Threads.cpp revision f6ed423af92a56ef54bba23eba883b1f21448b54
1bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant/* 2bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** 3f5256e16dfc425c1d466f6308d4026d529ce9e0bHoward Hinnant** Copyright 2012, The Android Open Source Project 4bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** 5b64f8b07c104c6cc986570ac8ee0ed16a9f23976Howard Hinnant** Licensed under the Apache License, Version 2.0 (the "License"); 6b64f8b07c104c6cc986570ac8ee0ed16a9f23976Howard Hinnant** you may not use this file except in compliance with the License. 7bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** You may obtain a copy of the License at 8bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** 9bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** http://www.apache.org/licenses/LICENSE-2.0 10bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** 11bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** Unless required by applicable law or agreed to in writing, software 12eb564e76cc3904d811c981a50ecce0659f444cc9Howard Hinnant** distributed under the License is distributed on an "AS IS" BASIS, 1398e5d974006989c505d7b2ec7b9e4b20b0f01e26Howard Hinnant** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** See the License for the specific language governing permissions and 15bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant** limitations under the License. 16bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant*/ 17bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant 18bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant 19bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant#define LOG_TAG "AudioFlinger" 20bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant//#define LOG_NDEBUG 0 21e3e3291f3ab4af96b0403cf6e255c833143ae3f1Howard Hinnant#define ATRACE_TAG ATRACE_TAG_AUDIO 2298e5d974006989c505d7b2ec7b9e4b20b0f01e26Howard Hinnant 2398e5d974006989c505d7b2ec7b9e4b20b0f01e26Howard Hinnant#include "Configuration.h" 2498e5d974006989c505d7b2ec7b9e4b20b0f01e26Howard Hinnant#include <math.h> 2598e5d974006989c505d7b2ec7b9e4b20b0f01e26Howard Hinnant#include <fcntl.h> 2698e5d974006989c505d7b2ec7b9e4b20b0f01e26Howard Hinnant#include <sys/stat.h> 2798e5d974006989c505d7b2ec7b9e4b20b0f01e26Howard Hinnant#include <cutils/properties.h> 289d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <cutils/compiler.h> 299d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <media/AudioParameter.h> 309d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <utils/Log.h> 319d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <utils/Trace.h> 329d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow 339d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <private/media/AudioTrackShared.h> 349d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <hardware/audio.h> 359d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <audio_effects/effect_ns.h> 369d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <audio_effects/effect_aec.h> 379d9463a3555aa559884809b8a7fc842a3968193eMarshall Clow#include <audio_utils/primitives.h> 38e3e3291f3ab4af96b0403cf6e255c833143ae3f1Howard Hinnant 39bc8d3f97eb5c958007f2713238472e0c1c8fe02Howard Hinnant// NBAIO implementations 40#include <media/nbaio/AudioStreamOutSink.h> 41#include <media/nbaio/MonoPipe.h> 42#include <media/nbaio/MonoPipeReader.h> 43#include <media/nbaio/Pipe.h> 44#include <media/nbaio/PipeReader.h> 45#include <media/nbaio/SourceAudioBufferProvider.h> 46 47#include <powermanager/PowerManager.h> 48 49#include <common_time/cc_helper.h> 50#include <common_time/local_clock.h> 51 52#include "AudioFlinger.h" 53#include "AudioMixer.h" 54#include "FastMixer.h" 55#include "ServiceUtilities.h" 56#include "SchedulingPolicyService.h" 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63#ifdef DEBUG_CPU_USAGE 64#include <cpustats/CentralTendencyStatistics.h> 65#include <cpustats/ThreadCpuUsage.h> 66#endif 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85// retry counts for buffer fill timeout 86// 50 * ~20msecs = 1 second 87static const int8_t kMaxTrackRetries = 50; 88static const int8_t kMaxTrackStartupRetries = 50; 89// allow less retry attempts on direct output thread. 90// direct outputs can be a scarce resource in audio hardware and should 91// be released as quickly as possible. 92static const int8_t kMaxTrackRetriesDirect = 2; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait for setParameters to complete 101static const nsecs_t kSetParametersTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108// minimum normal mix buffer size, expressed in milliseconds rather than frames 109static const uint32_t kMinNormalMixBufferSizeMs = 20; 110// maximum normal mix buffer size 111static const uint32_t kMaxNormalMixBufferSizeMs = 24; 112 113// Whether to use fast mixer 114static const enum { 115 FastMixer_Never, // never initialize or use: for debugging only 116 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 117 // normal mixer multiplier is 1 118 FastMixer_Static, // initialize if needed, then use all the time if initialized, 119 // multiplier is calculated based on min & max normal mixer buffer size 120 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 // FIXME for FastMixer_Dynamic: 123 // Supporting this option will require fixing HALs that can't handle large writes. 124 // For example, one HAL implementation returns an error from a large write, 125 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 126 // We could either fix the HAL implementations, or provide a wrapper that breaks 127 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 128} kUseFastMixer = FastMixer_Static; 129 130// Priorities for requestPriority 131static const int kPriorityAudioApp = 2; 132static const int kPriorityFastMixer = 3; 133 134// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 135// for the track. The client then sub-divides this into smaller buffers for its use. 136// Currently the client uses double-buffering by default, but doesn't tell us about that. 137// So for now we just assume that client is double-buffered. 138// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 139// N-buffering, so AudioFlinger could allocate the right amount of memory. 140// See the client's minBufCount and mNotificationFramesAct calculations for details. 141static const int kFastTrackMultiplier = 1; 142 143// ---------------------------------------------------------------------------- 144 145#ifdef ADD_BATTERY_DATA 146// To collect the amplifier usage 147static void addBatteryData(uint32_t params) { 148 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 149 if (service == NULL) { 150 // it already logged 151 return; 152 } 153 154 service->addBatteryData(params); 155} 156#endif 157 158 159// ---------------------------------------------------------------------------- 160// CPU Stats 161// ---------------------------------------------------------------------------- 162 163class CpuStats { 164public: 165 CpuStats(); 166 void sample(const String8 &title); 167#ifdef DEBUG_CPU_USAGE 168private: 169 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 170 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 171 172 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 173 174 int mCpuNum; // thread's current CPU number 175 int mCpukHz; // frequency of thread's current CPU in kHz 176#endif 177}; 178 179CpuStats::CpuStats() 180#ifdef DEBUG_CPU_USAGE 181 : mCpuNum(-1), mCpukHz(-1) 182#endif 183{ 184} 185 186void CpuStats::sample(const String8 &title) { 187#ifdef DEBUG_CPU_USAGE 188 // get current thread's delta CPU time in wall clock ns 189 double wcNs; 190 bool valid = mCpuUsage.sampleAndEnable(wcNs); 191 192 // record sample for wall clock statistics 193 if (valid) { 194 mWcStats.sample(wcNs); 195 } 196 197 // get the current CPU number 198 int cpuNum = sched_getcpu(); 199 200 // get the current CPU frequency in kHz 201 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 202 203 // check if either CPU number or frequency changed 204 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 205 mCpuNum = cpuNum; 206 mCpukHz = cpukHz; 207 // ignore sample for purposes of cycles 208 valid = false; 209 } 210 211 // if no change in CPU number or frequency, then record sample for cycle statistics 212 if (valid && mCpukHz > 0) { 213 double cycles = wcNs * cpukHz * 0.000001; 214 mHzStats.sample(cycles); 215 } 216 217 unsigned n = mWcStats.n(); 218 // mCpuUsage.elapsed() is expensive, so don't call it every loop 219 if ((n & 127) == 1) { 220 long long elapsed = mCpuUsage.elapsed(); 221 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 222 double perLoop = elapsed / (double) n; 223 double perLoop100 = perLoop * 0.01; 224 double perLoop1k = perLoop * 0.001; 225 double mean = mWcStats.mean(); 226 double stddev = mWcStats.stddev(); 227 double minimum = mWcStats.minimum(); 228 double maximum = mWcStats.maximum(); 229 double meanCycles = mHzStats.mean(); 230 double stddevCycles = mHzStats.stddev(); 231 double minCycles = mHzStats.minimum(); 232 double maxCycles = mHzStats.maximum(); 233 mCpuUsage.resetElapsed(); 234 mWcStats.reset(); 235 mHzStats.reset(); 236 ALOGD("CPU usage for %s over past %.1f secs\n" 237 " (%u mixer loops at %.1f mean ms per loop):\n" 238 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 239 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 240 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 241 title.string(), 242 elapsed * .000000001, n, perLoop * .000001, 243 mean * .001, 244 stddev * .001, 245 minimum * .001, 246 maximum * .001, 247 mean / perLoop100, 248 stddev / perLoop100, 249 minimum / perLoop100, 250 maximum / perLoop100, 251 meanCycles / perLoop1k, 252 stddevCycles / perLoop1k, 253 minCycles / perLoop1k, 254 maxCycles / perLoop1k); 255 256 } 257 } 258#endif 259}; 260 261// ---------------------------------------------------------------------------- 262// ThreadBase 263// ---------------------------------------------------------------------------- 264 265AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 266 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 267 : Thread(false /*canCallJava*/), 268 mType(type), 269 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 270 // mChannelMask 271 mChannelCount(0), 272 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 273 mParamStatus(NO_ERROR), 274 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 275 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 276 // mName will be set by concrete (non-virtual) subclass 277 mDeathRecipient(new PMDeathRecipient(this)) 278{ 279} 280 281AudioFlinger::ThreadBase::~ThreadBase() 282{ 283 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 284 for (size_t i = 0; i < mConfigEvents.size(); i++) { 285 delete mConfigEvents[i]; 286 } 287 mConfigEvents.clear(); 288 289 mParamCond.broadcast(); 290 // do not lock the mutex in destructor 291 releaseWakeLock_l(); 292 if (mPowerManager != 0) { 293 sp<IBinder> binder = mPowerManager->asBinder(); 294 binder->unlinkToDeath(mDeathRecipient); 295 } 296} 297 298void AudioFlinger::ThreadBase::exit() 299{ 300 ALOGV("ThreadBase::exit"); 301 // do any cleanup required for exit to succeed 302 preExit(); 303 { 304 // This lock prevents the following race in thread (uniprocessor for illustration): 305 // if (!exitPending()) { 306 // // context switch from here to exit() 307 // // exit() calls requestExit(), what exitPending() observes 308 // // exit() calls signal(), which is dropped since no waiters 309 // // context switch back from exit() to here 310 // mWaitWorkCV.wait(...); 311 // // now thread is hung 312 // } 313 AutoMutex lock(mLock); 314 requestExit(); 315 mWaitWorkCV.broadcast(); 316 } 317 // When Thread::requestExitAndWait is made virtual and this method is renamed to 318 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 319 requestExitAndWait(); 320} 321 322status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 323{ 324 status_t status; 325 326 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 327 Mutex::Autolock _l(mLock); 328 329 mNewParameters.add(keyValuePairs); 330 mWaitWorkCV.signal(); 331 // wait condition with timeout in case the thread loop has exited 332 // before the request could be processed 333 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 334 status = mParamStatus; 335 mWaitWorkCV.signal(); 336 } else { 337 status = TIMED_OUT; 338 } 339 return status; 340} 341 342void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 343{ 344 Mutex::Autolock _l(mLock); 345 sendIoConfigEvent_l(event, param); 346} 347 348// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 349void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 350{ 351 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 352 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 353 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 354 param); 355 mWaitWorkCV.signal(); 356} 357 358// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 359void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 360{ 361 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 362 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 363 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 364 mConfigEvents.size(), pid, tid, prio); 365 mWaitWorkCV.signal(); 366} 367 368void AudioFlinger::ThreadBase::processConfigEvents() 369{ 370 mLock.lock(); 371 while (!mConfigEvents.isEmpty()) { 372 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 373 ConfigEvent *event = mConfigEvents[0]; 374 mConfigEvents.removeAt(0); 375 // release mLock before locking AudioFlinger mLock: lock order is always 376 // AudioFlinger then ThreadBase to avoid cross deadlock 377 mLock.unlock(); 378 switch(event->type()) { 379 case CFG_EVENT_PRIO: { 380 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 381 // FIXME Need to understand why this has be done asynchronously 382 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 383 true /*asynchronous*/); 384 if (err != 0) { 385 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 386 "error %d", 387 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 388 } 389 } break; 390 case CFG_EVENT_IO: { 391 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 392 mAudioFlinger->mLock.lock(); 393 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 394 mAudioFlinger->mLock.unlock(); 395 } break; 396 default: 397 ALOGE("processConfigEvents() unknown event type %d", event->type()); 398 break; 399 } 400 delete event; 401 mLock.lock(); 402 } 403 mLock.unlock(); 404} 405 406void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 407{ 408 const size_t SIZE = 256; 409 char buffer[SIZE]; 410 String8 result; 411 412 bool locked = AudioFlinger::dumpTryLock(mLock); 413 if (!locked) { 414 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 415 write(fd, buffer, strlen(buffer)); 416 } 417 418 snprintf(buffer, SIZE, "io handle: %d\n", mId); 419 result.append(buffer); 420 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 437 result.append(buffer); 438 439 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 440 result.append(buffer); 441 result.append(" Index Command"); 442 for (size_t i = 0; i < mNewParameters.size(); ++i) { 443 snprintf(buffer, SIZE, "\n %02d ", i); 444 result.append(buffer); 445 result.append(mNewParameters[i]); 446 } 447 448 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 449 result.append(buffer); 450 for (size_t i = 0; i < mConfigEvents.size(); i++) { 451 mConfigEvents[i]->dump(buffer, SIZE); 452 result.append(buffer); 453 } 454 result.append("\n"); 455 456 write(fd, result.string(), result.size()); 457 458 if (locked) { 459 mLock.unlock(); 460 } 461} 462 463void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 464{ 465 const size_t SIZE = 256; 466 char buffer[SIZE]; 467 String8 result; 468 469 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 470 write(fd, buffer, strlen(buffer)); 471 472 for (size_t i = 0; i < mEffectChains.size(); ++i) { 473 sp<EffectChain> chain = mEffectChains[i]; 474 if (chain != 0) { 475 chain->dump(fd, args); 476 } 477 } 478} 479 480void AudioFlinger::ThreadBase::acquireWakeLock() 481{ 482 Mutex::Autolock _l(mLock); 483 acquireWakeLock_l(); 484} 485 486void AudioFlinger::ThreadBase::acquireWakeLock_l() 487{ 488 if (mPowerManager == 0) { 489 // use checkService() to avoid blocking if power service is not up yet 490 sp<IBinder> binder = 491 defaultServiceManager()->checkService(String16("power")); 492 if (binder == 0) { 493 ALOGW("Thread %s cannot connect to the power manager service", mName); 494 } else { 495 mPowerManager = interface_cast<IPowerManager>(binder); 496 binder->linkToDeath(mDeathRecipient); 497 } 498 } 499 if (mPowerManager != 0) { 500 sp<IBinder> binder = new BBinder(); 501 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 502 binder, 503 String16(mName), 504 String16("media")); 505 if (status == NO_ERROR) { 506 mWakeLockToken = binder; 507 } 508 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 509 } 510} 511 512void AudioFlinger::ThreadBase::releaseWakeLock() 513{ 514 Mutex::Autolock _l(mLock); 515 releaseWakeLock_l(); 516} 517 518void AudioFlinger::ThreadBase::releaseWakeLock_l() 519{ 520 if (mWakeLockToken != 0) { 521 ALOGV("releaseWakeLock_l() %s", mName); 522 if (mPowerManager != 0) { 523 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 524 } 525 mWakeLockToken.clear(); 526 } 527} 528 529void AudioFlinger::ThreadBase::clearPowerManager() 530{ 531 Mutex::Autolock _l(mLock); 532 releaseWakeLock_l(); 533 mPowerManager.clear(); 534} 535 536void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 537{ 538 sp<ThreadBase> thread = mThread.promote(); 539 if (thread != 0) { 540 thread->clearPowerManager(); 541 } 542 ALOGW("power manager service died !!!"); 543} 544 545void AudioFlinger::ThreadBase::setEffectSuspended( 546 const effect_uuid_t *type, bool suspend, int sessionId) 547{ 548 Mutex::Autolock _l(mLock); 549 setEffectSuspended_l(type, suspend, sessionId); 550} 551 552void AudioFlinger::ThreadBase::setEffectSuspended_l( 553 const effect_uuid_t *type, bool suspend, int sessionId) 554{ 555 sp<EffectChain> chain = getEffectChain_l(sessionId); 556 if (chain != 0) { 557 if (type != NULL) { 558 chain->setEffectSuspended_l(type, suspend); 559 } else { 560 chain->setEffectSuspendedAll_l(suspend); 561 } 562 } 563 564 updateSuspendedSessions_l(type, suspend, sessionId); 565} 566 567void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 568{ 569 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 570 if (index < 0) { 571 return; 572 } 573 574 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 575 mSuspendedSessions.valueAt(index); 576 577 for (size_t i = 0; i < sessionEffects.size(); i++) { 578 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 579 for (int j = 0; j < desc->mRefCount; j++) { 580 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 581 chain->setEffectSuspendedAll_l(true); 582 } else { 583 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 584 desc->mType.timeLow); 585 chain->setEffectSuspended_l(&desc->mType, true); 586 } 587 } 588 } 589} 590 591void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 592 bool suspend, 593 int sessionId) 594{ 595 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 596 597 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 598 599 if (suspend) { 600 if (index >= 0) { 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } else { 603 mSuspendedSessions.add(sessionId, sessionEffects); 604 } 605 } else { 606 if (index < 0) { 607 return; 608 } 609 sessionEffects = mSuspendedSessions.valueAt(index); 610 } 611 612 613 int key = EffectChain::kKeyForSuspendAll; 614 if (type != NULL) { 615 key = type->timeLow; 616 } 617 index = sessionEffects.indexOfKey(key); 618 619 sp<SuspendedSessionDesc> desc; 620 if (suspend) { 621 if (index >= 0) { 622 desc = sessionEffects.valueAt(index); 623 } else { 624 desc = new SuspendedSessionDesc(); 625 if (type != NULL) { 626 desc->mType = *type; 627 } 628 sessionEffects.add(key, desc); 629 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 630 } 631 desc->mRefCount++; 632 } else { 633 if (index < 0) { 634 return; 635 } 636 desc = sessionEffects.valueAt(index); 637 if (--desc->mRefCount == 0) { 638 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 639 sessionEffects.removeItemsAt(index); 640 if (sessionEffects.isEmpty()) { 641 ALOGV("updateSuspendedSessions_l() restore removing session %d", 642 sessionId); 643 mSuspendedSessions.removeItem(sessionId); 644 } 645 } 646 } 647 if (!sessionEffects.isEmpty()) { 648 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 649 } 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 Mutex::Autolock _l(mLock); 657 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 658} 659 660void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 661 bool enabled, 662 int sessionId) 663{ 664 if (mType != RECORD) { 665 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 666 // another session. This gives the priority to well behaved effect control panels 667 // and applications not using global effects. 