Threads.cpp revision 3051df27261e9952c0e642dec548515250e85f6a
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 mNBLogWriter->logf("createTrack_l isFast=%d caller=%d", 1200 (*flags & IAudioFlinger::TRACK_FAST) != 0, IPCThreadState::self()->getCallingPid()); 1201 1202 // all tracks in same audio session must share the same routing strategy otherwise 1203 // conflicts will happen when tracks are moved from one output to another by audio policy 1204 // manager 1205 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1206 for (size_t i = 0; i < mTracks.size(); ++i) { 1207 sp<Track> t = mTracks[i]; 1208 if (t != 0 && !t->isOutputTrack()) { 1209 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1210 if (sessionId == t->sessionId() && strategy != actual) { 1211 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1212 strategy, actual); 1213 lStatus = BAD_VALUE; 1214 goto Exit; 1215 } 1216 } 1217 } 1218 1219 if (!isTimed) { 1220 track = new Track(this, client, streamType, sampleRate, format, 1221 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1222 } else { 1223 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1224 channelMask, frameCount, sharedBuffer, sessionId); 1225 } 1226 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1227 lStatus = NO_MEMORY; 1228 goto Exit; 1229 } 1230 mTracks.add(track); 1231 1232 sp<EffectChain> chain = getEffectChain_l(sessionId); 1233 if (chain != 0) { 1234 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1235 track->setMainBuffer(chain->inBuffer()); 1236 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1237 chain->incTrackCnt(); 1238 } 1239 1240 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1241 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1242 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1243 // so ask activity manager to do this on our behalf 1244 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1245 } 1246 } 1247 1248 lStatus = NO_ERROR; 1249 1250Exit: 1251 if (status) { 1252 *status = lStatus; 1253 } 1254 return track; 1255} 1256 1257uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1258{ 1259 return latency; 1260} 1261 1262uint32_t AudioFlinger::PlaybackThread::latency() const 1263{ 1264 Mutex::Autolock _l(mLock); 1265 return latency_l(); 1266} 1267uint32_t AudioFlinger::PlaybackThread::latency_l() const 1268{ 1269 if (initCheck() == NO_ERROR) { 1270 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1271 } else { 1272 return 0; 1273 } 1274} 1275 1276void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1277{ 1278 Mutex::Autolock _l(mLock); 1279 // Don't apply master volume in SW if our HAL can do it for us. 1280 if (mOutput && mOutput->audioHwDev && 1281 mOutput->audioHwDev->canSetMasterVolume()) { 1282 mMasterVolume = 1.0; 1283 } else { 1284 mMasterVolume = value; 1285 } 1286} 1287 1288void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1289{ 1290 Mutex::Autolock _l(mLock); 1291 // Don't apply master mute in SW if our HAL can do it for us. 1292 if (mOutput && mOutput->audioHwDev && 1293 mOutput->audioHwDev->canSetMasterMute()) { 1294 mMasterMute = false; 1295 } else { 1296 mMasterMute = muted; 1297 } 1298} 1299 1300void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1301{ 1302 Mutex::Autolock _l(mLock); 1303 mStreamTypes[stream].volume = value; 1304} 1305 1306void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1307{ 1308 Mutex::Autolock _l(mLock); 1309 mStreamTypes[stream].mute = muted; 1310} 1311 1312float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1313{ 1314 Mutex::Autolock _l(mLock); 1315 return mStreamTypes[stream].volume; 1316} 1317 1318// addTrack_l() must be called with ThreadBase::mLock held 1319status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1320{ 1321 mNBLogWriter->logf("addTrack_l mName=%d mFastIndex=%d caller=%d", track->mName, 1322 track->mFastIndex, IPCThreadState::self()->getCallingPid()); 1323 status_t status = ALREADY_EXISTS; 1324 1325 // set retry count for buffer fill 1326 track->mRetryCount = kMaxTrackStartupRetries; 1327 if (mActiveTracks.indexOf(track) < 0) { 1328 // the track is newly added, make sure it fills up all its 1329 // buffers before playing. This is to ensure the client will 1330 // effectively get the latency it requested. 1331 track->mFillingUpStatus = Track::FS_FILLING; 1332 track->mResetDone = false; 1333 track->mPresentationCompleteFrames = 0; 1334 mActiveTracks.add(track); 1335 if (track->mainBuffer() != mMixBuffer) { 1336 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1337 if (chain != 0) { 1338 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1339 track->sessionId()); 1340 chain->incActiveTrackCnt(); 1341 } 1342 } 1343 1344 status = NO_ERROR; 1345 } 1346 1347 ALOGV("mWaitWorkCV.broadcast"); 1348 mWaitWorkCV.broadcast(); 1349 1350 return status; 1351} 1352 1353// destroyTrack_l() must be called with ThreadBase::mLock held 1354void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1355{ 1356 mNBLogWriter->logTimestamp(); 1357 mNBLogWriter->logf("destroyTrack_l mName=%d mFastIndex=%d mClientPid=%d", track->mName, 1358 track->mFastIndex, track->mClient != 0 ? track->mClient->pid() : -1); 1359 track->mState = TrackBase::TERMINATED; 1360 // active tracks are removed by threadLoop() 1361 if (mActiveTracks.indexOf(track) < 0) { 1362 removeTrack_l(track); 1363 } 1364} 1365 1366void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1367{ 1368 mNBLogWriter->logTimestamp(); 1369 mNBLogWriter->logf("removeTrack_l mName=%d mFastIndex=%d clientPid=%d", track->mName, 1370 track->mFastIndex, track->mClient != 0 ? track->mClient->pid() : -1); 1371 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1372 mTracks.remove(track); 1373 deleteTrackName_l(track->name()); 1374 // redundant as track is about to be destroyed, for dumpsys only 1375 track->mName = -1; 1376 if (track->isFastTrack()) { 1377 int index = track->mFastIndex; 1378 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1379 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1380 mFastTrackAvailMask |= 1 << index; 1381 // redundant as track is about to be destroyed, for dumpsys only 1382 track->mFastIndex = -1; 1383 } 1384 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1385 if (chain != 0) { 1386 chain->decTrackCnt(); 1387 } 1388} 1389 1390String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1391{ 1392 String8 out_s8 = String8(""); 1393 char *s; 1394 1395 Mutex::Autolock _l(mLock); 1396 if (initCheck() != NO_ERROR) { 1397 return out_s8; 1398 } 1399 1400 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1401 out_s8 = String8(s); 1402 free(s); 1403 return out_s8; 1404} 1405 1406// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1407void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1408 AudioSystem::OutputDescriptor desc; 1409 void *param2 = NULL; 1410 1411 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1412 param); 1413 1414 switch (event) { 1415 case AudioSystem::OUTPUT_OPENED: 1416 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1417 desc.channels = mChannelMask; 1418 desc.samplingRate = mSampleRate; 1419 desc.format = mFormat; 1420 desc.frameCount = mNormalFrameCount; // FIXME see 1421 // AudioFlinger::frameCount(audio_io_handle_t) 1422 desc.latency = latency(); 1423 param2 = &desc; 1424 break; 1425 1426 case AudioSystem::STREAM_CONFIG_CHANGED: 1427 param2 = ¶m; 1428 case AudioSystem::OUTPUT_CLOSED: 1429 default: 1430 break; 1431 } 1432 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1433} 1434 1435void AudioFlinger::PlaybackThread::readOutputParameters() 1436{ 1437 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1438 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1439 mChannelCount = (uint16_t)popcount(mChannelMask); 1440 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1441 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1442 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1443 if (mFrameCount & 15) { 1444 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1445 mFrameCount); 1446 } 1447 1448 // Calculate size of normal mix buffer relative to the HAL output buffer size 1449 double multiplier = 1.0; 1450 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1451 kUseFastMixer == FastMixer_Dynamic)) { 1452 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1453 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1454 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1455 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1456 maxNormalFrameCount = maxNormalFrameCount & ~15; 1457 if (maxNormalFrameCount < minNormalFrameCount) { 1458 maxNormalFrameCount = minNormalFrameCount; 1459 } 1460 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1461 if (multiplier <= 1.0) { 1462 multiplier = 1.0; 1463 } else if (multiplier <= 2.0) { 1464 if (2 * mFrameCount <= maxNormalFrameCount) { 1465 multiplier = 2.0; 1466 } else { 1467 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1468 } 1469 } else { 1470 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1471 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1472 // track, but we sometimes have to do this to satisfy the maximum frame count 1473 // constraint) 1474 // FIXME this rounding up should not be done if no HAL SRC 1475 uint32_t truncMult = (uint32_t) multiplier; 1476 if ((truncMult & 1)) { 1477 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1478 ++truncMult; 1479 } 1480 } 1481 multiplier = (double) truncMult; 1482 } 1483 } 1484 mNormalFrameCount = multiplier * mFrameCount; 1485 // round up to nearest 16 frames to satisfy AudioMixer 1486 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1487 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1488 mNormalFrameCount); 1489 1490 delete[] mMixBuffer; 1491 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1492 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1493 1494 // force reconfiguration of effect chains and engines to take new buffer size and audio 1495 // parameters into account 1496 // Note that mLock is not held when readOutputParameters() is called from the constructor 1497 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1498 // matter. 1499 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1500 Vector< sp<EffectChain> > effectChains = mEffectChains; 1501 for (size_t i = 0; i < effectChains.size(); i ++) { 1502 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1503 } 1504} 1505 1506 1507status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1508{ 1509 if (halFrames == NULL || dspFrames == NULL) { 1510 return BAD_VALUE; 1511 } 1512 Mutex::Autolock _l(mLock); 1513 if (initCheck() != NO_ERROR) { 1514 return INVALID_OPERATION; 1515 } 1516 size_t framesWritten = mBytesWritten / mFrameSize; 1517 *halFrames = framesWritten; 1518 1519 if (isSuspended()) { 1520 // return an estimation of rendered frames when the output is suspended 1521 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1522 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1523 return NO_ERROR; 1524 } else { 1525 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1526 } 1527} 1528 1529uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1530{ 1531 Mutex::Autolock _l(mLock); 1532 uint32_t result = 0; 1533 if (getEffectChain_l(sessionId) != 0) { 1534 result = EFFECT_SESSION; 1535 } 1536 1537 for (size_t i = 0; i < mTracks.size(); ++i) { 1538 sp<Track> track = mTracks[i]; 1539 if (sessionId == track->sessionId() && !track->isInvalid()) { 1540 result |= TRACK_SESSION; 1541 break; 1542 } 1543 } 1544 1545 return result; 1546} 1547 1548uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1549{ 1550 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1551 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1552 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1553 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1554 } 1555 for (size_t i = 0; i < mTracks.size(); i++) { 1556 sp<Track> track = mTracks[i]; 1557 if (sessionId == track->sessionId() && !track->isInvalid()) { 1558 return AudioSystem::getStrategyForStream(track->streamType()); 1559 } 1560 } 1561 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1562} 1563 1564 1565AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1566{ 1567 Mutex::Autolock _l(mLock); 1568 return mOutput; 1569} 1570 1571AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1572{ 1573 Mutex::Autolock _l(mLock); 1574 AudioStreamOut *output = mOutput; 1575 mOutput = NULL; 1576 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1577 // must push a NULL and wait for ack 1578 mOutputSink.clear(); 1579 mPipeSink.clear(); 1580 mNormalSink.clear(); 1581 return output; 1582} 1583 1584// this method must always be called either with ThreadBase mLock held or inside the thread loop 1585audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1586{ 1587 if (mOutput == NULL) { 1588 return NULL; 1589 } 1590 return &mOutput->stream->common; 1591} 1592 1593uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1594{ 1595 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1596} 1597 1598status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1599{ 1600 if (!isValidSyncEvent(event)) { 1601 return BAD_VALUE; 1602 } 1603 1604 Mutex::Autolock _l(mLock); 1605 1606 for (size_t i = 0; i < mTracks.size(); ++i) { 1607 sp<Track> track = mTracks[i]; 1608 if (event->triggerSession() == track->sessionId()) { 1609 (void) track->setSyncEvent(event); 1610 return NO_ERROR; 1611 } 1612 } 1613 1614 return NAME_NOT_FOUND; 1615} 1616 1617bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1618{ 1619 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1620} 1621 1622void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1623 const Vector< sp<Track> >& tracksToRemove) 1624{ 1625 size_t count = tracksToRemove.size(); 1626 if (CC_UNLIKELY(count)) { 1627 for (size_t i = 0 ; i < count ; i++) { 1628 const sp<Track>& track = tracksToRemove.itemAt(i); 1629 if ((track->sharedBuffer() != 0) && 1630 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1631 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1632 } 1633 } 1634 } 1635 1636} 1637 1638void AudioFlinger::PlaybackThread::checkSilentMode_l() 1639{ 1640 if (!mMasterMute) { 1641 char value[PROPERTY_VALUE_MAX]; 1642 if (property_get("ro.audio.silent", value, "0") > 0) { 1643 char *endptr; 1644 unsigned long ul = strtoul(value, &endptr, 0); 1645 if (*endptr == '\0' && ul != 0) { 1646 ALOGD("Silence is golden"); 1647 // The setprop command will not allow a property to be changed after 1648 // the first time it is set, so we don't have to worry about un-muting. 