThreadState::self()->flushCommands(); 160 AudioSystem::releaseAudioSessionId(mSessionId); 161 } 162} 163 164status_t AudioTrack::set( 165 audio_stream_type_t streamType, 166 uint32_t sampleRate, 167 audio_format_t format, 168 audio_channel_mask_t channelMask, 169 int frameCountInt, 170 audio_output_flags_t flags, 171 callback_t cbf, 172 void* user, 173 int notificationFrames, 174 const sp<IMemory>& sharedBuffer, 175 bool threadCanCallJava, 176 int sessionId) 177{ 178 // FIXME "int" here is legacy and will be replaced by size_t later 179 if (frameCountInt < 0) { 180 ALOGE("Invalid frame count %d", frameCountInt); 181 return BAD_VALUE; 182 } 183 size_t frameCount = frameCountInt; 184 185 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 186 sharedBuffer->size()); 187 188 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 189 190 AutoMutex lock(mLock); 191 if (mAudioTrack != 0) { 192 ALOGE("Track already in use"); 193 return INVALID_OPERATION; 194 } 195 196 // handle default values first. 197 if (streamType == AUDIO_STREAM_DEFAULT) { 198 streamType = AUDIO_STREAM_MUSIC; 199 } 200 201 if (sampleRate == 0) { 202 uint32_t afSampleRate; 203 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 204 return NO_INIT; 205 } 206 sampleRate = afSampleRate; 207 } 208 209 // these below should probably come from the audioFlinger too... 210 if (format == AUDIO_FORMAT_DEFAULT) { 211 format = AUDIO_FORMAT_PCM_16_BIT; 212 } 213 if (channelMask == 0) { 214 channelMask = AUDIO_CHANNEL_OUT_STEREO; 215 } 216 217 // validate parameters 218 if (!audio_is_valid_format(format)) { 219 ALOGE("Invalid format"); 220 return BAD_VALUE; 221 } 222 223 // AudioFlinger does not currently support 8-bit data in shared memory 224 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 225 ALOGE("8-bit data in shared memory is not supported"); 226 return BAD_VALUE; 227 } 228 229 // force direct flag if format is not linear PCM 230 if (!audio_is_linear_pcm(format)) { 231 flags = (audio_output_flags_t) 232 // FIXME why can't we allow direct AND fast? 233 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 234 } 235 // only allow deep buffering for music stream type 236 if (streamType != AUDIO_STREAM_MUSIC) { 237 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 238 } 239 240 if (!audio_is_output_channel(channelMask)) { 241 ALOGE("Invalid channel mask %#x", channelMask); 242 return BAD_VALUE; 243 } 244 uint32_t channelCount = popcount(channelMask); 245 246 audio_io_handle_t output = AudioSystem::getOutput( 247 streamType, 248 sampleRate, format, channelMask, 249 flags); 250 251 if (output == 0) { 252 ALOGE("Could not get audio output for stream type %d", streamType); 253 return BAD_VALUE; 254 } 255 256 mVolume[LEFT] = 1.0f; 257 mVolume[RIGHT] = 1.0f; 258 mSendLevel = 0.0f; 259 mFrameCount = frameCount; 260 mNotificationFramesReq = notificationFrames; 261 mSessionId = sessionId; 262 mAuxEffectId = 0; 263 mFlags = flags; 264 mCbf = cbf; 265 266 if (cbf != NULL) { 267 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 268 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 269 } 270 271 // create the IAudioTrack 272 status_t status = createTrack_l(streamType, 273 sampleRate, 274 format, 275 channelMask, 276 frameCount, 277 flags, 278 sharedBuffer, 279 output); 280 281 if (status != NO_ERROR) { 282 if (mAudioTrackThread != 0) { 283 mAudioTrackThread->requestExit(); 284 mAudioTrackThread.clear(); 285 } 286 return status; 287 } 288 289 mStatus = NO_ERROR; 290 291 mStreamType = streamType; 292 mFormat = format; 293 mChannelMask = channelMask; 294 mChannelCount = channelCount; 295 296 if (audio_is_linear_pcm(format)) { 297 mFrameSize = channelCount * audio_bytes_per_sample(format); 298 mFrameSizeAF = channelCount * sizeof(int16_t); 299 } else { 300 mFrameSize = sizeof(uint8_t); 301 mFrameSizeAF = sizeof(uint8_t); 302 } 303 304 mSharedBuffer = sharedBuffer; 305 mMuted = false; 306 mActive = false; 307 mUserData = user; 308 mLoopCount = 0; 309 mMarkerPosition = 0; 310 mMarkerReached = false; 311 mNewPosition = 0; 312 mUpdatePeriod = 0; 313 mFlushed = false; 314 AudioSystem::acquireAudioSessionId(mSessionId); 315 return NO_ERROR; 316} 317 318status_t AudioTrack::initCheck() const 319{ 320 return mStatus; 321} 322 323// ------------------------------------------------------------------------- 324 325uint32_t AudioTrack::latency() const 326{ 327 return mLatency; 328} 329 330audio_stream_type_t AudioTrack::streamType() const 331{ 332 return mStreamType; 333} 334 335audio_format_t AudioTrack::format() const 336{ 337 return mFormat; 338} 339 340int AudioTrack::channelCount() const 341{ 342 return mChannelCount; 343} 344 345size_t AudioTrack::frameCount() const 346{ 347 return mCblk->frameCount; 348} 349 350sp<IMemory>& AudioTrack::sharedBuffer() 351{ 352 return mSharedBuffer; 353} 354 355// ------------------------------------------------------------------------- 356 357void AudioTrack::start() 358{ 359 sp<AudioTrackThread> t = mAudioTrackThread; 360 361 ALOGV("start %p", this); 362 363 AutoMutex lock(mLock); 364 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 365 // while we are accessing the cblk 366 sp<IAudioTrack> audioTrack = mAudioTrack; 367 sp<IMemory> iMem = mCblkMemory; 368 audio_track_cblk_t* cblk = mCblk; 369 370 if (!mActive) { 371 mFlushed = false; 372 mActive = true; 373 mNewPosition = cblk->server + mUpdatePeriod; 374 cblk->lock.lock(); 375 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 376 cblk->waitTimeMs = 0; 377 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 378 if (t != 0) { 379 t->resume(); 380 } else { 381 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 382 get_sched_policy(0, &mPreviousSchedulingGroup); 383 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 384 } 385 386 ALOGV("start %p before lock cblk %p", this, cblk); 387 status_t status = NO_ERROR; 388 if (!