AudioTrack.cpp revision ba85098eb31bd2637db49816f0591361211024f2
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <stdint.h> 23#include <sys/types.h> 24#include <limits.h> 25 26#include <sched.h> 27#include <sys/resource.h> 28 29#include <private/media/AudioTrackShared.h> 30 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Log.h> 35#include <binder/Parcel.h> 36#include <binder/IPCThreadState.h> 37#include <utils/Timers.h> 38#include <utils/Atomic.h> 39 40#include <cutils/bitops.h> 41#include <cutils/compiler.h> 42 43#include <system/audio.h> 44#include <system/audio_policy.h> 45 46#include <audio_utils/primitives.h> 47 48namespace android { 49// --------------------------------------------------------------------------- 50 51// static 52status_t AudioTrack::getMinFrameCount( 53 int* frameCount, 54 audio_stream_type_t streamType, 55 uint32_t sampleRate) 56{ 57 if (frameCount == NULL) return BAD_VALUE; 58 59 // default to 0 in case of error 60 *frameCount = 0; 61 62 // FIXME merge with similar code in createTrack_l(), except we're missing 63 // some information here that is available in createTrack_l(): 64 // audio_io_handle_t output 65 // audio_format_t format 66 // audio_channel_mask_t channelMask 67 // audio_output_flags_t flags 68 int afSampleRate; 69 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 70 return NO_INIT; 71 } 72 int afFrameCount; 73 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 74 return NO_INIT; 75 } 76 uint32_t afLatency; 77 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 78 return NO_INIT; 79 } 80 81 // Ensure that buffer depth covers at least audio hardware latency 82 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 83 if (minBufCount < 2) minBufCount = 2; 84 85 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 86 afFrameCount * minBufCount * sampleRate / afSampleRate; 87 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 88 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 89 return NO_ERROR; 90} 91 92// --------------------------------------------------------------------------- 93 94AudioTrack::AudioTrack() 95 : mStatus(NO_INIT), 96 mIsTimed(false), 97 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 98 mPreviousSchedulingGroup(SP_DEFAULT) 99{ 100} 101 102AudioTrack::AudioTrack( 103 audio_stream_type_t streamType, 104 uint32_t sampleRate, 105 audio_format_t format, 106 audio_channel_mask_t channelMask, 107 int frameCount, 108 audio_output_flags_t flags, 109 callback_t cbf, 110 void* user, 111 int notificationFrames, 112 int sessionId) 113 : mStatus(NO_INIT), 114 mIsTimed(false), 115 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 116 mPreviousSchedulingGroup(SP_DEFAULT) 117{ 118 mStatus = set(streamType, sampleRate, format, channelMask, 119 frameCount, flags, cbf, user, notificationFrames, 120 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); 121} 122 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 frameCount, 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{ 178 179 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 180 sharedBuffer->size()); 181 182 ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags); 183 184 AutoMutex lock(mLock); 185 if (mAudioTrack != 0) { 186 ALOGE("Track already in use"); 187 return INVALID_OPERATION; 188 } 189 190 // handle default values first. 191 if (streamType == AUDIO_STREAM_DEFAULT) { 192 streamType = AUDIO_STREAM_MUSIC; 193 } 194 195 if (sampleRate == 0) { 196 int afSampleRate; 197 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 198 return NO_INIT; 199 } 200 sampleRate = afSampleRate; 201 } 202 203 // these below should probably come from the audioFlinger too... 204 if (format == AUDIO_FORMAT_DEFAULT) { 205 format = AUDIO_FORMAT_PCM_16_BIT; 206 } 207 if (channelMask == 0) { 208 channelMask = AUDIO_CHANNEL_OUT_STEREO; 209 } 210 211 // validate parameters 212 if (!audio_is_valid_format(format)) { 213 ALOGE("Invalid format"); 214 return BAD_VALUE; 215 } 216 217 // AudioFlinger does not currently support 8-bit data in shared memory 218 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 219 ALOGE("8-bit data in shared memory is not supported"); 220 return BAD_VALUE; 221 } 222 223 // force direct flag if format is not linear PCM 224 if (!audio_is_linear_pcm(format)) { 225 flags = (audio_output_flags_t) 226 // FIXME why can't we allow direct AND fast? 227 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 228 } 229 // only allow deep buffering for music stream type 230 if (streamType != AUDIO_STREAM_MUSIC) { 