Threads.cpp revision b6be7f22a82ee3bad8bcc709d21e72fc4727da09
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(false), 1161 mMixerBuffer(NULL), 1162 mMixerBufferSize(0), 1163 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1164 mMixerBufferValid(false), 1165 mEffectBufferEnabled(false), 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=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1405 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1406 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1407 audio_is_linear_pcm(format), 1408 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1409 *flags &= ~IAudioFlinger::TRACK_FAST; 1410 // For compatibility with AudioTrack calculation, buffer depth is forced 1411 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1412 // This is probably too conservative, but legacy application code may depend on it. 1413 // If you change this calculation, also review the start threshold which is related. 1414 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1415 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1416 if (minBufCount < 2) { 1417 minBufCount = 2; 1418 } 1419 size_t minFrameCount = mNormalFrameCount * minBufCount; 1420 if (frameCount < minFrameCount) { 1421 frameCount = minFrameCount; 1422 } 1423 } 1424 } 1425 *pFrameCount = frameCount; 1426 1427 switch (mType) { 1428 1429 case DIRECT: 1430 if (audio_is_linear_pcm(format)) { 1431 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1432 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1433 "for output %p with format %#x", 1434 sampleRate, format, channelMask, mOutput, mFormat); 1435 lStatus = BAD_VALUE; 1436 goto Exit; 1437 } 1438 } 1439 break; 1440 1441 case OFFLOAD: 1442 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1443 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1444 "for output %p with format %#x", 1445 sampleRate, format, channelMask, mOutput, mFormat); 1446 lStatus = BAD_VALUE; 1447 goto Exit; 1448 } 1449 break; 1450 1451 default: 1452 if (!audio_is_linear_pcm(format)) { 1453 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1454 "for output %p with format %#x", 1455 format, mOutput, mFormat); 1456 lStatus = BAD_VALUE; 1457 goto Exit; 1458 } 1459 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1460 if (sampleRate > mSampleRate*2) { 1461 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1462 lStatus = BAD_VALUE; 1463 goto Exit; 1464 } 1465 break; 1466 1467 } 1468 1469 lStatus = initCheck(); 1470 if (lStatus != NO_ERROR) { 1471 ALOGE("createTrack_l() audio driver not initialized"); 1472 goto Exit; 1473 } 1474 1475 { // scope for mLock 1476 Mutex::Autolock _l(mLock); 1477 1478 // all tracks in same audio session must share the same routing strategy otherwise 1479 // conflicts will happen when tracks are moved from one output to another by audio policy 1480 // manager 1481 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1482 for (size_t i = 0; i < mTracks.size(); ++i) { 1483 sp<Track> t = mTracks[i]; 1484 if (t != 0 && !t->isOutputTrack()) { 1485 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1486 if (sessionId == t->sessionId() && strategy != actual) { 1487 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1488 strategy, actual); 1489 lStatus = BAD_VALUE; 1490 goto Exit; 1491 } 1492 } 1493 } 1494 1495 if (!isTimed) { 1496 track = new Track(this, client, streamType, sampleRate, format, 1497 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1498 } else { 1499 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1500 channelMask, frameCount, sharedBuffer, sessionId, uid); 1501 } 1502 1503 // new Track always returns non-NULL, 1504 // but TimedTrack::create() is a factory that could fail by returning NULL 1505 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1506 if (lStatus != NO_ERROR) { 1507 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1508 // track must be cleared from the caller as the caller has the AF lock 1509 goto Exit; 1510 } 1511 mTracks.add(track); 1512 1513 sp<EffectChain> chain = getEffectChain_l(sessionId); 1514 if (chain != 0) { 1515 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1516 track->setMainBuffer(chain->inBuffer()); 1517 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1518 chain->incTrackCnt(); 1519 } 1520 1521 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1522 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1523 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1524 // so ask activity manager to do this on our behalf 1525 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1526 } 1527 } 1528 1529 lStatus = NO_ERROR; 1530 1531Exit: 1532 *status = lStatus; 1533 return track; 1534} 1535 1536uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1537{ 1538 return latency; 1539} 1540 1541uint32_t AudioFlinger::PlaybackThread::latency() const 1542{ 1543 Mutex::Autolock _l(mLock); 1544 return latency_l(); 1545} 1546uint32_t AudioFlinger::PlaybackThread::latency_l() const 1547{ 1548 if (initCheck() == NO_ERROR) { 1549 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1550 } else { 1551 return 0; 1552 } 1553} 1554 1555void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1556{ 1557 Mutex::Autolock _l(mLock); 1558 // Don't apply master volume in SW if our HAL can do it for us. 1559 if (mOutput && mOutput->audioHwDev && 1560 mOutput->audioHwDev->canSetMasterVolume()) { 1561 mMasterVolume = 1.0; 1562 } else { 1563 mMasterVolume = value; 1564 } 1565} 1566 1567void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1568{ 1569 Mutex::Autolock _l(mLock); 1570 // Don't apply master mute in SW if our HAL can do it for us. 1571 if (mOutput && mOutput->audioHwDev && 1572 mOutput->audioHwDev->canSetMasterMute()) { 1573 mMasterMute = false; 1574 } else { 1575 mMasterMute = muted; 1576 } 1577} 1578 1579void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1580{ 1581 Mutex::Autolock _l(mLock); 1582 mStreamTypes[stream].volume = value; 1583 broadcast_l(); 1584} 1585 1586void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1587{ 1588 Mutex::Autolock _l(mLock); 1589 mStreamTypes[stream].mute = muted; 1590 broadcast_l(); 1591} 1592 1593float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1594{ 1595 Mutex::Autolock _l(mLock); 1596 return mStreamTypes[stream].volume; 1597} 1598 1599// addTrack_l() must be called with ThreadBase::mLock held 1600status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1601{ 1602 status_t status = ALREADY_EXISTS; 1603 1604 // set retry count for buffer fill 1605 track->mRetryCount = kMaxTrackStartupRetries; 1606 if (mActiveTracks.indexOf(track) < 0) { 1607 // the track is newly added, make sure it fills up all its 1608 // buffers before playing. This is to ensure the client will 1609 // effectively get the latency it requested. 1610 if (!track->isOutputTrack()) { 1611 TrackBase::track_state state = track->mState; 1612 mLock.unlock(); 1613 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1614 mLock.lock(); 1615 // abort track was stopped/paused while we released the lock 1616 if (state != track->mState) { 1617 if (status == NO_ERROR) { 1618 mLock.unlock(); 1619 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1620 mLock.lock(); 1621 } 1622 return INVALID_OPERATION; 1623 } 1624 // abort if start is rejected by audio policy manager 1625 if (status != NO_ERROR) { 1626 return PERMISSION_DENIED; 1627 } 1628#ifdef ADD_BATTERY_DATA 1629 // to track the speaker usage 1630 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1631#endif 1632 } 1633 1634 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1635 track->mResetDone = false; 1636 track->mPresentationCompleteFrames = 0; 1637 mActiveTracks.add(track); 1638 mWakeLockUids.add(track->uid()); 1639 mActiveTracksGeneration++; 1640 mLatestActiveTrack = track; 1641 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1642 if (chain != 0) { 1643 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1644 track->sessionId()); 1645 chain->incActiveTrackCnt(); 1646 } 1647 1648 status = NO_ERROR; 1649 } 1650 1651 onAddNewTrack_l(); 1652 return status; 1653} 1654 1655bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1656{ 1657 track->terminate(); 1658 // active tracks are removed by threadLoop() 1659 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1660 track->mState = TrackBase::STOPPED; 1661 if (!trackActive) { 1662 removeTrack_l(track); 1663 } else if (track->isFastTrack() || track->isOffloaded()) { 1664 track->mState = TrackBase::STOPPING_1; 1665 } 1666 1667 return trackActive; 1668} 1669 1670void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1671{ 1672 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1673 mTracks.remove(track); 1674 deleteTrackName_l(track->name()); 1675 // redundant as track is about to be destroyed, for dumpsys only 1676 track->mName = -1; 1677 if (track->isFastTrack()) { 1678 int index = track->mFastIndex; 1679 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1680 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1681 mFastTrackAvailMask |= 1 << index; 1682 // redundant as track is about to be destroyed, for dumpsys only 1683 track->mFastIndex = -1; 1684 } 1685 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1686 if (chain != 0) { 1687 chain->decTrackCnt(); 1688 } 1689} 1690 1691void AudioFlinger::PlaybackThread::broadcast_l() 1692{ 1693 // Thread could be blocked waiting for async 1694 // so signal it to handle state changes immediately 1695 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1696 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1697 mSignalPending = true; 1698 mWaitWorkCV.broadcast(); 1699} 1700 1701String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1702{ 1703 Mutex::Autolock _l(mLock); 1704 if (initCheck() != NO_ERROR) { 1705 return String8(); 1706 } 1707 1708 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1709 const String8 out_s8(s); 1710 free(s); 1711 return out_s8; 1712} 1713 1714void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1715 AudioSystem::OutputDescriptor desc; 1716 void *param2 = NULL; 1717 1718 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1719 param); 1720 1721 switch (event) { 1722 case AudioSystem::OUTPUT_OPENED: 1723 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1724 desc.channelMask = mChannelMask; 1725 desc.samplingRate = mSampleRate; 1726 desc.format = mFormat; 1727 desc.frameCount = mNormalFrameCount; // FIXME see 1728 // AudioFlinger::frameCount(audio_io_handle_t) 1729 desc.latency = latency_l(); 1730 param2 = &desc; 1731 break; 1732 1733 case AudioSystem::STREAM_CONFIG_CHANGED: 1734 param2 = ¶m; 1735 case AudioSystem::OUTPUT_CLOSED: 1736 default: 1737 break; 1738 } 1739 mAudioFlinger->audioConfigChanged(event, mId, param2); 1740} 1741 1742void AudioFlinger::PlaybackThread::writeCallback() 1743{ 1744 ALOG_ASSERT(mCallbackThread != 0); 1745 mCallbackThread->resetWriteBlocked(); 1746} 1747 1748void AudioFlinger::PlaybackThread::drainCallback() 1749{ 1750 ALOG_ASSERT(mCallbackThread != 0); 1751 mCallbackThread->resetDraining(); 1752} 1753 1754void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1755{ 1756 Mutex::Autolock _l(mLock); 1757 // reject out of sequence requests 1758 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1759 mWriteAckSequence &= ~1; 1760 mWaitWorkCV.signal(); 1761 } 1762} 1763 1764void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1765{ 1766 Mutex::Autolock _l(mLock); 1767 // reject out of sequence requests 1768 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1769 mDrainSequence &= ~1; 1770 mWaitWorkCV.signal(); 1771 } 1772} 1773 1774// static 1775int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1776 void *param __unused, 1777 void *cookie) 1778{ 1779 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1780 ALOGV("asyncCallback() event %d", event); 1781 switch (event) { 1782 case STREAM_CBK_EVENT_WRITE_READY: 1783 me->writeCallback(); 1784 break; 1785 case STREAM_CBK_EVENT_DRAIN_READY: 1786 me->drainCallback(); 1787 break; 1788 default: 1789 ALOGW("asyncCallback() unknown event %d", event); 1790 break; 1791 } 1792 return 0; 1793} 1794 1795void AudioFlinger::PlaybackThread::readOutputParameters_l() 1796{ 1797 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1798 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1799 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1800 if (!audio_is_output_channel(mChannelMask)) { 1801 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1802 } 1803 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1804 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1805 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1806 } 1807 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1808 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1809 if (!audio_is_valid_format(mFormat)) { 1810 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1811 } 1812 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1813 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1814 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1815 } 1816 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1817 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1818 mFrameCount = mBufferSize / mFrameSize; 1819 if (mFrameCount & 15) { 1820 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1821 mFrameCount); 1822 } 1823 1824 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1825 (mOutput->stream->set_callback != NULL)) { 1826 if (mOutput->stream->set_callback(mOutput->stream, 1827 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1828 mUseAsyncWrite = true; 1829 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1830 } 1831 } 1832 1833 // Calculate size of normal sink buffer relative to the HAL output buffer size 1834 double multiplier = 1.0; 1835 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1836 kUseFastMixer == FastMixer_Dynamic)) { 1837 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1838 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1839 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1840 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1841 maxNormalFrameCount = maxNormalFrameCount & ~15; 1842 if (maxNormalFrameCount < minNormalFrameCount) { 1843 maxNormalFrameCount = minNormalFrameCount; 1844 } 1845 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1846 if (multiplier <= 1.0) { 1847 multiplier = 1.0; 1848 } else if (multiplier <= 2.0) { 1849 if (2 * mFrameCount <= maxNormalFrameCount) { 1850 multiplier = 2.0; 1851 } else { 1852 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1853 } 1854 } else { 1855 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1856 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1857 // track, but we sometimes have to do this to satisfy the maximum frame count 1858 // constraint) 1859 // FIXME this rounding up should not be done if no HAL SRC 1860 uint32_t truncMult = (uint32_t) multiplier; 1861 if ((truncMult & 1)) { 1862 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1863 ++truncMult; 1864 } 1865 } 1866 multiplier = (double) truncMult; 1867 } 1868 } 1869 mNormalFrameCount = multiplier * mFrameCount; 1870 // round up to nearest 16 frames to satisfy AudioMixer 1871 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1872 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1873 mNormalFrameCount); 1874 1875 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1876 // Originally this was int16_t[] array, need to remove legacy implications. 