(cblk->flags & CBLK_INVALID)) { 389 cblk->lock.unlock(); 390 ALOGV("mAudioTrack->start()"); 391 status = mAudioTrack->start(); 392 cblk->lock.lock(); 393 if (status == DEAD_OBJECT) { 394 android_atomic_or(CBLK_INVALID, &cblk->flags); 395 } 396 } 397 if (cblk->flags & CBLK_INVALID) { 398 audio_track_cblk_t* temp = cblk; 399 status = restoreTrack_l(temp, true /*fromStart*/); 400 cblk = temp; 401 } 402 cblk->lock.unlock(); 403 if (status != NO_ERROR) { 404 ALOGV("start() failed"); 405 mActive = false; 406 if (t != 0) { 407 t->pause(); 408 } else { 409 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 410 set_sched_policy(0, mPreviousSchedulingGroup); 411 } 412 } 413 } 414 415} 416 417void AudioTrack::stop() 418{ 419 sp<AudioTrackThread> t = mAudioTrackThread; 420 421 ALOGV("stop %p", this); 422 423 AutoMutex lock(mLock); 424 if (mActive) { 425 mActive = false; 426 mCblk->cv.signal(); 427 mAudioTrack->stop(); 428 // Cancel loops (If we are in the middle of a loop, playback 429 // would not stop until loopCount reaches 0). 430 setLoop_l(0, 0, 0); 431 // the playback head position will reset to 0, so if a marker is set, we need 432 // to activate it again 433 mMarkerReached = false; 434 // Force flush if a shared buffer is used otherwise audioflinger 435 // will not stop before end of buffer is reached. 436 if (mSharedBuffer != 0) { 437 flush_l(); 438 } 439 if (t != 0) { 440 t->pause(); 441 } else { 442 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 443 set_sched_policy(0, mPreviousSchedulingGroup); 444 } 445 } 446 447} 448 449bool AudioTrack::stopped() const 450{ 451 AutoMutex lock(mLock); 452 return stopped_l(); 453} 454 455void AudioTrack::flush() 456{ 457 AutoMutex lock(mLock); 458 flush_l(); 459} 460 461// must be called with mLock held 462void AudioTrack::flush_l() 463{ 464 ALOGV("flush"); 465 466 // clear playback marker and periodic update counter 467 mMarkerPosition = 0; 468 mMarkerReached = false; 469 mUpdatePeriod = 0; 470 471 if (!mActive) { 472 mFlushed = true; 473 mAudioTrack->flush(); 474 // Release AudioTrack callback thread in case it was waiting for new buffers 475 // in AudioTrack::obtainBuffer() 476 mCblk->cv.signal(); 477 } 478} 479 480void AudioTrack::pause() 481{ 482 ALOGV("pause"); 483 AutoMutex lock(mLock); 484 if (mActive) { 485 mActive = false; 486 mCblk->cv.signal(); 487 mAudioTrack->pause(); 488 } 489} 490 491void AudioTrack::mute(bool e) 492{ 493 mAudioTrack->mute(e); 494 mMuted = e; 495} 496 497bool AudioTrack::muted() const 498{ 499 return mMuted; 500} 501 502status_t AudioTrack::setVolume(float left, float right) 503{ 504 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 505 return BAD_VALUE; 506 } 507 508 AutoMutex lock(mLock); 509 mVolume[LEFT] = left; 510 mVolume[RIGHT] = right; 511 512 mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 513 514 return NO_ERROR; 515} 516 517status_t AudioTrack::setVolume(float volume) 518{ 519 return setVolume(volume, volume); 520} 521 522status_t AudioTrack::setAuxEffectSendLevel(float level) 523{ 524 ALOGV("setAuxEffectSendLevel(%f)", level); 525 if (level < 0.0f || level > 1.0f) { 526 return BAD_VALUE; 527 } 528 AutoMutex lock(mLock); 529 530 mSendLevel = level; 531 532 mCblk->setSendLevel(level); 533 534 return NO_ERROR; 535} 536 537void AudioTrack::getAuxEffectSendLevel(float* level) const 538{ 539 if (level != NULL) { 540 *level = mSendLevel; 541 } 542} 543 544status_t AudioTrack::setSampleRate(uint32_t rate) 545{ 546 uint32_t afSamplingRate; 547 548 if (mIsTimed) { 549 return INVALID_OPERATION; 550 } 551 552 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 553 return NO_INIT; 554 } 555 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 556 if (rate == 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 557 558 AutoMutex lock(mLock); 559 mCblk->sampleRate = rate; 560 return NO_ERROR; 561} 562 563uint32_t AudioTrack::getSampleRate() const 564{ 565 if (mIsTimed) { 566 return 0; 567 } 568 569 AutoMutex lock(mLock); 570 return mCblk->sampleRate; 571} 572 573status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 574{ 575 AutoMutex lock(mLock); 576 return setLoop_l(loopStart, loopEnd, loopCount); 577} 578 579// must be called with mLock held 580status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 581{ 582 audio_track_cblk_t* cblk = mCblk; 583 584 Mutex::Autolock _l(cblk->lock); 585 586 if (loopCount == 0) { 587 cblk->loopStart = UINT_MAX; 588 cblk->loopEnd = UINT_MAX; 589 cblk->loopCount = 0; 590 mLoopCount = 0; 591 return NO_ERROR; 592 } 593 594 if (mIsTimed) { 595 return INVALID_OPERATION; 596 } 597 598 if (loopStart >= loopEnd || 599 loopEnd - loopStart > cblk->frameCount || 600 cblk->server > loopStart) { 601 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, " 