668 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 669 // global effects 670 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 671 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 672 } 673 } 674 675 sp<EffectChain> chain = getEffectChain_l(sessionId); 676 if (chain != 0) { 677 chain->checkSuspendOnEffectEnabled(effect, enabled); 678 } 679} 680 681// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 682sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 683 const sp<AudioFlinger::Client>& client, 684 const sp<IEffectClient>& effectClient, 685 int32_t priority, 686 int sessionId, 687 effect_descriptor_t *desc, 688 int *enabled, 689 status_t *status 690 ) 691{ 692 sp<EffectModule> effect; 693 sp<EffectHandle> handle; 694 status_t lStatus; 695 sp<EffectChain> chain; 696 bool chainCreated = false; 697 bool effectCreated = false; 698 bool effectRegistered = false; 699 700 lStatus = initCheck(); 701 if (lStatus != NO_ERROR) { 702 ALOGW("createEffect_l() Audio driver not initialized."); 703 goto Exit; 704 } 705 706 // Do not allow effects with session ID 0 on direct output or duplicating threads 707 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 708 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 709 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 710 desc->name, sessionId); 711 lStatus = BAD_VALUE; 712 goto Exit; 713 } 714 // Only Pre processor effects are allowed on input threads and only on input threads 715 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 716 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 717 desc->name, desc->flags, mType); 718 lStatus = BAD_VALUE; 719 goto Exit; 720 } 721 722 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 723 724 { // scope for mLock 725 Mutex::Autolock _l(mLock); 726 727 // check for existing effect chain with the requested audio session 728 chain = getEffectChain_l(sessionId); 729 if (chain == 0) { 730 // create a new chain for this session 731 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 732 chain = new EffectChain(this, sessionId); 733 addEffectChain_l(chain); 734 chain->setStrategy(getStrategyForSession_l(sessionId)); 735 chainCreated = true; 736 } else { 737 effect = chain->getEffectFromDesc_l(desc); 738 } 739 740 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 741 742 if (effect == 0) { 743 int id = mAudioFlinger->nextUniqueId(); 744 // Check CPU and memory usage 745 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 746 if (lStatus != NO_ERROR) { 747 goto Exit; 748 } 749 effectRegistered = true; 750 // create a new effect module if none present in the chain 751 effect = new EffectModule(this, chain, desc, id, sessionId); 752 lStatus = effect->status(); 753 if (lStatus != NO_ERROR) { 754 goto Exit; 755 } 756 lStatus = chain->addEffect_l(effect); 757 if (lStatus != NO_ERROR) { 758 goto Exit; 759 } 760 effectCreated = true; 761 762 effect->setDevice(mOutDevice); 763 effect->setDevice(mInDevice); 764 effect->setMode(mAudioFlinger->getMode()); 765 effect->setAudioSource(mAudioSource); 766 } 767 // create effect handle and connect it to effect module 768 handle = new EffectHandle(effect, client, effectClient, priority); 769 lStatus = effect->addHandle(handle.get()); 770 if (enabled != NULL) { 771 *enabled = (int)effect->isEnabled(); 772 } 773 } 774 775Exit: 776 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 777 Mutex::Autolock _l(mLock); 778 if (effectCreated) { 779 chain->removeEffect_l(effect); 780 } 781 if (effectRegistered) { 782 AudioSystem::unregisterEffect(effect->id()); 783 } 784 if (chainCreated) { 785 removeEffectChain_l(chain); 786 } 787 handle.clear(); 788 } 789 790 if (status != NULL) { 791 *status = lStatus; 792 } 793 return handle; 794} 795 796sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 797{ 798 Mutex::Autolock _l(mLock); 799 return getEffect_l(sessionId, effectId); 800} 801 802sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 803{ 804 sp<EffectChain> chain = getEffectChain_l(sessionId); 805 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 806} 807 808// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 809// PlaybackThread::mLock held 810status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 811{ 812 // check for existing effect chain with the requested audio session 813 int sessionId = effect->sessionId(); 814 sp<EffectChain> chain = getEffectChain_l(sessionId); 815 bool chainCreated = false; 816 817 if (chain == 0) { 818 // create a new chain for this session 819 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 820 chain = new EffectChain(this, sessionId); 821 addEffectChain_l(chain); 822 chain->setStrategy(getStrategyForSession_l(sessionId)); 823 chainCreated = true; 824 } 825 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 826 827 if (chain->getEffectFromId_l(effect->id()) != 0) { 828 ALOGW("addEffect_l() %p effect %s already present in chain %p", 829 this, effect->desc().name, chain.get()); 830 return BAD_VALUE; 831 } 832 833 status_t status = chain->addEffect_l(effect); 834 if (status != NO_ERROR) { 835 if (chainCreated) { 836 removeEffectChain_l(chain); 837 } 838 return status; 839 } 840 841 effect->setDevice(mOutDevice); 842 effect->setDevice(mInDevice); 843 effect->setMode(mAudioFlinger->getMode()); 844 effect->setAudioSource(mAudioSource); 845 return NO_ERROR; 846} 847 848void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 849 850 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 851 effect_descriptor_t desc = effect->desc(); 852 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 853 detachAuxEffect_l(effect->id()); 854 } 855 856 sp<EffectChain> chain = effect->chain().promote(); 857 if (chain != 0) { 858 // remove effect chain if removing last effect 859 if (chain->removeEffect_l(effect) == 0) { 860 removeEffectChain_l(chain); 861 } 862 } else { 863 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 864 } 865} 866 867void AudioFlinger::ThreadBase::lockEffectChains_l( 868 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 869{ 870 effectChains = mEffectChains; 871 for (size_t i = 0; i < mEffectChains.size(); i++) { 872 mEffectChains[i]->lock(); 873 } 874} 875 876void AudioFlinger::ThreadBase::unlockEffectChains( 877 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 878{ 879 for (size_t i = 0; i < effectChains.size(); i++) { 880 effectChains[i]->unlock(); 881 } 882} 883 884sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 885{ 886 Mutex::Autolock _l(mLock); 887 return getEffectChain_l(sessionId); 888} 889 890sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 891{ 892 size_t size = mEffectChains.size(); 893 for (size_t i = 0; i < size; i++) { 894 if (mEffectChains[i]->sessionId() == sessionId) { 895 return mEffectChains[i]; 896 } 897 } 898 return 0; 899} 900 901void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 902{ 903 Mutex::Autolock _l(mLock); 904 size_t size = mEffectChains.size(); 905 for (size_t i = 0; i < size; i++) { 906 mEffectChains[i]->setMode_l(mode); 907 } 908} 909 910void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 911 EffectHandle *handle, 912 bool unpinIfLast) { 913 914 Mutex::Autolock _l(mLock); 915 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 916 // delete the effect module if removing last handle on it 917 if (effect->removeHandle(handle) == 0) { 918 if (!effect->isPinned() || unpinIfLast) { 919 removeEffect_l(effect); 920 AudioSystem::unregisterEffect(effect->id()); 921 } 922 } 923} 924 925// ---------------------------------------------------------------------------- 926// Playback 927// ---------------------------------------------------------------------------- 928 929AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 930 AudioStreamOut* output, 931 audio_io_handle_t id, 932 audio_devices_t device, 933 type_t type) 934 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 935 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 936 // mStreamTypes[] initialized in constructor body 937 mOutput(output), 938 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 939 mMixerStatus(MIXER_IDLE), 940 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 941 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 942 mBytesRemaining(0), 943 mCurrentWriteLength(0), 944 mUseAsyncWrite(false), 945 mWriteBlocked(false), 946 mDraining(false), 947 mScreenState(AudioFlinger::mScreenState), 948 // index 0 is reserved for normal mixer's submix 949 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 950{ 951 snprintf(mName, kNameLength, "AudioOut_%X", id); 952 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 953 954 // Assumes constructor is called by AudioFlinger with it's mLock held, but 955 // it would be safer to explicitly pass initial masterVolume/masterMute as 956 // parameter. 957 // 958 // If the HAL we are using has support for master volume or master mute, 959 // then do not attenuate or mute during mixing (just leave the volume at 1.0 960 // and the mute set to false). 961 mMasterVolume = audioFlinger->masterVolume_l(); 962 mMasterMute = audioFlinger->masterMute_l(); 963 if (mOutput && mOutput->audioHwDev) { 964 if (mOutput->audioHwDev->canSetMasterVolume()) { 965 mMasterVolume = 1.0; 966 } 967 968 if (mOutput->audioHwDev->canSetMasterMute()) { 969 mMasterMute = false; 970 } 971 } 972 973 readOutputParameters(); 974 975 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 976 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 977 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 978 stream = (audio_stream_type_t) (stream + 1)) { 979 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 980 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 981 } 982 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 983 // because mAudioFlinger doesn't have one to copy from 984} 985 986AudioFlinger::PlaybackThread::~PlaybackThread() 987{ 988 mAudioFlinger->unregisterWriter(mNBLogWriter); 989 delete [] mAllocMixBuffer; 990} 991 992void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 993{ 994 dumpInternals(fd, args); 995 dumpTracks(fd, args); 996 dumpEffectChains(fd, args); 997} 998 999void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1000{ 1001 const size_t SIZE = 256; 1002 char buffer[SIZE]; 1003 String8 result; 1004 1005 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1006 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1007 const stream_type_t *st = &mStreamTypes[i]; 1008 if (i > 0) { 1009 result.appendFormat(", "); 1010 } 1011 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1012 if (st->mute) { 1013 result.append("M"); 1014 } 1015 } 1016 result.append("\n"); 1017 write(fd, result.string(), result.length()); 1018 result.clear(); 1019 1020 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1021 result.append(buffer); 1022 Track::appendDumpHeader(result); 1023 for (size_t i = 0; i < mTracks.size(); ++i) { 1024 sp<Track> track = mTracks[i]; 1025 if (track != 0) { 1026 track->dump(buffer, SIZE); 1027 result.append(buffer); 1028 } 1029 } 1030 1031 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1032 result.append(buffer); 1033 Track::appendDumpHeader(result); 1034 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1035 sp<Track> track = mActiveTracks[i].promote(); 1036 if (track != 0) { 1037 track->dump(buffer, SIZE); 1038 result.append(buffer); 1039 } 1040 } 1041 write(fd, result.string(), result.size()); 1042 1043 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1044 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1045 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1046 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1047} 1048 1049void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1050{ 1051 const size_t SIZE = 256; 1052 char buffer[SIZE]; 1053 String8 result; 1054 1055 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1056 result.append(buffer); 1057 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1058 ns2ms(systemTime() - mLastWriteTime)); 1059 result.append(buffer); 1060 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1061 result.append(buffer); 1062 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1063 result.append(buffer); 1064 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1065 result.append(buffer); 1066 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1067 result.append(buffer); 1068 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1069 result.append(buffer); 1070 write(fd, result.string(), result.size()); 1071 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1072 1073 dumpBase(fd, args); 1074} 1075 1076// Thread virtuals 1077status_t AudioFlinger::PlaybackThread::readyToRun() 1078{ 1079 status_t status = initCheck(); 1080 if (status == NO_ERROR) { 1081 ALOGI("AudioFlinger's thread %p ready to run", this); 1082 } else { 1083 ALOGE("No working audio driver found."); 1084 } 1085 return status; 1086} 1087 1088void AudioFlinger::PlaybackThread::onFirstRef() 1089{ 1090 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1091} 1092 1093// ThreadBase virtuals 1094void AudioFlinger::PlaybackThread::preExit() 1095{ 1096 ALOGV(" preExit()"); 1097 // FIXME this is using hard-coded strings but in the future, this functionality will be 1098 // converted to use audio HAL extensions required to support tunneling 1099 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1100} 1101 1102// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1103sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1104 const sp<AudioFlinger::Client>& client, 1105 audio_stream_type_t streamType, 1106 uint32_t sampleRate, 1107 audio_format_t format, 1108 audio_channel_mask_t channelMask, 1109 size_t frameCount, 1110 const sp<IMemory>& sharedBuffer, 1111 int sessionId, 1112 IAudioFlinger::track_flags_t *flags, 1113 pid_t tid, 1114 status_t *status) 1115{ 1116 sp<Track> track; 1117 status_t lStatus; 1118 1119 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1120 1121 // client expresses a preference for FAST, but we get the final say 1122 if (*flags & IAudioFlinger::TRACK_FAST) { 1123 if ( 1124 // not timed 1125 (!isTimed) && 1126 // either of these use cases: 1127 ( 1128 // use case 1: shared buffer with any frame count 1129 ( 1130 (sharedBuffer != 0) 1131 ) || 1132 // use case 2: callback handler and frame count is default or at least as large as HAL 1133 ( 1134 (tid != -1) && 1135 ((frameCount == 0) || 1136 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1137 ) 1138 ) && 1139 // PCM data 1140 audio_is_linear_pcm(format) && 1141 // mono or stereo 1142 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1143 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1144#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1145 // hardware sample rate 1146 (sampleRate == mSampleRate) && 1147#endif 1148 // normal mixer has an associated fast mixer 1149 hasFastMixer() && 1150 // there are sufficient fast track slots available 1151 (mFastTrackAvailMask != 0) 1152 // FIXME test that MixerThread for this fast track has a capable output HAL 1153 // FIXME add a permission test also? 1154 ) { 1155 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1156 if (frameCount == 0) { 1157 frameCount = mFrameCount * kFastTrackMultiplier; 1158 } 1159 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1160 frameCount, mFrameCount); 1161 } else { 1162 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1163 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1164 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1165 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1166 audio_is_linear_pcm(format), 1167 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1168 *flags &= ~IAudioFlinger::TRACK_FAST; 1169 // For compatibility with AudioTrack calculation, buffer depth is forced 1170 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1171 // This is probably too conservative, but legacy application code may depend on it. 1172 // If you change this calculation, also review the start threshold which is related. 