1649 setMasterMute_l(true); 1650 } 1651 } 1652 } 1653} 1654 1655// shared by MIXER and DIRECT, overridden by DUPLICATING 1656void AudioFlinger::PlaybackThread::threadLoop_write() 1657{ 1658 // FIXME rewrite to reduce number of system calls 1659 mLastWriteTime = systemTime(); 1660 mInWrite = true; 1661 int bytesWritten; 1662 1663 // If an NBAIO sink is present, use it to write the normal mixer's submix 1664 if (mNormalSink != 0) { 1665#define mBitShift 2 // FIXME 1666 size_t count = mixBufferSize >> mBitShift; 1667 ATRACE_BEGIN("write"); 1668 // update the setpoint when AudioFlinger::mScreenState changes 1669 uint32_t screenState = AudioFlinger::mScreenState; 1670 if (screenState != mScreenState) { 1671 mScreenState = screenState; 1672 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1673 if (pipe != NULL) { 1674 pipe->setAvgFrames((mScreenState & 1) ? 1675 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1676 } 1677 } 1678 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1679 ATRACE_END(); 1680 if (framesWritten > 0) { 1681 bytesWritten = framesWritten << mBitShift; 1682 } else { 1683 bytesWritten = framesWritten; 1684 } 1685 // otherwise use the HAL / AudioStreamOut directly 1686 } else { 1687 // Direct output thread. 1688 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1689 } 1690 1691 if (bytesWritten > 0) { 1692 mBytesWritten += mixBufferSize; 1693 } 1694 mNumWrites++; 1695 mInWrite = false; 1696} 1697 1698/* 1699The derived values that are cached: 1700 - mixBufferSize from frame count * frame size 1701 - activeSleepTime from activeSleepTimeUs() 1702 - idleSleepTime from idleSleepTimeUs() 1703 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1704 - maxPeriod from frame count and sample rate (MIXER only) 1705 1706The parameters that affect these derived values are: 1707 - frame count 1708 - frame size 1709 - sample rate 1710 - device type: A2DP or not 1711 - device latency 1712 - format: PCM or not 1713 - active sleep time 1714 - idle sleep time 1715*/ 1716 1717void AudioFlinger::PlaybackThread::cacheParameters_l() 1718{ 1719 mixBufferSize = mNormalFrameCount * mFrameSize; 1720 activeSleepTime = activeSleepTimeUs(); 1721 idleSleepTime = idleSleepTimeUs(); 1722} 1723 1724void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1725{ 1726 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1727 this, streamType, mTracks.size()); 1728 Mutex::Autolock _l(mLock); 1729 1730 size_t size = mTracks.size(); 1731 for (size_t i = 0; i < size; i++) { 1732 sp<Track> t = mTracks[i]; 1733 if (t->streamType() == streamType) { 1734 t->invalidate(); 1735 } 1736 } 1737} 1738 1739status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1740{ 1741 int session = chain->sessionId(); 1742 int16_t *buffer = mMixBuffer; 1743 bool ownsBuffer = false; 1744 1745 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1746 if (session > 0) { 1747 // Only one effect chain can be present in direct output thread and it uses 1748 // the mix buffer as input 1749 if (mType != DIRECT) { 1750 size_t numSamples = mNormalFrameCount * mChannelCount; 1751 buffer = new int16_t[numSamples]; 1752 memset(buffer, 0, numSamples * sizeof(int16_t)); 1753 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1754 ownsBuffer = true; 1755 } 1756 1757 // Attach all tracks with same session ID to this chain. 1758 for (size_t i = 0; i < mTracks.size(); ++i) { 1759 sp<Track> track = mTracks[i]; 1760 if (session == track->sessionId()) { 1761 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1762 buffer); 1763 track->setMainBuffer(buffer); 1764 chain->incTrackCnt(); 1765 } 1766 } 1767 1768 // indicate all active tracks in the chain 1769 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1770 sp<Track> track = mActiveTracks[i].promote(); 1771 if (track == 0) { 1772 continue; 1773 } 1774 if (session == track->sessionId()) { 1775 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1776 chain->incActiveTrackCnt(); 1777 } 1778 } 1779 } 1780 1781 chain->setInBuffer(buffer, ownsBuffer); 1782 chain->setOutBuffer(mMixBuffer); 1783 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1784 // chains list in order to be processed last as it contains output stage effects 1785 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1786 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1787 // after track specific effects and before output stage 1788 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1789 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1790 // Effect chain for other sessions are inserted at beginning of effect 1791 // chains list to be processed before output mix effects. Relative order between other 1792 // sessions is not important 1793 size_t size = mEffectChains.size(); 1794 size_t i = 0; 1795 for (i = 0; i < size; i++) { 1796 if (mEffectChains[i]->sessionId() < session) { 1797 break; 1798 } 1799 } 1800 mEffectChains.insertAt(chain, i); 1801 checkSuspendOnAddEffectChain_l(chain); 1802 1803 return NO_ERROR; 1804} 1805 1806size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1807{ 1808 int session = chain->sessionId(); 1809 1810 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1811 1812 for (size_t i = 0; i < mEffectChains.size(); i++) { 1813 if (chain == mEffectChains[i]) { 1814 mEffectChains.removeAt(i); 1815 // detach all active tracks from the chain 1816 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1817 sp<Track> track = mActiveTracks[i].promote(); 1818 if (track == 0) { 1819 continue; 1820 } 1821 if (session == track->sessionId()) { 1822 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1823 chain.get(), session); 1824 chain->decActiveTrackCnt(); 1825 } 1826 } 1827 1828 // detach all tracks with same session ID from this chain 1829 for (size_t i = 0; i < mTracks.size(); ++i) { 1830 sp<Track> track = mTracks[i]; 1831 if (session == track->sessionId()) { 1832 track->setMainBuffer(mMixBuffer); 1833 chain->decTrackCnt(); 1834 } 1835 } 1836 break; 1837 } 1838 } 1839 return mEffectChains.size(); 1840} 1841 1842status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1843 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 return attachAuxEffect_l(track, EffectId); 1847} 1848 1849status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1850 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1851{ 1852 status_t status = NO_ERROR; 1853 1854 if (EffectId == 0) { 1855 track->setAuxBuffer(0, NULL); 1856 } else { 1857 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1858 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1859 if (effect != 0) { 1860 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1861 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1862 } else { 1863 status = INVALID_OPERATION; 1864 } 1865 } else { 1866 status = BAD_VALUE; 1867 } 1868 } 1869 return status; 1870} 1871 1872void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1873{ 1874 for (size_t i = 0; i < mTracks.size(); ++i) { 1875 sp<Track> track = mTracks[i]; 1876 if (track->auxEffectId() == effectId) { 1877 attachAuxEffect_l(track, 0); 1878 } 1879 } 1880} 1881 1882bool AudioFlinger::PlaybackThread::threadLoop() 1883{ 1884 Vector< sp<Track> > tracksToRemove; 1885 1886 standbyTime = systemTime(); 1887 1888 // MIXER 1889 nsecs_t lastWarning = 0; 1890 1891 // DUPLICATING 1892 // FIXME could this be made local to while loop? 1893 writeFrames = 0; 1894 1895 cacheParameters_l(); 1896 sleepTime = idleSleepTime; 1897 1898 if (mType == MIXER) { 1899 sleepTimeShift = 0; 1900 } 1901 1902 CpuStats cpuStats; 1903 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1904 1905 acquireWakeLock(); 1906 1907 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1908 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1909 // and then that string will be logged at the next convenient opportunity. 1910 const char *logString = NULL; 1911 1912 while (!exitPending()) 1913 { 1914 cpuStats.sample(myName); 1915 1916 Vector< sp<EffectChain> > effectChains; 1917 1918 processConfigEvents(); 1919 1920 { // scope for mLock 1921 1922 Mutex::Autolock _l(mLock); 1923 1924 if (logString != NULL) { 1925 mNBLogWriter->logTimestamp(); 1926 mNBLogWriter->log(logString); 1927 logString = NULL; 1928 } 1929 1930 if (checkForNewParameters_l()) { 1931 cacheParameters_l(); 1932 } 1933 1934 saveOutputTracks(); 1935 1936 // put audio hardware into standby after short delay 1937 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1938 isSuspended())) { 1939 if (!mStandby) { 1940 1941 threadLoop_standby(); 1942 1943 mNBLogWriter->log("standby"); 1944 mStandby = true; 1945 } 1946 1947 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1948 // we're about to wait, flush the binder command buffer 1949 IPCThreadState::self()->flushCommands(); 1950 1951 clearOutputTracks(); 1952 1953 if (exitPending()) { 1954 break; 1955 } 1956 1957 releaseWakeLock_l(); 1958 // wait until we have something to do... 1959 ALOGV("%s going to sleep", myName.string()); 1960 mWaitWorkCV.wait(mLock); 1961 ALOGV("%s waking up", myName.string()); 1962 acquireWakeLock_l(); 1963 1964 mMixerStatus = MIXER_IDLE; 1965 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1966 mBytesWritten = 0; 1967 1968 checkSilentMode_l(); 1969 1970 standbyTime = systemTime() + standbyDelay; 1971 sleepTime = idleSleepTime; 1972 if (mType == MIXER) { 1973 sleepTimeShift = 0; 1974 } 1975 1976 continue; 1977 } 1978 } 1979 1980 // mMixerStatusIgnoringFastTracks is also updated internally 1981 mMixerStatus = prepareTracks_l(&tracksToRemove); 1982 1983 // prevent any changes in effect chain list and in each effect chain 1984 // during mixing and effect process as the audio buffers could be deleted 1985 // or modified if an effect is created or deleted 1986 lockEffectChains_l(effectChains); 1987 } 1988 1989 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1990 threadLoop_mix(); 1991 } else { 1992 threadLoop_sleepTime(); 1993 } 1994 1995 if (isSuspended()) { 1996 sleepTime = suspendSleepTimeUs(); 1997 mBytesWritten += mixBufferSize; 1998 } 1999 2000 // only process effects if we're going to write 2001 if (sleepTime == 0) { 2002 for (size_t i = 0; i < effectChains.size(); i ++) { 2003 effectChains[i]->process_l(); 2004 } 2005 } 2006 2007 // enable changes in effect chain 2008 unlockEffectChains(effectChains); 2009 2010 // sleepTime == 0 means we must write to audio hardware 2011 if (sleepTime == 0) { 2012 2013 threadLoop_write(); 2014 2015if (mType == MIXER) { 2016 // write blocked detection 2017 nsecs_t now = systemTime(); 2018 nsecs_t delta = now - mLastWriteTime; 2019 if (!mStandby && delta > maxPeriod) { 2020 mNumDelayedWrites++; 2021 if ((now - lastWarning) > kWarningThrottleNs) { 2022 ATRACE_NAME("underrun"); 2023 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2024 ns2ms(delta), mNumDelayedWrites, this); 2025 lastWarning = now; 2026 } 2027 } 2028} 2029 2030 mStandby = false; 2031 } else { 2032 usleep(sleepTime); 2033 } 2034 2035 // Finally let go of removed track(s), without the lock held 2036 // since we can't guarantee the destructors won't acquire that 2037 // same lock. This will also mutate and push a new fast mixer state. 