1877 free(mSinkBuffer); 1878 mSinkBuffer = NULL; 1879 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1880 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1881 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1882 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1883 1884 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1885 // drives the output. 1886 free(mMixerBuffer); 1887 mMixerBuffer = NULL; 1888 if (mMixerBufferEnabled) { 1889 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1890 mMixerBufferSize = mNormalFrameCount * mChannelCount 1891 * audio_bytes_per_sample(mMixerBufferFormat); 1892 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1893 } 1894 free(mEffectBuffer); 1895 mEffectBuffer = NULL; 1896 if (mEffectBufferEnabled) { 1897 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1898 mEffectBufferSize = mNormalFrameCount * mChannelCount 1899 * audio_bytes_per_sample(mEffectBufferFormat); 1900 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1901 } 1902 1903 // force reconfiguration of effect chains and engines to take new buffer size and audio 1904 // parameters into account 1905 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1906 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1907 // matter. 1908 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1909 Vector< sp<EffectChain> > effectChains = mEffectChains; 1910 for (size_t i = 0; i < effectChains.size(); i ++) { 1911 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1912 } 1913} 1914 1915 1916status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1917{ 1918 if (halFrames == NULL || dspFrames == NULL) { 1919 return BAD_VALUE; 1920 } 1921 Mutex::Autolock _l(mLock); 1922 if (initCheck() != NO_ERROR) { 1923 return INVALID_OPERATION; 1924 } 1925 size_t framesWritten = mBytesWritten / mFrameSize; 1926 *halFrames = framesWritten; 1927 1928 if (isSuspended()) { 1929 // return an estimation of rendered frames when the output is suspended 1930 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1931 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1932 return NO_ERROR; 1933 } else { 1934 status_t status; 1935 uint32_t frames; 1936 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1937 *dspFrames = (size_t)frames; 1938 return status; 1939 } 1940} 1941 1942uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1943{ 1944 Mutex::Autolock _l(mLock); 1945 uint32_t result = 0; 1946 if (getEffectChain_l(sessionId) != 0) { 1947 result = EFFECT_SESSION; 1948 } 1949 1950 for (size_t i = 0; i < mTracks.size(); ++i) { 1951 sp<Track> track = mTracks[i]; 1952 if (sessionId == track->sessionId() && !track->isInvalid()) { 1953 result |= TRACK_SESSION; 1954 break; 1955 } 1956 } 1957 1958 return result; 1959} 1960 1961uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1962{ 1963 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1964 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1965 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1966 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1967 } 1968 for (size_t i = 0; i < mTracks.size(); i++) { 1969 sp<Track> track = mTracks[i]; 1970 if (sessionId == track->sessionId() && !track->isInvalid()) { 1971 return AudioSystem::getStrategyForStream(track->streamType()); 1972 } 1973 } 1974 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1975} 1976 1977 1978AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1979{ 1980 Mutex::Autolock _l(mLock); 1981 return mOutput; 1982} 1983 1984AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1985{ 1986 Mutex::Autolock _l(mLock); 1987 AudioStreamOut *output = mOutput; 1988 mOutput = NULL; 1989 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1990 // must push a NULL and wait for ack 1991 mOutputSink.clear(); 1992 mPipeSink.clear(); 1993 mNormalSink.clear(); 1994 return output; 1995} 1996 1997// this method must always be called either with ThreadBase mLock held or inside the thread loop 1998audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1999{ 2000 if (mOutput == NULL) { 2001 return NULL; 2002 } 2003 return &mOutput->stream->common; 2004} 2005 2006uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2007{ 2008 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2009} 2010 2011status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2012{ 2013 if (!isValidSyncEvent(event)) { 2014 return BAD_VALUE; 2015 } 2016 2017 Mutex::Autolock _l(mLock); 2018 2019 for (size_t i = 0; i < mTracks.size(); ++i) { 2020 sp<Track> track = mTracks[i]; 2021 if (event->triggerSession() == track->sessionId()) { 2022 (void) track->setSyncEvent(event); 2023 return NO_ERROR; 2024 } 2025 } 2026 2027 return NAME_NOT_FOUND; 2028} 2029 2030bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2031{ 2032 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2033} 2034 2035void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2036 const Vector< sp<Track> >& tracksToRemove) 2037{ 2038 size_t count = tracksToRemove.size(); 2039 if (count > 0) { 2040 for (size_t i = 0 ; i < count ; i++) { 2041 const sp<Track>& track = tracksToRemove.itemAt(i); 2042 if (!track->isOutputTrack()) { 2043 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2044#ifdef ADD_BATTERY_DATA 2045 // to track the speaker usage 2046 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2047#endif 2048 if (track->isTerminated()) { 2049 AudioSystem::releaseOutput(mId); 2050 } 2051 } 2052 } 2053 } 2054} 2055 2056void AudioFlinger::PlaybackThread::checkSilentMode_l() 2057{ 2058 if (!mMasterMute) { 2059 char value[PROPERTY_VALUE_MAX]; 2060 if (property_get("ro.audio.silent", value, "0") > 0) { 2061 char *endptr; 2062 unsigned long ul = strtoul(value, &endptr, 0); 2063 if (*endptr == '\0' && ul != 0) { 2064 ALOGD("Silence is golden"); 2065 // The setprop command will not allow a property to be changed after 2066 // the first time it is set, so we don't have to worry about un-muting. 2067 setMasterMute_l(true); 2068 } 2069 } 2070 } 2071} 2072 2073// shared by MIXER and DIRECT, overridden by DUPLICATING 2074ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2075{ 2076 // FIXME rewrite to reduce number of system calls 2077 mLastWriteTime = systemTime(); 2078 mInWrite = true; 2079 ssize_t bytesWritten; 2080 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2081 2082 // If an NBAIO sink is present, use it to write the normal mixer's submix 2083 if (mNormalSink != 0) { 2084 const size_t count = mBytesRemaining / mFrameSize; 2085 2086 ATRACE_BEGIN("write"); 2087 // update the setpoint when AudioFlinger::mScreenState changes 2088 uint32_t screenState = AudioFlinger::mScreenState; 2089 if (screenState != mScreenState) { 2090 mScreenState = screenState; 2091 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2092 if (pipe != NULL) { 2093 pipe->setAvgFrames((mScreenState & 1) ? 2094 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2095 } 2096 } 2097 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2098 ATRACE_END(); 2099 if (framesWritten > 0) { 2100 bytesWritten = framesWritten * mFrameSize; 2101 } else { 2102 bytesWritten = framesWritten; 2103 } 2104 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2105 if (status == NO_ERROR) { 2106 size_t totalFramesWritten = mNormalSink->framesWritten(); 2107 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2108 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2109 mLatchDValid = true; 2110 } 2111 } 2112 // otherwise use the HAL / AudioStreamOut directly 2113 } else { 2114 // Direct output and offload threads 2115 2116 if (mUseAsyncWrite) { 2117 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2118 mWriteAckSequence += 2; 2119 mWriteAckSequence |= 1; 2120 ALOG_ASSERT(mCallbackThread != 0); 2121 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2122 } 2123 // FIXME We should have an implementation of timestamps for direct output threads. 2124 // They are used e.g for multichannel PCM playback over HDMI. 2125 bytesWritten = mOutput->stream->write(mOutput->stream, 2126 (char *)mSinkBuffer + offset, mBytesRemaining); 2127 if (mUseAsyncWrite && 2128 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2129 // do not wait for async callback in case of error of full write 2130 mWriteAckSequence &= ~1; 2131 ALOG_ASSERT(mCallbackThread != 0); 2132 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2133 } 2134 } 2135 2136 mNumWrites++; 2137 mInWrite = false; 2138 mStandby = false; 2139 return bytesWritten; 2140} 2141 2142void AudioFlinger::PlaybackThread::threadLoop_drain() 2143{ 2144 if (mOutput->stream->drain) { 2145 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2146 if (mUseAsyncWrite) { 2147 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2148 mDrainSequence |= 1; 2149 ALOG_ASSERT(mCallbackThread != 0); 2150 mCallbackThread->setDraining(mDrainSequence); 2151 } 2152 mOutput->stream->drain(mOutput->stream, 2153 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2154 : AUDIO_DRAIN_ALL); 2155 } 2156} 2157 2158void AudioFlinger::PlaybackThread::threadLoop_exit() 2159{ 2160 // Default implementation has nothing to do 2161} 2162 2163/* 2164The derived values that are cached: 2165 - mSinkBufferSize from frame count * frame size 2166 - activeSleepTime from activeSleepTimeUs() 2167 - idleSleepTime from idleSleepTimeUs() 2168 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2169 - maxPeriod from frame count and sample rate (MIXER only) 2170 2171The parameters that affect these derived values are: 2172 - frame count 2173 - frame size 2174 - sample rate 2175 - device type: A2DP or not 2176 - device latency 2177 - format: PCM or not 2178 - active sleep time 2179 - idle sleep time 2180*/ 2181 2182void AudioFlinger::PlaybackThread::cacheParameters_l() 2183{ 2184 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2185 activeSleepTime = activeSleepTimeUs(); 2186 idleSleepTime = idleSleepTimeUs(); 2187} 2188 2189void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2190{ 2191 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2192 this, streamType, mTracks.size()); 2193 Mutex::Autolock _l(mLock); 2194 2195 size_t size = mTracks.size(); 2196 for (size_t i = 0; i < size; i++) { 2197 sp<Track> t = mTracks[i]; 2198 if (t->streamType() == streamType) { 2199 t->invalidate(); 2200 } 2201 } 2202} 2203 2204status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2205{ 2206 int session = chain->sessionId(); 2207 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2208 ? mEffectBuffer : mSinkBuffer); 2209 bool ownsBuffer = false; 2210 2211 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2212 if (session > 0) { 2213 // Only one effect chain can be present in direct output thread and it uses 2214 // the sink buffer as input 2215 if (mType != DIRECT) { 2216 size_t numSamples = mNormalFrameCount * mChannelCount; 2217 buffer = new int16_t[numSamples]; 2218 memset(buffer, 0, numSamples * sizeof(int16_t)); 2219 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2220 ownsBuffer = true; 2221 } 2222 2223 // Attach all tracks with same session ID to this chain. 2224 for (size_t i = 0; i < mTracks.size(); ++i) { 2225 sp<Track> track = mTracks[i]; 2226 if (session == track->sessionId()) { 2227 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2228 buffer); 2229 track->setMainBuffer(buffer); 2230 chain->incTrackCnt(); 2231 } 2232 } 2233 2234 // indicate all active tracks in the chain 2235 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2236 sp<Track> track = mActiveTracks[i].promote(); 2237 if (track == 0) { 2238 continue; 2239 } 2240 if (session == track->sessionId()) { 2241 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2242 chain->incActiveTrackCnt(); 2243 } 2244 } 2245 } 2246 2247 chain->setInBuffer(buffer, ownsBuffer); 2248 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2249 ? mEffectBuffer : mSinkBuffer)); 2250 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2251 // chains list in order to be processed last as it contains output stage effects 2252 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2253 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2254 // after track specific effects and before output stage 2255 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2256 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2257 // Effect chain for other sessions are inserted at beginning of effect 2258 // chains list to be processed before output mix effects. Relative order between other 2259 // sessions is not important 2260 size_t size = mEffectChains.size(); 2261 size_t i = 0; 2262 for (i = 0; i < size; i++) { 2263 if (mEffectChains[i]->sessionId() < session) { 2264 break; 2265 } 2266 } 2267 mEffectChains.insertAt(chain, i); 2268 checkSuspendOnAddEffectChain_l(chain); 2269 2270 return NO_ERROR; 2271} 2272 2273size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2274{ 2275 int session = chain->sessionId(); 2276 2277 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2278 2279 for (size_t i = 0; i < mEffectChains.size(); i++) { 2280 if (chain == mEffectChains[i]) { 2281 mEffectChains.removeAt(i); 2282 // detach all active tracks from the chain 2283 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2284 sp<Track> track = mActiveTracks[i].promote(); 2285 if (track == 0) { 2286 continue; 2287 } 2288 if (session == track->sessionId()) { 2289 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2290 chain.get(), session); 2291 chain->decActiveTrackCnt(); 2292 } 2293 } 2294 2295 // detach all tracks with same session ID from this chain 2296 for (size_t i = 0; i < mTracks.size(); ++i) { 2297 sp<Track> track = mTracks[i]; 2298 if (session == track->sessionId()) { 2299 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2300 chain->decTrackCnt(); 2301 } 2302 } 2303 break; 2304 } 2305 } 2306 return mEffectChains.size(); 2307} 2308 2309status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2310 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2311{ 2312 Mutex::Autolock _l(mLock); 2313 return attachAuxEffect_l(track, EffectId); 2314} 2315 2316status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2317 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2318{ 2319 status_t status = NO_ERROR; 2320 2321 if (EffectId == 0) { 2322 track->setAuxBuffer(0, NULL); 2323 } else { 2324 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2325 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2326 if (effect != 0) { 2327 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2328 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2329 } else { 2330 status = INVALID_OPERATION; 2331 } 2332 } else { 2333 status = BAD_VALUE; 2334 } 2335 } 2336 return status; 2337} 2338 2339void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2340{ 2341 for (size_t i = 0; i < mTracks.size(); ++i) { 2342 sp<Track> track = mTracks[i]; 2343 if (track->auxEffectId() == effectId) { 2344 attachAuxEffect_l(track, 0); 2345 } 2346 } 2347} 2348 2349bool AudioFlinger::PlaybackThread::threadLoop() 2350{ 2351 Vector< sp<Track> > tracksToRemove; 2352 2353 standbyTime = systemTime(); 2354 2355 // MIXER 2356 nsecs_t lastWarning = 0; 2357 2358 // DUPLICATING 2359 // FIXME could this be made local to while loop? 2360 writeFrames = 0; 2361 2362 int lastGeneration = 0; 2363 2364 cacheParameters_l(); 2365 sleepTime = idleSleepTime; 2366 2367 if (mType == MIXER) { 2368 sleepTimeShift = 0; 2369 } 2370 2371 CpuStats cpuStats; 2372 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2373 2374 acquireWakeLock(); 2375 2376 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2377 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2378 // and then that string will be logged at the next convenient opportunity. 