602 "user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 603 return BAD_VALUE; 604 } 605 606 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 607 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " 608 "framecount %d", 609 loopStart, loopEnd, cblk->frameCount); 610 return BAD_VALUE; 611 } 612 613 cblk->loopStart = loopStart; 614 cblk->loopEnd = loopEnd; 615 cblk->loopCount = loopCount; 616 mLoopCount = loopCount; 617 618 return NO_ERROR; 619} 620 621status_t AudioTrack::setMarkerPosition(uint32_t marker) 622{ 623 if (mCbf == NULL) return INVALID_OPERATION; 624 625 mMarkerPosition = marker; 626 mMarkerReached = false; 627 628 return NO_ERROR; 629} 630 631status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 632{ 633 if (marker == NULL) return BAD_VALUE; 634 635 *marker = mMarkerPosition; 636 637 return NO_ERROR; 638} 639 640status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 641{ 642 if (mCbf == NULL) return INVALID_OPERATION; 643 644 uint32_t curPosition; 645 getPosition(&curPosition); 646 mNewPosition = curPosition + updatePeriod; 647 mUpdatePeriod = updatePeriod; 648 649 return NO_ERROR; 650} 651 652status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 653{ 654 if (updatePeriod == NULL) return BAD_VALUE; 655 656 *updatePeriod = mUpdatePeriod; 657 658 return NO_ERROR; 659} 660 661status_t AudioTrack::setPosition(uint32_t position) 662{ 663 if (mIsTimed) return INVALID_OPERATION; 664 665 AutoMutex lock(mLock); 666 667 if (!stopped_l()) return INVALID_OPERATION; 668 669 audio_track_cblk_t* cblk = mCblk; 670 Mutex::Autolock _l(cblk->lock); 671 672 if (position > cblk->user) return BAD_VALUE; 673 674 cblk->server = position; 675 android_atomic_or(CBLK_FORCEREADY, &cblk->flags); 676 677 return NO_ERROR; 678} 679 680status_t AudioTrack::getPosition(uint32_t *position) 681{ 682 if (position == NULL) return BAD_VALUE; 683 AutoMutex lock(mLock); 684 *position = mFlushed ? 0 : mCblk->server; 685 686 return NO_ERROR; 687} 688 689status_t AudioTrack::reload() 690{ 691 AutoMutex lock(mLock); 692 693 if (!stopped_l()) return INVALID_OPERATION; 694 695 flush_l(); 696 697 audio_track_cblk_t* cblk = mCblk; 698 cblk->stepUserOut(cblk->frameCount); 699 700 return NO_ERROR; 701} 702 703audio_io_handle_t AudioTrack::getOutput() 704{ 705 AutoMutex lock(mLock); 706 return getOutput_l(); 707} 708 709// must be called with mLock held 710audio_io_handle_t AudioTrack::getOutput_l() 711{ 712 return AudioSystem::getOutput(mStreamType, 713 mCblk->sampleRate, mFormat, mChannelMask, mFlags); 714} 715 716int AudioTrack::getSessionId() const 717{ 718 return mSessionId; 719} 720 721status_t AudioTrack::attachAuxEffect(int effectId) 722{ 723 ALOGV("attachAuxEffect(%d)", effectId); 724 status_t status = mAudioTrack->attachAuxEffect(effectId); 725 if (status == NO_ERROR) { 726 mAuxEffectId = effectId; 727 } 728 return status; 729} 730 731// ------------------------------------------------------------------------- 732 733// must be called with mLock held 734status_t AudioTrack::createTrack_l( 735 audio_stream_type_t streamType, 736 uint32_t sampleRate, 737 audio_format_t format, 738 audio_channel_mask_t channelMask, 739 size_t frameCount, 740 audio_output_flags_t flags, 741 const sp<IMemory>& sharedBuffer, 742 audio_io_handle_t output) 743{ 744 status_t status; 745 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 746 if (audioFlinger == 0) { 747 ALOGE("Could not get audioflinger"); 748 return NO_INIT; 749 } 750 751 uint32_t afLatency; 752 if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { 753 return NO_INIT; 754 } 755 756 // Client decides whether the track is TIMED (see below), but can only express a preference 757 // for FAST. Server will perform additional tests. 758 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 759 // either of these use cases: 760 // use case 1: shared buffer 761 (sharedBuffer != 0) || 762 // use case 2: callback handler 763 (mCbf != NULL))) { 764 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 765 // once denied, do not request again if IAudioTrack is re-created 766 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 767 mFlags = flags; 768 } 769 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 770 771 mNotificationFramesAct = mNotificationFramesReq; 772 773 if (!audio_is_linear_pcm(format)) { 774 775 if (sharedBuffer != 0) { 776 // Same comment as below about ignoring frameCount parameter for set() 777 frameCount = sharedBuffer->size(); 778 } else if (frameCount == 0) { 779 size_t afFrameCount; 780 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 781 return NO_INIT; 782 } 783 frameCount = afFrameCount; 784 } 785 786 } else if (sharedBuffer != 0) { 787 788 // Ensure that buffer alignment matches channelCount 789 int channelCount = popcount(channelMask); 790 // 8-bit data in shared memory is not currently supported by AudioFlinger 791 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 792 if (channelCount > 1) { 793 // More than 2 channels does not require stronger alignment than stereo 794 alignment <<= 1; 795 } 796 if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 797 ALOGE("Invalid buffer alignment: address %p, channelCount %d", 798 sharedBuffer->pointer(), channelCount); 799 return BAD_VALUE; 800 } 801 802 // When initializing a shared buffer AudioTrack via constructors, 803 // there's no frameCount parameter. 