231 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 232 } 233 234 if (!audio_is_output_channel(channelMask)) { 235 ALOGE("Invalid channel mask %#x", channelMask); 236 return BAD_VALUE; 237 } 238 uint32_t channelCount = popcount(channelMask); 239 240 audio_io_handle_t output = AudioSystem::getOutput( 241 streamType, 242 sampleRate, format, channelMask, 243 flags); 244 245 if (output == 0) { 246 ALOGE("Could not get audio output for stream type %d", streamType); 247 return BAD_VALUE; 248 } 249 250 mVolume[LEFT] = 1.0f; 251 mVolume[RIGHT] = 1.0f; 252 mSendLevel = 0.0f; 253 mFrameCount = frameCount; 254 mNotificationFramesReq = notificationFrames; 255 mSessionId = sessionId; 256 mAuxEffectId = 0; 257 mFlags = flags; 258 mCbf = cbf; 259 260 if (cbf != NULL) { 261 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 262 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 263 } 264 265 // create the IAudioTrack 266 status_t status = createTrack_l(streamType, 267 sampleRate, 268 format, 269 channelMask, 270 frameCount, 271 flags, 272 sharedBuffer, 273 output); 274 275 if (status != NO_ERROR) { 276 if (mAudioTrackThread != 0) { 277 mAudioTrackThread->requestExit(); 278 mAudioTrackThread.clear(); 279 } 280 return status; 281 } 282 283 mStatus = NO_ERROR; 284 285 mStreamType = streamType; 286 mFormat = format; 287 mChannelMask = channelMask; 288 mChannelCount = channelCount; 289 mSharedBuffer = sharedBuffer; 290 mMuted = false; 291 mActive = false; 292 mUserData = user; 293 mLoopCount = 0; 294 mMarkerPosition = 0; 295 mMarkerReached = false; 296 mNewPosition = 0; 297 mUpdatePeriod = 0; 298 mFlushed = false; 299 AudioSystem::acquireAudioSessionId(mSessionId); 300 mRestoreStatus = NO_ERROR; 301 return NO_ERROR; 302} 303 304status_t AudioTrack::initCheck() const 305{ 306 return mStatus; 307} 308 309// ------------------------------------------------------------------------- 310 311uint32_t AudioTrack::latency() const 312{ 313 return mLatency; 314} 315 316audio_stream_type_t AudioTrack::streamType() const 317{ 318 return mStreamType; 319} 320 321audio_format_t AudioTrack::format() const 322{ 323 return mFormat; 324} 325 326int AudioTrack::channelCount() const 327{ 328 return mChannelCount; 329} 330 331uint32_t AudioTrack::frameCount() const 332{ 333 return mCblk->frameCount; 334} 335 336size_t AudioTrack::frameSize() const 337{ 338 if (audio_is_linear_pcm(mFormat)) { 339 return channelCount()*audio_bytes_per_sample(mFormat); 340 } else { 341 return sizeof(uint8_t); 342 } 343} 344 345sp<IMemory>& AudioTrack::sharedBuffer() 346{ 347 return mSharedBuffer; 348} 349 350// ------------------------------------------------------------------------- 351 352void AudioTrack::start() 353{ 354 sp<AudioTrackThread> t = mAudioTrackThread; 355 356 ALOGV("start %p", this); 357 358 AutoMutex lock(mLock); 359 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 360 // while we are accessing the cblk 361 sp<IAudioTrack> audioTrack = mAudioTrack; 362 sp<IMemory> iMem = mCblkMemory; 363 audio_track_cblk_t* cblk = mCblk; 364 365 if (!mActive) { 366 mFlushed = false; 367 mActive = true; 368 mNewPosition = cblk->server + mUpdatePeriod; 369 cblk->lock.lock(); 370 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 371 cblk->waitTimeMs = 0; 372 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 373 if (t != 0) { 374 t->resume(); 375 } else { 376 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 377 get_sched_policy(0, &mPreviousSchedulingGroup); 378 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 379 } 380 381 ALOGV("start %p before lock cblk %p", this, cblk); 382 status_t status = NO_ERROR; 383 if (!(cblk->flags & CBLK_INVALID)) { 384 cblk->lock.unlock(); 385 ALOGV("mAudioTrack->start()"); 386 status = mAudioTrack->start(); 387 cblk->lock.lock(); 388 if (status == DEAD_OBJECT) { 389 android_atomic_or(CBLK_INVALID, &cblk->flags); 390 } 391 } 392 if (cblk->flags & CBLK_INVALID) { 393 audio_track_cblk_t* temp = cblk; 394 status = restoreTrack_l(temp, true); 395 cblk = temp; 396 } 397 cblk->lock.unlock(); 398 if (status != NO_ERROR) { 399 ALOGV("start() failed"); 400 mActive = false; 401 if (t != 0) { 402 t->pause(); 403 } else { 404 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 405 set_sched_policy(0, mPreviousSchedulingGroup); 406 } 407 } 408 } 409 410} 411 412void AudioTrack::stop() 413{ 414 sp<AudioTrackThread> t = mAudioTrackThread; 415 416 ALOGV("stop %p", this); 417 418 AutoMutex lock(mLock); 419 if (mActive) { 420 mActive = false; 421 mCblk->cv.signal(); 422 mAudioTrack->stop(); 423 // Cancel loops (If we are in the middle of a loop, playback 424 // would not stop until loopCount reaches 0). 