2038 threadLoop_removeTracks(tracksToRemove); 2039 if (tracksToRemove.size() > 0) { 2040 logString = "remove"; 2041 } 2042 tracksToRemove.clear(); 2043 2044 // FIXME I don't understand the need for this here; 2045 // it was in the original code but maybe the 2046 // assignment in saveOutputTracks() makes this unnecessary? 2047 clearOutputTracks(); 2048 2049 // Effect chains will be actually deleted here if they were removed from 2050 // mEffectChains list during mixing or effects processing 2051 effectChains.clear(); 2052 2053 // FIXME Note that the above .clear() is no longer necessary since effectChains 2054 // is now local to this block, but will keep it for now (at least until merge done). 2055 } 2056 2057 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2058 if (mType == MIXER || mType == DIRECT) { 2059 // put output stream into standby mode 2060 if (!mStandby) { 2061 mOutput->stream->common.standby(&mOutput->stream->common); 2062 } 2063 } 2064 2065 releaseWakeLock(); 2066 2067 ALOGV("Thread %p type %d exiting", this, mType); 2068 return false; 2069} 2070 2071 2072// ---------------------------------------------------------------------------- 2073 2074AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2075 audio_io_handle_t id, audio_devices_t device, type_t type) 2076 : PlaybackThread(audioFlinger, output, id, device, type), 2077 // mAudioMixer below 2078 // mFastMixer below 2079 mFastMixerFutex(0) 2080 // mOutputSink below 2081 // mPipeSink below 2082 // mNormalSink below 2083{ 2084 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2085 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2086 "mFrameCount=%d, mNormalFrameCount=%d", 2087 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2088 mNormalFrameCount); 2089 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2090 2091 // FIXME - Current mixer implementation only supports stereo output 2092 if (mChannelCount != FCC_2) { 2093 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2094 } 2095 2096 // create an NBAIO sink for the HAL output stream, and negotiate 2097 mOutputSink = new AudioStreamOutSink(output->stream); 2098 size_t numCounterOffers = 0; 2099 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2100 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2101 ALOG_ASSERT(index == 0); 2102 2103 // initialize fast mixer depending on configuration 2104 bool initFastMixer; 2105 switch (kUseFastMixer) { 2106 case FastMixer_Never: 2107 initFastMixer = false; 2108 break; 2109 case FastMixer_Always: 2110 initFastMixer = true; 2111 break; 2112 case FastMixer_Static: 2113 case FastMixer_Dynamic: 2114 initFastMixer = mFrameCount < mNormalFrameCount; 2115 break; 2116 } 2117 if (initFastMixer) { 2118 2119 // create a MonoPipe to connect our submix to FastMixer 2120 NBAIO_Format format = mOutputSink->format(); 2121 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2122 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2123 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2124 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2125 const NBAIO_Format offers[1] = {format}; 2126 size_t numCounterOffers = 0; 2127 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2128 ALOG_ASSERT(index == 0); 2129 monoPipe->setAvgFrames((mScreenState & 1) ? 2130 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2131 mPipeSink = monoPipe; 2132 2133#ifdef TEE_SINK_FRAMES 2134 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2135 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2136 numCounterOffers = 0; 2137 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2138 ALOG_ASSERT(index == 0); 2139 mTeeSink = teeSink; 2140 PipeReader *teeSource = new PipeReader(*teeSink); 2141 numCounterOffers = 0; 2142 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2143 ALOG_ASSERT(index == 0); 2144 mTeeSource = teeSource; 2145#endif 2146 2147 // create fast mixer and configure it initially with just one fast track for our submix 2148 mFastMixer = new FastMixer(); 2149 FastMixerStateQueue *sq = mFastMixer->sq(); 2150#ifdef STATE_QUEUE_DUMP 2151 sq->setObserverDump(&mStateQueueObserverDump); 2152 sq->setMutatorDump(&mStateQueueMutatorDump); 2153#endif 2154 FastMixerState *state = sq->begin(); 2155 FastTrack *fastTrack = &state->mFastTracks[0]; 2156 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2157 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2158 fastTrack->mVolumeProvider = NULL; 2159 fastTrack->mGeneration++; 2160 state->mFastTracksGen++; 2161 state->mTrackMask = 1; 2162 // fast mixer will use the HAL output sink 2163 state->mOutputSink = mOutputSink.get(); 2164 state->mOutputSinkGen++; 2165 state->mFrameCount = mFrameCount; 2166 state->mCommand = FastMixerState::COLD_IDLE; 2167 // already done in constructor initialization list 2168 //mFastMixerFutex = 0; 2169 state->mColdFutexAddr = &mFastMixerFutex; 2170 state->mColdGen++; 2171 state->mDumpState = &mFastMixerDumpState; 2172 state->mTeeSink = mTeeSink.get(); 2173 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2174 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2175 sq->end(); 2176 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2177 2178 // start the fast mixer 2179 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2180 pid_t tid = mFastMixer->getTid(); 2181 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2182 if (err != 0) { 2183 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2184 kPriorityFastMixer, getpid_cached, tid, err); 2185 } 2186 2187#ifdef AUDIO_WATCHDOG 2188 // create and start the watchdog 2189 mAudioWatchdog = new AudioWatchdog(); 2190 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2191 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2192 tid = mAudioWatchdog->getTid(); 2193 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2194 if (err != 0) { 2195 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2196 kPriorityFastMixer, getpid_cached, tid, err); 2197 } 2198#endif 2199 2200 } else { 2201 mFastMixer = NULL; 2202 } 2203 2204 switch (kUseFastMixer) { 2205 case FastMixer_Never: 2206 case FastMixer_Dynamic: 2207 mNormalSink = mOutputSink; 2208 break; 2209 case FastMixer_Always: 2210 mNormalSink = mPipeSink; 2211 break; 2212 case FastMixer_Static: 2213 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2214 break; 2215 } 2216} 2217 2218AudioFlinger::MixerThread::~MixerThread() 2219{ 2220 if (mFastMixer != NULL) { 2221 FastMixerStateQueue *sq = mFastMixer->sq(); 2222 FastMixerState *state = sq->begin(); 2223 if (state->mCommand == FastMixerState::COLD_IDLE) { 2224 int32_t old = android_atomic_inc(&mFastMixerFutex); 2225 if (old == -1) { 2226 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2227 } 2228 } 2229 state->mCommand = FastMixerState::EXIT; 2230 sq->end(); 2231 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2232 mFastMixer->join(); 2233 // Though the fast mixer thread has exited, it's state queue is still valid. 2234 // We'll use that extract the final state which contains one remaining fast track 2235 // corresponding to our sub-mix. 2236 state = sq->begin(); 2237 ALOG_ASSERT(state->mTrackMask == 1); 2238 FastTrack *fastTrack = &state->mFastTracks[0]; 2239 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2240 delete fastTrack->mBufferProvider; 2241 sq->end(false /*didModify*/); 2242 delete mFastMixer; 2243#ifdef AUDIO_WATCHDOG 2244 if (mAudioWatchdog != 0) { 2245 mAudioWatchdog->requestExit(); 2246 mAudioWatchdog->requestExitAndWait(); 2247 mAudioWatchdog.clear(); 2248 } 2249#endif 2250 } 2251 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2252 delete mAudioMixer; 2253} 2254 2255 2256uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2257{ 2258 if (mFastMixer != NULL) { 2259 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2260 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2261 } 2262 return latency; 2263} 2264 2265 2266void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2267{ 2268 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2269} 2270 2271void AudioFlinger::MixerThread::threadLoop_write() 2272{ 2273 // FIXME we should only do one push per cycle; confirm this is true 2274 // Start the fast mixer if it's not already running 2275 if (mFastMixer != NULL) { 2276 FastMixerStateQueue *sq = mFastMixer->sq(); 2277 FastMixerState *state = sq->begin(); 2278 if (state->mCommand != FastMixerState::MIX_WRITE && 2279 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2280 if (state->mCommand == FastMixerState::COLD_IDLE) { 2281 int32_t old = android_atomic_inc(&mFastMixerFutex); 2282 if (old == -1) { 2283 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2284 } 2285#ifdef AUDIO_WATCHDOG 2286 if (mAudioWatchdog != 0) { 2287 mAudioWatchdog->resume(); 2288 } 2289#endif 2290 } 2291 state->mCommand = FastMixerState::MIX_WRITE; 2292 sq->end(); 2293 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2294 if (kUseFastMixer == FastMixer_Dynamic) { 2295 mNormalSink = mPipeSink; 2296 } 2297 } else { 2298 sq->end(false /*didModify*/); 2299 } 2300 } 2301 PlaybackThread::threadLoop_write(); 2302} 2303 2304void AudioFlinger::MixerThread::threadLoop_standby() 2305{ 2306 // Idle the fast mixer if it's currently running 2307 if (mFastMixer != NULL) { 2308 FastMixerStateQueue *sq = mFastMixer->sq(); 2309 FastMixerState *state = sq->begin(); 2310 if (!(state->mCommand & FastMixerState::IDLE)) { 2311 state->mCommand = FastMixerState::COLD_IDLE; 2312 state->mColdFutexAddr = &mFastMixerFutex; 2313 state->mColdGen++; 2314 mFastMixerFutex = 0; 2315 sq->end(); 2316 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2317 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2318 if (kUseFastMixer == FastMixer_Dynamic) { 2319 mNormalSink = mOutputSink; 2320 } 2321#ifdef AUDIO_WATCHDOG 2322 if (mAudioWatchdog != 0) { 2323 mAudioWatchdog->pause(); 2324 } 2325#endif 2326 } else { 2327 sq->end(false /*didModify*/); 2328 } 2329 } 2330 PlaybackThread::threadLoop_standby(); 2331} 2332 2333// shared by MIXER and DIRECT, overridden by DUPLICATING 2334void AudioFlinger::PlaybackThread::threadLoop_standby() 2335{ 2336 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2337 mOutput->stream->common.standby(&mOutput->stream->common); 2338} 2339 2340void AudioFlinger::MixerThread::threadLoop_mix() 2341{ 2342 // obtain the presentation timestamp of the next output buffer 2343 int64_t pts; 2344 status_t status = INVALID_OPERATION; 2345 2346 if (mNormalSink != 0) { 2347 status = mNormalSink->getNextWriteTimestamp(&pts); 2348 } else { 2349 status = mOutputSink->getNextWriteTimestamp(&pts); 2350 } 2351 2352 if (status != NO_ERROR) { 2353 pts = AudioBufferProvider::kInvalidPTS; 2354 } 2355 2356 // mix buffers... 2357 mAudioMixer->process(pts); 2358 // increase sleep time progressively when application underrun condition clears. 2359 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2360 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2361 // such that we would underrun the audio HAL. 2362 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2363 sleepTimeShift--; 2364 } 2365 sleepTime = 0; 2366 standbyTime = systemTime() + standbyDelay; 2367 //TODO: delay standby when effects have a tail 2368} 2369 2370void AudioFlinger::MixerThread::threadLoop_sleepTime() 2371{ 2372 // If no tracks are ready, sleep once for the duration of an output 2373 // buffer size, then write 0s to the output 2374 if (sleepTime == 0) { 2375 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2376 sleepTime = activeSleepTime >> sleepTimeShift; 2377 if (sleepTime < kMinThreadSleepTimeUs) { 2378 sleepTime = kMinThreadSleepTimeUs; 2379 } 2380 // reduce sleep time in case of consecutive application underruns to avoid 2381 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2382 // duration we would end up writing less data than needed by the audio HAL if 2383 // the condition persists. 