Threads.cpp revision 5c68f959eaa2e02fed5643c78e281fff42bcc0a2
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38#include <audio_utils/minifloat.h> 39 40// NBAIO implementations 41#include <media/nbaio/AudioStreamInSource.h> 42#include <media/nbaio/AudioStreamOutSink.h> 43#include <media/nbaio/MonoPipe.h> 44#include <media/nbaio/MonoPipeReader.h> 45#include <media/nbaio/Pipe.h> 46#include <media/nbaio/PipeReader.h> 47#include <media/nbaio/SourceAudioBufferProvider.h> 48 49#include <powermanager/PowerManager.h> 50 51#include <common_time/cc_helper.h> 52#include <common_time/local_clock.h> 53 54#include "AudioFlinger.h" 55#include "AudioMixer.h" 56#include "FastMixer.h" 57#include "FastCapture.h" 58#include "ServiceUtilities.h" 59#include "SchedulingPolicyService.h" 60 61#ifdef ADD_BATTERY_DATA 62#include <media/IMediaPlayerService.h> 63#include <media/IMediaDeathNotifier.h> 64#endif 65 66#ifdef DEBUG_CPU_USAGE 67#include <cpustats/CentralTendencyStatistics.h> 68#include <cpustats/ThreadCpuUsage.h> 69#endif 70 71// ---------------------------------------------------------------------------- 72 73// Note: the following macro is used for extremely verbose logging message. In 74// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 75// 0; but one side effect of this is to turn all LOGV's as well. Some messages 76// are so verbose that we want to suppress them even when we have ALOG_ASSERT 77// turned on. Do not uncomment the #def below unless you really know what you 78// are doing and want to see all of the extremely verbose messages. 79//#define VERY_VERY_VERBOSE_LOGGING 80#ifdef VERY_VERY_VERBOSE_LOGGING 81#define ALOGVV ALOGV 82#else 83#define ALOGVV(a...) do { } while(0) 84#endif 85 86namespace android { 87 88// retry counts for buffer fill timeout 89// 50 * ~20msecs = 1 second 90static const int8_t kMaxTrackRetries = 50; 91static const int8_t kMaxTrackStartupRetries = 50; 92// allow less retry attempts on direct output thread. 93// direct outputs can be a scarce resource in audio hardware and should 94// be released as quickly as possible. 95static const int8_t kMaxTrackRetriesDirect = 2; 96 97// don't warn about blocked writes or record buffer overflows more often than this 98static const nsecs_t kWarningThrottleNs = seconds(5); 99 100// RecordThread loop sleep time upon application overrun or audio HAL read error 101static const int kRecordThreadSleepUs = 5000; 102 103// maximum time to wait in sendConfigEvent_l() for a status to be received 104static const nsecs_t kConfigEventTimeoutNs = seconds(2); 105 106// minimum sleep time for the mixer thread loop when tracks are active but in underrun 107static const uint32_t kMinThreadSleepTimeUs = 5000; 108// maximum divider applied to the active sleep time in the mixer thread loop 109static const uint32_t kMaxThreadSleepTimeShift = 2; 110 111// minimum normal sink buffer size, expressed in milliseconds rather than frames 112static const uint32_t kMinNormalSinkBufferSizeMs = 20; 113// maximum normal sink buffer size 114static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 115 116// Offloaded output thread standby delay: allows track transition without going to standby 117static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 118 119// Whether to use fast mixer 120static const enum { 121 FastMixer_Never, // never initialize or use: for debugging only 122 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 123 // normal mixer multiplier is 1 124 FastMixer_Static, // initialize if needed, then use all the time if initialized, 125 // multiplier is calculated based on min & max normal mixer buffer size 126 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 127 // multiplier is calculated based on min & max normal mixer buffer size 128 // FIXME for FastMixer_Dynamic: 129 // Supporting this option will require fixing HALs that can't handle large writes. 130 // For example, one HAL implementation returns an error from a large write, 131 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 132 // We could either fix the HAL implementations, or provide a wrapper that breaks 133 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 134} kUseFastMixer = FastMixer_Static; 135 136// Whether to use fast capture 137static const enum { 138 FastCapture_Never, // never initialize or use: for debugging only 139 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 140 FastCapture_Static, // initialize if needed, then use all the time if initialized 141} kUseFastCapture = FastCapture_Static; 142 143// Priorities for requestPriority 144static const int kPriorityAudioApp = 2; 145static const int kPriorityFastMixer = 3; 146static const int kPriorityFastCapture = 3; 147 148// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 149// for the track. The client then sub-divides this into smaller buffers for its use. 150// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 151// So for now we just assume that client is double-buffered for fast tracks. 152// FIXME It would be better for client to tell AudioFlinger the value of N, 153// so AudioFlinger could allocate the right amount of memory. 154// See the client's minBufCount and mNotificationFramesAct calculations for details. 155 156// This is the default value, if not specified by property. 157static const int kFastTrackMultiplier = 2; 158 159// The minimum and maximum allowed values 160static const int kFastTrackMultiplierMin = 1; 161static const int kFastTrackMultiplierMax = 2; 162 163// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 164static int sFastTrackMultiplier = kFastTrackMultiplier; 165 166// See Thread::readOnlyHeap(). 167// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 168// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 169// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 170static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 171 172// ---------------------------------------------------------------------------- 173 174static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 175 176static void sFastTrackMultiplierInit() 177{ 178 char value[PROPERTY_VALUE_MAX]; 179 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 180 char *endptr; 181 unsigned long ul = strtoul(value, &endptr, 0); 182 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 183 sFastTrackMultiplier = (int) ul; 184 } 185 } 186} 187 188// ---------------------------------------------------------------------------- 189 190#ifdef ADD_BATTERY_DATA 191// To collect the amplifier usage 192static void addBatteryData(uint32_t params) { 193 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 194 if (service == NULL) { 195 // it already logged 196 return; 197 } 198 199 service->addBatteryData(params); 200} 201#endif 202 203 204// ---------------------------------------------------------------------------- 205// CPU Stats 206// ---------------------------------------------------------------------------- 207 208class CpuStats { 209public: 210 CpuStats(); 211 void sample(const String8 &title); 212#ifdef DEBUG_CPU_USAGE 213private: 214 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 215 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 216 217 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 218 219 int mCpuNum; // thread's current CPU number 220 int mCpukHz; // frequency of thread's current CPU in kHz 221#endif 222}; 223 224CpuStats::CpuStats() 225#ifdef DEBUG_CPU_USAGE 226 : mCpuNum(-1), mCpukHz(-1) 227#endif 228{ 229} 230 231void CpuStats::sample(const String8 &title 232#ifndef DEBUG_CPU_USAGE 233 __unused 234#endif 235 ) { 236#ifdef DEBUG_CPU_USAGE 237 // get current thread's delta CPU time in wall clock ns 238 double wcNs; 239 bool valid = mCpuUsage.sampleAndEnable(wcNs); 240 241 // record sample for wall clock statistics 242 if (valid) { 243 mWcStats.sample(wcNs); 244 } 245 246 // get the current CPU number 247 int cpuNum = sched_getcpu(); 248 249 // get the current CPU frequency in kHz 250 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 251 252 // check if either CPU number or frequency changed 253 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 254 mCpuNum = cpuNum; 255 mCpukHz = cpukHz; 256 // ignore sample for purposes of cycles 257 valid = false; 258 } 259 260 // if no change in CPU number or frequency, then record sample for cycle statistics 261 if (valid && mCpukHz > 0) { 262 double cycles = wcNs * cpukHz * 0.000001; 263 mHzStats.sample(cycles); 264 } 265 266 unsigned n = mWcStats.n(); 267 // mCpuUsage.elapsed() is expensive, so don't call it every loop 268 if ((n & 127) == 1) { 269 long long elapsed = mCpuUsage.elapsed(); 270 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 271 double perLoop = elapsed / (double) n; 272 double perLoop100 = perLoop * 0.01; 273 double perLoop1k = perLoop * 0.001; 274 double mean = mWcStats.mean(); 275 double stddev = mWcStats.stddev(); 276 double minimum = mWcStats.minimum(); 277 double maximum = mWcStats.maximum(); 278 double meanCycles = mHzStats.mean(); 279 double stddevCycles = mHzStats.stddev(); 280 double minCycles = mHzStats.minimum(); 281 double maxCycles = mHzStats.maximum(); 282 mCpuUsage.resetElapsed(); 283 mWcStats.reset(); 284 mHzStats.reset(); 285 ALOGD("CPU usage for %s over past %.1f secs\n" 286 " (%u mixer loops at %.1f mean ms per loop):\n" 287 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 288 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 289 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 290 title.string(), 291 elapsed * .000000001, n, perLoop * .000001, 292 mean * .001, 293 stddev * .001, 294 minimum * .001, 295 maximum * .001, 296 mean / perLoop100, 297 stddev / perLoop100, 298 minimum / perLoop100, 299 maximum / perLoop100, 300 meanCycles / perLoop1k, 301 stddevCycles / perLoop1k, 302 minCycles / perLoop1k, 303 maxCycles / perLoop1k); 304 305 } 306 } 307#endif 308}; 309 310// ---------------------------------------------------------------------------- 311// ThreadBase 312// ---------------------------------------------------------------------------- 313 314AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 315 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 316 : Thread(false /*canCallJava*/), 317 mType(type), 318 mAudioFlinger(audioFlinger), 319 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 320 // are set by PlaybackThread::readOutputParameters_l() or 321 // RecordThread::readInputParameters_l() 322 //FIXME: mStandby should be true here. Is this some kind of hack? 323 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 324 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 325 // mName will be set by concrete (non-virtual) subclass 326 mDeathRecipient(new PMDeathRecipient(this)) 327{ 328} 329 330AudioFlinger::ThreadBase::~ThreadBase() 331{ 332 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 333 mConfigEvents.clear(); 334 335 // do not lock the mutex in destructor 336 releaseWakeLock_l(); 337 if (mPowerManager != 0) { 338 sp<IBinder> binder = mPowerManager->asBinder(); 339 binder->unlinkToDeath(mDeathRecipient); 340 } 341} 342 343status_t AudioFlinger::ThreadBase::readyToRun() 344{ 345 status_t status = initCheck(); 346 if (status == NO_ERROR) { 347 ALOGI("AudioFlinger's thread %p ready to run", this); 348 } else { 349 ALOGE("No working audio driver found."); 350 } 351 return status; 352} 353 354void AudioFlinger::ThreadBase::exit() 355{ 356 ALOGV("ThreadBase::exit"); 357 // do any cleanup required for exit to succeed 358 preExit(); 359 { 360 // This lock prevents the following race in thread (uniprocessor for illustration): 361 // if (!exitPending()) { 362 // // context switch from here to exit() 363 // // exit() calls requestExit(), what exitPending() observes 364 // // exit() calls signal(), which is dropped since no waiters 365 // // context switch back from exit() to here 366 // mWaitWorkCV.wait(...); 367 // // now thread is hung 368 // } 369 AutoMutex lock(mLock); 370 requestExit(); 371 mWaitWorkCV.broadcast(); 372 } 373 // When Thread::requestExitAndWait is made virtual and this method is renamed to 374 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 375 requestExitAndWait(); 376} 377 378status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 379{ 380 status_t status; 381 382 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 383 Mutex::Autolock _l(mLock); 384 385 return sendSetParameterConfigEvent_l(keyValuePairs); 386} 387 388// sendConfigEvent_l() must be called with ThreadBase::mLock held 389// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 390status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 391{ 392 status_t status = NO_ERROR; 393 394 mConfigEvents.add(event); 395 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 396 mWaitWorkCV.signal(); 397 mLock.unlock(); 398 { 399 Mutex::Autolock _l(event->mLock); 400 while (event->mWaitStatus) { 401 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 402 event->mStatus = TIMED_OUT; 403 event->mWaitStatus = false; 404 } 405 } 406 status = event->mStatus; 407 } 408 mLock.lock(); 409 return status; 410} 411 412void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 413{ 414 Mutex::Autolock _l(mLock); 415 sendIoConfigEvent_l(event, param); 416} 417 418// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 419void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 420{ 421 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 422 sendConfigEvent_l(configEvent); 423} 424 425// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 426void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 427{ 428 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 429 sendConfigEvent_l(configEvent); 430} 431 432// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 433status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 434{ 435 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 436 return sendConfigEvent_l(configEvent); 437} 438 439status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 440 const struct audio_patch *patch, 441 audio_patch_handle_t *handle) 442{ 443 Mutex::Autolock _l(mLock); 444 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 445 status_t status = sendConfigEvent_l(configEvent); 446 if (status == NO_ERROR) { 447 CreateAudioPatchConfigEventData *data = 448 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 449 *handle = data->mHandle; 450 } 451 return status; 452} 453 454status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 455 const audio_patch_handle_t handle) 456{ 457 Mutex::Autolock _l(mLock); 458 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 459 return sendConfigEvent_l(configEvent); 460} 461 462 463// post condition: mConfigEvents.isEmpty() 464void AudioFlinger::ThreadBase::processConfigEvents_l() 465{ 466 bool configChanged = false; 467 468 while (!mConfigEvents.isEmpty()) { 469 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 470 sp<ConfigEvent> event = mConfigEvents[0]; 471 mConfigEvents.removeAt(0); 472 switch (event->mType) { 473 case CFG_EVENT_PRIO: { 474 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 475 // FIXME Need to understand why this has to be done asynchronously 476 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 477 true /*asynchronous*/); 478 if (err != 0) { 479 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 480 data->mPrio, data->mPid, data->mTid, err); 481 } 482 } break; 483 case CFG_EVENT_IO: { 484 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 485 audioConfigChanged(data->mEvent, data->mParam); 486 } break; 487 case CFG_EVENT_SET_PARAMETER: { 488 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 489 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 490 configChanged = true; 491 } 492 } break; 493 case CFG_EVENT_CREATE_AUDIO_PATCH: { 494 CreateAudioPatchConfigEventData *data = 495 (CreateAudioPatchConfigEventData *)event->mData.get(); 496 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 497 } break; 498 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 499 ReleaseAudioPatchConfigEventData *data = 500 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 501 event->mStatus = releaseAudioPatch_l(data->mHandle); 502 } break; 503 default: 504 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 505 break; 506 } 507 { 508 Mutex::Autolock _l(event->mLock); 509 if (event->mWaitStatus) { 510 event->mWaitStatus = false; 511 event->mCond.signal(); 512 } 513 } 514 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 515 } 516 517 if (configChanged) { 518 cacheParameters_l(); 519 } 520} 521 522String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 523 String8 s; 524 if (output) { 525 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 526 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 527 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 528 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 529 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 530 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 531 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 532 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 533 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 534 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 535 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 536 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 537 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 538 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 539 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 540 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 541 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 542 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 543 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 544 } else { 545 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 546 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 547 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 548 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 549 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 550 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 551 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 552 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 553 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 554 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 555 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 556 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 557 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 558 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 559 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 560 } 561 int len = s.length(); 562 if (s.length() > 2) { 563 char *str = s.lockBuffer(len); 564 s.unlockBuffer(len - 2); 565 } 566 return s; 567} 568 569void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 570{ 571 const size_t SIZE = 256; 572 char buffer[SIZE]; 573 String8 result; 574 575 bool locked = AudioFlinger::dumpTryLock(mLock); 576 if (!locked) { 577 dprintf(fd, "thread %p maybe dead locked\n", this); 578 } 579 580 dprintf(fd, " I/O handle: %d\n", mId); 581 dprintf(fd, " TID: %d\n", getTid()); 582 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 583 dprintf(fd, " Sample rate: %u\n", mSampleRate); 584 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 585 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 586 dprintf(fd, " Channel Count: %u\n", mChannelCount); 587 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 588 channelMaskToString(mChannelMask, mType != RECORD).string()); 589 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 590 dprintf(fd, " Frame size: %zu\n", mFrameSize); 591 dprintf(fd, " Pending config events:"); 592 size_t numConfig = mConfigEvents.size(); 593 if (numConfig) { 594 for (size_t i = 0; i < numConfig; i++) { 595 mConfigEvents[i]->dump(buffer, SIZE); 596 dprintf(fd, "\n %s", buffer); 597 } 598 dprintf(fd, "\n"); 599 } else { 600 dprintf(fd, " none\n"); 601 } 602 603 if (locked) { 604 mLock.unlock(); 605 } 606} 607 608void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 609{ 610 const size_t SIZE = 256; 611 char buffer[SIZE]; 612 String8 result; 613 614 size_t numEffectChains = mEffectChains.size(); 615 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 616 write(fd, buffer, strlen(buffer)); 617 618 for (size_t i = 0; i < numEffectChains; ++i) { 619 sp<EffectChain> chain = mEffectChains[i]; 620 if (chain != 0) { 621 chain->dump(fd, args); 622 } 623 } 624} 625 626void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 627{ 628 Mutex::Autolock _l(mLock); 629 acquireWakeLock_l(uid); 630} 631 632String16 AudioFlinger::ThreadBase::getWakeLockTag() 633{ 634 switch (mType) { 635 case MIXER: 636 return String16("AudioMix"); 637 case DIRECT: 638 return String16("AudioDirectOut"); 639 case DUPLICATING: 640 return String16("AudioDup"); 641 case RECORD: 642 return String16("AudioIn"); 643 case OFFLOAD: 644 return String16("AudioOffload"); 645 default: 646 ALOG_ASSERT(false); 647 return String16("AudioUnknown"); 648 } 649} 650 651void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 652{ 653 getPowerManager_l(); 654 if (mPowerManager != 0) { 655 sp<IBinder> binder = new BBinder(); 656 status_t status; 657 if (uid >= 0) { 658 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 659 binder, 660 getWakeLockTag(), 661 String16("media"), 662 uid); 663 } else { 664 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 665 binder, 666 getWakeLockTag(), 667 String16("media")); 668 } 669 if (status == NO_ERROR) { 670 mWakeLockToken = binder; 671 } 672 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 673 } 674} 675 676void AudioFlinger::ThreadBase::releaseWakeLock() 677{ 678 Mutex::Autolock _l(mLock); 679 releaseWakeLock_l(); 680} 681 682void AudioFlinger::ThreadBase::releaseWakeLock_l() 683{ 684 if (mWakeLockToken != 0) { 685 ALOGV("releaseWakeLock_l() %s", mName); 686 if (mPowerManager != 0) { 687 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 688 } 689 mWakeLockToken.clear(); 690 } 691} 692 693void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 694 Mutex::Autolock _l(mLock); 695 updateWakeLockUids_l(uids); 696} 697 698void AudioFlinger::ThreadBase::getPowerManager_l() { 699 700 if (mPowerManager == 0) { 701 // use checkService() to avoid blocking if power service is not up yet 702 sp<IBinder> binder = 703 defaultServiceManager()->checkService(String16("power")); 704 if (binder == 0) { 705 ALOGW("Thread %s cannot connect to the power manager service", mName); 706 } else { 707 mPowerManager = interface_cast<IPowerManager>(binder); 708 binder->linkToDeath(mDeathRecipient); 709 } 710 } 711} 712 713void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 714 715 getPowerManager_l(); 716 if (mWakeLockToken == NULL) { 717 ALOGE("no wake lock to update!"); 718 return; 719 } 720 if (mPowerManager != 0) { 721 sp<IBinder> binder = new BBinder(); 722 status_t status; 723 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 724 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 725 } 726} 727 728void AudioFlinger::ThreadBase::clearPowerManager() 729{ 730 Mutex::Autolock _l(mLock); 731 releaseWakeLock_l(); 732 mPowerManager.clear(); 733} 734 735void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 736{ 737 sp<ThreadBase> thread = mThread.promote(); 738 if (thread != 0) { 739 thread->clearPowerManager(); 740 } 741 ALOGW("power manager service died !!!"); 742} 743 744void AudioFlinger::ThreadBase::setEffectSuspended( 745 const effect_uuid_t *type, bool suspend, int sessionId) 746{ 747 Mutex::Autolock _l(mLock); 748 setEffectSuspended_l(type, suspend, sessionId); 749} 750 751void AudioFlinger::ThreadBase::setEffectSuspended_l( 752 const effect_uuid_t *type, bool suspend, int sessionId) 753{ 754 sp<EffectChain> chain = getEffectChain_l(sessionId); 755 if (chain != 0) { 756 if (type != NULL) { 757 chain->setEffectSuspended_l(type, suspend); 758 } else { 759 chain->setEffectSuspendedAll_l(suspend); 760 } 761 } 762 763 updateSuspendedSessions_l(type, suspend, sessionId); 764} 765 766void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 767{ 768 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 769 if (index < 0) { 770 return; 771 } 772 773 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 774 mSuspendedSessions.valueAt(index); 775 776 for (size_t i = 0; i < sessionEffects.size(); i++) { 777 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 778 for (int j = 0; j < desc->mRefCount; j++) { 779 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 780 chain->setEffectSuspendedAll_l(true); 781 } else { 782 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 783 desc->mType.timeLow); 784 chain->setEffectSuspended_l(&desc->mType, true); 785 } 786 } 787 } 788} 789 790void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 791 bool suspend, 792 int sessionId) 793{ 794 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 795 796 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 797 798 if (suspend) { 799 if (index >= 0) { 800 sessionEffects = mSuspendedSessions.valueAt(index); 801 } else { 802 mSuspendedSessions.add(sessionId, sessionEffects); 803 } 804 } else { 805 if (index < 0) { 806 return; 807 } 808 sessionEffects = mSuspendedSessions.valueAt(index); 809 } 810 811 812 int key = EffectChain::kKeyForSuspendAll; 813 if (type != NULL) { 814 key = type->timeLow; 815 } 816 index = sessionEffects.indexOfKey(key); 817 818 sp<SuspendedSessionDesc> desc; 819 if (suspend) { 820 if (index >= 0) { 821 desc = sessionEffects.valueAt(index); 822 } else { 823 desc = new SuspendedSessionDesc(); 824 if (type != NULL) { 825 desc->mType = *type; 826 } 827 sessionEffects.add(key, desc); 828 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 829 } 830 desc->mRefCount++; 831 } else { 832 if (index < 0) { 833 return; 834 } 835 desc = sessionEffects.valueAt(index); 836 if (--desc->mRefCount == 0) { 837 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 838 sessionEffects.removeItemsAt(index); 839 if (sessionEffects.isEmpty()) { 840 ALOGV("updateSuspendedSessions_l() restore removing session %d", 841 sessionId); 842 mSuspendedSessions.removeItem(sessionId); 843 } 844 } 845 } 846 if (!sessionEffects.isEmpty()) { 847 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 848 } 849} 850 851void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 852 bool enabled, 853 int sessionId) 854{ 855 Mutex::Autolock _l(mLock); 856 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 857} 858 859void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 860 bool enabled, 861 int sessionId) 862{ 863 if (mType != RECORD) { 864 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 865 // another session. This gives the priority to well behaved effect control panels 866 // and applications not using global effects. 