2379 const char *logString = NULL; 2380 2381 checkSilentMode_l(); 2382 2383 while (!exitPending()) 2384 { 2385 cpuStats.sample(myName); 2386 2387 Vector< sp<EffectChain> > effectChains; 2388 2389 { // scope for mLock 2390 2391 Mutex::Autolock _l(mLock); 2392 2393 processConfigEvents_l(); 2394 2395 if (logString != NULL) { 2396 mNBLogWriter->logTimestamp(); 2397 mNBLogWriter->log(logString); 2398 logString = NULL; 2399 } 2400 2401 if (mLatchDValid) { 2402 mLatchQ = mLatchD; 2403 mLatchDValid = false; 2404 mLatchQValid = true; 2405 } 2406 2407 saveOutputTracks(); 2408 if (mSignalPending) { 2409 // A signal was raised while we were unlocked 2410 mSignalPending = false; 2411 } else if (waitingAsyncCallback_l()) { 2412 if (exitPending()) { 2413 break; 2414 } 2415 releaseWakeLock_l(); 2416 mWakeLockUids.clear(); 2417 mActiveTracksGeneration++; 2418 ALOGV("wait async completion"); 2419 mWaitWorkCV.wait(mLock); 2420 ALOGV("async completion/wake"); 2421 acquireWakeLock_l(); 2422 standbyTime = systemTime() + standbyDelay; 2423 sleepTime = 0; 2424 2425 continue; 2426 } 2427 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2428 isSuspended()) { 2429 // put audio hardware into standby after short delay 2430 if (shouldStandby_l()) { 2431 2432 threadLoop_standby(); 2433 2434 mStandby = true; 2435 } 2436 2437 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2438 // we're about to wait, flush the binder command buffer 2439 IPCThreadState::self()->flushCommands(); 2440 2441 clearOutputTracks(); 2442 2443 if (exitPending()) { 2444 break; 2445 } 2446 2447 releaseWakeLock_l(); 2448 mWakeLockUids.clear(); 2449 mActiveTracksGeneration++; 2450 // wait until we have something to do... 2451 ALOGV("%s going to sleep", myName.string()); 2452 mWaitWorkCV.wait(mLock); 2453 ALOGV("%s waking up", myName.string()); 2454 acquireWakeLock_l(); 2455 2456 mMixerStatus = MIXER_IDLE; 2457 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2458 mBytesWritten = 0; 2459 mBytesRemaining = 0; 2460 checkSilentMode_l(); 2461 2462 standbyTime = systemTime() + standbyDelay; 2463 sleepTime = idleSleepTime; 2464 if (mType == MIXER) { 2465 sleepTimeShift = 0; 2466 } 2467 2468 continue; 2469 } 2470 } 2471 // mMixerStatusIgnoringFastTracks is also updated internally 2472 mMixerStatus = prepareTracks_l(&tracksToRemove); 2473 2474 // compare with previously applied list 2475 if (lastGeneration != mActiveTracksGeneration) { 2476 // update wakelock 2477 updateWakeLockUids_l(mWakeLockUids); 2478 lastGeneration = mActiveTracksGeneration; 2479 } 2480 2481 // prevent any changes in effect chain list and in each effect chain 2482 // during mixing and effect process as the audio buffers could be deleted 2483 // or modified if an effect is created or deleted 2484 lockEffectChains_l(effectChains); 2485 } // mLock scope ends 2486 2487 if (mBytesRemaining == 0) { 2488 mCurrentWriteLength = 0; 2489 if (mMixerStatus == MIXER_TRACKS_READY) { 2490 // threadLoop_mix() sets mCurrentWriteLength 2491 threadLoop_mix(); 2492 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2493 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2494 // threadLoop_sleepTime sets sleepTime to 0 if data 2495 // must be written to HAL 2496 threadLoop_sleepTime(); 2497 if (sleepTime == 0) { 2498 mCurrentWriteLength = mSinkBufferSize; 2499 } 2500 } 2501 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2502 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2503 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2504 // or mSinkBuffer (if there are no effects). 2505 // 2506 // This is done pre-effects computation; if effects change to 2507 // support higher precision, this needs to move. 2508 // 2509 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2510 // TODO use sleepTime == 0 as an additional condition. 2511 if (mMixerBufferValid) { 2512 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2513 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2514 2515 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2516 mNormalFrameCount * mChannelCount); 2517 } 2518 2519 mBytesRemaining = mCurrentWriteLength; 2520 if (isSuspended()) { 2521 sleepTime = suspendSleepTimeUs(); 2522 // simulate write to HAL when suspended 2523 mBytesWritten += mSinkBufferSize; 2524 mBytesRemaining = 0; 2525 } 2526 2527 // only process effects if we're going to write 2528 if (sleepTime == 0 && mType != OFFLOAD) { 2529 for (size_t i = 0; i < effectChains.size(); i ++) { 2530 effectChains[i]->process_l(); 2531 } 2532 } 2533 } 2534 // Process effect chains for offloaded thread even if no audio 2535 // was read from audio track: process only updates effect state 2536 // and thus does have to be synchronized with audio writes but may have 2537 // to be called while waiting for async write callback 2538 if (mType == OFFLOAD) { 2539 for (size_t i = 0; i < effectChains.size(); i ++) { 2540 effectChains[i]->process_l(); 2541 } 2542 } 2543 2544 // Only if the Effects buffer is enabled and there is data in the 2545 // Effects buffer (buffer valid), we need to 2546 // copy into the sink buffer. 2547 // TODO use sleepTime == 0 as an additional condition. 2548 if (mEffectBufferValid) { 2549 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2550 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2551 mNormalFrameCount * mChannelCount); 2552 } 2553 2554 // enable changes in effect chain 2555 unlockEffectChains(effectChains); 2556 2557 if (!waitingAsyncCallback()) { 2558 // sleepTime == 0 means we must write to audio hardware 2559 if (sleepTime == 0) { 2560 if (mBytesRemaining) { 2561 ssize_t ret = threadLoop_write(); 2562 if (ret < 0) { 2563 mBytesRemaining = 0; 2564 } else { 2565 mBytesWritten += ret; 2566 mBytesRemaining -= ret; 2567 } 2568 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2569 (mMixerStatus == MIXER_DRAIN_ALL)) { 2570 threadLoop_drain(); 2571 } 2572 if (mType == MIXER) { 2573 // write blocked detection 2574 nsecs_t now = systemTime(); 2575 nsecs_t delta = now - mLastWriteTime; 2576 if (!mStandby && delta > maxPeriod) { 2577 mNumDelayedWrites++; 2578 if ((now - lastWarning) > kWarningThrottleNs) { 2579 ATRACE_NAME("underrun"); 2580 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2581 ns2ms(delta), mNumDelayedWrites, this); 2582 lastWarning = now; 2583 } 2584 } 2585 } 2586 2587 } else { 2588 usleep(sleepTime); 2589 } 2590 } 2591 2592 // Finally let go of removed track(s), without the lock held 2593 // since we can't guarantee the destructors won't acquire that 2594 // same lock. This will also mutate and push a new fast mixer state. 2595 threadLoop_removeTracks(tracksToRemove); 2596 tracksToRemove.clear(); 2597 2598 // FIXME I don't understand the need for this here; 2599 // it was in the original code but maybe the 2600 // assignment in saveOutputTracks() makes this unnecessary? 2601 clearOutputTracks(); 2602 2603 // Effect chains will be actually deleted here if they were removed from 2604 // mEffectChains list during mixing or effects processing 2605 effectChains.clear(); 2606 2607 // FIXME Note that the above .clear() is no longer necessary since effectChains 2608 // is now local to this block, but will keep it for now (at least until merge done). 2609 } 2610 2611 threadLoop_exit(); 2612 2613 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2614 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2615 // put output stream into standby mode 2616 if (!mStandby) { 2617 mOutput->stream->common.standby(&mOutput->stream->common); 2618 } 2619 } 2620 2621 releaseWakeLock(); 2622 mWakeLockUids.clear(); 2623 mActiveTracksGeneration++; 2624 2625 ALOGV("Thread %p type %d exiting", this, mType); 2626 return false; 2627} 2628 2629// removeTracks_l() must be called with ThreadBase::mLock held 2630void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2631{ 2632 size_t count = tracksToRemove.size(); 2633 if (count > 0) { 2634 for (size_t i=0 ; i<count ; i++) { 2635 const sp<Track>& track = tracksToRemove.itemAt(i); 2636 mActiveTracks.remove(track); 2637 mWakeLockUids.remove(track->uid()); 2638 mActiveTracksGeneration++; 2639 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2640 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2641 if (chain != 0) { 2642 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2643 track->sessionId()); 2644 chain->decActiveTrackCnt(); 2645 } 2646 if (track->isTerminated()) { 2647 removeTrack_l(track); 2648 } 2649 } 2650 } 2651 2652} 2653 2654status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2655{ 2656 if (mNormalSink != 0) { 2657 return mNormalSink->getTimestamp(timestamp); 2658 } 2659 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2660 uint64_t position64; 2661 int ret = mOutput->stream->get_presentation_position( 2662 mOutput->stream, &position64, ×tamp.mTime); 2663 if (ret == 0) { 2664 timestamp.mPosition = (uint32_t)position64; 2665 return NO_ERROR; 2666 } 2667 } 2668 return INVALID_OPERATION; 2669} 2670 2671status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2672 audio_patch_handle_t *handle) 2673{ 2674 status_t status = NO_ERROR; 2675 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2676 // store new device and send to effects 2677 audio_devices_t type = AUDIO_DEVICE_NONE; 2678 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2679 type |= patch->sinks[i].ext.device.type; 2680 } 2681 mOutDevice = type; 2682 for (size_t i = 0; i < mEffectChains.size(); i++) { 2683 mEffectChains[i]->setDevice_l(mOutDevice); 2684 } 2685 2686 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2687 status = hwDevice->create_audio_patch(hwDevice, 2688 patch->num_sources, 2689 patch->sources, 2690 patch->num_sinks, 2691 patch->sinks, 2692 handle); 2693 } else { 2694 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2695 } 2696 return status; 2697} 2698 2699status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2700{ 2701 status_t status = NO_ERROR; 2702 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2703 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2704 status = hwDevice->release_audio_patch(hwDevice, handle); 2705 } else { 2706 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2707 } 2708 return status; 2709} 2710 2711// ---------------------------------------------------------------------------- 2712 2713AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2714 audio_io_handle_t id, audio_devices_t device, type_t type) 2715 : PlaybackThread(audioFlinger, output, id, device, type), 2716 // mAudioMixer below 2717 // mFastMixer below 2718 mFastMixerFutex(0) 2719 // mOutputSink below 2720 // mPipeSink below 2721 // mNormalSink below 2722{ 2723 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2724 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2725 "mFrameCount=%d, mNormalFrameCount=%d", 2726 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2727 mNormalFrameCount); 2728 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2729 2730 // FIXME - Current mixer implementation only supports stereo output 2731 if (mChannelCount != FCC_2) { 2732 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2733 } 2734 2735 // create an NBAIO sink for the HAL output stream, and negotiate 2736 mOutputSink = new AudioStreamOutSink(output->stream); 2737 size_t numCounterOffers = 0; 2738 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2739 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2740 ALOG_ASSERT(index == 0); 2741 2742 // initialize fast mixer depending on configuration 2743 bool initFastMixer; 2744 switch (kUseFastMixer) { 2745 case FastMixer_Never: 2746 initFastMixer = false; 2747 break; 2748 case FastMixer_Always: 2749 initFastMixer = true; 2750 break; 2751 case FastMixer_Static: 2752 case FastMixer_Dynamic: 2753 initFastMixer = mFrameCount < mNormalFrameCount; 2754 break; 2755 } 2756 if (initFastMixer) { 2757 audio_format_t fastMixerFormat; 2758 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2759 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2760 } else { 2761 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2762 } 2763 if (mFormat != fastMixerFormat) { 2764 // change our Sink format to accept our intermediate precision 2765 mFormat = fastMixerFormat; 2766 free(mSinkBuffer); 2767 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2768 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2769 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2770 } 2771 2772 // create a MonoPipe to connect our submix to FastMixer 2773 NBAIO_Format format = mOutputSink->format(); 2774 // adjust format to match that of the Fast Mixer 2775 format.mFormat = fastMixerFormat; 2776 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2777 2778 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2779 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2780 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2781 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2782 const NBAIO_Format offers[1] = {format}; 2783 size_t numCounterOffers = 0; 2784 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2785 ALOG_ASSERT(index == 0); 2786 monoPipe->setAvgFrames((mScreenState & 1) ? 