1173 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1174 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1175 if (minBufCount < 2) { 1176 minBufCount = 2; 1177 } 1178 size_t minFrameCount = mNormalFrameCount * minBufCount; 1179 if (frameCount < minFrameCount) { 1180 frameCount = minFrameCount; 1181 } 1182 } 1183 } 1184 1185 if (mType == DIRECT) { 1186 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1187 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1188 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1189 "for output %p with format %d", 1190 sampleRate, format, channelMask, mOutput, mFormat); 1191 lStatus = BAD_VALUE; 1192 goto Exit; 1193 } 1194 } 1195 } else if (mType == OFFLOAD) { 1196 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1197 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1198 "for output %p with format %d", 1199 sampleRate, format, channelMask, mOutput, mFormat); 1200 lStatus = BAD_VALUE; 1201 goto Exit; 1202 } 1203 } else { 1204 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1205 ALOGE("createTrack_l() Bad parameter: format %d \"" 1206 "for output %p with format %d", 1207 format, mOutput, mFormat); 1208 lStatus = BAD_VALUE; 1209 goto Exit; 1210 } 1211 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1212 if (sampleRate > mSampleRate*2) { 1213 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1214 lStatus = BAD_VALUE; 1215 goto Exit; 1216 } 1217 } 1218 1219 lStatus = initCheck(); 1220 if (lStatus != NO_ERROR) { 1221 ALOGE("Audio driver not initialized."); 1222 goto Exit; 1223 } 1224 1225 { // scope for mLock 1226 Mutex::Autolock _l(mLock); 1227 1228 // all tracks in same audio session must share the same routing strategy otherwise 1229 // conflicts will happen when tracks are moved from one output to another by audio policy 1230 // manager 1231 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1232 for (size_t i = 0; i < mTracks.size(); ++i) { 1233 sp<Track> t = mTracks[i]; 1234 if (t != 0 && !t->isOutputTrack()) { 1235 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1236 if (sessionId == t->sessionId() && strategy != actual) { 1237 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1238 strategy, actual); 1239 lStatus = BAD_VALUE; 1240 goto Exit; 1241 } 1242 } 1243 } 1244 1245 if (!isTimed) { 1246 track = new Track(this, client, streamType, sampleRate, format, 1247 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1248 } else { 1249 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1250 channelMask, frameCount, sharedBuffer, sessionId); 1251 } 1252 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1253 lStatus = NO_MEMORY; 1254 goto Exit; 1255 } 1256 1257 mTracks.add(track); 1258 1259 sp<EffectChain> chain = getEffectChain_l(sessionId); 1260 if (chain != 0) { 1261 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1262 track->setMainBuffer(chain->inBuffer()); 1263 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1264 chain->incTrackCnt(); 1265 } 1266 1267 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1268 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1269 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1270 // so ask activity manager to do this on our behalf 1271 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1272 } 1273 } 1274 1275 lStatus = NO_ERROR; 1276 1277Exit: 1278 if (status) { 1279 *status = lStatus; 1280 } 1281 return track; 1282} 1283 1284uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1285{ 1286 return latency; 1287} 1288 1289uint32_t AudioFlinger::PlaybackThread::latency() const 1290{ 1291 Mutex::Autolock _l(mLock); 1292 return latency_l(); 1293} 1294uint32_t AudioFlinger::PlaybackThread::latency_l() const 1295{ 1296 if (initCheck() == NO_ERROR) { 1297 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1298 } else { 1299 return 0; 1300 } 1301} 1302 1303void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1304{ 1305 Mutex::Autolock _l(mLock); 1306 // Don't apply master volume in SW if our HAL can do it for us. 1307 if (mOutput && mOutput->audioHwDev && 1308 mOutput->audioHwDev->canSetMasterVolume()) { 1309 mMasterVolume = 1.0; 1310 } else { 1311 mMasterVolume = value; 1312 } 1313} 1314 1315void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1316{ 1317 Mutex::Autolock _l(mLock); 1318 // Don't apply master mute in SW if our HAL can do it for us. 1319 if (mOutput && mOutput->audioHwDev && 1320 mOutput->audioHwDev->canSetMasterMute()) { 1321 mMasterMute = false; 1322 } else { 1323 mMasterMute = muted; 1324 } 1325} 1326 1327void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1328{ 1329 Mutex::Autolock _l(mLock); 1330 mStreamTypes[stream].volume = value; 1331 signal_l(); 1332} 1333 1334void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1335{ 1336 Mutex::Autolock _l(mLock); 1337 mStreamTypes[stream].mute = muted; 1338 signal_l(); 1339} 1340 1341float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1342{ 1343 Mutex::Autolock _l(mLock); 1344 return mStreamTypes[stream].volume; 1345} 1346 1347// addTrack_l() must be called with ThreadBase::mLock held 1348status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1349{ 1350 status_t status = ALREADY_EXISTS; 1351 1352 // set retry count for buffer fill 1353 track->mRetryCount = kMaxTrackStartupRetries; 1354 if (mActiveTracks.indexOf(track) < 0) { 1355 // the track is newly added, make sure it fills up all its 1356 // buffers before playing. This is to ensure the client will 1357 // effectively get the latency it requested. 1358 if (!track->isOutputTrack()) { 1359 TrackBase::track_state state = track->mState; 1360 mLock.unlock(); 1361 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1362 mLock.lock(); 1363 // abort track was stopped/paused while we released the lock 1364 if (state != track->mState) { 1365 if (status == NO_ERROR) { 1366 mLock.unlock(); 1367 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1368 mLock.lock(); 1369 } 1370 return INVALID_OPERATION; 1371 } 1372 // abort if start is rejected by audio policy manager 1373 if (status != NO_ERROR) { 1374 return PERMISSION_DENIED; 1375 } 1376#ifdef ADD_BATTERY_DATA 1377 // to track the speaker usage 1378 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1379#endif 1380 } 1381 1382 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1383 track->mResetDone = false; 1384 track->mPresentationCompleteFrames = 0; 1385 mActiveTracks.add(track); 1386 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1387 if (chain != 0) { 1388 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1389 track->sessionId()); 1390 chain->incActiveTrackCnt(); 1391 } 1392 1393 status = NO_ERROR; 1394 } 1395 1396 ALOGV("mWaitWorkCV.broadcast"); 1397 mWaitWorkCV.broadcast(); 1398 1399 return status; 1400} 1401 1402bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1403{ 1404 track->terminate(); 1405 // active tracks are removed by threadLoop() 1406 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1407 track->mState = TrackBase::STOPPED; 1408 if (!trackActive) { 1409 removeTrack_l(track); 1410 } else if (track->isFastTrack() || track->isOffloaded()) { 1411 track->mState = TrackBase::STOPPING_1; 1412 } 1413 1414 return trackActive; 1415} 1416 1417void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1418{ 1419 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1420 mTracks.remove(track); 1421 deleteTrackName_l(track->name()); 1422 // redundant as track is about to be destroyed, for dumpsys only 1423 track->mName = -1; 1424 if (track->isFastTrack()) { 1425 int index = track->mFastIndex; 1426 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1427 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1428 mFastTrackAvailMask |= 1 << index; 1429 // redundant as track is about to be destroyed, for dumpsys only 1430 track->mFastIndex = -1; 1431 } 1432 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1433 if (chain != 0) { 1434 chain->decTrackCnt(); 1435 } 1436} 1437 1438void AudioFlinger::PlaybackThread::signal_l() 1439{ 1440 // Thread could be blocked waiting for async 1441 // so signal it to handle state changes immediately 1442 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1443 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1444 mSignalPending = true; 1445 mWaitWorkCV.signal(); 1446} 1447 1448String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1449{ 1450 String8 out_s8 = String8(""); 1451 char *s; 1452 1453 Mutex::Autolock _l(mLock); 1454 if (initCheck() != NO_ERROR) { 1455 return out_s8; 1456 } 1457 1458 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1459 out_s8 = String8(s); 1460 free(s); 1461 return out_s8; 1462} 1463 1464// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1465void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1466 AudioSystem::OutputDescriptor desc; 1467 void *param2 = NULL; 1468 1469 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1470 param); 1471 1472 switch (event) { 1473 case AudioSystem::OUTPUT_OPENED: 1474 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1475 desc.channels = mChannelMask; 1476 desc.samplingRate = mSampleRate; 1477 desc.format = mFormat; 1478 desc.frameCount = mNormalFrameCount; // FIXME see 1479 // AudioFlinger::frameCount(audio_io_handle_t) 1480 desc.latency = latency(); 1481 param2 = &desc; 1482 break; 1483 1484 case AudioSystem::STREAM_CONFIG_CHANGED: 1485 param2 = ¶m; 1486 case AudioSystem::OUTPUT_CLOSED: 1487 default: 1488 break; 1489 } 1490 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1491} 1492 1493void AudioFlinger::PlaybackThread::writeCallback() 1494{ 1495 ALOG_ASSERT(mCallbackThread != 0); 1496 mCallbackThread->setWriteBlocked(false); 1497} 1498 1499void AudioFlinger::PlaybackThread::drainCallback() 1500{ 1501 ALOG_ASSERT(mCallbackThread != 0); 1502 mCallbackThread->setDraining(false); 1503} 1504 1505void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1506{ 1507 Mutex::Autolock _l(mLock); 1508 mWriteBlocked = value; 1509 if (!value) { 1510 mWaitWorkCV.signal(); 1511 } 1512} 1513 1514void AudioFlinger::PlaybackThread::setDraining(bool value) 1515{ 1516 Mutex::Autolock _l(mLock); 1517 mDraining = value; 1518 if (!value) { 1519 mWaitWorkCV.signal(); 1520 } 1521} 1522 1523// static 1524int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1525 void *param, 1526 void *cookie) 1527{ 1528 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1529 ALOGV("asyncCallback() event %d", event); 1530 switch (event) { 1531 case STREAM_CBK_EVENT_WRITE_READY: 1532 me->writeCallback(); 1533 break; 1534 case STREAM_CBK_EVENT_DRAIN_READY: 1535 me->drainCallback(); 1536 break; 1537 default: 1538 ALOGW("asyncCallback() unknown event %d", event); 1539 break; 1540 } 1541 return 0; 1542} 1543 1544void AudioFlinger::PlaybackThread::readOutputParameters() 1545{ 1546 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1547 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1548 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1549 if (!audio_is_output_channel(mChannelMask)) { 1550 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1551 } 1552 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1553 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1554 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1555 } 1556 mChannelCount = popcount(mChannelMask); 1557 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1558 if (!audio_is_valid_format(mFormat)) { 1559 LOG_FATAL("HAL format %d not valid for output", mFormat); 1560 } 1561 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1562 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1563 mFormat); 1564 } 1565 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1566 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1567 if (mFrameCount & 15) { 1568 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1569 mFrameCount); 1570 } 1571 1572 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1573 (mOutput->stream->set_callback != NULL)) { 1574 if (mOutput->stream->set_callback(mOutput->stream, 1575 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1576 mUseAsyncWrite = true; 1577 } 1578 } 1579 1580 // Calculate size of normal mix buffer relative to the HAL output buffer size 1581 double multiplier = 1.0; 1582 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1583 kUseFastMixer == FastMixer_Dynamic)) { 1584 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1585 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1586 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1587 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1588 maxNormalFrameCount = maxNormalFrameCount & ~15; 1589 if (maxNormalFrameCount < minNormalFrameCount) { 1590 maxNormalFrameCount = minNormalFrameCount; 1591 } 1592 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1593 if (multiplier <= 1.0) { 1594 multiplier = 1.0; 1595 } else if (multiplier <= 2.0) { 1596 if (2 * mFrameCount <= maxNormalFrameCount) { 1597 multiplier = 2.0; 1598 } else { 1599 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1600 } 1601 } else { 1602 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1603 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1604 // track, but we sometimes have to do this to satisfy the maximum frame count 1605 // constraint) 1606 // FIXME this rounding up should not be done if no HAL SRC 1607 uint32_t truncMult = (uint32_t) multiplier; 1608 if ((truncMult & 1)) { 1609 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1610 ++truncMult; 1611 } 1612 } 1613 multiplier = (double) truncMult; 1614 } 1615 } 1616 mNormalFrameCount = multiplier * mFrameCount; 1617 // round up to nearest 16 frames to satisfy AudioMixer 1618 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1619 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1620 mNormalFrameCount); 1621 1622 delete[] mAllocMixBuffer; 1623 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1624 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1625 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1626 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1627 1628 // force reconfiguration of effect chains and engines to take new buffer size and audio 1629 // parameters into account 1630 // Note that mLock is not held when readOutputParameters() is called from the constructor 1631 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1632 // matter. 1633 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1634 Vector< sp<EffectChain> > effectChains = mEffectChains; 1635 for (size_t i = 0; i < effectChains.size(); i ++) { 1636 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1637 } 1638} 1639 1640 1641status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1642{ 1643 if (halFrames == NULL || dspFrames == NULL) { 1644 return BAD_VALUE; 1645 } 1646 Mutex::Autolock _l(mLock); 1647 if (initCheck() != NO_ERROR) { 1648 return INVALID_OPERATION; 1649 } 1650 size_t framesWritten = mBytesWritten / mFrameSize; 1651 *halFrames = framesWritten; 1652 1653 if (isSuspended()) { 1654 // return an estimation of rendered frames when the output is suspended 1655 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1656 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1657 return NO_ERROR; 1658 } else { 1659 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1660 } 1661} 1662 1663uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1664{ 1665 Mutex::Autolock _l(mLock); 1666 uint32_t result = 0; 1667 if (getEffectChain_l(sessionId) != 0) { 1668 result = EFFECT_SESSION; 1669 } 1670 1671 for (size_t i = 0; i < mTracks.size(); ++i) { 1672 sp<Track> track = mTracks[i]; 1673 if (sessionId == track->sessionId() && !track->isInvalid()) { 1674 result |= TRACK_SESSION; 1675 break; 1676 } 1677 } 1678 1679 return result; 1680} 1681 1682uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1683{ 1684 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1685 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1686 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1687 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1688 } 1689 for (size_t i = 0; i < mTracks.size(); i++) { 1690 sp<Track> track = mTracks[i]; 1691 if (sessionId == track->sessionId() && !track->isInvalid()) { 1692 return AudioSystem::getStrategyForStream(track->streamType()); 1693 } 1694 } 1695 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1696} 1697 1698 1699AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1700{ 1701 Mutex::Autolock _l(mLock); 1702 return mOutput; 1703} 1704 1705AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1706{ 1707 Mutex::Autolock _l(mLock); 1708 AudioStreamOut *output = mOutput; 1709 mOutput = NULL; 1710 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1711 // must push a NULL and wait for ack 1712 mOutputSink.clear(); 1713 mPipeSink.clear(); 1714 mNormalSink.clear(); 1715 return output; 1716} 1717 1718// this method must always be called either with ThreadBase mLock held or inside the thread loop 1719audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1720{ 1721 if (mOutput == NULL) { 1722 return NULL; 1723 } 1724 return &mOutput->stream->common; 1725} 1726 1727uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1728{ 1729 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1730} 1731 1732status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1733{ 1734 if (!isValidSyncEvent(event)) { 1735 return BAD_VALUE; 1736 } 1737 1738 Mutex::Autolock _l(mLock); 1739 1740 for (size_t i = 0; i < mTracks.size(); ++i) { 1741 sp<Track> track = mTracks[i]; 1742 if (event->triggerSession() == track->sessionId()) { 1743 (void) track->setSyncEvent(event); 1744 return NO_ERROR; 1745 } 1746 } 1747 1748 return NAME_NOT_FOUND; 1749} 1750 1751bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1752{ 1753 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1754} 1755 1756void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1757 const Vector< sp<Track> >& tracksToRemove) 1758{ 1759 size_t count = tracksToRemove.size(); 1760 if (CC_UNLIKELY(count)) { 1761 for (size_t i = 0 ; i < count ; i++) { 1762 const sp<Track>& track = tracksToRemove.itemAt(i); 1763 if (!track->isOutputTrack()) { 1764 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1765#ifdef ADD_BATTERY_DATA 1766 // to track the speaker usage 1767 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1768#endif 1769 if (track->isTerminated()) { 1770 AudioSystem::releaseOutput(mId); 1771 } 1772 } 1773 } 1774 } 1775} 1776 1777void AudioFlinger::PlaybackThread::checkSilentMode_l() 1778{ 1779 if (!mMasterMute) { 1780 char value[PROPERTY_VALUE_MAX]; 1781 if (property_get("ro.audio.silent", value, "0") > 0) { 1782 char *endptr; 1783 unsigned long ul = strtoul(value, &endptr, 0); 1784 if (*endptr == '\0' && ul != 0) { 1785 ALOGD("Silence is golden"); 1786 // The setprop command will not allow a property to be changed after 1787 // the first time it is set, so we don't have to worry about un-muting. 1788 setMasterMute_l(true); 1789 } 1790 } 1791 } 1792} 1793 1794// shared by MIXER and DIRECT, overridden by DUPLICATING 1795ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1796{ 1797 // FIXME rewrite to reduce number of system calls 1798 mLastWriteTime = systemTime(); 1799 mInWrite = true; 1800 ssize_t bytesWritten; 1801 1802 // If an NBAIO sink is present, use it to write the normal mixer's submix 1803 if (mNormalSink != 0) { 1804#define mBitShift 2 // FIXME 1805 size_t count = mBytesRemaining >> mBitShift; 1806 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1807 ATRACE_BEGIN("write"); 1808 // update the setpoint when AudioFlinger::mScreenState changes 1809 uint32_t screenState = AudioFlinger::mScreenState; 1810 if (screenState != mScreenState) { 1811 mScreenState = screenState; 1812 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1813 if (pipe != NULL) { 1814 pipe->setAvgFrames((mScreenState & 1) ? 1815 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1816 } 1817 } 1818 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1819 ATRACE_END(); 1820 if (framesWritten > 0) { 1821 bytesWritten = framesWritten << mBitShift; 1822 } else { 1823 bytesWritten = framesWritten; 1824 } 1825 // otherwise use the HAL / AudioStreamOut directly 1826 } else { 1827 // Direct output and offload threads 1828 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1829 if (mUseAsyncWrite) { 1830 mWriteBlocked = true; 1831 ALOG_ASSERT(mCallbackThread != 0); 1832 mCallbackThread->setWriteBlocked(true); 1833 } 1834 bytesWritten = mOutput->stream->write(mOutput->stream, 1835 mMixBuffer + offset, mBytesRemaining); 1836 if (mUseAsyncWrite && 1837 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1838 // do not wait for async callback in case of error of full write 1839 mWriteBlocked = false; 1840 ALOG_ASSERT(mCallbackThread != 0); 1841 mCallbackThread->setWriteBlocked(false); 1842 } 1843 } 1844 1845 mNumWrites++; 1846 mInWrite = false; 1847 1848 return bytesWritten; 1849} 1850 1851void AudioFlinger::PlaybackThread::threadLoop_drain() 1852{ 1853 if (mOutput->stream->drain) { 1854 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1855 if (mUseAsyncWrite) { 1856 mDraining = true; 1857 ALOG_ASSERT(mCallbackThread != 0); 1858 mCallbackThread->setDraining(true); 1859 } 1860 mOutput->stream->drain(mOutput->stream, 1861 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1862 : AUDIO_DRAIN_ALL); 1863 } 1864} 1865 1866void AudioFlinger::PlaybackThread::threadLoop_exit() 1867{ 1868 // Default implementation has nothing to do 1869} 1870 1871/* 1872The derived values that are cached: 1873 - mixBufferSize from frame count * frame size 1874 - activeSleepTime from activeSleepTimeUs() 1875 - idleSleepTime from idleSleepTimeUs() 1876 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1877 - maxPeriod from frame count and sample rate (MIXER only) 1878 1879The parameters that affect these derived values are: 1880 - frame count 1881 - frame size 1882 - sample rate 1883 - device type: A2DP or not 1884 - device latency 1885 - format: PCM or not 1886 - active sleep time 1887 - idle sleep time 1888*/ 1889 1890void AudioFlinger::PlaybackThread::cacheParameters_l() 1891{ 1892 mixBufferSize = mNormalFrameCount * mFrameSize; 1893 activeSleepTime = activeSleepTimeUs(); 1894 idleSleepTime = idleSleepTimeUs(); 1895} 1896 1897void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1898{ 1899 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1900 this, streamType, mTracks.size()); 1901 Mutex::Autolock _l(mLock); 1902 1903 size_t size = mTracks.size(); 1904 for (size_t i = 0; i < size; i++) { 1905 sp<Track> t = mTracks[i]; 1906 if (t->streamType() == streamType) { 1907 t->invalidate(); 1908 } 1909 } 1910} 1911 1912status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1913{ 1914 int session = chain->sessionId(); 1915 int16_t *buffer = mMixBuffer; 1916 bool ownsBuffer = false; 1917 1918 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1919 if (session > 0) { 1920 // Only one effect chain can be present in direct output thread and it uses 1921 // the mix buffer as input 1922 if (mType != DIRECT) { 1923 size_t numSamples = mNormalFrameCount * mChannelCount; 1924 buffer = new int16_t[numSamples]; 1925 memset(buffer, 0, numSamples * sizeof(int16_t)); 1926 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1927 ownsBuffer = true; 1928 } 1929 1930 // Attach all tracks with same session ID to this chain. 