425 setLoop_l(0, 0, 0); 426 // the playback head position will reset to 0, so if a marker is set, we need 427 // to activate it again 428 mMarkerReached = false; 429 // Force flush if a shared buffer is used otherwise audioflinger 430 // will not stop before end of buffer is reached. 431 if (mSharedBuffer != 0) { 432 flush_l(); 433 } 434 if (t != 0) { 435 t->pause(); 436 } else { 437 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 438 set_sched_policy(0, mPreviousSchedulingGroup); 439 } 440 } 441 442} 443 444bool AudioTrack::stopped() const 445{ 446 AutoMutex lock(mLock); 447 return stopped_l(); 448} 449 450void AudioTrack::flush() 451{ 452 AutoMutex lock(mLock); 453 flush_l(); 454} 455 456// must be called with mLock held 457void AudioTrack::flush_l() 458{ 459 ALOGV("flush"); 460 461 // clear playback marker and periodic update counter 462 mMarkerPosition = 0; 463 mMarkerReached = false; 464 mUpdatePeriod = 0; 465 466 if (!mActive) { 467 mFlushed = true; 468 mAudioTrack->flush(); 469 // Release AudioTrack callback thread in case it was waiting for new buffers 470 // in AudioTrack::obtainBuffer() 471 mCblk->cv.signal(); 472 } 473} 474 475void AudioTrack::pause() 476{ 477 ALOGV("pause"); 478 AutoMutex lock(mLock); 479 if (mActive) { 480 mActive = false; 481 mCblk->cv.signal(); 482 mAudioTrack->pause(); 483 } 484} 485 486void AudioTrack::mute(bool e) 487{ 488 mAudioTrack->mute(e); 489 mMuted = e; 490} 491 492bool AudioTrack::muted() const 493{ 494 return mMuted; 495} 496 497status_t AudioTrack::setVolume(float left, float right) 498{ 499 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 500 return BAD_VALUE; 501 } 502 503 AutoMutex lock(mLock); 504 mVolume[LEFT] = left; 505 mVolume[RIGHT] = right; 506 507 mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 508 509 return NO_ERROR; 510} 511 512void AudioTrack::getVolume(float* left, float* right) const 513{ 514 if (left != NULL) { 515 *left = mVolume[LEFT]; 516 } 517 if (right != NULL) { 518 *right = mVolume[RIGHT]; 519 } 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(int rate) 545{ 546 int 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 INVALID_OPERATION; 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 int 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 int 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 int afSampleRate; 812 if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { 813 return NO_INIT; 814 } 815 int 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 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 825 ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d" 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 > (uint32_t)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 format, 868 channelMask, 869 frameCount, 870 &trackFlags, 871 sharedBuffer, 872 output, 873 tid, 874 &mSessionId, 875 &status); 876 877 if (track == 0) { 878 ALOGE("AudioFlinger could not create track, status: %d", status); 879 return status; 880 } 881 sp<IMemory> iMem = track->getCblk(); 882 if (iMem == 0) { 883 ALOGE("Could not get control block"); 884 return NO_INIT; 885 } 886 mAudioTrack = track; 887 mCblkMemory = iMem; 888 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 889 mCblk = cblk; 890 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 891 if (trackFlags & IAudioFlinger::TRACK_FAST) { 892 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", cblk->frameCount); 893 } else { 894 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", cblk->frameCount); 895 // once denied, do not request again if IAudioTrack is re-created 896 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 897 mFlags = flags; 898 } 899 if (sharedBuffer == 0) { 900 mNotificationFramesAct = cblk->frameCount/2; 901 } 902 } 903 if (sharedBuffer == 0) { 904 cblk->buffers = (char*)cblk + sizeof(audio_track_cblk_t); 905 } else { 906 cblk->buffers = sharedBuffer->pointer(); 907 // Force buffer full condition as data is already present in shared memory 908 cblk->stepUserOut(cblk->frameCount); 909 } 910 911 cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 912 uint16_t(mVolume[LEFT] * 0x1000)); 913 cblk->setSendLevel(mSendLevel); 914 mAudioTrack->attachAuxEffect(mAuxEffectId); 915 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 916 cblk->waitTimeMs = 0; 917 mRemainingFrames = mNotificationFramesAct; 918 // FIXME don't believe this lie 919 mLatency = afLatency + (1000*cblk->frameCount) / sampleRate; 920 // If IAudioTrack is re-created, don't let the requested frameCount 921 // decrease. This can confuse clients that cache frameCount(). 