867 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 868 // global effects 869 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 870 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 871 } 872 } 873 874 sp<EffectChain> chain = getEffectChain_l(sessionId); 875 if (chain != 0) { 876 chain->checkSuspendOnEffectEnabled(effect, enabled); 877 } 878} 879 880// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 881sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 882 const sp<AudioFlinger::Client>& client, 883 const sp<IEffectClient>& effectClient, 884 int32_t priority, 885 int sessionId, 886 effect_descriptor_t *desc, 887 int *enabled, 888 status_t *status) 889{ 890 sp<EffectModule> effect; 891 sp<EffectHandle> handle; 892 status_t lStatus; 893 sp<EffectChain> chain; 894 bool chainCreated = false; 895 bool effectCreated = false; 896 bool effectRegistered = false; 897 898 lStatus = initCheck(); 899 if (lStatus != NO_ERROR) { 900 ALOGW("createEffect_l() Audio driver not initialized."); 901 goto Exit; 902 } 903 904 // Reject any effect on Direct output threads for now, since the format of 905 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 906 if (mType == DIRECT) { 907 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 908 desc->name, mName); 909 lStatus = BAD_VALUE; 910 goto Exit; 911 } 912 913 // Allow global effects only on offloaded and mixer threads 914 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 915 switch (mType) { 916 case MIXER: 917 case OFFLOAD: 918 break; 919 case DIRECT: 920 case DUPLICATING: 921 case RECORD: 922 default: 923 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 924 lStatus = BAD_VALUE; 925 goto Exit; 926 } 927 } 928 929 // Only Pre processor effects are allowed on input threads and only on input threads 930 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 931 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 932 desc->name, desc->flags, mType); 933 lStatus = BAD_VALUE; 934 goto Exit; 935 } 936 937 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 938 939 { // scope for mLock 940 Mutex::Autolock _l(mLock); 941 942 // check for existing effect chain with the requested audio session 943 chain = getEffectChain_l(sessionId); 944 if (chain == 0) { 945 // create a new chain for this session 946 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 947 chain = new EffectChain(this, sessionId); 948 addEffectChain_l(chain); 949 chain->setStrategy(getStrategyForSession_l(sessionId)); 950 chainCreated = true; 951 } else { 952 effect = chain->getEffectFromDesc_l(desc); 953 } 954 955 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 956 957 if (effect == 0) { 958 int id = mAudioFlinger->nextUniqueId(); 959 // Check CPU and memory usage 960 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 961 if (lStatus != NO_ERROR) { 962 goto Exit; 963 } 964 effectRegistered = true; 965 // create a new effect module if none present in the chain 966 effect = new EffectModule(this, chain, desc, id, sessionId); 967 lStatus = effect->status(); 968 if (lStatus != NO_ERROR) { 969 goto Exit; 970 } 971 effect->setOffloaded(mType == OFFLOAD, mId); 972 973 lStatus = chain->addEffect_l(effect); 974 if (lStatus != NO_ERROR) { 975 goto Exit; 976 } 977 effectCreated = true; 978 979 effect->setDevice(mOutDevice); 980 effect->setDevice(mInDevice); 981 effect->setMode(mAudioFlinger->getMode()); 982 effect->setAudioSource(mAudioSource); 983 } 984 // create effect handle and connect it to effect module 985 handle = new EffectHandle(effect, client, effectClient, priority); 986 lStatus = handle->initCheck(); 987 if (lStatus == OK) { 988 lStatus = effect->addHandle(handle.get()); 989 } 990 if (enabled != NULL) { 991 *enabled = (int)effect->isEnabled(); 992 } 993 } 994 995Exit: 996 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 997 Mutex::Autolock _l(mLock); 998 if (effectCreated) { 999 chain->removeEffect_l(effect); 1000 } 1001 if (effectRegistered) { 1002 AudioSystem::unregisterEffect(effect->id()); 1003 } 1004 if (chainCreated) { 1005 removeEffectChain_l(chain); 1006 } 1007 handle.clear(); 1008 } 1009 1010 *status = lStatus; 1011 return handle; 1012} 1013 1014sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1015{ 1016 Mutex::Autolock _l(mLock); 1017 return getEffect_l(sessionId, effectId); 1018} 1019 1020sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1021{ 1022 sp<EffectChain> chain = getEffectChain_l(sessionId); 1023 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1024} 1025 1026// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1027// PlaybackThread::mLock held 1028status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1029{ 1030 // check for existing effect chain with the requested audio session 1031 int sessionId = effect->sessionId(); 1032 sp<EffectChain> chain = getEffectChain_l(sessionId); 1033 bool chainCreated = false; 1034 1035 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1036 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1037 this, effect->desc().name, effect->desc().flags); 1038 1039 if (chain == 0) { 1040 // create a new chain for this session 1041 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1042 chain = new EffectChain(this, sessionId); 1043 addEffectChain_l(chain); 1044 chain->setStrategy(getStrategyForSession_l(sessionId)); 1045 chainCreated = true; 1046 } 1047 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1048 1049 if (chain->getEffectFromId_l(effect->id()) != 0) { 1050 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1051 this, effect->desc().name, chain.get()); 1052 return BAD_VALUE; 1053 } 1054 1055 effect->setOffloaded(mType == OFFLOAD, mId); 1056 1057 status_t status = chain->addEffect_l(effect); 1058 if (status != NO_ERROR) { 1059 if (chainCreated) { 1060 removeEffectChain_l(chain); 1061 } 1062 return status; 1063 } 1064 1065 effect->setDevice(mOutDevice); 1066 effect->setDevice(mInDevice); 1067 effect->setMode(mAudioFlinger->getMode()); 1068 effect->setAudioSource(mAudioSource); 1069 return NO_ERROR; 1070} 1071 1072void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1073 1074 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1075 effect_descriptor_t desc = effect->desc(); 1076 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1077 detachAuxEffect_l(effect->id()); 1078 } 1079 1080 sp<EffectChain> chain = effect->chain().promote(); 1081 if (chain != 0) { 1082 // remove effect chain if removing last effect 1083 if (chain->removeEffect_l(effect) == 0) { 1084 removeEffectChain_l(chain); 1085 } 1086 } else { 1087 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1088 } 1089} 1090 1091void AudioFlinger::ThreadBase::lockEffectChains_l( 1092 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1093{ 1094 effectChains = mEffectChains; 1095 for (size_t i = 0; i < mEffectChains.size(); i++) { 1096 mEffectChains[i]->lock(); 1097 } 1098} 1099 1100void AudioFlinger::ThreadBase::unlockEffectChains( 1101 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1102{ 1103 for (size_t i = 0; i < effectChains.size(); i++) { 1104 effectChains[i]->unlock(); 1105 } 1106} 1107 1108sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1109{ 1110 Mutex::Autolock _l(mLock); 1111 return getEffectChain_l(sessionId); 1112} 1113 1114sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1115{ 1116 size_t size = mEffectChains.size(); 1117 for (size_t i = 0; i < size; i++) { 1118 if (mEffectChains[i]->sessionId() == sessionId) { 1119 return mEffectChains[i]; 1120 } 1121 } 1122 return 0; 1123} 1124 1125void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1126{ 1127 Mutex::Autolock _l(mLock); 1128 size_t size = mEffectChains.size(); 1129 for (size_t i = 0; i < size; i++) { 1130 mEffectChains[i]->setMode_l(mode); 1131 } 1132} 1133 1134void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1135 EffectHandle *handle, 1136 bool unpinIfLast) { 1137 1138 Mutex::Autolock _l(mLock); 1139 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1140 // delete the effect module if removing last handle on it 1141 if (effect->removeHandle(handle) == 0) { 1142 if (!effect->isPinned() || unpinIfLast) { 1143 removeEffect_l(effect); 1144 AudioSystem::unregisterEffect(effect->id()); 1145 } 1146 } 1147} 1148 1149// ---------------------------------------------------------------------------- 1150// Playback 1151// ---------------------------------------------------------------------------- 1152 1153AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1154 AudioStreamOut* output, 1155 audio_io_handle_t id, 1156 audio_devices_t device, 1157 type_t type) 1158 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1159 mNormalFrameCount(0), mSinkBuffer(NULL), 1160 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1161 mMixerBuffer(NULL), 1162 mMixerBufferSize(0), 1163 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1164 mMixerBufferValid(false), 1165 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1166 mEffectBuffer(NULL), 1167 mEffectBufferSize(0), 1168 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1169 mEffectBufferValid(false), 1170 mSuspended(0), mBytesWritten(0), 1171 mActiveTracksGeneration(0), 1172 // mStreamTypes[] initialized in constructor body 1173 mOutput(output), 1174 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1175 mMixerStatus(MIXER_IDLE), 1176 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1177 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1178 mBytesRemaining(0), 1179 mCurrentWriteLength(0), 1180 mUseAsyncWrite(false), 1181 mWriteAckSequence(0), 1182 mDrainSequence(0), 1183 mSignalPending(false), 1184 mScreenState(AudioFlinger::mScreenState), 1185 // index 0 is reserved for normal mixer's submix 1186 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1187 // mLatchD, mLatchQ, 1188 mLatchDValid(false), mLatchQValid(false) 1189{ 1190 snprintf(mName, kNameLength, "AudioOut_%X", id); 1191 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1192 1193 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1194 // it would be safer to explicitly pass initial masterVolume/masterMute as 1195 // parameter. 1196 // 1197 // If the HAL we are using has support for master volume or master mute, 1198 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1199 // and the mute set to false). 1200 mMasterVolume = audioFlinger->masterVolume_l(); 1201 mMasterMute = audioFlinger->masterMute_l(); 1202 if (mOutput && mOutput->audioHwDev) { 1203 if (mOutput->audioHwDev->canSetMasterVolume()) { 1204 mMasterVolume = 1.0; 1205 } 1206 1207 if (mOutput->audioHwDev->canSetMasterMute()) { 1208 mMasterMute = false; 1209 } 1210 } 1211 1212 readOutputParameters_l(); 1213 1214 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1215 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1216 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1217 stream = (audio_stream_type_t) (stream + 1)) { 1218 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1219 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1220 } 1221 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1222 // because mAudioFlinger doesn't have one to copy from 1223} 1224 1225AudioFlinger::PlaybackThread::~PlaybackThread() 1226{ 1227 mAudioFlinger->unregisterWriter(mNBLogWriter); 1228 free(mSinkBuffer); 1229 free(mMixerBuffer); 1230 free(mEffectBuffer); 1231} 1232 1233void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1234{ 1235 dumpInternals(fd, args); 1236 dumpTracks(fd, args); 1237 dumpEffectChains(fd, args); 1238} 1239 1240void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1241{ 1242 const size_t SIZE = 256; 1243 char buffer[SIZE]; 1244 String8 result; 1245 1246 result.appendFormat(" Stream volumes in dB: "); 1247 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1248 const stream_type_t *st = &mStreamTypes[i]; 1249 if (i > 0) { 1250 result.appendFormat(", "); 1251 } 1252 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1253 if (st->mute) { 1254 result.append("M"); 1255 } 1256 } 1257 result.append("\n"); 1258 write(fd, result.string(), result.length()); 1259 result.clear(); 1260 1261 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1262 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1263 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1264 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1265 1266 size_t numtracks = mTracks.size(); 1267 size_t numactive = mActiveTracks.size(); 1268 dprintf(fd, " %d Tracks", numtracks); 1269 size_t numactiveseen = 0; 1270 if (numtracks) { 1271 dprintf(fd, " of which %d are active\n", numactive); 1272 Track::appendDumpHeader(result); 1273 for (size_t i = 0; i < numtracks; ++i) { 1274 sp<Track> track = mTracks[i]; 1275 if (track != 0) { 1276 bool active = mActiveTracks.indexOf(track) >= 0; 1277 if (active) { 1278 numactiveseen++; 1279 } 1280 track->dump(buffer, SIZE, active); 1281 result.append(buffer); 1282 } 1283 } 1284 } else { 1285 result.append("\n"); 1286 } 1287 if (numactiveseen != numactive) { 1288 // some tracks in the active list were not in the tracks list 1289 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1290 " not in the track list\n"); 1291 result.append(buffer); 1292 Track::appendDumpHeader(result); 1293 for (size_t i = 0; i < numactive; ++i) { 1294 sp<Track> track = mActiveTracks[i].promote(); 1295 if (track != 0 && mTracks.indexOf(track) < 0) { 1296 track->dump(buffer, SIZE, true); 1297 result.append(buffer); 1298 } 1299 } 1300 } 1301 1302 write(fd, result.string(), result.size()); 1303} 1304 1305void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1306{ 1307 dprintf(fd, "\nOutput thread %p:\n", this); 1308 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1309 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1310 dprintf(fd, " Total writes: %d\n", mNumWrites); 1311 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1312 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1313 dprintf(fd, " Suspend count: %d\n", mSuspended); 1314 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1315 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1316 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1317 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1318 1319 dumpBase(fd, args); 1320} 1321 1322// Thread virtuals 1323 1324void AudioFlinger::PlaybackThread::onFirstRef() 1325{ 1326 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1327} 1328 1329// ThreadBase virtuals 1330void AudioFlinger::PlaybackThread::preExit() 1331{ 1332 ALOGV(" preExit()"); 1333 // FIXME this is using hard-coded strings but in the future, this functionality will be 1334 // converted to use audio HAL extensions required to support tunneling 1335 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1336} 1337 1338// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1339sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1340 const sp<AudioFlinger::Client>& client, 1341 audio_stream_type_t streamType, 1342 uint32_t sampleRate, 1343 audio_format_t format, 1344 audio_channel_mask_t channelMask, 1345 size_t *pFrameCount, 1346 const sp<IMemory>& sharedBuffer, 1347 int sessionId, 1348 IAudioFlinger::track_flags_t *flags, 1349 pid_t tid, 1350 int uid, 1351 status_t *status) 1352{ 1353 size_t frameCount = *pFrameCount; 1354 sp<Track> track; 1355 status_t lStatus; 1356 1357 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1358 1359 // client expresses a preference for FAST, but we get the final say 1360 if (*flags & IAudioFlinger::TRACK_FAST) { 1361 if ( 1362 // not timed 1363 (!isTimed) && 1364 // either of these use cases: 1365 ( 1366 // use case 1: shared buffer with any frame count 1367 ( 1368 (sharedBuffer != 0) 1369 ) || 1370 // use case 2: callback handler and frame count is default or at least as large as HAL 1371 ( 1372 (tid != -1) && 1373 ((frameCount == 0) || 1374 (frameCount >= mFrameCount)) 1375 ) 1376 ) && 1377 // PCM data 1378 audio_is_linear_pcm(format) && 1379 // mono or stereo 1380 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1381 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1382 // hardware sample rate 1383 (sampleRate == mSampleRate) && 1384 // normal mixer has an associated fast mixer 1385 hasFastMixer() && 1386 // there are sufficient fast track slots available 1387 (mFastTrackAvailMask != 0) 1388 // FIXME test that MixerThread for this fast track has a capable output HAL 1389 // FIXME add a permission test also? 1390 ) { 1391 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1392 if (frameCount == 0) { 1393 // read the fast track multiplier property the first time it is needed 1394 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1395 if (ok != 0) { 1396 ALOGE("%s pthread_once failed: %d", __func__, ok); 1397 } 1398 frameCount = mFrameCount * sFastTrackMultiplier; 1399 } 1400 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1401 frameCount, mFrameCount); 1402 } else { 1403 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1404 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1405 "sampleRate=%u mSampleRate=%u " 1406 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1407 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1408 audio_is_linear_pcm(format), 1409 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1410 *flags &= ~IAudioFlinger::TRACK_FAST; 1411 // For compatibility with AudioTrack calculation, buffer depth is forced 1412 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1413 // This is probably too conservative, but legacy application code may depend on it. 1414 // If you change this calculation, also review the start threshold which is related. 1415 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1416 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1417 if (minBufCount < 2) { 1418 minBufCount = 2; 1419 } 1420 size_t minFrameCount = mNormalFrameCount * minBufCount; 1421 if (frameCount < minFrameCount) { 1422 frameCount = minFrameCount; 1423 } 1424 } 1425 } 1426 *pFrameCount = frameCount; 1427 1428 switch (mType) { 1429 1430 case DIRECT: 1431 if (audio_is_linear_pcm(format)) { 1432 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1433 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1434 "for output %p with format %#x", 1435 sampleRate, format, channelMask, mOutput, mFormat); 1436 lStatus = BAD_VALUE; 1437 goto Exit; 1438 } 1439 } 1440 break; 1441 1442 case OFFLOAD: 1443 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1444 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1445 "for output %p with format %#x", 1446 sampleRate, format, channelMask, mOutput, mFormat); 1447 lStatus = BAD_VALUE; 1448 goto Exit; 1449 } 1450 break; 1451 1452 default: 1453 if (!audio_is_linear_pcm(format)) { 1454 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1455 "for output %p with format %#x", 1456 format, mOutput, mFormat); 1457 lStatus = BAD_VALUE; 1458 goto Exit; 1459 } 1460 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1461 if (sampleRate > mSampleRate*2) { 1462 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1463 lStatus = BAD_VALUE; 1464 goto Exit; 1465 } 1466 break; 1467 1468 } 1469 1470 lStatus = initCheck(); 1471 if (lStatus != NO_ERROR) { 1472 ALOGE("createTrack_l() audio driver not initialized"); 1473 goto Exit; 1474 } 1475 1476 { // scope for mLock 1477 Mutex::Autolock _l(mLock); 1478 1479 // all tracks in same audio session must share the same routing strategy otherwise 1480 // conflicts will happen when tracks are moved from one output to another by audio policy 1481 // manager 1482 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1483 for (size_t i = 0; i < mTracks.size(); ++i) { 1484 sp<Track> t = mTracks[i]; 1485 if (t != 0 && !t->isOutputTrack()) { 1486 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1487 if (sessionId == t->sessionId() && strategy != actual) { 1488 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1489 strategy, actual); 1490 lStatus = BAD_VALUE; 1491 goto Exit; 1492 } 1493 } 1494 } 1495 1496 if (!isTimed) { 1497 track = new Track(this, client, streamType, sampleRate, format, 1498 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1499 } else { 1500 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1501 channelMask, frameCount, sharedBuffer, sessionId, uid); 1502 } 1503 1504 // new Track always returns non-NULL, 1505 // but TimedTrack::create() is a factory that could fail by returning NULL 1506 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1507 if (lStatus != NO_ERROR) { 1508 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1509 // track must be cleared from the caller as the caller has the AF lock 1510 goto Exit; 1511 } 1512 mTracks.add(track); 1513 1514 sp<EffectChain> chain = getEffectChain_l(sessionId); 1515 if (chain != 0) { 1516 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1517 track->setMainBuffer(chain->inBuffer()); 1518 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1519 chain->incTrackCnt(); 1520 } 1521 1522 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1523 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1524 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1525 // so ask activity manager to do this on our behalf 1526 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1527 } 1528 } 1529 1530 lStatus = NO_ERROR; 1531 1532Exit: 1533 *status = lStatus; 1534 return track; 1535} 1536 1537uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1538{ 1539 return latency; 1540} 1541 1542uint32_t AudioFlinger::PlaybackThread::latency() const 1543{ 1544 Mutex::Autolock _l(mLock); 1545 return latency_l(); 1546} 1547uint32_t AudioFlinger::PlaybackThread::latency_l() const 1548{ 1549 if (initCheck() == NO_ERROR) { 1550 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1551 } else { 1552 return 0; 1553 } 1554} 1555 1556void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1557{ 1558 Mutex::Autolock _l(mLock); 1559 // Don't apply master volume in SW if our HAL can do it for us. 1560 if (mOutput && mOutput->audioHwDev && 1561 mOutput->audioHwDev->canSetMasterVolume()) { 1562 mMasterVolume = 1.0; 1563 } else { 1564 mMasterVolume = value; 1565 } 1566} 1567 1568void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1569{ 1570 Mutex::Autolock _l(mLock); 1571 // Don't apply master mute in SW if our HAL can do it for us. 1572 if (mOutput && mOutput->audioHwDev && 1573 mOutput->audioHwDev->canSetMasterMute()) { 1574 mMasterMute = false; 1575 } else { 1576 mMasterMute = muted; 1577 } 1578} 1579 1580void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1581{ 1582 Mutex::Autolock _l(mLock); 1583 mStreamTypes[stream].volume = value; 1584 broadcast_l(); 1585} 1586 1587void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1588{ 1589 Mutex::Autolock _l(mLock); 1590 mStreamTypes[stream].mute = muted; 1591 broadcast_l(); 1592} 1593 1594float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1595{ 1596 Mutex::Autolock _l(mLock); 1597 return mStreamTypes[stream].volume; 1598} 1599 1600// addTrack_l() must be called with ThreadBase::mLock held 1601status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1602{ 1603 status_t status = ALREADY_EXISTS; 1604 1605 // set retry count for buffer fill 1606 track->mRetryCount = kMaxTrackStartupRetries; 1607 if (mActiveTracks.indexOf(track) < 0) { 1608 // the track is newly added, make sure it fills up all its 1609 // buffers before playing. This is to ensure the client will 1610 // effectively get the latency it requested. 