1931 for (size_t i = 0; i < mTracks.size(); ++i) { 1932 sp<Track> track = mTracks[i]; 1933 if (session == track->sessionId()) { 1934 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1935 buffer); 1936 track->setMainBuffer(buffer); 1937 chain->incTrackCnt(); 1938 } 1939 } 1940 1941 // indicate all active tracks in the chain 1942 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1943 sp<Track> track = mActiveTracks[i].promote(); 1944 if (track == 0) { 1945 continue; 1946 } 1947 if (session == track->sessionId()) { 1948 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1949 chain->incActiveTrackCnt(); 1950 } 1951 } 1952 } 1953 1954 chain->setInBuffer(buffer, ownsBuffer); 1955 chain->setOutBuffer(mMixBuffer); 1956 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1957 // chains list in order to be processed last as it contains output stage effects 1958 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1959 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1960 // after track specific effects and before output stage 1961 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1962 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1963 // Effect chain for other sessions are inserted at beginning of effect 1964 // chains list to be processed before output mix effects. Relative order between other 1965 // sessions is not important 1966 size_t size = mEffectChains.size(); 1967 size_t i = 0; 1968 for (i = 0; i < size; i++) { 1969 if (mEffectChains[i]->sessionId() < session) { 1970 break; 1971 } 1972 } 1973 mEffectChains.insertAt(chain, i); 1974 checkSuspendOnAddEffectChain_l(chain); 1975 1976 return NO_ERROR; 1977} 1978 1979size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1980{ 1981 int session = chain->sessionId(); 1982 1983 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1984 1985 for (size_t i = 0; i < mEffectChains.size(); i++) { 1986 if (chain == mEffectChains[i]) { 1987 mEffectChains.removeAt(i); 1988 // detach all active tracks from the chain 1989 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1990 sp<Track> track = mActiveTracks[i].promote(); 1991 if (track == 0) { 1992 continue; 1993 } 1994 if (session == track->sessionId()) { 1995 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1996 chain.get(), session); 1997 chain->decActiveTrackCnt(); 1998 } 1999 } 2000 2001 // detach all tracks with same session ID from this chain 2002 for (size_t i = 0; i < mTracks.size(); ++i) { 2003 sp<Track> track = mTracks[i]; 2004 if (session == track->sessionId()) { 2005 track->setMainBuffer(mMixBuffer); 2006 chain->decTrackCnt(); 2007 } 2008 } 2009 break; 2010 } 2011 } 2012 return mEffectChains.size(); 2013} 2014 2015status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2016 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2017{ 2018 Mutex::Autolock _l(mLock); 2019 return attachAuxEffect_l(track, EffectId); 2020} 2021 2022status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2023 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2024{ 2025 status_t status = NO_ERROR; 2026 2027 if (EffectId == 0) { 2028 track->setAuxBuffer(0, NULL); 2029 } else { 2030 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2031 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2032 if (effect != 0) { 2033 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2034 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2035 } else { 2036 status = INVALID_OPERATION; 2037 } 2038 } else { 2039 status = BAD_VALUE; 2040 } 2041 } 2042 return status; 2043} 2044 2045void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2046{ 2047 for (size_t i = 0; i < mTracks.size(); ++i) { 2048 sp<Track> track = mTracks[i]; 2049 if (track->auxEffectId() == effectId) { 2050 attachAuxEffect_l(track, 0); 2051 } 2052 } 2053} 2054 2055bool AudioFlinger::PlaybackThread::threadLoop() 2056{ 2057 Vector< sp<Track> > tracksToRemove; 2058 2059 standbyTime = systemTime(); 2060 2061 // MIXER 2062 nsecs_t lastWarning = 0; 2063 2064 // DUPLICATING 2065 // FIXME could this be made local to while loop? 2066 writeFrames = 0; 2067 2068 cacheParameters_l(); 2069 sleepTime = idleSleepTime; 2070 2071 if (mType == MIXER) { 2072 sleepTimeShift = 0; 2073 } 2074 2075 CpuStats cpuStats; 2076 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2077 2078 acquireWakeLock(); 2079 2080 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2081 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2082 // and then that string will be logged at the next convenient opportunity. 2083 const char *logString = NULL; 2084 2085 while (!exitPending()) 2086 { 2087 cpuStats.sample(myName); 2088 2089 Vector< sp<EffectChain> > effectChains; 2090 2091 processConfigEvents(); 2092 2093 { // scope for mLock 2094 2095 Mutex::Autolock _l(mLock); 2096 2097 if (logString != NULL) { 2098 mNBLogWriter->logTimestamp(); 2099 mNBLogWriter->log(logString); 2100 logString = NULL; 2101 } 2102 2103 if (checkForNewParameters_l()) { 2104 cacheParameters_l(); 2105 } 2106 2107 saveOutputTracks(); 2108 2109 if (mSignalPending) { 2110 // A signal was raised while we were unlocked 2111 mSignalPending = false; 2112 } else if (waitingAsyncCallback_l()) { 2113 if (exitPending()) { 2114 break; 2115 } 2116 releaseWakeLock_l(); 2117 ALOGV("wait async completion"); 2118 mWaitWorkCV.wait(mLock); 2119 ALOGV("async completion/wake"); 2120 acquireWakeLock_l(); 2121 if (exitPending()) { 2122 break; 2123 } 2124 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2125 continue; 2126 } 2127 sleepTime = 0; 2128 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2129 isSuspended()) { 2130 // put audio hardware into standby after short delay 2131 if (shouldStandby_l()) { 2132 2133 threadLoop_standby(); 2134 2135 mStandby = true; 2136 } 2137 2138 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2139 // we're about to wait, flush the binder command buffer 2140 IPCThreadState::self()->flushCommands(); 2141 2142 clearOutputTracks(); 2143 2144 if (exitPending()) { 2145 break; 2146 } 2147 2148 releaseWakeLock_l(); 2149 // wait until we have something to do... 2150 ALOGV("%s going to sleep", myName.string()); 2151 mWaitWorkCV.wait(mLock); 2152 ALOGV("%s waking up", myName.string()); 2153 acquireWakeLock_l(); 2154 2155 mMixerStatus = MIXER_IDLE; 2156 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2157 mBytesWritten = 0; 2158 mBytesRemaining = 0; 2159 checkSilentMode_l(); 2160 2161 standbyTime = systemTime() + standbyDelay; 2162 sleepTime = idleSleepTime; 2163 if (mType == MIXER) { 2164 sleepTimeShift = 0; 2165 } 2166 2167 continue; 2168 } 2169 } 2170 2171 // mMixerStatusIgnoringFastTracks is also updated internally 2172 mMixerStatus = prepareTracks_l(&tracksToRemove); 2173 2174 // prevent any changes in effect chain list and in each effect chain 2175 // during mixing and effect process as the audio buffers could be deleted 2176 // or modified if an effect is created or deleted 2177 lockEffectChains_l(effectChains); 2178 } 2179 2180 if (mBytesRemaining == 0) { 2181 mCurrentWriteLength = 0; 2182 if (mMixerStatus == MIXER_TRACKS_READY) { 2183 // threadLoop_mix() sets mCurrentWriteLength 2184 threadLoop_mix(); 2185 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2186 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2187 // threadLoop_sleepTime sets sleepTime to 0 if data 2188 // must be written to HAL 2189 threadLoop_sleepTime(); 2190 if (sleepTime == 0) { 2191 mCurrentWriteLength = mixBufferSize; 2192 } 2193 } 2194 mBytesRemaining = mCurrentWriteLength; 2195 if (isSuspended()) { 2196 sleepTime = suspendSleepTimeUs(); 2197 // simulate write to HAL when suspended 2198 mBytesWritten += mixBufferSize; 2199 mBytesRemaining = 0; 2200 } 2201 2202 // only process effects if we're going to write 2203 if (sleepTime == 0) { 2204 for (size_t i = 0; i < effectChains.size(); i ++) { 2205 effectChains[i]->process_l(); 2206 } 2207 } 2208 } 2209 2210 // enable changes in effect chain 2211 unlockEffectChains(effectChains); 2212 2213 if (!waitingAsyncCallback()) { 2214 // sleepTime == 0 means we must write to audio hardware 2215 if (sleepTime == 0) { 2216 if (mBytesRemaining) { 2217 ssize_t ret = threadLoop_write(); 2218 if (ret < 0) { 2219 mBytesRemaining = 0; 2220 } else { 2221 mBytesWritten += ret; 2222 mBytesRemaining -= ret; 2223 } 2224 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2225 (mMixerStatus == MIXER_DRAIN_ALL)) { 2226 threadLoop_drain(); 2227 } 2228if (mType == MIXER) { 2229 // write blocked detection 2230 nsecs_t now = systemTime(); 2231 nsecs_t delta = now - mLastWriteTime; 2232 if (!mStandby && delta > maxPeriod) { 2233 mNumDelayedWrites++; 2234 if ((now - lastWarning) > kWarningThrottleNs) { 2235 ATRACE_NAME("underrun"); 2236 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2237 ns2ms(delta), mNumDelayedWrites, this); 2238 lastWarning = now; 2239 } 2240 } 2241} 2242 2243 mStandby = false; 2244 } else { 2245 usleep(sleepTime); 2246 } 2247 } 2248 2249 // Finally let go of removed track(s), without the lock held 2250 // since we can't guarantee the destructors won't acquire that 2251 // same lock. This will also mutate and push a new fast mixer state. 2252 threadLoop_removeTracks(tracksToRemove); 2253 tracksToRemove.clear(); 2254 2255 // FIXME I don't understand the need for this here; 2256 // it was in the original code but maybe the 2257 // assignment in saveOutputTracks() makes this unnecessary? 2258 clearOutputTracks(); 2259 2260 // Effect chains will be actually deleted here if they were removed from 2261 // mEffectChains list during mixing or effects processing 2262 effectChains.clear(); 2263 2264 // FIXME Note that the above .clear() is no longer necessary since effectChains 2265 // is now local to this block, but will keep it for now (at least until merge done). 2266 } 2267 2268 threadLoop_exit(); 2269 2270 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2271 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2272 // put output stream into standby mode 2273 if (!mStandby) { 2274 mOutput->stream->common.standby(&mOutput->stream->common); 2275 } 2276 } 2277 2278 releaseWakeLock(); 2279 2280 ALOGV("Thread %p type %d exiting", this, mType); 2281 return false; 2282} 2283 2284// removeTracks_l() must be called with ThreadBase::mLock held 2285void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2286{ 2287 size_t count = tracksToRemove.size(); 2288 if (CC_UNLIKELY(count)) { 2289 for (size_t i=0 ; i<count ; i++) { 2290 const sp<Track>& track = tracksToRemove.itemAt(i); 2291 mActiveTracks.remove(track); 2292 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2293 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2294 if (chain != 0) { 2295 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2296 track->sessionId()); 2297 chain->decActiveTrackCnt(); 2298 } 2299 if (track->isTerminated()) { 2300 removeTrack_l(track); 2301 } 2302 } 2303 } 2304 2305} 2306 2307// ---------------------------------------------------------------------------- 2308 2309AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2310 audio_io_handle_t id, audio_devices_t device, type_t type) 2311 : PlaybackThread(audioFlinger, output, id, device, type), 2312 // mAudioMixer below 2313 // mFastMixer below 2314 mFastMixerFutex(0) 2315 // mOutputSink below 2316 // mPipeSink below 2317 // mNormalSink below 2318{ 2319 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2320 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2321 "mFrameCount=%d, mNormalFrameCount=%d", 2322 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2323 mNormalFrameCount); 2324 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2325 2326 // FIXME - Current mixer implementation only supports stereo output 2327 if (mChannelCount != FCC_2) { 2328 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2329 } 2330 2331 // create an NBAIO sink for the HAL output stream, and negotiate 2332 mOutputSink = new AudioStreamOutSink(output->stream); 2333 size_t numCounterOffers = 0; 2334 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2335 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2336 ALOG_ASSERT(index == 0); 2337 2338 // initialize fast mixer depending on configuration 2339 bool initFastMixer; 2340 switch (kUseFastMixer) { 2341 case FastMixer_Never: 2342 initFastMixer = false; 2343 break; 2344 case FastMixer_Always: 2345 initFastMixer = true; 2346 break; 2347 case FastMixer_Static: 2348 case FastMixer_Dynamic: 2349 initFastMixer = mFrameCount < mNormalFrameCount; 2350 break; 2351 } 2352 if (initFastMixer) { 2353 2354 // create a MonoPipe to connect our submix to FastMixer 2355 NBAIO_Format format = mOutputSink->format(); 2356 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2357 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2358 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2359 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2360 const NBAIO_Format offers[1] = {format}; 2361 size_t numCounterOffers = 0; 2362 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2363 ALOG_ASSERT(index == 0); 2364 monoPipe->setAvgFrames((mScreenState & 1) ? 2365 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2366 mPipeSink = monoPipe; 2367 2368#ifdef TEE_SINK 2369 if (mTeeSinkOutputEnabled) { 2370 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2371 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2372 numCounterOffers = 0; 2373 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2374 ALOG_ASSERT(index == 0); 2375 mTeeSink = teeSink; 2376 PipeReader *teeSource = new PipeReader(*teeSink); 2377 numCounterOffers = 0; 2378 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2379 ALOG_ASSERT(index == 0); 2380 mTeeSource = teeSource; 2381 } 2382#endif 2383 2384 // create fast mixer and configure it initially with just one fast track for our submix 2385 mFastMixer = new FastMixer(); 2386 FastMixerStateQueue *sq = mFastMixer->sq(); 2387#ifdef STATE_QUEUE_DUMP 2388 sq->setObserverDump(&mStateQueueObserverDump); 2389 sq->setMutatorDump(&mStateQueueMutatorDump); 2390#endif 2391 FastMixerState *state = sq->begin(); 2392 FastTrack *fastTrack = &state->mFastTracks[0]; 2393 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2394 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2395 fastTrack->mVolumeProvider = NULL; 2396 fastTrack->mGeneration++; 2397 state->mFastTracksGen++; 2398 state->mTrackMask = 1; 2399 // fast mixer will use the HAL output sink 2400 state->mOutputSink = mOutputSink.get(); 2401 state->mOutputSinkGen++; 2402 state->mFrameCount = mFrameCount; 2403 state->mCommand = FastMixerState::COLD_IDLE; 2404 // already done in constructor initialization list 2405 //mFastMixerFutex = 0; 2406 state->mColdFutexAddr = &mFastMixerFutex; 2407 state->mColdGen++; 2408 state->mDumpState = &mFastMixerDumpState; 2409#ifdef TEE_SINK 2410 state->mTeeSink = mTeeSink.get(); 2411#endif 2412 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2413 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2414 sq->end(); 2415 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2416 2417 // start the fast mixer 2418 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2419 pid_t tid = mFastMixer->getTid(); 2420 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2421 if (err != 0) { 2422 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2423 kPriorityFastMixer, getpid_cached, tid, err); 2424 } 2425 2426#ifdef AUDIO_WATCHDOG 2427 // create and start the watchdog 2428 mAudioWatchdog = new AudioWatchdog(); 2429 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2430 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2431 tid = mAudioWatchdog->getTid(); 2432 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2433 if (err != 0) { 2434 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2435 kPriorityFastMixer, getpid_cached, tid, err); 2436 } 2437#endif 2438 2439 } else { 2440 mFastMixer = NULL; 2441 } 2442 2443 switch (kUseFastMixer) { 2444 case FastMixer_Never: 2445 case FastMixer_Dynamic: 2446 mNormalSink = mOutputSink; 2447 break; 2448 case FastMixer_Always: 2449 mNormalSink = mPipeSink; 2450 break; 2451 case FastMixer_Static: 2452 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2453 break; 2454 } 2455} 2456 2457AudioFlinger::MixerThread::~MixerThread() 2458{ 2459 if (mFastMixer != NULL) { 2460 FastMixerStateQueue *sq = mFastMixer->sq(); 2461 FastMixerState *state = sq->begin(); 2462 if (state->mCommand == FastMixerState::COLD_IDLE) { 2463 int32_t old = android_atomic_inc(&mFastMixerFutex); 2464 if (old == -1) { 2465 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2466 } 2467 } 2468 state->mCommand = FastMixerState::EXIT; 2469 sq->end(); 2470 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2471 mFastMixer->join(); 2472 // Though the fast mixer thread has exited, it's state queue is still valid. 2473 // We'll use that extract the final state which contains one remaining fast track 2474 // corresponding to our sub-mix. 2475 state = sq->begin(); 2476 ALOG_ASSERT(state->mTrackMask == 1); 2477 FastTrack *fastTrack = &state->mFastTracks[0]; 2478 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2479 delete fastTrack->mBufferProvider; 2480 sq->end(false /*didModify*/); 2481 delete mFastMixer; 2482#ifdef AUDIO_WATCHDOG 2483 if (mAudioWatchdog != 0) { 2484 mAudioWatchdog->requestExit(); 2485 mAudioWatchdog->requestExitAndWait(); 2486 mAudioWatchdog.clear(); 2487 } 2488#endif 2489 } 2490 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2491 delete mAudioMixer; 2492} 2493 2494 2495uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2496{ 2497 if (mFastMixer != NULL) { 2498 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2499 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2500 } 2501 return latency; 2502} 2503 2504 2505void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2506{ 2507 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2508} 2509 2510ssize_t AudioFlinger::MixerThread::threadLoop_write() 2511{ 2512 // FIXME we should only do one push per cycle; confirm this is true 2513 // Start the fast mixer if it's not already running 2514 if (mFastMixer != NULL) { 2515 FastMixerStateQueue *sq = mFastMixer->sq(); 2516 FastMixerState *state = sq->begin(); 2517 if (state->mCommand != FastMixerState::MIX_WRITE && 2518 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2519 if (state->mCommand == FastMixerState::COLD_IDLE) { 2520 int32_t old = android_atomic_inc(&mFastMixerFutex); 2521 if (old == -1) { 2522 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2523 } 2524#ifdef AUDIO_WATCHDOG 2525 if (mAudioWatchdog != 0) { 2526 mAudioWatchdog->resume(); 2527 } 2528#endif 2529 } 2530 state->mCommand = FastMixerState::MIX_WRITE; 2531 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2532 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2533 sq->end(); 2534 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2535 if (kUseFastMixer == FastMixer_Dynamic) { 2536 mNormalSink = mPipeSink; 2537 } 2538 } else { 2539 sq->end(false /*didModify*/); 2540 } 2541 } 2542 return PlaybackThread::threadLoop_write(); 2543} 2544 2545void AudioFlinger::MixerThread::threadLoop_standby() 2546{ 2547 // Idle the fast mixer if it's currently running 2548 if (mFastMixer != NULL) { 2549 FastMixerStateQueue *sq = mFastMixer->sq(); 2550 FastMixerState *state = sq->begin(); 2551 if (!