2384 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2385 sleepTimeShift++; 2386 } 2387 } else { 2388 sleepTime = idleSleepTime; 2389 } 2390 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2391 memset (mMixBuffer, 0, mixBufferSize); 2392 sleepTime = 0; 2393 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2394 "anticipated start"); 2395 } 2396 // TODO add standby time extension fct of effect tail 2397} 2398 2399// prepareTracks_l() must be called with ThreadBase::mLock held 2400AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2401 Vector< sp<Track> > *tracksToRemove) 2402{ 2403 2404 mixer_state mixerStatus = MIXER_IDLE; 2405 // find out which tracks need to be processed 2406 size_t count = mActiveTracks.size(); 2407 size_t mixedTracks = 0; 2408 size_t tracksWithEffect = 0; 2409 // counts only _active_ fast tracks 2410 size_t fastTracks = 0; 2411 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2412 2413 float masterVolume = mMasterVolume; 2414 bool masterMute = mMasterMute; 2415 2416 if (masterMute) { 2417 masterVolume = 0; 2418 } 2419 // Delegate master volume control to effect in output mix effect chain if needed 2420 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2421 if (chain != 0) { 2422 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2423 chain->setVolume_l(&v, &v); 2424 masterVolume = (float)((v + (1 << 23)) >> 24); 2425 chain.clear(); 2426 } 2427 2428 // prepare a new state to push 2429 FastMixerStateQueue *sq = NULL; 2430 FastMixerState *state = NULL; 2431 bool didModify = false; 2432 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2433 if (mFastMixer != NULL) { 2434 sq = mFastMixer->sq(); 2435 state = sq->begin(); 2436 } 2437 2438 for (size_t i=0 ; i<count ; i++) { 2439 sp<Track> t = mActiveTracks[i].promote(); 2440 if (t == 0) { 2441 continue; 2442 } 2443 2444 // this const just means the local variable doesn't change 2445 Track* const track = t.get(); 2446 2447 // process fast tracks 2448 if (track->isFastTrack()) { 2449 2450 // It's theoretically possible (though unlikely) for a fast track to be created 2451 // and then removed within the same normal mix cycle. This is not a problem, as 2452 // the track never becomes active so it's fast mixer slot is never touched. 2453 // The converse, of removing an (active) track and then creating a new track 2454 // at the identical fast mixer slot within the same normal mix cycle, 2455 // is impossible because the slot isn't marked available until the end of each cycle. 2456 int j = track->mFastIndex; 2457 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2458 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2459 FastTrack *fastTrack = &state->mFastTracks[j]; 2460 2461 // Determine whether the track is currently in underrun condition, 2462 // and whether it had a recent underrun. 2463 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2464 FastTrackUnderruns underruns = ftDump->mUnderruns; 2465 uint32_t recentFull = (underruns.mBitFields.mFull - 2466 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2467 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2468 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2469 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2470 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2471 uint32_t recentUnderruns = recentPartial + recentEmpty; 2472 track->mObservedUnderruns = underruns; 2473 // don't count underruns that occur while stopping or pausing 2474 // or stopped which can occur when flush() is called while active 2475 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2476 track->mUnderrunCount += recentUnderruns; 2477 } 2478 2479 // This is similar to the state machine for normal tracks, 2480 // with a few modifications for fast tracks. 2481 bool isActive = true; 2482 switch (track->mState) { 2483 case TrackBase::STOPPING_1: 2484 // track stays active in STOPPING_1 state until first underrun 2485 if (recentUnderruns > 0) { 2486 track->mState = TrackBase::STOPPING_2; 2487 } 2488 break; 2489 case TrackBase::PAUSING: 2490 // ramp down is not yet implemented 2491 track->setPaused(); 2492 break; 2493 case TrackBase::RESUMING: 2494 // ramp up is not yet implemented 2495 track->mState = TrackBase::ACTIVE; 2496 break; 2497 case TrackBase::ACTIVE: 2498 if (recentFull > 0 || recentPartial > 0) { 2499 // track has provided at least some frames recently: reset retry count 2500 track->mRetryCount = kMaxTrackRetries; 2501 } 2502 if (recentUnderruns == 0) { 2503 // no recent underruns: stay active 2504 break; 2505 } 2506 // there has recently been an underrun of some kind 2507 if (track->sharedBuffer() == 0) { 2508 // were any of the recent underruns "empty" (no frames available)? 2509 if (recentEmpty == 0) { 2510 // no, then ignore the partial underruns as they are allowed indefinitely 2511 break; 2512 } 2513 // there has recently been an "empty" underrun: decrement the retry counter 2514 if (--(track->mRetryCount) > 0) { 2515 break; 2516 } 2517 // indicate to client process that the track was disabled because of underrun; 2518 // it will then automatically call start() when data is available 2519 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2520 // remove from active list, but state remains ACTIVE [confusing but true] 2521 isActive = false; 2522 break; 2523 } 2524 // fall through 2525 case TrackBase::STOPPING_2: 2526 case TrackBase::PAUSED: 2527 case TrackBase::TERMINATED: 2528 case TrackBase::STOPPED: 2529 case TrackBase::FLUSHED: // flush() while active 2530 // Check for presentation complete if track is inactive 2531 // We have consumed all the buffers of this track. 2532 // This would be incomplete if we auto-paused on underrun 2533 { 2534 size_t audioHALFrames = 2535 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2536 size_t framesWritten = mBytesWritten / mFrameSize; 2537 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2538 // track stays in active list until presentation is complete 2539 break; 2540 } 2541 } 2542 if (track->isStopping_2()) { 2543 track->mState = TrackBase::STOPPED; 2544 } 2545 if (track->isStopped()) { 2546 // Can't reset directly, as fast mixer is still polling this track 2547 // track->reset(); 2548 // So instead mark this track as needing to be reset after push with ack 2549 resetMask |= 1 << i; 2550 } 2551 isActive = false; 2552 break; 2553 case TrackBase::IDLE: 2554 default: 2555 LOG_FATAL("unexpected track state %d", track->mState); 2556 } 2557 2558 if (isActive) { 2559 // was it previously inactive? 2560 if (!(state->mTrackMask & (1 << j))) { 2561 ExtendedAudioBufferProvider *eabp = track; 2562 VolumeProvider *vp = track; 2563 fastTrack->mBufferProvider = eabp; 2564 fastTrack->mVolumeProvider = vp; 2565 fastTrack->mSampleRate = track->mSampleRate; 2566 fastTrack->mChannelMask = track->mChannelMask; 2567 fastTrack->mGeneration++; 2568 state->mTrackMask |= 1 << j; 2569 didModify = true; 2570 // no acknowledgement required for newly active tracks 2571 } 2572 // cache the combined master volume and stream type volume for fast mixer; this 2573 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2574 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2575 ++fastTracks; 2576 } else { 2577 // was it previously active? 2578 if (state->mTrackMask & (1 << j)) { 2579 fastTrack->mBufferProvider = NULL; 2580 fastTrack->mGeneration++; 2581 state->mTrackMask &= ~(1 << j); 2582 didModify = true; 2583 // If any fast tracks were removed, we must wait for acknowledgement 2584 // because we're about to decrement the last sp<> on those tracks. 2585 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2586 } else { 2587 LOG_FATAL("fast track %d should have been active", j); 2588 } 2589 tracksToRemove->add(track); 2590 // Avoids a misleading display in dumpsys 2591 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2592 } 2593 continue; 2594 } 2595 2596 { // local variable scope to avoid goto warning 2597 2598 audio_track_cblk_t* cblk = track->cblk(); 2599 2600 // The first time a track is added we wait 2601 // for all its buffers to be filled before processing it 2602 int name = track->name(); 2603 // make sure that we have enough frames to mix one full buffer. 2604 // enforce this condition only once to enable draining the buffer in case the client 2605 // app does not call stop() and relies on underrun to stop: 2606 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2607 // during last round 2608 uint32_t minFrames = 1; 2609 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2610 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2611 if (t->sampleRate() == mSampleRate) { 2612 minFrames = mNormalFrameCount; 2613 } else { 2614 // +1 for rounding and +1 for additional sample needed for interpolation 2615 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2616 // add frames already consumed but not yet released by the resampler 2617 // because cblk->framesReady() will include these frames 2618 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2619 // the minimum track buffer size is normally twice the number of frames necessary 2620 // to fill one buffer and the resampler should not leave more than one buffer worth 2621 // of unreleased frames after each pass, but just in case... 2622 ALOG_ASSERT(minFrames <= cblk->frameCount_); 2623 } 2624 } 2625 if ((track->framesReady() >= minFrames) && track->isReady() && 2626 !track->isPaused() && !track->isTerminated()) 2627 { 2628 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2629 this); 2630 2631 mixedTracks++; 2632 2633 // track->mainBuffer() != mMixBuffer means there is an effect chain 2634 // connected to the track 2635 chain.clear(); 2636 if (track->mainBuffer() != mMixBuffer) { 2637 chain = getEffectChain_l(track->sessionId()); 2638 // Delegate volume control to effect in track effect chain if needed 2639 if (chain != 0) { 2640 tracksWithEffect++; 2641 } else { 2642 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2643 "session %d", 2644 name, track->sessionId()); 2645 } 2646 } 2647 2648 2649 int param = AudioMixer::VOLUME; 2650 if (track->mFillingUpStatus == Track::FS_FILLED) { 2651 // no ramp for the first volume setting 2652 track->mFillingUpStatus = Track::FS_ACTIVE; 2653 if (track->mState == TrackBase::RESUMING) { 2654 track->mState = TrackBase::ACTIVE; 2655 param = AudioMixer::RAMP_VOLUME; 2656 } 2657 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2658 } else if (cblk->server != 0) { 2659 // If the track is stopped before the first frame was mixed, 2660 // do not apply ramp 2661 param = AudioMixer::RAMP_VOLUME; 2662 } 2663 2664 // compute volume for this track 2665 uint32_t vl, vr, va; 2666 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2667 vl = vr = va = 0; 2668 if (track->isPausing()) { 2669 track->setPaused(); 2670 } 2671 } else { 2672 2673 // read original volumes with volume control 2674 float typeVolume = mStreamTypes[track->streamType()].volume; 2675 float v = masterVolume * typeVolume; 2676 ServerProxy *proxy = track->mServerProxy; 2677 uint32_t vlr = proxy->getVolumeLR(); 2678 vl = vlr & 0xFFFF; 2679 vr = vlr >> 16; 2680 // track volumes come from shared memory, so can't be trusted and must be clamped 2681 if (vl > MAX_GAIN_INT) { 2682 ALOGV("Track left volume out of range: %04X", vl); 2683 vl = MAX_GAIN_INT; 2684 } 2685 if (vr > MAX_GAIN_INT) { 2686 ALOGV("Track right volume out of range: %04X", vr); 2687 vr = MAX_GAIN_INT; 2688 } 2689 // now apply the master volume and stream type volume 2690 vl = (uint32_t)(v * vl) << 12; 2691 vr = (uint32_t)(v * vr) << 12; 2692 // assuming master volume and stream type volume each go up to 1.0, 2693 // vl and vr are now in 8.24 format 2694 2695 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2696 // send level comes from shared memory and so may be corrupt 2697 if (sendLevel > MAX_GAIN_INT) { 2698 ALOGV("Track send level out of range: %04X", sendLevel); 2699 sendLevel = MAX_GAIN_INT; 2700 } 2701 va = (uint32_t)(v * sendLevel); 2702 } 2703 // Delegate volume control to effect in track effect chain if needed 2704 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2705 // Do not ramp volume if volume is controlled by effect 2706 param = AudioMixer::VOLUME; 2707 track->mHasVolumeController = true; 2708 } else { 2709 // force no volume ramp when volume controller was just disabled or removed 2710 // from effect chain to avoid volume spike 2711 if (track->mHasVolumeController) { 2712 param = AudioMixer::VOLUME; 2713 } 2714 track->mHasVolumeController = false; 2715 } 2716 2717 // Convert volumes from 8.24 to 4.12 format 2718 // This additional clamping is needed in case chain->setVolume_l() overshot 2719 vl = (vl + (1 << 11)) >> 12; 2720 if (vl > MAX_GAIN_INT) { 2721 vl = MAX_GAIN_INT; 2722 } 2723 vr = (vr + (1 << 11)) >> 12; 2724 if (vr > MAX_GAIN_INT) { 2725 vr = MAX_GAIN_INT; 2726 } 2727 2728 if (va > MAX_GAIN_INT) { 2729 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2730 } 2731 2732 // XXX: these things DON'T need to be done each time 2733 mAudioMixer->setBufferProvider(name, track); 2734 mAudioMixer->enable(name); 2735 2736 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2737 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2738 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2739 mAudioMixer->setParameter( 2740 name, 2741 AudioMixer::TRACK, 2742 AudioMixer::FORMAT, (void *)track->format()); 2743 mAudioMixer->setParameter( 2744 name, 2745 AudioMixer::TRACK, 2746 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2747 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2748 uint32_t maxSampleRate = mSampleRate * 2; 2749 uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); 2750 if (reqSampleRate == 0) { 2751 reqSampleRate = mSampleRate; 2752 } else if (reqSampleRate > maxSampleRate) { 2753 reqSampleRate = maxSampleRate; 2754 } 2755 mAudioMixer->setParameter( 2756 name, 2757 AudioMixer::RESAMPLE, 2758 AudioMixer::SAMPLE_RATE, 2759 (void *)reqSampleRate); 2760 mAudioMixer->setParameter( 2761 name, 2762 AudioMixer::TRACK, 2763 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2764 mAudioMixer->setParameter( 2765 name, 2766 AudioMixer::TRACK, 2767 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2768 2769 // reset retry count 2770 track->mRetryCount = kMaxTrackRetries; 2771 2772 // If one track is ready, set the mixer ready if: 2773 // - the mixer was not ready during previous round OR 2774 // - no other track is not ready 2775 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2776 mixerStatus != MIXER_TRACKS_ENABLED) { 2777 mixerStatus = MIXER_TRACKS_READY; 2778 } 2779 } else { 2780 // clear effect chain input buffer if an active track underruns to avoid sending 2781 // previous audio buffer again to effects 2782 chain = getEffectChain_l(track->sessionId()); 2783 if (chain != 0) { 2784 chain->clearInputBuffer(); 2785 } 2786 2787 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2788 cblk->server, this); 2789 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2790 track->isStopped() || track->isPaused()) { 2791 // We have consumed all the buffers of this track. 