Threads.cpp revision ab7d72f0804fbb7e91ad9d2a16f826d97e20e5d0
1/* 2** 3** Copyright 2012, 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_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include <math.h> 24#include <fcntl.h> 25#include <sys/stat.h> 26#include <cutils/properties.h> 27#include <cutils/compiler.h> 28#include <utils/Log.h> 29#include <utils/Trace.h> 30 31#include <private/media/AudioTrackShared.h> 32#include <hardware/audio.h> 33#include <audio_effects/effect_ns.h> 34#include <audio_effects/effect_aec.h> 35#include <audio_utils/primitives.h> 36 37// NBAIO implementations 38#include <media/nbaio/AudioStreamOutSink.h> 39#include <media/nbaio/MonoPipe.h> 40#include <media/nbaio/MonoPipeReader.h> 41#include <media/nbaio/Pipe.h> 42#include <media/nbaio/PipeReader.h> 43#include <media/nbaio/SourceAudioBufferProvider.h> 44 45#include <powermanager/PowerManager.h> 46 47#include <common_time/cc_helper.h> 48#include <common_time/local_clock.h> 49 50#include "AudioFlinger.h" 51#include "AudioMixer.h" 52#include "FastMixer.h" 53#include "ServiceUtilities.h" 54#include "SchedulingPolicyService.h" 55 56#undef ADD_BATTERY_DATA 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait for setParameters to complete 102static const nsecs_t kSetParametersTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal mix buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalMixBufferSizeMs = 20; 111// maximum normal mix buffer size 112static const uint32_t kMaxNormalMixBufferSizeMs = 24; 113 114// Whether to use fast mixer 115static const enum { 116 FastMixer_Never, // never initialize or use: for debugging only 117 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 118 // normal mixer multiplier is 1 119 FastMixer_Static, // initialize if needed, then use all the time if initialized, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 // FIXME for FastMixer_Dynamic: 124 // Supporting this option will require fixing HALs that can't handle large writes. 125 // For example, one HAL implementation returns an error from a large write, 126 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 127 // We could either fix the HAL implementations, or provide a wrapper that breaks 128 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 129} kUseFastMixer = FastMixer_Static; 130 131// Priorities for requestPriority 132static const int kPriorityAudioApp = 2; 133static const int kPriorityFastMixer = 3; 134 135// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 136// for the track. The client then sub-divides this into smaller buffers for its use. 137// Currently the client uses double-buffering by default, but doesn't tell us about that. 138// So for now we just assume that client is double-buffered. 139// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 140// N-buffering, so AudioFlinger could allocate the right amount of memory. 141// See the client's minBufCount and mNotificationFramesAct calculations for details. 142static const int kFastTrackMultiplier = 2; 143 144// ---------------------------------------------------------------------------- 145 146#ifdef ADD_BATTERY_DATA 147// To collect the amplifier usage 148static void addBatteryData(uint32_t params) { 149 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 150 if (service == NULL) { 151 // it already logged 152 return; 153 } 154 155 service->addBatteryData(params); 156} 157#endif 158 159 160// ---------------------------------------------------------------------------- 161// CPU Stats 162// ---------------------------------------------------------------------------- 163 164class CpuStats { 165public: 166 CpuStats(); 167 void sample(const String8 &title); 168#ifdef DEBUG_CPU_USAGE 169private: 170 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 171 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 172 173 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 174 175 int mCpuNum; // thread's current CPU number 176 int mCpukHz; // frequency of thread's current CPU in kHz 177#endif 178}; 179 180CpuStats::CpuStats() 181#ifdef DEBUG_CPU_USAGE 182 : mCpuNum(-1), mCpukHz(-1) 183#endif 184{ 185} 186 187void CpuStats::sample(const String8 &title) { 188#ifdef DEBUG_CPU_USAGE 189 // get current thread's delta CPU time in wall clock ns 190 double wcNs; 191 bool valid = mCpuUsage.sampleAndEnable(wcNs); 192 193 // record sample for wall clock statistics 194 if (valid) { 195 mWcStats.sample(wcNs); 196 } 197 198 // get the current CPU number 199 int cpuNum = sched_getcpu(); 200 201 // get the current CPU frequency in kHz 202 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 203 204 // check if either CPU number or frequency changed 205 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 206 mCpuNum = cpuNum; 207 mCpukHz = cpukHz; 208 // ignore sample for purposes of cycles 209 valid = false; 210 } 211 212 // if no change in CPU number or frequency, then record sample for cycle statistics 213 if (valid && mCpukHz > 0) { 214 double cycles = wcNs * cpukHz * 0.000001; 215 mHzStats.sample(cycles); 216 } 217 218 unsigned n = mWcStats.n(); 219 // mCpuUsage.elapsed() is expensive, so don't call it every loop 220 if ((n & 127) == 1) { 221 long long elapsed = mCpuUsage.elapsed(); 222 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 223 double perLoop = elapsed / (double) n; 224 double perLoop100 = perLoop * 0.01; 225 double perLoop1k = perLoop * 0.001; 226 double mean = mWcStats.mean(); 227 double stddev = mWcStats.stddev(); 228 double minimum = mWcStats.minimum(); 229 double maximum = mWcStats.maximum(); 230 double meanCycles = mHzStats.mean(); 231 double stddevCycles = mHzStats.stddev(); 232 double minCycles = mHzStats.minimum(); 233 double maxCycles = mHzStats.maximum(); 234 mCpuUsage.resetElapsed(); 235 mWcStats.reset(); 236 mHzStats.reset(); 237 ALOGD("CPU usage for %s over past %.1f secs\n" 238 " (%u mixer loops at %.1f mean ms per loop):\n" 239 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 240 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 241 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 242 title.string(), 243 elapsed * .000000001, n, perLoop * .000001, 244 mean * .001, 245 stddev * .001, 246 minimum * .001, 247 maximum * .001, 248 mean / perLoop100, 249 stddev / perLoop100, 250 minimum / perLoop100, 251 maximum / perLoop100, 252 meanCycles / perLoop1k, 253 stddevCycles / perLoop1k, 254 minCycles / perLoop1k, 255 maxCycles / perLoop1k); 256 257 } 258 } 259#endif 260}; 261 262// ---------------------------------------------------------------------------- 263// ThreadBase 264// ---------------------------------------------------------------------------- 265 266AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 267 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 268 : Thread(false /*canCallJava*/), 269 mType(type), 270 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 271 // mChannelMask 272 mChannelCount(0), 273 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 mParamCond.broadcast(); 285 // do not lock the mutex in destructor 286 releaseWakeLock_l(); 287 if (mPowerManager != 0) { 288 sp<IBinder> binder = mPowerManager->asBinder(); 289 binder->unlinkToDeath(mDeathRecipient); 290 } 291} 292 293void AudioFlinger::ThreadBase::exit() 294{ 295 ALOGV("ThreadBase::exit"); 296 // do any cleanup required for exit to succeed 297 preExit(); 298 { 299 // This lock prevents the following race in thread (uniprocessor for illustration): 300 // if (!exitPending()) { 301 // // context switch from here to exit() 302 // // exit() calls requestExit(), what exitPending() observes 303 // // exit() calls signal(), which is dropped since no waiters 304 // // context switch back from exit() to here 305 // mWaitWorkCV.wait(...); 306 // // now thread is hung 307 // } 308 AutoMutex lock(mLock); 309 requestExit(); 310 mWaitWorkCV.broadcast(); 311 } 312 // When Thread::requestExitAndWait is made virtual and this method is renamed to 313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 314 requestExitAndWait(); 315} 316 317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 318{ 319 status_t status; 320 321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 322 Mutex::Autolock _l(mLock); 323 324 mNewParameters.add(keyValuePairs); 325 mWaitWorkCV.signal(); 326 // wait condition with timeout in case the thread loop has exited 327 // before the request could be processed 328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 329 status = mParamStatus; 330 mWaitWorkCV.signal(); 331 } else { 332 status = TIMED_OUT; 333 } 334 return status; 335} 336 337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 338{ 339 Mutex::Autolock _l(mLock); 340 sendIoConfigEvent_l(event, param); 341} 342 343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 345{ 346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 349 param); 350 mWaitWorkCV.signal(); 351} 352 353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 355{ 356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 359 mConfigEvents.size(), pid, tid, prio); 360 mWaitWorkCV.signal(); 361} 362 363void AudioFlinger::ThreadBase::processConfigEvents() 364{ 365 mLock.lock(); 366 while (!mConfigEvents.isEmpty()) { 367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 368 ConfigEvent *event = mConfigEvents[0]; 369 mConfigEvents.removeAt(0); 370 // release mLock before locking AudioFlinger mLock: lock order is always 371 // AudioFlinger then ThreadBase to avoid cross deadlock 372 mLock.unlock(); 373 switch(event->type()) { 374 case CFG_EVENT_PRIO: { 375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 376 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 377 if (err != 0) { 378 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 379 "error %d", 380 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 381 } 382 } break; 383 case CFG_EVENT_IO: { 384 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 385 mAudioFlinger->mLock.lock(); 386 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 387 mAudioFlinger->mLock.unlock(); 388 } break; 389 default: 390 ALOGE("processConfigEvents() unknown event type %d", event->type()); 391 break; 392 } 393 delete event; 394 mLock.lock(); 395 } 396 mLock.unlock(); 397} 398 399void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 400{ 401 const size_t SIZE = 256; 402 char buffer[SIZE]; 403 String8 result; 404 405 bool locked = AudioFlinger::dumpTryLock(mLock); 406 if (!locked) { 407 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 408 write(fd, buffer, strlen(buffer)); 409 } 410 411 snprintf(buffer, SIZE, "io handle: %d\n", mId); 412 result.append(buffer); 413 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 430 result.append(buffer); 431 432 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 433 result.append(buffer); 434 result.append(" Index Command"); 435 for (size_t i = 0; i < mNewParameters.size(); ++i) { 436 snprintf(buffer, SIZE, "\n %02d ", i); 437 result.append(buffer); 438 result.append(mNewParameters[i]); 439 } 440 441 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 442 result.append(buffer); 443 for (size_t i = 0; i < mConfigEvents.size(); i++) { 444 mConfigEvents[i]->dump(buffer, SIZE); 445 result.append(buffer); 446 } 447 result.append("\n"); 448 449 write(fd, result.string(), result.size()); 450 451 if (locked) { 452 mLock.unlock(); 453 } 454} 455 456void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 457{ 458 const size_t SIZE = 256; 459 char buffer[SIZE]; 460 String8 result; 461 462 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 463 write(fd, buffer, strlen(buffer)); 464 465 for (size_t i = 0; i < mEffectChains.size(); ++i) { 466 sp<EffectChain> chain = mEffectChains[i]; 467 if (chain != 0) { 468 chain->dump(fd, args); 469 } 470 } 471} 472 473void AudioFlinger::ThreadBase::acquireWakeLock() 474{ 475 Mutex::Autolock _l(mLock); 476 acquireWakeLock_l(); 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock_l() 480{ 481 if (mPowerManager == 0) { 482 // use checkService() to avoid blocking if power service is not up yet 483 sp<IBinder> binder = 484 defaultServiceManager()->checkService(String16("power")); 485 if (binder == 0) { 486 ALOGW("Thread %s cannot connect to the power manager service", mName); 487 } else { 488 mPowerManager = interface_cast<IPowerManager>(binder); 489 binder->linkToDeath(mDeathRecipient); 490 } 491 } 492 if (mPowerManager != 0) { 493 sp<IBinder> binder = new BBinder(); 494 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 495 binder, 496 String16(mName)); 497 if (status == NO_ERROR) { 498 mWakeLockToken = binder; 499 } 500 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 501 } 502} 503 504void AudioFlinger::ThreadBase::releaseWakeLock() 505{ 506 Mutex::Autolock _l(mLock); 507 releaseWakeLock_l(); 508} 509 510void AudioFlinger::ThreadBase::releaseWakeLock_l() 511{ 512 if (mWakeLockToken != 0) { 513 ALOGV("releaseWakeLock_l() %s", mName); 514 if (mPowerManager != 0) { 515 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 516 } 517 mWakeLockToken.clear(); 518 } 519} 520 521void AudioFlinger::ThreadBase::clearPowerManager() 522{ 523 Mutex::Autolock _l(mLock); 524 releaseWakeLock_l(); 525 mPowerManager.clear(); 526} 527 528void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 529{ 530 sp<ThreadBase> thread = mThread.promote(); 531 if (thread != 0) { 532 thread->clearPowerManager(); 533 } 534 ALOGW("power manager service died !!!"); 535} 536 537void AudioFlinger::ThreadBase::setEffectSuspended( 538 const effect_uuid_t *type, bool suspend, int sessionId) 539{ 540 Mutex::Autolock _l(mLock); 541 setEffectSuspended_l(type, suspend, sessionId); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended_l( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 sp<EffectChain> chain = getEffectChain_l(sessionId); 548 if (chain != 0) { 549 if (type != NULL) { 550 chain->setEffectSuspended_l(type, suspend); 551 } else { 552 chain->setEffectSuspendedAll_l(suspend); 553 } 554 } 555 556 updateSuspendedSessions_l(type, suspend, sessionId); 557} 558 559void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 560{ 561 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 562 if (index < 0) { 563 return; 564 } 565 566 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 567 mSuspendedSessions.valueAt(index); 568 569 for (size_t i = 0; i < sessionEffects.size(); i++) { 570 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 571 for (int j = 0; j < desc->mRefCount; j++) { 572 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 573 chain->setEffectSuspendedAll_l(true); 574 } else { 575 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 576 desc->mType.timeLow); 577 chain->setEffectSuspended_l(&desc->mType, true); 578 } 579 } 580 } 581} 582 583void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId) 586{ 587 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 588 589 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 590 591 if (suspend) { 592 if (index >= 0) { 593 sessionEffects = mSuspendedSessions.valueAt(index); 594 } else { 595 mSuspendedSessions.add(sessionId, sessionEffects); 596 } 597 } else { 598 if (index < 0) { 599 return; 600 } 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } 603 604 605 int key = EffectChain::kKeyForSuspendAll; 606 if (type != NULL) { 607 key = type->timeLow; 608 } 609 index = sessionEffects.indexOfKey(key); 610 611 sp<SuspendedSessionDesc> desc; 612 if (suspend) { 613 if (index >= 0) { 614 desc = sessionEffects.valueAt(index); 615 } else { 616 desc = new SuspendedSessionDesc(); 617 if (type != NULL) { 618 desc->mType = *type; 619 } 620 sessionEffects.add(key, desc); 621 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 622 } 623 desc->mRefCount++; 624 } else { 625 if (index < 0) { 626 return; 627 } 628 desc = sessionEffects.valueAt(index); 629 if (--desc->mRefCount == 0) { 630 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 631 sessionEffects.removeItemsAt(index); 632 if (sessionEffects.isEmpty()) { 633 ALOGV("updateSuspendedSessions_l() restore removing session %d", 634 sessionId); 635 mSuspendedSessions.removeItem(sessionId); 636 } 637 } 638 } 639 if (!sessionEffects.isEmpty()) { 640 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 641 } 642} 643 644void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 645 bool enabled, 646 int sessionId) 647{ 648 Mutex::Autolock _l(mLock); 649 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 if (mType != RECORD) { 657 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 658 // another session. This gives the priority to well behaved effect control panels 659 // and applications not using global effects. 