1611 if (!track->isOutputTrack()) { 1612 TrackBase::track_state state = track->mState; 1613 mLock.unlock(); 1614 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1615 mLock.lock(); 1616 // abort track was stopped/paused while we released the lock 1617 if (state != track->mState) { 1618 if (status == NO_ERROR) { 1619 mLock.unlock(); 1620 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1621 mLock.lock(); 1622 } 1623 return INVALID_OPERATION; 1624 } 1625 // abort if start is rejected by audio policy manager 1626 if (status != NO_ERROR) { 1627 return PERMISSION_DENIED; 1628 } 1629#ifdef ADD_BATTERY_DATA 1630 // to track the speaker usage 1631 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1632#endif 1633 } 1634 1635 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1636 track->mResetDone = false; 1637 track->mPresentationCompleteFrames = 0; 1638 mActiveTracks.add(track); 1639 mWakeLockUids.add(track->uid()); 1640 mActiveTracksGeneration++; 1641 mLatestActiveTrack = track; 1642 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1643 if (chain != 0) { 1644 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1645 track->sessionId()); 1646 chain->incActiveTrackCnt(); 1647 } 1648 1649 status = NO_ERROR; 1650 } 1651 1652 onAddNewTrack_l(); 1653 return status; 1654} 1655 1656bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1657{ 1658 track->terminate(); 1659 // active tracks are removed by threadLoop() 1660 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1661 track->mState = TrackBase::STOPPED; 1662 if (!trackActive) { 1663 removeTrack_l(track); 1664 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1665 track->mState = TrackBase::STOPPING_1; 1666 } 1667 1668 return trackActive; 1669} 1670 1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1672{ 1673 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1674 mTracks.remove(track); 1675 deleteTrackName_l(track->name()); 1676 // redundant as track is about to be destroyed, for dumpsys only 1677 track->mName = -1; 1678 if (track->isFastTrack()) { 1679 int index = track->mFastIndex; 1680 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1681 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1682 mFastTrackAvailMask |= 1 << index; 1683 // redundant as track is about to be destroyed, for dumpsys only 1684 track->mFastIndex = -1; 1685 } 1686 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1687 if (chain != 0) { 1688 chain->decTrackCnt(); 1689 } 1690} 1691 1692void AudioFlinger::PlaybackThread::broadcast_l() 1693{ 1694 // Thread could be blocked waiting for async 1695 // so signal it to handle state changes immediately 1696 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1697 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1698 mSignalPending = true; 1699 mWaitWorkCV.broadcast(); 1700} 1701 1702String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1703{ 1704 Mutex::Autolock _l(mLock); 1705 if (initCheck() != NO_ERROR) { 1706 return String8(); 1707 } 1708 1709 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1710 const String8 out_s8(s); 1711 free(s); 1712 return out_s8; 1713} 1714 1715void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1716 AudioSystem::OutputDescriptor desc; 1717 void *param2 = NULL; 1718 1719 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1720 param); 1721 1722 switch (event) { 1723 case AudioSystem::OUTPUT_OPENED: 1724 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1725 desc.channelMask = mChannelMask; 1726 desc.samplingRate = mSampleRate; 1727 desc.format = mFormat; 1728 desc.frameCount = mNormalFrameCount; // FIXME see 1729 // AudioFlinger::frameCount(audio_io_handle_t) 1730 desc.latency = latency_l(); 1731 param2 = &desc; 1732 break; 1733 1734 case AudioSystem::STREAM_CONFIG_CHANGED: 1735 param2 = ¶m; 1736 case AudioSystem::OUTPUT_CLOSED: 1737 default: 1738 break; 1739 } 1740 mAudioFlinger->audioConfigChanged(event, mId, param2); 1741} 1742 1743void AudioFlinger::PlaybackThread::writeCallback() 1744{ 1745 ALOG_ASSERT(mCallbackThread != 0); 1746 mCallbackThread->resetWriteBlocked(); 1747} 1748 1749void AudioFlinger::PlaybackThread::drainCallback() 1750{ 1751 ALOG_ASSERT(mCallbackThread != 0); 1752 mCallbackThread->resetDraining(); 1753} 1754 1755void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1756{ 1757 Mutex::Autolock _l(mLock); 1758 // reject out of sequence requests 1759 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1760 mWriteAckSequence &= ~1; 1761 mWaitWorkCV.signal(); 1762 } 1763} 1764 1765void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1766{ 1767 Mutex::Autolock _l(mLock); 1768 // reject out of sequence requests 1769 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1770 mDrainSequence &= ~1; 1771 mWaitWorkCV.signal(); 1772 } 1773} 1774 1775// static 1776int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1777 void *param __unused, 1778 void *cookie) 1779{ 1780 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1781 ALOGV("asyncCallback() event %d", event); 1782 switch (event) { 1783 case STREAM_CBK_EVENT_WRITE_READY: 1784 me->writeCallback(); 1785 break; 1786 case STREAM_CBK_EVENT_DRAIN_READY: 1787 me->drainCallback(); 1788 break; 1789 default: 1790 ALOGW("asyncCallback() unknown event %d", event); 1791 break; 1792 } 1793 return 0; 1794} 1795 1796void AudioFlinger::PlaybackThread::readOutputParameters_l() 1797{ 1798 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1799 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1800 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1801 if (!audio_is_output_channel(mChannelMask)) { 1802 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1803 } 1804 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1805 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1806 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1807 } 1808 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1809 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1810 if (!audio_is_valid_format(mFormat)) { 1811 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1812 } 1813 if ((mType == MIXER || mType == DUPLICATING) 1814 && !isValidPcmSinkFormat(mFormat)) { 1815 LOG_FATAL("HAL format %#x not supported for mixed output", 1816 mFormat); 1817 } 1818 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1819 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1820 mFrameCount = mBufferSize / mFrameSize; 1821 if (mFrameCount & 15) { 1822 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1823 mFrameCount); 1824 } 1825 1826 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1827 (mOutput->stream->set_callback != NULL)) { 1828 if (mOutput->stream->set_callback(mOutput->stream, 1829 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1830 mUseAsyncWrite = true; 1831 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1832 } 1833 } 1834 1835 // Calculate size of normal sink buffer relative to the HAL output buffer size 1836 double multiplier = 1.0; 1837 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1838 kUseFastMixer == FastMixer_Dynamic)) { 1839 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1840 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1841 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1842 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1843 maxNormalFrameCount = maxNormalFrameCount & ~15; 1844 if (maxNormalFrameCount < minNormalFrameCount) { 1845 maxNormalFrameCount = minNormalFrameCount; 1846 } 1847 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1848 if (multiplier <= 1.0) { 1849 multiplier = 1.0; 1850 } else if (multiplier <= 2.0) { 1851 if (2 * mFrameCount <= maxNormalFrameCount) { 1852 multiplier = 2.0; 1853 } else { 1854 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1855 } 1856 } else { 1857 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1858 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1859 // track, but we sometimes have to do this to satisfy the maximum frame count 1860 // constraint) 1861 // FIXME this rounding up should not be done if no HAL SRC 1862 uint32_t truncMult = (uint32_t) multiplier; 1863 if ((truncMult & 1)) { 1864 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1865 ++truncMult; 1866 } 1867 } 1868 multiplier = (double) truncMult; 1869 } 1870 } 1871 mNormalFrameCount = multiplier * mFrameCount; 1872 // round up to nearest 16 frames to satisfy AudioMixer 1873 if (mType == MIXER || mType == DUPLICATING) { 1874 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1875 } 1876 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1877 mNormalFrameCount); 1878 1879 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1880 // Originally this was int16_t[] array, need to remove legacy implications. 1881 free(mSinkBuffer); 1882 mSinkBuffer = NULL; 1883 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1884 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1885 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1886 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1887 1888 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1889 // drives the output. 1890 free(mMixerBuffer); 1891 mMixerBuffer = NULL; 1892 if (mMixerBufferEnabled) { 1893 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1894 mMixerBufferSize = mNormalFrameCount * mChannelCount 1895 * audio_bytes_per_sample(mMixerBufferFormat); 1896 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1897 } 1898 free(mEffectBuffer); 1899 mEffectBuffer = NULL; 1900 if (mEffectBufferEnabled) { 1901 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1902 mEffectBufferSize = mNormalFrameCount * mChannelCount 1903 * audio_bytes_per_sample(mEffectBufferFormat); 1904 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1905 } 1906 1907 // force reconfiguration of effect chains and engines to take new buffer size and audio 1908 // parameters into account 1909 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1910 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1911 // matter. 1912 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1913 Vector< sp<EffectChain> > effectChains = mEffectChains; 1914 for (size_t i = 0; i < effectChains.size(); i ++) { 1915 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1916 } 1917} 1918 1919 1920status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1921{ 1922 if (halFrames == NULL || dspFrames == NULL) { 1923 return BAD_VALUE; 1924 } 1925 Mutex::Autolock _l(mLock); 1926 if (initCheck() != NO_ERROR) { 1927 return INVALID_OPERATION; 1928 } 1929 size_t framesWritten = mBytesWritten / mFrameSize; 1930 *halFrames = framesWritten; 1931 1932 if (isSuspended()) { 1933 // return an estimation of rendered frames when the output is suspended 1934 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1935 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1936 return NO_ERROR; 1937 } else { 1938 status_t status; 1939 uint32_t frames; 1940 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1941 *dspFrames = (size_t)frames; 1942 return status; 1943 } 1944} 1945 1946uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1947{ 1948 Mutex::Autolock _l(mLock); 1949 uint32_t result = 0; 1950 if (getEffectChain_l(sessionId) != 0) { 1951 result = EFFECT_SESSION; 1952 } 1953 1954 for (size_t i = 0; i < mTracks.size(); ++i) { 1955 sp<Track> track = mTracks[i]; 1956 if (sessionId == track->sessionId() && !track->isInvalid()) { 1957 result |= TRACK_SESSION; 1958 break; 1959 } 1960 } 1961 1962 return result; 1963} 1964 1965uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1966{ 1967 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1968 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1969 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1970 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1971 } 1972 for (size_t i = 0; i < mTracks.size(); i++) { 1973 sp<Track> track = mTracks[i]; 1974 if (sessionId == track->sessionId() && !track->isInvalid()) { 1975 return AudioSystem::getStrategyForStream(track->streamType()); 1976 } 1977 } 1978 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1979} 1980 1981 1982AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1983{ 1984 Mutex::Autolock _l(mLock); 1985 return mOutput; 1986} 1987 1988AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1989{ 1990 Mutex::Autolock _l(mLock); 1991 AudioStreamOut *output = mOutput; 1992 mOutput = NULL; 1993 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1994 // must push a NULL and wait for ack 1995 mOutputSink.clear(); 1996 mPipeSink.clear(); 1997 mNormalSink.clear(); 1998 return output; 1999} 2000 2001// this method must always be called either with ThreadBase mLock held or inside the thread loop 2002audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2003{ 2004 if (mOutput == NULL) { 2005 return NULL; 2006 } 2007 return &mOutput->stream->common; 2008} 2009 2010uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2011{ 2012 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2013} 2014 2015status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2016{ 2017 if (!isValidSyncEvent(event)) { 2018 return BAD_VALUE; 2019 } 2020 2021 Mutex::Autolock _l(mLock); 2022 2023 for (size_t i = 0; i < mTracks.size(); ++i) { 2024 sp<Track> track = mTracks[i]; 2025 if (event->triggerSession() == track->sessionId()) { 2026 (void) track->setSyncEvent(event); 2027 return NO_ERROR; 2028 } 2029 } 2030 2031 return NAME_NOT_FOUND; 2032} 2033 2034bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2035{ 2036 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2037} 2038 2039void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2040 const Vector< sp<Track> >& tracksToRemove) 2041{ 2042 size_t count = tracksToRemove.size(); 2043 if (count > 0) { 2044 for (size_t i = 0 ; i < count ; i++) { 2045 const sp<Track>& track = tracksToRemove.itemAt(i); 2046 if (!track->isOutputTrack()) { 2047 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2048#ifdef ADD_BATTERY_DATA 2049 // to track the speaker usage 2050 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2051#endif 2052 if (track->isTerminated()) { 2053 AudioSystem::releaseOutput(mId); 2054 } 2055 } 2056 } 2057 } 2058} 2059 2060void AudioFlinger::PlaybackThread::checkSilentMode_l() 2061{ 2062 if (!mMasterMute) { 2063 char value[PROPERTY_VALUE_MAX]; 2064 if (property_get("ro.audio.silent", value, "0") > 0) { 2065 char *endptr; 2066 unsigned long ul = strtoul(value, &endptr, 0); 2067 if (*endptr == '\0' && ul != 0) { 2068 ALOGD("Silence is golden"); 2069 // The setprop command will not allow a property to be changed after 2070 // the first time it is set, so we don't have to worry about un-muting. 2071 setMasterMute_l(true); 2072 } 2073 } 2074 } 2075} 2076 2077// shared by MIXER and DIRECT, overridden by DUPLICATING 2078ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2079{ 2080 // FIXME rewrite to reduce number of system calls 2081 mLastWriteTime = systemTime(); 2082 mInWrite = true; 2083 ssize_t bytesWritten; 2084 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2085 2086 // If an NBAIO sink is present, use it to write the normal mixer's submix 2087 if (mNormalSink != 0) { 2088 const size_t count = mBytesRemaining / mFrameSize; 2089 2090 ATRACE_BEGIN("write"); 2091 // update the setpoint when AudioFlinger::mScreenState changes 2092 uint32_t screenState = AudioFlinger::mScreenState; 2093 if (screenState != mScreenState) { 2094 mScreenState = screenState; 2095 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2096 if (pipe != NULL) { 2097 pipe->setAvgFrames((mScreenState & 1) ? 2098 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2099 } 2100 } 2101 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2102 ATRACE_END(); 2103 if (framesWritten > 0) { 2104 bytesWritten = framesWritten * mFrameSize; 2105 } else { 2106 bytesWritten = framesWritten; 2107 } 2108 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2109 if (status == NO_ERROR) { 2110 size_t totalFramesWritten = mNormalSink->framesWritten(); 2111 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2112 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2113 mLatchDValid = true; 2114 } 2115 } 2116 // otherwise use the HAL / AudioStreamOut directly 2117 } else { 2118 // Direct output and offload threads 2119 2120 if (mUseAsyncWrite) { 2121 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2122 mWriteAckSequence += 2; 2123 mWriteAckSequence |= 1; 2124 ALOG_ASSERT(mCallbackThread != 0); 2125 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2126 } 2127 // FIXME We should have an implementation of timestamps for direct output threads. 2128 // They are used e.g for multichannel PCM playback over HDMI. 2129 bytesWritten = mOutput->stream->write(mOutput->stream, 2130 (char *)mSinkBuffer + offset, mBytesRemaining); 2131 if (mUseAsyncWrite && 2132 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2133 // do not wait for async callback in case of error of full write 2134 mWriteAckSequence &= ~1; 2135 ALOG_ASSERT(mCallbackThread != 0); 2136 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2137 } 2138 } 2139 2140 mNumWrites++; 2141 mInWrite = false; 2142 mStandby = false; 2143 return bytesWritten; 2144} 2145 2146void AudioFlinger::PlaybackThread::threadLoop_drain() 2147{ 2148 if (mOutput->stream->drain) { 2149 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2150 if (mUseAsyncWrite) { 2151 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2152 mDrainSequence |= 1; 2153 ALOG_ASSERT(mCallbackThread != 0); 2154 mCallbackThread->setDraining(mDrainSequence); 2155 } 2156 mOutput->stream->drain(mOutput->stream, 2157 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2158 : AUDIO_DRAIN_ALL); 2159 } 2160} 2161 2162void AudioFlinger::PlaybackThread::threadLoop_exit() 2163{ 2164 // Default implementation has nothing to do 2165} 2166 2167/* 2168The derived values that are cached: 2169 - mSinkBufferSize from frame count * frame size 2170 - activeSleepTime from activeSleepTimeUs() 2171 - idleSleepTime from idleSleepTimeUs() 2172 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2173 - maxPeriod from frame count and sample rate (MIXER only) 2174 2175The parameters that affect these derived values are: 2176 - frame count 2177 - frame size 2178 - sample rate 2179 - device type: A2DP or not 2180 - device latency 2181 - format: PCM or not 2182 - active sleep time 2183 - idle sleep time 2184*/ 2185 2186void AudioFlinger::PlaybackThread::cacheParameters_l() 2187{ 2188 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2189 activeSleepTime = activeSleepTimeUs(); 2190 idleSleepTime = idleSleepTimeUs(); 2191} 2192 2193void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2194{ 2195 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2196 this, streamType, mTracks.size()); 2197 Mutex::Autolock _l(mLock); 2198 2199 size_t size = mTracks.size(); 2200 for (size_t i = 0; i < size; i++) { 2201 sp<Track> t = mTracks[i]; 2202 if (t->streamType() == streamType) { 2203 t->invalidate(); 2204 } 2205 } 2206} 2207 2208status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2209{ 2210 int session = chain->sessionId(); 2211 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2212 ? mEffectBuffer : mSinkBuffer); 2213 bool ownsBuffer = false; 2214 2215 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2216 if (session > 0) { 2217 // Only one effect chain can be present in direct output thread and it uses 2218 // the sink buffer as input 2219 if (mType != DIRECT) { 2220 size_t numSamples = mNormalFrameCount * mChannelCount; 2221 buffer = new int16_t[numSamples]; 2222 memset(buffer, 0, numSamples * sizeof(int16_t)); 2223 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2224 ownsBuffer = true; 2225 } 2226 2227 // Attach all tracks with same session ID to this chain. 2228 for (size_t i = 0; i < mTracks.size(); ++i) { 2229 sp<Track> track = mTracks[i]; 2230 if (session == track->sessionId()) { 2231 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2232 buffer); 2233 track->setMainBuffer(buffer); 2234 chain->incTrackCnt(); 2235 } 2236 } 2237 2238 // indicate all active tracks in the chain 2239 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2240 sp<Track> track = mActiveTracks[i].promote(); 2241 if (track == 0) { 2242 continue; 2243 } 2244 if (session == track->sessionId()) { 2245 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2246 chain->incActiveTrackCnt(); 2247 } 2248 } 2249 } 2250 2251 chain->setInBuffer(buffer, ownsBuffer); 2252 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2253 ? mEffectBuffer : mSinkBuffer)); 2254 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2255 // chains list in order to be processed last as it contains output stage effects 2256 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2257 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2258 // after track specific effects and before output stage 2259 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2260 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2261 // Effect chain for other sessions are inserted at beginning of effect 2262 // chains list to be processed before output mix effects. Relative order between other 2263 // sessions is not important 2264 size_t size = mEffectChains.size(); 2265 size_t i = 0; 2266 for (i = 0; i < size; i++) { 2267 if (mEffectChains[i]->sessionId() < session) { 2268 break; 2269 } 2270 } 2271 mEffectChains.insertAt(chain, i); 2272 checkSuspendOnAddEffectChain_l(chain); 2273 2274 return NO_ERROR; 2275} 2276 2277size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2278{ 2279 int session = chain->sessionId(); 2280 2281 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2282 2283 for (size_t i = 0; i < mEffectChains.size(); i++) { 2284 if (chain == mEffectChains[i]) { 2285 mEffectChains.removeAt(i); 2286 // detach all active tracks from the chain 2287 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2288 sp<Track> track = mActiveTracks[i].promote(); 2289 if (track == 0) { 2290 continue; 2291 } 2292 if (session == track->sessionId()) { 2293 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2294 chain.get(), session); 2295 chain->decActiveTrackCnt(); 2296 } 2297 } 2298 2299 // detach all tracks with same session ID from this chain 2300 for (size_t i = 0; i < mTracks.size(); ++i) { 2301 sp<Track> track = mTracks[i]; 2302 if (session == track->sessionId()) { 2303 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2304 chain->decTrackCnt(); 2305 } 2306 } 2307 break; 2308 } 2309 } 2310 return mEffectChains.size(); 2311} 2312 2313status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2314 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2315{ 2316 Mutex::Autolock _l(mLock); 2317 return attachAuxEffect_l(track, EffectId); 2318} 2319 2320status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2321 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2322{ 2323 status_t status = NO_ERROR; 2324 2325 if (EffectId == 0) { 2326 track->setAuxBuffer(0, NULL); 2327 } else { 2328 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2329 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2330 if (effect != 0) { 2331 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2332 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2333 } else { 2334 status = INVALID_OPERATION; 2335 } 2336 } else { 2337 status = BAD_VALUE; 2338 } 2339 } 2340 return status; 2341} 2342 2343void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2344{ 2345 for (size_t i = 0; i < mTracks.size(); ++i) { 2346 sp<Track> track = mTracks[i]; 2347 if (track->auxEffectId() == effectId) { 2348 attachAuxEffect_l(track, 0); 2349 } 2350 } 2351} 2352 2353bool AudioFlinger::PlaybackThread::threadLoop() 2354{ 2355 Vector< sp<Track> > tracksToRemove; 2356 2357 standbyTime = systemTime(); 2358 2359 // MIXER 2360 nsecs_t lastWarning = 0; 2361 2362 // DUPLICATING 2363 // FIXME could this be made local to while loop? 2364 writeFrames = 0; 2365 2366 int lastGeneration = 0; 2367 2368 cacheParameters_l(); 2369 sleepTime = idleSleepTime; 2370 2371 if (mType == MIXER) { 2372 sleepTimeShift = 0; 2373 } 2374 2375 CpuStats cpuStats; 2376 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2377 2378 acquireWakeLock(); 2379 2380 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2381 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2382 // and then that string will be logged at the next convenient opportunity. 