2792 // Remove it from the list of active tracks. 2793 // TODO: use actual buffer filling status instead of latency when available from 2794 // audio HAL 2795 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2796 size_t framesWritten = mBytesWritten / mFrameSize; 2797 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2798 if (track->isStopped()) { 2799 track->reset(); 2800 } 2801 tracksToRemove->add(track); 2802 } 2803 } else { 2804 track->mUnderrunCount++; 2805 // No buffers for this track. Give it a few chances to 2806 // fill a buffer, then remove it from active list. 2807 if (--(track->mRetryCount) <= 0) { 2808 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2809 tracksToRemove->add(track); 2810 // indicate to client process that the track was disabled because of underrun; 2811 // it will then automatically call start() when data is available 2812 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2813 // If one track is not ready, mark the mixer also not ready if: 2814 // - the mixer was ready during previous round OR 2815 // - no other track is ready 2816 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2817 mixerStatus != MIXER_TRACKS_READY) { 2818 mixerStatus = MIXER_TRACKS_ENABLED; 2819 } 2820 } 2821 mAudioMixer->disable(name); 2822 } 2823 2824 } // local variable scope to avoid goto warning 2825track_is_ready: ; 2826 2827 } 2828 2829 // Push the new FastMixer state if necessary 2830 bool pauseAudioWatchdog = false; 2831 if (didModify) { 2832 state->mFastTracksGen++; 2833 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2834 if (kUseFastMixer == FastMixer_Dynamic && 2835 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2836 state->mCommand = FastMixerState::COLD_IDLE; 2837 state->mColdFutexAddr = &mFastMixerFutex; 2838 state->mColdGen++; 2839 mFastMixerFutex = 0; 2840 if (kUseFastMixer == FastMixer_Dynamic) { 2841 mNormalSink = mOutputSink; 2842 } 2843 // If we go into cold idle, need to wait for acknowledgement 2844 // so that fast mixer stops doing I/O. 2845 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2846 pauseAudioWatchdog = true; 2847 } 2848 sq->end(); 2849 } 2850 if (sq != NULL) { 2851 sq->end(didModify); 2852 sq->push(block); 2853 } 2854#ifdef AUDIO_WATCHDOG 2855 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2856 mAudioWatchdog->pause(); 2857 } 2858#endif 2859 2860 // Now perform the deferred reset on fast tracks that have stopped 2861 while (resetMask != 0) { 2862 size_t i = __builtin_ctz(resetMask); 2863 ALOG_ASSERT(i < count); 2864 resetMask &= ~(1 << i); 2865 sp<Track> t = mActiveTracks[i].promote(); 2866 if (t == 0) { 2867 continue; 2868 } 2869 Track* track = t.get(); 2870 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2871 track->reset(); 2872 } 2873 2874 // remove all the tracks that need to be... 2875 count = tracksToRemove->size(); 2876 if (CC_UNLIKELY(count)) { 2877 for (size_t i=0 ; i<count ; i++) { 2878 const sp<Track>& track = tracksToRemove->itemAt(i); 2879 mNBLogWriter->logTimestamp(); 2880 mNBLogWriter->logf("prepareTracks_l remove name=%u mFastIndex=%d", track->name(), 2881 track->mFastIndex); 2882 mActiveTracks.remove(track); 2883 if (track->mainBuffer() != mMixBuffer) { 2884 chain = getEffectChain_l(track->sessionId()); 2885 if (chain != 0) { 2886 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2887 track->sessionId()); 2888 chain->decActiveTrackCnt(); 2889 } 2890 } 2891 if (track->isTerminated()) { 2892 removeTrack_l(track); 2893 } 2894 } 2895 } 2896 2897 // mix buffer must be cleared if all tracks are connected to an 2898 // effect chain as in this case the mixer will not write to 2899 // mix buffer and track effects will accumulate into it 2900 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2901 (mixedTracks == 0 && fastTracks > 0)) { 2902 // FIXME as a performance optimization, should remember previous zero status 2903 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2904 } 2905 2906 // if any fast tracks, then status is ready 2907 mMixerStatusIgnoringFastTracks = mixerStatus; 2908 if (fastTracks > 0) { 2909 mixerStatus = MIXER_TRACKS_READY; 2910 } 2911 return mixerStatus; 2912} 2913 2914// getTrackName_l() must be called with ThreadBase::mLock held 2915int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2916{ 2917 return mAudioMixer->getTrackName(channelMask, sessionId); 2918} 2919 2920// deleteTrackName_l() must be called with ThreadBase::mLock held 2921void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2922{ 2923 ALOGV("remove track (%d) and delete from mixer", name); 2924 mAudioMixer->deleteTrackName(name); 2925} 2926 2927// checkForNewParameters_l() must be called with ThreadBase::mLock held 2928bool AudioFlinger::MixerThread::checkForNewParameters_l() 2929{ 2930 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2931 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2932 bool reconfig = false; 2933 2934 while (!mNewParameters.isEmpty()) { 2935 2936 if (mFastMixer != NULL) { 2937 FastMixerStateQueue *sq = mFastMixer->sq(); 2938 FastMixerState *state = sq->begin(); 2939 if (!(state->mCommand & FastMixerState::IDLE)) { 2940 previousCommand = state->mCommand; 2941 state->mCommand = FastMixerState::HOT_IDLE; 2942 sq->end(); 2943 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2944 } else { 2945 sq->end(false /*didModify*/); 2946 } 2947 } 2948 2949 status_t status = NO_ERROR; 2950 String8 keyValuePair = mNewParameters[0]; 2951 AudioParameter param = AudioParameter(keyValuePair); 2952 int value; 2953 2954 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2955 reconfig = true; 2956 } 2957 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2958 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2959 status = BAD_VALUE; 2960 } else { 2961 reconfig = true; 2962 } 2963 } 2964 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2965 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2966 status = BAD_VALUE; 2967 } else { 2968 reconfig = true; 2969 } 2970 } 2971 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2972 // do not accept frame count changes if tracks are open as the track buffer 2973 // size depends on frame count and correct behavior would not be guaranteed 2974 // if frame count is changed after track creation 2975 if (!mTracks.isEmpty()) { 2976 status = INVALID_OPERATION; 2977 } else { 2978 reconfig = true; 2979 } 2980 } 2981 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2982#ifdef ADD_BATTERY_DATA 2983 // when changing the audio output device, call addBatteryData to notify 2984 // the change 2985 if (mOutDevice != value) { 2986 uint32_t params = 0; 2987 // check whether speaker is on 2988 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2989 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2990 } 2991 2992 audio_devices_t deviceWithoutSpeaker 2993 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2994 // check if any other device (except speaker) is on 2995 if (value & deviceWithoutSpeaker ) { 2996 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2997 } 2998 2999 if (params != 0) { 3000 addBatteryData(params); 3001 } 3002 } 3003#endif 3004 3005 // forward device change to effects that have requested to be 3006 // aware of attached audio device. 3007 mOutDevice = value; 3008 for (size_t i = 0; i < mEffectChains.size(); i++) { 3009 mEffectChains[i]->setDevice_l(mOutDevice); 3010 } 3011 } 3012 3013 if (status == NO_ERROR) { 3014 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3015 keyValuePair.string()); 3016 if (!mStandby && status == INVALID_OPERATION) { 3017 mOutput->stream->common.standby(&mOutput->stream->common); 3018 mStandby = true; 3019 mBytesWritten = 0; 3020 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3021 keyValuePair.string()); 3022 } 3023 if (status == NO_ERROR && reconfig) { 3024 delete mAudioMixer; 3025 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3026 mAudioMixer = NULL; 3027 readOutputParameters(); 3028 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3029 for (size_t i = 0; i < mTracks.size() ; i++) { 3030 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3031 if (name < 0) { 3032 break; 3033 } 3034 mTracks[i]->mName = name; 3035 } 3036 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3037 } 3038 } 3039 3040 mNewParameters.removeAt(0); 3041 3042 mParamStatus = status; 3043 mParamCond.signal(); 3044 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3045 // already timed out waiting for the status and will never signal the condition. 3046 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3047 } 3048 3049 if (!(previousCommand & FastMixerState::IDLE)) { 3050 ALOG_ASSERT(mFastMixer != NULL); 3051 FastMixerStateQueue *sq = mFastMixer->sq(); 3052 FastMixerState *state = sq->begin(); 3053 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3054 state->mCommand = previousCommand; 3055 sq->end(); 3056 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3057 } 3058 3059 return reconfig; 3060} 3061 3062 3063void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3064{ 3065 const size_t SIZE = 256; 3066 char buffer[SIZE]; 3067 String8 result; 3068 3069 PlaybackThread::dumpInternals(fd, args); 3070 3071 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3072 result.append(buffer); 3073 write(fd, result.string(), result.size()); 3074 3075 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3076 FastMixerDumpState copy = mFastMixerDumpState; 3077 copy.dump(fd); 3078 3079#ifdef STATE_QUEUE_DUMP 3080 // Similar for state queue 3081 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3082 observerCopy.dump(fd); 3083 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3084 mutatorCopy.dump(fd); 3085#endif 3086 3087 // Write the tee output to a .wav file 3088 dumpTee(fd, mTeeSource, mId); 3089 3090#ifdef AUDIO_WATCHDOG 3091 if (mAudioWatchdog != 0) { 3092 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3093 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3094 wdCopy.dump(fd); 3095 } 3096#endif 3097} 3098 3099uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3100{ 3101 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3102} 3103 3104uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3105{ 3106 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3107} 3108 3109void AudioFlinger::MixerThread::cacheParameters_l() 3110{ 3111 PlaybackThread::cacheParameters_l(); 3112 3113 // FIXME: Relaxed timing because of a certain device that can't meet latency 3114 // Should be reduced to 2x after the vendor fixes the driver issue 3115 // increase threshold again due to low power audio mode. The way this warning 3116 // threshold is calculated and its usefulness should be reconsidered anyway. 