(state->mCommand & FastMixerState::IDLE)) { 2552 state->mCommand = FastMixerState::COLD_IDLE; 2553 state->mColdFutexAddr = &mFastMixerFutex; 2554 state->mColdGen++; 2555 mFastMixerFutex = 0; 2556 sq->end(); 2557 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2558 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2559 if (kUseFastMixer == FastMixer_Dynamic) { 2560 mNormalSink = mOutputSink; 2561 } 2562#ifdef AUDIO_WATCHDOG 2563 if (mAudioWatchdog != 0) { 2564 mAudioWatchdog->pause(); 2565 } 2566#endif 2567 } else { 2568 sq->end(false /*didModify*/); 2569 } 2570 } 2571 PlaybackThread::threadLoop_standby(); 2572} 2573 2574// Empty implementation for standard mixer 2575// Overridden for offloaded playback 2576void AudioFlinger::PlaybackThread::flushOutput_l() 2577{ 2578} 2579 2580bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2581{ 2582 return false; 2583} 2584 2585bool AudioFlinger::PlaybackThread::shouldStandby_l() 2586{ 2587 return !mStandby; 2588} 2589 2590bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2591{ 2592 Mutex::Autolock _l(mLock); 2593 return waitingAsyncCallback_l(); 2594} 2595 2596// shared by MIXER and DIRECT, overridden by DUPLICATING 2597void AudioFlinger::PlaybackThread::threadLoop_standby() 2598{ 2599 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2600 mOutput->stream->common.standby(&mOutput->stream->common); 2601 if (mUseAsyncWrite != 0) { 2602 mWriteBlocked = false; 2603 mDraining = false; 2604 ALOG_ASSERT(mCallbackThread != 0); 2605 mCallbackThread->setWriteBlocked(false); 2606 mCallbackThread->setDraining(false); 2607 } 2608} 2609 2610void AudioFlinger::MixerThread::threadLoop_mix() 2611{ 2612 // obtain the presentation timestamp of the next output buffer 2613 int64_t pts; 2614 status_t status = INVALID_OPERATION; 2615 2616 if (mNormalSink != 0) { 2617 status = mNormalSink->getNextWriteTimestamp(&pts); 2618 } else { 2619 status = mOutputSink->getNextWriteTimestamp(&pts); 2620 } 2621 2622 if (status != NO_ERROR) { 2623 pts = AudioBufferProvider::kInvalidPTS; 2624 } 2625 2626 // mix buffers... 2627 mAudioMixer->process(pts); 2628 mCurrentWriteLength = mixBufferSize; 2629 // increase sleep time progressively when application underrun condition clears. 2630 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2631 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2632 // such that we would underrun the audio HAL. 2633 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2634 sleepTimeShift--; 2635 } 2636 sleepTime = 0; 2637 standbyTime = systemTime() + standbyDelay; 2638 //TODO: delay standby when effects have a tail 2639} 2640 2641void AudioFlinger::MixerThread::threadLoop_sleepTime() 2642{ 2643 // If no tracks are ready, sleep once for the duration of an output 2644 // buffer size, then write 0s to the output 2645 if (sleepTime == 0) { 2646 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2647 sleepTime = activeSleepTime >> sleepTimeShift; 2648 if (sleepTime < kMinThreadSleepTimeUs) { 2649 sleepTime = kMinThreadSleepTimeUs; 2650 } 2651 // reduce sleep time in case of consecutive application underruns to avoid 2652 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2653 // duration we would end up writing less data than needed by the audio HAL if 2654 // the condition persists. 2655 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2656 sleepTimeShift++; 2657 } 2658 } else { 2659 sleepTime = idleSleepTime; 2660 } 2661 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2662 memset (mMixBuffer, 0, mixBufferSize); 2663 sleepTime = 0; 2664 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2665 "anticipated start"); 2666 } 2667 // TODO add standby time extension fct of effect tail 2668} 2669 2670// prepareTracks_l() must be called with ThreadBase::mLock held 2671AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2672 Vector< sp<Track> > *tracksToRemove) 2673{ 2674 2675 mixer_state mixerStatus = MIXER_IDLE; 2676 // find out which tracks need to be processed 2677 size_t count = mActiveTracks.size(); 2678 size_t mixedTracks = 0; 2679 size_t tracksWithEffect = 0; 2680 // counts only _active_ fast tracks 2681 size_t fastTracks = 0; 2682 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2683 2684 float masterVolume = mMasterVolume; 2685 bool masterMute = mMasterMute; 2686 2687 if (masterMute) { 2688 masterVolume = 0; 2689 } 2690 // Delegate master volume control to effect in output mix effect chain if needed 2691 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2692 if (chain != 0) { 2693 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2694 chain->setVolume_l(&v, &v); 2695 masterVolume = (float)((v + (1 << 23)) >> 24); 2696 chain.clear(); 2697 } 2698 2699 // prepare a new state to push 2700 FastMixerStateQueue *sq = NULL; 2701 FastMixerState *state = NULL; 2702 bool didModify = false; 2703 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2704 if (mFastMixer != NULL) { 2705 sq = mFastMixer->sq(); 2706 state = sq->begin(); 2707 } 2708 2709 for (size_t i=0 ; i<count ; i++) { 2710 sp<Track> t = mActiveTracks[i].promote(); 2711 if (t == 0) { 2712 continue; 2713 } 2714 2715 // this const just means the local variable doesn't change 2716 Track* const track = t.get(); 2717 2718 // process fast tracks 2719 if (track->isFastTrack()) { 2720 2721 // It's theoretically possible (though unlikely) for a fast track to be created 2722 // and then removed within the same normal mix cycle. This is not a problem, as 2723 // the track never becomes active so it's fast mixer slot is never touched. 2724 // The converse, of removing an (active) track and then creating a new track 2725 // at the identical fast mixer slot within the same normal mix cycle, 2726 // is impossible because the slot isn't marked available until the end of each cycle. 2727 int j = track->mFastIndex; 2728 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2729 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2730 FastTrack *fastTrack = &state->mFastTracks[j]; 2731 2732 // Determine whether the track is currently in underrun condition, 2733 // and whether it had a recent underrun. 2734 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2735 FastTrackUnderruns underruns = ftDump->mUnderruns; 2736 uint32_t recentFull = (underruns.mBitFields.mFull - 2737 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2738 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2739 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2740 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2741 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2742 uint32_t recentUnderruns = recentPartial + recentEmpty; 2743 track->mObservedUnderruns = underruns; 2744 // don't count underruns that occur while stopping or pausing 2745 // or stopped which can occur when flush() is called while active 2746 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2747 track->mUnderrunCount += recentUnderruns; 2748 } 2749 2750 // This is similar to the state machine for normal tracks, 2751 // with a few modifications for fast tracks. 2752 bool isActive = true; 2753 switch (track->mState) { 2754 case TrackBase::STOPPING_1: 2755 // track stays active in STOPPING_1 state until first underrun 2756 if (recentUnderruns > 0 || track->isTerminated()) { 2757 track->mState = TrackBase::STOPPING_2; 2758 } 2759 break; 2760 case TrackBase::PAUSING: 2761 // ramp down is not yet implemented 2762 track->setPaused(); 2763 break; 2764 case TrackBase::RESUMING: 2765 // ramp up is not yet implemented 2766 track->mState = TrackBase::ACTIVE; 2767 break; 2768 case TrackBase::ACTIVE: 2769 if (recentFull > 0 || recentPartial > 0) { 2770 // track has provided at least some frames recently: reset retry count 2771 track->mRetryCount = kMaxTrackRetries; 2772 } 2773 if (recentUnderruns == 0) { 2774 // no recent underruns: stay active 2775 break; 2776 } 2777 // there has recently been an underrun of some kind 2778 if (track->sharedBuffer() == 0) { 2779 // were any of the recent underruns "empty" (no frames available)? 2780 if (recentEmpty == 0) { 2781 // no, then ignore the partial underruns as they are allowed indefinitely 2782 break; 2783 } 2784 // there has recently been an "empty" underrun: decrement the retry counter 2785 if (--(track->mRetryCount) > 0) { 2786 break; 2787 } 2788 // indicate to client process that the track was disabled because of underrun; 2789 // it will then automatically call start() when data is available 2790 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2791 // remove from active list, but state remains ACTIVE [confusing but true] 2792 isActive = false; 2793 break; 2794 } 2795 // fall through 2796 case TrackBase::STOPPING_2: 2797 case TrackBase::PAUSED: 2798 case TrackBase::STOPPED: 2799 case TrackBase::FLUSHED: // flush() while active 2800 // Check for presentation complete if track is inactive 2801 // We have consumed all the buffers of this track. 2802 // This would be incomplete if we auto-paused on underrun 2803 { 2804 size_t audioHALFrames = 2805 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2806 size_t framesWritten = mBytesWritten / mFrameSize; 2807 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2808 // track stays in active list until presentation is complete 2809 break; 2810 } 2811 } 2812 if (track->isStopping_2()) { 2813 track->mState = TrackBase::STOPPED; 2814 } 2815 if (track->isStopped()) { 2816 // Can't reset directly, as fast mixer is still polling this track 2817 // track->reset(); 2818 // So instead mark this track as needing to be reset after push with ack 2819 resetMask |= 1 << i; 2820 } 2821 isActive = false; 2822 break; 2823 case TrackBase::IDLE: 2824 default: 2825 LOG_FATAL("unexpected track state %d", track->mState); 2826 } 2827 2828 if (isActive) { 2829 // was it previously inactive? 2830 if (!(state->mTrackMask & (1 << j))) { 2831 ExtendedAudioBufferProvider *eabp = track; 2832 VolumeProvider *vp = track; 2833 fastTrack->mBufferProvider = eabp; 2834 fastTrack->mVolumeProvider = vp; 2835 fastTrack->mSampleRate = track->mSampleRate; 2836 fastTrack->mChannelMask = track->mChannelMask; 2837 fastTrack->mGeneration++; 2838 state->mTrackMask |= 1 << j; 2839 didModify = true; 2840 // no acknowledgement required for newly active tracks 2841 } 2842 // cache the combined master volume and stream type volume for fast mixer; this 2843 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2844 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2845 ++fastTracks; 2846 } else { 2847 // was it previously active? 2848 if (state->mTrackMask & (1 << j)) { 2849 fastTrack->mBufferProvider = NULL; 2850 fastTrack->mGeneration++; 2851 state->mTrackMask &= ~(1 << j); 2852 didModify = true; 2853 // If any fast tracks were removed, we must wait for acknowledgement 2854 // because we're about to decrement the last sp<> on those tracks. 2855 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2856 } else { 2857 LOG_FATAL("fast track %d should have been active", j); 2858 } 2859 tracksToRemove->add(track); 2860 // Avoids a misleading display in dumpsys 2861 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2862 } 2863 continue; 2864 } 2865 2866 { // local variable scope to avoid goto warning 2867 2868 audio_track_cblk_t* cblk = track->cblk(); 2869 2870 // The first time a track is added we wait 2871 // for all its buffers to be filled before processing it 2872 int name = track->name(); 2873 // make sure that we have enough frames to mix one full buffer. 2874 // enforce this condition only once to enable draining the buffer in case the client 2875 // app does not call stop() and relies on underrun to stop: 2876 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2877 // during last round 2878 size_t desiredFrames; 2879 if (t->sampleRate() == mSampleRate) { 2880 desiredFrames = mNormalFrameCount; 2881 } else { 2882 // +1 for rounding and +1 for additional sample needed for interpolation 2883 desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2884 // add frames already consumed but not yet released by the resampler 2885 // because cblk->framesReady() will include these frames 2886 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2887 // the minimum track buffer size is normally twice the number of frames necessary 2888 // to fill one buffer and the resampler should not leave more than one buffer worth 2889 // of unreleased frames after each pass, but just in case... 2890 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2891 } 2892 uint32_t minFrames = 1; 2893 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2894 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2895 minFrames = desiredFrames; 2896 } 2897 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2898 size_t framesReady; 2899 if (track->sharedBuffer() == 0) { 2900 framesReady = track->framesReady(); 2901 } else if (track->isStopped()) { 2902 framesReady = 0; 2903 } else { 2904 framesReady = 1; 2905 } 2906 if ((framesReady >= minFrames) && track->isReady() && 2907 !track->isPaused() && !track->isTerminated()) 2908 { 2909 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->server, this); 2910 2911 mixedTracks++; 2912 2913 // track->mainBuffer() != mMixBuffer means there is an effect chain 2914 // connected to the track 2915 chain.clear(); 2916 if (track->mainBuffer() != mMixBuffer) { 2917 chain = getEffectChain_l(track->sessionId()); 2918 // Delegate volume control to effect in track effect chain if needed 2919 if (chain != 0) { 2920 tracksWithEffect++; 2921 } else { 2922 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2923 "session %d", 2924 name, track->sessionId()); 2925 } 2926 } 2927 2928 2929 int param = AudioMixer::VOLUME; 2930 if (track->mFillingUpStatus == Track::FS_FILLED) { 2931 // no ramp for the first volume setting 2932 track->mFillingUpStatus = Track::FS_ACTIVE; 2933 if (track->mState == TrackBase::RESUMING) { 2934 track->mState = TrackBase::ACTIVE; 2935 param = AudioMixer::RAMP_VOLUME; 2936 } 2937 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2938 } else if (cblk->server != 0) { 2939 // If the track is stopped before the first frame was mixed, 2940 // do not apply ramp 2941 param = AudioMixer::RAMP_VOLUME; 2942 } 2943 2944 // compute volume for this track 2945 uint32_t vl, vr, va; 2946 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2947 vl = vr = va = 0; 2948 if (track->isPausing()) { 2949 track->setPaused(); 2950 } 2951 } else { 2952 2953 // read original volumes with volume control 2954 float typeVolume = mStreamTypes[track->streamType()].volume; 2955 float v = masterVolume * typeVolume; 2956 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2957 uint32_t vlr = proxy->getVolumeLR(); 2958 vl = vlr & 0xFFFF; 2959 vr = vlr >> 16; 2960 // track volumes come from shared memory, so can't be trusted and must be clamped 2961 if (vl > MAX_GAIN_INT) { 2962 ALOGV("Track left volume out of range: %04X", vl); 2963 vl = MAX_GAIN_INT; 2964 } 2965 if (vr > MAX_GAIN_INT) { 2966 ALOGV("Track right volume out of range: %04X", vr); 2967 vr = MAX_GAIN_INT; 2968 } 2969 // now apply the master volume and stream type volume 2970 vl = (uint32_t)(v * vl) << 12; 2971 vr = (uint32_t)(v * vr) << 12; 2972 // assuming master volume and stream type volume each go up to 1.0, 2973 // vl and vr are now in 8.24 format 2974 2975 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2976 // send level comes from shared memory and so may be corrupt 2977 if (sendLevel > MAX_GAIN_INT) { 2978 ALOGV("Track send level out of range: %04X", sendLevel); 2979 sendLevel = MAX_GAIN_INT; 2980 } 2981 va = (uint32_t)(v * sendLevel); 2982 } 2983 2984 // Delegate volume control to effect in track effect chain if needed 2985 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2986 // Do not ramp volume if volume is controlled by effect 2987 param = AudioMixer::VOLUME; 2988 track->mHasVolumeController = true; 2989 } else { 2990 // force no volume ramp when volume controller was just disabled or removed 2991 // from effect chain to avoid volume spike 2992 if (track->mHasVolumeController) { 2993 param = AudioMixer::VOLUME; 2994 } 2995 track->mHasVolumeController = false; 2996 } 2997 2998 // Convert volumes from 8.24 to 4.12 format 2999 // This additional clamping is needed in case chain->setVolume_l() overshot 3000 vl = (vl + (1 << 11)) >> 12; 3001 if (vl > MAX_GAIN_INT) { 3002 vl = MAX_GAIN_INT; 3003 } 3004 vr = (vr + (1 << 11)) >> 12; 3005 if (vr > MAX_GAIN_INT) { 3006 vr = MAX_GAIN_INT; 3007 } 3008 3009 if (va > MAX_GAIN_INT) { 3010 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3011 } 3012 3013 // XXX: these things DON'T need to be done each time 3014 mAudioMixer->setBufferProvider(name, track); 3015 mAudioMixer->enable(name); 3016 3017 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3018 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3019 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3020 mAudioMixer->setParameter( 3021 name, 3022 AudioMixer::TRACK, 3023 AudioMixer::FORMAT, (void *)track->format()); 3024 mAudioMixer->setParameter( 3025 name, 3026 AudioMixer::TRACK, 3027 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3028 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3029 uint32_t maxSampleRate = mSampleRate * 2; 3030 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3031 if (reqSampleRate == 0) { 3032 reqSampleRate = mSampleRate; 3033 } else if (reqSampleRate > maxSampleRate) { 3034 reqSampleRate = maxSampleRate; 3035 } 3036 mAudioMixer->setParameter( 3037 name, 3038 AudioMixer::RESAMPLE, 3039 AudioMixer::SAMPLE_RATE, 3040 (void *)reqSampleRate); 3041 mAudioMixer->setParameter( 3042 name, 3043 AudioMixer::TRACK, 3044 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3045 mAudioMixer->setParameter( 3046 name, 3047 AudioMixer::TRACK, 3048 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3049 3050 // reset retry count 3051 track->mRetryCount = kMaxTrackRetries; 3052 3053 // If one track is ready, set the mixer ready if: 3054 // - the mixer was not ready during previous round OR 3055 // - no other track is not ready 3056 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3057 mixerStatus != MIXER_TRACKS_ENABLED) { 3058 mixerStatus = MIXER_TRACKS_READY; 3059 } 3060 } else { 3061 // only implemented for normal tracks, not fast tracks 3062 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3063 // we missed desiredFrames whatever the actual number of frames missing was 3064 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 3065 // FIXME also wake futex so that underrun is noticed more quickly 3066 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 3067 } 3068 // clear effect chain input buffer if an active track underruns to avoid sending 3069 // previous audio buffer again to effects 3070 chain = getEffectChain_l(track->sessionId()); 3071 if (chain != 0) { 3072 chain->clearInputBuffer(); 3073 } 3074 3075 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->server, this); 3076 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3077 track->isStopped() || track->isPaused()) { 3078 // We have consumed all the buffers of this track. 