804 // But when initializing a shared buffer AudioTrack via set(), 805 // there _is_ a frameCount parameter. We silently ignore it. 806 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 807 808 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 809 810 // FIXME move these calculations and associated checks to server 811 uint32_t afSampleRate; 812 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 813 return NO_INIT; 814 } 815 size_t afFrameCount; 816 if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { 817 return NO_INIT; 818 } 819 820 // Ensure that buffer depth covers at least audio hardware latency 821 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 822 if (minBufCount < 2) minBufCount = 2; 823 824 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 825 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 826 ", afLatency=%d", 827 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 828 829 if (frameCount == 0) { 830 frameCount = minFrameCount; 831 } 832 if (mNotificationFramesAct == 0) { 833 mNotificationFramesAct = frameCount/2; 834 } 835 // Make sure that application is notified with sufficient margin 836 // before underrun 837 if (mNotificationFramesAct > frameCount/2) { 838 mNotificationFramesAct = frameCount/2; 839 } 840 if (frameCount < minFrameCount) { 841 // not ALOGW because it happens all the time when playing key clicks over A2DP 842 ALOGV("Minimum buffer size corrected from %d to %d", 843 frameCount, minFrameCount); 844 frameCount = minFrameCount; 845 } 846 847 } else { 848 // For fast tracks, the frame count calculations and checks are done by server 849 } 850 851 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 852 if (mIsTimed) { 853 trackFlags |= IAudioFlinger::TRACK_TIMED; 854 } 855 856 pid_t tid = -1; 857 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 858 trackFlags |= IAudioFlinger::TRACK_FAST; 859 if (mAudioTrackThread != 0) { 860 tid = mAudioTrackThread->getTid(); 861 } 862 } 863 864 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 865 streamType, 866 sampleRate, 867 // AudioFlinger only sees 16-bit PCM 868 format == AUDIO_FORMAT_PCM_8_BIT ? 869 AUDIO_FORMAT_PCM_16_BIT : format, 870 channelMask, 871 frameCount, 872 &trackFlags, 873 sharedBuffer, 874 output, 875 tid, 876 &mSessionId, 877 &status); 878 879 if (track == 0) { 880 ALOGE("AudioFlinger could not create track, status: %d", status); 881 return status; 882 } 883 sp<IMemory> iMem = track->getCblk(); 884 if (iMem == 0) { 885 ALOGE("Could not get control block"); 886 return NO_INIT; 887 } 888 mAudioTrack = track; 889 mCblkMemory = iMem; 890 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 891 mCblk = cblk; 892 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 893 if (trackFlags & IAudioFlinger::TRACK_FAST) { 894 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", cblk->frameCount); 895 } else { 896 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", cblk->frameCount); 897 // once denied, do not request again if IAudioTrack is re-created 898 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 899 mFlags = flags; 900 } 901 if (sharedBuffer == 0) { 902 mNotificationFramesAct = cblk->frameCount/2; 903 } 904 } 905 if (sharedBuffer == 0) { 906 mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); 907 } else { 908 mBuffers = sharedBuffer->pointer(); 909 // Force buffer full condition as data is already present in shared memory 910 cblk->stepUserOut(cblk->frameCount); 911 } 912 913 cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 914 uint16_t(mVolume[LEFT] * 0x1000)); 915 cblk->setSendLevel(mSendLevel); 916 mAudioTrack->attachAuxEffect(mAuxEffectId); 917 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 918 cblk->waitTimeMs = 0; 919 mRemainingFrames = mNotificationFramesAct; 920 // FIXME don't believe this lie 921 mLatency = afLatency + (1000*cblk->frameCount) / sampleRate; 922 // If IAudioTrack is re-created, don't let the requested frameCount 923 // decrease. This can confuse clients that cache frameCount(). 924 if (cblk->frameCount > mFrameCount) { 925 mFrameCount = cblk->frameCount; 926 } 927 return NO_ERROR; 928} 929 930status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 931{ 932 AutoMutex lock(mLock); 933 bool active; 934 status_t result = NO_ERROR; 935 audio_track_cblk_t* cblk = mCblk; 936 uint32_t framesReq = audioBuffer->frameCount; 937 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 938 939 audioBuffer->frameCount = 0; 940 audioBuffer->size = 0; 941 942 uint32_t framesAvail = cblk->framesAvailableOut(); 943 944 cblk->lock.lock(); 945 if (cblk->flags & CBLK_INVALID) { 946 goto create_new_track; 947 } 948 cblk->lock.unlock(); 949 950 if (framesAvail == 0) { 951 cblk->lock.lock(); 952 goto start_loop_here; 953 while (framesAvail == 0) { 954 active = mActive; 955 if (CC_UNLIKELY(!active)) { 956 ALOGV("Not active and NO_MORE_BUFFERS"); 957 cblk->lock.unlock(); 958 return NO_MORE_BUFFERS; 959 } 960 if (CC_UNLIKELY(!waitCount)) { 961 cblk->lock.unlock(); 962 return WOULD_BLOCK; 963 } 964 if (!