922 if (cblk->frameCount > mFrameCount) { 923 mFrameCount = cblk->frameCount; 924 } 925 return NO_ERROR; 926} 927 928status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 929{ 930 AutoMutex lock(mLock); 931 bool active; 932 status_t result = NO_ERROR; 933 audio_track_cblk_t* cblk = mCblk; 934 uint32_t framesReq = audioBuffer->frameCount; 935 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 936 937 audioBuffer->frameCount = 0; 938 audioBuffer->size = 0; 939 940 uint32_t framesAvail = cblk->framesAvailableOut(); 941 942 cblk->lock.lock(); 943 if (cblk->flags & CBLK_INVALID) { 944 goto create_new_track; 945 } 946 cblk->lock.unlock(); 947 948 if (framesAvail == 0) { 949 cblk->lock.lock(); 950 goto start_loop_here; 951 while (framesAvail == 0) { 952 active = mActive; 953 if (CC_UNLIKELY(!active)) { 954 ALOGV("Not active and NO_MORE_BUFFERS"); 955 cblk->lock.unlock(); 956 return NO_MORE_BUFFERS; 957 } 958 if (CC_UNLIKELY(!waitCount)) { 959 cblk->lock.unlock(); 960 return WOULD_BLOCK; 961 } 962 if (!(cblk->flags & CBLK_INVALID)) { 963 mLock.unlock(); 964 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 965 cblk->lock.unlock(); 966 mLock.lock(); 967 if (!mActive) { 968 return status_t(STOPPED); 969 } 970 cblk->lock.lock(); 971 } 972 973 if (cblk->flags & CBLK_INVALID) { 974 goto create_new_track; 975 } 976 if (CC_UNLIKELY(result != NO_ERROR)) { 977 cblk->waitTimeMs += waitTimeMs; 978 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 979 // timing out when a loop has been set and we have already written upto loop end 980 // is a normal condition: no need to wake AudioFlinger up. 981 if (cblk->user < cblk->loopEnd) { 982 ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " 983 "server=%08x", this, cblk->mName, cblk->user, cblk->server); 984 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 985 cblk->lock.unlock(); 986 result = mAudioTrack->start(); 987 cblk->lock.lock(); 988 if (result == DEAD_OBJECT) { 989 android_atomic_or(CBLK_INVALID, &cblk->flags); 990create_new_track: 991 audio_track_cblk_t* temp = cblk; 992 result = restoreTrack_l(temp, false); 993 cblk = temp; 994 } 995 if (result != NO_ERROR) { 996 ALOGW("obtainBuffer create Track error %d", result); 997 cblk->lock.unlock(); 998 return result; 999 } 1000 } 1001 cblk->waitTimeMs = 0; 1002 } 1003 1004 if (--waitCount == 0) { 1005 cblk->lock.unlock(); 1006 return TIMED_OUT; 1007 } 1008 } 1009 // read the server count again 1010 start_loop_here: 1011 framesAvail = cblk->framesAvailableOut_l(); 1012 } 1013 cblk->lock.unlock(); 1014 } 1015 1016 cblk->waitTimeMs = 0; 1017 1018 if (framesReq > framesAvail) { 1019 framesReq = framesAvail; 1020 } 1021 1022 uint32_t u = cblk->user; 1023 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 1024 1025 if (framesReq > bufferEnd - u) { 1026 framesReq = bufferEnd - u; 1027 } 1028 1029 audioBuffer->frameCount = framesReq; 1030 audioBuffer->size = framesReq * cblk->frameSize; 1031 audioBuffer->raw = (int8_t *)cblk->buffer(u); 1032 active = mActive; 1033 return active ? status_t(NO_ERROR) : status_t(STOPPED); 1034} 1035 1036void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1037{ 1038 AutoMutex lock(mLock); 1039 audio_track_cblk_t* cblk = mCblk; 1040 cblk->stepUserOut(audioBuffer->frameCount); 1041 if (audioBuffer->frameCount > 0) { 1042 // restart track if it was disabled by audioflinger due to previous underrun 1043 if (mActive && (cblk->flags & CBLK_DISABLED)) { 1044 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1045 ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName); 1046 mAudioTrack->start(); 1047 } 1048 } 1049} 1050 1051// ------------------------------------------------------------------------- 1052 1053ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1054{ 1055 1056 if (mSharedBuffer != 0) return INVALID_OPERATION; 1057 if (mIsTimed) return INVALID_OPERATION; 1058 1059 if (ssize_t(userSize) < 0) { 1060 // Sanity-check: user is most-likely passing an error code, and it would 1061 // make the return value ambiguous (actualSize vs error). 