660 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 661 // global effects 662 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 663 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 664 } 665 } 666 667 sp<EffectChain> chain = getEffectChain_l(sessionId); 668 if (chain != 0) { 669 chain->checkSuspendOnEffectEnabled(effect, enabled); 670 } 671} 672 673// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 674sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 675 const sp<AudioFlinger::Client>& client, 676 const sp<IEffectClient>& effectClient, 677 int32_t priority, 678 int sessionId, 679 effect_descriptor_t *desc, 680 int *enabled, 681 status_t *status 682 ) 683{ 684 sp<EffectModule> effect; 685 sp<EffectHandle> handle; 686 status_t lStatus; 687 sp<EffectChain> chain; 688 bool chainCreated = false; 689 bool effectCreated = false; 690 bool effectRegistered = false; 691 692 lStatus = initCheck(); 693 if (lStatus != NO_ERROR) { 694 ALOGW("createEffect_l() Audio driver not initialized."); 695 goto Exit; 696 } 697 698 // Do not allow effects with session ID 0 on direct output or duplicating threads 699 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 700 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 701 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 702 desc->name, sessionId); 703 lStatus = BAD_VALUE; 704 goto Exit; 705 } 706 // Only Pre processor effects are allowed on input threads and only on input threads 707 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 708 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 709 desc->name, desc->flags, mType); 710 lStatus = BAD_VALUE; 711 goto Exit; 712 } 713 714 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 715 716 { // scope for mLock 717 Mutex::Autolock _l(mLock); 718 719 // check for existing effect chain with the requested audio session 720 chain = getEffectChain_l(sessionId); 721 if (chain == 0) { 722 // create a new chain for this session 723 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 724 chain = new EffectChain(this, sessionId); 725 addEffectChain_l(chain); 726 chain->setStrategy(getStrategyForSession_l(sessionId)); 727 chainCreated = true; 728 } else { 729 effect = chain->getEffectFromDesc_l(desc); 730 } 731 732 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 733 734 if (effect == 0) { 735 int id = mAudioFlinger->nextUniqueId(); 736 // Check CPU and memory usage 737 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 738 if (lStatus != NO_ERROR) { 739 goto Exit; 740 } 741 effectRegistered = true; 742 // create a new effect module if none present in the chain 743 effect = new EffectModule(this, chain, desc, id, sessionId); 744 lStatus = effect->status(); 745 if (lStatus != NO_ERROR) { 746 goto Exit; 747 } 748 lStatus = chain->addEffect_l(effect); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 effectCreated = true; 753 754 effect->setDevice(mOutDevice); 755 effect->setDevice(mInDevice); 756 effect->setMode(mAudioFlinger->getMode()); 757 effect->setAudioSource(mAudioSource); 758 } 759 // create effect handle and connect it to effect module 760 handle = new EffectHandle(effect, client, effectClient, priority); 761 lStatus = effect->addHandle(handle.get()); 762 if (enabled != NULL) { 763 *enabled = (int)effect->isEnabled(); 764 } 765 } 766 767Exit: 768 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 769 Mutex::Autolock _l(mLock); 770 if (effectCreated) { 771 chain->removeEffect_l(effect); 772 } 773 if (effectRegistered) { 774 AudioSystem::unregisterEffect(effect->id()); 775 } 776 if (chainCreated) { 777 removeEffectChain_l(chain); 778 } 779 handle.clear(); 780 } 781 782 if (status != NULL) { 783 *status = lStatus; 784 } 785 return handle; 786} 787 788sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 789{ 790 Mutex::Autolock _l(mLock); 791 return getEffect_l(sessionId, effectId); 792} 793 794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 795{ 796 sp<EffectChain> chain = getEffectChain_l(sessionId); 797 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 798} 799 800// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 801// PlaybackThread::mLock held 802status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 803{ 804 // check for existing effect chain with the requested audio session 805 int sessionId = effect->sessionId(); 806 sp<EffectChain> chain = getEffectChain_l(sessionId); 807 bool chainCreated = false; 808 809 if (chain == 0) { 810 // create a new chain for this session 811 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 812 chain = new EffectChain(this, sessionId); 813 addEffectChain_l(chain); 814 chain->setStrategy(getStrategyForSession_l(sessionId)); 815 chainCreated = true; 816 } 817 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 818 819 if (chain->getEffectFromId_l(effect->id()) != 0) { 820 ALOGW("addEffect_l() %p effect %s already present in chain %p", 821 this, effect->desc().name, chain.get()); 822 return BAD_VALUE; 823 } 824 825 status_t status = chain->addEffect_l(effect); 826 if (status != NO_ERROR) { 827 if (chainCreated) { 828 removeEffectChain_l(chain); 829 } 830 return status; 831 } 832 833 effect->setDevice(mOutDevice); 834 effect->setDevice(mInDevice); 835 effect->setMode(mAudioFlinger->getMode()); 836 effect->setAudioSource(mAudioSource); 837 return NO_ERROR; 838} 839 840void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 841 842 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 843 effect_descriptor_t desc = effect->desc(); 844 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 845 detachAuxEffect_l(effect->id()); 846 } 847 848 sp<EffectChain> chain = effect->chain().promote(); 849 if (chain != 0) { 850 // remove effect chain if removing last effect 851 if (chain->removeEffect_l(effect) == 0) { 852 removeEffectChain_l(chain); 853 } 854 } else { 855 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 856 } 857} 858 859void AudioFlinger::ThreadBase::lockEffectChains_l( 860 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 861{ 862 effectChains = mEffectChains; 863 for (size_t i = 0; i < mEffectChains.size(); i++) { 864 mEffectChains[i]->lock(); 865 } 866} 867 868void AudioFlinger::ThreadBase::unlockEffectChains( 869 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 870{ 871 for (size_t i = 0; i < effectChains.size(); i++) { 872 effectChains[i]->unlock(); 873 } 874} 875 876sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 877{ 878 Mutex::Autolock _l(mLock); 879 return getEffectChain_l(sessionId); 880} 881 882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 883{ 884 size_t size = mEffectChains.size(); 885 for (size_t i = 0; i < size; i++) { 886 if (mEffectChains[i]->sessionId() == sessionId) { 887 return mEffectChains[i]; 888 } 889 } 890 return 0; 891} 892 893void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 894{ 895 Mutex::Autolock _l(mLock); 896 size_t size = mEffectChains.size(); 897 for (size_t i = 0; i < size; i++) { 898 mEffectChains[i]->setMode_l(mode); 899 } 900} 901 902void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 903 EffectHandle *handle, 904 bool unpinIfLast) { 905 906 Mutex::Autolock _l(mLock); 907 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 908 // delete the effect module if removing last handle on it 909 if (effect->removeHandle(handle) == 0) { 910 if (!effect->isPinned() || unpinIfLast) { 911 removeEffect_l(effect); 912 AudioSystem::unregisterEffect(effect->id()); 913 } 914 } 915} 916 917// ---------------------------------------------------------------------------- 918// Playback 919// ---------------------------------------------------------------------------- 920 921AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 922 AudioStreamOut* output, 923 audio_io_handle_t id, 924 audio_devices_t device, 925 type_t type) 926 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 927 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 928 // mStreamTypes[] initialized in constructor body 929 mOutput(output), 930 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 931 mMixerStatus(MIXER_IDLE), 932 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 933 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 934 mScreenState(AudioFlinger::mScreenState), 935 // index 0 is reserved for normal mixer's submix 936 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 937{ 938 snprintf(mName, kNameLength, "AudioOut_%X", id); 939 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 940 941 // Assumes constructor is called by AudioFlinger with it's mLock held, but 942 // it would be safer to explicitly pass initial masterVolume/masterMute as 943 // parameter. 944 // 945 // If the HAL we are using has support for master volume or master mute, 946 // then do not attenuate or mute during mixing (just leave the volume at 1.0 947 // and the mute set to false). 948 mMasterVolume = audioFlinger->masterVolume_l(); 949 mMasterMute = audioFlinger->masterMute_l(); 950 if (mOutput && mOutput->audioHwDev) { 951 if (mOutput->audioHwDev->canSetMasterVolume()) { 952 mMasterVolume = 1.0; 953 } 954 955 if (mOutput->audioHwDev->canSetMasterMute()) { 956 mMasterMute = false; 957 } 958 } 959 960 readOutputParameters(); 961 962 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 963 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 964 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 965 stream = (audio_stream_type_t) (stream + 1)) { 966 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 967 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 968 } 969 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 970 // because mAudioFlinger doesn't have one to copy from 971} 972 973AudioFlinger::PlaybackThread::~PlaybackThread() 974{ 975 mAudioFlinger->unregisterWriter(mNBLogWriter); 976 delete [] mMixBuffer; 977} 978 979void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 980{ 981 dumpInternals(fd, args); 982 dumpTracks(fd, args); 983 dumpEffectChains(fd, args); 984} 985 986void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 987{ 988 const size_t SIZE = 256; 989 char buffer[SIZE]; 990 String8 result; 991 992 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 993 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 994 const stream_type_t *st = &mStreamTypes[i]; 995 if (i > 0) { 996 result.appendFormat(", "); 997 } 998 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 999 if (st->mute) { 1000 result.append("M"); 1001 } 1002 } 1003 result.append("\n"); 1004 write(fd, result.string(), result.length()); 1005 result.clear(); 1006 1007 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1008 result.append(buffer); 1009 Track::appendDumpHeader(result); 1010 for (size_t i = 0; i < mTracks.size(); ++i) { 1011 sp<Track> track = mTracks[i]; 1012 if (track != 0) { 1013 track->dump(buffer, SIZE); 1014 result.append(buffer); 1015 } 1016 } 1017 1018 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1019 result.append(buffer); 1020 Track::appendDumpHeader(result); 1021 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1022 sp<Track> track = mActiveTracks[i].promote(); 1023 if (track != 0) { 1024 track->dump(buffer, SIZE); 1025 result.append(buffer); 1026 } 1027 } 1028 write(fd, result.string(), result.size()); 1029 1030 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1031 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1032 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1033 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1034} 1035 1036void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1037{ 1038 const size_t SIZE = 256; 1039 char buffer[SIZE]; 1040 String8 result; 1041 1042 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1043 result.append(buffer); 1044 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1045 ns2ms(systemTime() - mLastWriteTime)); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1056 result.append(buffer); 1057 write(fd, result.string(), result.size()); 1058 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1059 1060 dumpBase(fd, args); 1061} 1062 1063// Thread virtuals 1064status_t AudioFlinger::PlaybackThread::readyToRun() 1065{ 1066 status_t status = initCheck(); 1067 if (status == NO_ERROR) { 1068 ALOGI("AudioFlinger's thread %p ready to run", this); 1069 } else { 1070 ALOGE("No working audio driver found."); 1071 } 1072 return status; 1073} 1074 1075void AudioFlinger::PlaybackThread::onFirstRef() 1076{ 1077 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1078} 1079 1080// ThreadBase virtuals 1081void AudioFlinger::PlaybackThread::preExit() 1082{ 1083 ALOGV(" preExit()"); 1084 // FIXME this is using hard-coded strings but in the future, this functionality will be 1085 // converted to use audio HAL extensions required to support tunneling 1086 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1087} 1088 1089// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1090sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1091 const sp<AudioFlinger::Client>& client, 1092 audio_stream_type_t streamType, 1093 uint32_t sampleRate, 1094 audio_format_t format, 1095 audio_channel_mask_t channelMask, 1096 size_t frameCount, 1097 const sp<IMemory>& sharedBuffer, 1098 int sessionId, 1099 IAudioFlinger::track_flags_t *flags, 1100 pid_t tid, 1101 status_t *status) 1102{ 1103 sp<Track> track; 1104 status_t lStatus; 1105 1106 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1107 1108 // client expresses a preference for FAST, but we get the final say 1109 if (*flags & IAudioFlinger::TRACK_FAST) { 1110 if ( 1111 // not timed 1112 (!isTimed) && 1113 // either of these use cases: 1114 ( 1115 // use case 1: shared buffer with any frame count 1116 ( 1117 (sharedBuffer != 0) 1118 ) || 1119 // use case 2: callback handler and frame count is default or at least as large as HAL 1120 ( 1121 (tid != -1) && 1122 ((frameCount == 0) || 1123 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1124 ) 1125 ) && 1126 // PCM data 1127 audio_is_linear_pcm(format) && 1128 // mono or stereo 1129 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1130 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1131#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1132 // hardware sample rate 1133 (sampleRate == mSampleRate) && 1134#endif 1135 // normal mixer has an associated fast mixer 1136 hasFastMixer() && 1137 // there are sufficient fast track slots available 1138 (mFastTrackAvailMask != 0) 1139 // FIXME test that MixerThread for this fast track has a capable output HAL 1140 // FIXME add a permission test also? 1141 ) { 1142 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1143 if (frameCount == 0) { 1144 frameCount = mFrameCount * kFastTrackMultiplier; 1145 } 1146 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1147 frameCount, mFrameCount); 1148 } else { 1149 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1150 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1151 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1152 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1153 audio_is_linear_pcm(format), 1154 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1155 *flags &= ~IAudioFlinger::TRACK_FAST; 1156 // For compatibility with AudioTrack calculation, buffer depth is forced 1157 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1158 // This is probably too conservative, but legacy application code may depend on it. 1159 // If you change this calculation, also review the start threshold which is related. 