(cblk->flags & CBLK_INVALID)) { 965 mLock.unlock(); 966 // this condition is in shared memory, so if IAudioTrack and control block 967 // are replaced due to mediaserver death or IAudioTrack invalidation then 968 // cv won't be signalled, but fortunately the timeout will limit the wait 969 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 970 cblk->lock.unlock(); 971 mLock.lock(); 972 if (!mActive) { 973 return status_t(STOPPED); 974 } 975 // IAudioTrack may have been re-created while mLock was unlocked 976 cblk = mCblk; 977 cblk->lock.lock(); 978 } 979 980 if (cblk->flags & CBLK_INVALID) { 981 goto create_new_track; 982 } 983 if (CC_UNLIKELY(result != NO_ERROR)) { 984 cblk->waitTimeMs += waitTimeMs; 985 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 986 // timing out when a loop has been set and we have already written upto loop end 987 // is a normal condition: no need to wake AudioFlinger up. 988 if (cblk->user < cblk->loopEnd) { 989 ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " 990 "server=%08x", this, cblk->mName, cblk->user, cblk->server); 991 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 992 cblk->lock.unlock(); 993 result = mAudioTrack->start(); 994 cblk->lock.lock(); 995 if (result == DEAD_OBJECT) { 996 android_atomic_or(CBLK_INVALID, &cblk->flags); 997create_new_track: 998 audio_track_cblk_t* temp = cblk; 999 result = restoreTrack_l(temp, false /*fromStart*/); 1000 cblk = temp; 1001 } 1002 if (result != NO_ERROR) { 1003 ALOGW("obtainBuffer create Track error %d", result); 1004 cblk->lock.unlock(); 1005 return result; 1006 } 1007 } 1008 cblk->waitTimeMs = 0; 1009 } 1010 1011 if (--waitCount == 0) { 1012 cblk->lock.unlock(); 1013 return TIMED_OUT; 1014 } 1015 } 1016 // read the server count again 1017 start_loop_here: 1018 framesAvail = cblk->framesAvailableOut_l(); 1019 } 1020 cblk->lock.unlock(); 1021 } 1022 1023 cblk->waitTimeMs = 0; 1024 1025 if (framesReq > framesAvail) { 1026 framesReq = framesAvail; 1027 } 1028 1029 uint32_t u = cblk->user; 1030 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 1031 1032 if (framesReq > bufferEnd - u) { 1033 framesReq = bufferEnd - u; 1034 } 1035 1036 audioBuffer->frameCount = framesReq; 1037 audioBuffer->size = framesReq * mFrameSizeAF; 1038 audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u); 1039 active = mActive; 1040 return active ? status_t(NO_ERROR) : status_t(STOPPED); 1041} 1042 1043void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1044{ 1045 AutoMutex lock(mLock); 1046 audio_track_cblk_t* cblk = mCblk; 1047 cblk->stepUserOut(audioBuffer->frameCount); 1048 if (audioBuffer->frameCount > 0) { 1049 // restart track if it was disabled by audioflinger due to previous underrun 1050 if (mActive && (cblk->flags & CBLK_DISABLED)) { 1051 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1052 ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName); 1053 mAudioTrack->start(); 1054 } 1055 } 1056} 1057 1058// ------------------------------------------------------------------------- 1059 1060ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1061{ 1062 1063 if (mSharedBuffer != 0) return INVALID_OPERATION; 1064 if (mIsTimed) return INVALID_OPERATION; 1065 1066 if (ssize_t(userSize) < 0) { 1067 // Sanity-check: user is most-likely passing an error code, and it would 1068 // make the return value ambiguous (actualSize vs error). 1069 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1070 buffer, userSize, userSize); 1071 return BAD_VALUE; 1072 } 1073 1074 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1075 1076 if (userSize == 0) { 1077 return 0; 1078 } 1079 1080 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1081 // while we are accessing the cblk 1082 mLock.lock(); 1083 sp<IAudioTrack> audioTrack = mAudioTrack; 1084 sp<IMemory> iMem = mCblkMemory; 1085 mLock.unlock(); 1086 1087 // since mLock is unlocked the IAudioTrack and shared memory may be re-created, 1088 // so all cblk references might still refer to old shared memory, but that should be benign 1089 1090 ssize_t written = 0; 1091 const int8_t *src = (const int8_t *)buffer; 1092 Buffer audioBuffer; 1093 size_t frameSz = frameSize(); 1094 1095 do { 1096 audioBuffer.frameCount = userSize/frameSz; 1097 1098 status_t err = obtainBuffer(&audioBuffer, -1); 1099 if (err < 0) { 1100 // out of buffers, return #bytes written 1101 if (err == status_t(NO_MORE_BUFFERS)) 1102 break; 1103 return ssize_t(err); 1104 } 1105 1106 size_t toWrite; 1107 1108 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1109 // Divide capacity by 2 to take expansion into account 1110 toWrite = audioBuffer.size>>1; 1111 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1112 } else { 1113 toWrite = audioBuffer.size; 1114 memcpy(audioBuffer.i8, src, toWrite); 1115 } 1116 src += toWrite; 1117 userSize -= toWrite; 1118 written += toWrite; 1119 1120 releaseBuffer(&audioBuffer); 1121 } while (userSize >= frameSz); 1122 1123 return written; 1124} 1125 1126// ------------------------------------------------------------------------- 1127 1128TimedAudioTrack::TimedAudioTrack() { 1129 mIsTimed = true; 1130} 1131 1132status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1133{ 1134 AutoMutex lock(mLock); 1135 status_t result = UNKNOWN_ERROR; 1136 1137 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1138 // while we are accessing the cblk 1139 sp<IAudioTrack> audioTrack = mAudioTrack; 1140 sp<IMemory> iMem = mCblkMemory; 1141 1142 // If the track is not invalid already, try to allocate a buffer. alloc 1143 // fails indicating that the server is dead, flag the track as invalid so 1144 // we can attempt to restore in just a bit. 1145 audio_track_cblk_t* cblk = mCblk; 1146 if (!