3117 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3118} 3119 3120// ---------------------------------------------------------------------------- 3121 3122AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3123 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3124 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3125 // mLeftVolFloat, mRightVolFloat 3126{ 3127} 3128 3129AudioFlinger::DirectOutputThread::~DirectOutputThread() 3130{ 3131} 3132 3133AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3134 Vector< sp<Track> > *tracksToRemove 3135) 3136{ 3137 sp<Track> trackToRemove; 3138 3139 mixer_state mixerStatus = MIXER_IDLE; 3140 3141 // find out which tracks need to be processed 3142 if (mActiveTracks.size() != 0) { 3143 sp<Track> t = mActiveTracks[0].promote(); 3144 // The track died recently 3145 if (t == 0) { 3146 return MIXER_IDLE; 3147 } 3148 3149 Track* const track = t.get(); 3150 audio_track_cblk_t* cblk = track->cblk(); 3151 3152 // The first time a track is added we wait 3153 // for all its buffers to be filled before processing it 3154 uint32_t minFrames; 3155 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3156 minFrames = mNormalFrameCount; 3157 } else { 3158 minFrames = 1; 3159 } 3160 if ((track->framesReady() >= minFrames) && track->isReady() && 3161 !track->isPaused() && !track->isTerminated()) 3162 { 3163 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3164 3165 if (track->mFillingUpStatus == Track::FS_FILLED) { 3166 track->mFillingUpStatus = Track::FS_ACTIVE; 3167 mLeftVolFloat = mRightVolFloat = 0; 3168 if (track->mState == TrackBase::RESUMING) { 3169 track->mState = TrackBase::ACTIVE; 3170 } 3171 } 3172 3173 // compute volume for this track 3174 float left, right; 3175 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3176 left = right = 0; 3177 if (track->isPausing()) { 3178 track->setPaused(); 3179 } 3180 } else { 3181 float typeVolume = mStreamTypes[track->streamType()].volume; 3182 float v = mMasterVolume * typeVolume; 3183 uint32_t vlr = track->mServerProxy->getVolumeLR(); 3184 float v_clamped = v * (vlr & 0xFFFF); 3185 if (v_clamped > MAX_GAIN) { 3186 v_clamped = MAX_GAIN; 3187 } 3188 left = v_clamped/MAX_GAIN; 3189 v_clamped = v * (vlr >> 16); 3190 if (v_clamped > MAX_GAIN) { 3191 v_clamped = MAX_GAIN; 3192 } 3193 right = v_clamped/MAX_GAIN; 3194 } 3195 3196 if (left != mLeftVolFloat || right != mRightVolFloat) { 3197 mLeftVolFloat = left; 3198 mRightVolFloat = right; 3199 3200 // Convert volumes from float to 8.24 3201 uint32_t vl = (uint32_t)(left * (1 << 24)); 3202 uint32_t vr = (uint32_t)(right * (1 << 24)); 3203 3204 // Delegate volume control to effect in track effect chain if needed 3205 // only one effect chain can be present on DirectOutputThread, so if 3206 // there is one, the track is connected to it 3207 if (!mEffectChains.isEmpty()) { 3208 // Do not ramp volume if volume is controlled by effect 3209 mEffectChains[0]->setVolume_l(&vl, &vr); 3210 left = (float)vl / (1 << 24); 3211 right = (float)vr / (1 << 24); 3212 } 3213 mOutput->stream->set_volume(mOutput->stream, left, right); 3214 } 3215 3216 // reset retry count 3217 track->mRetryCount = kMaxTrackRetriesDirect; 3218 mActiveTrack = t; 3219 mixerStatus = MIXER_TRACKS_READY; 3220 } else { 3221 // clear effect chain input buffer if an active track underruns to avoid sending 3222 // previous audio buffer again to effects 3223 if (!mEffectChains.isEmpty()) { 3224 mEffectChains[0]->clearInputBuffer(); 3225 } 3226 3227 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3228 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3229 track->isStopped() || track->isPaused()) { 3230 // We have consumed all the buffers of this track. 3231 // Remove it from the list of active tracks. 3232 // TODO: implement behavior for compressed audio 3233 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3234 size_t framesWritten = mBytesWritten / mFrameSize; 3235 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3236 if (track->isStopped()) { 3237 track->reset(); 3238 } 3239 trackToRemove = track; 3240 } 3241 } else { 3242 // No buffers for this track. Give it a few chances to 3243 // fill a buffer, then remove it from active list. 3244 if (--(track->mRetryCount) <= 0) { 3245 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3246 trackToRemove = track; 3247 } else { 3248 mixerStatus = MIXER_TRACKS_ENABLED; 3249 } 3250 } 3251 } 3252 } 3253 3254 // FIXME merge this with similar code for removing multiple tracks 3255 // remove all the tracks that need to be... 3256 if (CC_UNLIKELY(trackToRemove != 0)) { 3257 tracksToRemove->add(trackToRemove); 3258#if 0 3259 mNBLogWriter->logf("prepareTracks_l remove name=%u", trackToRemove->name()); 3260#endif 3261 mActiveTracks.remove(trackToRemove); 3262 if (!mEffectChains.isEmpty()) { 3263 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3264 trackToRemove->sessionId()); 3265 mEffectChains[0]->decActiveTrackCnt(); 3266 } 3267 if (trackToRemove->isTerminated()) { 3268 removeTrack_l(trackToRemove); 3269 } 3270 } 3271 3272 return mixerStatus; 3273} 3274 3275void AudioFlinger::DirectOutputThread::threadLoop_mix() 3276{ 3277 AudioBufferProvider::Buffer buffer; 3278 size_t frameCount = mFrameCount; 3279 int8_t *curBuf = (int8_t *)mMixBuffer; 3280 // output audio to hardware 3281 while (frameCount) { 3282 buffer.frameCount = frameCount; 3283 mActiveTrack->getNextBuffer(&buffer); 3284 if (CC_UNLIKELY(buffer.raw == NULL)) { 3285 memset(curBuf, 0, frameCount * mFrameSize); 3286 break; 3287 } 3288 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3289 frameCount -= buffer.frameCount; 3290 curBuf += buffer.frameCount * mFrameSize; 3291 mActiveTrack->releaseBuffer(&buffer); 3292 } 3293 sleepTime = 0; 3294 standbyTime = systemTime() + standbyDelay; 3295 mActiveTrack.clear(); 3296 3297} 3298 3299void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3300{ 3301 if (sleepTime == 0) { 3302 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3303 sleepTime = activeSleepTime; 3304 } else { 3305 sleepTime = idleSleepTime; 3306 } 3307 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3308 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3309 sleepTime = 0; 3310 } 3311} 3312 3313// getTrackName_l() must be called with ThreadBase::mLock held 3314int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3315 int sessionId) 3316{ 3317 return 0; 3318} 3319 3320// deleteTrackName_l() must be called with ThreadBase::mLock held 3321void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3322{ 3323} 3324 3325// checkForNewParameters_l() must be called with ThreadBase::mLock held 3326bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3327{ 3328 bool reconfig = false; 3329 3330 while (!mNewParameters.isEmpty()) { 3331 status_t status = NO_ERROR; 3332 String8 keyValuePair = mNewParameters[0]; 3333 AudioParameter param = AudioParameter(keyValuePair); 3334 int value; 3335 3336 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3337 // do not accept frame count changes if tracks are open as the track buffer 3338 // size depends on frame count and correct behavior would not be garantied 3339 // if frame count is changed after track creation 3340 if (!mTracks.isEmpty()) { 3341 status = INVALID_OPERATION; 3342 } else { 3343 reconfig = true; 3344 } 3345 } 3346 if (status == NO_ERROR) { 3347 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3348 keyValuePair.string()); 3349 if (!mStandby && status == INVALID_OPERATION) { 3350 mOutput->stream->common.standby(&mOutput->stream->common); 3351 mStandby = true; 3352 mBytesWritten = 0; 3353 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3354 keyValuePair.string()); 3355 } 3356 if (status == NO_ERROR && reconfig) { 3357 readOutputParameters(); 3358 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3359 } 3360 } 3361 3362 mNewParameters.removeAt(0); 3363 3364 mParamStatus = status; 3365 mParamCond.signal(); 3366 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3367 // already timed out waiting for the status and will never signal the condition. 3368 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3369 } 3370 return reconfig; 3371} 3372 3373uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3374{ 3375 uint32_t time; 3376 if (audio_is_linear_pcm(mFormat)) { 3377 time = PlaybackThread::activeSleepTimeUs(); 3378 } else { 3379 time = 10000; 3380 } 3381 return time; 3382} 3383 3384uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3385{ 3386 uint32_t time; 3387 if (audio_is_linear_pcm(mFormat)) { 3388 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3389 } else { 3390 time = 10000; 3391 } 3392 return time; 3393} 3394 3395uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3396{ 3397 uint32_t time; 3398 if (audio_is_linear_pcm(mFormat)) { 3399 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3400 } else { 3401 time = 10000; 3402 } 3403 return time; 3404} 3405 3406void AudioFlinger::DirectOutputThread::cacheParameters_l() 3407{ 3408 PlaybackThread::cacheParameters_l(); 3409 3410 // use shorter standby delay as on normal output to release 3411 // hardware resources as soon as possible 3412 standbyDelay = microseconds(activeSleepTime*2); 3413} 3414 3415// ---------------------------------------------------------------------------- 3416 3417AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3418 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3419 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3420 DUPLICATING), 3421 mWaitTimeMs(UINT_MAX) 3422{ 3423 addOutputTrack(mainThread); 3424} 3425 3426AudioFlinger::DuplicatingThread::~DuplicatingThread() 3427{ 3428 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3429 mOutputTracks[i]->destroy(); 3430 } 3431} 3432 3433void AudioFlinger::DuplicatingThread::threadLoop_mix() 3434{ 3435 // mix buffers... 3436 if (outputsReady(outputTracks)) { 3437 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3438 } else { 3439 memset(mMixBuffer, 0, mixBufferSize); 3440 } 3441 sleepTime = 0; 3442 writeFrames = mNormalFrameCount; 3443 standbyTime = systemTime() + standbyDelay; 3444} 3445 3446void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3447{ 3448 if (sleepTime == 0) { 3449 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3450 sleepTime = activeSleepTime; 3451 } else { 3452 sleepTime = idleSleepTime; 3453 } 3454 } else if (mBytesWritten != 0) { 3455 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3456 writeFrames = mNormalFrameCount; 3457 memset(mMixBuffer, 0, mixBufferSize); 3458 } else { 3459 // flush remaining overflow buffers in output tracks 3460 writeFrames = 0; 3461 } 3462 sleepTime = 0; 3463 } 3464} 3465 3466void AudioFlinger::DuplicatingThread::threadLoop_write() 3467{ 3468 for (size_t i = 0; i < outputTracks.size(); i++) { 3469 outputTracks[i]->write(mMixBuffer, writeFrames); 3470 } 3471 mBytesWritten += mixBufferSize; 3472} 3473 3474void AudioFlinger::DuplicatingThread::threadLoop_standby() 3475{ 3476 // DuplicatingThread implements standby by stopping all tracks 3477 for (size_t i = 0; i < outputTracks.size(); i++) { 3478 outputTracks[i]->stop(); 3479 } 3480} 3481 3482void AudioFlinger::DuplicatingThread::saveOutputTracks() 3483{ 3484 outputTracks = mOutputTracks; 3485} 3486 3487void AudioFlinger::DuplicatingThread::clearOutputTracks() 3488{ 3489 outputTracks.clear(); 3490} 3491 3492void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3493{ 3494 Mutex::Autolock _l(mLock); 3495 // FIXME explain this formula 3496 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3497 OutputTrack *outputTrack = new OutputTrack(thread, 3498 this, 3499 mSampleRate, 3500 mFormat, 3501 mChannelMask, 3502 frameCount); 3503 if (outputTrack->cblk() != NULL) { 3504 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3505 mOutputTracks.add(outputTrack); 3506 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3507 updateWaitTime_l(); 3508 } 3509} 3510 3511void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3512{ 3513 Mutex::Autolock _l(mLock); 3514 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3515 if (mOutputTracks[i]->thread() == thread) { 3516 mOutputTracks[i]->destroy(); 3517 mOutputTracks.removeAt(i); 3518 updateWaitTime_l(); 3519 return; 3520 } 3521 } 3522 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3523} 3524 3525// caller must hold mLock 3526void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3527{ 3528 mWaitTimeMs = UINT_MAX; 3529 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3530 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3531 if (strong != 0) { 3532 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3533 if (waitTimeMs < mWaitTimeMs) { 3534 mWaitTimeMs = waitTimeMs; 3535 } 3536 } 3537 } 3538} 3539 3540 3541bool AudioFlinger::DuplicatingThread::outputsReady( 3542 const SortedVector< sp<OutputTrack> > &outputTracks) 3543{ 3544 for (size_t i = 0; i < outputTracks.size(); i++) { 3545 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3546 if (thread == 0) { 3547 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3548 outputTracks[i].get()); 3549 return false; 3550 } 3551 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3552 // see note at standby() declaration 3553 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3554 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3555 thread.get()); 3556 return false; 3557 } 3558 } 3559 return true; 3560} 3561 3562uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3563{ 3564 return (mWaitTimeMs * 1000) / 2; 3565} 3566 3567void AudioFlinger::DuplicatingThread::cacheParameters_l() 3568{ 3569 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3570 updateWaitTime_l(); 3571 3572 MixerThread::cacheParameters_l(); 3573} 3574 