1160 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1161 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1162 if (minBufCount < 2) { 1163 minBufCount = 2; 1164 } 1165 size_t minFrameCount = mNormalFrameCount * minBufCount; 1166 if (frameCount < minFrameCount) { 1167 frameCount = minFrameCount; 1168 } 1169 } 1170 } 1171 1172 if (mType == DIRECT) { 1173 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1174 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1175 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1176 "for output %p with format %d", 1177 sampleRate, format, channelMask, mOutput, mFormat); 1178 lStatus = BAD_VALUE; 1179 goto Exit; 1180 } 1181 } 1182 } else { 1183 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1184 if (sampleRate > mSampleRate*2) { 1185 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 } 1190 1191 lStatus = initCheck(); 1192 if (lStatus != NO_ERROR) { 1193 ALOGE("Audio driver not initialized."); 1194 goto Exit; 1195 } 1196 1197 { // scope for mLock 1198 Mutex::Autolock _l(mLock); 1199 1200 // all tracks in same audio session must share the same routing strategy otherwise 1201 // conflicts will happen when tracks are moved from one output to another by audio policy 1202 // manager 1203 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1204 for (size_t i = 0; i < mTracks.size(); ++i) { 1205 sp<Track> t = mTracks[i]; 1206 if (t != 0 && !t->isOutputTrack()) { 1207 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1208 if (sessionId == t->sessionId() && strategy != actual) { 1209 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1210 strategy, actual); 1211 lStatus = BAD_VALUE; 1212 goto Exit; 1213 } 1214 } 1215 } 1216 1217 if (!isTimed) { 1218 track = new Track(this, client, streamType, sampleRate, format, 1219 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1220 } else { 1221 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1222 channelMask, frameCount, sharedBuffer, sessionId); 1223 } 1224 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1225 lStatus = NO_MEMORY; 1226 goto Exit; 1227 } 1228 mTracks.add(track); 1229 1230 sp<EffectChain> chain = getEffectChain_l(sessionId); 1231 if (chain != 0) { 1232 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1233 track->setMainBuffer(chain->inBuffer()); 1234 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1235 chain->incTrackCnt(); 1236 } 1237 1238 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1239 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1240 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1241 // so ask activity manager to do this on our behalf 1242 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1243 } 1244 } 1245 1246 lStatus = NO_ERROR; 1247 1248Exit: 1249 if (status) { 1250 *status = lStatus; 1251 } 1252 return track; 1253} 1254 1255uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1256{ 1257 return latency; 1258} 1259 1260uint32_t AudioFlinger::PlaybackThread::latency() const 1261{ 1262 Mutex::Autolock _l(mLock); 1263 return latency_l(); 1264} 1265uint32_t AudioFlinger::PlaybackThread::latency_l() const 1266{ 1267 if (initCheck() == NO_ERROR) { 1268 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1269 } else { 1270 return 0; 1271 } 1272} 1273 1274void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1275{ 1276 Mutex::Autolock _l(mLock); 1277 // Don't apply master volume in SW if our HAL can do it for us. 1278 if (mOutput && mOutput->audioHwDev && 1279 mOutput->audioHwDev->canSetMasterVolume()) { 1280 mMasterVolume = 1.0; 1281 } else { 1282 mMasterVolume = value; 1283 } 1284} 1285 1286void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1287{ 1288 Mutex::Autolock _l(mLock); 1289 // Don't apply master mute in SW if our HAL can do it for us. 1290 if (mOutput && mOutput->audioHwDev && 1291 mOutput->audioHwDev->canSetMasterMute()) { 1292 mMasterMute = false; 1293 } else { 1294 mMasterMute = muted; 1295 } 1296} 1297 1298void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1299{ 1300 Mutex::Autolock _l(mLock); 1301 mStreamTypes[stream].volume = value; 1302} 1303 1304void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1305{ 1306 Mutex::Autolock _l(mLock); 1307 mStreamTypes[stream].mute = muted; 1308} 1309 1310float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1311{ 1312 Mutex::Autolock _l(mLock); 1313 return mStreamTypes[stream].volume; 1314} 1315 1316// addTrack_l() must be called with ThreadBase::mLock held 1317status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1318{ 1319 status_t status = ALREADY_EXISTS; 1320 1321 // set retry count for buffer fill 1322 track->mRetryCount = kMaxTrackStartupRetries; 1323 if (mActiveTracks.indexOf(track) < 0) { 1324 // the track is newly added, make sure it fills up all its 1325 // buffers before playing. This is to ensure the client will 1326 // effectively get the latency it requested. 1327 track->mFillingUpStatus = Track::FS_FILLING; 1328 track->mResetDone = false; 1329 track->mPresentationCompleteFrames = 0; 1330 mActiveTracks.add(track); 1331 if (track->mainBuffer() != mMixBuffer) { 1332 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1333 if (chain != 0) { 1334 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1335 track->sessionId()); 1336 chain->incActiveTrackCnt(); 1337 } 1338 } 1339 1340 status = NO_ERROR; 1341 } 1342 1343 ALOGV("mWaitWorkCV.broadcast"); 1344 mWaitWorkCV.broadcast(); 1345 1346 return status; 1347} 1348 1349// destroyTrack_l() must be called with ThreadBase::mLock held 1350void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1351{ 1352 track->mState = TrackBase::TERMINATED; 1353 // active tracks are removed by threadLoop() 1354 if (mActiveTracks.indexOf(track) < 0) { 1355 removeTrack_l(track); 1356 } 1357} 1358 1359void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1360{ 1361 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1362 mTracks.remove(track); 1363 deleteTrackName_l(track->name()); 1364 // redundant as track is about to be destroyed, for dumpsys only 1365 track->mName = -1; 1366 if (track->isFastTrack()) { 1367 int index = track->mFastIndex; 1368 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1369 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1370 mFastTrackAvailMask |= 1 << index; 1371 // redundant as track is about to be destroyed, for dumpsys only 1372 track->mFastIndex = -1; 1373 } 1374 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1375 if (chain != 0) { 1376 chain->decTrackCnt(); 1377 } 1378} 1379 1380String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1381{ 1382 String8 out_s8 = String8(""); 1383 char *s; 1384 1385 Mutex::Autolock _l(mLock); 1386 if (initCheck() != NO_ERROR) { 1387 return out_s8; 1388 } 1389 1390 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1391 out_s8 = String8(s); 1392 free(s); 1393 return out_s8; 1394} 1395 1396// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1397void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1398 AudioSystem::OutputDescriptor desc; 1399 void *param2 = NULL; 1400 1401 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1402 param); 1403 1404 switch (event) { 1405 case AudioSystem::OUTPUT_OPENED: 1406 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1407 desc.channels = mChannelMask; 1408 desc.samplingRate = mSampleRate; 1409 desc.format = mFormat; 1410 desc.frameCount = mNormalFrameCount; // FIXME see 1411 // AudioFlinger::frameCount(audio_io_handle_t) 1412 desc.latency = latency(); 1413 param2 = &desc; 1414 break; 1415 1416 case AudioSystem::STREAM_CONFIG_CHANGED: 1417 param2 = ¶m; 1418 case AudioSystem::OUTPUT_CLOSED: 1419 default: 1420 break; 1421 } 1422 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1423} 1424 1425void AudioFlinger::PlaybackThread::readOutputParameters() 1426{ 1427 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1428 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1429 mChannelCount = (uint16_t)popcount(mChannelMask); 1430 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1431 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1432 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1433 if (mFrameCount & 15) { 1434 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1435 mFrameCount); 1436 } 1437 1438 // Calculate size of normal mix buffer relative to the HAL output buffer size 1439 double multiplier = 1.0; 1440 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1441 kUseFastMixer == FastMixer_Dynamic)) { 1442 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1443 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1444 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1445 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1446 maxNormalFrameCount = maxNormalFrameCount & ~15; 1447 if (maxNormalFrameCount < minNormalFrameCount) { 1448 maxNormalFrameCount = minNormalFrameCount; 1449 } 1450 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1451 if (multiplier <= 1.0) { 1452 multiplier = 1.0; 1453 } else if (multiplier <= 2.0) { 1454 if (2 * mFrameCount <= maxNormalFrameCount) { 1455 multiplier = 2.0; 1456 } else { 1457 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1458 } 1459 } else { 1460 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1461 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1462 // track, but we sometimes have to do this to satisfy the maximum frame count 1463 // constraint) 1464 // FIXME this rounding up should not be done if no HAL SRC 1465 uint32_t truncMult = (uint32_t) multiplier; 1466 if ((truncMult & 1)) { 1467 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1468 ++truncMult; 1469 } 1470 } 1471 multiplier = (double) truncMult; 1472 } 1473 } 1474 mNormalFrameCount = multiplier * mFrameCount; 1475 // round up to nearest 16 frames to satisfy AudioMixer 1476 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1477 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1478 mNormalFrameCount); 1479 1480 delete[] mMixBuffer; 1481 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1482 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1483 1484 // force reconfiguration of effect chains and engines to take new buffer size and audio 1485 // parameters into account 1486 // Note that mLock is not held when readOutputParameters() is called from the constructor 1487 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1488 // matter. 1489 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1490 Vector< sp<EffectChain> > effectChains = mEffectChains; 1491 for (size_t i = 0; i < effectChains.size(); i ++) { 1492 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1493 } 1494} 1495 1496 1497status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1498{ 1499 if (halFrames == NULL || dspFrames == NULL) { 1500 return BAD_VALUE; 1501 } 1502 Mutex::Autolock _l(mLock); 1503 if (initCheck() != NO_ERROR) { 1504 return INVALID_OPERATION; 1505 } 1506 size_t framesWritten = mBytesWritten / mFrameSize; 1507 *halFrames = framesWritten; 1508 1509 if (isSuspended()) { 1510 // return an estimation of rendered frames when the output is suspended 1511 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1512 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1513 return NO_ERROR; 1514 } else { 1515 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1516 } 1517} 1518 1519uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1520{ 1521 Mutex::Autolock _l(mLock); 1522 uint32_t result = 0; 1523 if (getEffectChain_l(sessionId) != 0) { 1524 result = EFFECT_SESSION; 1525 } 1526 1527 for (size_t i = 0; i < mTracks.size(); ++i) { 1528 sp<Track> track = mTracks[i]; 1529 if (sessionId == track->sessionId() && !track->isInvalid()) { 1530 result |= TRACK_SESSION; 1531 break; 1532 } 1533 } 1534 1535 return result; 1536} 1537 1538uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1539{ 1540 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1541 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1542 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1543 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1544 } 1545 for (size_t i = 0; i < mTracks.size(); i++) { 1546 sp<Track> track = mTracks[i]; 1547 if (sessionId == track->sessionId() && !track->isInvalid()) { 1548 return AudioSystem::getStrategyForStream(track->streamType()); 1549 } 1550 } 1551 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1552} 1553 1554 1555AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1556{ 1557 Mutex::Autolock _l(mLock); 1558 return mOutput; 1559} 1560 1561AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1562{ 1563 Mutex::Autolock _l(mLock); 1564 AudioStreamOut *output = mOutput; 1565 mOutput = NULL; 1566 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1567 // must push a NULL and wait for ack 1568 mOutputSink.clear(); 1569 mPipeSink.clear(); 1570 mNormalSink.clear(); 1571 return output; 1572} 1573 1574// this method must always be called either with ThreadBase mLock held or inside the thread loop 1575audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1576{ 1577 if (mOutput == NULL) { 1578 return NULL; 1579 } 1580 return &mOutput->stream->common; 1581} 1582 1583uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1584{ 1585 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1586} 1587 1588status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1589{ 1590 if (!isValidSyncEvent(event)) { 1591 return BAD_VALUE; 1592 } 1593 1594 Mutex::Autolock _l(mLock); 1595 1596 for (size_t i = 0; i < mTracks.size(); ++i) { 1597 sp<Track> track = mTracks[i]; 1598 if (event->triggerSession() == track->sessionId()) { 1599 (void) track->setSyncEvent(event); 1600 return NO_ERROR; 1601 } 1602 } 1603 1604 return NAME_NOT_FOUND; 1605} 1606 1607bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1608{ 1609 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1610} 1611 1612void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1613 const Vector< sp<Track> >& tracksToRemove) 1614{ 1615 size_t count = tracksToRemove.size(); 1616 if (CC_UNLIKELY(count)) { 1617 for (size_t i = 0 ; i < count ; i++) { 1618 const sp<Track>& track = tracksToRemove.itemAt(i); 1619 if ((track->sharedBuffer() != 0) && 1620 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1621 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1622 } 1623 } 1624 } 1625 1626} 1627 1628void AudioFlinger::PlaybackThread::checkSilentMode_l() 1629{ 1630 if (!mMasterMute) { 1631 char value[PROPERTY_VALUE_MAX]; 1632 if (property_get("ro.audio.silent", value, "0") > 0) { 1633 char *endptr; 1634 unsigned long ul = strtoul(value, &endptr, 0); 1635 if (*endptr == '\0' && ul != 0) { 1636 ALOGD("Silence is golden"); 1637 // The setprop command will not allow a property to be changed after 1638 // the first time it is set, so we don't have to worry about un-muting. 1639 setMasterMute_l(true); 1640 } 1641 } 1642 } 1643} 1644 1645// shared by MIXER and DIRECT, overridden by DUPLICATING 1646void AudioFlinger::PlaybackThread::threadLoop_write() 1647{ 1648 // FIXME rewrite to reduce number of system calls 1649 mLastWriteTime = systemTime(); 1650 mInWrite = true; 1651 int bytesWritten; 1652 1653 // If an NBAIO sink is present, use it to write the normal mixer's submix 1654 if (mNormalSink != 0) { 1655#define mBitShift 2 // FIXME 1656 size_t count = mixBufferSize >> mBitShift; 1657 ATRACE_BEGIN("write"); 1658 // update the setpoint when AudioFlinger::mScreenState changes 1659 uint32_t screenState = AudioFlinger::mScreenState; 1660 if (screenState != mScreenState) { 1661 mScreenState = screenState; 1662 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1663 if (pipe != NULL) { 1664 pipe->setAvgFrames((mScreenState & 1) ? 1665 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1666 } 1667 } 1668 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1669 ATRACE_END(); 1670 if (framesWritten > 0) { 1671 bytesWritten = framesWritten << mBitShift; 1672 } else { 1673 bytesWritten = framesWritten; 1674 } 1675 // otherwise use the HAL / AudioStreamOut directly 1676 } else { 1677 // Direct output thread. 1678 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1679 } 1680 1681 if (bytesWritten > 0) { 1682 mBytesWritten += mixBufferSize; 1683 } 1684 mNumWrites++; 1685 mInWrite = false; 1686} 1687 1688/* 1689The derived values that are cached: 1690 - mixBufferSize from frame count * frame size 1691 - activeSleepTime from activeSleepTimeUs() 1692 - idleSleepTime from idleSleepTimeUs() 1693 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1694 - maxPeriod from frame count and sample rate (MIXER only) 1695 1696The parameters that affect these derived values are: 1697 - frame count 1698 - frame size 1699 - sample rate 1700 - device type: A2DP or not 1701 - device latency 1702 - format: PCM or not 1703 - active sleep time 1704 - idle sleep time 1705*/ 1706 1707void AudioFlinger::PlaybackThread::cacheParameters_l() 1708{ 1709 mixBufferSize = mNormalFrameCount * mFrameSize; 1710 activeSleepTime = activeSleepTimeUs(); 1711 idleSleepTime = idleSleepTimeUs(); 1712} 1713 1714void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1715{ 1716 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1717 this, streamType, mTracks.size()); 1718 Mutex::Autolock _l(mLock); 1719 1720 size_t size = mTracks.size(); 1721 for (size_t i = 0; i < size; i++) { 1722 sp<Track> t = mTracks[i]; 1723 if (t->streamType() == streamType) { 1724 t->invalidate(); 1725 } 1726 } 1727} 1728 1729status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1730{ 1731 int session = chain->sessionId(); 1732 int16_t *buffer = mMixBuffer; 1733 bool ownsBuffer = false; 1734 1735 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1736 if (session > 0) { 1737 // Only one effect chain can be present in direct output thread and it uses 1738 // the mix buffer as input 1739 if (mType != DIRECT) { 1740 size_t numSamples = mNormalFrameCount * mChannelCount; 1741 buffer = new int16_t[numSamples]; 1742 memset(buffer, 0, numSamples * sizeof(int16_t)); 1743 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1744 ownsBuffer = true; 1745 } 1746 1747 // Attach all tracks with same session ID to this chain. 