(cblk->flags & CBLK_INVALID)) { 1147 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1148 if (result == DEAD_OBJECT) { 1149 android_atomic_or(CBLK_INVALID, &cblk->flags); 1150 } 1151 } 1152 1153 // If the track is invalid at this point, attempt to restore it. and try the 1154 // allocation one more time. 1155 if (cblk->flags & CBLK_INVALID) { 1156 cblk->lock.lock(); 1157 audio_track_cblk_t* temp = cblk; 1158 result = restoreTrack_l(temp, false /*fromStart*/); 1159 cblk = temp; 1160 cblk->lock.unlock(); 1161 1162 if (result == OK) 1163 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1164 } 1165 1166 return result; 1167} 1168 1169status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1170 int64_t pts) 1171{ 1172 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1173 { 1174 AutoMutex lock(mLock); 1175 audio_track_cblk_t* cblk = mCblk; 1176 // restart track if it was disabled by audioflinger due to previous underrun 1177 if (buffer->size() != 0 && status == NO_ERROR && 1178 mActive && (cblk->flags & CBLK_DISABLED)) { 1179 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1180 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1181 mAudioTrack->start(); 1182 } 1183 } 1184 return status; 1185} 1186 1187status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1188 TargetTimeline target) 1189{ 1190 return mAudioTrack->setMediaTimeTransform(xform, target); 1191} 1192 1193// ------------------------------------------------------------------------- 1194 1195bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1196{ 1197 Buffer audioBuffer; 1198 uint32_t frames; 1199 size_t writtenSize; 1200 1201 mLock.lock(); 1202 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1203 // while we are accessing the cblk 1204 sp<IAudioTrack> audioTrack = mAudioTrack; 1205 sp<IMemory> iMem = mCblkMemory; 1206 audio_track_cblk_t* cblk = mCblk; 1207 bool active = mActive; 1208 mLock.unlock(); 1209 1210 // since mLock is unlocked the IAudioTrack and shared memory may be re-created, 1211 // so all cblk references might still refer to old shared memory, but that should be benign 1212 1213 // Manage underrun callback 1214 if (active && (cblk->framesAvailableOut() == cblk->frameCount)) { 1215 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1216 if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { 1217 mCbf(EVENT_UNDERRUN, mUserData, 0); 1218 if (cblk->server == cblk->frameCount) { 1219 mCbf(EVENT_BUFFER_END, mUserData, 0); 1220 } 1221 if (mSharedBuffer != 0) return false; 1222 } 1223 } 1224 1225 // Manage loop end callback 1226 while (mLoopCount > cblk->loopCount) { 1227 int loopCount = -1; 1228 mLoopCount--; 1229 if (mLoopCount >= 0) loopCount = mLoopCount; 1230 1231 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1232 } 1233 1234 // Manage marker callback 1235 if (!mMarkerReached && (mMarkerPosition > 0)) { 1236 if (cblk->server >= mMarkerPosition) { 1237 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1238 mMarkerReached = true; 1239 } 1240 } 1241 1242 // Manage new position callback 1243 if (mUpdatePeriod > 0) { 1244 while (cblk->server >= mNewPosition) { 1245 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1246 mNewPosition += mUpdatePeriod; 1247 } 1248 } 1249 1250 // If Shared buffer is used, no data is requested from client. 1251 if (mSharedBuffer != 0) { 1252 frames = 0; 1253 } else { 1254 frames = mRemainingFrames; 1255 } 1256 1257 // See description of waitCount parameter at declaration of obtainBuffer(). 1258 // The logic below prevents us from being stuck below at obtainBuffer() 1259 // not being able to handle timed events (position, markers, loops). 1260 int32_t waitCount = -1; 1261 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1262 waitCount = 1; 1263 } 1264 1265 do { 1266 1267 audioBuffer.frameCount = frames; 1268 1269 status_t err = obtainBuffer(&audioBuffer, waitCount); 1270 if (err < NO_ERROR) { 1271 if (err != TIMED_OUT) { 1272 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), 1273 "Error obtaining an audio buffer, giving up."); 1274 return false; 1275 } 1276 break; 1277 } 1278 if (err == status_t(STOPPED)) return false; 1279 1280 // Divide buffer size by 2 to take into account the expansion 1281 // due to 8 to 16 bit conversion: the callback must fill only half 1282 // of the destination buffer 1283 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1284 audioBuffer.size >>= 1; 1285 } 1286 1287 size_t reqSize = audioBuffer.size; 1288 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1289 writtenSize = audioBuffer.size; 1290 1291 // Sanity check on returned size 1292 if (ssize_t(writtenSize) <= 0) { 1293 // The callback is done filling buffers 1294 // Keep this thread going to handle timed events and 1295 // still try to get more data in intervals of WAIT_PERIOD_MS 1296 // but don't just loop and block the CPU, so wait 1297 usleep(WAIT_PERIOD_MS*1000); 1298 break; 1299 } 1300 1301 if (writtenSize > reqSize) writtenSize = reqSize; 1302 1303 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1304 // 8 to 16 bit