2383 const char *logString = NULL; 2384 2385 checkSilentMode_l(); 2386 2387 while (!exitPending()) 2388 { 2389 cpuStats.sample(myName); 2390 2391 Vector< sp<EffectChain> > effectChains; 2392 2393 { // scope for mLock 2394 2395 Mutex::Autolock _l(mLock); 2396 2397 processConfigEvents_l(); 2398 2399 if (logString != NULL) { 2400 mNBLogWriter->logTimestamp(); 2401 mNBLogWriter->log(logString); 2402 logString = NULL; 2403 } 2404 2405 if (mLatchDValid) { 2406 mLatchQ = mLatchD; 2407 mLatchDValid = false; 2408 mLatchQValid = true; 2409 } 2410 2411 saveOutputTracks(); 2412 if (mSignalPending) { 2413 // A signal was raised while we were unlocked 2414 mSignalPending = false; 2415 } else if (waitingAsyncCallback_l()) { 2416 if (exitPending()) { 2417 break; 2418 } 2419 releaseWakeLock_l(); 2420 mWakeLockUids.clear(); 2421 mActiveTracksGeneration++; 2422 ALOGV("wait async completion"); 2423 mWaitWorkCV.wait(mLock); 2424 ALOGV("async completion/wake"); 2425 acquireWakeLock_l(); 2426 standbyTime = systemTime() + standbyDelay; 2427 sleepTime = 0; 2428 2429 continue; 2430 } 2431 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2432 isSuspended()) { 2433 // put audio hardware into standby after short delay 2434 if (shouldStandby_l()) { 2435 2436 threadLoop_standby(); 2437 2438 mStandby = true; 2439 } 2440 2441 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2442 // we're about to wait, flush the binder command buffer 2443 IPCThreadState::self()->flushCommands(); 2444 2445 clearOutputTracks(); 2446 2447 if (exitPending()) { 2448 break; 2449 } 2450 2451 releaseWakeLock_l(); 2452 mWakeLockUids.clear(); 2453 mActiveTracksGeneration++; 2454 // wait until we have something to do... 2455 ALOGV("%s going to sleep", myName.string()); 2456 mWaitWorkCV.wait(mLock); 2457 ALOGV("%s waking up", myName.string()); 2458 acquireWakeLock_l(); 2459 2460 mMixerStatus = MIXER_IDLE; 2461 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2462 mBytesWritten = 0; 2463 mBytesRemaining = 0; 2464 checkSilentMode_l(); 2465 2466 standbyTime = systemTime() + standbyDelay; 2467 sleepTime = idleSleepTime; 2468 if (mType == MIXER) { 2469 sleepTimeShift = 0; 2470 } 2471 2472 continue; 2473 } 2474 } 2475 // mMixerStatusIgnoringFastTracks is also updated internally 2476 mMixerStatus = prepareTracks_l(&tracksToRemove); 2477 2478 // compare with previously applied list 2479 if (lastGeneration != mActiveTracksGeneration) { 2480 // update wakelock 2481 updateWakeLockUids_l(mWakeLockUids); 2482 lastGeneration = mActiveTracksGeneration; 2483 } 2484 2485 // prevent any changes in effect chain list and in each effect chain 2486 // during mixing and effect process as the audio buffers could be deleted 2487 // or modified if an effect is created or deleted 2488 lockEffectChains_l(effectChains); 2489 } // mLock scope ends 2490 2491 if (mBytesRemaining == 0) { 2492 mCurrentWriteLength = 0; 2493 if (mMixerStatus == MIXER_TRACKS_READY) { 2494 // threadLoop_mix() sets mCurrentWriteLength 2495 threadLoop_mix(); 2496 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2497 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2498 // threadLoop_sleepTime sets sleepTime to 0 if data 2499 // must be written to HAL 2500 threadLoop_sleepTime(); 2501 if (sleepTime == 0) { 2502 mCurrentWriteLength = mSinkBufferSize; 2503 } 2504 } 2505 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2506 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2507 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2508 // or mSinkBuffer (if there are no effects). 2509 // 2510 // This is done pre-effects computation; if effects change to 2511 // support higher precision, this needs to move. 2512 // 2513 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2514 // TODO use sleepTime == 0 as an additional condition. 2515 if (mMixerBufferValid) { 2516 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2517 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2518 2519 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2520 mNormalFrameCount * mChannelCount); 2521 } 2522 2523 mBytesRemaining = mCurrentWriteLength; 2524 if (isSuspended()) { 2525 sleepTime = suspendSleepTimeUs(); 2526 // simulate write to HAL when suspended 2527 mBytesWritten += mSinkBufferSize; 2528 mBytesRemaining = 0; 2529 } 2530 2531 // only process effects if we're going to write 2532 if (sleepTime == 0 && mType != OFFLOAD) { 2533 for (size_t i = 0; i < effectChains.size(); i ++) { 2534 effectChains[i]->process_l(); 2535 } 2536 } 2537 } 2538 // Process effect chains for offloaded thread even if no audio 2539 // was read from audio track: process only updates effect state 2540 // and thus does have to be synchronized with audio writes but may have 2541 // to be called while waiting for async write callback 2542 if (mType == OFFLOAD) { 2543 for (size_t i = 0; i < effectChains.size(); i ++) { 2544 effectChains[i]->process_l(); 2545 } 2546 } 2547 2548 // Only if the Effects buffer is enabled and there is data in the 2549 // Effects buffer (buffer valid), we need to 2550 // copy into the sink buffer. 2551 // TODO use sleepTime == 0 as an additional condition. 2552 if (mEffectBufferValid) { 2553 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2554 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2555 mNormalFrameCount * mChannelCount); 2556 } 2557 2558 // enable changes in effect chain 2559 unlockEffectChains(effectChains); 2560 2561 if (!waitingAsyncCallback()) { 2562 // sleepTime == 0 means we must write to audio hardware 2563 if (sleepTime == 0) { 2564 if (mBytesRemaining) { 2565 ssize_t ret = threadLoop_write(); 2566 if (ret < 0) { 2567 mBytesRemaining = 0; 2568 } else { 2569 mBytesWritten += ret; 2570 mBytesRemaining -= ret; 2571 } 2572 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2573 (mMixerStatus == MIXER_DRAIN_ALL)) { 2574 threadLoop_drain(); 2575 } 2576 if (mType == MIXER) { 2577 // write blocked detection 2578 nsecs_t now = systemTime(); 2579 nsecs_t delta = now - mLastWriteTime; 2580 if (!mStandby && delta > maxPeriod) { 2581 mNumDelayedWrites++; 2582 if ((now - lastWarning) > kWarningThrottleNs) { 2583 ATRACE_NAME("underrun"); 2584 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2585 ns2ms(delta), mNumDelayedWrites, this); 2586 lastWarning = now; 2587 } 2588 } 2589 } 2590 2591 } else { 2592 usleep(sleepTime); 2593 } 2594 } 2595 2596 // Finally let go of removed track(s), without the lock held 2597 // since we can't guarantee the destructors won't acquire that 2598 // same lock. This will also mutate and push a new fast mixer state. 2599 threadLoop_removeTracks(tracksToRemove); 2600 tracksToRemove.clear(); 2601 2602 // FIXME I don't understand the need for this here; 2603 // it was in the original code but maybe the 2604 // assignment in saveOutputTracks() makes this unnecessary? 2605 clearOutputTracks(); 2606 2607 // Effect chains will be actually deleted here if they were removed from 2608 // mEffectChains list during mixing or effects processing 2609 effectChains.clear(); 2610 2611 // FIXME Note that the above .clear() is no longer necessary since effectChains 2612 // is now local to this block, but will keep it for now (at least until merge done). 2613 } 2614 2615 threadLoop_exit(); 2616 2617 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2618 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2619 // put output stream into standby mode 2620 if (!mStandby) { 2621 mOutput->stream->common.standby(&mOutput->stream->common); 2622 } 2623 } 2624 2625 releaseWakeLock(); 2626 mWakeLockUids.clear(); 2627 mActiveTracksGeneration++; 2628 2629 ALOGV("Thread %p type %d exiting", this, mType); 2630 return false; 2631} 2632 2633// removeTracks_l() must be called with ThreadBase::mLock held 2634void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2635{ 2636 size_t count = tracksToRemove.size(); 2637 if (count > 0) { 2638 for (size_t i=0 ; i<count ; i++) { 2639 const sp<Track>& track = tracksToRemove.itemAt(i); 2640 mActiveTracks.remove(track); 2641 mWakeLockUids.remove(track->uid()); 2642 mActiveTracksGeneration++; 2643 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2644 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2645 if (chain != 0) { 2646 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2647 track->sessionId()); 2648 chain->decActiveTrackCnt(); 2649 } 2650 if (track->isTerminated()) { 2651 removeTrack_l(track); 2652 } 2653 } 2654 } 2655 2656} 2657 2658status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2659{ 2660 if (mNormalSink != 0) { 2661 return mNormalSink->getTimestamp(timestamp); 2662 } 2663 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2664 uint64_t position64; 2665 int ret = mOutput->stream->get_presentation_position( 2666 mOutput->stream, &position64, ×tamp.mTime); 2667 if (ret == 0) { 2668 timestamp.mPosition = (uint32_t)position64; 2669 return NO_ERROR; 2670 } 2671 } 2672 return INVALID_OPERATION; 2673} 2674 2675status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2676 audio_patch_handle_t *handle) 2677{ 2678 status_t status = NO_ERROR; 2679 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2680 // store new device and send to effects 2681 audio_devices_t type = AUDIO_DEVICE_NONE; 2682 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2683 type |= patch->sinks[i].ext.device.type; 2684 } 2685 mOutDevice = type; 2686 for (size_t i = 0; i < mEffectChains.size(); i++) { 2687 mEffectChains[i]->setDevice_l(mOutDevice); 2688 } 2689 2690 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2691 status = hwDevice->create_audio_patch(hwDevice, 2692 patch->num_sources, 2693 patch->sources, 2694 patch->num_sinks, 2695 patch->sinks, 2696 handle); 2697 } else { 2698 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2699 } 2700 return status; 2701} 2702 2703status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2704{ 2705 status_t status = NO_ERROR; 2706 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2707 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2708 status = hwDevice->release_audio_patch(hwDevice, handle); 2709 } else { 2710 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2711 } 2712 return status; 2713} 2714 2715// ---------------------------------------------------------------------------- 2716 2717AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2718 audio_io_handle_t id, audio_devices_t device, type_t type) 2719 : PlaybackThread(audioFlinger, output, id, device, type), 2720 // mAudioMixer below 2721 // mFastMixer below 2722 mFastMixerFutex(0) 2723 // mOutputSink below 2724 // mPipeSink below 2725 // mNormalSink below 2726{ 2727 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2728 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2729 "mFrameCount=%d, mNormalFrameCount=%d", 2730 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2731 mNormalFrameCount); 2732 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2733 2734 // FIXME - Current mixer implementation only supports stereo output 2735 if (mChannelCount != FCC_2) { 2736 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2737 } 2738 2739 // create an NBAIO sink for the HAL output stream, and negotiate 2740 mOutputSink = new AudioStreamOutSink(output->stream); 2741 size_t numCounterOffers = 0; 2742 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2743 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2744 ALOG_ASSERT(index == 0); 2745 2746 // initialize fast mixer depending on configuration 2747 bool initFastMixer; 2748 switch (kUseFastMixer) { 2749 case FastMixer_Never: 2750 initFastMixer = false; 2751 break; 2752 case FastMixer_Always: 2753 initFastMixer = true; 2754 break; 2755 case FastMixer_Static: 2756 case FastMixer_Dynamic: 2757 initFastMixer = mFrameCount < mNormalFrameCount; 2758 break; 2759 } 2760 if (initFastMixer) { 2761 audio_format_t fastMixerFormat; 2762 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2763 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2764 } else { 2765 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2766 } 2767 if (mFormat != fastMixerFormat) { 2768 // change our Sink format to accept our intermediate precision 2769 mFormat = fastMixerFormat; 2770 free(mSinkBuffer); 2771 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2772 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2773 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2774 } 2775 2776 // create a MonoPipe to connect our submix to FastMixer 2777 NBAIO_Format format = mOutputSink->format(); 2778 // adjust format to match that of the Fast Mixer 2779 format.mFormat = fastMixerFormat; 2780 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2781 2782 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2783 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2784 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2785 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2786 const NBAIO_Format offers[1] = {format}; 2787 size_t numCounterOffers = 0; 2788 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2789 ALOG_ASSERT(index == 0); 2790 monoPipe->setAvgFrames((mScreenState & 1) ? 2791 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2792 mPipeSink = monoPipe; 2793 2794#ifdef TEE_SINK 2795 if (mTeeSinkOutputEnabled) { 2796 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2797 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2798 numCounterOffers = 0; 2799 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2800 ALOG_ASSERT(index == 0); 2801 mTeeSink = teeSink; 2802 PipeReader *teeSource = new PipeReader(*teeSink); 2803 numCounterOffers = 0; 2804 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2805 ALOG_ASSERT(index == 0); 2806 mTeeSource = teeSource; 2807 } 2808#endif 2809 2810 // create fast mixer and configure it initially with just one fast track for our submix 2811 mFastMixer = new FastMixer(); 2812 FastMixerStateQueue *sq = mFastMixer->sq(); 2813#ifdef STATE_QUEUE_DUMP 2814 sq->setObserverDump(&mStateQueueObserverDump); 2815 sq->setMutatorDump(&mStateQueueMutatorDump); 2816#endif 2817 FastMixerState *state = sq->begin(); 2818 FastTrack *fastTrack = &state->mFastTracks[0]; 2819 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2820 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2821 fastTrack->mVolumeProvider = NULL; 2822 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2823 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2824 fastTrack->mGeneration++; 2825 state->mFastTracksGen++; 2826 state->mTrackMask = 1; 2827 // fast mixer will use the HAL output sink 2828 state->mOutputSink = mOutputSink.get(); 2829 state->mOutputSinkGen++; 2830 state->mFrameCount = mFrameCount; 2831 state->mCommand = FastMixerState::COLD_IDLE; 2832 // already done in constructor initialization list 2833 //mFastMixerFutex = 0; 2834 state->mColdFutexAddr = &mFastMixerFutex; 2835 state->mColdGen++; 2836 state->mDumpState = &mFastMixerDumpState; 2837#ifdef TEE_SINK 2838 state->mTeeSink = mTeeSink.get(); 2839#endif 2840 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2841 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2842 sq->end(); 2843 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2844 2845 // start the fast mixer 2846 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2847 pid_t tid = mFastMixer->getTid(); 2848 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2849 if (err != 0) { 2850 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2851 kPriorityFastMixer, getpid_cached, tid, err); 2852 } 2853 2854#ifdef AUDIO_WATCHDOG 2855 // create and start the watchdog 2856 mAudioWatchdog = new AudioWatchdog(); 2857 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2858 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2859 tid = mAudioWatchdog->getTid(); 2860 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2861 if (err != 0) { 2862 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2863 kPriorityFastMixer, getpid_cached, tid, err); 2864 } 2865#endif 2866 2867 } 2868 2869 switch (kUseFastMixer) { 2870 case FastMixer_Never: 2871 case FastMixer_Dynamic: 2872 mNormalSink = mOutputSink; 2873 break; 2874 case FastMixer_Always: 2875 mNormalSink = mPipeSink; 2876 break; 2877 case FastMixer_Static: 2878 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2879 break; 2880 } 2881} 2882 2883AudioFlinger::MixerThread::~MixerThread() 2884{ 2885 if (mFastMixer != 0) { 2886 FastMixerStateQueue *sq = mFastMixer->sq(); 2887 FastMixerState *state = sq->begin(); 2888 if (state->mCommand == FastMixerState::COLD_IDLE) { 2889 int32_t old = android_atomic_inc(&mFastMixerFutex); 2890 if (old == -1) { 2891 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2892 } 2893 } 2894 state->mCommand = FastMixerState::EXIT; 2895 sq->end(); 2896 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2897 mFastMixer->join(); 2898 // Though the fast mixer thread has exited, it's state queue is still valid. 2899 // We'll use that extract the final state which contains one remaining fast track 2900 // corresponding to our sub-mix. 2901 state = sq->begin(); 2902 ALOG_ASSERT(state->mTrackMask == 1); 2903 FastTrack *fastTrack = &state->mFastTracks[0]; 2904 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2905 delete fastTrack->mBufferProvider; 2906 sq->end(false /*didModify*/); 2907 mFastMixer.clear(); 2908#ifdef AUDIO_WATCHDOG 2909 if (mAudioWatchdog != 0) { 2910 mAudioWatchdog->requestExit(); 2911 mAudioWatchdog->requestExitAndWait(); 2912 mAudioWatchdog.clear(); 2913 } 2914#endif 2915 } 2916 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2917 delete mAudioMixer; 2918} 2919 2920 2921uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2922{ 2923 if (mFastMixer != 0) { 2924 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2925 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2926 } 2927 return latency; 2928} 2929 2930 2931void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2932{ 2933 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2934} 2935 2936ssize_t AudioFlinger::MixerThread::threadLoop_write() 2937{ 2938 // FIXME we should only do one push per cycle; confirm this is true 2939 // Start the fast mixer if it's not already running 2940 if (mFastMixer != 0) { 2941 FastMixerStateQueue *sq = mFastMixer->sq(); 2942 FastMixerState *state = sq->begin(); 2943 if (state->mCommand != FastMixerState::MIX_WRITE && 2944 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2945 if (state->mCommand == FastMixerState::COLD_IDLE) { 2946 int32_t old = android_atomic_inc(&mFastMixerFutex); 2947 if (old == -1) { 2948 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2949 } 2950#ifdef AUDIO_WATCHDOG 2951 if (mAudioWatchdog != 0) { 2952 mAudioWatchdog->resume(); 2953 } 2954#endif 2955 } 2956 state->mCommand = FastMixerState::MIX_WRITE; 2957 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2958 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2959 sq->end(); 2960 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2961 if (kUseFastMixer == FastMixer_Dynamic) { 2962 mNormalSink = mPipeSink; 2963 } 2964 } else { 2965 sq->end(false /*didModify*/); 2966 } 2967 } 2968 return PlaybackThread::threadLoop_write(); 2969} 2970 2971void AudioFlinger::MixerThread::threadLoop_standby() 2972{ 2973 // Idle the fast mixer if it's currently running 2974 if (mFastMixer != 0) { 2975 FastMixerStateQueue *sq = mFastMixer->sq(); 2976 FastMixerState *state = sq->begin(); 2977 if (!(state->mCommand & FastMixerState::IDLE)) { 2978 state->mCommand = FastMixerState::COLD_IDLE; 2979 state->mColdFutexAddr = &mFastMixerFutex; 2980 state->mColdGen++; 2981 mFastMixerFutex = 0; 2982 sq->end(); 2983 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2984 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2985 if (kUseFastMixer == FastMixer_Dynamic) { 2986 mNormalSink = mOutputSink; 2987 } 2988#ifdef AUDIO_WATCHDOG 2989 if (mAudioWatchdog != 0) { 2990 mAudioWatchdog->pause(); 2991 } 2992#endif 2993 } else { 2994 sq->end(false /*didModify*/); 2995 } 2996 } 2997 PlaybackThread::threadLoop_standby(); 2998} 2999 3000bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3001{ 3002 return false; 3003} 3004 3005bool AudioFlinger::PlaybackThread::shouldStandby_l() 3006{ 3007 return !mStandby; 3008} 3009 3010bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3011{ 3012 Mutex::Autolock _l(mLock); 3013 return waitingAsyncCallback_l(); 3014} 3015 3016// shared by MIXER and DIRECT, overridden by DUPLICATING 3017void AudioFlinger::PlaybackThread::threadLoop_standby() 3018{ 3019 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3020 mOutput->stream->common.standby(&mOutput->stream->common); 3021 if (mUseAsyncWrite != 0) { 3022 // discard any pending drain or write ack by incrementing sequence 3023 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3024 mDrainSequence = (mDrainSequence + 2) & ~1; 3025 ALOG_ASSERT(mCallbackThread != 0); 3026 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3027 mCallbackThread->setDraining(mDrainSequence); 3028 } 3029} 3030 3031void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3032{ 3033 ALOGV("signal playback thread"); 3034 broadcast_l(); 3035} 3036 3037void AudioFlinger::MixerThread::threadLoop_mix() 3038{ 3039 // obtain the presentation timestamp of the next output buffer 3040 int64_t pts; 3041 status_t status = INVALID_OPERATION; 3042 3043 if (mNormalSink != 0) { 3044 status = mNormalSink->getNextWriteTimestamp(&pts); 3045 } else { 3046 status = mOutputSink->getNextWriteTimestamp(&pts); 3047 } 3048 3049 if (status != NO_ERROR) { 3050 pts = AudioBufferProvider::kInvalidPTS; 3051 } 3052 3053 // mix buffers... 3054 mAudioMixer->process(pts); 3055 mCurrentWriteLength = mSinkBufferSize; 3056 // increase sleep time progressively when application underrun condition clears. 3057 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3058 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3059 // such that we would underrun the audio HAL. 3060 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3061 sleepTimeShift--; 3062 } 3063 sleepTime = 0; 3064 standbyTime = systemTime() + standbyDelay; 3065 //TODO: delay standby when effects have a tail 3066} 3067 3068void AudioFlinger::MixerThread::threadLoop_sleepTime() 3069{ 3070 // If no tracks are ready, sleep once for the duration of an output 3071 // buffer size, then write 0s to the output 3072 if (sleepTime == 0) { 3073 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3074 sleepTime = activeSleepTime >> sleepTimeShift; 3075 if (sleepTime < kMinThreadSleepTimeUs) { 3076 sleepTime = kMinThreadSleepTimeUs; 3077 } 3078 // reduce sleep time in case of consecutive application underruns to avoid 3079 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3080 // duration we would end up writing less data than needed by the audio HAL if 3081 // the condition persists. 3082 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3083 sleepTimeShift++; 3084 } 3085 } else { 3086 sleepTime = idleSleepTime; 3087 } 3088 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3089 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3090 // before effects processing or output. 3091 if (mMixerBufferValid) { 3092 memset(mMixerBuffer, 0, mMixerBufferSize); 3093 } else { 3094 memset(mSinkBuffer, 0, mSinkBufferSize); 3095 } 3096 sleepTime = 0; 3097 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3098 "anticipated start"); 3099 } 3100 // TODO add standby time extension fct of effect tail 3101} 3102 3103// prepareTracks_l() must be called with ThreadBase::mLock held 3104AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3105 Vector< sp<Track> > *tracksToRemove) 3106{ 3107 3108 mixer_state mixerStatus = MIXER_IDLE; 3109 // find out which tracks need to be processed 3110 size_t count = mActiveTracks.size(); 3111 size_t mixedTracks = 0; 3112 size_t tracksWithEffect = 0; 3113 // counts only _active_ fast tracks 3114 size_t fastTracks = 0; 3115 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3116 3117 float masterVolume = mMasterVolume; 3118 bool masterMute = mMasterMute; 3119 3120 if (masterMute) { 3121 masterVolume = 0; 3122 } 3123 // Delegate master volume control to effect in output mix effect chain if needed 3124 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3125 if (chain != 0) { 3126 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3127 chain->setVolume_l(&v, &v); 3128 masterVolume = (float)((v + (1 << 23)) >> 24); 3129 chain.clear(); 3130 } 3131 3132 // prepare a new state to push 3133 FastMixerStateQueue *sq = NULL; 3134 FastMixerState *state = NULL; 3135 bool didModify = false; 3136 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3137 if (mFastMixer != 0) { 3138 sq = mFastMixer->sq(); 3139 state = sq->begin(); 3140 } 3141 3142 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3143 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3144 3145 for (size_t i=0 ; i<count ; i++) { 3146 const sp<Track> t = mActiveTracks[i].promote(); 3147 if (t == 0) { 3148 continue; 3149 } 3150 3151 // this const just means the local variable doesn't change 3152 Track* const track = t.get(); 3153 3154 // process fast tracks 3155 if (track->isFastTrack()) { 3156 3157 // It's theoretically possible (though unlikely) for a fast track to be created 3158 // and then removed within the same normal mix cycle. This is not a problem, as 3159 // the track never becomes active so it's fast mixer slot is never touched. 