2787 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2788 mPipeSink = monoPipe; 2789 2790#ifdef TEE_SINK 2791 if (mTeeSinkOutputEnabled) { 2792 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2793 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2794 numCounterOffers = 0; 2795 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2796 ALOG_ASSERT(index == 0); 2797 mTeeSink = teeSink; 2798 PipeReader *teeSource = new PipeReader(*teeSink); 2799 numCounterOffers = 0; 2800 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2801 ALOG_ASSERT(index == 0); 2802 mTeeSource = teeSource; 2803 } 2804#endif 2805 2806 // create fast mixer and configure it initially with just one fast track for our submix 2807 mFastMixer = new FastMixer(); 2808 FastMixerStateQueue *sq = mFastMixer->sq(); 2809#ifdef STATE_QUEUE_DUMP 2810 sq->setObserverDump(&mStateQueueObserverDump); 2811 sq->setMutatorDump(&mStateQueueMutatorDump); 2812#endif 2813 FastMixerState *state = sq->begin(); 2814 FastTrack *fastTrack = &state->mFastTracks[0]; 2815 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2816 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2817 fastTrack->mVolumeProvider = NULL; 2818 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2819 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2820 fastTrack->mGeneration++; 2821 state->mFastTracksGen++; 2822 state->mTrackMask = 1; 2823 // fast mixer will use the HAL output sink 2824 state->mOutputSink = mOutputSink.get(); 2825 state->mOutputSinkGen++; 2826 state->mFrameCount = mFrameCount; 2827 state->mCommand = FastMixerState::COLD_IDLE; 2828 // already done in constructor initialization list 2829 //mFastMixerFutex = 0; 2830 state->mColdFutexAddr = &mFastMixerFutex; 2831 state->mColdGen++; 2832 state->mDumpState = &mFastMixerDumpState; 2833#ifdef TEE_SINK 2834 state->mTeeSink = mTeeSink.get(); 2835#endif 2836 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2837 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2838 sq->end(); 2839 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2840 2841 // start the fast mixer 2842 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2843 pid_t tid = mFastMixer->getTid(); 2844 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2845 if (err != 0) { 2846 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2847 kPriorityFastMixer, getpid_cached, tid, err); 2848 } 2849 2850#ifdef AUDIO_WATCHDOG 2851 // create and start the watchdog 2852 mAudioWatchdog = new AudioWatchdog(); 2853 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2854 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2855 tid = mAudioWatchdog->getTid(); 2856 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2857 if (err != 0) { 2858 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2859 kPriorityFastMixer, getpid_cached, tid, err); 2860 } 2861#endif 2862 2863 } else { 2864 mFastMixer = NULL; 2865 } 2866 2867 switch (kUseFastMixer) { 2868 case FastMixer_Never: 2869 case FastMixer_Dynamic: 2870 mNormalSink = mOutputSink; 2871 break; 2872 case FastMixer_Always: 2873 mNormalSink = mPipeSink; 2874 break; 2875 case FastMixer_Static: 2876 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2877 break; 2878 } 2879} 2880 2881AudioFlinger::MixerThread::~MixerThread() 2882{ 2883 if (mFastMixer != NULL) { 2884 FastMixerStateQueue *sq = mFastMixer->sq(); 2885 FastMixerState *state = sq->begin(); 2886 if (state->mCommand == FastMixerState::COLD_IDLE) { 2887 int32_t old = android_atomic_inc(&mFastMixerFutex); 2888 if (old == -1) { 2889 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2890 } 2891 } 2892 state->mCommand = FastMixerState::EXIT; 2893 sq->end(); 2894 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2895 mFastMixer->join(); 2896 // Though the fast mixer thread has exited, it's state queue is still valid. 2897 // We'll use that extract the final state which contains one remaining fast track 2898 // corresponding to our sub-mix. 2899 state = sq->begin(); 2900 ALOG_ASSERT(state->mTrackMask == 1); 2901 FastTrack *fastTrack = &state->mFastTracks[0]; 2902 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2903 delete fastTrack->mBufferProvider; 2904 sq->end(false /*didModify*/); 2905 delete mFastMixer; 2906#ifdef AUDIO_WATCHDOG 2907 if (mAudioWatchdog != 0) { 2908 mAudioWatchdog->requestExit(); 2909 mAudioWatchdog->requestExitAndWait(); 2910 mAudioWatchdog.clear(); 2911 } 2912#endif 2913 } 2914 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2915 delete mAudioMixer; 2916} 2917 2918 2919uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2920{ 2921 if (mFastMixer != NULL) { 2922 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2923 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2924 } 2925 return latency; 2926} 2927 2928 2929void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2930{ 2931 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2932} 2933 2934ssize_t AudioFlinger::MixerThread::threadLoop_write() 2935{ 2936 // FIXME we should only do one push per cycle; confirm this is true 2937 // Start the fast mixer if it's not already running 2938 if (mFastMixer != NULL) { 2939 FastMixerStateQueue *sq = mFastMixer->sq(); 2940 FastMixerState *state = sq->begin(); 2941 if (state->mCommand != FastMixerState::MIX_WRITE && 2942 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2943 if (state->mCommand == FastMixerState::COLD_IDLE) { 2944 int32_t old = android_atomic_inc(&mFastMixerFutex); 2945 if (old == -1) { 2946 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2947 } 2948#ifdef AUDIO_WATCHDOG 2949 if (mAudioWatchdog != 0) { 2950 mAudioWatchdog->resume(); 2951 } 2952#endif 2953 } 2954 state->mCommand = FastMixerState::MIX_WRITE; 2955 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2956 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2957 sq->end(); 2958 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2959 if (kUseFastMixer == FastMixer_Dynamic) { 2960 mNormalSink = mPipeSink; 2961 } 2962 } else { 2963 sq->end(false /*didModify*/); 2964 } 2965 } 2966 return PlaybackThread::threadLoop_write(); 2967} 2968 2969void AudioFlinger::MixerThread::threadLoop_standby() 2970{ 2971 // Idle the fast mixer if it's currently running 2972 if (mFastMixer != NULL) { 2973 FastMixerStateQueue *sq = mFastMixer->sq(); 2974 FastMixerState *state = sq->begin(); 2975 if (!(state->mCommand & FastMixerState::IDLE)) { 2976 state->mCommand = FastMixerState::COLD_IDLE; 2977 state->mColdFutexAddr = &mFastMixerFutex; 2978 state->mColdGen++; 2979 mFastMixerFutex = 0; 2980 sq->end(); 2981 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2982 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2983 if (kUseFastMixer == FastMixer_Dynamic) { 2984 mNormalSink = mOutputSink; 2985 } 2986#ifdef AUDIO_WATCHDOG 2987 if (mAudioWatchdog != 0) { 2988 mAudioWatchdog->pause(); 2989 } 2990#endif 2991 } else { 2992 sq->end(false /*didModify*/); 2993 } 2994 } 2995 PlaybackThread::threadLoop_standby(); 2996} 2997 2998bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2999{ 3000 return false; 3001} 3002 3003bool AudioFlinger::PlaybackThread::shouldStandby_l() 3004{ 3005 return !mStandby; 3006} 3007 3008bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3009{ 3010 Mutex::Autolock _l(mLock); 3011 return waitingAsyncCallback_l(); 3012} 3013 3014// shared by MIXER and DIRECT, overridden by DUPLICATING 3015void AudioFlinger::PlaybackThread::threadLoop_standby() 3016{ 3017 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3018 mOutput->stream->common.standby(&mOutput->stream->common); 3019 if (mUseAsyncWrite != 0) { 3020 // discard any pending drain or write ack by incrementing sequence 3021 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3022 mDrainSequence = (mDrainSequence + 2) & ~1; 3023 ALOG_ASSERT(mCallbackThread != 0); 3024 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3025 mCallbackThread->setDraining(mDrainSequence); 3026 } 3027} 3028 3029void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3030{ 3031 ALOGV("signal playback thread"); 3032 broadcast_l(); 3033} 3034 3035void AudioFlinger::MixerThread::threadLoop_mix() 3036{ 3037 // obtain the presentation timestamp of the next output buffer 3038 int64_t pts; 3039 status_t status = INVALID_OPERATION; 3040 3041 if (mNormalSink != 0) { 3042 status = mNormalSink->getNextWriteTimestamp(&pts); 3043 } else { 3044 status = mOutputSink->getNextWriteTimestamp(&pts); 3045 } 3046 3047 if (status != NO_ERROR) { 3048 pts = AudioBufferProvider::kInvalidPTS; 3049 } 3050 3051 // mix buffers... 3052 mAudioMixer->process(pts); 3053 mCurrentWriteLength = mSinkBufferSize; 3054 // increase sleep time progressively when application underrun condition clears. 3055 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3056 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3057 // such that we would underrun the audio HAL. 3058 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3059 sleepTimeShift--; 3060 } 3061 sleepTime = 0; 3062 standbyTime = systemTime() + standbyDelay; 3063 //TODO: delay standby when effects have a tail 3064} 3065 3066void AudioFlinger::MixerThread::threadLoop_sleepTime() 3067{ 3068 // If no tracks are ready, sleep once for the duration of an output 3069 // buffer size, then write 0s to the output 3070 if (sleepTime == 0) { 3071 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3072 sleepTime = activeSleepTime >> sleepTimeShift; 3073 if (sleepTime < kMinThreadSleepTimeUs) { 3074 sleepTime = kMinThreadSleepTimeUs; 3075 } 3076 // reduce sleep time in case of consecutive application underruns to avoid 3077 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3078 // duration we would end up writing less data than needed by the audio HAL if 3079 // the condition persists. 3080 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3081 sleepTimeShift++; 3082 } 3083 } else { 3084 sleepTime = idleSleepTime; 3085 } 3086 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3087 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3088 // before effects processing or output. 3089 if (mMixerBufferValid) { 3090 memset(mMixerBuffer, 0, mMixerBufferSize); 3091 } else { 3092 memset(mSinkBuffer, 0, mSinkBufferSize); 3093 } 3094 sleepTime = 0; 3095 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3096 "anticipated start"); 3097 } 3098 // TODO add standby time extension fct of effect tail 3099} 3100 3101// prepareTracks_l() must be called with ThreadBase::mLock held 3102AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3103 Vector< sp<Track> > *tracksToRemove) 3104{ 3105 3106 mixer_state mixerStatus = MIXER_IDLE; 3107 // find out which tracks need to be processed 3108 size_t count = mActiveTracks.size(); 3109 size_t mixedTracks = 0; 3110 size_t tracksWithEffect = 0; 3111 // counts only _active_ fast tracks 3112 size_t fastTracks = 0; 3113 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3114 3115 float masterVolume = mMasterVolume; 3116 bool masterMute = mMasterMute; 3117 3118 if (masterMute) { 3119 masterVolume = 0; 3120 } 3121 // Delegate master volume control to effect in output mix effect chain if needed 3122 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3123 if (chain != 0) { 3124 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3125 chain->setVolume_l(&v, &v); 3126 masterVolume = (float)((v + (1 << 23)) >> 24); 3127 chain.clear(); 3128 } 3129 3130 // prepare a new state to push 3131 FastMixerStateQueue *sq = NULL; 3132 FastMixerState *state = NULL; 3133 bool didModify = false; 3134 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3135 if (mFastMixer != NULL) { 3136 sq = mFastMixer->sq(); 3137 state = sq->begin(); 3138 } 3139 3140 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3141 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3142 3143 for (size_t i=0 ; i<count ; i++) { 3144 const sp<Track> t = mActiveTracks[i].promote(); 3145 if (t == 0) { 3146 continue; 3147 } 3148 3149 // this const just means the local variable doesn't change 3150 Track* const track = t.get(); 3151 3152 // process fast tracks 3153 if (track->isFastTrack()) { 3154 3155 // It's theoretically possible (though unlikely) for a fast track to be created 3156 // and then removed within the same normal mix cycle. This is not a problem, as 3157 // the track never becomes active so it's fast mixer slot is never touched. 3158 // The converse, of removing an (active) track and then creating a new track 3159 // at the identical fast mixer slot within the same normal mix cycle, 3160 // is impossible because the slot isn't marked available until the end of each cycle. 3161 int j = track->mFastIndex; 3162 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3163 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3164 FastTrack *fastTrack = &state->mFastTracks[j]; 3165 3166 // Determine whether the track is currently in underrun condition, 3167 // and whether it had a recent underrun. 3168 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3169 FastTrackUnderruns underruns = ftDump->mUnderruns; 3170 uint32_t recentFull = (underruns.mBitFields.mFull - 3171 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3172 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3173 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3174 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3175 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3176 uint32_t recentUnderruns = recentPartial + recentEmpty; 3177 track->mObservedUnderruns = underruns; 3178 // don't count underruns that occur while stopping or pausing 3179 // or stopped which can occur when flush() is called while active 3180 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3181 recentUnderruns > 0) { 3182 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3183 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3184 } 3185 3186 // This is similar to the state machine for normal tracks, 3187 // with a few modifications for fast tracks. 3188 bool isActive = true; 3189 switch (track->mState) { 3190 case TrackBase::STOPPING_1: 3191 // track stays active in STOPPING_1 state until first underrun 3192 if (recentUnderruns > 0 || track->isTerminated()) { 3193 track->mState = TrackBase::STOPPING_2; 3194 } 3195 break; 3196 case TrackBase::PAUSING: 3197 // ramp down is not yet implemented 3198 track->setPaused(); 3199 break; 3200 case TrackBase::RESUMING: 3201 // ramp up is not yet implemented 3202 track->mState = TrackBase::ACTIVE; 3203 break; 3204 case TrackBase::ACTIVE: 3205 if (recentFull > 0 || recentPartial > 0) { 3206 // track has provided at least some frames recently: reset retry count 3207 track->mRetryCount = kMaxTrackRetries; 3208 } 3209 if (recentUnderruns == 0) { 3210 // no recent underruns: stay active 3211 break; 3212 } 3213 // there has recently been an underrun of some kind 3214 if (track->sharedBuffer() == 0) { 3215 // were any of the recent underruns "empty" (no frames available)? 3216 if (recentEmpty == 0) { 3217 // no, then ignore the partial underruns as they are allowed indefinitely 3218 break; 3219 } 3220 // there has recently been an "empty" underrun: decrement the retry counter 3221 if (--(track->mRetryCount) > 0) { 3222 break; 3223 } 3224 // indicate to client process that the track was disabled because of underrun; 3225 // it will then automatically call start() when data is available 3226 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3227 // remove from active list, but state remains ACTIVE [confusing but true] 3228 isActive = false; 3229 break; 3230 } 3231 // fall through 3232 case TrackBase::STOPPING_2: 3233 case TrackBase::PAUSED: 3234 case TrackBase::STOPPED: 3235 case TrackBase::FLUSHED: // flush() while active 3236 // Check for presentation complete if track is inactive 3237 // We have consumed all the buffers of this track. 3238 // This would be incomplete if we auto-paused on underrun 3239 { 3240 size_t audioHALFrames = 3241 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3242 size_t framesWritten = mBytesWritten / mFrameSize; 3243 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3244 // track stays in active list until presentation is complete 3245 break; 3246 } 3247 } 3248 if (track->isStopping_2()) { 3249 track->mState = TrackBase::STOPPED; 3250 } 3251 if (track->isStopped()) { 3252 // Can't reset directly, as fast mixer is still polling this track 3253 // track->reset(); 3254 // So instead mark this track as needing to be reset after push with ack 3255 resetMask |= 1 << i; 3256 } 3257 isActive = false; 3258 break; 3259 case TrackBase::IDLE: 3260 default: 3261 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3262 } 3263 3264 if (isActive) { 3265 // was it previously inactive? 