1062 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 1063 buffer, userSize, userSize); 1064 return BAD_VALUE; 1065 } 1066 1067 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 1068 1069 if (userSize == 0) { 1070 return 0; 1071 } 1072 1073 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1074 // while we are accessing the cblk 1075 mLock.lock(); 1076 sp<IAudioTrack> audioTrack = mAudioTrack; 1077 sp<IMemory> iMem = mCblkMemory; 1078 mLock.unlock(); 1079 1080 ssize_t written = 0; 1081 const int8_t *src = (const int8_t *)buffer; 1082 Buffer audioBuffer; 1083 size_t frameSz = frameSize(); 1084 1085 do { 1086 audioBuffer.frameCount = userSize/frameSz; 1087 1088 status_t err = obtainBuffer(&audioBuffer, -1); 1089 if (err < 0) { 1090 // out of buffers, return #bytes written 1091 if (err == status_t(NO_MORE_BUFFERS)) 1092 break; 1093 return ssize_t(err); 1094 } 1095 1096 size_t toWrite; 1097 1098 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1099 // Divide capacity by 2 to take expansion into account 1100 toWrite = audioBuffer.size>>1; 1101 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1102 } else { 1103 toWrite = audioBuffer.size; 1104 memcpy(audioBuffer.i8, src, toWrite); 1105 src += toWrite; 1106 } 1107 userSize -= toWrite; 1108 written += toWrite; 1109 1110 releaseBuffer(&audioBuffer); 1111 } while (userSize >= frameSz); 1112 1113 return written; 1114} 1115 1116// ------------------------------------------------------------------------- 1117 1118TimedAudioTrack::TimedAudioTrack() { 1119 mIsTimed = true; 1120} 1121 1122status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1123{ 1124 status_t result = UNKNOWN_ERROR; 1125 1126 // If the track is not invalid already, try to allocate a buffer. alloc 1127 // fails indicating that the server is dead, flag the track as invalid so 1128 // we can attempt to restore in just a bit. 1129 audio_track_cblk_t* cblk = mCblk; 1130 if (!(cblk->flags & CBLK_INVALID)) { 1131 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1132 if (result == DEAD_OBJECT) { 1133 android_atomic_or(CBLK_INVALID, &cblk->flags); 1134 } 1135 } 1136 1137 // If the track is invalid at this point, attempt to restore it. and try the 1138 // allocation one more time. 1139 if (cblk->flags & CBLK_INVALID) { 1140 cblk->lock.lock(); 1141 audio_track_cblk_t* temp = cblk; 1142 result = restoreTrack_l(temp, false); 1143 cblk = temp; 1144 cblk->lock.unlock(); 1145 1146 if (result == OK) 1147 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1148 } 1149 1150 return result; 1151} 1152 1153status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1154 int64_t pts) 1155{ 1156 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1157 { 1158 AutoMutex lock(mLock); 1159 audio_track_cblk_t* cblk = mCblk; 1160 // restart track if it was disabled by audioflinger due to previous underrun 1161 if (buffer->size() != 0 && status == NO_ERROR && 1162 mActive && (cblk->flags & CBLK_DISABLED)) { 1163 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1164 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1165 mAudioTrack->start(); 1166 } 1167 } 1168 return status; 1169} 1170 1171status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1172 TargetTimeline target) 1173{ 1174 return mAudioTrack->setMediaTimeTransform(xform, target); 1175} 1176 1177// ------------------------------------------------------------------------- 1178 1179bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1180{ 1181 Buffer audioBuffer; 1182 uint32_t frames; 1183 size_t writtenSize; 1184 1185 mLock.lock(); 1186 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1187 // while we are accessing the cblk 1188 sp<IAudioTrack> audioTrack = mAudioTrack; 1189 sp<IMemory> iMem = mCblkMemory; 1190 audio_track_cblk_t* cblk = mCblk; 1191 bool active = mActive; 1192 mLock.unlock(); 1193 1194 // Manage underrun callback 1195 if (active && (cblk->framesAvailableOut() == cblk->frameCount)) { 1196 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1197 if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { 1198 mCbf(EVENT_UNDERRUN, mUserData, 0); 1199 if (cblk->server == cblk->frameCount) { 1200 mCbf(EVENT_BUFFER_END, mUserData, 0); 1201 } 1202 if (mSharedBuffer != 0) return false; 1203 } 1204 } 1205 1206 // Manage loop end callback 1207 while (mLoopCount > cblk->loopCount) { 1208 int loopCount = -1; 1209 mLoopCount--; 1210 if (mLoopCount >= 0) loopCount = mLoopCount; 1211 1212 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1213 } 1214 1215 // Manage marker callback 1216 if (!mMarkerReached && (mMarkerPosition > 0)) { 1217 if (cblk->server >= mMarkerPosition) { 1218 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1219 mMarkerReached = true; 1220 } 1221 } 1222 1223 // Manage new position callback 1224 if (mUpdatePeriod > 0) { 1225 while (cblk->server >= mNewPosition) { 1226 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1227 mNewPosition += mUpdatePeriod; 1228 } 1229 } 1230 1231 // If Shared buffer is used, no data is requested from client. 