3160 // The converse, of removing an (active) track and then creating a new track 3161 // at the identical fast mixer slot within the same normal mix cycle, 3162 // is impossible because the slot isn't marked available until the end of each cycle. 3163 int j = track->mFastIndex; 3164 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3165 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3166 FastTrack *fastTrack = &state->mFastTracks[j]; 3167 3168 // Determine whether the track is currently in underrun condition, 3169 // and whether it had a recent underrun. 3170 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3171 FastTrackUnderruns underruns = ftDump->mUnderruns; 3172 uint32_t recentFull = (underruns.mBitFields.mFull - 3173 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3174 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3175 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3176 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3177 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3178 uint32_t recentUnderruns = recentPartial + recentEmpty; 3179 track->mObservedUnderruns = underruns; 3180 // don't count underruns that occur while stopping or pausing 3181 // or stopped which can occur when flush() is called while active 3182 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3183 recentUnderruns > 0) { 3184 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3185 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3186 } 3187 3188 // This is similar to the state machine for normal tracks, 3189 // with a few modifications for fast tracks. 3190 bool isActive = true; 3191 switch (track->mState) { 3192 case TrackBase::STOPPING_1: 3193 // track stays active in STOPPING_1 state until first underrun 3194 if (recentUnderruns > 0 || track->isTerminated()) { 3195 track->mState = TrackBase::STOPPING_2; 3196 } 3197 break; 3198 case TrackBase::PAUSING: 3199 // ramp down is not yet implemented 3200 track->setPaused(); 3201 break; 3202 case TrackBase::RESUMING: 3203 // ramp up is not yet implemented 3204 track->mState = TrackBase::ACTIVE; 3205 break; 3206 case TrackBase::ACTIVE: 3207 if (recentFull > 0 || recentPartial > 0) { 3208 // track has provided at least some frames recently: reset retry count 3209 track->mRetryCount = kMaxTrackRetries; 3210 } 3211 if (recentUnderruns == 0) { 3212 // no recent underruns: stay active 3213 break; 3214 } 3215 // there has recently been an underrun of some kind 3216 if (track->sharedBuffer() == 0) { 3217 // were any of the recent underruns "empty" (no frames available)? 3218 if (recentEmpty == 0) { 3219 // no, then ignore the partial underruns as they are allowed indefinitely 3220 break; 3221 } 3222 // there has recently been an "empty" underrun: decrement the retry counter 3223 if (--(track->mRetryCount) > 0) { 3224 break; 3225 } 3226 // indicate to client process that the track was disabled because of underrun; 3227 // it will then automatically call start() when data is available 3228 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3229 // remove from active list, but state remains ACTIVE [confusing but true] 3230 isActive = false; 3231 break; 3232 } 3233 // fall through 3234 case TrackBase::STOPPING_2: 3235 case TrackBase::PAUSED: 3236 case TrackBase::STOPPED: 3237 case TrackBase::FLUSHED: // flush() while active 3238 // Check for presentation complete if track is inactive 3239 // We have consumed all the buffers of this track. 3240 // This would be incomplete if we auto-paused on underrun 3241 { 3242 size_t audioHALFrames = 3243 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3244 size_t framesWritten = mBytesWritten / mFrameSize; 3245 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3246 // track stays in active list until presentation is complete 3247 break; 3248 } 3249 } 3250 if (track->isStopping_2()) { 3251 track->mState = TrackBase::STOPPED; 3252 } 3253 if (track->isStopped()) { 3254 // Can't reset directly, as fast mixer is still polling this track 3255 // track->reset(); 3256 // So instead mark this track as needing to be reset after push with ack 3257 resetMask |= 1 << i; 3258 } 3259 isActive = false; 3260 break; 3261 case TrackBase::IDLE: 3262 default: 3263 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3264 } 3265 3266 if (isActive) { 3267 // was it previously inactive? 3268 if (!(state->mTrackMask & (1 << j))) { 3269 ExtendedAudioBufferProvider *eabp = track; 3270 VolumeProvider *vp = track; 3271 fastTrack->mBufferProvider = eabp; 3272 fastTrack->mVolumeProvider = vp; 3273 fastTrack->mChannelMask = track->mChannelMask; 3274 fastTrack->mFormat = track->mFormat; 3275 fastTrack->mGeneration++; 3276 state->mTrackMask |= 1 << j; 3277 didModify = true; 3278 // no acknowledgement required for newly active tracks 3279 } 3280 // cache the combined master volume and stream type volume for fast mixer; this 3281 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3282 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3283 ++fastTracks; 3284 } else { 3285 // was it previously active? 3286 if (state->mTrackMask & (1 << j)) { 3287 fastTrack->mBufferProvider = NULL; 3288 fastTrack->mGeneration++; 3289 state->mTrackMask &= ~(1 << j); 3290 didModify = true; 3291 // If any fast tracks were removed, we must wait for acknowledgement 3292 // because we're about to decrement the last sp<> on those tracks. 3293 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3294 } else { 3295 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3296 } 3297 tracksToRemove->add(track); 3298 // Avoids a misleading display in dumpsys 3299 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3300 } 3301 continue; 3302 } 3303 3304 { // local variable scope to avoid goto warning 3305 3306 audio_track_cblk_t* cblk = track->cblk(); 3307 3308 // The first time a track is added we wait 3309 // for all its buffers to be filled before processing it 3310 int name = track->name(); 3311 // make sure that we have enough frames to mix one full buffer. 3312 // enforce this condition only once to enable draining the buffer in case the client 3313 // app does not call stop() and relies on underrun to stop: 3314 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3315 // during last round 3316 size_t desiredFrames; 3317 uint32_t sr = track->sampleRate(); 3318 if (sr == mSampleRate) { 3319 desiredFrames = mNormalFrameCount; 3320 } else { 3321 // +1 for rounding and +1 for additional sample needed for interpolation 3322 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3323 // add frames already consumed but not yet released by the resampler 3324 // because mAudioTrackServerProxy->framesReady() will include these frames 3325 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3326#if 0 3327 // the minimum track buffer size is normally twice the number of frames necessary 3328 // to fill one buffer and the resampler should not leave more than one buffer worth 3329 // of unreleased frames after each pass, but just in case... 3330 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3331#endif 3332 } 3333 uint32_t minFrames = 1; 3334 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3335 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3336 minFrames = desiredFrames; 3337 } 3338 3339 size_t framesReady = track->framesReady(); 3340 if ((framesReady >= minFrames) && track->isReady() && 3341 !track->isPaused() && !track->isTerminated()) 3342 { 3343 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3344 3345 mixedTracks++; 3346 3347 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3348 // there is an effect chain connected to the track 3349 chain.clear(); 3350 if (track->mainBuffer() != mSinkBuffer && 3351 track->mainBuffer() != mMixerBuffer) { 3352 if (mEffectBufferEnabled) { 3353 mEffectBufferValid = true; // Later can set directly. 3354 } 3355 chain = getEffectChain_l(track->sessionId()); 3356 // Delegate volume control to effect in track effect chain if needed 3357 if (chain != 0) { 3358 tracksWithEffect++; 3359 } else { 3360 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3361 "session %d", 3362 name, track->sessionId()); 3363 } 3364 } 3365 3366 3367 int param = AudioMixer::VOLUME; 3368 if (track->mFillingUpStatus == Track::FS_FILLED) { 3369 // no ramp for the first volume setting 3370 track->mFillingUpStatus = Track::FS_ACTIVE; 3371 if (track->mState == TrackBase::RESUMING) { 3372 track->mState = TrackBase::ACTIVE; 3373 param = AudioMixer::RAMP_VOLUME; 3374 } 3375 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3376 // FIXME should not make a decision based on mServer 3377 } else if (cblk->mServer != 0) { 3378 // If the track is stopped before the first frame was mixed, 3379 // do not apply ramp 3380 param = AudioMixer::RAMP_VOLUME; 3381 } 3382 3383 // compute volume for this track 3384 uint32_t vl, vr; // in U8.24 integer format 3385 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3386 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3387 vl = vr = 0; 3388 vlf = vrf = vaf = 0.; 3389 if (track->isPausing()) { 3390 track->setPaused(); 3391 } 3392 } else { 3393 3394 // read original volumes with volume control 3395 float typeVolume = mStreamTypes[track->streamType()].volume; 3396 float v = masterVolume * typeVolume; 3397 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3398 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3399 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3400 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3401 // track volumes come from shared memory, so can't be trusted and must be clamped 3402 if (vlf > GAIN_FLOAT_UNITY) { 3403 ALOGV("Track left volume out of range: %.3g", vlf); 3404 vlf = GAIN_FLOAT_UNITY; 3405 } 3406 if (vrf > GAIN_FLOAT_UNITY) { 3407 ALOGV("Track right volume out of range: %.3g", vrf); 3408 vrf = GAIN_FLOAT_UNITY; 3409 } 3410 // now apply the master volume and stream type volume 3411 vlf *= v; 3412 vrf *= v; 3413 // assuming master volume and stream type volume each go up to 1.0, 3414 // then derive vl and vr as U8.24 versions for the effect chain 3415 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3416 vl = (uint32_t) (scaleto8_24 * vlf); 3417 vr = (uint32_t) (scaleto8_24 * vrf); 3418 // vl and vr are now in U8.24 format 3419 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3420 // send level comes from shared memory and so may be corrupt 3421 if (sendLevel > MAX_GAIN_INT) { 3422 ALOGV("Track send level out of range: %04X", sendLevel); 3423 sendLevel = MAX_GAIN_INT; 3424 } 3425 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3426 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3427 } 3428 3429 // Delegate volume control to effect in track effect chain if needed 3430 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3431 // Do not ramp volume if volume is controlled by effect 3432 param = AudioMixer::VOLUME; 3433 // Update remaining floating point volume levels 3434 vlf = (float)vl / (1 << 24); 3435 vrf = (float)vr / (1 << 24); 3436 track->mHasVolumeController = true; 3437 } else { 3438 // force no volume ramp when volume controller was just disabled or removed 3439 // from effect chain to avoid volume spike 3440 if (track->mHasVolumeController) { 3441 param = AudioMixer::VOLUME; 3442 } 3443 track->mHasVolumeController = false; 3444 } 3445 3446 // XXX: these things DON'T need to be done each time 3447 mAudioMixer->setBufferProvider(name, track); 3448 mAudioMixer->enable(name); 3449 3450 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3451 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3452 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3453 mAudioMixer->setParameter( 3454 name, 3455 AudioMixer::TRACK, 3456 AudioMixer::FORMAT, (void *)track->format()); 3457 mAudioMixer->setParameter( 3458 name, 3459 AudioMixer::TRACK, 3460 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3461 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3462 uint32_t maxSampleRate = mSampleRate * 2; 3463 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3464 if (reqSampleRate == 0) { 3465 reqSampleRate = mSampleRate; 3466 } else if (reqSampleRate > maxSampleRate) { 3467 reqSampleRate = maxSampleRate; 3468 } 3469 mAudioMixer->setParameter( 3470 name, 3471 AudioMixer::RESAMPLE, 3472 AudioMixer::SAMPLE_RATE, 3473 (void *)(uintptr_t)reqSampleRate); 3474 /* 3475 * Select the appropriate output buffer for the track. 3476 * 3477 * Tracks with effects go into their own effects chain buffer 3478 * and from there into either mEffectBuffer or mSinkBuffer. 3479 * 3480 * Other tracks can use mMixerBuffer for higher precision 3481 * channel accumulation. If this buffer is enabled 3482 * (mMixerBufferEnabled true), then selected tracks will accumulate 3483 * into it. 3484 * 3485 */ 3486 if (mMixerBufferEnabled 3487 && (track->mainBuffer() == mSinkBuffer 3488 || track->mainBuffer() == mMixerBuffer)) { 3489 mAudioMixer->setParameter( 3490 name, 3491 AudioMixer::TRACK, 3492 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3493 mAudioMixer->setParameter( 3494 name, 3495 AudioMixer::TRACK, 3496 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3497 // TODO: override track->mainBuffer()? 3498 mMixerBufferValid = true; 3499 } else { 3500 mAudioMixer->setParameter( 3501 name, 3502 AudioMixer::TRACK, 3503 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3504 mAudioMixer->setParameter( 3505 name, 3506 AudioMixer::TRACK, 3507 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3508 } 3509 mAudioMixer->setParameter( 3510 name, 3511 AudioMixer::TRACK, 3512 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3513 3514 // reset retry count 3515 track->mRetryCount = kMaxTrackRetries; 3516 3517 // If one track is ready, set the mixer ready if: 3518 // - the mixer was not ready during previous round OR 3519 // - no other track is not ready 3520 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3521 mixerStatus != MIXER_TRACKS_ENABLED) { 3522 mixerStatus = MIXER_TRACKS_READY; 3523 } 3524 } else { 3525 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3526 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3527 } 3528 // clear effect chain input buffer if an active track underruns to avoid sending 3529 // previous audio buffer again to effects 3530 chain = getEffectChain_l(track->sessionId()); 3531 if (chain != 0) { 3532 chain->clearInputBuffer(); 3533 } 3534 3535 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3536 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3537 track->isStopped() || track->isPaused()) { 3538 // We have consumed all the buffers of this track. 3539 // Remove it from the list of active tracks. 3540 // TODO: use actual buffer filling status instead of latency when available from 3541 // audio HAL 3542 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3543 size_t framesWritten = mBytesWritten / mFrameSize; 3544 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3545 if (track->isStopped()) { 3546 track->reset(); 3547 } 3548 tracksToRemove->add(track); 3549 } 3550 } else { 3551 // No buffers for this track. Give it a few chances to 3552 // fill a buffer, then remove it from active list. 3553 if (--(track->mRetryCount) <= 0) { 3554 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3555 tracksToRemove->add(track); 3556 // indicate to client process that the track was disabled because of underrun; 3557 // it will then automatically call start() when data is available 3558 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3559 // If one track is not ready, mark the mixer also not ready if: 3560 // - the mixer was ready during previous round OR 3561 // - no other track is ready 3562 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3563 mixerStatus != MIXER_TRACKS_READY) { 3564 mixerStatus = MIXER_TRACKS_ENABLED; 3565 } 3566 } 3567 mAudioMixer->disable(name); 3568 } 3569 3570 } // local variable scope to avoid goto warning 3571track_is_ready: ; 3572 3573 } 3574 3575 // Push the new FastMixer state if necessary 3576 bool pauseAudioWatchdog = false; 3577 if (didModify) { 3578 state->mFastTracksGen++; 3579 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3580 if (kUseFastMixer == FastMixer_Dynamic && 3581 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3582 state->mCommand = FastMixerState::COLD_IDLE; 3583 state->mColdFutexAddr = &mFastMixerFutex; 3584 state->mColdGen++; 3585 mFastMixerFutex = 0; 3586 if (kUseFastMixer == FastMixer_Dynamic) { 3587 mNormalSink = mOutputSink; 3588 } 3589 // If we go into cold idle, need to wait for acknowledgement 3590 // so that fast mixer stops doing I/O. 3591 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3592 pauseAudioWatchdog = true; 3593 } 3594 } 3595 if (sq != NULL) { 3596 sq->end(didModify); 3597 sq->push(block); 3598 } 3599#ifdef AUDIO_WATCHDOG 3600 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3601 mAudioWatchdog->pause(); 3602 } 3603#endif 3604 3605 // Now perform the deferred reset on fast tracks that have stopped 3606 while (resetMask != 0) { 3607 size_t i = __builtin_ctz(resetMask); 3608 ALOG_ASSERT(i < count); 3609 resetMask &= ~(1 << i); 3610 sp<Track> t = mActiveTracks[i].promote(); 3611 if (t == 0) { 3612 continue; 3613 } 3614 Track* track = t.get(); 3615 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3616 track->reset(); 3617 } 3618 3619 // remove all the tracks that need to be... 3620 removeTracks_l(*tracksToRemove); 3621 3622 // sink or mix buffer must be cleared if all tracks are connected to an 3623 // effect chain as in this case the mixer will not write to the sink or mix buffer 3624 // and track effects will accumulate into it 3625 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3626 (mixedTracks == 0 && fastTracks > 0))) { 3627 // FIXME as a performance optimization, should remember previous zero status 3628 if (mMixerBufferValid) { 3629 memset(mMixerBuffer, 0, mMixerBufferSize); 3630 // TODO: In testing, mSinkBuffer below need not be cleared because 3631 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3632 // after mixing. 3633 // 3634 // To enforce this guarantee: 3635 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3636 // (mixedTracks == 0 && fastTracks > 0)) 3637 // must imply MIXER_TRACKS_READY. 3638 // Later, we may clear buffers regardless, and skip much of this logic. 3639 } 3640 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3641 if (mEffectBufferValid) { 3642 memset(mEffectBuffer, 0, mEffectBufferSize); 3643 } 3644 // FIXME as a performance optimization, should remember previous zero status 3645 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3646 } 3647 3648 // if any fast tracks, then status is ready 3649 mMixerStatusIgnoringFastTracks = mixerStatus; 3650 if (fastTracks > 0) { 3651 mixerStatus = MIXER_TRACKS_READY; 3652 } 3653 return mixerStatus; 3654} 3655 3656// getTrackName_l() must be called with ThreadBase::mLock held 3657int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3658 audio_format_t format, int sessionId) 3659{ 3660 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3661} 3662 3663// deleteTrackName_l() must be called with ThreadBase::mLock held 3664void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3665{ 3666 ALOGV("remove track (%d) and delete from mixer", name); 3667 mAudioMixer->deleteTrackName(name); 3668} 3669 3670// checkForNewParameter_l() must be called with ThreadBase::mLock held 3671bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3672 status_t& status) 3673{ 3674 bool reconfig = false; 3675 3676 status = NO_ERROR; 3677 3678 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3679 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3680 if (mFastMixer != 0) { 3681 FastMixerStateQueue *sq = mFastMixer->sq(); 3682 FastMixerState *state = sq->begin(); 3683 if (!(state->mCommand & FastMixerState::IDLE)) { 3684 previousCommand = state->mCommand; 3685 state->mCommand = FastMixerState::HOT_IDLE; 3686 sq->end(); 3687 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3688 } else { 3689 sq->end(false /*didModify*/); 3690 } 3691 } 3692 3693 AudioParameter param = AudioParameter(keyValuePair); 3694 int value; 3695 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3696 reconfig = true; 3697 } 3698 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3699 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3700 status = BAD_VALUE; 3701 } else { 3702 // no need to save value, since it's constant 3703 reconfig = true; 3704 } 3705 } 3706 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3707 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3708 status = BAD_VALUE; 3709 } else { 3710 // no need to save value, since it's constant 3711 reconfig = true; 3712 } 3713 } 3714 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3715 // do not accept frame count changes if tracks are open as the track buffer 3716 // size depends on frame count and correct behavior would not be guaranteed 3717 // if frame count is changed after track creation 3718 if (!mTracks.isEmpty()) { 3719 status = INVALID_OPERATION; 3720 } else { 3721 reconfig = true; 3722 } 3723 } 3724 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3725#ifdef ADD_BATTERY_DATA 3726 // when changing the audio output device, call addBatteryData to notify 3727 // the change 3728 if (mOutDevice != value) { 3729 uint32_t params = 0; 3730 // check whether speaker is on 3731 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3732 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3733 } 3734 3735 audio_devices_t deviceWithoutSpeaker 3736 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3737 // check if any other device (except speaker) is on 3738 if (value & deviceWithoutSpeaker ) { 3739 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3740 } 3741 3742 if (params != 0) { 3743 addBatteryData(params); 3744 } 3745 } 3746#endif 3747 3748 // forward device change to effects that have requested to be 3749 // aware of attached audio device. 3750 if (value != AUDIO_DEVICE_NONE) { 3751 mOutDevice = value; 3752 for (size_t i = 0; i < mEffectChains.size(); i++) { 3753 mEffectChains[i]->setDevice_l(mOutDevice); 3754 } 3755 } 3756 } 3757 3758 if (status == NO_ERROR) { 3759 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3760 keyValuePair.string()); 3761 if (!mStandby && status == INVALID_OPERATION) { 3762 mOutput->stream->common.standby(&mOutput->stream->common); 3763 mStandby = true; 3764 mBytesWritten = 0; 3765 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3766 keyValuePair.string()); 3767 } 3768 if (status == NO_ERROR && reconfig) { 3769 readOutputParameters_l(); 3770 delete mAudioMixer; 3771 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3772 for (size_t i = 0; i < mTracks.size() ; i++) { 3773 int name = getTrackName_l(mTracks[i]->mChannelMask, 3774 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3775 if (name < 0) { 3776 break; 3777 } 3778 mTracks[i]->mName = name; 3779 } 3780 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3781 } 3782 } 3783 3784 if (!(previousCommand & FastMixerState::IDLE)) { 3785 ALOG_ASSERT(mFastMixer != 0); 3786 FastMixerStateQueue *sq = mFastMixer->sq(); 3787 FastMixerState *state = sq->begin(); 3788 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3789 state->mCommand = previousCommand; 3790 sq->end(); 3791 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3792 } 3793 3794 return reconfig; 3795} 3796 3797 3798void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3799{ 3800 const size_t SIZE = 256; 3801 char buffer[SIZE]; 3802 String8 result; 3803 3804 PlaybackThread::dumpInternals(fd, args); 3805 3806 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3807 3808 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3809 const FastMixerDumpState copy(mFastMixerDumpState); 3810 copy.dump(fd); 3811 3812#ifdef STATE_QUEUE_DUMP 3813 // Similar for state queue 3814 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3815 observerCopy.dump(fd); 3816 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3817 mutatorCopy.dump(fd); 3818#endif 3819 3820#ifdef TEE_SINK 3821 // Write the tee output to a .wav file 3822 dumpTee(fd, mTeeSource, mId); 3823#endif 3824 3825#ifdef AUDIO_WATCHDOG 3826 if (mAudioWatchdog != 0) { 3827 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3828 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3829 wdCopy.dump(fd); 3830 } 3831#endif 3832} 3833 3834uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3835{ 3836 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3837} 3838 3839uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3840{ 3841 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3842} 3843 3844void AudioFlinger::MixerThread::cacheParameters_l() 3845{ 3846 PlaybackThread::cacheParameters_l(); 3847 3848 // FIXME: Relaxed timing because of a certain device that can't meet latency 3849 // Should be reduced to 2x after the vendor fixes the driver issue 3850 // increase threshold again due to low power audio mode. The way this warning 3851 // threshold is calculated and its usefulness should be reconsidered anyway. 