1748 for (size_t i = 0; i < mTracks.size(); ++i) { 1749 sp<Track> track = mTracks[i]; 1750 if (session == track->sessionId()) { 1751 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1752 buffer); 1753 track->setMainBuffer(buffer); 1754 chain->incTrackCnt(); 1755 } 1756 } 1757 1758 // indicate all active tracks in the chain 1759 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1760 sp<Track> track = mActiveTracks[i].promote(); 1761 if (track == 0) { 1762 continue; 1763 } 1764 if (session == track->sessionId()) { 1765 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1766 chain->incActiveTrackCnt(); 1767 } 1768 } 1769 } 1770 1771 chain->setInBuffer(buffer, ownsBuffer); 1772 chain->setOutBuffer(mMixBuffer); 1773 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1774 // chains list in order to be processed last as it contains output stage effects 1775 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1776 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1777 // after track specific effects and before output stage 1778 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1779 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1780 // Effect chain for other sessions are inserted at beginning of effect 1781 // chains list to be processed before output mix effects. Relative order between other 1782 // sessions is not important 1783 size_t size = mEffectChains.size(); 1784 size_t i = 0; 1785 for (i = 0; i < size; i++) { 1786 if (mEffectChains[i]->sessionId() < session) { 1787 break; 1788 } 1789 } 1790 mEffectChains.insertAt(chain, i); 1791 checkSuspendOnAddEffectChain_l(chain); 1792 1793 return NO_ERROR; 1794} 1795 1796size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1797{ 1798 int session = chain->sessionId(); 1799 1800 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1801 1802 for (size_t i = 0; i < mEffectChains.size(); i++) { 1803 if (chain == mEffectChains[i]) { 1804 mEffectChains.removeAt(i); 1805 // detach all active tracks from the chain 1806 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1807 sp<Track> track = mActiveTracks[i].promote(); 1808 if (track == 0) { 1809 continue; 1810 } 1811 if (session == track->sessionId()) { 1812 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1813 chain.get(), session); 1814 chain->decActiveTrackCnt(); 1815 } 1816 } 1817 1818 // detach all tracks with same session ID from this chain 1819 for (size_t i = 0; i < mTracks.size(); ++i) { 1820 sp<Track> track = mTracks[i]; 1821 if (session == track->sessionId()) { 1822 track->setMainBuffer(mMixBuffer); 1823 chain->decTrackCnt(); 1824 } 1825 } 1826 break; 1827 } 1828 } 1829 return mEffectChains.size(); 1830} 1831 1832status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1833 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1834{ 1835 Mutex::Autolock _l(mLock); 1836 return attachAuxEffect_l(track, EffectId); 1837} 1838 1839status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1840 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1841{ 1842 status_t status = NO_ERROR; 1843 1844 if (EffectId == 0) { 1845 track->setAuxBuffer(0, NULL); 1846 } else { 1847 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1848 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1849 if (effect != 0) { 1850 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1851 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1852 } else { 1853 status = INVALID_OPERATION; 1854 } 1855 } else { 1856 status = BAD_VALUE; 1857 } 1858 } 1859 return status; 1860} 1861 1862void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1863{ 1864 for (size_t i = 0; i < mTracks.size(); ++i) { 1865 sp<Track> track = mTracks[i]; 1866 if (track->auxEffectId() == effectId) { 1867 attachAuxEffect_l(track, 0); 1868 } 1869 } 1870} 1871 1872bool AudioFlinger::PlaybackThread::threadLoop() 1873{ 1874 Vector< sp<Track> > tracksToRemove; 1875 1876 standbyTime = systemTime(); 1877 1878 // MIXER 1879 nsecs_t lastWarning = 0; 1880 1881 // DUPLICATING 1882 // FIXME could this be made local to while loop? 1883 writeFrames = 0; 1884 1885 cacheParameters_l(); 1886 sleepTime = idleSleepTime; 1887 1888 if (mType == MIXER) { 1889 sleepTimeShift = 0; 1890 } 1891 1892 CpuStats cpuStats; 1893 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1894 1895 acquireWakeLock(); 1896 1897 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1898 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1899 // and then that string will be logged at the next convenient opportunity. 1900 const char *logString = NULL; 1901 1902 while (!exitPending()) 1903 { 1904 cpuStats.sample(myName); 1905 1906 Vector< sp<EffectChain> > effectChains; 1907 1908 processConfigEvents(); 1909 1910 { // scope for mLock 1911 1912 Mutex::Autolock _l(mLock); 1913 1914 if (logString != NULL) { 1915 mNBLogWriter->logTimestamp(); 1916 mNBLogWriter->log(logString); 1917 logString = NULL; 1918 } 1919 1920 if (checkForNewParameters_l()) { 1921 cacheParameters_l(); 1922 } 1923 1924 saveOutputTracks(); 1925 1926 // put audio hardware into standby after short delay 1927 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1928 isSuspended())) { 1929 if (!mStandby) { 1930 1931 threadLoop_standby(); 1932 1933 mStandby = true; 1934 } 1935 1936 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1937 // we're about to wait, flush the binder command buffer 1938 IPCThreadState::self()->flushCommands(); 1939 1940 clearOutputTracks(); 1941 1942 if (exitPending()) { 1943 break; 1944 } 1945 1946 releaseWakeLock_l(); 1947 // wait until we have something to do... 1948 ALOGV("%s going to sleep", myName.string()); 1949 mWaitWorkCV.wait(mLock); 1950 ALOGV("%s waking up", myName.string()); 1951 acquireWakeLock_l(); 1952 1953 mMixerStatus = MIXER_IDLE; 1954 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1955 mBytesWritten = 0; 1956 1957 checkSilentMode_l(); 1958 1959 standbyTime = systemTime() + standbyDelay; 1960 sleepTime = idleSleepTime; 1961 if (mType == MIXER) { 1962 sleepTimeShift = 0; 1963 } 1964 1965 continue; 1966 } 1967 } 1968 1969 // mMixerStatusIgnoringFastTracks is also updated internally 1970 mMixerStatus = prepareTracks_l(&tracksToRemove); 1971 1972 // prevent any changes in effect chain list and in each effect chain 1973 // during mixing and effect process as the audio buffers could be deleted 1974 // or modified if an effect is created or deleted 1975 lockEffectChains_l(effectChains); 1976 } 1977 1978 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1979 threadLoop_mix(); 1980 } else { 1981 threadLoop_sleepTime(); 1982 } 1983 1984 if (isSuspended()) { 1985 sleepTime = suspendSleepTimeUs(); 1986 mBytesWritten += mixBufferSize; 1987 } 1988 1989 // only process effects if we're going to write 1990 if (sleepTime == 0) { 1991 for (size_t i = 0; i < effectChains.size(); i ++) { 1992 effectChains[i]->process_l(); 1993 } 1994 } 1995 1996 // enable changes in effect chain 1997 unlockEffectChains(effectChains); 1998 1999 // sleepTime == 0 means we must write to audio hardware 2000 if (sleepTime == 0) { 2001 2002 threadLoop_write(); 2003 2004if (mType == MIXER) { 2005 // write blocked detection 2006 nsecs_t now = systemTime(); 2007 nsecs_t delta = now - mLastWriteTime; 2008 if (!mStandby && delta > maxPeriod) { 2009 mNumDelayedWrites++; 2010 if ((now - lastWarning) > kWarningThrottleNs) { 2011 ATRACE_NAME("underrun"); 2012 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2013 ns2ms(delta), mNumDelayedWrites, this); 2014 lastWarning = now; 2015 } 2016 } 2017} 2018 2019 mStandby = false; 2020 } else { 2021 usleep(sleepTime); 2022 } 2023 2024 // Finally let go of removed track(s), without the lock held 2025 // since we can't guarantee the destructors won't acquire that 2026 // same lock. This will also mutate and push a new fast mixer state. 2027 threadLoop_removeTracks(tracksToRemove); 2028 tracksToRemove.clear(); 2029 2030 // FIXME I don't understand the need for this here; 2031 // it was in the original code but maybe the 2032 // assignment in saveOutputTracks() makes this unnecessary? 2033 clearOutputTracks(); 2034 2035 // Effect chains will be actually deleted here if they were removed from 2036 // mEffectChains list during mixing or effects processing 2037 effectChains.clear(); 2038 2039 // FIXME Note that the above .clear() is no longer necessary since effectChains 2040 // is now local to this block, but will keep it for now (at least until merge done). 2041 } 2042 2043 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2044 if (mType == MIXER || mType == DIRECT) { 2045 // put output stream into standby mode 2046 if (!mStandby) { 2047 mOutput->stream->common.standby(&mOutput->stream->common); 2048 } 2049 } 2050 2051 releaseWakeLock(); 2052 2053 ALOGV("Thread %p type %d exiting", this, mType); 2054 return false; 2055} 2056 2057 2058// ---------------------------------------------------------------------------- 2059 2060AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2061 audio_io_handle_t id, audio_devices_t device, type_t type) 2062 : PlaybackThread(audioFlinger, output, id, device, type), 2063 // mAudioMixer below 2064 // mFastMixer below 2065 mFastMixerFutex(0) 2066 // mOutputSink below 2067 // mPipeSink below 2068 // mNormalSink below 2069{ 2070 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2071 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2072 "mFrameCount=%d, mNormalFrameCount=%d", 2073 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2074 mNormalFrameCount); 2075 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2076 2077 // FIXME - Current mixer implementation only supports stereo output 2078 if (mChannelCount != FCC_2) { 2079 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2080 } 2081 2082 // create an NBAIO sink for the HAL output stream, and negotiate 2083 mOutputSink = new AudioStreamOutSink(output->stream); 2084 size_t numCounterOffers = 0; 2085 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2086 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2087 ALOG_ASSERT(index == 0); 2088 2089 // initialize fast mixer depending on configuration 2090 bool initFastMixer; 2091 switch (kUseFastMixer) { 2092 case FastMixer_Never: 2093 initFastMixer = false; 2094 break; 2095 case FastMixer_Always: 2096 initFastMixer = true; 2097 break; 2098 case FastMixer_Static: 2099 case FastMixer_Dynamic: 2100 initFastMixer = mFrameCount < mNormalFrameCount; 2101 break; 2102 } 2103 if (initFastMixer) { 2104 2105 // create a MonoPipe to connect our submix to FastMixer 2106 NBAIO_Format format = mOutputSink->format(); 2107 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2108 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2109 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2110 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2111 const NBAIO_Format offers[1] = {format}; 2112 size_t numCounterOffers = 0; 2113 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2114 ALOG_ASSERT(index == 0); 2115 monoPipe->setAvgFrames((mScreenState & 1) ? 2116 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2117 mPipeSink = monoPipe; 2118 2119 if (mTeeSinkOutputEnabled) { 2120 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2121 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2122 numCounterOffers = 0; 2123 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2124 ALOG_ASSERT(index == 0); 2125 mTeeSink = teeSink; 2126 PipeReader *teeSource = new PipeReader(*teeSink); 2127 numCounterOffers = 0; 2128 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2129 ALOG_ASSERT(index == 0); 2130 mTeeSource = teeSource; 2131 } 2132 2133 // create fast mixer and configure it initially with just one fast track for our submix 2134 mFastMixer = new FastMixer(); 2135 FastMixerStateQueue *sq = mFastMixer->sq(); 2136#ifdef STATE_QUEUE_DUMP 2137 sq->setObserverDump(&mStateQueueObserverDump); 2138 sq->setMutatorDump(&mStateQueueMutatorDump); 2139#endif 2140 FastMixerState *state = sq->begin(); 2141 FastTrack *fastTrack = &state->mFastTracks[0]; 2142 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2143 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2144 fastTrack->mVolumeProvider = NULL; 2145 fastTrack->mGeneration++; 2146 state->mFastTracksGen++; 2147 state->mTrackMask = 1; 2148 // fast mixer will use the HAL output sink 2149 state->mOutputSink = mOutputSink.get(); 2150 state->mOutputSinkGen++; 2151 state->mFrameCount = mFrameCount; 2152 state->mCommand = FastMixerState::COLD_IDLE; 2153 // already done in constructor initialization list 2154 //mFastMixerFutex = 0; 2155 state->mColdFutexAddr = &mFastMixerFutex; 2156 state->mColdGen++; 2157 state->mDumpState = &mFastMixerDumpState; 2158 state->mTeeSink = mTeeSink.get(); 2159 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2160 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2161 sq->end(); 2162 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2163 2164 // start the fast mixer 2165 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2166 pid_t tid = mFastMixer->getTid(); 2167 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2168 if (err != 0) { 2169 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2170 kPriorityFastMixer, getpid_cached, tid, err); 2171 } 2172 2173#ifdef AUDIO_WATCHDOG 2174 // create and start the watchdog 2175 mAudioWatchdog = new AudioWatchdog(); 2176 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2177 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2178 tid = mAudioWatchdog->getTid(); 2179 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2180 if (err != 0) { 2181 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2182 kPriorityFastMixer, getpid_cached, tid, err); 2183 } 2184#endif 2185 2186 } else { 2187 mFastMixer = NULL; 2188 } 2189 2190 switch (kUseFastMixer) { 2191 case FastMixer_Never: 2192 case FastMixer_Dynamic: 2193 mNormalSink = mOutputSink; 2194 break; 2195 case FastMixer_Always: 2196 mNormalSink = mPipeSink; 2197 break; 2198 case FastMixer_Static: 2199 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2200 break; 2201 } 2202} 2203 2204AudioFlinger::MixerThread::~MixerThread() 2205{ 2206 if (mFastMixer != NULL) { 2207 FastMixerStateQueue *sq = mFastMixer->sq(); 2208 FastMixerState *state = sq->begin(); 2209 if (state->mCommand == FastMixerState::COLD_IDLE) { 2210 int32_t old = android_atomic_inc(&mFastMixerFutex); 2211 if (old == -1) { 2212 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2213 } 2214 } 2215 state->mCommand = FastMixerState::EXIT; 2216 sq->end(); 2217 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2218 mFastMixer->join(); 2219 // Though the fast mixer thread has exited, it's state queue is still valid. 2220 // We'll use that extract the final state which contains one remaining fast track 2221 // corresponding to our sub-mix. 2222 state = sq->begin(); 2223 ALOG_ASSERT(state->mTrackMask == 1); 2224 FastTrack *fastTrack = &state->mFastTracks[0]; 2225 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2226 delete fastTrack->mBufferProvider; 2227 sq->end(false /*didModify*/); 2228 delete mFastMixer; 2229#ifdef AUDIO_WATCHDOG 2230 if (mAudioWatchdog != 0) { 2231 mAudioWatchdog->requestExit(); 2232 mAudioWatchdog->requestExitAndWait(); 2233 mAudioWatchdog.clear(); 2234 } 2235#endif 2236 } 2237 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2238 delete mAudioMixer; 2239} 2240 2241 2242uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2243{ 2244 if (mFastMixer != NULL) { 2245 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2246 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2247 } 2248 return latency; 2249} 2250 2251 2252void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2253{ 2254 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2255} 2256 2257void AudioFlinger::MixerThread::threadLoop_write() 2258{ 2259 // FIXME we should only do one push per cycle; confirm this is true 2260 // Start the fast mixer if it's not already running 2261 if (mFastMixer != NULL) { 2262 FastMixerStateQueue *sq = mFastMixer->sq(); 2263 FastMixerState *state = sq->begin(); 2264 if (state->mCommand != FastMixerState::MIX_WRITE && 2265 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2266 if (state->mCommand == FastMixerState::COLD_IDLE) { 2267 int32_t old = android_atomic_inc(&mFastMixerFutex); 2268 if (old == -1) { 2269 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2270 } 2271#ifdef AUDIO_WATCHDOG 2272 if (mAudioWatchdog != 0) { 2273 mAudioWatchdog->resume(); 2274 } 2275#endif 2276 } 2277 state->mCommand = FastMixerState::MIX_WRITE; 2278 sq->end(); 2279 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2280 if (kUseFastMixer == FastMixer_Dynamic) { 2281 mNormalSink = mPipeSink; 2282 } 2283 } else { 2284 sq->end(false /*didModify*/); 2285 } 2286 } 2287 PlaybackThread::threadLoop_write(); 2288} 2289 2290void AudioFlinger::MixerThread::threadLoop_standby() 2291{ 2292 // Idle the fast mixer if it's currently running 2293 if (mFastMixer != NULL) { 2294 FastMixerStateQueue *sq = mFastMixer->sq(); 2295 FastMixerState *state = sq->begin(); 2296 if (!