3575// ---------------------------------------------------------------------------- 3576// Record 3577// ---------------------------------------------------------------------------- 3578 3579AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3580 AudioStreamIn *input, 3581 uint32_t sampleRate, 3582 audio_channel_mask_t channelMask, 3583 audio_io_handle_t id, 3584 audio_devices_t outDevice, 3585 audio_devices_t inDevice, 3586 const sp<NBAIO_Sink>& teeSink) : 3587 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3588 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3589 // mRsmpInIndex and mInputBytes set by readInputParameters() 3590 mReqChannelCount(popcount(channelMask)), 3591 mReqSampleRate(sampleRate), 3592 // mBytesRead is only meaningful while active, and so is cleared in start() 3593 // (but might be better to also clear here for dump?) 3594 mTeeSink(teeSink) 3595{ 3596 snprintf(mName, kNameLength, "AudioIn_%X", id); 3597 3598 readInputParameters(); 3599 3600} 3601 3602 3603AudioFlinger::RecordThread::~RecordThread() 3604{ 3605 delete[] mRsmpInBuffer; 3606 delete mResampler; 3607 delete[] mRsmpOutBuffer; 3608} 3609 3610void AudioFlinger::RecordThread::onFirstRef() 3611{ 3612 run(mName, PRIORITY_URGENT_AUDIO); 3613} 3614 3615status_t AudioFlinger::RecordThread::readyToRun() 3616{ 3617 status_t status = initCheck(); 3618 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3619 return status; 3620} 3621 3622bool AudioFlinger::RecordThread::threadLoop() 3623{ 3624 AudioBufferProvider::Buffer buffer; 3625 sp<RecordTrack> activeTrack; 3626 Vector< sp<EffectChain> > effectChains; 3627 3628 nsecs_t lastWarning = 0; 3629 3630 inputStandBy(); 3631 acquireWakeLock(); 3632 3633 // used to verify we've read at least once before evaluating how many bytes were read 3634 bool readOnce = false; 3635 3636 // start recording 3637 while (!exitPending()) { 3638 3639 processConfigEvents(); 3640 3641 { // scope for mLock 3642 Mutex::Autolock _l(mLock); 3643 checkForNewParameters_l(); 3644 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3645 standby(); 3646 3647 if (exitPending()) { 3648 break; 3649 } 3650 3651 releaseWakeLock_l(); 3652 ALOGV("RecordThread: loop stopping"); 3653 // go to sleep 3654 mWaitWorkCV.wait(mLock); 3655 ALOGV("RecordThread: loop starting"); 3656 acquireWakeLock_l(); 3657 continue; 3658 } 3659 if (mActiveTrack != 0) { 3660 if (mActiveTrack->mState == TrackBase::PAUSING) { 3661 standby(); 3662 mActiveTrack.clear(); 3663 mStartStopCond.broadcast(); 3664 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3665 if (mReqChannelCount != mActiveTrack->channelCount()) { 3666 mActiveTrack.clear(); 3667 mStartStopCond.broadcast(); 3668 } else if (readOnce) { 3669 // record start succeeds only if first read from audio input 3670 // succeeds 3671 if (mBytesRead >= 0) { 3672 mActiveTrack->mState = TrackBase::ACTIVE; 3673 } else { 3674 mActiveTrack.clear(); 3675 } 3676 mStartStopCond.broadcast(); 3677 } 3678 mStandby = false; 3679 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3680 removeTrack_l(mActiveTrack); 3681 mActiveTrack.clear(); 3682 } 3683 } 3684 lockEffectChains_l(effectChains); 3685 } 3686 3687 if (mActiveTrack != 0) { 3688 if (mActiveTrack->mState != TrackBase::ACTIVE && 3689 mActiveTrack->mState != TrackBase::RESUMING) { 3690 unlockEffectChains(effectChains); 3691 usleep(kRecordThreadSleepUs); 3692 continue; 3693 } 3694 for (size_t i = 0; i < effectChains.size(); i ++) { 3695 effectChains[i]->process_l(); 3696 } 3697 3698 buffer.frameCount = mFrameCount; 3699 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3700 readOnce = true; 3701 size_t framesOut = buffer.frameCount; 3702 if (mResampler == NULL) { 3703 // no resampling 3704 while (framesOut) { 3705 size_t framesIn = mFrameCount - mRsmpInIndex; 3706 if (framesIn) { 3707 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3708 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3709 mActiveTrack->mFrameSize; 3710 if (framesIn > framesOut) 3711 framesIn = framesOut; 3712 mRsmpInIndex += framesIn; 3713 framesOut -= framesIn; 3714 if (mChannelCount == mReqChannelCount || 3715 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3716 memcpy(dst, src, framesIn * mFrameSize); 3717 } else { 3718 if (mChannelCount == 1) { 3719 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3720 (int16_t *)src, framesIn); 3721 } else { 3722 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3723 (int16_t *)src, framesIn); 3724 } 3725 } 3726 } 3727 if (framesOut && mFrameCount == mRsmpInIndex) { 3728 void *readInto; 3729 if (framesOut == mFrameCount && 3730 (mChannelCount == mReqChannelCount || 3731 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3732 readInto = buffer.raw; 3733 framesOut = 0; 3734 } else { 3735 readInto = mRsmpInBuffer; 3736 mRsmpInIndex = 0; 3737 } 3738 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3739 if (mBytesRead <= 0) { 3740 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3741 { 3742 ALOGE("Error reading audio input"); 3743 // Force input into standby so that it tries to 3744 // recover at next read attempt 3745 inputStandBy(); 3746 usleep(kRecordThreadSleepUs); 3747 } 3748 mRsmpInIndex = mFrameCount; 3749 framesOut = 0; 3750 buffer.frameCount = 0; 3751 } else if (mTeeSink != 0) { 3752 (void) mTeeSink->write(readInto, 3753 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3754 } 3755 } 3756 } 3757 } else { 3758 // resampling 3759 3760 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3761 // alter output frame count as if we were expecting stereo samples 3762 if (mChannelCount == 1 && mReqChannelCount == 1) { 3763 framesOut >>= 1; 3764 } 3765 mResampler->resample(mRsmpOutBuffer, framesOut, 3766 this /* AudioBufferProvider* */); 3767 // ditherAndClamp() works as long as all buffers returned by 3768 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3769 if (mChannelCount == 2 && mReqChannelCount == 1) { 3770 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3771 // the resampler always outputs stereo samples: 3772 // do post stereo to mono conversion 3773 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3774 framesOut); 3775 } else { 3776 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3777 } 3778 3779 } 3780 if (mFramestoDrop == 0) { 3781 mActiveTrack->releaseBuffer(&buffer); 3782 } else { 3783 if (mFramestoDrop > 0) { 3784 mFramestoDrop -= buffer.frameCount; 3785 if (mFramestoDrop <= 0) { 3786 clearSyncStartEvent(); 3787 } 3788 } else { 3789 mFramestoDrop += buffer.frameCount; 3790 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3791 mSyncStartEvent->isCancelled()) { 3792 ALOGW("Synced record %s, session %d, trigger session %d", 3793 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3794 mActiveTrack->sessionId(), 3795 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3796 clearSyncStartEvent(); 3797 } 3798 } 3799 } 3800 mActiveTrack->clearOverflow(); 3801 } 3802 // client isn't retrieving buffers fast enough 3803 else { 3804 if (!mActiveTrack->setOverflow()) { 3805 nsecs_t now = systemTime(); 3806 if ((now - lastWarning) > kWarningThrottleNs) { 3807 ALOGW("RecordThread: buffer overflow"); 3808 lastWarning = now; 3809 } 3810 } 3811 // Release the processor for a while before asking for a new buffer. 3812 // This will give the application more chance to read from the buffer and 3813 // clear the overflow. 3814 usleep(kRecordThreadSleepUs); 3815 } 3816 } 3817 // enable changes in effect chain 3818 unlockEffectChains(effectChains); 3819 effectChains.clear(); 3820 } 3821 3822 standby(); 3823 3824 { 3825 Mutex::Autolock _l(mLock); 3826 mActiveTrack.clear(); 3827 mStartStopCond.broadcast(); 3828 } 3829 3830 releaseWakeLock(); 3831 3832 ALOGV("RecordThread %p exiting", this); 3833 return false; 3834} 3835 3836void AudioFlinger::RecordThread::standby() 3837{ 3838 if (!mStandby) { 3839 inputStandBy(); 3840 mStandby = true; 3841 } 3842} 3843 3844void AudioFlinger::RecordThread::inputStandBy() 3845{ 3846 mInput->stream->common.standby(&mInput->stream->common); 3847} 3848 3849sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3850 const sp<AudioFlinger::Client>& client, 3851 uint32_t sampleRate, 3852 audio_format_t format, 3853 audio_channel_mask_t channelMask, 3854 size_t frameCount, 3855 int sessionId, 3856 IAudioFlinger::track_flags_t flags, 3857 pid_t tid, 3858 status_t *status) 3859{ 3860 sp<RecordTrack> track; 3861 status_t lStatus; 3862 3863 lStatus = initCheck(); 3864 if (lStatus != NO_ERROR) { 3865 ALOGE("Audio driver not initialized."); 3866 goto Exit; 3867 } 3868 3869 // FIXME use flags and tid similar to createTrack_l() 3870 3871 { // scope for mLock 3872 Mutex::Autolock _l(mLock); 3873 3874 track = new RecordTrack(this, client, sampleRate, 3875 format, channelMask, frameCount, sessionId); 3876 3877 if (track->getCblk() == 0) { 3878 lStatus = NO_MEMORY; 3879 goto Exit; 3880 } 3881 mTracks.add(track); 3882 3883 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3884 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3885 mAudioFlinger->btNrecIsOff(); 3886 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3887 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3888 } 3889 lStatus = NO_ERROR; 3890 3891Exit: 3892 if (status) { 3893 *status = lStatus; 3894 } 3895 return track; 3896} 3897 3898status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3899 AudioSystem::sync_event_t event, 3900 int triggerSession) 3901{ 3902 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3903 sp<ThreadBase> strongMe = this; 3904 status_t status = NO_ERROR; 3905 3906 if (event == AudioSystem::SYNC_EVENT_NONE) { 3907 clearSyncStartEvent(); 3908 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3909 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3910 triggerSession, 3911 recordTrack->sessionId(), 3912 syncStartEventCallback, 3913 this); 3914 // Sync event can be cancelled by the trigger session if the track is not in a 3915 // compatible state in which case we start record immediately 3916 if (mSyncStartEvent->isCancelled()) { 3917 clearSyncStartEvent(); 3918 } else { 3919 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3920 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3921 } 3922 } 3923 3924 { 3925 AutoMutex lock(mLock); 3926 if (mActiveTrack != 0) { 3927 if (recordTrack != mActiveTrack.get()) { 3928 status = -EBUSY; 3929 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3930 mActiveTrack->mState = TrackBase::ACTIVE; 3931 } 3932 return status; 3933 } 3934 3935 recordTrack->mState = TrackBase::IDLE; 3936 mActiveTrack = recordTrack; 3937 mLock.unlock(); 3938 status_t status = AudioSystem::startInput(mId); 3939 mLock.lock(); 3940 if (status != NO_ERROR) { 3941 mActiveTrack.clear(); 3942 clearSyncStartEvent(); 3943 return status; 3944 } 3945 mRsmpInIndex = mFrameCount; 3946 mBytesRead = 0; 3947 if (mResampler != NULL) { 3948 mResampler->reset(); 3949 } 3950 mActiveTrack->mState = TrackBase::RESUMING; 3951 // signal thread to start 3952 ALOGV("Signal record thread"); 3953 mWaitWorkCV.broadcast(); 3954 // do not wait for mStartStopCond if exiting 3955 if (exitPending()) { 3956 mActiveTrack.clear(); 3957 status = INVALID_OPERATION; 3958 goto startError; 3959 } 3960 mStartStopCond.wait(mLock); 3961 if (mActiveTrack == 0) { 3962 ALOGV("Record failed to start"); 3963 status = BAD_VALUE; 3964 goto startError; 3965 } 3966 ALOGV("Record started OK"); 3967 return status; 3968 } 3969startError: 3970 AudioSystem::stopInput(mId); 3971 clearSyncStartEvent(); 3972 return status; 3973} 3974 3975void AudioFlinger::RecordThread::clearSyncStartEvent() 3976{ 3977 if (mSyncStartEvent != 0) { 3978 mSyncStartEvent->cancel(); 3979 } 3980 mSyncStartEvent.clear(); 3981 mFramestoDrop = 0; 3982} 3983 3984void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3985{ 3986 sp<SyncEvent> strongEvent = event.promote(); 3987 3988 if (strongEvent != 0) { 3989 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3990 me->handleSyncStartEvent(strongEvent); 3991 } 3992} 3993 3994void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 3995{ 3996 if (event == mSyncStartEvent) { 3997 // TODO: use actual buffer filling status instead of 2 buffers when info is available 3998 // from audio HAL 3999 mFramestoDrop = mFrameCount * 2; 4000 } 4001} 4002 4003bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4004 ALOGV("RecordThread::stop"); 4005 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4006 return false; 4007 } 4008 recordTrack->mState = TrackBase::PAUSING; 4009 // do not wait for mStartStopCond if exiting 4010 if (exitPending()) { 4011 return true; 4012 } 4013 mStartStopCond.wait(mLock); 4014 // if we have been restarted, recordTrack == mActiveTrack.get() here 4015 if (exitPending() || recordTrack != mActiveTrack.get()) { 4016 ALOGV("Record stopped OK"); 4017 return true; 4018 } 4019 return false; 4020} 4021 