1232 if (mSharedBuffer != 0) { 1233 frames = 0; 1234 } else { 1235 frames = mRemainingFrames; 1236 } 1237 1238 // See description of waitCount parameter at declaration of obtainBuffer(). 1239 // The logic below prevents us from being stuck below at obtainBuffer() 1240 // not being able to handle timed events (position, markers, loops). 1241 int32_t waitCount = -1; 1242 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1243 waitCount = 1; 1244 } 1245 1246 do { 1247 1248 audioBuffer.frameCount = frames; 1249 1250 status_t err = obtainBuffer(&audioBuffer, waitCount); 1251 if (err < NO_ERROR) { 1252 if (err != TIMED_OUT) { 1253 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), 1254 "Error obtaining an audio buffer, giving up."); 1255 return false; 1256 } 1257 break; 1258 } 1259 if (err == status_t(STOPPED)) return false; 1260 1261 // Divide buffer size by 2 to take into account the expansion 1262 // due to 8 to 16 bit conversion: the callback must fill only half 1263 // of the destination buffer 1264 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1265 audioBuffer.size >>= 1; 1266 } 1267 1268 size_t reqSize = audioBuffer.size; 1269 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1270 writtenSize = audioBuffer.size; 1271 1272 // Sanity check on returned size 1273 if (ssize_t(writtenSize) <= 0) { 1274 // The callback is done filling buffers 1275 // Keep this thread going to handle timed events and 1276 // still try to get more data in intervals of WAIT_PERIOD_MS 1277 // but don't just loop and block the CPU, so wait 1278 usleep(WAIT_PERIOD_MS*1000); 1279 break; 1280 } 1281 1282 if (writtenSize > reqSize) writtenSize = reqSize; 1283 1284 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1285 // 8 to 16 bit conversion, note that source and destination are the same address 1286 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1287 writtenSize <<= 1; 1288 } 1289 1290 audioBuffer.size = writtenSize; 1291 // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for 1292 // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of 1293 // 16 bit. 1294 audioBuffer.frameCount = writtenSize/cblk->frameSize; 1295 1296 frames -= audioBuffer.frameCount; 1297 1298 releaseBuffer(&audioBuffer); 1299 } 1300 while (frames); 1301 1302 if (frames == 0) { 1303 mRemainingFrames = mNotificationFramesAct; 1304 } else { 1305 mRemainingFrames = frames; 1306 } 1307 return true; 1308} 1309 1310// must be called with mLock and refCblk.lock held. Callers must also hold strong references on 1311// the IAudioTrack and IMemory in case they are recreated here. 1312// If the IAudioTrack is successfully restored, the refCblk pointer is updated 1313// FIXME Don't depend on caller to hold strong references. 1314status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart) 1315{ 1316 status_t result; 1317 1318 audio_track_cblk_t* cblk = refCblk; 1319 audio_track_cblk_t* newCblk = cblk; 1320 if (!(android_atomic_or(CBLK_RESTORING, &cblk->flags) & CBLK_RESTORING)) { 1321 ALOGW("dead IAudioTrack, creating a new one from %s TID %d", 1322 fromStart ? "start()" : "obtainBuffer()", gettid()); 1323 1324 // signal old cblk condition so that other threads waiting for available buffers stop 1325 // waiting now 1326 cblk->cv.broadcast(); 1327 cblk->lock.unlock(); 1328 1329 // refresh the audio configuration cache in this process to make sure we get new 1330 // output parameters in getOutput_l() and createTrack_l() 1331 AudioSystem::clearAudioConfigCache(); 1332 1333 // if the new IAudioTrack is created, createTrack_l() will modify the 1334 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1335 // It will also delete the strong references on previous IAudioTrack and IMemory 1336 result = createTrack_l(mStreamType, 1337 cblk->sampleRate, 1338 mFormat, 1339 mChannelMask, 1340 mFrameCount, 1341 mFlags, 1342 mSharedBuffer, 1343 getOutput_l()); 1344 1345 if (result == NO_ERROR) { 1346 uint32_t user = cblk->user; 1347 uint32_t server = cblk->server; 1348 // restore write index and set other indexes to reflect empty buffer status 1349 newCblk = mCblk; 1350 newCblk->user = user; 1351 newCblk->server = user; 1352 newCblk->userBase = user; 1353 newCblk->serverBase = user; 1354 // restore loop: this is not guaranteed to succeed if new frame count is not 1355 // compatible with loop length 1356 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1357 if (!fromStart) { 1358 newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1359 // Make sure that a client relying on callback events indicating underrun or 1360 // the actual amount of audio frames played (e.g SoundPool) receives them. 1361 if (mSharedBuffer == 0) { 1362 uint32_t frames = 0; 1363 if (user > server) { 1364 frames = ((user - server) > newCblk->frameCount) ? 