3079 // Remove it from the list of active tracks. 3080 // TODO: use actual buffer filling status instead of latency when available from 3081 // audio HAL 3082 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3083 size_t framesWritten = mBytesWritten / mFrameSize; 3084 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3085 if (track->isStopped()) { 3086 track->reset(); 3087 } 3088 tracksToRemove->add(track); 3089 } 3090 } else { 3091 track->mUnderrunCount++; 3092 // No buffers for this track. Give it a few chances to 3093 // fill a buffer, then remove it from active list. 3094 if (--(track->mRetryCount) <= 0) { 3095 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3096 tracksToRemove->add(track); 3097 // indicate to client process that the track was disabled because of underrun; 3098 // it will then automatically call start() when data is available 3099 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3100 // If one track is not ready, mark the mixer also not ready if: 3101 // - the mixer was ready during previous round OR 3102 // - no other track is ready 3103 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3104 mixerStatus != MIXER_TRACKS_READY) { 3105 mixerStatus = MIXER_TRACKS_ENABLED; 3106 } 3107 } 3108 mAudioMixer->disable(name); 3109 } 3110 3111 } // local variable scope to avoid goto warning 3112track_is_ready: ; 3113 3114 } 3115 3116 // Push the new FastMixer state if necessary 3117 bool pauseAudioWatchdog = false; 3118 if (didModify) { 3119 state->mFastTracksGen++; 3120 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3121 if (kUseFastMixer == FastMixer_Dynamic && 3122 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3123 state->mCommand = FastMixerState::COLD_IDLE; 3124 state->mColdFutexAddr = &mFastMixerFutex; 3125 state->mColdGen++; 3126 mFastMixerFutex = 0; 3127 if (kUseFastMixer == FastMixer_Dynamic) { 3128 mNormalSink = mOutputSink; 3129 } 3130 // If we go into cold idle, need to wait for acknowledgement 3131 // so that fast mixer stops doing I/O. 3132 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3133 pauseAudioWatchdog = true; 3134 } 3135 } 3136 if (sq != NULL) { 3137 sq->end(didModify); 3138 sq->push(block); 3139 } 3140#ifdef AUDIO_WATCHDOG 3141 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3142 mAudioWatchdog->pause(); 3143 } 3144#endif 3145 3146 // Now perform the deferred reset on fast tracks that have stopped 3147 while (resetMask != 0) { 3148 size_t i = __builtin_ctz(resetMask); 3149 ALOG_ASSERT(i < count); 3150 resetMask &= ~(1 << i); 3151 sp<Track> t = mActiveTracks[i].promote(); 3152 if (t == 0) { 3153 continue; 3154 } 3155 Track* track = t.get(); 3156 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3157 track->reset(); 3158 } 3159 3160 // remove all the tracks that need to be... 3161 removeTracks_l(*tracksToRemove); 3162 3163 // mix buffer must be cleared if all tracks are connected to an 3164 // effect chain as in this case the mixer will not write to 3165 // mix buffer and track effects will accumulate into it 3166 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3167 (mixedTracks == 0 && fastTracks > 0))) { 3168 // FIXME as a performance optimization, should remember previous zero status 3169 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3170 } 3171 3172 // if any fast tracks, then status is ready 3173 mMixerStatusIgnoringFastTracks = mixerStatus; 3174 if (fastTracks > 0) { 3175 mixerStatus = MIXER_TRACKS_READY; 3176 } 3177 return mixerStatus; 3178} 3179 3180// getTrackName_l() must be called with ThreadBase::mLock held 3181int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3182{ 3183 return mAudioMixer->getTrackName(channelMask, sessionId); 3184} 3185 3186// deleteTrackName_l() must be called with ThreadBase::mLock held 3187void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3188{ 3189 ALOGV("remove track (%d) and delete from mixer", name); 3190 mAudioMixer->deleteTrackName(name); 3191} 3192 3193// checkForNewParameters_l() must be called with ThreadBase::mLock held 3194bool AudioFlinger::MixerThread::checkForNewParameters_l() 3195{ 3196 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3197 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3198 bool reconfig = false; 3199 3200 while (!mNewParameters.isEmpty()) { 3201 3202 if (mFastMixer != NULL) { 3203 FastMixerStateQueue *sq = mFastMixer->sq(); 3204 FastMixerState *state = sq->begin(); 3205 if (!(state->mCommand & FastMixerState::IDLE)) { 3206 previousCommand = state->mCommand; 3207 state->mCommand = FastMixerState::HOT_IDLE; 3208 sq->end(); 3209 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3210 } else { 3211 sq->end(false /*didModify*/); 3212 } 3213 } 3214 3215 status_t status = NO_ERROR; 3216 String8 keyValuePair = mNewParameters[0]; 3217 AudioParameter param = AudioParameter(keyValuePair); 3218 int value; 3219 3220 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3221 reconfig = true; 3222 } 3223 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3224 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3225 status = BAD_VALUE; 3226 } else { 3227 reconfig = true; 3228 } 3229 } 3230 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3231 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3232 status = BAD_VALUE; 3233 } else { 3234 reconfig = true; 3235 } 3236 } 3237 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3238 // do not accept frame count changes if tracks are open as the track buffer 3239 // size depends on frame count and correct behavior would not be guaranteed 3240 // if frame count is changed after track creation 3241 if (!mTracks.isEmpty()) { 3242 status = INVALID_OPERATION; 3243 } else { 3244 reconfig = true; 3245 } 3246 } 3247 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3248#ifdef ADD_BATTERY_DATA 3249 // when changing the audio output device, call addBatteryData to notify 3250 // the change 3251 if (mOutDevice != value) { 3252 uint32_t params = 0; 3253 // check whether speaker is on 3254 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3255 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3256 } 3257 3258 audio_devices_t deviceWithoutSpeaker 3259 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3260 // check if any other device (except speaker) is on 3261 if (value & deviceWithoutSpeaker ) { 3262 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3263 } 3264 3265 if (params != 0) { 3266 addBatteryData(params); 3267 } 3268 } 3269#endif 3270 3271 // forward device change to effects that have requested to be 3272 // aware of attached audio device. 3273 if (value != AUDIO_DEVICE_NONE) { 3274 mOutDevice = value; 3275 for (size_t i = 0; i < mEffectChains.size(); i++) { 3276 mEffectChains[i]->setDevice_l(mOutDevice); 3277 } 3278 } 3279 } 3280 3281 if (status == NO_ERROR) { 3282 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3283 keyValuePair.string()); 3284 if (!mStandby && status == INVALID_OPERATION) { 3285 mOutput->stream->common.standby(&mOutput->stream->common); 3286 mStandby = true; 3287 mBytesWritten = 0; 3288 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3289 keyValuePair.string()); 3290 } 3291 if (status == NO_ERROR && reconfig) { 3292 readOutputParameters(); 3293 delete mAudioMixer; 3294 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3295 for (size_t i = 0; i < mTracks.size() ; i++) { 3296 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3297 if (name < 0) { 3298 break; 3299 } 3300 mTracks[i]->mName = name; 3301 } 3302 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3303 } 3304 } 3305 3306 mNewParameters.removeAt(0); 3307 3308 mParamStatus = status; 3309 mParamCond.signal(); 3310 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3311 // already timed out waiting for the status and will never signal the condition. 3312 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3313 } 3314 3315 if (!(previousCommand & FastMixerState::IDLE)) { 3316 ALOG_ASSERT(mFastMixer != NULL); 3317 FastMixerStateQueue *sq = mFastMixer->sq(); 3318 FastMixerState *state = sq->begin(); 3319 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3320 state->mCommand = previousCommand; 3321 sq->end(); 3322 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3323 } 3324 3325 return reconfig; 3326} 3327 3328 3329void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3330{ 3331 const size_t SIZE = 256; 3332 char buffer[SIZE]; 3333 String8 result; 3334 3335 PlaybackThread::dumpInternals(fd, args); 3336 3337 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3338 result.append(buffer); 3339 write(fd, result.string(), result.size()); 3340 3341 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3342 const FastMixerDumpState copy(mFastMixerDumpState); 3343 copy.dump(fd); 3344 3345#ifdef STATE_QUEUE_DUMP 3346 // Similar for state queue 3347 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3348 observerCopy.dump(fd); 3349 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3350 mutatorCopy.dump(fd); 3351#endif 3352 3353#ifdef TEE_SINK 3354 // Write the tee output to a .wav file 3355 dumpTee(fd, mTeeSource, mId); 3356#endif 3357 3358#ifdef AUDIO_WATCHDOG 3359 if (mAudioWatchdog != 0) { 3360 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3361 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3362 wdCopy.dump(fd); 3363 } 3364#endif 3365} 3366 3367uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3368{ 3369 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3370} 3371 3372uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3373{ 3374 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3375} 3376 3377void AudioFlinger::MixerThread::cacheParameters_l() 3378{ 3379 PlaybackThread::cacheParameters_l(); 3380 3381 // FIXME: Relaxed timing because of a certain device that can't meet latency 3382 // Should be reduced to 2x after the vendor fixes the driver issue 3383 // increase threshold again due to low power audio mode. The way this warning 3384 // threshold is calculated and its usefulness should be reconsidered anyway. 3385 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3386} 3387 3388// ---------------------------------------------------------------------------- 3389 3390AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3391 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3392 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3393 // mLeftVolFloat, mRightVolFloat 3394{ 3395} 3396 3397AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3398 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3399 ThreadBase::type_t type) 3400 : PlaybackThread(audioFlinger, output, id, device, type) 3401 // mLeftVolFloat, mRightVolFloat 3402{ 3403} 3404 3405AudioFlinger::DirectOutputThread::~DirectOutputThread() 3406{ 3407} 3408 3409void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3410{ 3411 audio_track_cblk_t* cblk = track->cblk(); 3412 float left, right; 3413 3414 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3415 left = right = 0; 3416 } else { 3417 float typeVolume = mStreamTypes[track->streamType()].volume; 3418 float v = mMasterVolume * typeVolume; 3419 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3420 uint32_t vlr = proxy->getVolumeLR(); 3421 float v_clamped = v * (vlr & 0xFFFF); 3422 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3423 left = v_clamped/MAX_GAIN; 3424 v_clamped = v * (vlr >> 16); 3425 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3426 right = v_clamped/MAX_GAIN; 3427 } 3428 3429 if (lastTrack) { 3430 if (left != mLeftVolFloat || right != mRightVolFloat) { 3431 mLeftVolFloat = left; 3432 mRightVolFloat = right; 3433 3434 // Convert volumes from float to 8.24 3435 uint32_t vl = (uint32_t)(left * (1 << 24)); 3436 uint32_t vr = (uint32_t)(right * (1 << 24)); 3437 3438 // Delegate volume control to effect in track effect chain if needed 3439 // only one effect chain can be present on DirectOutputThread, so if 3440 // there is one, the track is connected to it 3441 if (!mEffectChains.isEmpty()) { 3442 mEffectChains[0]->setVolume_l(&vl, &vr); 3443 left = (float)vl / (1 << 24); 3444 right = (float)vr / (1 << 24); 3445 } 3446 if (mOutput->stream->set_volume) { 3447 mOutput->stream->set_volume(mOutput->stream, left, right); 3448 } 3449 } 3450 } 3451} 3452 3453 3454AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3455 Vector< sp<Track> > *tracksToRemove 3456) 3457{ 3458 size_t count = mActiveTracks.size(); 3459 mixer_state mixerStatus = MIXER_IDLE; 3460 3461 // find out which tracks need to be processed 3462 for (size_t i = 0; i < count; i++) { 3463 sp<Track> t = mActiveTracks[i].promote(); 3464 // The track died recently 3465 if (t == 0) { 3466 continue; 3467 } 3468 3469 Track* const track = t.get(); 3470 audio_track_cblk_t* cblk = track->cblk(); 3471 3472 // The first time a track is added we wait 3473 // for all its buffers to be filled before processing it 3474 uint32_t minFrames; 3475 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3476 minFrames = mNormalFrameCount; 3477 } else { 3478 minFrames = 1; 3479 } 3480 // Only consider last track started for volume and mixer state control. 3481 // This is the last entry in mActiveTracks unless a track underruns. 3482 // As we only care about the transition phase between two tracks on a 3483 // direct output, it is not a problem to ignore the underrun case. 3484 bool last = (i == (count - 1)); 3485 3486 if ((track->framesReady() >= minFrames) && track->isReady() && 3487 !track->isPaused() && !track->isTerminated()) 3488 { 3489 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3490 3491 if (track->mFillingUpStatus == Track::FS_FILLED) { 3492 track->mFillingUpStatus = Track::FS_ACTIVE; 3493 mLeftVolFloat = mRightVolFloat = 0; 3494 if (track->mState == TrackBase::RESUMING) { 3495 track->mState = TrackBase::ACTIVE; 3496 } 3497 } 3498 3499 // compute volume for this track 3500 processVolume_l(track, last); 3501 if (last) { 3502 // reset retry count 3503 track->mRetryCount = kMaxTrackRetriesDirect; 3504 mActiveTrack = t; 3505 mixerStatus = MIXER_TRACKS_READY; 3506 } 3507 } else { 3508 // clear effect chain input buffer if the last active track started underruns 3509 // to avoid sending previous audio buffer again to effects 3510 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3511 mEffectChains[0]->clearInputBuffer(); 3512 } 3513 3514 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3515 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3516 track->isStopped() || track->isPaused()) { 3517 // We have consumed all the buffers of this track. 3518 // Remove it from the list of active tracks. 3519 // TODO: implement behavior for compressed audio 3520 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3521 size_t framesWritten = mBytesWritten / mFrameSize; 3522 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3523 if (track->isStopped()) { 3524 track->reset(); 3525 } 3526 tracksToRemove->add(track); 3527 } 3528 } else { 3529 // No buffers for this track. Give it a few chances to 3530 // fill a buffer, then remove it from active list. 3531 // Only consider last track started for mixer state control 3532 if (--(track->mRetryCount) <= 0) { 3533 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3534 tracksToRemove->add(track); 3535 } else if (last) { 3536 mixerStatus = MIXER_TRACKS_ENABLED; 3537 } 3538 } 3539 } 3540 } 3541 3542 // remove all the tracks that need to be... 3543 removeTracks_l(*tracksToRemove); 3544 3545 return mixerStatus; 3546} 3547 3548void AudioFlinger::DirectOutputThread::threadLoop_mix() 3549{ 3550 size_t frameCount = mFrameCount; 3551 int8_t *curBuf = (int8_t *)mMixBuffer; 3552 // output audio to hardware 3553 while (frameCount) { 3554 AudioBufferProvider::Buffer buffer; 3555 buffer.frameCount = frameCount; 3556 mActiveTrack->getNextBuffer(&buffer); 3557 if (CC_UNLIKELY(buffer.raw == NULL)) { 3558 memset(curBuf, 0, frameCount * mFrameSize); 3559 break; 3560 } 3561 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3562 frameCount -= buffer.frameCount; 3563 curBuf += buffer.frameCount * mFrameSize; 3564 mActiveTrack->releaseBuffer(&buffer); 3565 } 3566 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3567 sleepTime = 0; 3568 standbyTime = systemTime() + standbyDelay; 3569 mActiveTrack.clear(); 3570} 3571 3572void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3573{ 3574 if (sleepTime == 0) { 3575 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3576 sleepTime = activeSleepTime; 3577 } else { 3578 sleepTime = idleSleepTime; 3579 } 3580 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3581 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3582 sleepTime = 0; 3583 } 3584} 3585 3586// getTrackName_l() must be called with ThreadBase::mLock held 3587int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3588 int sessionId) 3589{ 3590 return 0; 3591} 3592 3593// deleteTrackName_l() must be called with ThreadBase::mLock held 3594void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3595{ 3596} 3597 3598// checkForNewParameters_l() must be called with ThreadBase::mLock held 3599bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3600{ 3601 bool reconfig = false; 3602 3603 while (!mNewParameters.isEmpty()) { 3604 status_t status = NO_ERROR; 3605 String8 keyValuePair = mNewParameters[0]; 3606 AudioParameter param = AudioParameter(keyValuePair); 3607 int value; 3608 3609 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3610 // do not accept frame count changes if tracks are open as the track buffer 3611 // size depends on frame count and correct behavior would not be garantied 3612 // if frame count is changed after track creation 3613 if (!mTracks.isEmpty()) { 3614 status = INVALID_OPERATION; 3615 } else { 3616 reconfig = true; 3617 } 3618 } 3619 if (status == NO_ERROR) { 3620 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3621 keyValuePair.string()); 3622 if (!mStandby && status == INVALID_OPERATION) { 3623 mOutput->stream->common.standby(&mOutput->stream->common); 3624 mStandby = true; 3625 mBytesWritten = 0; 3626 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3627 keyValuePair.string()); 3628 } 3629 if (status == NO_ERROR && reconfig) { 3630 readOutputParameters(); 3631 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3632 } 3633 } 3634 3635 mNewParameters.removeAt(0); 3636 3637 mParamStatus = status; 3638 mParamCond.signal(); 3639 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3640 // already timed out waiting for the status and will never signal the condition. 