conversion, note that source and destination are the same address 1305 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1306 writtenSize <<= 1; 1307 } 1308 1309 audioBuffer.size = writtenSize; 1310 // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for 1311 // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of 1312 // 16 bit. 1313 audioBuffer.frameCount = writtenSize / mFrameSizeAF; 1314 1315 frames -= audioBuffer.frameCount; 1316 1317 releaseBuffer(&audioBuffer); 1318 } 1319 while (frames); 1320 1321 if (frames == 0) { 1322 mRemainingFrames = mNotificationFramesAct; 1323 } else { 1324 mRemainingFrames = frames; 1325 } 1326 return true; 1327} 1328 1329// must be called with mLock and refCblk.lock held. Callers must also hold strong references on 1330// the IAudioTrack and IMemory in case they are recreated here. 1331// If the IAudioTrack is successfully restored, the refCblk pointer is updated 1332// FIXME Don't depend on caller to hold strong references. 1333status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart) 1334{ 1335 status_t result; 1336 1337 audio_track_cblk_t* cblk = refCblk; 1338 audio_track_cblk_t* newCblk = cblk; 1339 ALOGW("dead IAudioTrack, creating a new one from %s TID %d", 1340 fromStart ? "start()" : "obtainBuffer()", gettid()); 1341 1342 // signal old cblk condition so that other threads waiting for available buffers stop 1343 // waiting now 1344 cblk->cv.broadcast(); 1345 cblk->lock.unlock(); 1346 1347 // refresh the audio configuration cache in this process to make sure we get new 1348 // output parameters in getOutput_l() and createTrack_l() 1349 AudioSystem::clearAudioConfigCache(); 1350 1351 // if the new IAudioTrack is created, createTrack_l() will modify the 1352 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1353 // It will also delete the strong references on previous IAudioTrack and IMemory 1354 result = createTrack_l(mStreamType, 1355 cblk->sampleRate, 1356 mFormat, 1357 mChannelMask, 1358 mFrameCount, 1359 mFlags, 1360 mSharedBuffer, 1361 getOutput_l()); 1362 1363 if (result == NO_ERROR) { 1364 uint32_t user = cblk->user; 1365 uint32_t server = cblk->server; 1366 // restore write index and set other indexes to reflect empty buffer status 1367 newCblk = mCblk; 1368 newCblk->user = user; 1369 newCblk->server = user; 1370 newCblk->userBase = user; 1371 newCblk->serverBase = user; 1372 // restore loop: this is not guaranteed to succeed if new frame count is not 1373 // compatible with loop length 1374 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1375 if (!fromStart) { 1376 newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1377 // Make sure that a client relying on callback events indicating underrun or 1378 // the actual amount of audio frames played (e.g SoundPool) receives them. 1379 if (mSharedBuffer == 0) { 1380 uint32_t frames = 0; 1381 if (user > server) { 1382 frames = ((user - server) > newCblk->frameCount) ? 1383 newCblk->frameCount : (user - server); 1384 memset(mBuffers, 0, frames * mFrameSizeAF); 1385 } 1386 // restart playback even if buffer is not completely filled. 1387 android_atomic_or(CBLK_FORCEREADY, &newCblk->flags); 1388 // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to 1389 // the client 1390 newCblk->stepUserOut(frames); 1391 } 1392 } 1393 if (mSharedBuffer != 0) { 1394 newCblk->stepUserOut(newCblk->frameCount); 1395 } 1396 if (mActive) { 1397 result = mAudioTrack->start(); 1398 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1399 } 1400 if (fromStart && result == NO_ERROR) { 1401 mNewPosition = newCblk->server + mUpdatePeriod; 1402 } 1403 } 1404 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1405 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1406 result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); 1407 1408 if (result == NO_ERROR) { 1409 // from now on we switch to the newly created cblk 1410 refCblk = newCblk; 1411 } 1412 newCblk->lock.lock(); 1413 1414 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1415 1416 return result; 1417} 1418 1419status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1420{ 1421 1422 const size_t SIZE = 256; 1423 char buffer[SIZE]; 1424 String8 result; 1425 1426 audio_track_cblk_t* cblk = mCblk; 1427 result.append(" AudioTrack::dump\n"); 1428 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1429 mVolume[0], mVolume[1]); 1430 result.append(buffer); 1431 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1432 mChannelCount, cblk->frameCount); 1433 result.append(buffer); 1434 snprintf(buffer, 255, " sample rate(%u), status(%d), muted(%d)\n", 1435 (cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted); 1436 result.append(buffer); 1437 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1438 result.append(buffer); 1439 ::write(fd, result.string(), result.size()); 1440 return NO_ERROR; 1441} 1442 1443// ========================================================================= 1444 1445AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1446 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1447{ 1448} 1449 1450AudioTrack::AudioTrackThread::~AudioTrackThread() 1451{ 1452} 1453 1454bool AudioTrack::AudioTrackThread::threadLoop() 1455{ 1456 { 1457 AutoMutex _l(mMyLock); 1458 if (mPaused) { 1459 mMyCond.wait(mMyLock); 1460 // caller will check