1365 newCblk->frameCount : (user - server); 1366 memset(newCblk->buffers, 0, frames * newCblk->frameSize); 1367 } 1368 // restart playback even if buffer is not completely filled. 1369 android_atomic_or(CBLK_FORCEREADY, &newCblk->flags); 1370 // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to 1371 // the client 1372 newCblk->stepUserOut(frames); 1373 } 1374 } 1375 if (mSharedBuffer != 0) { 1376 newCblk->stepUserOut(newCblk->frameCount); 1377 } 1378 if (mActive) { 1379 result = mAudioTrack->start(); 1380 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1381 } 1382 if (fromStart && result == NO_ERROR) { 1383 mNewPosition = newCblk->server + mUpdatePeriod; 1384 } 1385 } 1386 if (result != NO_ERROR) { 1387 android_atomic_and(~CBLK_RESTORING, &cblk->flags); 1388 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1389 } 1390 mRestoreStatus = result; 1391 // signal old cblk condition for other threads waiting for restore completion 1392 android_atomic_or(CBLK_RESTORED, &cblk->flags); 1393 cblk->cv.broadcast(); 1394 } else { 1395 bool haveLogged = false; 1396 for (;;) { 1397 if (cblk->flags & CBLK_RESTORED) { 1398 ALOGW("dead IAudioTrack restored"); 1399 result = mRestoreStatus; 1400 cblk->lock.unlock(); 1401 break; 1402 } 1403 if (!haveLogged) { 1404 ALOGW("dead IAudioTrack, waiting for a new one"); 1405 haveLogged = true; 1406 } 1407 mLock.unlock(); 1408 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1409 cblk->lock.unlock(); 1410 mLock.lock(); 1411 if (result != NO_ERROR) { 1412 ALOGW("timed out"); 1413 break; 1414 } 1415 cblk->lock.lock(); 1416 } 1417 } 1418 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1419 result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); 1420 1421 if (result == NO_ERROR) { 1422 // from now on we switch to the newly created cblk 1423 refCblk = newCblk; 1424 } 1425 newCblk->lock.lock(); 1426 1427 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1428 1429 return result; 1430} 1431 1432status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1433{ 1434 1435 const size_t SIZE = 256; 1436 char buffer[SIZE]; 1437 String8 result; 1438 1439 audio_track_cblk_t* cblk = mCblk; 1440 result.append(" AudioTrack::dump\n"); 1441 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1442 mVolume[0], mVolume[1]); 1443 result.append(buffer); 1444 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1445 mChannelCount, cblk->frameCount); 1446 result.append(buffer); 1447 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", 1448 (cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted); 1449 result.append(buffer); 1450 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1451 result.append(buffer); 1452 ::write(fd, result.string(), result.size()); 1453 return NO_ERROR; 1454} 1455 1456// ========================================================================= 1457 1458AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1459 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) 1460{ 1461} 1462 1463AudioTrack::AudioTrackThread::~AudioTrackThread() 1464{ 1465} 1466 1467bool AudioTrack::AudioTrackThread::threadLoop() 1468{ 1469 { 1470 AutoMutex _l(mMyLock); 1471 if (mPaused) { 1472 mMyCond.wait(mMyLock); 1473 // caller will check for exitPending() 1474 return true; 1475 } 1476 } 1477 if (!mReceiver.processAudioBuffer(this)) { 1478 pause(); 1479 } 1480 return true; 1481} 1482 1483void AudioTrack::AudioTrackThread::requestExit() 1484{ 1485 // must be in this order to avoid a race condition 1486 Thread::requestExit(); 1487 resume(); 1488} 1489 1490void AudioTrack::AudioTrackThread::pause() 1491{ 1492 AutoMutex _l(mMyLock); 1493 mPaused = true; 1494} 1495 1496void AudioTrack::AudioTrackThread::resume() 1497{ 1498 AutoMutex _l(mMyLock); 1499 if (mPaused) { 1500 mPaused = false; 1501 mMyCond.signal(); 1502 } 1503} 1504 1505// ========================================================================= 1506 1507 1508audio_track_cblk_t::audio_track_cblk_t() 1509 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1510 