(state->mCommand & FastMixerState::IDLE)) { 2297 state->mCommand = FastMixerState::COLD_IDLE; 2298 state->mColdFutexAddr = &mFastMixerFutex; 2299 state->mColdGen++; 2300 mFastMixerFutex = 0; 2301 sq->end(); 2302 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2303 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2304 if (kUseFastMixer == FastMixer_Dynamic) { 2305 mNormalSink = mOutputSink; 2306 } 2307#ifdef AUDIO_WATCHDOG 2308 if (mAudioWatchdog != 0) { 2309 mAudioWatchdog->pause(); 2310 } 2311#endif 2312 } else { 2313 sq->end(false /*didModify*/); 2314 } 2315 } 2316 PlaybackThread::threadLoop_standby(); 2317} 2318 2319// shared by MIXER and DIRECT, overridden by DUPLICATING 2320void AudioFlinger::PlaybackThread::threadLoop_standby() 2321{ 2322 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2323 mOutput->stream->common.standby(&mOutput->stream->common); 2324} 2325 2326void AudioFlinger::MixerThread::threadLoop_mix() 2327{ 2328 // obtain the presentation timestamp of the next output buffer 2329 int64_t pts; 2330 status_t status = INVALID_OPERATION; 2331 2332 if (mNormalSink != 0) { 2333 status = mNormalSink->getNextWriteTimestamp(&pts); 2334 } else { 2335 status = mOutputSink->getNextWriteTimestamp(&pts); 2336 } 2337 2338 if (status != NO_ERROR) { 2339 pts = AudioBufferProvider::kInvalidPTS; 2340 } 2341 2342 // mix buffers... 2343 mAudioMixer->process(pts); 2344 // increase sleep time progressively when application underrun condition clears. 2345 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2346 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2347 // such that we would underrun the audio HAL. 2348 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2349 sleepTimeShift--; 2350 } 2351 sleepTime = 0; 2352 standbyTime = systemTime() + standbyDelay; 2353 //TODO: delay standby when effects have a tail 2354} 2355 2356void AudioFlinger::MixerThread::threadLoop_sleepTime() 2357{ 2358 // If no tracks are ready, sleep once for the duration of an output 2359 // buffer size, then write 0s to the output 2360 if (sleepTime == 0) { 2361 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2362 sleepTime = activeSleepTime >> sleepTimeShift; 2363 if (sleepTime < kMinThreadSleepTimeUs) { 2364 sleepTime = kMinThreadSleepTimeUs; 2365 } 2366 // reduce sleep time in case of consecutive application underruns to avoid 2367 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2368 // duration we would end up writing less data than needed by the audio HAL if 2369 // the condition persists. 2370 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2371 sleepTimeShift++; 2372 } 2373 } else { 2374 sleepTime = idleSleepTime; 2375 } 2376 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2377 memset (mMixBuffer, 0, mixBufferSize); 2378 sleepTime = 0; 2379 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2380 "anticipated start"); 2381 } 2382 // TODO add standby time extension fct of effect tail 2383} 2384 2385// prepareTracks_l() must be called with ThreadBase::mLock held 2386AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2387 Vector< sp<Track> > *tracksToRemove) 2388{ 2389 2390 mixer_state mixerStatus = MIXER_IDLE; 2391 // find out which tracks need to be processed 2392 size_t count = mActiveTracks.size(); 2393 size_t mixedTracks = 0; 2394 size_t tracksWithEffect = 0; 2395 // counts only _active_ fast tracks 2396 size_t fastTracks = 0; 2397 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2398 2399 float masterVolume = mMasterVolume; 2400 bool masterMute = mMasterMute; 2401 2402 if (masterMute) { 2403 masterVolume = 0; 2404 } 2405 // Delegate master volume control to effect in output mix effect chain if needed 2406 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2407 if (chain != 0) { 2408 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2409 chain->setVolume_l(&v, &v); 2410 masterVolume = (float)((v + (1 << 23)) >> 24); 2411 chain.clear(); 2412 } 2413 2414 // prepare a new state to push 2415 FastMixerStateQueue *sq = NULL; 2416 FastMixerState *state = NULL; 2417 bool didModify = false; 2418 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2419 if (mFastMixer != NULL) { 2420 sq = mFastMixer->sq(); 2421 state = sq->begin(); 2422 } 2423 2424 for (size_t i=0 ; i<count ; i++) { 2425 sp<Track> t = mActiveTracks[i].promote(); 2426 if (t == 0) { 2427 continue; 2428 } 2429 2430 // this const just means the local variable doesn't change 2431 Track* const track = t.get(); 2432 2433 // process fast tracks 2434 if (track->isFastTrack()) { 2435 2436 // It's theoretically possible (though unlikely) for a fast track to be created 2437 // and then removed within the same normal mix cycle. This is not a problem, as 2438 // the track never becomes active so it's fast mixer slot is never touched. 2439 // The converse, of removing an (active) track and then creating a new track 2440 // at the identical fast mixer slot within the same normal mix cycle, 2441 // is impossible because the slot isn't marked available until the end of each cycle. 2442 int j = track->mFastIndex; 2443 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2444 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2445 FastTrack *fastTrack = &state->mFastTracks[j]; 2446 2447 // Determine whether the track is currently in underrun condition, 2448 // and whether it had a recent underrun. 2449 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2450 FastTrackUnderruns underruns = ftDump->mUnderruns; 2451 uint32_t recentFull = (underruns.mBitFields.mFull - 2452 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2453 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2454 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2455 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2456 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2457 uint32_t recentUnderruns = recentPartial + recentEmpty; 2458 track->mObservedUnderruns = underruns; 2459 // don't count underruns that occur while stopping or pausing 2460 // or stopped which can occur when flush() is called while active 2461 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2462 track->mUnderrunCount += recentUnderruns; 2463 } 2464 2465 // This is similar to the state machine for normal tracks, 2466 // with a few modifications for fast tracks. 2467 bool isActive = true; 2468 switch (track->mState) { 2469 case TrackBase::STOPPING_1: 2470 // track stays active in STOPPING_1 state until first underrun 2471 if (recentUnderruns > 0) { 2472 track->mState = TrackBase::STOPPING_2; 2473 } 2474 break; 2475 case TrackBase::PAUSING: 2476 // ramp down is not yet implemented 2477 track->setPaused(); 2478 break; 2479 case TrackBase::RESUMING: 2480 // ramp up is not yet implemented 2481 track->mState = TrackBase::ACTIVE; 2482 break; 2483 case TrackBase::ACTIVE: 2484 if (recentFull > 0 || recentPartial > 0) { 2485 // track has provided at least some frames recently: reset retry count 2486 track->mRetryCount = kMaxTrackRetries; 2487 } 2488 if (recentUnderruns == 0) { 2489 // no recent underruns: stay active 2490 break; 2491 } 2492 // there has recently been an underrun of some kind 2493 if (track->sharedBuffer() == 0) { 2494 // were any of the recent underruns "empty" (no frames available)? 2495 if (recentEmpty == 0) { 2496 // no, then ignore the partial underruns as they are allowed indefinitely 2497 break; 2498 } 2499 // there has recently been an "empty" underrun: decrement the retry counter 2500 if (--(track->mRetryCount) > 0) { 2501 break; 2502 } 2503 // indicate to client process that the track was disabled because of underrun; 2504 // it will then automatically call start() when data is available 2505 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2506 // remove from active list, but state remains ACTIVE [confusing but true] 2507 isActive = false; 2508 break; 2509 } 2510 // fall through 2511 case TrackBase::STOPPING_2: 2512 case TrackBase::PAUSED: 2513 case TrackBase::TERMINATED: 2514 case TrackBase::STOPPED: 2515 case TrackBase::FLUSHED: // flush() while active 2516 // Check for presentation complete if track is inactive 2517 // We have consumed all the buffers of this track. 2518 // This would be incomplete if we auto-paused on underrun 2519 { 2520 size_t audioHALFrames = 2521 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2522 size_t framesWritten = mBytesWritten / mFrameSize; 2523 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2524 // track stays in active list until presentation is complete 2525 break; 2526 } 2527 } 2528 if (track->isStopping_2()) { 2529 track->mState = TrackBase::STOPPED; 2530 } 2531 if (track->isStopped()) { 2532 // Can't reset directly, as fast mixer is still polling this track 2533 // track->reset(); 2534 // So instead mark this track as needing to be reset after push with ack 2535 resetMask |= 1 << i; 2536 } 2537 isActive = false; 2538 break; 2539 case TrackBase::IDLE: 2540 default: 2541 LOG_FATAL("unexpected track state %d", track->mState); 2542 } 2543 2544 if (isActive) { 2545 // was it previously inactive? 2546 if (!(state->mTrackMask & (1 << j))) { 2547 ExtendedAudioBufferProvider *eabp = track; 2548 VolumeProvider *vp = track; 2549 fastTrack->mBufferProvider = eabp; 2550 fastTrack->mVolumeProvider = vp; 2551 fastTrack->mSampleRate = track->mSampleRate; 2552 fastTrack->mChannelMask = track->mChannelMask; 2553 fastTrack->mGeneration++; 2554 state->mTrackMask |= 1 << j; 2555 didModify = true; 2556 // no acknowledgement required for newly active tracks 2557 } 2558 // cache the combined master volume and stream type volume for fast mixer; this 2559 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2560 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2561 ++fastTracks; 2562 } else { 2563 // was it previously active? 2564 if (state->mTrackMask & (1 << j)) { 2565 fastTrack->mBufferProvider = NULL; 2566 fastTrack->mGeneration++; 2567 state->mTrackMask &= ~(1 << j); 2568 didModify = true; 2569 // If any fast tracks were removed, we must wait for acknowledgement 2570 // because we're about to decrement the last sp<> on those tracks. 2571 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2572 } else { 2573 LOG_FATAL("fast track %d should have been active", j); 2574 } 2575 tracksToRemove->add(track); 2576 // Avoids a misleading display in dumpsys 2577 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2578 } 2579 continue; 2580 } 2581 2582 { // local variable scope to avoid goto warning 2583 2584 audio_track_cblk_t* cblk = track->cblk(); 2585 2586 // The first time a track is added we wait 2587 // for all its buffers to be filled before processing it 2588 int name = track->name(); 2589 // make sure that we have enough frames to mix one full buffer. 2590 // enforce this condition only once to enable draining the buffer in case the client 2591 // app does not call stop() and relies on underrun to stop: 2592 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2593 // during last round 2594 uint32_t minFrames = 1; 2595 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2596 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2597 if (t->sampleRate() == mSampleRate) { 2598 minFrames = mNormalFrameCount; 2599 } else { 2600 // +1 for rounding and +1 for additional sample needed for interpolation 2601 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2602 // add frames already consumed but not yet released by the resampler 2603 // because cblk->framesReady() will include these frames 2604 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2605 // the minimum track buffer size is normally twice the number of frames necessary 2606 // to fill one buffer and the resampler should not leave more than one buffer worth 2607 // of unreleased frames after each pass, but just in case... 2608 ALOG_ASSERT(minFrames <= cblk->frameCount_); 2609 } 2610 } 2611 if ((track->framesReady() >= minFrames) && track->isReady() && 2612 !track->isPaused() && !track->isTerminated()) 2613 { 2614 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2615 this); 2616 2617 mixedTracks++; 2618 2619 // track->mainBuffer() != mMixBuffer means there is an effect chain 2620 // connected to the track 2621 chain.clear(); 2622 if (track->mainBuffer() != mMixBuffer) { 2623 chain = getEffectChain_l(track->sessionId()); 2624 // Delegate volume control to effect in track effect chain if needed 2625 if (chain != 0) { 2626 tracksWithEffect++; 2627 } else { 2628 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2629 "session %d", 2630 name, track->sessionId()); 2631 } 2632 } 2633 2634 2635 int param = AudioMixer::VOLUME; 2636 if (track->mFillingUpStatus == Track::FS_FILLED) { 2637 // no ramp for the first volume setting 2638 track->mFillingUpStatus = Track::FS_ACTIVE; 2639 if (track->mState == TrackBase::RESUMING) { 2640 track->mState = TrackBase::ACTIVE; 2641 param = AudioMixer::RAMP_VOLUME; 2642 } 2643 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2644 } else if (cblk->server != 0) { 2645 // If the track is stopped before the first frame was mixed, 2646 // do not apply ramp 2647 param = AudioMixer::RAMP_VOLUME; 2648 } 2649 2650 // compute volume for this track 2651 uint32_t vl, vr, va; 2652 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2653 vl = vr = va = 0; 2654 if (track->isPausing()) { 2655 track->setPaused(); 2656 } 2657 } else { 2658 2659 // read original volumes with volume control 2660 float typeVolume = mStreamTypes[track->streamType()].volume; 2661 float v = masterVolume * typeVolume; 2662 ServerProxy *proxy = track->mServerProxy; 2663 uint32_t vlr = proxy->getVolumeLR(); 2664 vl = vlr & 0xFFFF; 2665 vr = vlr >> 16; 2666 // track volumes come from shared memory, so can't be trusted and must be clamped 2667 if (vl > MAX_GAIN_INT) { 2668 ALOGV("Track left volume out of range: %04X", vl); 2669 vl = MAX_GAIN_INT; 2670 } 2671 if (vr > MAX_GAIN_INT) { 2672 ALOGV("Track right volume out of range: %04X", vr); 2673 vr = MAX_GAIN_INT; 2674 } 2675 // now apply the master volume and stream type volume 2676 vl = (uint32_t)(v * vl) << 12; 2677 vr = (uint32_t)(v * vr) << 12; 2678 // assuming master volume and stream type volume each go up to 1.0, 2679 // vl and vr are now in 8.24 format 2680 2681 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2682 // send level comes from shared memory and so may be corrupt 2683 if (sendLevel > MAX_GAIN_INT) { 2684 ALOGV("Track send level out of range: %04X", sendLevel); 2685 sendLevel = MAX_GAIN_INT; 2686 } 2687 va = (uint32_t)(v * sendLevel); 2688 } 2689 // Delegate volume control to effect in track effect chain if needed 2690 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2691 // Do not ramp volume if volume is controlled by effect 2692 param = AudioMixer::VOLUME; 2693 track->mHasVolumeController = true; 2694 } else { 2695 // force no volume ramp when volume controller was just disabled or removed 2696 // from effect chain to avoid volume spike 2697 if (track->mHasVolumeController) { 2698 param = AudioMixer::VOLUME; 2699 } 2700 track->mHasVolumeController = false; 2701 } 2702 2703 // Convert volumes from 8.24 to 4.12 format 2704 // This additional clamping is needed in case chain->setVolume_l() overshot 2705 vl = (vl + (1 << 11)) >> 12; 2706 if (vl > MAX_GAIN_INT) { 2707 vl = MAX_GAIN_INT; 2708 } 2709 vr = (vr + (1 << 11)) >> 12; 2710 if (vr > MAX_GAIN_INT) { 2711 vr = MAX_GAIN_INT; 2712 } 2713 2714 if (va > MAX_GAIN_INT) { 2715 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2716 } 2717 2718 // XXX: these things DON'T need to be done each time 2719 mAudioMixer->setBufferProvider(name, track); 2720 mAudioMixer->enable(name); 2721 2722 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2723 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2724 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2725 mAudioMixer->setParameter( 2726 name, 2727 AudioMixer::TRACK, 2728 AudioMixer::FORMAT, (void *)track->format()); 2729 mAudioMixer->setParameter( 2730 name, 2731 AudioMixer::TRACK, 2732 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2733 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2734 uint32_t maxSampleRate = mSampleRate * 2; 2735 uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); 2736 if (reqSampleRate == 0) { 2737 reqSampleRate = mSampleRate; 2738 } else if (reqSampleRate > maxSampleRate) { 2739 reqSampleRate = maxSampleRate; 2740 } 2741 mAudioMixer->setParameter( 2742 name, 2743 AudioMixer::RESAMPLE, 2744 AudioMixer::SAMPLE_RATE, 2745 (void *)reqSampleRate); 2746 mAudioMixer->setParameter( 2747 name, 2748 AudioMixer::TRACK, 2749 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2750 mAudioMixer->setParameter( 2751 name, 2752 AudioMixer::TRACK, 2753 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2754 2755 // reset retry count 2756 track->mRetryCount = kMaxTrackRetries; 2757 2758 // If one track is ready, set the mixer ready if: 2759 // - the mixer was not ready during previous round OR 2760 // - no other track is not ready 2761 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2762 mixerStatus != MIXER_TRACKS_ENABLED) { 2763 mixerStatus = MIXER_TRACKS_READY; 2764 } 2765 } else { 2766 // clear effect chain input buffer if an active track underruns to avoid sending 2767 // previous audio buffer again to effects 2768 chain = getEffectChain_l(track->sessionId()); 2769 if (chain != 0) { 2770 chain->clearInputBuffer(); 2771 } 2772 2773 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2774 cblk->server, this); 2775 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2776 track->isStopped() || track->isPaused()) { 2777 // We have consumed all the buffers of this track. 