3852 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3853} 3854 3855// ---------------------------------------------------------------------------- 3856 3857AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3858 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3859 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3860 // mLeftVolFloat, mRightVolFloat 3861{ 3862} 3863 3864AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3865 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3866 ThreadBase::type_t type) 3867 : PlaybackThread(audioFlinger, output, id, device, type) 3868 // mLeftVolFloat, mRightVolFloat 3869{ 3870} 3871 3872AudioFlinger::DirectOutputThread::~DirectOutputThread() 3873{ 3874} 3875 3876void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3877{ 3878 audio_track_cblk_t* cblk = track->cblk(); 3879 float left, right; 3880 3881 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3882 left = right = 0; 3883 } else { 3884 float typeVolume = mStreamTypes[track->streamType()].volume; 3885 float v = mMasterVolume * typeVolume; 3886 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3887 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3888 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3889 if (left > GAIN_FLOAT_UNITY) { 3890 left = GAIN_FLOAT_UNITY; 3891 } 3892 left *= v; 3893 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3894 if (right > GAIN_FLOAT_UNITY) { 3895 right = GAIN_FLOAT_UNITY; 3896 } 3897 right *= v; 3898 } 3899 3900 if (lastTrack) { 3901 if (left != mLeftVolFloat || right != mRightVolFloat) { 3902 mLeftVolFloat = left; 3903 mRightVolFloat = right; 3904 3905 // Convert volumes from float to 8.24 3906 uint32_t vl = (uint32_t)(left * (1 << 24)); 3907 uint32_t vr = (uint32_t)(right * (1 << 24)); 3908 3909 // Delegate volume control to effect in track effect chain if needed 3910 // only one effect chain can be present on DirectOutputThread, so if 3911 // there is one, the track is connected to it 3912 if (!mEffectChains.isEmpty()) { 3913 mEffectChains[0]->setVolume_l(&vl, &vr); 3914 left = (float)vl / (1 << 24); 3915 right = (float)vr / (1 << 24); 3916 } 3917 if (mOutput->stream->set_volume) { 3918 mOutput->stream->set_volume(mOutput->stream, left, right); 3919 } 3920 } 3921 } 3922} 3923 3924 3925AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3926 Vector< sp<Track> > *tracksToRemove 3927) 3928{ 3929 size_t count = mActiveTracks.size(); 3930 mixer_state mixerStatus = MIXER_IDLE; 3931 3932 // find out which tracks need to be processed 3933 for (size_t i = 0; i < count; i++) { 3934 sp<Track> t = mActiveTracks[i].promote(); 3935 // The track died recently 3936 if (t == 0) { 3937 continue; 3938 } 3939 3940 Track* const track = t.get(); 3941 audio_track_cblk_t* cblk = track->cblk(); 3942 // Only consider last track started for volume and mixer state control. 3943 // In theory an older track could underrun and restart after the new one starts 3944 // but as we only care about the transition phase between two tracks on a 3945 // direct output, it is not a problem to ignore the underrun case. 3946 sp<Track> l = mLatestActiveTrack.promote(); 3947 bool last = l.get() == track; 3948 3949 // The first time a track is added we wait 3950 // for all its buffers to be filled before processing it 3951 uint32_t minFrames; 3952 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 3953 minFrames = mNormalFrameCount; 3954 } else { 3955 minFrames = 1; 3956 } 3957 3958 ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ", 3959 minFrames, track->mState, track->framesReady()); 3960 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 3961 !track->isStopping_2() && !track->isStopped()) 3962 { 3963 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3964 3965 if (track->mFillingUpStatus == Track::FS_FILLED) { 3966 track->mFillingUpStatus = Track::FS_ACTIVE; 3967 // make sure processVolume_l() will apply new volume even if 0 3968 mLeftVolFloat = mRightVolFloat = -1.0; 3969 if (track->mState == TrackBase::RESUMING) { 3970 track->mState = TrackBase::ACTIVE; 3971 } 3972 } 3973 3974 // compute volume for this track 3975 processVolume_l(track, last); 3976 if (last) { 3977 // reset retry count 3978 track->mRetryCount = kMaxTrackRetriesDirect; 3979 mActiveTrack = t; 3980 mixerStatus = MIXER_TRACKS_READY; 3981 } 3982 } else { 3983 // clear effect chain input buffer if the last active track started underruns 3984 // to avoid sending previous audio buffer again to effects 3985 if (!mEffectChains.isEmpty() && last) { 3986 mEffectChains[0]->clearInputBuffer(); 3987 } 3988 if (track->isStopping_1()) { 3989 track->mState = TrackBase::STOPPING_2; 3990 } 3991 if ((track->sharedBuffer() != 0) || track->isStopped() || 3992 track->isStopping_2() || track->isPaused()) { 3993 // We have consumed all the buffers of this track. 3994 // Remove it from the list of active tracks. 3995 size_t audioHALFrames; 3996 if (audio_is_linear_pcm(mFormat)) { 3997 audioHALFrames = (latency_l() * mSampleRate) / 1000; 3998 } else { 3999 audioHALFrames = 0; 4000 } 4001 4002 size_t framesWritten = mBytesWritten / mFrameSize; 4003 if (mStandby || !last || 4004 track->presentationComplete(framesWritten, audioHALFrames)) { 4005 if (track->isStopping_2()) { 4006 track->mState = TrackBase::STOPPED; 4007 } 4008 if (track->isStopped()) { 4009 track->reset(); 4010 } 4011 tracksToRemove->add(track); 4012 } 4013 } else { 4014 // No buffers for this track. Give it a few chances to 4015 // fill a buffer, then remove it from active list. 4016 // Only consider last track started for mixer state control 4017 if (--(track->mRetryCount) <= 0) { 4018 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4019 tracksToRemove->add(track); 4020 // indicate to client process that the track was disabled because of underrun; 4021 // it will then automatically call start() when data is available 4022 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4023 } else if (last) { 4024 mixerStatus = MIXER_TRACKS_ENABLED; 4025 } 4026 } 4027 } 4028 } 4029 4030 // remove all the tracks that need to be... 4031 removeTracks_l(*tracksToRemove); 4032 4033 return mixerStatus; 4034} 4035 4036void AudioFlinger::DirectOutputThread::threadLoop_mix() 4037{ 4038 size_t frameCount = mFrameCount; 4039 int8_t *curBuf = (int8_t *)mSinkBuffer; 4040 // output audio to hardware 4041 while (frameCount) { 4042 AudioBufferProvider::Buffer buffer; 4043 buffer.frameCount = frameCount; 4044 mActiveTrack->getNextBuffer(&buffer); 4045 if (buffer.raw == NULL) { 4046 memset(curBuf, 0, frameCount * mFrameSize); 4047 break; 4048 } 4049 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4050 frameCount -= buffer.frameCount; 4051 curBuf += buffer.frameCount * mFrameSize; 4052 mActiveTrack->releaseBuffer(&buffer); 4053 } 4054 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4055 sleepTime = 0; 4056 standbyTime = systemTime() + standbyDelay; 4057 mActiveTrack.clear(); 4058} 4059 4060void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4061{ 4062 if (sleepTime == 0) { 4063 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4064 sleepTime = activeSleepTime; 4065 } else { 4066 sleepTime = idleSleepTime; 4067 } 4068 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4069 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4070 sleepTime = 0; 4071 } 4072} 4073 4074// getTrackName_l() must be called with ThreadBase::mLock held 4075int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4076 audio_format_t format __unused, int sessionId __unused) 4077{ 4078 return 0; 4079} 4080 4081// deleteTrackName_l() must be called with ThreadBase::mLock held 4082void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4083{ 4084} 4085 4086// checkForNewParameter_l() must be called with ThreadBase::mLock held 4087bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4088 status_t& status) 4089{ 4090 bool reconfig = false; 4091 4092 status = NO_ERROR; 4093 4094 AudioParameter param = AudioParameter(keyValuePair); 4095 int value; 4096 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4097 // forward device change to effects that have requested to be 4098 // aware of attached audio device. 4099 if (value != AUDIO_DEVICE_NONE) { 4100 mOutDevice = value; 4101 for (size_t i = 0; i < mEffectChains.size(); i++) { 4102 mEffectChains[i]->setDevice_l(mOutDevice); 4103 } 4104 } 4105 } 4106 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4107 // do not accept frame count changes if tracks are open as the track buffer 4108 // size depends on frame count and correct behavior would not be garantied 4109 // if frame count is changed after track creation 4110 if (!mTracks.isEmpty()) { 4111 status = INVALID_OPERATION; 4112 } else { 4113 reconfig = true; 4114 } 4115 } 4116 if (status == NO_ERROR) { 4117 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4118 keyValuePair.string()); 4119 if (!mStandby && status == INVALID_OPERATION) { 4120 mOutput->stream->common.standby(&mOutput->stream->common); 4121 mStandby = true; 4122 mBytesWritten = 0; 4123 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4124 keyValuePair.string()); 4125 } 4126 if (status == NO_ERROR && reconfig) { 4127 readOutputParameters_l(); 4128 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4129 } 4130 } 4131 4132 return reconfig; 4133} 4134 4135uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4136{ 4137 uint32_t time; 4138 if (audio_is_linear_pcm(mFormat)) { 4139 time = PlaybackThread::activeSleepTimeUs(); 4140 } else { 4141 time = 10000; 4142 } 4143 return time; 4144} 4145 4146uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4147{ 4148 uint32_t time; 4149 if (audio_is_linear_pcm(mFormat)) { 4150 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4151 } else { 4152 time = 10000; 4153 } 4154 return time; 4155} 4156 4157uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4158{ 4159 uint32_t time; 4160 if (audio_is_linear_pcm(mFormat)) { 4161 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4162 } else { 4163 time = 10000; 4164 } 4165 return time; 4166} 4167 4168void AudioFlinger::DirectOutputThread::cacheParameters_l() 4169{ 4170 PlaybackThread::cacheParameters_l(); 4171 4172 // use shorter standby delay as on normal output to release 4173 // hardware resources as soon as possible 4174 if (audio_is_linear_pcm(mFormat)) { 4175 standbyDelay = microseconds(activeSleepTime*2); 4176 } else { 4177 standbyDelay = kOffloadStandbyDelayNs; 4178 } 4179} 4180 4181// ---------------------------------------------------------------------------- 4182 4183AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4184 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4185 : Thread(false /*canCallJava*/), 4186 mPlaybackThread(playbackThread), 4187 mWriteAckSequence(0), 4188 mDrainSequence(0) 4189{ 4190} 4191 4192AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4193{ 4194} 4195 4196void AudioFlinger::AsyncCallbackThread::onFirstRef() 4197{ 4198 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4199} 4200 4201bool AudioFlinger::AsyncCallbackThread::threadLoop() 4202{ 4203 while (!exitPending()) { 4204 uint32_t writeAckSequence; 4205 uint32_t drainSequence; 4206 4207 { 4208 Mutex::Autolock _l(mLock); 4209 while (!((mWriteAckSequence & 1) || 4210 (mDrainSequence & 1) || 4211 exitPending())) { 4212 mWaitWorkCV.wait(mLock); 4213 } 4214 4215 if (exitPending()) { 4216 break; 4217 } 4218 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4219 mWriteAckSequence, mDrainSequence); 4220 writeAckSequence = mWriteAckSequence; 4221 mWriteAckSequence &= ~1; 4222 drainSequence = mDrainSequence; 4223 mDrainSequence &= ~1; 4224 } 4225 { 4226 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4227 if (playbackThread != 0) { 4228 if (writeAckSequence & 1) { 4229 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4230 } 4231 if (drainSequence & 1) { 4232 playbackThread->resetDraining(drainSequence >> 1); 4233 } 4234 } 4235 } 4236 } 4237 return false; 4238} 4239 4240void AudioFlinger::AsyncCallbackThread::exit() 4241{ 4242 ALOGV("AsyncCallbackThread::exit"); 4243 Mutex::Autolock _l(mLock); 4244 requestExit(); 4245 mWaitWorkCV.broadcast(); 4246} 4247 4248void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4249{ 4250 Mutex::Autolock _l(mLock); 4251 // bit 0 is cleared 4252 mWriteAckSequence = sequence << 1; 4253} 4254 4255void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4256{ 4257 Mutex::Autolock _l(mLock); 4258 // ignore unexpected callbacks 4259 if (mWriteAckSequence & 2) { 4260 mWriteAckSequence |= 1; 4261 mWaitWorkCV.signal(); 4262 } 4263} 4264 4265void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4266{ 4267 Mutex::Autolock _l(mLock); 4268 // bit 0 is cleared 4269 mDrainSequence = sequence << 1; 4270} 4271 4272void AudioFlinger::AsyncCallbackThread::resetDraining() 4273{ 4274 Mutex::Autolock _l(mLock); 4275 // ignore unexpected callbacks 4276 if (mDrainSequence & 2) { 4277 mDrainSequence |= 1; 4278 mWaitWorkCV.signal(); 4279 } 4280} 4281 4282 4283// ---------------------------------------------------------------------------- 4284AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4285 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4286 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4287 mHwPaused(false), 4288 mFlushPending(false), 4289 mPausedBytesRemaining(0) 4290{ 4291 //FIXME: mStandby should be set to true by ThreadBase constructor 4292 mStandby = true; 4293} 4294 4295void AudioFlinger::OffloadThread::threadLoop_exit() 4296{ 4297 if (mFlushPending || mHwPaused) { 4298 // If a flush is pending or track was paused, just discard buffered data 4299 flushHw_l(); 4300 } else { 4301 mMixerStatus = MIXER_DRAIN_ALL; 4302 threadLoop_drain(); 4303 } 4304 if (mUseAsyncWrite) { 4305 ALOG_ASSERT(mCallbackThread != 0); 4306 mCallbackThread->exit(); 4307 } 4308 PlaybackThread::threadLoop_exit(); 4309} 4310 4311AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4312 Vector< sp<Track> > *tracksToRemove 4313) 4314{ 4315 size_t count = mActiveTracks.size(); 4316 4317 mixer_state mixerStatus = MIXER_IDLE; 4318 bool doHwPause = false; 4319 bool doHwResume = false; 4320 4321 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4322 4323 // find out which tracks need to be processed 4324 for (size_t i = 0; i < count; i++) { 4325 sp<Track> t = mActiveTracks[i].promote(); 4326 // The track died recently 4327 if (t == 0) { 4328 continue; 4329 } 4330 Track* const track = t.get(); 4331 audio_track_cblk_t* cblk = track->cblk(); 4332 // Only consider last track started for volume and mixer state control. 4333 // In theory an older track could underrun and restart after the new one starts 4334 // but as we only care about the transition phase between two tracks on a 4335 // direct output, it is not a problem to ignore the underrun case. 4336 sp<Track> l = mLatestActiveTrack.promote(); 4337 bool last = l.get() == track; 4338 4339 if (track->isInvalid()) { 4340 ALOGW("An invalidated track shouldn't be in active list"); 4341 tracksToRemove->add(track); 4342 continue; 4343 } 4344 4345 if (track->mState == TrackBase::IDLE) { 4346 ALOGW("An idle track shouldn't be in active list"); 4347 continue; 4348 } 4349 4350 if (track->isPausing()) { 4351 track->setPaused(); 4352 if (last) { 4353 if (!mHwPaused) { 4354 doHwPause = true; 4355 mHwPaused = true; 4356 } 4357 // If we were part way through writing the mixbuffer to 4358 // the HAL we must save this until we resume 4359 // BUG - this will be wrong if a different track is made active, 4360 // in that case we want to discard the pending data in the 4361 // mixbuffer and tell the client to present it again when the 4362 // track is resumed 4363 mPausedWriteLength = mCurrentWriteLength; 4364 mPausedBytesRemaining = mBytesRemaining; 4365 mBytesRemaining = 0; // stop writing 4366 } 4367 tracksToRemove->add(track); 4368 } else if (track->isFlushPending()) { 4369 track->flushAck(); 4370 if (last) { 4371 mFlushPending = true; 4372 } 4373 } else if (track->isResumePending()){ 4374 track->resumeAck(); 4375 if (last) { 4376 if (mPausedBytesRemaining) { 4377 // Need to continue write that was interrupted 4378 mCurrentWriteLength = mPausedWriteLength; 4379 mBytesRemaining = mPausedBytesRemaining; 4380 mPausedBytesRemaining = 0; 4381 } 4382 if (mHwPaused) { 4383 doHwResume = true; 4384 mHwPaused = false; 4385 // threadLoop_mix() will handle the case that we need to 4386 // resume an interrupted write 4387 } 4388 // enable write to audio HAL 4389 sleepTime = 0; 4390 4391 // Do not handle new data in this iteration even if track->framesReady() 4392 mixerStatus = MIXER_TRACKS_ENABLED; 4393 } 4394 } else if (track->framesReady() && track->isReady() && 4395 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4396 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4397 if (track->mFillingUpStatus == Track::FS_FILLED) { 4398 track->mFillingUpStatus = Track::FS_ACTIVE; 4399 // make sure processVolume_l() will apply new volume even if 0 4400 mLeftVolFloat = mRightVolFloat = -1.0; 4401 } 4402 4403 if (last) { 4404 sp<Track> previousTrack = mPreviousTrack.promote(); 4405 if (previousTrack != 0) { 4406 if (track != previousTrack.get()) { 4407 // Flush any data still being written from last track 4408 mBytesRemaining = 0; 4409 if (mPausedBytesRemaining) { 4410 // Last track was paused so we also need to flush saved 4411 // mixbuffer state and invalidate track so that it will 4412 // re-submit that unwritten data when it is next resumed 4413 mPausedBytesRemaining = 0; 4414 // Invalidate is a bit drastic - would be more efficient 4415 // to have a flag to tell client that some of the 4416 // previously written data was lost 4417 previousTrack->invalidate(); 4418 } 4419 // flush data already sent to the DSP if changing audio session as audio 4420 // comes from a different source. Also invalidate previous track to force a 4421 // seek when resuming. 4422 if (previousTrack->sessionId() != track->sessionId()) { 4423 previousTrack->invalidate(); 4424 } 4425 } 4426 } 4427 mPreviousTrack = track; 4428 // reset retry count 4429 track->mRetryCount = kMaxTrackRetriesOffload; 4430 mActiveTrack = t; 4431 mixerStatus = MIXER_TRACKS_READY; 4432 } 4433 } else { 4434 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4435 if (track->isStopping_1()) { 4436 // Hardware buffer can hold a large amount of audio so we must 4437 // wait for all current track's data to drain before we say 4438 // that the track is stopped. 4439 if (mBytesRemaining == 0) { 4440 // Only start draining when all data in mixbuffer 4441 // has been written 4442 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4443 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4444 // do not drain if no data was ever sent to HAL (mStandby == true) 4445 if (last && !mStandby) { 4446 // do not modify drain sequence if we are already draining. This happens 4447 // when resuming from pause after drain. 4448 if ((mDrainSequence & 1) == 0) { 4449 sleepTime = 0; 4450 standbyTime = systemTime() + standbyDelay; 4451 mixerStatus = MIXER_DRAIN_TRACK; 4452 mDrainSequence += 2; 4453 } 4454 if (mHwPaused) { 4455 // It is possible to move from PAUSED to STOPPING_1 without 4456 // a resume so we must ensure hardware is running 4457 doHwResume = true; 4458 mHwPaused = false; 4459 } 4460 } 4461 } 4462 } else if (track->isStopping_2()) { 4463 // Drain has completed or we are in standby, signal presentation complete 4464 if (!(mDrainSequence & 1) || !last || mStandby) { 4465 track->mState = TrackBase::STOPPED; 4466 size_t audioHALFrames = 4467 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4468 size_t framesWritten = 4469 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4470 track->presentationComplete(framesWritten, audioHALFrames); 4471 track->reset(); 4472 tracksToRemove->add(track); 4473 } 4474 } else { 4475 // No buffers for this track. Give it a few chances to 4476 // fill a buffer, then remove it from active list. 4477 if (--(track->mRetryCount) <= 0) { 4478 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4479 track->name()); 4480 tracksToRemove->add(track); 4481 // indicate to client process that the track was disabled because of underrun; 4482 // it will then automatically call start() when data is available 4483 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4484 } else if (last){ 4485 mixerStatus = MIXER_TRACKS_ENABLED; 4486 } 4487 } 4488 } 4489 // compute volume for this track 4490 processVolume_l(track, last); 4491 } 4492 4493 // make sure the pause/flush/resume sequence is executed in the right order. 4494 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4495 // before flush and then resume HW. This can happen in case of pause/flush/resume 4496 // if resume is received before pause is executed. 4497 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4498 mOutput->stream->pause(mOutput->stream); 4499 } 4500 if (mFlushPending) { 4501 flushHw_l(); 4502 mFlushPending = false; 4503 } 4504 if (!mStandby && doHwResume) { 4505 mOutput->stream->resume(mOutput->stream); 4506 } 4507 4508 // remove all the tracks that need to be... 4509 removeTracks_l(*tracksToRemove); 4510 4511 return mixerStatus; 4512} 4513 4514// must be called with thread mutex locked 4515bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4516{ 4517 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4518 mWriteAckSequence, mDrainSequence); 4519 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4520 return true; 4521 } 4522 return false; 4523} 4524 4525// must be called with thread mutex locked 4526bool AudioFlinger::OffloadThread::shouldStandby_l() 4527{ 4528 bool trackPaused = false; 4529 4530 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4531 // after a timeout and we will enter standby then. 4532 if (mTracks.size() > 0) { 4533 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4534 } 4535 4536 return !mStandby && !trackPaused; 4537} 4538 4539 4540bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4541{ 4542 Mutex::Autolock _l(mLock); 4543 return waitingAsyncCallback_l(); 4544} 4545 4546void AudioFlinger::OffloadThread::flushHw_l() 4547{ 4548 mOutput->stream->flush(mOutput->stream); 4549 // Flush anything still waiting in the mixbuffer 4550 mCurrentWriteLength = 0; 4551 mBytesRemaining = 0; 4552 mPausedWriteLength = 0; 4553 mPausedBytesRemaining = 0; 4554 mHwPaused = false; 4555 4556 if (mUseAsyncWrite) { 4557 // discard any pending drain or write ack by incrementing sequence 4558 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4559 mDrainSequence = (mDrainSequence + 2) & ~1; 4560 ALOG_ASSERT(mCallbackThread != 0); 4561 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4562 mCallbackThread->setDraining(mDrainSequence); 4563 } 4564} 4565 4566void AudioFlinger::OffloadThread::onAddNewTrack_l() 4567{ 4568 sp<Track> previousTrack = mPreviousTrack.promote(); 4569 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4570 4571 if (previousTrack != 0 && latestTrack != 0 && 4572 (previousTrack->sessionId() != latestTrack->sessionId())) { 4573 mFlushPending = true; 4574 } 4575 PlaybackThread::onAddNewTrack_l(); 4576} 4577 4578// ---------------------------------------------------------------------------- 4579 4580AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4581 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4582 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4583 DUPLICATING), 4584 mWaitTimeMs(UINT_MAX) 4585{ 4586 addOutputTrack(mainThread); 4587} 4588 4589AudioFlinger::DuplicatingThread::~DuplicatingThread() 4590{ 4591 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4592 mOutputTracks[i]->destroy(); 4593 } 4594} 4595 4596void AudioFlinger::DuplicatingThread::threadLoop_mix() 4597{ 4598 // mix buffers... 