userBase(0), serverBase(0), buffers(NULL), frameCount(0), 1511 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), 1512 mSendLevel(0), flags(0) 1513{ 1514} 1515 1516uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount, bool isOut) 1517{ 1518 ALOGV("stepuser %08x %08x %d", user, server, frameCount); 1519 1520 uint32_t u = user; 1521 u += frameCount; 1522 // Ensure that user is never ahead of server for AudioRecord 1523 if (isOut) { 1524 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1525 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1526 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1527 } 1528 } else if (u > server) { 1529 ALOGW("stepUser occurred after track reset"); 1530 u = server; 1531 } 1532 1533 uint32_t fc = this->frameCount; 1534 if (u >= fc) { 1535 // common case, user didn't just wrap 1536 if (u - fc >= userBase ) { 1537 userBase += fc; 1538 } 1539 } else if (u >= userBase + fc) { 1540 // user just wrapped 1541 userBase += fc; 1542 } 1543 1544 user = u; 1545 1546 // Clear flow control error condition as new data has been written/read to/from buffer. 1547 if (flags & CBLK_UNDERRUN) { 1548 android_atomic_and(~CBLK_UNDERRUN, &flags); 1549 } 1550 1551 return u; 1552} 1553 1554bool audio_track_cblk_t::stepServer(uint32_t frameCount, bool isOut) 1555{ 1556 ALOGV("stepserver %08x %08x %d", user, server, frameCount); 1557 1558 if (!tryLock()) { 1559 ALOGW("stepServer() could not lock cblk"); 1560 return false; 1561 } 1562 1563 uint32_t s = server; 1564 bool flushed = (s == user); 1565 1566 s += frameCount; 1567 if (isOut) { 1568 // Mark that we have read the first buffer so that next time stepUser() is called 1569 // we switch to normal obtainBuffer() timeout period 1570 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1571 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1572 } 1573 // It is possible that we receive a flush() 1574 // while the mixer is processing a block: in this case, 1575 // stepServer() is called After the flush() has reset u & s and 1576 // we have s > u 1577 if (flushed) { 1578 ALOGW("stepServer occurred after track reset"); 1579 s = user; 1580 } 1581 } 1582 1583 if (s >= loopEnd) { 1584 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1585 s = loopStart; 1586 if (--loopCount == 0) { 1587 loopEnd = UINT_MAX; 1588 loopStart = UINT_MAX; 1589 } 1590 } 1591 1592 uint32_t fc = this->frameCount; 1593 if (s >= fc) { 1594 // common case, server didn't just wrap 1595 if (s - fc >= serverBase ) { 1596 serverBase += fc; 1597 } 1598 } else if (s >= serverBase + fc) { 1599 // server just wrapped 1600 serverBase += fc; 1601 } 1602 1603 server = s; 1604 1605 if (!(flags & CBLK_INVALID)) { 1606 cv.signal(); 1607 } 1608 lock.unlock(); 1609 return true; 1610} 1611 1612void* audio_track_cblk_t::buffer(uint32_t offset) const 1613{ 1614 return (int8_t *)buffers + (offset - userBase) * frameSize; 1615} 1616 1617uint32_t audio_track_cblk_t::framesAvailable(bool isOut) 1618{ 1619 Mutex::Autolock _l(lock); 1620 return framesAvailable_l(isOut); 1621} 1622 1623uint32_t audio_track_cblk_t::framesAvailable_l(bool isOut) 1624{ 1625 uint32_t u = user; 1626 uint32_t s = server; 1627 1628 if (isOut) { 1629 uint32_t limit = (s < loopStart) ? s : loopStart; 1630 return limit + frameCount - u; 1631 } else { 1632 return frameCount + u - s; 1633 } 1634} 1635 1636uint32_t audio_track_cblk_t::framesReady(bool isOut) 1637{ 1638 uint32_t u = user; 1639 uint32_t s = server; 1640 1641 if (isOut) { 1642 if (u < loopEnd) { 1643 return u - s; 1644 } else { 1645 // do not block on mutex shared with client on AudioFlinger side 1646 if (!tryLock()) { 1647 ALOGW("framesReady() could not lock cblk"); 1648 return 0; 1649 } 1650 uint32_t frames = UINT_MAX; 1651 if (loopCount >= 0) { 1652 frames = (loopEnd - loopStart)*loopCount + u - s; 1653 } 1654 lock.unlock(); 1655 return frames; 1656 } 1657 } else { 1658 return s - u; 1659 } 1660} 1661 1662bool audio_track_cblk_t::tryLock() 1663{ 1664 // the code below simulates lock-with-timeout 1665 // we MUST do this to protect the AudioFlinger server 1666 // as this lock is shared with the client. 1667 status_t err; 1668 1669 err = lock.tryLock(); 1670 if (err == -EBUSY) { // just wait a bit 1671 usleep(1000); 1672 err = lock.tryLock(); 1673 } 1674 if (err != NO_ERROR) { 1675 // probably, the client just died. 1676 return false; 1677 } 1678 return true; 1679} 1680 1681// ------------------------------------------------------------------------- 1682 1683}; // namespace android 1684