for exitPending() 1461 return true; 1462 } 1463 } 1464 if (!mReceiver.processAudioBuffer(this)) { 1465 pause(); 1466 } 1467 return true; 1468} 1469 1470void AudioTrack::AudioTrackThread::requestExit() 1471{ 1472 // must be in this order to avoid a race condition 1473 Thread::requestExit(); 1474 resume(); 1475} 1476 1477void AudioTrack::AudioTrackThread::pause() 1478{ 1479 AutoMutex _l(mMyLock); 1480 mPaused = true; 1481} 1482 1483void AudioTrack::AudioTrackThread::resume() 1484{ 1485 AutoMutex _l(mMyLock); 1486 if (mPaused) { 1487 mPaused = false; 1488 mMyCond.signal(); 1489 } 1490} 1491 1492// ========================================================================= 1493 1494 1495audio_track_cblk_t::audio_track_cblk_t() 1496 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1497 userBase(0), serverBase(0), frameCount(0), 1498 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), 1499 mSendLevel(0), flags(0) 1500{ 1501} 1502 1503uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount, bool isOut) 1504{ 1505 ALOGV("stepuser %08x %08x %d", user, server, frameCount); 1506 1507 uint32_t u = user; 1508 u += frameCount; 1509 // Ensure that user is never ahead of server for AudioRecord 1510 if (isOut) { 1511 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1512 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1513 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1514 } 1515 } else if (u > server) { 1516 ALOGW("stepUser occurred after track reset"); 1517 u = server; 1518 } 1519 1520 uint32_t fc = this->frameCount; 1521 if (u >= fc) { 1522 // common case, user didn't just wrap 1523 if (u - fc >= userBase ) { 1524 userBase += fc; 1525 } 1526 } else if (u >= userBase + fc) { 1527 // user just wrapped 1528 userBase += fc; 1529 } 1530 1531 user = u; 1532 1533 // Clear flow control error condition as new data has been written/read to/from buffer. 1534 if (flags & CBLK_UNDERRUN) { 1535 android_atomic_and(~CBLK_UNDERRUN, &flags); 1536 } 1537 1538 return u; 1539} 1540 1541bool audio_track_cblk_t::stepServer(uint32_t frameCount, bool isOut) 1542{ 1543 ALOGV("stepserver %08x %08x %d", user, server, frameCount); 1544 1545 if (!tryLock()) { 1546 ALOGW("stepServer() could not lock cblk"); 1547 return false; 1548 } 1549 1550 uint32_t s = server; 1551 bool flushed = (s == user); 1552 1553 s += frameCount; 1554 if (isOut) { 1555 // Mark that we have read the first buffer so that next time stepUser() is called 1556 // we switch to normal obtainBuffer() timeout period 1557 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1558 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1559 } 1560 // It is possible that we receive a flush() 1561 // while the mixer is processing a block: in this case, 1562 // stepServer() is called After the flush() has reset u & s and 1563 // we have s > u 1564 if (flushed) { 1565 ALOGW("stepServer occurred after track reset"); 1566 s = user; 1567 } 1568 } 1569 1570 if (s >= loopEnd) { 1571 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1572 s = loopStart; 1573 if (--loopCount == 0) { 1574 loopEnd = UINT_MAX; 1575 loopStart = UINT_MAX; 1576 } 1577 } 1578 1579 uint32_t fc = this->frameCount; 1580 if (s >= fc) { 1581 // common case, server didn't just wrap 1582 if (s - fc >= serverBase ) { 1583 serverBase += fc; 1584 } 1585 } else if (s >= serverBase + fc) { 1586 // server just wrapped 1587 serverBase += fc; 1588 } 1589 1590 server = s; 1591 1592 if (!(flags & CBLK_INVALID)) { 1593 cv.signal(); 1594 } 1595 lock.unlock(); 1596 return true; 1597} 1598 1599void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const 1600{ 1601 return (int8_t *)buffers + (offset - userBase) * frameSize; 1602} 1603 1604uint32_t audio_track_cblk_t::framesAvailable(bool isOut) 1605{ 1606 Mutex::Autolock _l(lock); 1607 return framesAvailable_l(isOut); 1608} 1609 1610uint32_t audio_track_cblk_t::framesAvailable_l(bool isOut) 1611{ 1612 uint32_t u = user; 1613 uint32_t s = server; 1614 1615 if (isOut) { 1616 uint32_t limit = (s < loopStart) ? s : loopStart; 1617 return limit + frameCount - u; 1618 } else { 1619 return frameCount + u - s; 1620 } 1621} 1622 1623uint32_t audio_track_cblk_t::framesReady(bool isOut) 1624{ 1625 uint32_t u = user; 1626 uint32_t s = server; 1627 1628 if (isOut) { 1629 if (u < loopEnd) { 1630 return u - s; 1631 } else { 1632 // do not block on mutex shared with client on AudioFlinger side 1633 if (!tryLock()) { 1634 ALOGW("framesReady() could not lock cblk"); 1635 return 0; 1636 } 1637 uint32_t frames = UINT_MAX; 1638 if (loopCount >= 0) { 1639 frames = (loopEnd - loopStart)*loopCount + u - s; 1640 } 1641 lock.unlock(); 1642 return frames; 1643 } 1644 } else { 1645 return s - u; 1646 } 1647} 1648 1649bool audio_track_cblk_t::tryLock() 1650{ 1651 // the code below simulates lock-with-timeout 1652 // we MUST do this to protect the AudioFlinger server 1653 // as this lock is shared with the client. 1654 status_t err; 1655 1656 err = lock.tryLock(); 1657 if (err == -EBUSY) { // just wait a bit 1658 usleep(1000); 1659 err = lock.tryLock(); 1660 } 1661 if (err != NO_ERROR) { 1662 // probably, the client just died. 1663 return false; 1664 } 1665 return true; 1666} 1667 1668// ------------------------------------------------------------------------- 1669 1670}; // namespace android 1671