3266 if (!(state->mTrackMask & (1 << j))) { 3267 ExtendedAudioBufferProvider *eabp = track; 3268 VolumeProvider *vp = track; 3269 fastTrack->mBufferProvider = eabp; 3270 fastTrack->mVolumeProvider = vp; 3271 fastTrack->mChannelMask = track->mChannelMask; 3272 fastTrack->mFormat = track->mFormat; 3273 fastTrack->mGeneration++; 3274 state->mTrackMask |= 1 << j; 3275 didModify = true; 3276 // no acknowledgement required for newly active tracks 3277 } 3278 // cache the combined master volume and stream type volume for fast mixer; this 3279 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3280 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3281 ++fastTracks; 3282 } else { 3283 // was it previously active? 3284 if (state->mTrackMask & (1 << j)) { 3285 fastTrack->mBufferProvider = NULL; 3286 fastTrack->mGeneration++; 3287 state->mTrackMask &= ~(1 << j); 3288 didModify = true; 3289 // If any fast tracks were removed, we must wait for acknowledgement 3290 // because we're about to decrement the last sp<> on those tracks. 3291 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3292 } else { 3293 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3294 } 3295 tracksToRemove->add(track); 3296 // Avoids a misleading display in dumpsys 3297 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3298 } 3299 continue; 3300 } 3301 3302 { // local variable scope to avoid goto warning 3303 3304 audio_track_cblk_t* cblk = track->cblk(); 3305 3306 // The first time a track is added we wait 3307 // for all its buffers to be filled before processing it 3308 int name = track->name(); 3309 // make sure that we have enough frames to mix one full buffer. 3310 // enforce this condition only once to enable draining the buffer in case the client 3311 // app does not call stop() and relies on underrun to stop: 3312 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3313 // during last round 3314 size_t desiredFrames; 3315 uint32_t sr = track->sampleRate(); 3316 if (sr == mSampleRate) { 3317 desiredFrames = mNormalFrameCount; 3318 } else { 3319 // +1 for rounding and +1 for additional sample needed for interpolation 3320 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3321 // add frames already consumed but not yet released by the resampler 3322 // because mAudioTrackServerProxy->framesReady() will include these frames 3323 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3324#if 0 3325 // the minimum track buffer size is normally twice the number of frames necessary 3326 // to fill one buffer and the resampler should not leave more than one buffer worth 3327 // of unreleased frames after each pass, but just in case... 3328 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3329#endif 3330 } 3331 uint32_t minFrames = 1; 3332 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3333 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3334 minFrames = desiredFrames; 3335 } 3336 3337 size_t framesReady = track->framesReady(); 3338 if ((framesReady >= minFrames) && track->isReady() && 3339 !track->isPaused() && !track->isTerminated()) 3340 { 3341 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3342 3343 mixedTracks++; 3344 3345 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3346 // there is an effect chain connected to the track 3347 chain.clear(); 3348 if (track->mainBuffer() != mSinkBuffer && 3349 track->mainBuffer() != mMixerBuffer) { 3350 if (mEffectBufferEnabled) { 3351 mEffectBufferValid = true; // Later can set directly. 3352 } 3353 chain = getEffectChain_l(track->sessionId()); 3354 // Delegate volume control to effect in track effect chain if needed 3355 if (chain != 0) { 3356 tracksWithEffect++; 3357 } else { 3358 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3359 "session %d", 3360 name, track->sessionId()); 3361 } 3362 } 3363 3364 3365 int param = AudioMixer::VOLUME; 3366 if (track->mFillingUpStatus == Track::FS_FILLED) { 3367 // no ramp for the first volume setting 3368 track->mFillingUpStatus = Track::FS_ACTIVE; 3369 if (track->mState == TrackBase::RESUMING) { 3370 track->mState = TrackBase::ACTIVE; 3371 param = AudioMixer::RAMP_VOLUME; 3372 } 3373 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3374 // FIXME should not make a decision based on mServer 3375 } else if (cblk->mServer != 0) { 3376 // If the track is stopped before the first frame was mixed, 3377 // do not apply ramp 3378 param = AudioMixer::RAMP_VOLUME; 3379 } 3380 3381 // compute volume for this track 3382 uint32_t vl, vr; // in U8.24 integer format 3383 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3384 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3385 vl = vr = 0; 3386 vlf = vrf = vaf = 0.; 3387 if (track->isPausing()) { 3388 track->setPaused(); 3389 } 3390 } else { 3391 3392 // read original volumes with volume control 3393 float typeVolume = mStreamTypes[track->streamType()].volume; 3394 float v = masterVolume * typeVolume; 3395 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3396 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3397 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3398 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3399 // track volumes come from shared memory, so can't be trusted and must be clamped 3400 if (vlf > GAIN_FLOAT_UNITY) { 3401 ALOGV("Track left volume out of range: %.3g", vlf); 3402 vlf = GAIN_FLOAT_UNITY; 3403 } 3404 if (vrf > GAIN_FLOAT_UNITY) { 3405 ALOGV("Track right volume out of range: %.3g", vrf); 3406 vrf = GAIN_FLOAT_UNITY; 3407 } 3408 // now apply the master volume and stream type volume 3409 vlf *= v; 3410 vrf *= v; 3411 // assuming master volume and stream type volume each go up to 1.0, 3412 // then derive vl and vr as U8.24 versions for the effect chain 3413 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3414 vl = (uint32_t) (scaleto8_24 * vlf); 3415 vr = (uint32_t) (scaleto8_24 * vrf); 3416 // vl and vr are now in U8.24 format 3417 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3418 // send level comes from shared memory and so may be corrupt 3419 if (sendLevel > MAX_GAIN_INT) { 3420 ALOGV("Track send level out of range: %04X", sendLevel); 3421 sendLevel = MAX_GAIN_INT; 3422 } 3423 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3424 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3425 } 3426 3427 // Delegate volume control to effect in track effect chain if needed 3428 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3429 // Do not ramp volume if volume is controlled by effect 3430 param = AudioMixer::VOLUME; 3431 // Update remaining floating point volume levels 3432 vlf = (float)vl / (1 << 24); 3433 vrf = (float)vr / (1 << 24); 3434 track->mHasVolumeController = true; 3435 } else { 3436 // force no volume ramp when volume controller was just disabled or removed 3437 // from effect chain to avoid volume spike 3438 if (track->mHasVolumeController) { 3439 param = AudioMixer::VOLUME; 3440 } 3441 track->mHasVolumeController = false; 3442 } 3443 3444 // XXX: these things DON'T need to be done each time 3445 mAudioMixer->setBufferProvider(name, track); 3446 mAudioMixer->enable(name); 3447 3448 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3449 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3450 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3451 mAudioMixer->setParameter( 3452 name, 3453 AudioMixer::TRACK, 3454 AudioMixer::FORMAT, (void *)track->format()); 3455 mAudioMixer->setParameter( 3456 name, 3457 AudioMixer::TRACK, 3458 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3459 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3460 uint32_t maxSampleRate = mSampleRate * 2; 3461 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3462 if (reqSampleRate == 0) { 3463 reqSampleRate = mSampleRate; 3464 } else if (reqSampleRate > maxSampleRate) { 3465 reqSampleRate = maxSampleRate; 3466 } 3467 mAudioMixer->setParameter( 3468 name, 3469 AudioMixer::RESAMPLE, 3470 AudioMixer::SAMPLE_RATE, 3471 (void *)(uintptr_t)reqSampleRate); 3472 /* 3473 * Select the appropriate output buffer for the track. 3474 * 3475 * Tracks with effects go into their own effects chain buffer 3476 * and from there into either mEffectBuffer or mSinkBuffer. 3477 * 3478 * Other tracks can use mMixerBuffer for higher precision 3479 * channel accumulation. If this buffer is enabled 3480 * (mMixerBufferEnabled true), then selected tracks will accumulate 3481 * into it. 3482 * 3483 */ 3484 if (mMixerBufferEnabled 3485 && (track->mainBuffer() == mSinkBuffer 3486 || track->mainBuffer() == mMixerBuffer)) { 3487 mAudioMixer->setParameter( 3488 name, 3489 AudioMixer::TRACK, 3490 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3491 mAudioMixer->setParameter( 3492 name, 3493 AudioMixer::TRACK, 3494 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3495 // TODO: override track->mainBuffer()? 3496 mMixerBufferValid = true; 3497 } else { 3498 mAudioMixer->setParameter( 3499 name, 3500 AudioMixer::TRACK, 3501 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3502 mAudioMixer->setParameter( 3503 name, 3504 AudioMixer::TRACK, 3505 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3506 } 3507 mAudioMixer->setParameter( 3508 name, 3509 AudioMixer::TRACK, 3510 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3511 3512 // reset retry count 3513 track->mRetryCount = kMaxTrackRetries; 3514 3515 // If one track is ready, set the mixer ready if: 3516 // - the mixer was not ready during previous round OR 3517 // - no other track is not ready 3518 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3519 mixerStatus != MIXER_TRACKS_ENABLED) { 3520 mixerStatus = MIXER_TRACKS_READY; 3521 } 3522 } else { 3523 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3524 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3525 } 3526 // clear effect chain input buffer if an active track underruns to avoid sending 3527 // previous audio buffer again to effects 3528 chain = getEffectChain_l(track->sessionId()); 3529 if (chain != 0) { 3530 chain->clearInputBuffer(); 3531 } 3532 3533 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3534 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3535 track->isStopped() || track->isPaused()) { 3536 // We have consumed all the buffers of this track. 3537 // Remove it from the list of active tracks. 3538 // TODO: use actual buffer filling status instead of latency when available from 3539 // audio HAL 3540 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3541 size_t framesWritten = mBytesWritten / mFrameSize; 3542 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3543 if (track->isStopped()) { 3544 track->reset(); 3545 } 3546 tracksToRemove->add(track); 3547 } 3548 } else { 3549 // No buffers for this track. Give it a few chances to 3550 // fill a buffer, then remove it from active list. 3551 if (--(track->mRetryCount) <= 0) { 3552 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3553 tracksToRemove->add(track); 3554 // indicate to client process that the track was disabled because of underrun; 3555 // it will then automatically call start() when data is available 3556 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3557 // If one track is not ready, mark the mixer also not ready if: 3558 // - the mixer was ready during previous round OR 3559 // - no other track is ready 3560 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3561 mixerStatus != MIXER_TRACKS_READY) { 3562 mixerStatus = MIXER_TRACKS_ENABLED; 3563 } 3564 } 3565 mAudioMixer->disable(name); 3566 } 3567 3568 } // local variable scope to avoid goto warning 3569track_is_ready: ; 3570 3571 } 3572 3573 // Push the new FastMixer state if necessary 3574 bool pauseAudioWatchdog = false; 3575 if (didModify) { 3576 state->mFastTracksGen++; 3577 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3578 if (kUseFastMixer == FastMixer_Dynamic && 3579 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3580 state->mCommand = FastMixerState::COLD_IDLE; 3581 state->mColdFutexAddr = &mFastMixerFutex; 3582 state->mColdGen++; 3583 mFastMixerFutex = 0; 3584 if (kUseFastMixer == FastMixer_Dynamic) { 3585 mNormalSink = mOutputSink; 3586 } 3587 // If we go into cold idle, need to wait for acknowledgement 3588 // so that fast mixer stops doing I/O. 3589 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3590 pauseAudioWatchdog = true; 3591 } 3592 } 3593 if (sq != NULL) { 3594 sq->end(didModify); 3595 sq->push(block); 3596 } 3597#ifdef AUDIO_WATCHDOG 3598 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3599 mAudioWatchdog->pause(); 3600 } 3601#endif 3602 3603 // Now perform the deferred reset on fast tracks that have stopped 3604 while (resetMask != 0) { 3605 size_t i = __builtin_ctz(resetMask); 3606 ALOG_ASSERT(i < count); 3607 resetMask &= ~(1 << i); 3608 sp<Track> t = mActiveTracks[i].promote(); 3609 if (t == 0) { 3610 continue; 3611 } 3612 Track* track = t.get(); 3613 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3614 track->reset(); 3615 } 3616 3617 // remove all the tracks that need to be... 3618 removeTracks_l(*tracksToRemove); 3619 3620 // sink or mix buffer must be cleared if all tracks are connected to an 3621 // effect chain as in this case the mixer will not write to the sink or mix buffer 3622 // and track effects will accumulate into it 3623 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3624 (mixedTracks == 0 && fastTracks > 0))) { 3625 // FIXME as a performance optimization, should remember previous zero status 3626 if (mMixerBufferValid) { 3627 memset(mMixerBuffer, 0, mMixerBufferSize); 3628 // TODO: In testing, mSinkBuffer below need not be cleared because 3629 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3630 // after mixing. 3631 // 3632 // To enforce this guarantee: 3633 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3634 // (mixedTracks == 0 && fastTracks > 0)) 3635 // must imply MIXER_TRACKS_READY. 3636 // Later, we may clear buffers regardless, and skip much of this logic. 3637 } 3638 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3639 if (mEffectBufferValid) { 3640 memset(mEffectBuffer, 0, mEffectBufferSize); 3641 } 3642 // FIXME as a performance optimization, should remember previous zero status 3643 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3644 } 3645 3646 // if any fast tracks, then status is ready 3647 mMixerStatusIgnoringFastTracks = mixerStatus; 3648 if (fastTracks > 0) { 3649 mixerStatus = MIXER_TRACKS_READY; 3650 } 3651 return mixerStatus; 3652} 3653 3654// getTrackName_l() must be called with ThreadBase::mLock held 3655int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3656 audio_format_t format, int sessionId) 3657{ 3658 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3659} 3660 3661// deleteTrackName_l() must be called with ThreadBase::mLock held 3662void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3663{ 3664 ALOGV("remove track (%d) and delete from mixer", name); 3665 mAudioMixer->deleteTrackName(name); 3666} 3667 3668// checkForNewParameter_l() must be called with ThreadBase::mLock held 3669bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3670 status_t& status) 3671{ 3672 bool reconfig = false; 3673 3674 status = NO_ERROR; 3675 3676 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3677 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3678 if (mFastMixer != NULL) { 3679 FastMixerStateQueue *sq = mFastMixer->sq(); 3680 FastMixerState *state = sq->begin(); 3681 if (!