3641 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3642 } 3643 return reconfig; 3644} 3645 3646uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3647{ 3648 uint32_t time; 3649 if (audio_is_linear_pcm(mFormat)) { 3650 time = PlaybackThread::activeSleepTimeUs(); 3651 } else { 3652 time = 10000; 3653 } 3654 return time; 3655} 3656 3657uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3658{ 3659 uint32_t time; 3660 if (audio_is_linear_pcm(mFormat)) { 3661 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3662 } else { 3663 time = 10000; 3664 } 3665 return time; 3666} 3667 3668uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3669{ 3670 uint32_t time; 3671 if (audio_is_linear_pcm(mFormat)) { 3672 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3673 } else { 3674 time = 10000; 3675 } 3676 return time; 3677} 3678 3679void AudioFlinger::DirectOutputThread::cacheParameters_l() 3680{ 3681 PlaybackThread::cacheParameters_l(); 3682 3683 // use shorter standby delay as on normal output to release 3684 // hardware resources as soon as possible 3685 standbyDelay = microseconds(activeSleepTime*2); 3686} 3687 3688// ---------------------------------------------------------------------------- 3689 3690AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3691 const sp<AudioFlinger::OffloadThread>& offloadThread) 3692 : Thread(false /*canCallJava*/), 3693 mOffloadThread(offloadThread), 3694 mWriteBlocked(false), 3695 mDraining(false) 3696{ 3697} 3698 3699AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3700{ 3701} 3702 3703void AudioFlinger::AsyncCallbackThread::onFirstRef() 3704{ 3705 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3706} 3707 3708bool AudioFlinger::AsyncCallbackThread::threadLoop() 3709{ 3710 while (!exitPending()) { 3711 bool writeBlocked; 3712 bool draining; 3713 3714 { 3715 Mutex::Autolock _l(mLock); 3716 mWaitWorkCV.wait(mLock); 3717 if (exitPending()) { 3718 break; 3719 } 3720 writeBlocked = mWriteBlocked; 3721 draining = mDraining; 3722 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3723 } 3724 { 3725 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3726 if (offloadThread != 0) { 3727 if (writeBlocked == false) { 3728 offloadThread->setWriteBlocked(false); 3729 } 3730 if (draining == false) { 3731 offloadThread->setDraining(false); 3732 } 3733 } 3734 } 3735 } 3736 return false; 3737} 3738 3739void AudioFlinger::AsyncCallbackThread::exit() 3740{ 3741 ALOGV("AsyncCallbackThread::exit"); 3742 Mutex::Autolock _l(mLock); 3743 requestExit(); 3744 mWaitWorkCV.broadcast(); 3745} 3746 3747void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3748{ 3749 Mutex::Autolock _l(mLock); 3750 mWriteBlocked = value; 3751 if (!value) { 3752 mWaitWorkCV.signal(); 3753 } 3754} 3755 3756void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3757{ 3758 Mutex::Autolock _l(mLock); 3759 mDraining = value; 3760 if (!value) { 3761 mWaitWorkCV.signal(); 3762 } 3763} 3764 3765 3766// ---------------------------------------------------------------------------- 3767AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3768 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3769 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3770 mHwPaused(false), 3771 mPausedBytesRemaining(0) 3772{ 3773 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3774} 3775 3776AudioFlinger::OffloadThread::~OffloadThread() 3777{ 3778 mPreviousTrack.clear(); 3779} 3780 3781void AudioFlinger::OffloadThread::threadLoop_exit() 3782{ 3783 if (mFlushPending || mHwPaused) { 3784 // If a flush is pending or track was paused, just discard buffered data 3785 flushHw_l(); 3786 } else { 3787 mMixerStatus = MIXER_DRAIN_ALL; 3788 threadLoop_drain(); 3789 } 3790 mCallbackThread->exit(); 3791 PlaybackThread::threadLoop_exit(); 3792} 3793 3794AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3795 Vector< sp<Track> > *tracksToRemove 3796) 3797{ 3798 ALOGV("OffloadThread::prepareTracks_l"); 3799 size_t count = mActiveTracks.size(); 3800 3801 mixer_state mixerStatus = MIXER_IDLE; 3802 if (mFlushPending) { 3803 flushHw_l(); 3804 mFlushPending = false; 3805 } 3806 // find out which tracks need to be processed 3807 for (size_t i = 0; i < count; i++) { 3808 sp<Track> t = mActiveTracks[i].promote(); 3809 // The track died recently 3810 if (t == 0) { 3811 continue; 3812 } 3813 Track* const track = t.get(); 3814 audio_track_cblk_t* cblk = track->cblk(); 3815 if (mPreviousTrack != NULL) { 3816 if (t != mPreviousTrack) { 3817 // Flush any data still being written from last track 3818 mBytesRemaining = 0; 3819 if (mPausedBytesRemaining) { 3820 // Last track was paused so we also need to flush saved 3821 // mixbuffer state and invalidate track so that it will 3822 // re-submit that unwritten data when it is next resumed 3823 mPausedBytesRemaining = 0; 3824 // Invalidate is a bit drastic - would be more efficient 3825 // to have a flag to tell client that some of the 3826 // previously written data was lost 3827 mPreviousTrack->invalidate(); 3828 } 3829 } 3830 } 3831 mPreviousTrack = t; 3832 bool last = (i == (count - 1)); 3833 if (track->isPausing()) { 3834 track->setPaused(); 3835 if (last) { 3836 if (!mHwPaused) { 3837 mOutput->stream->pause(mOutput->stream); 3838 mHwPaused = true; 3839 } 3840 // If we were part way through writing the mixbuffer to 3841 // the HAL we must save this until we resume 3842 // BUG - this will be wrong if a different track is made active, 3843 // in that case we want to discard the pending data in the 3844 // mixbuffer and tell the client to present it again when the 3845 // track is resumed 3846 mPausedWriteLength = mCurrentWriteLength; 3847 mPausedBytesRemaining = mBytesRemaining; 3848 mBytesRemaining = 0; // stop writing 3849 } 3850 tracksToRemove->add(track); 3851 } else if (track->framesReady() && track->isReady() && 3852 !track->isPaused() && !track->isTerminated()) { 3853 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->server); 3854 if (track->mFillingUpStatus == Track::FS_FILLED) { 3855 track->mFillingUpStatus = Track::FS_ACTIVE; 3856 mLeftVolFloat = mRightVolFloat = 0; 3857 if (track->mState == TrackBase::RESUMING) { 3858 if (CC_UNLIKELY(mPausedBytesRemaining)) { 3859 // Need to continue write that was interrupted 3860 mCurrentWriteLength = mPausedWriteLength; 3861 mBytesRemaining = mPausedBytesRemaining; 3862 mPausedBytesRemaining = 0; 3863 } 3864 track->mState = TrackBase::ACTIVE; 3865 } 3866 } 3867 3868 if (last) { 3869 if (mHwPaused) { 3870 mOutput->stream->resume(mOutput->stream); 3871 mHwPaused = false; 3872 // threadLoop_mix() will handle the case that we need to 3873 // resume an interrupted write 3874 } 3875 // reset retry count 3876 track->mRetryCount = kMaxTrackRetriesOffload; 3877 mActiveTrack = t; 3878 mixerStatus = MIXER_TRACKS_READY; 3879 } 3880 } else { 3881 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->server); 3882 if (track->isStopping_1()) { 3883 // Hardware buffer can hold a large amount of audio so we must 3884 // wait for all current track's data to drain before we say 3885 // that the track is stopped. 3886 if (mBytesRemaining == 0) { 3887 // Only start draining when all data in mixbuffer 3888 // has been written 3889 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3890 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3891 sleepTime = 0; 3892 standbyTime = systemTime() + standbyDelay; 3893 if (last) { 3894 mixerStatus = MIXER_DRAIN_TRACK; 3895 if (mHwPaused) { 3896 // It is possible to move from PAUSED to STOPPING_1 without 3897 // a resume so we must ensure hardware is running 3898 mOutput->stream->resume(mOutput->stream); 3899 mHwPaused = false; 3900 } 3901 } 3902 } 3903 } else if (track->isStopping_2()) { 3904 // Drain has completed, signal presentation complete 3905 if (!mDraining || !last) { 3906 track->mState = TrackBase::STOPPED; 3907 size_t audioHALFrames = 3908 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3909 size_t framesWritten = 3910 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3911 track->presentationComplete(framesWritten, audioHALFrames); 3912 track->reset(); 3913 tracksToRemove->add(track); 3914 } 3915 } else { 3916 // No buffers for this track. Give it a few chances to 3917 // fill a buffer, then remove it from active list. 3918 if (--(track->mRetryCount) <= 0) { 3919 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3920 track->name()); 3921 tracksToRemove->add(track); 3922 } else if (last){ 3923 mixerStatus = MIXER_TRACKS_ENABLED; 3924 } 3925 } 3926 } 3927 // compute volume for this track 3928 processVolume_l(track, last); 3929 } 3930 // remove all the tracks that need to be... 3931 removeTracks_l(*tracksToRemove); 3932 3933 return mixerStatus; 3934} 3935 3936void AudioFlinger::OffloadThread::flushOutput_l() 3937{ 3938 mFlushPending = true; 3939} 3940 3941// must be called with thread mutex locked 3942bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3943{ 3944 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3945 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3946 return true; 3947 } 3948 return false; 3949} 3950 3951// must be called with thread mutex locked 3952bool AudioFlinger::OffloadThread::shouldStandby_l() 3953{ 3954 bool TrackPaused = false; 3955 3956 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3957 // after a timeout and we will enter standby then. 3958 if (mTracks.size() > 0) { 3959 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3960 } 3961 3962 return !mStandby && !TrackPaused; 3963} 3964 3965 3966bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3967{ 3968 Mutex::Autolock _l(mLock); 3969 return waitingAsyncCallback_l(); 3970} 3971 3972void AudioFlinger::OffloadThread::flushHw_l() 3973{ 3974 mOutput->stream->flush(mOutput->stream); 3975 // Flush anything still waiting in the mixbuffer 3976 mCurrentWriteLength = 0; 3977 mBytesRemaining = 0; 3978 mPausedWriteLength = 0; 3979 mPausedBytesRemaining = 0; 3980 if (mUseAsyncWrite) { 3981 mWriteBlocked = false; 3982 mDraining = false; 3983 ALOG_ASSERT(mCallbackThread != 0); 3984 mCallbackThread->setWriteBlocked(false); 3985 mCallbackThread->setDraining(false); 3986 } 3987} 3988 3989// ---------------------------------------------------------------------------- 3990 3991AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3992 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3993 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3994 DUPLICATING), 3995 mWaitTimeMs(UINT_MAX) 3996{ 3997 addOutputTrack(mainThread); 3998} 3999 4000AudioFlinger::DuplicatingThread::~DuplicatingThread() 4001{ 4002 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4003 mOutputTracks[i]->destroy(); 4004 } 4005} 4006 4007void AudioFlinger::DuplicatingThread::threadLoop_mix() 4008{ 4009 // mix buffers... 4010 if (outputsReady(outputTracks)) { 4011 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4012 } else { 4013 memset(mMixBuffer, 0, mixBufferSize); 4014 } 4015 sleepTime = 0; 4016 writeFrames = mNormalFrameCount; 4017 mCurrentWriteLength = mixBufferSize; 4018 standbyTime = systemTime() + standbyDelay; 4019} 4020 4021void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4022{ 4023 if (sleepTime == 0) { 4024 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4025 sleepTime = activeSleepTime; 4026 } else { 4027 sleepTime = idleSleepTime; 4028 } 4029 } else if (mBytesWritten != 0) { 4030 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4031 writeFrames = mNormalFrameCount; 4032 memset(mMixBuffer, 0, mixBufferSize); 4033 } else { 4034 // flush remaining overflow buffers in output tracks 4035 writeFrames = 0; 4036 } 4037 sleepTime = 0; 4038 } 4039} 4040 4041ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4042{ 4043 for (size_t i = 0; i < outputTracks.size(); i++) { 4044 outputTracks[i]->write(mMixBuffer, writeFrames); 4045 } 4046 return (ssize_t)mixBufferSize; 4047} 4048 4049void AudioFlinger::DuplicatingThread::threadLoop_standby() 4050{ 4051 // DuplicatingThread implements standby by stopping all tracks 4052 for (size_t i = 0; i < outputTracks.size(); i++) { 4053 outputTracks[i]->stop(); 4054 } 4055} 4056 4057void AudioFlinger::DuplicatingThread::saveOutputTracks() 4058{ 4059 outputTracks = mOutputTracks; 4060} 4061 4062void AudioFlinger::DuplicatingThread::clearOutputTracks() 4063{ 4064 outputTracks.clear(); 4065} 4066 4067void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4068{ 4069 Mutex::Autolock _l(mLock); 4070 // FIXME explain this formula 4071 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4072 OutputTrack *outputTrack = new OutputTrack(thread, 4073 this, 4074 mSampleRate, 4075 mFormat, 4076 mChannelMask, 4077 frameCount); 4078 if (outputTrack->cblk() != NULL) { 4079 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4080 mOutputTracks.add(outputTrack); 4081 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4082 updateWaitTime_l(); 4083 } 4084} 4085 4086void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4087{ 4088 Mutex::Autolock _l(mLock); 4089 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4090 if (mOutputTracks[i]->thread() == thread) { 4091 mOutputTracks[i]->destroy(); 4092 mOutputTracks.removeAt(i); 4093 updateWaitTime_l(); 4094 return; 4095 } 4096 } 4097 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4098} 4099 4100// caller must hold mLock 4101void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4102{ 4103 mWaitTimeMs = UINT_MAX; 4104 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4105 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4106 if (strong != 0) { 4107 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4108 if (waitTimeMs < mWaitTimeMs) { 4109 mWaitTimeMs = waitTimeMs; 4110 } 4111 } 4112 } 4113} 4114 4115 4116bool AudioFlinger::DuplicatingThread::outputsReady( 4117 const SortedVector< sp<OutputTrack> > &outputTracks) 4118{ 4119 for (size_t i = 0; i < outputTracks.size(); i++) { 4120 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4121 if (thread == 0) { 4122 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4123 outputTracks[i].get()); 4124 return false; 4125 } 4126 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4127 // see note at standby() declaration 4128 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4129 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4130 thread.get()); 4131 return false; 4132 } 4133 } 4134 return true; 4135} 4136 4137uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4138{ 4139 return (mWaitTimeMs * 1000) / 2; 4140} 4141 4142void AudioFlinger::DuplicatingThread::cacheParameters_l() 4143{ 4144 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4145 updateWaitTime_l(); 4146 4147 MixerThread::cacheParameters_l(); 4148} 4149 4150// ---------------------------------------------------------------------------- 4151// Record 4152// ---------------------------------------------------------------------------- 4153 4154AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4155 AudioStreamIn *input, 4156 uint32_t sampleRate, 4157 audio_channel_mask_t channelMask, 4158 audio_io_handle_t id, 4159 audio_devices_t outDevice, 4160 audio_devices_t inDevice 4161#ifdef TEE_SINK 4162 , const sp<NBAIO_Sink>& teeSink 4163#endif 4164 ) : 4165 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4166 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4167 // mRsmpInIndex and mInputBytes set by readInputParameters() 4168 mReqChannelCount(popcount(channelMask)), 4169 mReqSampleRate(sampleRate) 4170 // mBytesRead is only meaningful while active, and so is cleared in start() 4171 // (but might be better to also clear here for dump?) 4172#ifdef TEE_SINK 4173 , mTeeSink(teeSink) 4174#endif 4175{ 4176 snprintf(mName, kNameLength, "AudioIn_%X", id); 4177 4178 readInputParameters(); 4179 4180} 4181 4182 4183AudioFlinger::RecordThread::~RecordThread() 4184{ 4185 delete[] mRsmpInBuffer; 4186 delete mResampler; 4187 delete[] mRsmpOutBuffer; 4188} 4189 4190void AudioFlinger::RecordThread::onFirstRef() 4191{ 4192 run(mName, PRIORITY_URGENT_AUDIO); 4193} 4194 4195status_t AudioFlinger::RecordThread::readyToRun() 4196{ 4197 status_t status = initCheck(); 4198 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4199 return status; 4200} 4201 4202bool AudioFlinger::RecordThread::threadLoop() 4203{ 4204 AudioBufferProvider::Buffer buffer; 4205 sp<RecordTrack> activeTrack; 4206 Vector< sp<EffectChain> > effectChains; 4207 4208 nsecs_t lastWarning = 0; 4209 4210 inputStandBy(); 4211 acquireWakeLock(); 4212 4213 // used to verify we've read at least once before evaluating how many bytes were read 4214 bool readOnce = false; 4215 4216 // start recording 4217 while (!exitPending()) { 4218 4219 processConfigEvents(); 4220 4221 { // scope for mLock 4222 Mutex::Autolock _l(mLock); 4223 checkForNewParameters_l(); 4224 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4225 standby(); 4226 4227 if (exitPending()) { 4228 break; 4229 } 4230 4231 releaseWakeLock_l(); 4232 ALOGV("RecordThread: loop stopping"); 4233 // go to sleep 4234 mWaitWorkCV.wait(mLock); 4235 ALOGV("RecordThread: loop starting"); 4236 acquireWakeLock_l(); 4237 continue; 4238 } 4239 if (mActiveTrack != 0) { 4240 if (mActiveTrack->isTerminated()) { 4241 removeTrack_l(mActiveTrack); 4242 mActiveTrack.clear(); 4243 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4244 standby(); 4245 mActiveTrack.clear(); 4246 mStartStopCond.broadcast(); 4247 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4248 if (mReqChannelCount != mActiveTrack->channelCount()) { 4249 mActiveTrack.clear(); 4250 mStartStopCond.broadcast(); 4251 } else if (readOnce) { 4252 // record start succeeds only if first read from audio input 4253 // succeeds 4254 if (mBytesRead >= 0) { 4255 mActiveTrack->mState = TrackBase::ACTIVE; 4256 } else { 4257 mActiveTrack.clear(); 4258 } 4259 mStartStopCond.broadcast(); 4260 } 4261 mStandby = false; 4262 } 4263 } 4264 lockEffectChains_l(effectChains); 4265 } 4266 4267 if (mActiveTrack != 0) { 4268 if (mActiveTrack->mState != TrackBase::ACTIVE && 4269 mActiveTrack->mState != TrackBase::RESUMING) { 4270 unlockEffectChains(effectChains); 4271 usleep(kRecordThreadSleepUs); 4272 continue; 4273 } 4274 for (size_t i = 0; i < effectChains.size(); i ++) { 4275 effectChains[i]->process_l(); 4276 } 4277 4278 buffer.frameCount = mFrameCount; 4279 status_t status = mActiveTrack->getNextBuffer(&buffer); 4280 if (CC_LIKELY(status == NO_ERROR)) { 4281 readOnce = true; 4282 size_t framesOut = buffer.frameCount; 4283 if (mResampler == NULL) { 4284 // no resampling 4285 while (framesOut) { 4286 size_t framesIn = mFrameCount - mRsmpInIndex; 4287 if (framesIn) { 4288 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4289 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4290 mActiveTrack->mFrameSize; 4291 if (framesIn > framesOut) 4292 framesIn = framesOut; 4293 mRsmpInIndex += framesIn; 4294 framesOut -= framesIn; 4295 if (mChannelCount == mReqChannelCount || 4296 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4297 memcpy(dst, src, framesIn * mFrameSize); 4298 } else { 4299 if (mChannelCount == 1) { 4300 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4301 (int16_t *)src, framesIn); 4302 } else { 4303 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4304 (int16_t *)src, framesIn); 4305 } 4306 } 4307 } 4308 if (framesOut && mFrameCount == mRsmpInIndex) { 4309 void *readInto; 4310 if (framesOut == mFrameCount && 4311 (mChannelCount == mReqChannelCount || 4312 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4313 readInto = buffer.raw; 4314 framesOut = 0; 4315 } else { 4316 readInto = mRsmpInBuffer; 4317 mRsmpInIndex = 0; 4318 } 4319 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4320 mInputBytes); 4321 if (mBytesRead <= 0) { 4322 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4323 { 4324 ALOGE("Error reading audio input"); 4325 // Force input into standby so that it tries to 4326 // recover at next read attempt 4327 inputStandBy(); 4328 usleep(kRecordThreadSleepUs); 4329 } 4330 mRsmpInIndex = mFrameCount; 4331 framesOut = 0; 4332 buffer.frameCount = 0; 4333 } 4334#ifdef TEE_SINK 4335 else if (mTeeSink != 0) { 4336 (void) mTeeSink->write(readInto, 4337 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4338 } 4339#endif 4340 } 4341 } 4342 } else { 4343 // resampling 4344 4345 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4346 // alter output frame count as if we were expecting stereo samples 4347 if (mChannelCount == 1 && mReqChannelCount == 1) { 4348 framesOut >>= 1; 4349 } 4350 mResampler->resample(mRsmpOutBuffer, framesOut, 4351 this /* AudioBufferProvider* */); 4352 // ditherAndClamp() works as long as all buffers returned by 4353 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4354 if (mChannelCount == 2 && mReqChannelCount == 1) { 4355 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4356 // the resampler always outputs stereo samples: 4357 // do post stereo to mono conversion 4358 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4359 framesOut); 4360 } else { 4361 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4362 } 4363 4364 } 4365 if (mFramestoDrop == 0) { 4366 mActiveTrack->releaseBuffer(&buffer); 4367 } else { 4368 if (mFramestoDrop > 0) { 4369 mFramestoDrop -= buffer.frameCount; 4370 if (mFramestoDrop <= 0) { 4371 clearSyncStartEvent(); 4372 } 4373 } else { 4374 mFramestoDrop += buffer.frameCount; 4375 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4376 mSyncStartEvent->isCancelled()) { 4377 ALOGW("Synced record %s, session %d, trigger session %d", 4378 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4379 mActiveTrack->sessionId(), 4380 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4381 clearSyncStartEvent(); 4382 } 4383 } 4384 } 4385 mActiveTrack->clearOverflow(); 4386 } 4387 // client isn't retrieving buffers fast enough 4388 else { 4389 if (!mActiveTrack->setOverflow()) { 4390 nsecs_t now = systemTime(); 4391 if ((now - lastWarning) > kWarningThrottleNs) { 4392 ALOGW("RecordThread: buffer overflow"); 4393 lastWarning = now; 4394 } 4395 } 4396 // Release the processor for a while before asking for a new buffer. 