4599 if (outputsReady(outputTracks)) { 4600 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4601 } else { 4602 memset(mSinkBuffer, 0, mSinkBufferSize); 4603 } 4604 sleepTime = 0; 4605 writeFrames = mNormalFrameCount; 4606 mCurrentWriteLength = mSinkBufferSize; 4607 standbyTime = systemTime() + standbyDelay; 4608} 4609 4610void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4611{ 4612 if (sleepTime == 0) { 4613 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4614 sleepTime = activeSleepTime; 4615 } else { 4616 sleepTime = idleSleepTime; 4617 } 4618 } else if (mBytesWritten != 0) { 4619 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4620 writeFrames = mNormalFrameCount; 4621 memset(mSinkBuffer, 0, mSinkBufferSize); 4622 } else { 4623 // flush remaining overflow buffers in output tracks 4624 writeFrames = 0; 4625 } 4626 sleepTime = 0; 4627 } 4628} 4629 4630ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4631{ 4632 for (size_t i = 0; i < outputTracks.size(); i++) { 4633 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4634 // for delivery downstream as needed. This in-place conversion is safe as 4635 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4636 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4637 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4638 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4639 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4640 } 4641 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4642 } 4643 mStandby = false; 4644 return (ssize_t)mSinkBufferSize; 4645} 4646 4647void AudioFlinger::DuplicatingThread::threadLoop_standby() 4648{ 4649 // DuplicatingThread implements standby by stopping all tracks 4650 for (size_t i = 0; i < outputTracks.size(); i++) { 4651 outputTracks[i]->stop(); 4652 } 4653} 4654 4655void AudioFlinger::DuplicatingThread::saveOutputTracks() 4656{ 4657 outputTracks = mOutputTracks; 4658} 4659 4660void AudioFlinger::DuplicatingThread::clearOutputTracks() 4661{ 4662 outputTracks.clear(); 4663} 4664 4665void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4666{ 4667 Mutex::Autolock _l(mLock); 4668 // FIXME explain this formula 4669 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4670 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4671 // due to current usage case and restrictions on the AudioBufferProvider. 4672 // Actual buffer conversion is done in threadLoop_write(). 4673 // 4674 // TODO: This may change in the future, depending on multichannel 4675 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4676 OutputTrack *outputTrack = new OutputTrack(thread, 4677 this, 4678 mSampleRate, 4679 AUDIO_FORMAT_PCM_16_BIT, 4680 mChannelMask, 4681 frameCount, 4682 IPCThreadState::self()->getCallingUid()); 4683 if (outputTrack->cblk() != NULL) { 4684 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4685 mOutputTracks.add(outputTrack); 4686 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4687 updateWaitTime_l(); 4688 } 4689} 4690 4691void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4692{ 4693 Mutex::Autolock _l(mLock); 4694 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4695 if (mOutputTracks[i]->thread() == thread) { 4696 mOutputTracks[i]->destroy(); 4697 mOutputTracks.removeAt(i); 4698 updateWaitTime_l(); 4699 return; 4700 } 4701 } 4702 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4703} 4704 4705// caller must hold mLock 4706void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4707{ 4708 mWaitTimeMs = UINT_MAX; 4709 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4710 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4711 if (strong != 0) { 4712 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4713 if (waitTimeMs < mWaitTimeMs) { 4714 mWaitTimeMs = waitTimeMs; 4715 } 4716 } 4717 } 4718} 4719 4720 4721bool AudioFlinger::DuplicatingThread::outputsReady( 4722 const SortedVector< sp<OutputTrack> > &outputTracks) 4723{ 4724 for (size_t i = 0; i < outputTracks.size(); i++) { 4725 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4726 if (thread == 0) { 4727 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4728 outputTracks[i].get()); 4729 return false; 4730 } 4731 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4732 // see note at standby() declaration 4733 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4734 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4735 thread.get()); 4736 return false; 4737 } 4738 } 4739 return true; 4740} 4741 4742uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4743{ 4744 return (mWaitTimeMs * 1000) / 2; 4745} 4746 4747void AudioFlinger::DuplicatingThread::cacheParameters_l() 4748{ 4749 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4750 updateWaitTime_l(); 4751 4752 MixerThread::cacheParameters_l(); 4753} 4754 4755// ---------------------------------------------------------------------------- 4756// Record 4757// ---------------------------------------------------------------------------- 4758 4759AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4760 AudioStreamIn *input, 4761 audio_io_handle_t id, 4762 audio_devices_t outDevice, 4763 audio_devices_t inDevice 4764#ifdef TEE_SINK 4765 , const sp<NBAIO_Sink>& teeSink 4766#endif 4767 ) : 4768 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4769 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4770 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4771 mRsmpInRear(0) 4772#ifdef TEE_SINK 4773 , mTeeSink(teeSink) 4774#endif 4775 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4776 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4777 // mFastCapture below 4778 , mFastCaptureFutex(0) 4779 // mInputSource 4780 // mPipeSink 4781 // mPipeSource 4782 , mPipeFramesP2(0) 4783 // mPipeMemory 4784 // mFastCaptureNBLogWriter 4785 , mFastTrackAvail(true) 4786{ 4787 snprintf(mName, kNameLength, "AudioIn_%X", id); 4788 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4789 4790 readInputParameters_l(); 4791 4792 // create an NBAIO source for the HAL input stream, and negotiate 4793 mInputSource = new AudioStreamInSource(input->stream); 4794 size_t numCounterOffers = 0; 4795 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4796 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4797 ALOG_ASSERT(index == 0); 4798 4799 // initialize fast capture depending on configuration 4800 bool initFastCapture; 4801 switch (kUseFastCapture) { 4802 case FastCapture_Never: 4803 initFastCapture = false; 4804 break; 4805 case FastCapture_Always: 4806 initFastCapture = true; 4807 break; 4808 case FastCapture_Static: 4809 uint32_t primaryOutputSampleRate; 4810 { 4811 AutoMutex _l(audioFlinger->mHardwareLock); 4812 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4813 } 4814 initFastCapture = 4815 // either capture sample rate is same as (a reasonable) primary output sample rate 4816 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4817 (mSampleRate == primaryOutputSampleRate)) || 4818 // or primary output sample rate is unknown, and capture sample rate is reasonable 4819 ((primaryOutputSampleRate == 0) && 4820 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4821 // and the buffer size is < 10 ms 4822 (mFrameCount * 1000) / mSampleRate < 10; 4823 break; 4824 // case FastCapture_Dynamic: 4825 } 4826 4827 if (initFastCapture) { 4828 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4829 NBAIO_Format format = mInputSource->format(); 4830 size_t pipeFramesP2 = roundup(mFrameCount * 8); 4831 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4832 void *pipeBuffer; 4833 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4834 sp<IMemory> pipeMemory; 4835 if ((roHeap == 0) || 4836 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4837 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4838 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4839 goto failed; 4840 } 4841 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4842 memset(pipeBuffer, 0, pipeSize); 4843 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4844 const NBAIO_Format offers[1] = {format}; 4845 size_t numCounterOffers = 0; 4846 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4847 ALOG_ASSERT(index == 0); 4848 mPipeSink = pipe; 4849 PipeReader *pipeReader = new PipeReader(*pipe); 4850 numCounterOffers = 0; 4851 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4852 ALOG_ASSERT(index == 0); 4853 mPipeSource = pipeReader; 4854 mPipeFramesP2 = pipeFramesP2; 4855 mPipeMemory = pipeMemory; 4856 4857 // create fast capture 4858 mFastCapture = new FastCapture(); 4859 FastCaptureStateQueue *sq = mFastCapture->sq(); 4860#ifdef STATE_QUEUE_DUMP 4861 // FIXME 4862#endif 4863 FastCaptureState *state = sq->begin(); 4864 state->mCblk = NULL; 4865 state->mInputSource = mInputSource.get(); 4866 state->mInputSourceGen++; 4867 state->mPipeSink = pipe; 4868 state->mPipeSinkGen++; 4869 state->mFrameCount = mFrameCount; 4870 state->mCommand = FastCaptureState::COLD_IDLE; 4871 // already done in constructor initialization list 4872 //mFastCaptureFutex = 0; 4873 state->mColdFutexAddr = &mFastCaptureFutex; 4874 state->mColdGen++; 4875 state->mDumpState = &mFastCaptureDumpState; 4876#ifdef TEE_SINK 4877 // FIXME 4878#endif 4879 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4880 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4881 sq->end(); 4882 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4883 4884 // start the fast capture 4885 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4886 pid_t tid = mFastCapture->getTid(); 4887 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4888 if (err != 0) { 4889 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4890 kPriorityFastCapture, getpid_cached, tid, err); 4891 } 4892 4893#ifdef AUDIO_WATCHDOG 4894 // FIXME 4895#endif 4896 4897 } 4898failed: ; 4899 4900 // FIXME mNormalSource 4901} 4902 4903 4904AudioFlinger::RecordThread::~RecordThread() 4905{ 4906 if (mFastCapture != 0) { 4907 FastCaptureStateQueue *sq = mFastCapture->sq(); 4908 FastCaptureState *state = sq->begin(); 4909 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4910 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4911 if (old == -1) { 4912 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4913 } 4914 } 4915 state->mCommand = FastCaptureState::EXIT; 4916 sq->end(); 4917 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4918 mFastCapture->join(); 4919 mFastCapture.clear(); 4920 } 4921 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4922 mAudioFlinger->unregisterWriter(mNBLogWriter); 4923 delete[] mRsmpInBuffer; 4924} 4925 4926void AudioFlinger::RecordThread::onFirstRef() 4927{ 4928 run(mName, PRIORITY_URGENT_AUDIO); 4929} 4930 4931bool AudioFlinger::RecordThread::threadLoop() 4932{ 4933 nsecs_t lastWarning = 0; 4934 4935 inputStandBy(); 4936 4937reacquire_wakelock: 4938 sp<RecordTrack> activeTrack; 4939 int activeTracksGen; 4940 { 4941 Mutex::Autolock _l(mLock); 4942 size_t size = mActiveTracks.size(); 4943 activeTracksGen = mActiveTracksGen; 4944 if (size > 0) { 4945 // FIXME an arbitrary choice 4946 activeTrack = mActiveTracks[0]; 4947 acquireWakeLock_l(activeTrack->uid()); 4948 if (size > 1) { 4949 SortedVector<int> tmp; 4950 for (size_t i = 0; i < size; i++) { 4951 tmp.add(mActiveTracks[i]->uid()); 4952 } 4953 updateWakeLockUids_l(tmp); 4954 } 4955 } else { 4956 acquireWakeLock_l(-1); 4957 } 4958 } 4959 4960 // used to request a deferred sleep, to be executed later while mutex is unlocked 4961 uint32_t sleepUs = 0; 4962 4963 // loop while there is work to do 4964 for (;;) { 4965 Vector< sp<EffectChain> > effectChains; 4966 4967 // sleep with mutex unlocked 4968 if (sleepUs > 0) { 4969 usleep(sleepUs); 4970 sleepUs = 0; 4971 } 4972 4973 // activeTracks accumulates a copy of a subset of mActiveTracks 4974 Vector< sp<RecordTrack> > activeTracks; 4975 4976 // reference to the (first and only) fast track 4977 sp<RecordTrack> fastTrack; 4978 4979 { // scope for mLock 4980 Mutex::Autolock _l(mLock); 4981 4982 processConfigEvents_l(); 4983 4984 // check exitPending here because checkForNewParameters_l() and 4985 // checkForNewParameters_l() can temporarily release mLock 4986 if (exitPending()) { 4987 break; 4988 } 4989 4990 // if no active track(s), then standby and release wakelock 4991 size_t size = mActiveTracks.size(); 4992 if (size == 0) { 4993 standbyIfNotAlreadyInStandby(); 4994 // exitPending() can't become true here 4995 releaseWakeLock_l(); 4996 ALOGV("RecordThread: loop stopping"); 4997 // go to sleep 4998 mWaitWorkCV.wait(mLock); 4999 ALOGV("RecordThread: loop starting"); 5000 goto reacquire_wakelock; 5001 } 5002 5003 if (mActiveTracksGen != activeTracksGen) { 5004 activeTracksGen = mActiveTracksGen; 5005 SortedVector<int> tmp; 5006 for (size_t i = 0; i < size; i++) { 5007 tmp.add(mActiveTracks[i]->uid()); 5008 } 5009 updateWakeLockUids_l(tmp); 5010 } 5011 5012 bool doBroadcast = false; 5013 for (size_t i = 0; i < size; ) { 5014 5015 activeTrack = mActiveTracks[i]; 5016 if (activeTrack->isTerminated()) { 5017 removeTrack_l(activeTrack); 5018 mActiveTracks.remove(activeTrack); 5019 mActiveTracksGen++; 5020 size--; 5021 continue; 5022 } 5023 5024 TrackBase::track_state activeTrackState = activeTrack->mState; 5025 switch (activeTrackState) { 5026 5027 case TrackBase::PAUSING: 5028 mActiveTracks.remove(activeTrack); 5029 mActiveTracksGen++; 5030 doBroadcast = true; 5031 size--; 5032 continue; 5033 5034 case TrackBase::STARTING_1: 5035 sleepUs = 10000; 5036 i++; 5037 continue; 5038 5039 case TrackBase::STARTING_2: 5040 doBroadcast = true; 5041 mStandby = false; 5042 activeTrack->mState = TrackBase::ACTIVE; 5043 break; 5044 5045 case TrackBase::ACTIVE: 5046 break; 5047 5048 case TrackBase::IDLE: 5049 i++; 5050 continue; 5051 5052 default: 5053 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5054 } 5055 5056 activeTracks.add(activeTrack); 5057 i++; 5058 5059 if (activeTrack->isFastTrack()) { 5060 ALOG_ASSERT(!mFastTrackAvail); 5061 ALOG_ASSERT(fastTrack == 0); 5062 fastTrack = activeTrack; 5063 } 5064 } 5065 if (doBroadcast) { 5066 mStartStopCond.broadcast(); 5067 } 5068 5069 // sleep if there are no active tracks to process 5070 if (activeTracks.size() == 0) { 5071 if (sleepUs == 0) { 5072 sleepUs = kRecordThreadSleepUs; 5073 } 5074 continue; 5075 } 5076 sleepUs = 0; 5077 5078 lockEffectChains_l(effectChains); 5079 } 5080 5081 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5082 5083 size_t size = effectChains.size(); 5084 for (size_t i = 0; i < size; i++) { 5085 // thread mutex is not locked, but effect chain is locked 5086 effectChains[i]->process_l(); 5087 } 5088 5089 // Start the fast capture if it's not already running 5090 if (mFastCapture != 0) { 5091 FastCaptureStateQueue *sq = mFastCapture->sq(); 5092 FastCaptureState *state = sq->begin(); 5093 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5094 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5095 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5096 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5097 if (old == -1) { 5098 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5099 } 5100 } 5101 state->mCommand = FastCaptureState::READ_WRITE; 5102#if 0 // FIXME 5103 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5104 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5105#endif 5106 state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL; 5107 sq->end(); 5108 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5109#if 0 5110 if (kUseFastCapture == FastCapture_Dynamic) { 5111 mNormalSource = mPipeSource; 5112 } 5113#endif 5114 } else { 5115 sq->end(false /*didModify*/); 5116 } 5117 } 5118 5119 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5120 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5121 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5122 // If destination is non-contiguous, first read past the nominal end of buffer, then 5123 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5124 5125 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5126 ssize_t framesRead; 5127 5128 // If an NBAIO source is present, use it to read the normal capture's data 5129 if (mPipeSource != 0) { 5130 size_t framesToRead = mBufferSize / mFrameSize; 5131 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5132 framesToRead, AudioBufferProvider::kInvalidPTS); 5133 if (framesRead == 0) { 5134 // since pipe is non-blocking, simulate blocking input 5135 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5136 } 5137 // otherwise use the HAL / AudioStreamIn directly 5138 } else { 5139 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5140 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5141 if (bytesRead < 0) { 5142 framesRead = bytesRead; 5143 } else { 5144 framesRead = bytesRead / mFrameSize; 5145 } 5146 } 5147 5148 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5149 ALOGE("read failed: framesRead=%d", framesRead); 5150 // Force input into standby so that it tries to recover at next read attempt 5151 inputStandBy(); 5152 sleepUs = kRecordThreadSleepUs; 5153 } 5154 if (framesRead <= 0) { 5155 goto unlock; 5156 } 5157 ALOG_ASSERT(framesRead > 0); 5158 5159 if (mTeeSink != 0) { 5160 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5161 } 5162 // If destination is non-contiguous, we now correct for reading past end of buffer. 5163 { 5164 size_t part1 = mRsmpInFramesP2 - rear; 5165 if ((size_t) framesRead > part1) { 5166 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5167 (framesRead - part1) * mFrameSize); 5168 } 5169 } 5170 rear = mRsmpInRear += framesRead; 5171 5172 size = activeTracks.size(); 5173 // loop over each active track 5174 for (size_t i = 0; i < size; i++) { 5175 activeTrack = activeTracks[i]; 5176 5177 // skip fast tracks, as those are handled directly by FastCapture 5178 if (activeTrack->isFastTrack()) { 5179 continue; 5180 } 5181 5182 enum { 5183 OVERRUN_UNKNOWN, 5184 OVERRUN_TRUE, 5185 OVERRUN_FALSE 5186 } overrun = OVERRUN_UNKNOWN; 5187 5188 // loop over getNextBuffer to handle circular sink 5189 for (;;) { 5190 5191 activeTrack->mSink.frameCount = ~0; 5192 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5193 size_t framesOut = activeTrack->mSink.frameCount; 5194 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5195 5196 int32_t front = activeTrack->mRsmpInFront; 5197 ssize_t filled = rear - front; 5198 size_t framesIn; 5199 5200 if (filled < 0) { 5201 // should not happen, but treat like a massive overrun and re-sync 5202 framesIn = 0; 5203 activeTrack->mRsmpInFront = rear; 5204 overrun = OVERRUN_TRUE; 5205 } else if ((size_t) filled <= mRsmpInFrames) { 5206 framesIn = (size_t) filled; 5207 } else { 5208 // client is not keeping up with server, but give it latest data 5209 framesIn = mRsmpInFrames; 5210 activeTrack->mRsmpInFront = front = rear - framesIn; 5211 overrun = OVERRUN_TRUE; 5212 } 5213 5214 if (framesOut == 0 || framesIn == 0) { 5215 break; 5216 } 5217 5218 if (activeTrack->mResampler == NULL) { 5219 // no resampling 5220 if (framesIn > framesOut) { 5221 framesIn = framesOut; 5222 } else { 5223 framesOut = framesIn; 5224 } 5225 int8_t *dst = activeTrack->mSink.i8; 5226 while (framesIn > 0) { 5227 front &= mRsmpInFramesP2 - 1; 5228 size_t part1 = mRsmpInFramesP2 - front; 5229 if (part1 > framesIn) { 5230 part1 = framesIn; 5231 } 5232 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5233 if (mChannelCount == activeTrack->mChannelCount) { 5234 memcpy(dst, src, part1 * mFrameSize); 5235 } else if (mChannelCount == 1) { 5236 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 5237 part1); 5238 } else { 5239 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 5240 part1); 5241 } 5242 dst += part1 * activeTrack->mFrameSize; 5243 front += part1; 5244 framesIn -= part1; 5245 } 5246 activeTrack->mRsmpInFront += framesOut; 5247 5248 } else { 5249 // resampling 5250 // FIXME framesInNeeded should really be part of resampler API, and should 5251 // depend on the SRC ratio 5252 // to keep mRsmpInBuffer full so resampler always has sufficient input 5253 size_t framesInNeeded; 5254 // FIXME only re-calculate when it changes, and optimize for common ratios 5255 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 5256 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 5257 framesInNeeded = ceil(framesOut * inOverOut) + 1; 5258 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5259 framesInNeeded, framesOut, inOverOut); 5260 // Although we theoretically have framesIn in circular buffer, some of those are 5261 // unreleased frames, and thus must be discounted for purpose of budgeting. 5262 size_t unreleased = activeTrack->mRsmpInUnrel; 5263 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5264 if (framesIn < framesInNeeded) { 5265 ALOGV("not enough to resample: have %u frames in but need %u in to " 5266 "produce %u out given in/out ratio of %.4g", 5267 framesIn, framesInNeeded, framesOut, inOverOut); 5268 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 5269 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5270 if (newFramesOut == 0) { 5271 break; 5272 } 5273 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 5274 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5275 framesInNeeded, newFramesOut, outOverIn); 5276 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5277 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5278 "given in/out ratio of %.4g", 5279 framesIn, framesInNeeded, newFramesOut, inOverOut); 5280 framesOut = newFramesOut; 5281 } else { 5282 ALOGV("success 1: have %u in and need %u in to produce %u out " 5283 "given in/out ratio of %.4g", 5284 framesIn, framesInNeeded, framesOut, inOverOut); 5285 } 5286 5287 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5288 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5289 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5290 delete[] activeTrack->mRsmpOutBuffer; 5291 // resampler always outputs stereo 5292 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5293 activeTrack->mRsmpOutFrameCount = framesOut; 5294 } 5295 5296 // resampler accumulates, but we only have one source track 5297 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5298 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5299 // FIXME how about having activeTrack implement this interface itself? 5300 activeTrack->mResamplerBufferProvider 5301 /*this*/ /* AudioBufferProvider* */); 5302 // ditherAndClamp() works as long as all buffers returned by 5303 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5304 if (activeTrack->mChannelCount == 1) { 5305 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5306 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5307 framesOut); 5308 // the resampler always outputs stereo samples: 5309 // do post stereo to mono conversion 5310 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5311 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5312 } else { 5313 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5314 activeTrack->mRsmpOutBuffer, framesOut); 5315 } 5316 // now done with mRsmpOutBuffer 5317 5318 } 5319 5320 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5321 overrun = OVERRUN_FALSE; 5322 } 5323 5324 if (activeTrack->mFramesToDrop == 0) { 5325 if (framesOut > 0) { 5326 activeTrack->mSink.frameCount = framesOut; 5327 activeTrack->releaseBuffer(&activeTrack->mSink); 5328 } 5329 } else { 5330 // FIXME could do a partial drop of framesOut 5331 if (activeTrack->mFramesToDrop > 0) { 5332 activeTrack->mFramesToDrop -= framesOut; 5333 if (activeTrack->mFramesToDrop <= 0) { 5334 activeTrack->clearSyncStartEvent(); 5335 } 5336 } else { 5337 activeTrack->mFramesToDrop += framesOut; 5338 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5339 activeTrack->mSyncStartEvent->isCancelled()) { 5340 ALOGW("Synced record %s, session %d, trigger session %d", 5341 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5342 activeTrack->sessionId(), 5343 (activeTrack->mSyncStartEvent != 0) ? 5344 activeTrack->mSyncStartEvent->triggerSession() : 0); 5345 activeTrack->clearSyncStartEvent(); 5346 } 5347 } 5348 } 5349 5350 if (framesOut == 0) { 5351 break; 5352 } 5353 } 5354 5355 switch (overrun) { 5356 case OVERRUN_TRUE: 5357 // client isn't retrieving buffers fast enough 5358 if (!activeTrack->setOverflow()) { 5359 nsecs_t now = systemTime(); 5360 // FIXME should lastWarning per track? 5361 if ((now - lastWarning) > kWarningThrottleNs) { 5362 ALOGW("RecordThread: buffer overflow"); 5363 lastWarning = now; 5364 } 5365 } 5366 break; 5367 case OVERRUN_FALSE: 5368 activeTrack->clearOverflow(); 5369 break; 5370 case OVERRUN_UNKNOWN: 5371 break; 5372 } 5373 5374 } 5375 5376unlock: 5377 // enable changes in effect chain 5378 unlockEffectChains(effectChains); 5379 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5380 } 5381 5382 standbyIfNotAlreadyInStandby(); 5383 5384 { 5385 Mutex::Autolock _l(mLock); 5386 for (size_t i = 0; i < mTracks.size(); i++) { 5387 sp<RecordTrack> track = mTracks[i]; 5388 track->invalidate(); 5389 } 5390 mActiveTracks.clear(); 5391 mActiveTracksGen++; 5392 mStartStopCond.broadcast(); 5393 } 5394 5395 releaseWakeLock(); 5396 5397 ALOGV("RecordThread %p exiting", this); 5398 return false; 5399} 5400 5401void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5402{ 5403 if (!mStandby) { 5404 inputStandBy(); 5405 mStandby = true; 5406 } 5407} 5408 5409void AudioFlinger::RecordThread::inputStandBy() 5410{ 5411 // Idle the fast capture if it's currently running 5412 if (mFastCapture != 0) { 5413 FastCaptureStateQueue *sq = mFastCapture->sq(); 5414 FastCaptureState *state = sq->begin(); 5415 if (!