(state->mCommand & FastMixerState::IDLE)) { 3682 previousCommand = state->mCommand; 3683 state->mCommand = FastMixerState::HOT_IDLE; 3684 sq->end(); 3685 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3686 } else { 3687 sq->end(false /*didModify*/); 3688 } 3689 } 3690 3691 AudioParameter param = AudioParameter(keyValuePair); 3692 int value; 3693 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3694 reconfig = true; 3695 } 3696 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3697 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3698 status = BAD_VALUE; 3699 } else { 3700 // no need to save value, since it's constant 3701 reconfig = true; 3702 } 3703 } 3704 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3705 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3706 status = BAD_VALUE; 3707 } else { 3708 // no need to save value, since it's constant 3709 reconfig = true; 3710 } 3711 } 3712 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3713 // do not accept frame count changes if tracks are open as the track buffer 3714 // size depends on frame count and correct behavior would not be guaranteed 3715 // if frame count is changed after track creation 3716 if (!mTracks.isEmpty()) { 3717 status = INVALID_OPERATION; 3718 } else { 3719 reconfig = true; 3720 } 3721 } 3722 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3723#ifdef ADD_BATTERY_DATA 3724 // when changing the audio output device, call addBatteryData to notify 3725 // the change 3726 if (mOutDevice != value) { 3727 uint32_t params = 0; 3728 // check whether speaker is on 3729 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3730 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3731 } 3732 3733 audio_devices_t deviceWithoutSpeaker 3734 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3735 // check if any other device (except speaker) is on 3736 if (value & deviceWithoutSpeaker ) { 3737 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3738 } 3739 3740 if (params != 0) { 3741 addBatteryData(params); 3742 } 3743 } 3744#endif 3745 3746 // forward device change to effects that have requested to be 3747 // aware of attached audio device. 3748 if (value != AUDIO_DEVICE_NONE) { 3749 mOutDevice = value; 3750 for (size_t i = 0; i < mEffectChains.size(); i++) { 3751 mEffectChains[i]->setDevice_l(mOutDevice); 3752 } 3753 } 3754 } 3755 3756 if (status == NO_ERROR) { 3757 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3758 keyValuePair.string()); 3759 if (!mStandby && status == INVALID_OPERATION) { 3760 mOutput->stream->common.standby(&mOutput->stream->common); 3761 mStandby = true; 3762 mBytesWritten = 0; 3763 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3764 keyValuePair.string()); 3765 } 3766 if (status == NO_ERROR && reconfig) { 3767 readOutputParameters_l(); 3768 delete mAudioMixer; 3769 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3770 for (size_t i = 0; i < mTracks.size() ; i++) { 3771 int name = getTrackName_l(mTracks[i]->mChannelMask, 3772 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3773 if (name < 0) { 3774 break; 3775 } 3776 mTracks[i]->mName = name; 3777 } 3778 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3779 } 3780 } 3781 3782 if (!(previousCommand & FastMixerState::IDLE)) { 3783 ALOG_ASSERT(mFastMixer != NULL); 3784 FastMixerStateQueue *sq = mFastMixer->sq(); 3785 FastMixerState *state = sq->begin(); 3786 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3787 state->mCommand = previousCommand; 3788 sq->end(); 3789 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3790 } 3791 3792 return reconfig; 3793} 3794 3795 3796void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3797{ 3798 const size_t SIZE = 256; 3799 char buffer[SIZE]; 3800 String8 result; 3801 3802 PlaybackThread::dumpInternals(fd, args); 3803 3804 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3805 3806 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3807 const FastMixerDumpState copy(mFastMixerDumpState); 3808 copy.dump(fd); 3809 3810#ifdef STATE_QUEUE_DUMP 3811 // Similar for state queue 3812 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3813 observerCopy.dump(fd); 3814 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3815 mutatorCopy.dump(fd); 3816#endif 3817 3818#ifdef TEE_SINK 3819 // Write the tee output to a .wav file 3820 dumpTee(fd, mTeeSource, mId); 3821#endif 3822 3823#ifdef AUDIO_WATCHDOG 3824 if (mAudioWatchdog != 0) { 3825 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3826 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3827 wdCopy.dump(fd); 3828 } 3829#endif 3830} 3831 3832uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3833{ 3834 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3835} 3836 3837uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3838{ 3839 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3840} 3841 3842void AudioFlinger::MixerThread::cacheParameters_l() 3843{ 3844 PlaybackThread::cacheParameters_l(); 3845 3846 // FIXME: Relaxed timing because of a certain device that can't meet latency 3847 // Should be reduced to 2x after the vendor fixes the driver issue 3848 // increase threshold again due to low power audio mode. The way this warning 3849 // threshold is calculated and its usefulness should be reconsidered anyway. 3850 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3851} 3852 3853// ---------------------------------------------------------------------------- 3854 3855AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3856 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3857 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3858 // mLeftVolFloat, mRightVolFloat 3859{ 3860} 3861 3862AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3863 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3864 ThreadBase::type_t type) 3865 : PlaybackThread(audioFlinger, output, id, device, type) 3866 // mLeftVolFloat, mRightVolFloat 3867{ 3868} 3869 3870AudioFlinger::DirectOutputThread::~DirectOutputThread() 3871{ 3872} 3873 3874void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3875{ 3876 audio_track_cblk_t* cblk = track->cblk(); 3877 float left, right; 3878 3879 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3880 left = right = 0; 3881 } else { 3882 float typeVolume = mStreamTypes[track->streamType()].volume; 3883 float v = mMasterVolume * typeVolume; 3884 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3885 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3886 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3887 if (left > GAIN_FLOAT_UNITY) { 3888 left = GAIN_FLOAT_UNITY; 3889 } 3890 left *= v; 3891 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3892 if (right > GAIN_FLOAT_UNITY) { 3893 right = GAIN_FLOAT_UNITY; 3894 } 3895 right *= v; 3896 } 3897 3898 if (lastTrack) { 3899 if (left != mLeftVolFloat || right != mRightVolFloat) { 3900 mLeftVolFloat = left; 3901 mRightVolFloat = right; 3902 3903 // Convert volumes from float to 8.24 3904 uint32_t vl = (uint32_t)(left * (1 << 24)); 3905 uint32_t vr = (uint32_t)(right * (1 << 24)); 3906 3907 // Delegate volume control to effect in track effect chain if needed 3908 // only one effect chain can be present on DirectOutputThread, so if 3909 // there is one, the track is connected to it 3910 if (!mEffectChains.isEmpty()) { 3911 mEffectChains[0]->setVolume_l(&vl, &vr); 3912 left = (float)vl / (1 << 24); 3913 right = (float)vr / (1 << 24); 3914 } 3915 if (mOutput->stream->set_volume) { 3916 mOutput->stream->set_volume(mOutput->stream, left, right); 3917 } 3918 } 3919 } 3920} 3921 3922 3923AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3924 Vector< sp<Track> > *tracksToRemove 3925) 3926{ 3927 size_t count = mActiveTracks.size(); 3928 mixer_state mixerStatus = MIXER_IDLE; 3929 3930 // find out which tracks need to be processed 3931 for (size_t i = 0; i < count; i++) { 3932 sp<Track> t = mActiveTracks[i].promote(); 3933 // The track died recently 3934 if (t == 0) { 3935 continue; 3936 } 3937 3938 Track* const track = t.get(); 3939 audio_track_cblk_t* cblk = track->cblk(); 3940 // Only consider last track started for volume and mixer state control. 3941 // In theory an older track could underrun and restart after the new one starts 3942 // but as we only care about the transition phase between two tracks on a 3943 // direct output, it is not a problem to ignore the underrun case. 3944 sp<Track> l = mLatestActiveTrack.promote(); 3945 bool last = l.get() == track; 3946 3947 // The first time a track is added we wait 3948 // for all its buffers to be filled before processing it 3949 uint32_t minFrames; 3950 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3951 minFrames = mNormalFrameCount; 3952 } else { 3953 minFrames = 1; 3954 } 3955 3956 if ((track->framesReady() >= minFrames) && track->isReady() && 3957 !track->isPaused() && !track->isTerminated()) 3958 { 3959 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3960 3961 if (track->mFillingUpStatus == Track::FS_FILLED) { 3962 track->mFillingUpStatus = Track::FS_ACTIVE; 3963 // make sure processVolume_l() will apply new volume even if 0 3964 mLeftVolFloat = mRightVolFloat = -1.0; 3965 if (track->mState == TrackBase::RESUMING) { 3966 track->mState = TrackBase::ACTIVE; 3967 } 3968 } 3969 3970 // compute volume for this track 3971 processVolume_l(track, last); 3972 if (last) { 3973 // reset retry count 3974 track->mRetryCount = kMaxTrackRetriesDirect; 3975 mActiveTrack = t; 3976 mixerStatus = MIXER_TRACKS_READY; 3977 } 3978 } else { 3979 // clear effect chain input buffer if the last active track started underruns 3980 // to avoid sending previous audio buffer again to effects 3981 if (!mEffectChains.isEmpty() && last) { 3982 mEffectChains[0]->clearInputBuffer(); 3983 } 3984 3985 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3986 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3987 track->isStopped() || track->isPaused()) { 3988 // We have consumed all the buffers of this track. 3989 // Remove it from the list of active tracks. 3990 // TODO: implement behavior for compressed audio 3991 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3992 size_t framesWritten = mBytesWritten / mFrameSize; 3993 if (mStandby || !last || 3994 track->presentationComplete(framesWritten, audioHALFrames)) { 3995 if (track->isStopped()) { 3996 track->reset(); 3997 } 3998 tracksToRemove->add(track); 3999 } 4000 } else { 4001 // No buffers for this track. Give it a few chances to 4002 // fill a buffer, then remove it from active list. 4003 // Only consider last track started for mixer state control 4004 if (--(track->mRetryCount) <= 0) { 4005 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4006 tracksToRemove->add(track); 4007 // indicate to client process that the track was disabled because of underrun; 4008 // it will then automatically call start() when data is available 4009 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4010 } else if (last) { 4011 mixerStatus = MIXER_TRACKS_ENABLED; 4012 } 4013 } 4014 } 4015 } 4016 4017 // remove all the tracks that need to be... 4018 removeTracks_l(*tracksToRemove); 4019 4020 return mixerStatus; 4021} 4022 4023void AudioFlinger::DirectOutputThread::threadLoop_mix() 4024{ 4025 size_t frameCount = mFrameCount; 4026 int8_t *curBuf = (int8_t *)mSinkBuffer; 4027 // output audio to hardware 4028 while (frameCount) { 4029 AudioBufferProvider::Buffer buffer; 4030 buffer.frameCount = frameCount; 4031 mActiveTrack->getNextBuffer(&buffer); 4032 if (buffer.raw == NULL) { 4033 memset(curBuf, 0, frameCount * mFrameSize); 4034 break; 4035 } 4036 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4037 frameCount -= buffer.frameCount; 4038 curBuf += buffer.frameCount * mFrameSize; 4039 mActiveTrack->releaseBuffer(&buffer); 4040 } 4041 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4042 sleepTime = 0; 4043 standbyTime = systemTime() + standbyDelay; 4044 mActiveTrack.clear(); 4045} 4046 4047void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4048{ 4049 if (sleepTime == 0) { 4050 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4051 sleepTime = activeSleepTime; 4052 } else { 4053 sleepTime = idleSleepTime; 4054 } 4055 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4056 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4057 sleepTime = 0; 4058 } 4059} 4060 4061// getTrackName_l() must be called with ThreadBase::mLock held 4062int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4063 audio_format_t format __unused, int sessionId __unused) 4064{ 4065 return 0; 4066} 4067 4068// deleteTrackName_l() must be called with ThreadBase::mLock held 4069void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4070{ 4071} 4072 4073// checkForNewParameter_l() must be called with ThreadBase::mLock held 4074bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4075 status_t& status) 4076{ 4077 bool reconfig = false; 4078 4079 status = NO_ERROR; 4080 4081 AudioParameter param = AudioParameter(keyValuePair); 4082 int value; 4083 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4084 // forward device change to effects that have requested to be 4085 // aware of attached audio device. 