2778 // Remove it from the list of active tracks. 2779 // TODO: use actual buffer filling status instead of latency when available from 2780 // audio HAL 2781 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2782 size_t framesWritten = mBytesWritten / mFrameSize; 2783 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2784 if (track->isStopped()) { 2785 track->reset(); 2786 } 2787 tracksToRemove->add(track); 2788 } 2789 } else { 2790 track->mUnderrunCount++; 2791 // No buffers for this track. Give it a few chances to 2792 // fill a buffer, then remove it from active list. 2793 if (--(track->mRetryCount) <= 0) { 2794 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2795 tracksToRemove->add(track); 2796 // indicate to client process that the track was disabled because of underrun; 2797 // it will then automatically call start() when data is available 2798 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2799 // If one track is not ready, mark the mixer also not ready if: 2800 // - the mixer was ready during previous round OR 2801 // - no other track is ready 2802 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2803 mixerStatus != MIXER_TRACKS_READY) { 2804 mixerStatus = MIXER_TRACKS_ENABLED; 2805 } 2806 } 2807 mAudioMixer->disable(name); 2808 } 2809 2810 } // local variable scope to avoid goto warning 2811track_is_ready: ; 2812 2813 } 2814 2815 // Push the new FastMixer state if necessary 2816 bool pauseAudioWatchdog = false; 2817 if (didModify) { 2818 state->mFastTracksGen++; 2819 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2820 if (kUseFastMixer == FastMixer_Dynamic && 2821 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2822 state->mCommand = FastMixerState::COLD_IDLE; 2823 state->mColdFutexAddr = &mFastMixerFutex; 2824 state->mColdGen++; 2825 mFastMixerFutex = 0; 2826 if (kUseFastMixer == FastMixer_Dynamic) { 2827 mNormalSink = mOutputSink; 2828 } 2829 // If we go into cold idle, need to wait for acknowledgement 2830 // so that fast mixer stops doing I/O. 2831 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2832 pauseAudioWatchdog = true; 2833 } 2834 } 2835 if (sq != NULL) { 2836 sq->end(didModify); 2837 sq->push(block); 2838 } 2839#ifdef AUDIO_WATCHDOG 2840 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2841 mAudioWatchdog->pause(); 2842 } 2843#endif 2844 2845 // Now perform the deferred reset on fast tracks that have stopped 2846 while (resetMask != 0) { 2847 size_t i = __builtin_ctz(resetMask); 2848 ALOG_ASSERT(i < count); 2849 resetMask &= ~(1 << i); 2850 sp<Track> t = mActiveTracks[i].promote(); 2851 if (t == 0) { 2852 continue; 2853 } 2854 Track* track = t.get(); 2855 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2856 track->reset(); 2857 } 2858 2859 // remove all the tracks that need to be... 2860 count = tracksToRemove->size(); 2861 if (CC_UNLIKELY(count)) { 2862 for (size_t i=0 ; i<count ; i++) { 2863 const sp<Track>& track = tracksToRemove->itemAt(i); 2864 mActiveTracks.remove(track); 2865 if (track->mainBuffer() != mMixBuffer) { 2866 chain = getEffectChain_l(track->sessionId()); 2867 if (chain != 0) { 2868 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2869 track->sessionId()); 2870 chain->decActiveTrackCnt(); 2871 } 2872 } 2873 if (track->isTerminated()) { 2874 removeTrack_l(track); 2875 } 2876 } 2877 } 2878 2879 // mix buffer must be cleared if all tracks are connected to an 2880 // effect chain as in this case the mixer will not write to 2881 // mix buffer and track effects will accumulate into it 2882 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2883 (mixedTracks == 0 && fastTracks > 0)) { 2884 // FIXME as a performance optimization, should remember previous zero status 2885 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2886 } 2887 2888 // if any fast tracks, then status is ready 2889 mMixerStatusIgnoringFastTracks = mixerStatus; 2890 if (fastTracks > 0) { 2891 mixerStatus = MIXER_TRACKS_READY; 2892 } 2893 return mixerStatus; 2894} 2895 2896// getTrackName_l() must be called with ThreadBase::mLock held 2897int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2898{ 2899 return mAudioMixer->getTrackName(channelMask, sessionId); 2900} 2901 2902// deleteTrackName_l() must be called with ThreadBase::mLock held 2903void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2904{ 2905 ALOGV("remove track (%d) and delete from mixer", name); 2906 mAudioMixer->deleteTrackName(name); 2907} 2908 2909// checkForNewParameters_l() must be called with ThreadBase::mLock held 2910bool AudioFlinger::MixerThread::checkForNewParameters_l() 2911{ 2912 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2913 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2914 bool reconfig = false; 2915 2916 while (!mNewParameters.isEmpty()) { 2917 2918 if (mFastMixer != NULL) { 2919 FastMixerStateQueue *sq = mFastMixer->sq(); 2920 FastMixerState *state = sq->begin(); 2921 if (!(state->mCommand & FastMixerState::IDLE)) { 2922 previousCommand = state->mCommand; 2923 state->mCommand = FastMixerState::HOT_IDLE; 2924 sq->end(); 2925 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2926 } else { 2927 sq->end(false /*didModify*/); 2928 } 2929 } 2930 2931 status_t status = NO_ERROR; 2932 String8 keyValuePair = mNewParameters[0]; 2933 AudioParameter param = AudioParameter(keyValuePair); 2934 int value; 2935 2936 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2937 reconfig = true; 2938 } 2939 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2940 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2941 status = BAD_VALUE; 2942 } else { 2943 reconfig = true; 2944 } 2945 } 2946 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2947 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2948 status = BAD_VALUE; 2949 } else { 2950 reconfig = true; 2951 } 2952 } 2953 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2954 // do not accept frame count changes if tracks are open as the track buffer 2955 // size depends on frame count and correct behavior would not be guaranteed 2956 // if frame count is changed after track creation 2957 if (!mTracks.isEmpty()) { 2958 status = INVALID_OPERATION; 2959 } else { 2960 reconfig = true; 2961 } 2962 } 2963 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2964#ifdef ADD_BATTERY_DATA 2965 // when changing the audio output device, call addBatteryData to notify 2966 // the change 2967 if (mOutDevice != value) { 2968 uint32_t params = 0; 2969 // check whether speaker is on 2970 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2971 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2972 } 2973 2974 audio_devices_t deviceWithoutSpeaker 2975 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2976 // check if any other device (except speaker) is on 2977 if (value & deviceWithoutSpeaker ) { 2978 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2979 } 2980 2981 if (params != 0) { 2982 addBatteryData(params); 2983 } 2984 } 2985#endif 2986 2987 // forward device change to effects that have requested to be 2988 // aware of attached audio device. 2989 mOutDevice = value; 2990 for (size_t i = 0; i < mEffectChains.size(); i++) { 2991 mEffectChains[i]->setDevice_l(mOutDevice); 2992 } 2993 } 2994 2995 if (status == NO_ERROR) { 2996 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2997 keyValuePair.string()); 2998 if (!mStandby && status == INVALID_OPERATION) { 2999 mOutput->stream->common.standby(&mOutput->stream->common); 3000 mStandby = true; 3001 mBytesWritten = 0; 3002 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3003 keyValuePair.string()); 3004 } 3005 if (status == NO_ERROR && reconfig) { 3006 delete mAudioMixer; 3007 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3008 mAudioMixer = NULL; 3009 readOutputParameters(); 3010 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3011 for (size_t i = 0; i < mTracks.size() ; i++) { 3012 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3013 if (name < 0) { 3014 break; 3015 } 3016 mTracks[i]->mName = name; 3017 } 3018 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3019 } 3020 } 3021 3022 mNewParameters.removeAt(0); 3023 3024 mParamStatus = status; 3025 mParamCond.signal(); 3026 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3027 // already timed out waiting for the status and will never signal the condition. 3028 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3029 } 3030 3031 if (!(previousCommand & FastMixerState::IDLE)) { 3032 ALOG_ASSERT(mFastMixer != NULL); 3033 FastMixerStateQueue *sq = mFastMixer->sq(); 3034 FastMixerState *state = sq->begin(); 3035 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3036 state->mCommand = previousCommand; 3037 sq->end(); 3038 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3039 } 3040 3041 return reconfig; 3042} 3043 3044 3045void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3046{ 3047 const size_t SIZE = 256; 3048 char buffer[SIZE]; 3049 String8 result; 3050 3051 PlaybackThread::dumpInternals(fd, args); 3052 3053 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3054 result.append(buffer); 3055 write(fd, result.string(), result.size()); 3056 3057 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3058 FastMixerDumpState copy = mFastMixerDumpState; 3059 copy.dump(fd); 3060 3061#ifdef STATE_QUEUE_DUMP 3062 // Similar for state queue 3063 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3064 observerCopy.dump(fd); 3065 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3066 mutatorCopy.dump(fd); 3067#endif 3068 3069 // Write the tee output to a .wav file 3070 dumpTee(fd, mTeeSource, mId); 3071 3072#ifdef AUDIO_WATCHDOG 3073 if (mAudioWatchdog != 0) { 3074 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3075 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3076 wdCopy.dump(fd); 3077 } 3078#endif 3079} 3080 3081uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3082{ 3083 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3084} 3085 3086uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3087{ 3088 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3089} 3090 3091void AudioFlinger::MixerThread::cacheParameters_l() 3092{ 3093 PlaybackThread::cacheParameters_l(); 3094 3095 // FIXME: Relaxed timing because of a certain device that can't meet latency 3096 // Should be reduced to 2x after the vendor fixes the driver issue 3097 // increase threshold again due to low power audio mode. The way this warning 3098 // threshold is calculated and its usefulness should be reconsidered anyway. 3099 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3100} 3101 3102// ---------------------------------------------------------------------------- 3103 3104AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3105 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3106 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3107 // mLeftVolFloat, mRightVolFloat 3108{ 3109} 3110 3111AudioFlinger::DirectOutputThread::~DirectOutputThread() 3112{ 3113} 3114 3115AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3116 Vector< sp<Track> > *tracksToRemove 3117) 3118{ 3119 sp<Track> trackToRemove; 3120 3121 mixer_state mixerStatus = MIXER_IDLE; 3122 3123 // find out which tracks need to be processed 3124 if (mActiveTracks.size() != 0) { 3125 sp<Track> t = mActiveTracks[0].promote(); 3126 // The track died recently 3127 if (t == 0) { 3128 return MIXER_IDLE; 3129 } 3130 3131 Track* const track = t.get(); 3132 audio_track_cblk_t* cblk = track->cblk(); 3133 3134 // The first time a track is added we wait 3135 // for all its buffers to be filled before processing it 3136 uint32_t minFrames; 3137 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3138 minFrames = mNormalFrameCount; 3139 } else { 3140 minFrames = 1; 3141 } 3142 if ((track->framesReady() >= minFrames) && track->isReady() && 3143 !track->isPaused() && !track->isTerminated()) 3144 { 3145 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3146 3147 if (track->mFillingUpStatus == Track::FS_FILLED) { 3148 track->mFillingUpStatus = Track::FS_ACTIVE; 3149 mLeftVolFloat = mRightVolFloat = 0; 3150 if (track->mState == TrackBase::RESUMING) { 3151 track->mState = TrackBase::ACTIVE; 3152 } 3153 } 3154 3155 // compute volume for this track 3156 float left, right; 3157 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3158 left = right = 0; 3159 if (track->isPausing()) { 3160 track->setPaused(); 3161 } 3162 } else { 3163 float typeVolume = mStreamTypes[track->streamType()].volume; 3164 float v = mMasterVolume * typeVolume; 3165 uint32_t vlr = track->mServerProxy->getVolumeLR(); 3166 float v_clamped = v * (vlr & 0xFFFF); 3167 if (v_clamped > MAX_GAIN) { 3168 v_clamped = MAX_GAIN; 3169 } 3170 left = v_clamped/MAX_GAIN; 3171 v_clamped = v * (vlr >> 16); 3172 if (v_clamped > MAX_GAIN) { 3173 v_clamped = MAX_GAIN; 3174 } 3175 right = v_clamped/MAX_GAIN; 3176 } 3177 3178 if (left != mLeftVolFloat || right != mRightVolFloat) { 3179 mLeftVolFloat = left; 3180 mRightVolFloat = right; 3181 3182 // Convert volumes from float to 8.24 3183 uint32_t vl = (uint32_t)(left * (1 << 24)); 3184 uint32_t vr = (uint32_t)(right * (1 << 24)); 3185 3186 // Delegate volume control to effect in track effect chain if needed 3187 // only one effect chain can be present on DirectOutputThread, so if 3188 // there is one, the track is connected to it 3189 if (!mEffectChains.isEmpty()) { 3190 // Do not ramp volume if volume is controlled by effect 3191 mEffectChains[0]->setVolume_l(&vl, &vr); 3192 left = (float)vl / (1 << 24); 3193 right = (float)vr / (1 << 24); 3194 } 3195 mOutput->stream->set_volume(mOutput->stream, left, right); 3196 } 3197 3198 // reset retry count 3199 track->mRetryCount = kMaxTrackRetriesDirect; 3200 mActiveTrack = t; 3201 mixerStatus = MIXER_TRACKS_READY; 3202 } else { 3203 // clear effect chain input buffer if an active track underruns to avoid sending 3204 // previous audio buffer again to effects 3205 if (!mEffectChains.isEmpty()) { 3206 mEffectChains[0]->clearInputBuffer(); 3207 } 3208 3209 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3210 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3211 track->isStopped() || track->isPaused()) { 3212 // We have consumed all the buffers of this track. 3213 // Remove it from the list of active tracks. 