(state->mCommand & FastCaptureState::IDLE)) { 5416 state->mCommand = FastCaptureState::COLD_IDLE; 5417 state->mColdFutexAddr = &mFastCaptureFutex; 5418 state->mColdGen++; 5419 mFastCaptureFutex = 0; 5420 sq->end(); 5421 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5422 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5423#if 0 5424 if (kUseFastCapture == FastCapture_Dynamic) { 5425 // FIXME 5426 } 5427#endif 5428#ifdef AUDIO_WATCHDOG 5429 // FIXME 5430#endif 5431 } else { 5432 sq->end(false /*didModify*/); 5433 } 5434 } 5435 mInput->stream->common.standby(&mInput->stream->common); 5436} 5437 5438// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5439sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5440 const sp<AudioFlinger::Client>& client, 5441 uint32_t sampleRate, 5442 audio_format_t format, 5443 audio_channel_mask_t channelMask, 5444 size_t *pFrameCount, 5445 int sessionId, 5446 size_t *notificationFrames, 5447 int uid, 5448 IAudioFlinger::track_flags_t *flags, 5449 pid_t tid, 5450 status_t *status) 5451{ 5452 size_t frameCount = *pFrameCount; 5453 sp<RecordTrack> track; 5454 status_t lStatus; 5455 5456 // client expresses a preference for FAST, but we get the final say 5457 if (*flags & IAudioFlinger::TRACK_FAST) { 5458 if ( 5459 // use case: callback handler and frame count is default or at least as large as HAL 5460 ( 5461 (tid != -1) && 5462 ((frameCount == 0) /*|| 5463 // FIXME must be equal to pipe depth, so don't allow it to be specified by client 5464 // FIXME not necessarily true, should be native frame count for native SR! 5465 (frameCount >= mFrameCount)*/) 5466 ) && 5467 // PCM data 5468 audio_is_linear_pcm(format) && 5469 // native format 5470 (format == mFormat) && 5471 // mono or stereo 5472 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5473 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5474 // native channel mask 5475 (channelMask == mChannelMask) && 5476 // native hardware sample rate 5477 (sampleRate == mSampleRate) && 5478 // record thread has an associated fast capture 5479 hasFastCapture() && 5480 // there are sufficient fast track slots available 5481 mFastTrackAvail 5482 ) { 5483 // if frameCount not specified, then it defaults to pipe frame count 5484 if (frameCount == 0) { 5485 frameCount = mPipeFramesP2; 5486 } 5487 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5488 frameCount, mFrameCount); 5489 } else { 5490 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5491 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5492 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5493 frameCount, mFrameCount, format, 5494 audio_is_linear_pcm(format), 5495 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid, mFastTrackAvail); 5496 *flags &= ~IAudioFlinger::TRACK_FAST; 5497 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5498 // For compatibility with AudioRecord calculation, buffer depth is forced 5499 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5500 // This is probably too conservative, but legacy application code may depend on it. 5501 // If you change this calculation, also review the start threshold which is related. 5502 // FIXME It's not clear how input latency actually matters. Perhaps this should be 0. 5503 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5504 size_t mNormalFrameCount = 2048; // FIXME 5505 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5506 if (minBufCount < 2) { 5507 minBufCount = 2; 5508 } 5509 size_t minFrameCount = mNormalFrameCount * minBufCount; 5510 if (frameCount < minFrameCount) { 5511 frameCount = minFrameCount; 5512 } 5513 } 5514 } 5515 *pFrameCount = frameCount; 5516 *notificationFrames = 0; // FIXME implement 5517 5518 lStatus = initCheck(); 5519 if (lStatus != NO_ERROR) { 5520 ALOGE("createRecordTrack_l() audio driver not initialized"); 5521 goto Exit; 5522 } 5523 5524 { // scope for mLock 5525 Mutex::Autolock _l(mLock); 5526 5527 track = new RecordTrack(this, client, sampleRate, 5528 format, channelMask, frameCount, sessionId, uid, 5529 *flags); 5530 5531 lStatus = track->initCheck(); 5532 if (lStatus != NO_ERROR) { 5533 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5534 // track must be cleared from the caller as the caller has the AF lock 5535 goto Exit; 5536 } 5537 mTracks.add(track); 5538 5539 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5540 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5541 mAudioFlinger->btNrecIsOff(); 5542 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5543 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5544 5545 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5546 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5547 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5548 // so ask activity manager to do this on our behalf 5549 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5550 } 5551 } 5552 5553 lStatus = NO_ERROR; 5554 5555Exit: 5556 *status = lStatus; 5557 return track; 5558} 5559 5560status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5561 AudioSystem::sync_event_t event, 5562 int triggerSession) 5563{ 5564 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5565 sp<ThreadBase> strongMe = this; 5566 status_t status = NO_ERROR; 5567 5568 if (event == AudioSystem::SYNC_EVENT_NONE) { 5569 recordTrack->clearSyncStartEvent(); 5570 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5571 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5572 triggerSession, 5573 recordTrack->sessionId(), 5574 syncStartEventCallback, 5575 recordTrack); 5576 // Sync event can be cancelled by the trigger session if the track is not in a 5577 // compatible state in which case we start record immediately 5578 if (recordTrack->mSyncStartEvent->isCancelled()) { 5579 recordTrack->clearSyncStartEvent(); 5580 } else { 5581 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5582 recordTrack->mFramesToDrop = - 5583 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5584 } 5585 } 5586 5587 { 5588 // This section is a rendezvous between binder thread executing start() and RecordThread 5589 AutoMutex lock(mLock); 5590 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5591 if (recordTrack->mState == TrackBase::PAUSING) { 5592 ALOGV("active record track PAUSING -> ACTIVE"); 5593 recordTrack->mState = TrackBase::ACTIVE; 5594 } else { 5595 ALOGV("active record track state %d", recordTrack->mState); 5596 } 5597 return status; 5598 } 5599 5600 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5601 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5602 // or using a separate command thread 5603 recordTrack->mState = TrackBase::STARTING_1; 5604 mActiveTracks.add(recordTrack); 5605 mActiveTracksGen++; 5606 mLock.unlock(); 5607 status_t status = AudioSystem::startInput(mId); 5608 mLock.lock(); 5609 // FIXME should verify that recordTrack is still in mActiveTracks 5610 if (status != NO_ERROR) { 5611 mActiveTracks.remove(recordTrack); 5612 mActiveTracksGen++; 5613 recordTrack->clearSyncStartEvent(); 5614 return status; 5615 } 5616 // Catch up with current buffer indices if thread is already running. 5617 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5618 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5619 // see previously buffered data before it called start(), but with greater risk of overrun. 5620 5621 recordTrack->mRsmpInFront = mRsmpInRear; 5622 recordTrack->mRsmpInUnrel = 0; 5623 // FIXME why reset? 5624 if (recordTrack->mResampler != NULL) { 5625 recordTrack->mResampler->reset(); 5626 } 5627 recordTrack->mState = TrackBase::STARTING_2; 5628 // signal thread to start 5629 mWaitWorkCV.broadcast(); 5630 if (mActiveTracks.indexOf(recordTrack) < 0) { 5631 ALOGV("Record failed to start"); 5632 status = BAD_VALUE; 5633 goto startError; 5634 } 5635 return status; 5636 } 5637 5638startError: 5639 AudioSystem::stopInput(mId); 5640 recordTrack->clearSyncStartEvent(); 5641 // FIXME I wonder why we do not reset the state here? 5642 return status; 5643} 5644 5645void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5646{ 5647 sp<SyncEvent> strongEvent = event.promote(); 5648 5649 if (strongEvent != 0) { 5650 sp<RefBase> ptr = strongEvent->cookie().promote(); 5651 if (ptr != 0) { 5652 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5653 recordTrack->handleSyncStartEvent(strongEvent); 5654 } 5655 } 5656} 5657 5658bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5659 ALOGV("RecordThread::stop"); 5660 AutoMutex _l(mLock); 5661 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5662 return false; 5663 } 5664 // note that threadLoop may still be processing the track at this point [without lock] 5665 recordTrack->mState = TrackBase::PAUSING; 5666 // do not wait for mStartStopCond if exiting 5667 if (exitPending()) { 5668 return true; 5669 } 5670 // FIXME incorrect usage of wait: no explicit predicate or loop 5671 mStartStopCond.wait(mLock); 5672 // if we have been restarted, recordTrack is in mActiveTracks here 5673 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5674 ALOGV("Record stopped OK"); 5675 return true; 5676 } 5677 return false; 5678} 5679 5680bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5681{ 5682 return false; 5683} 5684 5685status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5686{ 5687#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5688 if (!isValidSyncEvent(event)) { 5689 return BAD_VALUE; 5690 } 5691 5692 int eventSession = event->triggerSession(); 5693 status_t ret = NAME_NOT_FOUND; 5694 5695 Mutex::Autolock _l(mLock); 5696 5697 for (size_t i = 0; i < mTracks.size(); i++) { 5698 sp<RecordTrack> track = mTracks[i]; 5699 if (eventSession == track->sessionId()) { 5700 (void) track->setSyncEvent(event); 5701 ret = NO_ERROR; 5702 } 5703 } 5704 return ret; 5705#else 5706 return BAD_VALUE; 5707#endif 5708} 5709 5710// destroyTrack_l() must be called with ThreadBase::mLock held 5711void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5712{ 5713 track->terminate(); 5714 track->mState = TrackBase::STOPPED; 5715 // active tracks are removed by threadLoop() 5716 if (mActiveTracks.indexOf(track) < 0) { 5717 removeTrack_l(track); 5718 } 5719} 5720 5721void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5722{ 5723 mTracks.remove(track); 5724 // need anything related to effects here? 5725 if (track->isFastTrack()) { 5726 ALOG_ASSERT(!mFastTrackAvail); 5727 mFastTrackAvail = true; 5728 } 5729} 5730 5731void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5732{ 5733 dumpInternals(fd, args); 5734 dumpTracks(fd, args); 5735 dumpEffectChains(fd, args); 5736} 5737 5738void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5739{ 5740 dprintf(fd, "\nInput thread %p:\n", this); 5741 5742 if (mActiveTracks.size() > 0) { 5743 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5744 } else { 5745 dprintf(fd, " No active record clients\n"); 5746 } 5747 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5748 5749 dumpBase(fd, args); 5750} 5751 5752void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5753{ 5754 const size_t SIZE = 256; 5755 char buffer[SIZE]; 5756 String8 result; 5757 5758 size_t numtracks = mTracks.size(); 5759 size_t numactive = mActiveTracks.size(); 5760 size_t numactiveseen = 0; 5761 dprintf(fd, " %d Tracks", numtracks); 5762 if (numtracks) { 5763 dprintf(fd, " of which %d are active\n", numactive); 5764 RecordTrack::appendDumpHeader(result); 5765 for (size_t i = 0; i < numtracks ; ++i) { 5766 sp<RecordTrack> track = mTracks[i]; 5767 if (track != 0) { 5768 bool active = mActiveTracks.indexOf(track) >= 0; 5769 if (active) { 5770 numactiveseen++; 5771 } 5772 track->dump(buffer, SIZE, active); 5773 result.append(buffer); 5774 } 5775 } 5776 } else { 5777 dprintf(fd, "\n"); 5778 } 5779 5780 if (numactiveseen != numactive) { 5781 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5782 " not in the track list\n"); 5783 result.append(buffer); 5784 RecordTrack::appendDumpHeader(result); 5785 for (size_t i = 0; i < numactive; ++i) { 5786 sp<RecordTrack> track = mActiveTracks[i]; 5787 if (mTracks.indexOf(track) < 0) { 5788 track->dump(buffer, SIZE, true); 5789 result.append(buffer); 5790 } 5791 } 5792 5793 } 5794 write(fd, result.string(), result.size()); 5795} 5796 5797// AudioBufferProvider interface 5798status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5799 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5800{ 5801 RecordTrack *activeTrack = mRecordTrack; 5802 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5803 if (threadBase == 0) { 5804 buffer->frameCount = 0; 5805 buffer->raw = NULL; 5806 return NOT_ENOUGH_DATA; 5807 } 5808 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5809 int32_t rear = recordThread->mRsmpInRear; 5810 int32_t front = activeTrack->mRsmpInFront; 5811 ssize_t filled = rear - front; 5812 // FIXME should not be P2 (don't want to increase latency) 5813 // FIXME if client not keeping up, discard 5814 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5815 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5816 front &= recordThread->mRsmpInFramesP2 - 1; 5817 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5818 if (part1 > (size_t) filled) { 5819 part1 = filled; 5820 } 5821 size_t ask = buffer->frameCount; 5822 ALOG_ASSERT(ask > 0); 5823 if (part1 > ask) { 5824 part1 = ask; 5825 } 5826 if (part1 == 0) { 5827 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5828 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5829 buffer->raw = NULL; 5830 buffer->frameCount = 0; 5831 activeTrack->mRsmpInUnrel = 0; 5832 return NOT_ENOUGH_DATA; 5833 } 5834 5835 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5836 buffer->frameCount = part1; 5837 activeTrack->mRsmpInUnrel = part1; 5838 return NO_ERROR; 5839} 5840 5841// AudioBufferProvider interface 5842void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5843 AudioBufferProvider::Buffer* buffer) 5844{ 5845 RecordTrack *activeTrack = mRecordTrack; 5846 size_t stepCount = buffer->frameCount; 5847 if (stepCount == 0) { 5848 return; 5849 } 5850 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5851 activeTrack->mRsmpInUnrel -= stepCount; 5852 activeTrack->mRsmpInFront += stepCount; 5853 buffer->raw = NULL; 5854 buffer->frameCount = 0; 5855} 5856 5857bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5858 status_t& status) 5859{ 5860 bool reconfig = false; 5861 5862 status = NO_ERROR; 5863 5864 audio_format_t reqFormat = mFormat; 5865 uint32_t samplingRate = mSampleRate; 5866 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5867 5868 AudioParameter param = AudioParameter(keyValuePair); 5869 int value; 5870 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5871 // channel count change can be requested. Do we mandate the first client defines the 5872 // HAL sampling rate and channel count or do we allow changes on the fly? 5873 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5874 samplingRate = value; 5875 reconfig = true; 5876 } 5877 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5878 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5879 status = BAD_VALUE; 5880 } else { 5881 reqFormat = (audio_format_t) value; 5882 reconfig = true; 5883 } 5884 } 5885 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5886 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5887 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5888 status = BAD_VALUE; 5889 } else { 5890 channelMask = mask; 5891 reconfig = true; 5892 } 5893 } 5894 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5895 // do not accept frame count changes if tracks are open as the track buffer 5896 // size depends on frame count and correct behavior would not be guaranteed 5897 // if frame count is changed after track creation 5898 if (mActiveTracks.size() > 0) { 5899 status = INVALID_OPERATION; 5900 } else { 5901 reconfig = true; 5902 } 5903 } 5904 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5905 // forward device change to effects that have requested to be 5906 // aware of attached audio device. 5907 for (size_t i = 0; i < mEffectChains.size(); i++) { 5908 mEffectChains[i]->setDevice_l(value); 5909 } 5910 5911 // store input device and output device but do not forward output device to audio HAL. 5912 // Note that status is ignored by the caller for output device 5913 // (see AudioFlinger::setParameters() 5914 if (audio_is_output_devices(value)) { 5915 mOutDevice = value; 5916 status = BAD_VALUE; 5917 } else { 5918 mInDevice = value; 5919 // disable AEC and NS if the device is a BT SCO headset supporting those 5920 // pre processings 5921 if (mTracks.size() > 0) { 5922 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5923 mAudioFlinger->btNrecIsOff(); 5924 for (size_t i = 0; i < mTracks.size(); i++) { 5925 sp<RecordTrack> track = mTracks[i]; 5926 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5927 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5928 } 5929 } 5930 } 5931 } 5932 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5933 mAudioSource != (audio_source_t)value) { 5934 // forward device change to effects that have requested to be 5935 // aware of attached audio device. 5936 for (size_t i = 0; i < mEffectChains.size(); i++) { 5937 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5938 } 5939 mAudioSource = (audio_source_t)value; 5940 } 5941 5942 if (status == NO_ERROR) { 5943 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5944 keyValuePair.string()); 5945 if (status == INVALID_OPERATION) { 5946 inputStandBy(); 5947 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5948 keyValuePair.string()); 5949 } 5950 if (reconfig) { 5951 if (status == BAD_VALUE && 5952 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5953 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5954 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5955 <= (2 * samplingRate)) && 5956 audio_channel_count_from_in_mask( 5957 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5958 (channelMask == AUDIO_CHANNEL_IN_MONO || 5959 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5960 status = NO_ERROR; 5961 } 5962 if (status == NO_ERROR) { 5963 readInputParameters_l(); 5964 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5965 } 5966 } 5967 } 5968 5969 return reconfig; 5970} 5971 5972String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5973{ 5974 Mutex::Autolock _l(mLock); 5975 if (initCheck() != NO_ERROR) { 5976 return String8(); 5977 } 5978 5979 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5980 const String8 out_s8(s); 5981 free(s); 5982 return out_s8; 5983} 5984 5985void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5986 AudioSystem::OutputDescriptor desc; 5987 const void *param2 = NULL; 5988 5989 switch (event) { 5990 case AudioSystem::INPUT_OPENED: 5991 case AudioSystem::INPUT_CONFIG_CHANGED: 5992 desc.channelMask = mChannelMask; 5993 desc.samplingRate = mSampleRate; 5994 desc.format = mFormat; 5995 desc.frameCount = mFrameCount; 5996 desc.latency = 0; 5997 param2 = &desc; 5998 break; 5999 6000 case AudioSystem::INPUT_CLOSED: 6001 default: 6002 break; 6003 } 6004 mAudioFlinger->audioConfigChanged(event, mId, param2); 6005} 6006 6007void AudioFlinger::RecordThread::readInputParameters_l() 6008{ 6009 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6010 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6011 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6012 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6013 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6014 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6015 } 6016 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6017 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6018 mFrameCount = mBufferSize / mFrameSize; 6019 // This is the formula for calculating the temporary buffer size. 6020 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6021 // 1 full output buffer, regardless of the alignment of the available input. 6022 // The value is somewhat arbitrary, and could probably be even larger. 6023 // A larger value should allow more old data to be read after a track calls start(), 6024 // without increasing latency. 6025 mRsmpInFrames = mFrameCount * 7; 6026 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6027 delete[] mRsmpInBuffer; 6028 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6029 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6030 6031 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6032 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6033} 6034 6035uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6036{ 6037 Mutex::Autolock _l(mLock); 6038 if (initCheck() != NO_ERROR) { 6039 return 0; 6040 } 6041 6042 return mInput->stream->get_input_frames_lost(mInput->stream); 6043} 6044 6045uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6046{ 6047 Mutex::Autolock _l(mLock); 6048 uint32_t result = 0; 6049 if (getEffectChain_l(sessionId) != 0) { 6050 result = EFFECT_SESSION; 6051 } 6052 6053 for (size_t i = 0; i < mTracks.size(); ++i) { 6054 if (sessionId == mTracks[i]->sessionId()) { 6055 result |= TRACK_SESSION; 6056 break; 6057 } 6058 } 6059 6060 return result; 6061} 6062 6063KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6064{ 6065 KeyedVector<int, bool> ids; 6066 Mutex::Autolock _l(mLock); 6067 for (size_t j = 0; j < mTracks.size(); ++j) { 6068 sp<RecordThread::RecordTrack> track = mTracks[j]; 6069 int sessionId = track->sessionId(); 6070 if (ids.indexOfKey(sessionId) < 0) { 6071 ids.add(sessionId, true); 6072 } 6073 } 6074 return ids; 6075} 6076 6077AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6078{ 6079 Mutex::Autolock _l(mLock); 6080 AudioStreamIn *input = mInput; 6081 mInput = NULL; 6082 return input; 6083} 6084 6085// this method must always be called either with ThreadBase mLock held or inside the thread loop 6086audio_stream_t* AudioFlinger::RecordThread::stream() const 6087{ 6088 if (mInput == NULL) { 6089 return NULL; 6090 } 6091 return &mInput->stream->common; 6092} 6093 6094status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6095{ 6096 // only one chain per input thread 6097 if (mEffectChains.size() != 0) { 6098 return INVALID_OPERATION; 6099 } 6100 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6101 6102 chain->setInBuffer(NULL); 6103 chain->setOutBuffer(NULL); 6104 6105 checkSuspendOnAddEffectChain_l(chain); 6106 6107 mEffectChains.add(chain); 6108 6109 return NO_ERROR; 6110} 6111 6112size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6113{ 6114 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6115 ALOGW_IF(mEffectChains.size() != 1, 6116 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6117 chain.get(), mEffectChains.size(), this); 6118 if (mEffectChains.size() == 1) { 6119 mEffectChains.removeAt(0); 6120 } 6121 return 0; 6122} 6123 6124status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6125 audio_patch_handle_t *handle) 6126{ 6127 status_t status = NO_ERROR; 6128 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6129 // store new device and send to effects 6130 mInDevice = patch->sources[0].ext.device.type; 6131 for (size_t i = 0; i < mEffectChains.size(); i++) { 6132 mEffectChains[i]->setDevice_l(mInDevice); 6133 } 6134 6135 // disable AEC and NS if the device is a BT SCO headset supporting those 6136 // pre processings 6137 if (mTracks.size() > 0) { 6138 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6139 mAudioFlinger->btNrecIsOff(); 6140 for (size_t i = 0; i < mTracks.size(); i++) { 6141 sp<RecordTrack> track = mTracks[i]; 6142 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6143 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6144 } 6145 } 6146 6147 // store new source and send to effects 6148 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6149 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6150 for (size_t i = 0; i < mEffectChains.size(); i++) { 6151 mEffectChains[i]->setAudioSource_l(mAudioSource); 6152 } 6153 } 6154 6155 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6156 status = hwDevice->create_audio_patch(hwDevice, 6157 patch->num_sources, 6158 patch->sources, 6159 patch->num_sinks, 6160 patch->sinks, 6161 handle); 6162 } else { 6163 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6164 } 6165 return status; 6166} 6167 6168status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6169{ 6170 status_t status = NO_ERROR; 6171 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6172 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6173 status = hwDevice->release_audio_patch(hwDevice, handle); 6174 } else { 6175 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6176 } 6177 return status; 6178} 6179 6180 6181}; // namespace android 6182