4086 if (value != AUDIO_DEVICE_NONE) { 4087 mOutDevice = value; 4088 for (size_t i = 0; i < mEffectChains.size(); i++) { 4089 mEffectChains[i]->setDevice_l(mOutDevice); 4090 } 4091 } 4092 } 4093 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4094 // do not accept frame count changes if tracks are open as the track buffer 4095 // size depends on frame count and correct behavior would not be garantied 4096 // if frame count is changed after track creation 4097 if (!mTracks.isEmpty()) { 4098 status = INVALID_OPERATION; 4099 } else { 4100 reconfig = true; 4101 } 4102 } 4103 if (status == NO_ERROR) { 4104 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4105 keyValuePair.string()); 4106 if (!mStandby && status == INVALID_OPERATION) { 4107 mOutput->stream->common.standby(&mOutput->stream->common); 4108 mStandby = true; 4109 mBytesWritten = 0; 4110 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4111 keyValuePair.string()); 4112 } 4113 if (status == NO_ERROR && reconfig) { 4114 readOutputParameters_l(); 4115 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4116 } 4117 } 4118 4119 return reconfig; 4120} 4121 4122uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4123{ 4124 uint32_t time; 4125 if (audio_is_linear_pcm(mFormat)) { 4126 time = PlaybackThread::activeSleepTimeUs(); 4127 } else { 4128 time = 10000; 4129 } 4130 return time; 4131} 4132 4133uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4134{ 4135 uint32_t time; 4136 if (audio_is_linear_pcm(mFormat)) { 4137 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4138 } else { 4139 time = 10000; 4140 } 4141 return time; 4142} 4143 4144uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4145{ 4146 uint32_t time; 4147 if (audio_is_linear_pcm(mFormat)) { 4148 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4149 } else { 4150 time = 10000; 4151 } 4152 return time; 4153} 4154 4155void AudioFlinger::DirectOutputThread::cacheParameters_l() 4156{ 4157 PlaybackThread::cacheParameters_l(); 4158 4159 // use shorter standby delay as on normal output to release 4160 // hardware resources as soon as possible 4161 if (audio_is_linear_pcm(mFormat)) { 4162 standbyDelay = microseconds(activeSleepTime*2); 4163 } else { 4164 standbyDelay = kOffloadStandbyDelayNs; 4165 } 4166} 4167 4168// ---------------------------------------------------------------------------- 4169 4170AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4171 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4172 : Thread(false /*canCallJava*/), 4173 mPlaybackThread(playbackThread), 4174 mWriteAckSequence(0), 4175 mDrainSequence(0) 4176{ 4177} 4178 4179AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4180{ 4181} 4182 4183void AudioFlinger::AsyncCallbackThread::onFirstRef() 4184{ 4185 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4186} 4187 4188bool AudioFlinger::AsyncCallbackThread::threadLoop() 4189{ 4190 while (!exitPending()) { 4191 uint32_t writeAckSequence; 4192 uint32_t drainSequence; 4193 4194 { 4195 Mutex::Autolock _l(mLock); 4196 while (!((mWriteAckSequence & 1) || 4197 (mDrainSequence & 1) || 4198 exitPending())) { 4199 mWaitWorkCV.wait(mLock); 4200 } 4201 4202 if (exitPending()) { 4203 break; 4204 } 4205 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4206 mWriteAckSequence, mDrainSequence); 4207 writeAckSequence = mWriteAckSequence; 4208 mWriteAckSequence &= ~1; 4209 drainSequence = mDrainSequence; 4210 mDrainSequence &= ~1; 4211 } 4212 { 4213 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4214 if (playbackThread != 0) { 4215 if (writeAckSequence & 1) { 4216 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4217 } 4218 if (drainSequence & 1) { 4219 playbackThread->resetDraining(drainSequence >> 1); 4220 } 4221 } 4222 } 4223 } 4224 return false; 4225} 4226 4227void AudioFlinger::AsyncCallbackThread::exit() 4228{ 4229 ALOGV("AsyncCallbackThread::exit"); 4230 Mutex::Autolock _l(mLock); 4231 requestExit(); 4232 mWaitWorkCV.broadcast(); 4233} 4234 4235void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4236{ 4237 Mutex::Autolock _l(mLock); 4238 // bit 0 is cleared 4239 mWriteAckSequence = sequence << 1; 4240} 4241 4242void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4243{ 4244 Mutex::Autolock _l(mLock); 4245 // ignore unexpected callbacks 4246 if (mWriteAckSequence & 2) { 4247 mWriteAckSequence |= 1; 4248 mWaitWorkCV.signal(); 4249 } 4250} 4251 4252void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4253{ 4254 Mutex::Autolock _l(mLock); 4255 // bit 0 is cleared 4256 mDrainSequence = sequence << 1; 4257} 4258 4259void AudioFlinger::AsyncCallbackThread::resetDraining() 4260{ 4261 Mutex::Autolock _l(mLock); 4262 // ignore unexpected callbacks 4263 if (mDrainSequence & 2) { 4264 mDrainSequence |= 1; 4265 mWaitWorkCV.signal(); 4266 } 4267} 4268 4269 4270// ---------------------------------------------------------------------------- 4271AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4272 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4273 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4274 mHwPaused(false), 4275 mFlushPending(false), 4276 mPausedBytesRemaining(0) 4277{ 4278 //FIXME: mStandby should be set to true by ThreadBase constructor 4279 mStandby = true; 4280} 4281 4282void AudioFlinger::OffloadThread::threadLoop_exit() 4283{ 4284 if (mFlushPending || mHwPaused) { 4285 // If a flush is pending or track was paused, just discard buffered data 4286 flushHw_l(); 4287 } else { 4288 mMixerStatus = MIXER_DRAIN_ALL; 4289 threadLoop_drain(); 4290 } 4291 if (mUseAsyncWrite) { 4292 ALOG_ASSERT(mCallbackThread != 0); 4293 mCallbackThread->exit(); 4294 } 4295 PlaybackThread::threadLoop_exit(); 4296} 4297 4298AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4299 Vector< sp<Track> > *tracksToRemove 4300) 4301{ 4302 size_t count = mActiveTracks.size(); 4303 4304 mixer_state mixerStatus = MIXER_IDLE; 4305 bool doHwPause = false; 4306 bool doHwResume = false; 4307 4308 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4309 4310 // find out which tracks need to be processed 4311 for (size_t i = 0; i < count; i++) { 4312 sp<Track> t = mActiveTracks[i].promote(); 4313 // The track died recently 4314 if (t == 0) { 4315 continue; 4316 } 4317 Track* const track = t.get(); 4318 audio_track_cblk_t* cblk = track->cblk(); 4319 // Only consider last track started for volume and mixer state control. 4320 // In theory an older track could underrun and restart after the new one starts 4321 // but as we only care about the transition phase between two tracks on a 4322 // direct output, it is not a problem to ignore the underrun case. 4323 sp<Track> l = mLatestActiveTrack.promote(); 4324 bool last = l.get() == track; 4325 4326 if (track->isInvalid()) { 4327 ALOGW("An invalidated track shouldn't be in active list"); 4328 tracksToRemove->add(track); 4329 continue; 4330 } 4331 4332 if (track->mState == TrackBase::IDLE) { 4333 ALOGW("An idle track shouldn't be in active list"); 4334 continue; 4335 } 4336 4337 if (track->isPausing()) { 4338 track->setPaused(); 4339 if (last) { 4340 if (!mHwPaused) { 4341 doHwPause = true; 4342 mHwPaused = true; 4343 } 4344 // If we were part way through writing the mixbuffer to 4345 // the HAL we must save this until we resume 4346 // BUG - this will be wrong if a different track is made active, 4347 // in that case we want to discard the pending data in the 4348 // mixbuffer and tell the client to present it again when the 4349 // track is resumed 4350 mPausedWriteLength = mCurrentWriteLength; 4351 mPausedBytesRemaining = mBytesRemaining; 4352 mBytesRemaining = 0; // stop writing 4353 } 4354 tracksToRemove->add(track); 4355 } else if (track->isFlushPending()) { 4356 track->flushAck(); 4357 if (last) { 4358 mFlushPending = true; 4359 } 4360 } else if (track->isResumePending()){ 4361 track->resumeAck(); 4362 if (last) { 4363 if (mPausedBytesRemaining) { 4364 // Need to continue write that was interrupted 4365 mCurrentWriteLength = mPausedWriteLength; 4366 mBytesRemaining = mPausedBytesRemaining; 4367 mPausedBytesRemaining = 0; 4368 } 4369 if (mHwPaused) { 4370 doHwResume = true; 4371 mHwPaused = false; 4372 // threadLoop_mix() will handle the case that we need to 4373 // resume an interrupted write 4374 } 4375 // enable write to audio HAL 4376 sleepTime = 0; 4377 4378 // Do not handle new data in this iteration even if track->framesReady() 4379 mixerStatus = MIXER_TRACKS_ENABLED; 4380 } 4381 } else if (track->framesReady() && track->isReady() && 4382 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4383 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4384 if (track->mFillingUpStatus == Track::FS_FILLED) { 4385 track->mFillingUpStatus = Track::FS_ACTIVE; 4386 // make sure processVolume_l() will apply new volume even if 0 4387 mLeftVolFloat = mRightVolFloat = -1.0; 4388 } 4389 4390 if (last) { 4391 sp<Track> previousTrack = mPreviousTrack.promote(); 4392 if (previousTrack != 0) { 4393 if (track != previousTrack.get()) { 4394 // Flush any data still being written from last track 4395 mBytesRemaining = 0; 4396 if (mPausedBytesRemaining) { 4397 // Last track was paused so we also need to flush saved 4398 // mixbuffer state and invalidate track so that it will 4399 // re-submit that unwritten data when it is next resumed 4400 mPausedBytesRemaining = 0; 4401 // Invalidate is a bit drastic - would be more efficient 4402 // to have a flag to tell client that some of the 4403 // previously written data was lost 4404 previousTrack->invalidate(); 4405 } 4406 // flush data already sent to the DSP if changing audio session as audio 4407 // comes from a different source. Also invalidate previous track to force a 4408 // seek when resuming. 4409 if (previousTrack->sessionId() != track->sessionId()) { 4410 previousTrack->invalidate(); 4411 } 4412 } 4413 } 4414 mPreviousTrack = track; 4415 // reset retry count 4416 track->mRetryCount = kMaxTrackRetriesOffload; 4417 mActiveTrack = t; 4418 mixerStatus = MIXER_TRACKS_READY; 4419 } 4420 } else { 4421 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4422 if (track->isStopping_1()) { 4423 // Hardware buffer can hold a large amount of audio so we must 4424 // wait for all current track's data to drain before we say 4425 // that the track is stopped. 4426 if (mBytesRemaining == 0) { 4427 // Only start draining when all data in mixbuffer 4428 // has been written 4429 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4430 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4431 // do not drain if no data was ever sent to HAL (mStandby == true) 4432 if (last && !mStandby) { 4433 // do not modify drain sequence if we are already draining. This happens 4434 // when resuming from pause after drain. 4435 if ((mDrainSequence & 1) == 0) { 4436 sleepTime = 0; 4437 standbyTime = systemTime() + standbyDelay; 4438 mixerStatus = MIXER_DRAIN_TRACK; 4439 mDrainSequence += 2; 4440 } 4441 if (mHwPaused) { 4442 // It is possible to move from PAUSED to STOPPING_1 without 4443 // a resume so we must ensure hardware is running 4444 doHwResume = true; 4445 mHwPaused = false; 4446 } 4447 } 4448 } 4449 } else if (track->isStopping_2()) { 4450 // Drain has completed or we are in standby, signal presentation complete 4451 if (!(mDrainSequence & 1) || !last || mStandby) { 4452 track->mState = TrackBase::STOPPED; 4453 size_t audioHALFrames = 4454 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4455 size_t framesWritten = 4456 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4457 track->presentationComplete(framesWritten, audioHALFrames); 4458 track->reset(); 4459 tracksToRemove->add(track); 4460 } 4461 } else { 4462 // No buffers for this track. Give it a few chances to 4463 // fill a buffer, then remove it from active list. 4464 if (--(track->mRetryCount) <= 0) { 4465 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4466 track->name()); 4467 tracksToRemove->add(track); 4468 // indicate to client process that the track was disabled because of underrun; 4469 // it will then automatically call start() when data is available 4470 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4471 } else if (last){ 4472 mixerStatus = MIXER_TRACKS_ENABLED; 4473 } 4474 } 4475 } 4476 // compute volume for this track 4477 processVolume_l(track, last); 4478 } 4479 4480 // make sure the pause/flush/resume sequence is executed in the right order. 4481 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4482 // before flush and then resume HW. This can happen in case of pause/flush/resume 4483 // if resume is received before pause is executed. 4484 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4485 mOutput->stream->pause(mOutput->stream); 4486 } 4487 if (mFlushPending) { 4488 flushHw_l(); 4489 mFlushPending = false; 4490 } 4491 if (!mStandby && doHwResume) { 4492 mOutput->stream->resume(mOutput->stream); 4493 } 4494 4495 // remove all the tracks that need to be... 4496 removeTracks_l(*tracksToRemove); 4497 4498 return mixerStatus; 4499} 4500 4501// must be called with thread mutex locked 4502bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4503{ 4504 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4505 mWriteAckSequence, mDrainSequence); 4506 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4507 return true; 4508 } 4509 return false; 4510} 4511 4512// must be called with thread mutex locked 4513bool AudioFlinger::OffloadThread::shouldStandby_l() 4514{ 4515 bool trackPaused = false; 4516 4517 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4518 // after a timeout and we will enter standby then. 4519 if (mTracks.size() > 0) { 4520 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4521 } 4522 4523 return !mStandby && !trackPaused; 4524} 4525 4526 4527bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4528{ 4529 Mutex::Autolock _l(mLock); 4530 return waitingAsyncCallback_l(); 4531} 4532 4533void AudioFlinger::OffloadThread::flushHw_l() 4534{ 4535 mOutput->stream->flush(mOutput->stream); 4536 // Flush anything still waiting in the mixbuffer 4537 mCurrentWriteLength = 0; 4538 mBytesRemaining = 0; 4539 mPausedWriteLength = 0; 4540 mPausedBytesRemaining = 0; 4541 mHwPaused = false; 4542 4543 if (mUseAsyncWrite) { 4544 // discard any pending drain or write ack by incrementing sequence 4545 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4546 mDrainSequence = (mDrainSequence + 2) & ~1; 4547 ALOG_ASSERT(mCallbackThread != 0); 4548 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4549 mCallbackThread->setDraining(mDrainSequence); 4550 } 4551} 4552 4553void AudioFlinger::OffloadThread::onAddNewTrack_l() 4554{ 4555 sp<Track> previousTrack = mPreviousTrack.promote(); 4556 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4557 4558 if (previousTrack != 0 && latestTrack != 0 && 4559 (previousTrack->sessionId() != latestTrack->sessionId())) { 4560 mFlushPending = true; 4561 } 4562 PlaybackThread::onAddNewTrack_l(); 4563} 4564 4565// ---------------------------------------------------------------------------- 4566 4567AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4568 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4569 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4570 DUPLICATING), 4571 mWaitTimeMs(UINT_MAX) 4572{ 4573 addOutputTrack(mainThread); 4574} 4575 4576AudioFlinger::DuplicatingThread::~DuplicatingThread() 4577{ 4578 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4579 mOutputTracks[i]->destroy(); 4580 } 4581} 4582 4583void AudioFlinger::DuplicatingThread::threadLoop_mix() 4584{ 4585 // mix buffers... 