3214 // TODO: implement behavior for compressed audio 3215 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3216 size_t framesWritten = mBytesWritten / mFrameSize; 3217 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3218 if (track->isStopped()) { 3219 track->reset(); 3220 } 3221 trackToRemove = track; 3222 } 3223 } else { 3224 // No buffers for this track. Give it a few chances to 3225 // fill a buffer, then remove it from active list. 3226 if (--(track->mRetryCount) <= 0) { 3227 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3228 trackToRemove = track; 3229 } else { 3230 mixerStatus = MIXER_TRACKS_ENABLED; 3231 } 3232 } 3233 } 3234 } 3235 3236 // FIXME merge this with similar code for removing multiple tracks 3237 // remove all the tracks that need to be... 3238 if (CC_UNLIKELY(trackToRemove != 0)) { 3239 tracksToRemove->add(trackToRemove); 3240 mActiveTracks.remove(trackToRemove); 3241 if (!mEffectChains.isEmpty()) { 3242 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3243 trackToRemove->sessionId()); 3244 mEffectChains[0]->decActiveTrackCnt(); 3245 } 3246 if (trackToRemove->isTerminated()) { 3247 removeTrack_l(trackToRemove); 3248 } 3249 } 3250 3251 return mixerStatus; 3252} 3253 3254void AudioFlinger::DirectOutputThread::threadLoop_mix() 3255{ 3256 AudioBufferProvider::Buffer buffer; 3257 size_t frameCount = mFrameCount; 3258 int8_t *curBuf = (int8_t *)mMixBuffer; 3259 // output audio to hardware 3260 while (frameCount) { 3261 buffer.frameCount = frameCount; 3262 mActiveTrack->getNextBuffer(&buffer); 3263 if (CC_UNLIKELY(buffer.raw == NULL)) { 3264 memset(curBuf, 0, frameCount * mFrameSize); 3265 break; 3266 } 3267 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3268 frameCount -= buffer.frameCount; 3269 curBuf += buffer.frameCount * mFrameSize; 3270 mActiveTrack->releaseBuffer(&buffer); 3271 } 3272 sleepTime = 0; 3273 standbyTime = systemTime() + standbyDelay; 3274 mActiveTrack.clear(); 3275 3276} 3277 3278void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3279{ 3280 if (sleepTime == 0) { 3281 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3282 sleepTime = activeSleepTime; 3283 } else { 3284 sleepTime = idleSleepTime; 3285 } 3286 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3287 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3288 sleepTime = 0; 3289 } 3290} 3291 3292// getTrackName_l() must be called with ThreadBase::mLock held 3293int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3294 int sessionId) 3295{ 3296 return 0; 3297} 3298 3299// deleteTrackName_l() must be called with ThreadBase::mLock held 3300void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3301{ 3302} 3303 3304// checkForNewParameters_l() must be called with ThreadBase::mLock held 3305bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3306{ 3307 bool reconfig = false; 3308 3309 while (!mNewParameters.isEmpty()) { 3310 status_t status = NO_ERROR; 3311 String8 keyValuePair = mNewParameters[0]; 3312 AudioParameter param = AudioParameter(keyValuePair); 3313 int value; 3314 3315 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3316 // do not accept frame count changes if tracks are open as the track buffer 3317 // size depends on frame count and correct behavior would not be garantied 3318 // if frame count is changed after track creation 3319 if (!mTracks.isEmpty()) { 3320 status = INVALID_OPERATION; 3321 } else { 3322 reconfig = true; 3323 } 3324 } 3325 if (status == NO_ERROR) { 3326 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3327 keyValuePair.string()); 3328 if (!mStandby && status == INVALID_OPERATION) { 3329 mOutput->stream->common.standby(&mOutput->stream->common); 3330 mStandby = true; 3331 mBytesWritten = 0; 3332 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3333 keyValuePair.string()); 3334 } 3335 if (status == NO_ERROR && reconfig) { 3336 readOutputParameters(); 3337 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3338 } 3339 } 3340 3341 mNewParameters.removeAt(0); 3342 3343 mParamStatus = status; 3344 mParamCond.signal(); 3345 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3346 // already timed out waiting for the status and will never signal the condition. 3347 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3348 } 3349 return reconfig; 3350} 3351 3352uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3353{ 3354 uint32_t time; 3355 if (audio_is_linear_pcm(mFormat)) { 3356 time = PlaybackThread::activeSleepTimeUs(); 3357 } else { 3358 time = 10000; 3359 } 3360 return time; 3361} 3362 3363uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3364{ 3365 uint32_t time; 3366 if (audio_is_linear_pcm(mFormat)) { 3367 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3368 } else { 3369 time = 10000; 3370 } 3371 return time; 3372} 3373 3374uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3375{ 3376 uint32_t time; 3377 if (audio_is_linear_pcm(mFormat)) { 3378 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3379 } else { 3380 time = 10000; 3381 } 3382 return time; 3383} 3384 3385void AudioFlinger::DirectOutputThread::cacheParameters_l() 3386{ 3387 PlaybackThread::cacheParameters_l(); 3388 3389 // use shorter standby delay as on normal output to release 3390 // hardware resources as soon as possible 3391 standbyDelay = microseconds(activeSleepTime*2); 3392} 3393 3394// ---------------------------------------------------------------------------- 3395 3396AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3397 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3398 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3399 DUPLICATING), 3400 mWaitTimeMs(UINT_MAX) 3401{ 3402 addOutputTrack(mainThread); 3403} 3404 3405AudioFlinger::DuplicatingThread::~DuplicatingThread() 3406{ 3407 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3408 mOutputTracks[i]->destroy(); 3409 } 3410} 3411 3412void AudioFlinger::DuplicatingThread::threadLoop_mix() 3413{ 3414 // mix buffers... 3415 if (outputsReady(outputTracks)) { 3416 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3417 } else { 3418 memset(mMixBuffer, 0, mixBufferSize); 3419 } 3420 sleepTime = 0; 3421 writeFrames = mNormalFrameCount; 3422 standbyTime = systemTime() + standbyDelay; 3423} 3424 3425void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3426{ 3427 if (sleepTime == 0) { 3428 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3429 sleepTime = activeSleepTime; 3430 } else { 3431 sleepTime = idleSleepTime; 3432 } 3433 } else if (mBytesWritten != 0) { 3434 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3435 writeFrames = mNormalFrameCount; 3436 memset(mMixBuffer, 0, mixBufferSize); 3437 } else { 3438 // flush remaining overflow buffers in output tracks 3439 writeFrames = 0; 3440 } 3441 sleepTime = 0; 3442 } 3443} 3444 3445void AudioFlinger::DuplicatingThread::threadLoop_write() 3446{ 3447 for (size_t i = 0; i < outputTracks.size(); i++) { 3448 outputTracks[i]->write(mMixBuffer, writeFrames); 3449 } 3450 mBytesWritten += mixBufferSize; 3451} 3452 3453void AudioFlinger::DuplicatingThread::threadLoop_standby() 3454{ 3455 // DuplicatingThread implements standby by stopping all tracks 3456 for (size_t i = 0; i < outputTracks.size(); i++) { 3457 outputTracks[i]->stop(); 3458 } 3459} 3460 3461void AudioFlinger::DuplicatingThread::saveOutputTracks() 3462{ 3463 outputTracks = mOutputTracks; 3464} 3465 3466void AudioFlinger::DuplicatingThread::clearOutputTracks() 3467{ 3468 outputTracks.clear(); 3469} 3470 3471void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3472{ 3473 Mutex::Autolock _l(mLock); 3474 // FIXME explain this formula 3475 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3476 OutputTrack *outputTrack = new OutputTrack(thread, 3477 this, 3478 mSampleRate, 3479 mFormat, 3480 mChannelMask, 3481 frameCount); 3482 if (outputTrack->cblk() != NULL) { 3483 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3484 mOutputTracks.add(outputTrack); 3485 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3486 updateWaitTime_l(); 3487 } 3488} 3489 3490void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3491{ 3492 Mutex::Autolock _l(mLock); 3493 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3494 if (mOutputTracks[i]->thread() == thread) { 3495 mOutputTracks[i]->destroy(); 3496 mOutputTracks.removeAt(i); 3497 updateWaitTime_l(); 3498 return; 3499 } 3500 } 3501 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3502} 3503 3504// caller must hold mLock 3505void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3506{ 3507 mWaitTimeMs = UINT_MAX; 3508 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3509 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3510 if (strong != 0) { 3511 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3512 if (waitTimeMs < mWaitTimeMs) { 3513 mWaitTimeMs = waitTimeMs; 3514 } 3515 } 3516 } 3517} 3518 3519 3520bool AudioFlinger::DuplicatingThread::outputsReady( 3521 const SortedVector< sp<OutputTrack> > &outputTracks) 3522{ 3523 for (size_t i = 0; i < outputTracks.size(); i++) { 3524 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3525 if (thread == 0) { 3526 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3527 outputTracks[i].get()); 3528 return false; 3529 } 3530 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3531 // see note at standby() declaration 3532 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3533 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3534 thread.get()); 3535 return false; 3536 } 3537 } 3538 return true; 3539} 3540 3541uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3542{ 3543 return (mWaitTimeMs * 1000) / 2; 3544} 3545 3546void AudioFlinger::DuplicatingThread::cacheParameters_l() 3547{ 3548 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3549 updateWaitTime_l(); 3550 3551 MixerThread::cacheParameters_l(); 3552} 3553 3554// ---------------------------------------------------------------------------- 3555// Record 3556// ---------------------------------------------------------------------------- 3557 3558AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3559 AudioStreamIn *input, 3560 uint32_t sampleRate, 3561 audio_channel_mask_t channelMask, 3562 audio_io_handle_t id, 3563 audio_devices_t outDevice, 3564 audio_devices_t inDevice, 3565 const sp<NBAIO_Sink>& teeSink) : 3566 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3567 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3568 // mRsmpInIndex and mInputBytes set by readInputParameters() 3569 mReqChannelCount(popcount(channelMask)), 3570 mReqSampleRate(sampleRate), 3571 // mBytesRead is only meaningful while active, and so is cleared in start() 3572 // (but might be better to also clear here for dump?) 3573 mTeeSink(teeSink) 3574{ 3575 snprintf(mName, kNameLength, "AudioIn_%X", id); 3576 3577 readInputParameters(); 3578 3579} 3580 3581 3582AudioFlinger::RecordThread::~RecordThread() 3583{ 3584 delete[] mRsmpInBuffer; 3585 delete mResampler; 3586 delete[] mRsmpOutBuffer; 3587} 3588 3589void AudioFlinger::RecordThread::onFirstRef() 3590{ 3591 run(mName, PRIORITY_URGENT_AUDIO); 3592} 3593 3594status_t AudioFlinger::RecordThread::readyToRun() 3595{ 3596 status_t status = initCheck(); 3597 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3598 return status; 3599} 3600 3601bool AudioFlinger::RecordThread::threadLoop() 3602{ 3603 AudioBufferProvider::Buffer buffer; 3604 sp<RecordTrack> activeTrack; 3605 Vector< sp<EffectChain> > effectChains; 3606 3607 nsecs_t lastWarning = 0; 3608 3609 inputStandBy(); 3610 acquireWakeLock(); 3611 3612 // used to verify we've read at least once before evaluating how many bytes were read 3613 bool readOnce = false; 3614 3615 // start recording 3616 while (!exitPending()) { 3617 3618 processConfigEvents(); 3619 3620 { // scope for mLock 3621 Mutex::Autolock _l(mLock); 3622 checkForNewParameters_l(); 3623 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3624 standby(); 3625 3626 if (exitPending()) { 3627 break; 3628 } 3629 3630 releaseWakeLock_l(); 3631 ALOGV("RecordThread: loop stopping"); 3632 // go to sleep 3633 mWaitWorkCV.wait(mLock); 3634 ALOGV("RecordThread: loop starting"); 3635 acquireWakeLock_l(); 3636 continue; 3637 } 3638 if (mActiveTrack != 0) { 3639 if (mActiveTrack->mState == TrackBase::PAUSING) { 3640 standby(); 3641 mActiveTrack.clear(); 3642 mStartStopCond.broadcast(); 3643 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3644 if (mReqChannelCount != mActiveTrack->channelCount()) { 3645 mActiveTrack.clear(); 3646 mStartStopCond.broadcast(); 3647 } else if (readOnce) { 3648 // record start succeeds only if first read from audio input 3649 // succeeds 3650 if (mBytesRead >= 0) { 3651 mActiveTrack->mState = TrackBase::ACTIVE; 3652 } else { 3653 mActiveTrack.clear(); 3654 } 3655 mStartStopCond.broadcast(); 3656 } 3657 mStandby = false; 3658 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3659 removeTrack_l(mActiveTrack); 3660 mActiveTrack.clear(); 3661 } 3662 } 3663 lockEffectChains_l(effectChains); 3664 } 3665 3666 if (mActiveTrack != 0) { 3667 if (mActiveTrack->mState != TrackBase::ACTIVE && 3668 mActiveTrack->mState != TrackBase::RESUMING) { 3669 unlockEffectChains(effectChains); 3670 usleep(kRecordThreadSleepUs); 3671 continue; 3672 } 3673 for (size_t i = 0; i < effectChains.size(); i ++) { 3674 effectChains[i]->process_l(); 3675 } 3676 3677 buffer.frameCount = mFrameCount; 3678 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3679 readOnce = true; 3680 size_t framesOut = buffer.frameCount; 3681 if (mResampler == NULL) { 3682 // no resampling 3683 while (framesOut) { 3684 size_t framesIn = mFrameCount - mRsmpInIndex; 3685 if (framesIn) { 3686 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3687 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3688 mActiveTrack->mFrameSize; 3689 if (framesIn > framesOut) 3690 framesIn = framesOut; 3691 mRsmpInIndex += framesIn; 3692 framesOut -= framesIn; 3693 if (mChannelCount == mReqChannelCount || 3694 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3695 memcpy(dst, src, framesIn * mFrameSize); 3696 } else { 3697 if (mChannelCount == 1) { 3698 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3699 (int16_t *)src, framesIn); 3700 } else { 3701 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3702 (int16_t *)src, framesIn); 3703 } 3704 } 3705 } 3706 if (framesOut && mFrameCount == mRsmpInIndex) { 3707 void *readInto; 3708 if (framesOut == mFrameCount && 3709 (mChannelCount == mReqChannelCount || 3710 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3711 readInto = buffer.raw; 3712 framesOut = 0; 3713 } else { 3714 readInto = mRsmpInBuffer; 3715 mRsmpInIndex = 0; 3716 } 3717 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3718 if (mBytesRead <= 0) { 3719 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3720 { 3721 ALOGE("Error reading audio input"); 3722 // Force input into standby so that it tries to 3723 // recover at next read attempt 3724 inputStandBy(); 3725 usleep(kRecordThreadSleepUs); 3726 } 3727 mRsmpInIndex = mFrameCount; 3728 framesOut = 0; 3729 buffer.frameCount = 0; 3730 } else if (mTeeSink != 0) { 3731 (void) mTeeSink->write(readInto, 3732 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3733 } 3734 } 3735 } 3736 } else { 3737 // resampling 3738 3739 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3740 // alter output frame count as if we were expecting stereo samples 3741 if (mChannelCount == 1 && mReqChannelCount == 1) { 3742 framesOut >>= 1; 3743 } 3744 mResampler->resample(mRsmpOutBuffer, framesOut, 3745 this /* AudioBufferProvider* */); 3746 // ditherAndClamp() works as long as all buffers returned by 3747 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3748 if (mChannelCount == 2 && mReqChannelCount == 1) { 3749 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3750 // the resampler always outputs stereo samples: 3751 // do post stereo to mono conversion 3752 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3753 framesOut); 3754 } else { 3755 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3756 } 3757 3758 } 3759 if (mFramestoDrop == 0) { 3760 mActiveTrack->releaseBuffer(&buffer); 3761 } else { 3762 if (mFramestoDrop > 0) { 3763 mFramestoDrop -= buffer.frameCount; 3764 if (mFramestoDrop <= 0) { 3765 clearSyncStartEvent(); 3766 } 3767 } else { 3768 mFramestoDrop += buffer.frameCount; 3769 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3770 mSyncStartEvent->isCancelled()) { 3771 ALOGW("Synced record %s, session %d, trigger session %d", 3772 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3773 mActiveTrack->sessionId(), 3774 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3775 clearSyncStartEvent(); 3776 } 3777 } 3778 } 3779 mActiveTrack->clearOverflow(); 3780 } 3781 // client isn't retrieving buffers fast enough 3782 else { 3783 if (!mActiveTrack->setOverflow()) { 3784 nsecs_t now = systemTime(); 3785 if ((now - lastWarning) > kWarningThrottleNs) { 3786 ALOGW("RecordThread: buffer overflow"); 3787 lastWarning = now; 3788 } 3789 } 3790 // Release the processor for a while before asking for a new buffer. 3791 // This will give the application more chance to read from the buffer and 3792 // clear the overflow. 