4022bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4023{ 4024 return false; 4025} 4026 4027status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4028{ 4029#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4030 if (!isValidSyncEvent(event)) { 4031 return BAD_VALUE; 4032 } 4033 4034 int eventSession = event->triggerSession(); 4035 status_t ret = NAME_NOT_FOUND; 4036 4037 Mutex::Autolock _l(mLock); 4038 4039 for (size_t i = 0; i < mTracks.size(); i++) { 4040 sp<RecordTrack> track = mTracks[i]; 4041 if (eventSession == track->sessionId()) { 4042 (void) track->setSyncEvent(event); 4043 ret = NO_ERROR; 4044 } 4045 } 4046 return ret; 4047#else 4048 return BAD_VALUE; 4049#endif 4050} 4051 4052// destroyTrack_l() must be called with ThreadBase::mLock held 4053void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4054{ 4055 track->mState = TrackBase::TERMINATED; 4056 // active tracks are removed by threadLoop() 4057 if (mActiveTrack != track) { 4058 removeTrack_l(track); 4059 } 4060} 4061 4062void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4063{ 4064 mTracks.remove(track); 4065 // need anything related to effects here? 4066} 4067 4068void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4069{ 4070 dumpInternals(fd, args); 4071 dumpTracks(fd, args); 4072 dumpEffectChains(fd, args); 4073} 4074 4075void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4076{ 4077 const size_t SIZE = 256; 4078 char buffer[SIZE]; 4079 String8 result; 4080 4081 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4082 result.append(buffer); 4083 4084 if (mActiveTrack != 0) { 4085 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4086 result.append(buffer); 4087 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4088 result.append(buffer); 4089 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4090 result.append(buffer); 4091 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4092 result.append(buffer); 4093 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4094 result.append(buffer); 4095 } else { 4096 result.append("No active record client\n"); 4097 } 4098 4099 write(fd, result.string(), result.size()); 4100 4101 dumpBase(fd, args); 4102} 4103 4104void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4105{ 4106 const size_t SIZE = 256; 4107 char buffer[SIZE]; 4108 String8 result; 4109 4110 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4111 result.append(buffer); 4112 RecordTrack::appendDumpHeader(result); 4113 for (size_t i = 0; i < mTracks.size(); ++i) { 4114 sp<RecordTrack> track = mTracks[i]; 4115 if (track != 0) { 4116 track->dump(buffer, SIZE); 4117 result.append(buffer); 4118 } 4119 } 4120 4121 if (mActiveTrack != 0) { 4122 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4123 result.append(buffer); 4124 RecordTrack::appendDumpHeader(result); 4125 mActiveTrack->dump(buffer, SIZE); 4126 result.append(buffer); 4127 4128 } 4129 write(fd, result.string(), result.size()); 4130} 4131 4132// AudioBufferProvider interface 4133status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4134{ 4135 size_t framesReq = buffer->frameCount; 4136 size_t framesReady = mFrameCount - mRsmpInIndex; 4137 int channelCount; 4138 4139 if (framesReady == 0) { 4140 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4141 if (mBytesRead <= 0) { 4142 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4143 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4144 // Force input into standby so that it tries to 4145 // recover at next read attempt 4146 inputStandBy(); 4147 usleep(kRecordThreadSleepUs); 4148 } 4149 buffer->raw = NULL; 4150 buffer->frameCount = 0; 4151 return NOT_ENOUGH_DATA; 4152 } 4153 mRsmpInIndex = 0; 4154 framesReady = mFrameCount; 4155 } 4156 4157 if (framesReq > framesReady) { 4158 framesReq = framesReady; 4159 } 4160 4161 if (mChannelCount == 1 && mReqChannelCount == 2) { 4162 channelCount = 1; 4163 } else { 4164 channelCount = 2; 4165 } 4166 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4167 buffer->frameCount = framesReq; 4168 return NO_ERROR; 4169} 4170 4171// AudioBufferProvider interface 4172void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4173{ 4174 mRsmpInIndex += buffer->frameCount; 4175 buffer->frameCount = 0; 4176} 4177 4178bool AudioFlinger::RecordThread::checkForNewParameters_l() 4179{ 4180 bool reconfig = false; 4181 4182 while (!mNewParameters.isEmpty()) { 4183 status_t status = NO_ERROR; 4184 String8 keyValuePair = mNewParameters[0]; 4185 AudioParameter param = AudioParameter(keyValuePair); 4186 int value; 4187 audio_format_t reqFormat = mFormat; 4188 uint32_t reqSamplingRate = mReqSampleRate; 4189 uint32_t reqChannelCount = mReqChannelCount; 4190 4191 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4192 reqSamplingRate = value; 4193 reconfig = true; 4194 } 4195 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4196 reqFormat = (audio_format_t) value; 4197 reconfig = true; 4198 } 4199 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4200 reqChannelCount = popcount(value); 4201 reconfig = true; 4202 } 4203 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4204 // do not accept frame count changes if tracks are open as the track buffer 4205 // size depends on frame count and correct behavior would not be guaranteed 4206 // if frame count is changed after track creation 4207 if (mActiveTrack != 0) { 4208 status = INVALID_OPERATION; 4209 } else { 4210 reconfig = true; 4211 } 4212 } 4213 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4214 // forward device change to effects that have requested to be 4215 // aware of attached audio device. 4216 for (size_t i = 0; i < mEffectChains.size(); i++) { 4217 mEffectChains[i]->setDevice_l(value); 4218 } 4219 4220 // store input device and output device but do not forward output device to audio HAL. 4221 // Note that status is ignored by the caller for output device 4222 // (see AudioFlinger::setParameters() 4223 if (audio_is_output_devices(value)) { 4224 mOutDevice = value; 4225 status = BAD_VALUE; 4226 } else { 4227 mInDevice = value; 4228 // disable AEC and NS if the device is a BT SCO headset supporting those 4229 // pre processings 4230 if (mTracks.size() > 0) { 4231 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4232 mAudioFlinger->btNrecIsOff(); 4233 for (size_t i = 0; i < mTracks.size(); i++) { 4234 sp<RecordTrack> track = mTracks[i]; 4235 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4236 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4237 } 4238 } 4239 } 4240 } 4241 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4242 mAudioSource != (audio_source_t)value) { 4243 // forward device change to effects that have requested to be 4244 // aware of attached audio device. 4245 for (size_t i = 0; i < mEffectChains.size(); i++) { 4246 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4247 } 4248 mAudioSource = (audio_source_t)value; 4249 } 4250 if (status == NO_ERROR) { 4251 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4252 keyValuePair.string()); 4253 if (status == INVALID_OPERATION) { 4254 inputStandBy(); 4255 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4256 keyValuePair.string()); 4257 } 4258 if (reconfig) { 4259 if (status == BAD_VALUE && 4260 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4261 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4262 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4263 <= (2 * reqSamplingRate)) && 4264 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4265 <= FCC_2 && 4266 (reqChannelCount <= FCC_2)) { 4267 status = NO_ERROR; 4268 } 4269 if (status == NO_ERROR) { 4270 readInputParameters(); 4271 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4272 } 4273 } 4274 } 4275 4276 mNewParameters.removeAt(0); 4277 4278 mParamStatus = status; 4279 mParamCond.signal(); 4280 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4281 // already timed out waiting for the status and will never signal the condition. 4282 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4283 } 4284 return reconfig; 4285} 4286 4287String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4288{ 4289 char *s; 4290 String8 out_s8 = String8(); 4291 4292 Mutex::Autolock _l(mLock); 4293 if (initCheck() != NO_ERROR) { 4294 return out_s8; 4295 } 4296 4297 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4298 out_s8 = String8(s); 4299 free(s); 4300 return out_s8; 4301} 4302 4303void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4304 AudioSystem::OutputDescriptor desc; 4305 void *param2 = NULL; 4306 4307 switch (event) { 4308 case AudioSystem::INPUT_OPENED: 4309 case AudioSystem::INPUT_CONFIG_CHANGED: 4310 desc.channels = mChannelMask; 4311 desc.samplingRate = mSampleRate; 4312 desc.format = mFormat; 4313 desc.frameCount = mFrameCount; 4314 desc.latency = 0; 4315 param2 = &desc; 4316 break; 4317 4318 case AudioSystem::INPUT_CLOSED: 4319 default: 4320 break; 4321 } 4322 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4323} 4324 4325void AudioFlinger::RecordThread::readInputParameters() 4326{ 4327 delete mRsmpInBuffer; 4328 // mRsmpInBuffer is always assigned a new[] below 4329 delete mRsmpOutBuffer; 4330 mRsmpOutBuffer = NULL; 4331 delete mResampler; 4332 mResampler = NULL; 4333 4334 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4335 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4336 mChannelCount = (uint16_t)popcount(mChannelMask); 4337 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4338 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4339 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4340 mFrameCount = mInputBytes / mFrameSize; 4341 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4342 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4343 4344 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4345 { 4346 int channelCount; 4347 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4348 // stereo to mono post process as the resampler always outputs stereo. 4349 if (mChannelCount == 1 && mReqChannelCount == 2) { 4350 channelCount = 1; 4351 } else { 4352 channelCount = 2; 4353 } 4354 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4355 mResampler->setSampleRate(mSampleRate); 4356 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4357 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4358 4359 // optmization: if mono to mono, alter input frame count as if we were inputing 4360 // stereo samples 4361 if (mChannelCount == 1 && mReqChannelCount == 1) { 4362 mFrameCount >>= 1; 4363 } 4364 4365 } 4366 mRsmpInIndex = mFrameCount; 4367} 4368 4369unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4370{ 4371 Mutex::Autolock _l(mLock); 4372 if (initCheck() != NO_ERROR) { 4373 return 0; 4374 } 4375 4376 return mInput->stream->get_input_frames_lost(mInput->stream); 4377} 4378 4379uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4380{ 4381 Mutex::Autolock _l(mLock); 4382 uint32_t result = 0; 4383 if (getEffectChain_l(sessionId) != 0) { 4384 result = EFFECT_SESSION; 4385 } 4386 4387 for (size_t i = 0; i < mTracks.size(); ++i) { 4388 if (sessionId == mTracks[i]->sessionId()) { 4389 result |= TRACK_SESSION; 4390 break; 4391 } 4392 } 4393 4394 return result; 4395} 4396 4397KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4398{ 4399 KeyedVector<int, bool> ids; 4400 Mutex::Autolock _l(mLock); 4401 for (size_t j = 0; j < mTracks.size(); ++j) { 4402 sp<RecordThread::RecordTrack> track = mTracks[j]; 4403 int sessionId = track->sessionId(); 4404 if (ids.indexOfKey(sessionId) < 0) { 4405 ids.add(sessionId, true); 4406 } 4407 } 4408 return ids; 4409} 4410 4411AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4412{ 4413 Mutex::Autolock _l(mLock); 4414 AudioStreamIn *input = mInput; 4415 mInput = NULL; 4416 return input; 4417} 4418 4419// this method must always be called either with ThreadBase mLock held or inside the thread loop 4420audio_stream_t* AudioFlinger::RecordThread::stream() const 4421{ 4422 if (mInput == NULL) { 4423 return NULL; 4424 } 4425 return &mInput->stream->common; 4426} 4427 4428status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4429{ 4430 // only one chain per input thread 4431 if (mEffectChains.size() != 0) { 4432 return INVALID_OPERATION; 4433 } 4434 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4435 4436 chain->setInBuffer(NULL); 4437 chain->setOutBuffer(NULL); 4438 4439 checkSuspendOnAddEffectChain_l(chain); 4440 4441 mEffectChains.add(chain); 4442 4443 return NO_ERROR; 4444} 4445 4446size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4447{ 4448 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4449 ALOGW_IF(mEffectChains.size() != 1, 4450 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4451 chain.get(), mEffectChains.size(), this); 4452 if (mEffectChains.size() == 1) { 4453 mEffectChains.removeAt(0); 4454 } 4455 return 0; 4456} 4457 4458}; // namespace android 4459