4397 // This will give the application more chance to read from the buffer and 4398 // clear the overflow. 4399 usleep(kRecordThreadSleepUs); 4400 } 4401 } 4402 // enable changes in effect chain 4403 unlockEffectChains(effectChains); 4404 effectChains.clear(); 4405 } 4406 4407 standby(); 4408 4409 { 4410 Mutex::Autolock _l(mLock); 4411 mActiveTrack.clear(); 4412 mStartStopCond.broadcast(); 4413 } 4414 4415 releaseWakeLock(); 4416 4417 ALOGV("RecordThread %p exiting", this); 4418 return false; 4419} 4420 4421void AudioFlinger::RecordThread::standby() 4422{ 4423 if (!mStandby) { 4424 inputStandBy(); 4425 mStandby = true; 4426 } 4427} 4428 4429void AudioFlinger::RecordThread::inputStandBy() 4430{ 4431 mInput->stream->common.standby(&mInput->stream->common); 4432} 4433 4434sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4435 const sp<AudioFlinger::Client>& client, 4436 uint32_t sampleRate, 4437 audio_format_t format, 4438 audio_channel_mask_t channelMask, 4439 size_t frameCount, 4440 int sessionId, 4441 IAudioFlinger::track_flags_t flags, 4442 pid_t tid, 4443 status_t *status) 4444{ 4445 sp<RecordTrack> track; 4446 status_t lStatus; 4447 4448 lStatus = initCheck(); 4449 if (lStatus != NO_ERROR) { 4450 ALOGE("Audio driver not initialized."); 4451 goto Exit; 4452 } 4453 4454 // FIXME use flags and tid similar to createTrack_l() 4455 4456 { // scope for mLock 4457 Mutex::Autolock _l(mLock); 4458 4459 track = new RecordTrack(this, client, sampleRate, 4460 format, channelMask, frameCount, sessionId); 4461 4462 if (track->getCblk() == 0) { 4463 lStatus = NO_MEMORY; 4464 goto Exit; 4465 } 4466 mTracks.add(track); 4467 4468 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4469 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4470 mAudioFlinger->btNrecIsOff(); 4471 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4472 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4473 } 4474 lStatus = NO_ERROR; 4475 4476Exit: 4477 if (status) { 4478 *status = lStatus; 4479 } 4480 return track; 4481} 4482 4483status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4484 AudioSystem::sync_event_t event, 4485 int triggerSession) 4486{ 4487 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4488 sp<ThreadBase> strongMe = this; 4489 status_t status = NO_ERROR; 4490 4491 if (event == AudioSystem::SYNC_EVENT_NONE) { 4492 clearSyncStartEvent(); 4493 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4494 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4495 triggerSession, 4496 recordTrack->sessionId(), 4497 syncStartEventCallback, 4498 this); 4499 // Sync event can be cancelled by the trigger session if the track is not in a 4500 // compatible state in which case we start record immediately 4501 if (mSyncStartEvent->isCancelled()) { 4502 clearSyncStartEvent(); 4503 } else { 4504 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4505 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4506 } 4507 } 4508 4509 { 4510 AutoMutex lock(mLock); 4511 if (mActiveTrack != 0) { 4512 if (recordTrack != mActiveTrack.get()) { 4513 status = -EBUSY; 4514 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4515 mActiveTrack->mState = TrackBase::ACTIVE; 4516 } 4517 return status; 4518 } 4519 4520 recordTrack->mState = TrackBase::IDLE; 4521 mActiveTrack = recordTrack; 4522 mLock.unlock(); 4523 status_t status = AudioSystem::startInput(mId); 4524 mLock.lock(); 4525 if (status != NO_ERROR) { 4526 mActiveTrack.clear(); 4527 clearSyncStartEvent(); 4528 return status; 4529 } 4530 mRsmpInIndex = mFrameCount; 4531 mBytesRead = 0; 4532 if (mResampler != NULL) { 4533 mResampler->reset(); 4534 } 4535 mActiveTrack->mState = TrackBase::RESUMING; 4536 // signal thread to start 4537 ALOGV("Signal record thread"); 4538 mWaitWorkCV.broadcast(); 4539 // do not wait for mStartStopCond if exiting 4540 if (exitPending()) { 4541 mActiveTrack.clear(); 4542 status = INVALID_OPERATION; 4543 goto startError; 4544 } 4545 mStartStopCond.wait(mLock); 4546 if (mActiveTrack == 0) { 4547 ALOGV("Record failed to start"); 4548 status = BAD_VALUE; 4549 goto startError; 4550 } 4551 ALOGV("Record started OK"); 4552 return status; 4553 } 4554 4555startError: 4556 AudioSystem::stopInput(mId); 4557 clearSyncStartEvent(); 4558 return status; 4559} 4560 4561void AudioFlinger::RecordThread::clearSyncStartEvent() 4562{ 4563 if (mSyncStartEvent != 0) { 4564 mSyncStartEvent->cancel(); 4565 } 4566 mSyncStartEvent.clear(); 4567 mFramestoDrop = 0; 4568} 4569 4570void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4571{ 4572 sp<SyncEvent> strongEvent = event.promote(); 4573 4574 if (strongEvent != 0) { 4575 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4576 me->handleSyncStartEvent(strongEvent); 4577 } 4578} 4579 4580void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4581{ 4582 if (event == mSyncStartEvent) { 4583 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4584 // from audio HAL 4585 mFramestoDrop = mFrameCount * 2; 4586 } 4587} 4588 4589bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4590 ALOGV("RecordThread::stop"); 4591 AutoMutex _l(mLock); 4592 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4593 return false; 4594 } 4595 recordTrack->mState = TrackBase::PAUSING; 4596 // do not wait for mStartStopCond if exiting 4597 if (exitPending()) { 4598 return true; 4599 } 4600 mStartStopCond.wait(mLock); 4601 // if we have been restarted, recordTrack == mActiveTrack.get() here 4602 if (exitPending() || recordTrack != mActiveTrack.get()) { 4603 ALOGV("Record stopped OK"); 4604 return true; 4605 } 4606 return false; 4607} 4608 4609bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4610{ 4611 return false; 4612} 4613 4614status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4615{ 4616#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4617 if (!isValidSyncEvent(event)) { 4618 return BAD_VALUE; 4619 } 4620 4621 int eventSession = event->triggerSession(); 4622 status_t ret = NAME_NOT_FOUND; 4623 4624 Mutex::Autolock _l(mLock); 4625 4626 for (size_t i = 0; i < mTracks.size(); i++) { 4627 sp<RecordTrack> track = mTracks[i]; 4628 if (eventSession == track->sessionId()) { 4629 (void) track->setSyncEvent(event); 4630 ret = NO_ERROR; 4631 } 4632 } 4633 return ret; 4634#else 4635 return BAD_VALUE; 4636#endif 4637} 4638 4639// destroyTrack_l() must be called with ThreadBase::mLock held 4640void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4641{ 4642 track->terminate(); 4643 track->mState = TrackBase::STOPPED; 4644 // active tracks are removed by threadLoop() 4645 if (mActiveTrack != track) { 4646 removeTrack_l(track); 4647 } 4648} 4649 4650void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4651{ 4652 mTracks.remove(track); 4653 // need anything related to effects here? 4654} 4655 4656void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4657{ 4658 dumpInternals(fd, args); 4659 dumpTracks(fd, args); 4660 dumpEffectChains(fd, args); 4661} 4662 4663void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4664{ 4665 const size_t SIZE = 256; 4666 char buffer[SIZE]; 4667 String8 result; 4668 4669 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4670 result.append(buffer); 4671 4672 if (mActiveTrack != 0) { 4673 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4674 result.append(buffer); 4675 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4676 result.append(buffer); 4677 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4678 result.append(buffer); 4679 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4680 result.append(buffer); 4681 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4682 result.append(buffer); 4683 } else { 4684 result.append("No active record client\n"); 4685 } 4686 4687 write(fd, result.string(), result.size()); 4688 4689 dumpBase(fd, args); 4690} 4691 4692void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4693{ 4694 const size_t SIZE = 256; 4695 char buffer[SIZE]; 4696 String8 result; 4697 4698 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4699 result.append(buffer); 4700 RecordTrack::appendDumpHeader(result); 4701 for (size_t i = 0; i < mTracks.size(); ++i) { 4702 sp<RecordTrack> track = mTracks[i]; 4703 if (track != 0) { 4704 track->dump(buffer, SIZE); 4705 result.append(buffer); 4706 } 4707 } 4708 4709 if (mActiveTrack != 0) { 4710 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4711 result.append(buffer); 4712 RecordTrack::appendDumpHeader(result); 4713 mActiveTrack->dump(buffer, SIZE); 4714 result.append(buffer); 4715 4716 } 4717 write(fd, result.string(), result.size()); 4718} 4719 4720// AudioBufferProvider interface 4721status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4722{ 4723 size_t framesReq = buffer->frameCount; 4724 size_t framesReady = mFrameCount - mRsmpInIndex; 4725 int channelCount; 4726 4727 if (framesReady == 0) { 4728 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4729 if (mBytesRead <= 0) { 4730 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4731 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4732 // Force input into standby so that it tries to 4733 // recover at next read attempt 4734 inputStandBy(); 4735 usleep(kRecordThreadSleepUs); 4736 } 4737 buffer->raw = NULL; 4738 buffer->frameCount = 0; 4739 return NOT_ENOUGH_DATA; 4740 } 4741 mRsmpInIndex = 0; 4742 framesReady = mFrameCount; 4743 } 4744 4745 if (framesReq > framesReady) { 4746 framesReq = framesReady; 4747 } 4748 4749 if (mChannelCount == 1 && mReqChannelCount == 2) { 4750 channelCount = 1; 4751 } else { 4752 channelCount = 2; 4753 } 4754 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4755 buffer->frameCount = framesReq; 4756 return NO_ERROR; 4757} 4758 4759// AudioBufferProvider interface 4760void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4761{ 4762 mRsmpInIndex += buffer->frameCount; 4763 buffer->frameCount = 0; 4764} 4765 4766bool AudioFlinger::RecordThread::checkForNewParameters_l() 4767{ 4768 bool reconfig = false; 4769 4770 while (!mNewParameters.isEmpty()) { 4771 status_t status = NO_ERROR; 4772 String8 keyValuePair = mNewParameters[0]; 4773 AudioParameter param = AudioParameter(keyValuePair); 4774 int value; 4775 audio_format_t reqFormat = mFormat; 4776 uint32_t reqSamplingRate = mReqSampleRate; 4777 uint32_t reqChannelCount = mReqChannelCount; 4778 4779 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4780 reqSamplingRate = value; 4781 reconfig = true; 4782 } 4783 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4784 reqFormat = (audio_format_t) value; 4785 reconfig = true; 4786 } 4787 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4788 reqChannelCount = popcount(value); 4789 reconfig = true; 4790 } 4791 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4792 // do not accept frame count changes if tracks are open as the track buffer 4793 // size depends on frame count and correct behavior would not be guaranteed 4794 // if frame count is changed after track creation 4795 if (mActiveTrack != 0) { 4796 status = INVALID_OPERATION; 4797 } else { 4798 reconfig = true; 4799 } 4800 } 4801 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4802 // forward device change to effects that have requested to be 4803 // aware of attached audio device. 4804 for (size_t i = 0; i < mEffectChains.size(); i++) { 4805 mEffectChains[i]->setDevice_l(value); 4806 } 4807 4808 // store input device and output device but do not forward output device to audio HAL. 4809 // Note that status is ignored by the caller for output device 4810 // (see AudioFlinger::setParameters() 4811 if (audio_is_output_devices(value)) { 4812 mOutDevice = value; 4813 status = BAD_VALUE; 4814 } else { 4815 mInDevice = value; 4816 // disable AEC and NS if the device is a BT SCO headset supporting those 4817 // pre processings 4818 if (mTracks.size() > 0) { 4819 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4820 mAudioFlinger->btNrecIsOff(); 4821 for (size_t i = 0; i < mTracks.size(); i++) { 4822 sp<RecordTrack> track = mTracks[i]; 4823 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4824 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4825 } 4826 } 4827 } 4828 } 4829 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4830 mAudioSource != (audio_source_t)value) { 4831 // forward device change to effects that have requested to be 4832 // aware of attached audio device. 4833 for (size_t i = 0; i < mEffectChains.size(); i++) { 4834 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4835 } 4836 mAudioSource = (audio_source_t)value; 4837 } 4838 if (status == NO_ERROR) { 4839 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4840 keyValuePair.string()); 4841 if (status == INVALID_OPERATION) { 4842 inputStandBy(); 4843 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4844 keyValuePair.string()); 4845 } 4846 if (reconfig) { 4847 if (status == BAD_VALUE && 4848 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4849 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4850 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4851 <= (2 * reqSamplingRate)) && 4852 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4853 <= FCC_2 && 4854 (reqChannelCount <= FCC_2)) { 4855 status = NO_ERROR; 4856 } 4857 if (status == NO_ERROR) { 4858 readInputParameters(); 4859 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4860 } 4861 } 4862 } 4863 4864 mNewParameters.removeAt(0); 4865 4866 mParamStatus = status; 4867 mParamCond.signal(); 4868 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4869 // already timed out waiting for the status and will never signal the condition. 4870 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4871 } 4872 return reconfig; 4873} 4874 4875String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4876{ 4877 char *s; 4878 String8 out_s8 = String8(); 4879 4880 Mutex::Autolock _l(mLock); 4881 if (initCheck() != NO_ERROR) { 4882 return out_s8; 4883 } 4884 4885 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4886 out_s8 = String8(s); 4887 free(s); 4888 return out_s8; 4889} 4890 4891void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4892 AudioSystem::OutputDescriptor desc; 4893 void *param2 = NULL; 4894 4895 switch (event) { 4896 case AudioSystem::INPUT_OPENED: 4897 case AudioSystem::INPUT_CONFIG_CHANGED: 4898 desc.channels = mChannelMask; 4899 desc.samplingRate = mSampleRate; 4900 desc.format = mFormat; 4901 desc.frameCount = mFrameCount; 4902 desc.latency = 0; 4903 param2 = &desc; 4904 break; 4905 4906 case AudioSystem::INPUT_CLOSED: 4907 default: 4908 break; 4909 } 4910 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4911} 4912 4913void AudioFlinger::RecordThread::readInputParameters() 4914{ 4915 delete mRsmpInBuffer; 4916 // mRsmpInBuffer is always assigned a new[] below 4917 delete mRsmpOutBuffer; 4918 mRsmpOutBuffer = NULL; 4919 delete mResampler; 4920 mResampler = NULL; 4921 4922 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4923 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4924 mChannelCount = popcount(mChannelMask); 4925 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4926 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4927 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4928 mFrameCount = mInputBytes / mFrameSize; 4929 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4930 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4931 4932 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4933 { 4934 int channelCount; 4935 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4936 // stereo to mono post process as the resampler always outputs stereo. 4937 if (mChannelCount == 1 && mReqChannelCount == 2) { 4938 channelCount = 1; 4939 } else { 4940 channelCount = 2; 4941 } 4942 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4943 mResampler->setSampleRate(mSampleRate); 4944 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4945 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4946 4947 // optmization: if mono to mono, alter input frame count as if we were inputing 4948 // stereo samples 4949 if (mChannelCount == 1 && mReqChannelCount == 1) { 4950 mFrameCount >>= 1; 4951 } 4952 4953 } 4954 mRsmpInIndex = mFrameCount; 4955} 4956 4957unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4958{ 4959 Mutex::Autolock _l(mLock); 4960 if (initCheck() != NO_ERROR) { 4961 return 0; 4962 } 4963 4964 return mInput->stream->get_input_frames_lost(mInput->stream); 4965} 4966 4967uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4968{ 4969 Mutex::Autolock _l(mLock); 4970 uint32_t result = 0; 4971 if (getEffectChain_l(sessionId) != 0) { 4972 result = EFFECT_SESSION; 4973 } 4974 4975 for (size_t i = 0; i < mTracks.size(); ++i) { 4976 if (sessionId == mTracks[i]->sessionId()) { 4977 result |= TRACK_SESSION; 4978 break; 4979 } 4980 } 4981 4982 return result; 4983} 4984 4985KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4986{ 4987 KeyedVector<int, bool> ids; 4988 Mutex::Autolock _l(mLock); 4989 for (size_t j = 0; j < mTracks.size(); ++j) { 4990 sp<RecordThread::RecordTrack> track = mTracks[j]; 4991 int sessionId = track->sessionId(); 4992 if (ids.indexOfKey(sessionId) < 0) { 4993 ids.add(sessionId, true); 4994 } 4995 } 4996 return ids; 4997} 4998 4999AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5000{ 5001 Mutex::Autolock _l(mLock); 5002 AudioStreamIn *input = mInput; 5003 mInput = NULL; 5004 return input; 5005} 5006 5007// this method must always be called either with ThreadBase mLock held or inside the thread loop 5008audio_stream_t* AudioFlinger::RecordThread::stream() const 5009{ 5010 if (mInput == NULL) { 5011 return NULL; 5012 } 5013 return &mInput->stream->common; 5014} 5015 5016status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5017{ 5018 // only one chain per input thread 5019 if (mEffectChains.size() != 0) { 5020 return INVALID_OPERATION; 5021 } 5022 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5023 5024 chain->setInBuffer(NULL); 5025 chain->setOutBuffer(NULL); 5026 5027 checkSuspendOnAddEffectChain_l(chain); 5028 5029 mEffectChains.add(chain); 5030 5031 return NO_ERROR; 5032} 5033 5034size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5035{ 5036 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5037 ALOGW_IF(mEffectChains.size() != 1, 5038 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5039 chain.get(), mEffectChains.size(), this); 5040 if (mEffectChains.size() == 1) { 5041 mEffectChains.removeAt(0); 5042 } 5043 return 0; 5044} 5045 5046}; // namespace android 5047