3793 usleep(kRecordThreadSleepUs); 3794 } 3795 } 3796 // enable changes in effect chain 3797 unlockEffectChains(effectChains); 3798 effectChains.clear(); 3799 } 3800 3801 standby(); 3802 3803 { 3804 Mutex::Autolock _l(mLock); 3805 mActiveTrack.clear(); 3806 mStartStopCond.broadcast(); 3807 } 3808 3809 releaseWakeLock(); 3810 3811 ALOGV("RecordThread %p exiting", this); 3812 return false; 3813} 3814 3815void AudioFlinger::RecordThread::standby() 3816{ 3817 if (!mStandby) { 3818 inputStandBy(); 3819 mStandby = true; 3820 } 3821} 3822 3823void AudioFlinger::RecordThread::inputStandBy() 3824{ 3825 mInput->stream->common.standby(&mInput->stream->common); 3826} 3827 3828sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3829 const sp<AudioFlinger::Client>& client, 3830 uint32_t sampleRate, 3831 audio_format_t format, 3832 audio_channel_mask_t channelMask, 3833 size_t frameCount, 3834 int sessionId, 3835 IAudioFlinger::track_flags_t flags, 3836 pid_t tid, 3837 status_t *status) 3838{ 3839 sp<RecordTrack> track; 3840 status_t lStatus; 3841 3842 lStatus = initCheck(); 3843 if (lStatus != NO_ERROR) { 3844 ALOGE("Audio driver not initialized."); 3845 goto Exit; 3846 } 3847 3848 // FIXME use flags and tid similar to createTrack_l() 3849 3850 { // scope for mLock 3851 Mutex::Autolock _l(mLock); 3852 3853 track = new RecordTrack(this, client, sampleRate, 3854 format, channelMask, frameCount, sessionId); 3855 3856 if (track->getCblk() == 0) { 3857 lStatus = NO_MEMORY; 3858 goto Exit; 3859 } 3860 mTracks.add(track); 3861 3862 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3863 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3864 mAudioFlinger->btNrecIsOff(); 3865 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3866 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3867 } 3868 lStatus = NO_ERROR; 3869 3870Exit: 3871 if (status) { 3872 *status = lStatus; 3873 } 3874 return track; 3875} 3876 3877status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3878 AudioSystem::sync_event_t event, 3879 int triggerSession) 3880{ 3881 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3882 sp<ThreadBase> strongMe = this; 3883 status_t status = NO_ERROR; 3884 3885 if (event == AudioSystem::SYNC_EVENT_NONE) { 3886 clearSyncStartEvent(); 3887 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3888 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3889 triggerSession, 3890 recordTrack->sessionId(), 3891 syncStartEventCallback, 3892 this); 3893 // Sync event can be cancelled by the trigger session if the track is not in a 3894 // compatible state in which case we start record immediately 3895 if (mSyncStartEvent->isCancelled()) { 3896 clearSyncStartEvent(); 3897 } else { 3898 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3899 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3900 } 3901 } 3902 3903 { 3904 AutoMutex lock(mLock); 3905 if (mActiveTrack != 0) { 3906 if (recordTrack != mActiveTrack.get()) { 3907 status = -EBUSY; 3908 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3909 mActiveTrack->mState = TrackBase::ACTIVE; 3910 } 3911 return status; 3912 } 3913 3914 recordTrack->mState = TrackBase::IDLE; 3915 mActiveTrack = recordTrack; 3916 mLock.unlock(); 3917 status_t status = AudioSystem::startInput(mId); 3918 mLock.lock(); 3919 if (status != NO_ERROR) { 3920 mActiveTrack.clear(); 3921 clearSyncStartEvent(); 3922 return status; 3923 } 3924 mRsmpInIndex = mFrameCount; 3925 mBytesRead = 0; 3926 if (mResampler != NULL) { 3927 mResampler->reset(); 3928 } 3929 mActiveTrack->mState = TrackBase::RESUMING; 3930 // signal thread to start 3931 ALOGV("Signal record thread"); 3932 mWaitWorkCV.broadcast(); 3933 // do not wait for mStartStopCond if exiting 3934 if (exitPending()) { 3935 mActiveTrack.clear(); 3936 status = INVALID_OPERATION; 3937 goto startError; 3938 } 3939 mStartStopCond.wait(mLock); 3940 if (mActiveTrack == 0) { 3941 ALOGV("Record failed to start"); 3942 status = BAD_VALUE; 3943 goto startError; 3944 } 3945 ALOGV("Record started OK"); 3946 return status; 3947 } 3948startError: 3949 AudioSystem::stopInput(mId); 3950 clearSyncStartEvent(); 3951 return status; 3952} 3953 3954void AudioFlinger::RecordThread::clearSyncStartEvent() 3955{ 3956 if (mSyncStartEvent != 0) { 3957 mSyncStartEvent->cancel(); 3958 } 3959 mSyncStartEvent.clear(); 3960 mFramestoDrop = 0; 3961} 3962 3963void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3964{ 3965 sp<SyncEvent> strongEvent = event.promote(); 3966 3967 if (strongEvent != 0) { 3968 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3969 me->handleSyncStartEvent(strongEvent); 3970 } 3971} 3972 3973void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 3974{ 3975 if (event == mSyncStartEvent) { 3976 // TODO: use actual buffer filling status instead of 2 buffers when info is available 3977 // from audio HAL 3978 mFramestoDrop = mFrameCount * 2; 3979 } 3980} 3981 3982bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 3983 ALOGV("RecordThread::stop"); 3984 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 3985 return false; 3986 } 3987 recordTrack->mState = TrackBase::PAUSING; 3988 // do not wait for mStartStopCond if exiting 3989 if (exitPending()) { 3990 return true; 3991 } 3992 mStartStopCond.wait(mLock); 3993 // if we have been restarted, recordTrack == mActiveTrack.get() here 3994 if (exitPending() || recordTrack != mActiveTrack.get()) { 3995 ALOGV("Record stopped OK"); 3996 return true; 3997 } 3998 return false; 3999} 4000 4001bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4002{ 4003 return false; 4004} 4005 4006status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4007{ 4008#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4009 if (!isValidSyncEvent(event)) { 4010 return BAD_VALUE; 4011 } 4012 4013 int eventSession = event->triggerSession(); 4014 status_t ret = NAME_NOT_FOUND; 4015 4016 Mutex::Autolock _l(mLock); 4017 4018 for (size_t i = 0; i < mTracks.size(); i++) { 4019 sp<RecordTrack> track = mTracks[i]; 4020 if (eventSession == track->sessionId()) { 4021 (void) track->setSyncEvent(event); 4022 ret = NO_ERROR; 4023 } 4024 } 4025 return ret; 4026#else 4027 return BAD_VALUE; 4028#endif 4029} 4030 4031// destroyTrack_l() must be called with ThreadBase::mLock held 4032void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4033{ 4034 track->mState = TrackBase::TERMINATED; 4035 // active tracks are removed by threadLoop() 4036 if (mActiveTrack != track) { 4037 removeTrack_l(track); 4038 } 4039} 4040 4041void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4042{ 4043 mTracks.remove(track); 4044 // need anything related to effects here? 4045} 4046 4047void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4048{ 4049 dumpInternals(fd, args); 4050 dumpTracks(fd, args); 4051 dumpEffectChains(fd, args); 4052} 4053 4054void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4055{ 4056 const size_t SIZE = 256; 4057 char buffer[SIZE]; 4058 String8 result; 4059 4060 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4061 result.append(buffer); 4062 4063 if (mActiveTrack != 0) { 4064 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4065 result.append(buffer); 4066 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4067 result.append(buffer); 4068 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4069 result.append(buffer); 4070 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4071 result.append(buffer); 4072 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4073 result.append(buffer); 4074 } else { 4075 result.append("No active record client\n"); 4076 } 4077 4078 write(fd, result.string(), result.size()); 4079 4080 dumpBase(fd, args); 4081} 4082 4083void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4084{ 4085 const size_t SIZE = 256; 4086 char buffer[SIZE]; 4087 String8 result; 4088 4089 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4090 result.append(buffer); 4091 RecordTrack::appendDumpHeader(result); 4092 for (size_t i = 0; i < mTracks.size(); ++i) { 4093 sp<RecordTrack> track = mTracks[i]; 4094 if (track != 0) { 4095 track->dump(buffer, SIZE); 4096 result.append(buffer); 4097 } 4098 } 4099 4100 if (mActiveTrack != 0) { 4101 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4102 result.append(buffer); 4103 RecordTrack::appendDumpHeader(result); 4104 mActiveTrack->dump(buffer, SIZE); 4105 result.append(buffer); 4106 4107 } 4108 write(fd, result.string(), result.size()); 4109} 4110 4111// AudioBufferProvider interface 4112status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4113{ 4114 size_t framesReq = buffer->frameCount; 4115 size_t framesReady = mFrameCount - mRsmpInIndex; 4116 int channelCount; 4117 4118 if (framesReady == 0) { 4119 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4120 if (mBytesRead <= 0) { 4121 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4122 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4123 // Force input into standby so that it tries to 4124 // recover at next read attempt 4125 inputStandBy(); 4126 usleep(kRecordThreadSleepUs); 4127 } 4128 buffer->raw = NULL; 4129 buffer->frameCount = 0; 4130 return NOT_ENOUGH_DATA; 4131 } 4132 mRsmpInIndex = 0; 4133 framesReady = mFrameCount; 4134 } 4135 4136 if (framesReq > framesReady) { 4137 framesReq = framesReady; 4138 } 4139 4140 if (mChannelCount == 1 && mReqChannelCount == 2) { 4141 channelCount = 1; 4142 } else { 4143 channelCount = 2; 4144 } 4145 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4146 buffer->frameCount = framesReq; 4147 return NO_ERROR; 4148} 4149 4150// AudioBufferProvider interface 4151void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4152{ 4153 mRsmpInIndex += buffer->frameCount; 4154 buffer->frameCount = 0; 4155} 4156 4157bool AudioFlinger::RecordThread::checkForNewParameters_l() 4158{ 4159 bool reconfig = false; 4160 4161 while (!mNewParameters.isEmpty()) { 4162 status_t status = NO_ERROR; 4163 String8 keyValuePair = mNewParameters[0]; 4164 AudioParameter param = AudioParameter(keyValuePair); 4165 int value; 4166 audio_format_t reqFormat = mFormat; 4167 uint32_t reqSamplingRate = mReqSampleRate; 4168 uint32_t reqChannelCount = mReqChannelCount; 4169 4170 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4171 reqSamplingRate = value; 4172 reconfig = true; 4173 } 4174 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4175 reqFormat = (audio_format_t) value; 4176 reconfig = true; 4177 } 4178 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4179 reqChannelCount = popcount(value); 4180 reconfig = true; 4181 } 4182 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4183 // do not accept frame count changes if tracks are open as the track buffer 4184 // size depends on frame count and correct behavior would not be guaranteed 4185 // if frame count is changed after track creation 4186 if (mActiveTrack != 0) { 4187 status = INVALID_OPERATION; 4188 } else { 4189 reconfig = true; 4190 } 4191 } 4192 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4193 // forward device change to effects that have requested to be 4194 // aware of attached audio device. 4195 for (size_t i = 0; i < mEffectChains.size(); i++) { 4196 mEffectChains[i]->setDevice_l(value); 4197 } 4198 4199 // store input device and output device but do not forward output device to audio HAL. 4200 // Note that status is ignored by the caller for output device 4201 // (see AudioFlinger::setParameters() 4202 if (audio_is_output_devices(value)) { 4203 mOutDevice = value; 4204 status = BAD_VALUE; 4205 } else { 4206 mInDevice = value; 4207 // disable AEC and NS if the device is a BT SCO headset supporting those 4208 // pre processings 4209 if (mTracks.size() > 0) { 4210 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4211 mAudioFlinger->btNrecIsOff(); 4212 for (size_t i = 0; i < mTracks.size(); i++) { 4213 sp<RecordTrack> track = mTracks[i]; 4214 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4215 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4216 } 4217 } 4218 } 4219 } 4220 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4221 mAudioSource != (audio_source_t)value) { 4222 // forward device change to effects that have requested to be 4223 // aware of attached audio device. 4224 for (size_t i = 0; i < mEffectChains.size(); i++) { 4225 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4226 } 4227 mAudioSource = (audio_source_t)value; 4228 } 4229 if (status == NO_ERROR) { 4230 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4231 keyValuePair.string()); 4232 if (status == INVALID_OPERATION) { 4233 inputStandBy(); 4234 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4235 keyValuePair.string()); 4236 } 4237 if (reconfig) { 4238 if (status == BAD_VALUE && 4239 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4240 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4241 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4242 <= (2 * reqSamplingRate)) && 4243 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4244 <= FCC_2 && 4245 (reqChannelCount <= FCC_2)) { 4246 status = NO_ERROR; 4247 } 4248 if (status == NO_ERROR) { 4249 readInputParameters(); 4250 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4251 } 4252 } 4253 } 4254 4255 mNewParameters.removeAt(0); 4256 4257 mParamStatus = status; 4258 mParamCond.signal(); 4259 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4260 // already timed out waiting for the status and will never signal the condition. 4261 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4262 } 4263 return reconfig; 4264} 4265 4266String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4267{ 4268 char *s; 4269 String8 out_s8 = String8(); 4270 4271 Mutex::Autolock _l(mLock); 4272 if (initCheck() != NO_ERROR) { 4273 return out_s8; 4274 } 4275 4276 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4277 out_s8 = String8(s); 4278 free(s); 4279 return out_s8; 4280} 4281 4282void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4283 AudioSystem::OutputDescriptor desc; 4284 void *param2 = NULL; 4285 4286 switch (event) { 4287 case AudioSystem::INPUT_OPENED: 4288 case AudioSystem::INPUT_CONFIG_CHANGED: 4289 desc.channels = mChannelMask; 4290 desc.samplingRate = mSampleRate; 4291 desc.format = mFormat; 4292 desc.frameCount = mFrameCount; 4293 desc.latency = 0; 4294 param2 = &desc; 4295 break; 4296 4297 case AudioSystem::INPUT_CLOSED: 4298 default: 4299 break; 4300 } 4301 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4302} 4303 4304void AudioFlinger::RecordThread::readInputParameters() 4305{ 4306 delete mRsmpInBuffer; 4307 // mRsmpInBuffer is always assigned a new[] below 4308 delete mRsmpOutBuffer; 4309 mRsmpOutBuffer = NULL; 4310 delete mResampler; 4311 mResampler = NULL; 4312 4313 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4314 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4315 mChannelCount = (uint16_t)popcount(mChannelMask); 4316 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4317 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4318 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4319 mFrameCount = mInputBytes / mFrameSize; 4320 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4321 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4322 4323 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4324 { 4325 int channelCount; 4326 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4327 // stereo to mono post process as the resampler always outputs stereo. 4328 if (mChannelCount == 1 && mReqChannelCount == 2) { 4329 channelCount = 1; 4330 } else { 4331 channelCount = 2; 4332 } 4333 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4334 mResampler->setSampleRate(mSampleRate); 4335 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4336 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4337 4338 // optmization: if mono to mono, alter input frame count as if we were inputing 4339 // stereo samples 4340 if (mChannelCount == 1 && mReqChannelCount == 1) { 4341 mFrameCount >>= 1; 4342 } 4343 4344 } 4345 mRsmpInIndex = mFrameCount; 4346} 4347 4348unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4349{ 4350 Mutex::Autolock _l(mLock); 4351 if (initCheck() != NO_ERROR) { 4352 return 0; 4353 } 4354 4355 return mInput->stream->get_input_frames_lost(mInput->stream); 4356} 4357 4358uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4359{ 4360 Mutex::Autolock _l(mLock); 4361 uint32_t result = 0; 4362 if (getEffectChain_l(sessionId) != 0) { 4363 result = EFFECT_SESSION; 4364 } 4365 4366 for (size_t i = 0; i < mTracks.size(); ++i) { 4367 if (sessionId == mTracks[i]->sessionId()) { 4368 result |= TRACK_SESSION; 4369 break; 4370 } 4371 } 4372 4373 return result; 4374} 4375 4376KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4377{ 4378 KeyedVector<int, bool> ids; 4379 Mutex::Autolock _l(mLock); 4380 for (size_t j = 0; j < mTracks.size(); ++j) { 4381 sp<RecordThread::RecordTrack> track = mTracks[j]; 4382 int sessionId = track->sessionId(); 4383 if (ids.indexOfKey(sessionId) < 0) { 4384 ids.add(sessionId, true); 4385 } 4386 } 4387 return ids; 4388} 4389 4390AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4391{ 4392 Mutex::Autolock _l(mLock); 4393 AudioStreamIn *input = mInput; 4394 mInput = NULL; 4395 return input; 4396} 4397 4398// this method must always be called either with ThreadBase mLock held or inside the thread loop 4399audio_stream_t* AudioFlinger::RecordThread::stream() const 4400{ 4401 if (mInput == NULL) { 4402 return NULL; 4403 } 4404 return &mInput->stream->common; 4405} 4406 4407status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4408{ 4409 // only one chain per input thread 4410 if (mEffectChains.size() != 0) { 4411 return INVALID_OPERATION; 4412 } 4413 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4414 4415 chain->setInBuffer(NULL); 4416 chain->setOutBuffer(NULL); 4417 4418 checkSuspendOnAddEffectChain_l(chain); 4419 4420 mEffectChains.add(chain); 4421 4422 return NO_ERROR; 4423} 4424 4425size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4426{ 4427 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4428 ALOGW_IF(mEffectChains.size() != 1, 4429 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4430 chain.get(), mEffectChains.size(), this); 4431 if (mEffectChains.size() == 1) { 4432 mEffectChains.removeAt(0); 4433 } 4434 return 0; 4435} 4436 4437}; // namespace android 4438