4586 if (outputsReady(outputTracks)) { 4587 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4588 } else { 4589 memset(mSinkBuffer, 0, mSinkBufferSize); 4590 } 4591 sleepTime = 0; 4592 writeFrames = mNormalFrameCount; 4593 mCurrentWriteLength = mSinkBufferSize; 4594 standbyTime = systemTime() + standbyDelay; 4595} 4596 4597void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4598{ 4599 if (sleepTime == 0) { 4600 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4601 sleepTime = activeSleepTime; 4602 } else { 4603 sleepTime = idleSleepTime; 4604 } 4605 } else if (mBytesWritten != 0) { 4606 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4607 writeFrames = mNormalFrameCount; 4608 memset(mSinkBuffer, 0, mSinkBufferSize); 4609 } else { 4610 // flush remaining overflow buffers in output tracks 4611 writeFrames = 0; 4612 } 4613 sleepTime = 0; 4614 } 4615} 4616 4617ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4618{ 4619 for (size_t i = 0; i < outputTracks.size(); i++) { 4620 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4621 // for delivery downstream as needed. This in-place conversion is safe as 4622 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4623 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4624 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4625 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4626 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4627 } 4628 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4629 } 4630 mStandby = false; 4631 return (ssize_t)mSinkBufferSize; 4632} 4633 4634void AudioFlinger::DuplicatingThread::threadLoop_standby() 4635{ 4636 // DuplicatingThread implements standby by stopping all tracks 4637 for (size_t i = 0; i < outputTracks.size(); i++) { 4638 outputTracks[i]->stop(); 4639 } 4640} 4641 4642void AudioFlinger::DuplicatingThread::saveOutputTracks() 4643{ 4644 outputTracks = mOutputTracks; 4645} 4646 4647void AudioFlinger::DuplicatingThread::clearOutputTracks() 4648{ 4649 outputTracks.clear(); 4650} 4651 4652void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4653{ 4654 Mutex::Autolock _l(mLock); 4655 // FIXME explain this formula 4656 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4657 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4658 // due to current usage case and restrictions on the AudioBufferProvider. 4659 // Actual buffer conversion is done in threadLoop_write(). 4660 // 4661 // TODO: This may change in the future, depending on multichannel 4662 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4663 OutputTrack *outputTrack = new OutputTrack(thread, 4664 this, 4665 mSampleRate, 4666 AUDIO_FORMAT_PCM_16_BIT, 4667 mChannelMask, 4668 frameCount, 4669 IPCThreadState::self()->getCallingUid()); 4670 if (outputTrack->cblk() != NULL) { 4671 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4672 mOutputTracks.add(outputTrack); 4673 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4674 updateWaitTime_l(); 4675 } 4676} 4677 4678void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4679{ 4680 Mutex::Autolock _l(mLock); 4681 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4682 if (mOutputTracks[i]->thread() == thread) { 4683 mOutputTracks[i]->destroy(); 4684 mOutputTracks.removeAt(i); 4685 updateWaitTime_l(); 4686 return; 4687 } 4688 } 4689 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4690} 4691 4692// caller must hold mLock 4693void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4694{ 4695 mWaitTimeMs = UINT_MAX; 4696 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4697 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4698 if (strong != 0) { 4699 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4700 if (waitTimeMs < mWaitTimeMs) { 4701 mWaitTimeMs = waitTimeMs; 4702 } 4703 } 4704 } 4705} 4706 4707 4708bool AudioFlinger::DuplicatingThread::outputsReady( 4709 const SortedVector< sp<OutputTrack> > &outputTracks) 4710{ 4711 for (size_t i = 0; i < outputTracks.size(); i++) { 4712 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4713 if (thread == 0) { 4714 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4715 outputTracks[i].get()); 4716 return false; 4717 } 4718 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4719 // see note at standby() declaration 4720 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4721 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4722 thread.get()); 4723 return false; 4724 } 4725 } 4726 return true; 4727} 4728 4729uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4730{ 4731 return (mWaitTimeMs * 1000) / 2; 4732} 4733 4734void AudioFlinger::DuplicatingThread::cacheParameters_l() 4735{ 4736 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4737 updateWaitTime_l(); 4738 4739 MixerThread::cacheParameters_l(); 4740} 4741 4742// ---------------------------------------------------------------------------- 4743// Record 4744// ---------------------------------------------------------------------------- 4745 4746AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4747 AudioStreamIn *input, 4748 audio_io_handle_t id, 4749 audio_devices_t outDevice, 4750 audio_devices_t inDevice 4751#ifdef TEE_SINK 4752 , const sp<NBAIO_Sink>& teeSink 4753#endif 4754 ) : 4755 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4756 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4757 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4758 mRsmpInRear(0) 4759#ifdef TEE_SINK 4760 , mTeeSink(teeSink) 4761#endif 4762 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4763 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4764 // mFastCapture below 4765 , mFastCaptureFutex(0) 4766 // mInputSource 4767 // mPipeSink 4768 // mPipeSource 4769 , mPipeFramesP2(0) 4770 // mPipeMemory 4771 // mFastCaptureNBLogWriter 4772 , mFastTrackAvail(true) 4773{ 4774 snprintf(mName, kNameLength, "AudioIn_%X", id); 4775 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4776 4777 readInputParameters_l(); 4778 4779 // create an NBAIO source for the HAL input stream, and negotiate 4780 mInputSource = new AudioStreamInSource(input->stream); 4781 size_t numCounterOffers = 0; 4782 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4783 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4784 ALOG_ASSERT(index == 0); 4785 4786 // initialize fast capture depending on configuration 4787 bool initFastCapture; 4788 switch (kUseFastCapture) { 4789 case FastCapture_Never: 4790 initFastCapture = false; 4791 break; 4792 case FastCapture_Always: 4793 initFastCapture = true; 4794 break; 4795 case FastCapture_Static: 4796 uint32_t primaryOutputSampleRate; 4797 { 4798 AutoMutex _l(audioFlinger->mHardwareLock); 4799 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4800 } 4801 initFastCapture = 4802 // either capture sample rate is same as (a reasonable) primary output sample rate 4803 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4804 (mSampleRate == primaryOutputSampleRate)) || 4805 // or primary output sample rate is unknown, and capture sample rate is reasonable 4806 ((primaryOutputSampleRate == 0) && 4807 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4808 // and the buffer size is < 10 ms 4809 (mFrameCount * 1000) / mSampleRate < 10; 4810 break; 4811 // case FastCapture_Dynamic: 4812 } 4813 4814 if (initFastCapture) { 4815 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4816 NBAIO_Format format = mInputSource->format(); 4817 size_t pipeFramesP2 = roundup(mFrameCount * 8); 4818 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4819 void *pipeBuffer; 4820 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4821 sp<IMemory> pipeMemory; 4822 if ((roHeap == 0) || 4823 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4824 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4825 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4826 goto failed; 4827 } 4828 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4829 memset(pipeBuffer, 0, pipeSize); 4830 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4831 const NBAIO_Format offers[1] = {format}; 4832 size_t numCounterOffers = 0; 4833 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4834 ALOG_ASSERT(index == 0); 4835 mPipeSink = pipe; 4836 PipeReader *pipeReader = new PipeReader(*pipe); 4837 numCounterOffers = 0; 4838 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4839 ALOG_ASSERT(index == 0); 4840 mPipeSource = pipeReader; 4841 mPipeFramesP2 = pipeFramesP2; 4842 mPipeMemory = pipeMemory; 4843 4844 // create fast capture 4845 mFastCapture = new FastCapture(); 4846 FastCaptureStateQueue *sq = mFastCapture->sq(); 4847#ifdef STATE_QUEUE_DUMP 4848 // FIXME 4849#endif 4850 FastCaptureState *state = sq->begin(); 4851 state->mCblk = NULL; 4852 state->mInputSource = mInputSource.get(); 4853 state->mInputSourceGen++; 4854 state->mPipeSink = pipe; 4855 state->mPipeSinkGen++; 4856 state->mFrameCount = mFrameCount; 4857 state->mCommand = FastCaptureState::COLD_IDLE; 4858 // already done in constructor initialization list 4859 //mFastCaptureFutex = 0; 4860 state->mColdFutexAddr = &mFastCaptureFutex; 4861 state->mColdGen++; 4862 state->mDumpState = &mFastCaptureDumpState; 4863#ifdef TEE_SINK 4864 // FIXME 4865#endif 4866 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4867 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4868 sq->end(); 4869 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4870 4871 // start the fast capture 4872 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4873 pid_t tid = mFastCapture->getTid(); 4874 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4875 if (err != 0) { 4876 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4877 kPriorityFastCapture, getpid_cached, tid, err); 4878 } 4879 4880#ifdef AUDIO_WATCHDOG 4881 // FIXME 4882#endif 4883 4884 } 4885failed: ; 4886 4887 // FIXME mNormalSource 4888} 4889 4890 4891AudioFlinger::RecordThread::~RecordThread() 4892{ 4893 if (mFastCapture != 0) { 4894 FastCaptureStateQueue *sq = mFastCapture->sq(); 4895 FastCaptureState *state = sq->begin(); 4896 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4897 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4898 if (old == -1) { 4899 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4900 } 4901 } 4902 state->mCommand = FastCaptureState::EXIT; 4903 sq->end(); 4904 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4905 mFastCapture->join(); 4906 mFastCapture.clear(); 4907 } 4908 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4909 mAudioFlinger->unregisterWriter(mNBLogWriter); 4910 delete[] mRsmpInBuffer; 4911} 4912 4913void AudioFlinger::RecordThread::onFirstRef() 4914{ 4915 run(mName, PRIORITY_URGENT_AUDIO); 4916} 4917 4918bool AudioFlinger::RecordThread::threadLoop() 4919{ 4920 nsecs_t lastWarning = 0; 4921 4922 inputStandBy(); 4923 4924reacquire_wakelock: 4925 sp<RecordTrack> activeTrack; 4926 int activeTracksGen; 4927 { 4928 Mutex::Autolock _l(mLock); 4929 size_t size = mActiveTracks.size(); 4930 activeTracksGen = mActiveTracksGen; 4931 if (size > 0) { 4932 // FIXME an arbitrary choice 4933 activeTrack = mActiveTracks[0]; 4934 acquireWakeLock_l(activeTrack->uid()); 4935 if (size > 1) { 4936 SortedVector<int> tmp; 4937 for (size_t i = 0; i < size; i++) { 4938 tmp.add(mActiveTracks[i]->uid()); 4939 } 4940 updateWakeLockUids_l(tmp); 4941 } 4942 } else { 4943 acquireWakeLock_l(-1); 4944 } 4945 } 4946 4947 // used to request a deferred sleep, to be executed later while mutex is unlocked 4948 uint32_t sleepUs = 0; 4949 4950 // loop while there is work to do 4951 for (;;) { 4952 Vector< sp<EffectChain> > effectChains; 4953 4954 // sleep with mutex unlocked 4955 if (sleepUs > 0) { 4956 usleep(sleepUs); 4957 sleepUs = 0; 4958 } 4959 4960 // activeTracks accumulates a copy of a subset of mActiveTracks 4961 Vector< sp<RecordTrack> > activeTracks; 4962 4963 // reference to the (first and only) fast track 4964 sp<RecordTrack> fastTrack; 4965 4966 { // scope for mLock 4967 Mutex::Autolock _l(mLock); 4968 4969 processConfigEvents_l(); 4970 4971 // check exitPending here because checkForNewParameters_l() and 4972 // checkForNewParameters_l() can temporarily release mLock 4973 if (exitPending()) { 4974 break; 4975 } 4976 4977 // if no active track(s), then standby and release wakelock 4978 size_t size = mActiveTracks.size(); 4979 if (size == 0) { 4980 standbyIfNotAlreadyInStandby(); 4981 // exitPending() can't become true here 4982 releaseWakeLock_l(); 4983 ALOGV("RecordThread: loop stopping"); 4984 // go to sleep 4985 mWaitWorkCV.wait(mLock); 4986 ALOGV("RecordThread: loop starting"); 4987 goto reacquire_wakelock; 4988 } 4989 4990 if (mActiveTracksGen != activeTracksGen) { 4991 activeTracksGen = mActiveTracksGen; 4992 SortedVector<int> tmp; 4993 for (size_t i = 0; i < size; i++) { 4994 tmp.add(mActiveTracks[i]->uid()); 4995 } 4996 updateWakeLockUids_l(tmp); 4997 } 4998 4999 bool doBroadcast = false; 5000 for (size_t i = 0; i < size; ) { 5001 5002 activeTrack = mActiveTracks[i]; 5003 if (activeTrack->isTerminated()) { 5004 removeTrack_l(activeTrack); 5005 mActiveTracks.remove(activeTrack); 5006 mActiveTracksGen++; 5007 size--; 5008 continue; 5009 } 5010 5011 TrackBase::track_state activeTrackState = activeTrack->mState; 5012 switch (activeTrackState) { 5013 5014 case TrackBase::PAUSING: 5015 mActiveTracks.remove(activeTrack); 5016 mActiveTracksGen++; 5017 doBroadcast = true; 5018 size--; 5019 continue; 5020 5021 case TrackBase::STARTING_1: 5022 sleepUs = 10000; 5023 i++; 5024 continue; 5025 5026 case TrackBase::STARTING_2: 5027 doBroadcast = true; 5028 mStandby = false; 5029 activeTrack->mState = TrackBase::ACTIVE; 5030 break; 5031 5032 case TrackBase::ACTIVE: 5033 break; 5034 5035 case TrackBase::IDLE: 5036 i++; 5037 continue; 5038 5039 default: 5040 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5041 } 5042 5043 activeTracks.add(activeTrack); 5044 i++; 5045 5046 if (activeTrack->isFastTrack()) { 5047 ALOG_ASSERT(!mFastTrackAvail); 5048 ALOG_ASSERT(fastTrack == 0); 5049 fastTrack = activeTrack; 5050 } 5051 } 5052 if (doBroadcast) { 5053 mStartStopCond.broadcast(); 5054 } 5055 5056 // sleep if there are no active tracks to process 5057 if (activeTracks.size() == 0) { 5058 if (sleepUs == 0) { 5059 sleepUs = kRecordThreadSleepUs; 5060 } 5061 continue; 5062 } 5063 sleepUs = 0; 5064 5065 lockEffectChains_l(effectChains); 5066 } 5067 5068 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5069 5070 size_t size = effectChains.size(); 5071 for (size_t i = 0; i < size; i++) { 5072 // thread mutex is not locked, but effect chain is locked 5073 effectChains[i]->process_l(); 5074 } 5075 5076 // Start the fast capture if it's not already running 5077 if (mFastCapture != 0) { 5078 FastCaptureStateQueue *sq = mFastCapture->sq(); 5079 FastCaptureState *state = sq->begin(); 5080 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5081 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5082 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5083 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5084 if (old == -1) { 5085 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5086 } 5087 } 5088 state->mCommand = FastCaptureState::READ_WRITE; 5089#if 0 // FIXME 5090 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5091 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5092#endif 5093 state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL; 5094 sq->end(); 5095 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5096#if 0 5097 if (kUseFastCapture == FastCapture_Dynamic) { 5098 mNormalSource = mPipeSource; 5099 } 5100#endif 5101 } else { 5102 sq->end(false /*didModify*/); 5103 } 5104 } 5105 5106 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5107 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5108 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5109 // If destination is non-contiguous, first read past the nominal end of buffer, then 5110 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5111 5112 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5113 ssize_t framesRead; 5114 5115 // If an NBAIO source is present, use it to read the normal capture's data 5116 if (mPipeSource != 0) { 5117 size_t framesToRead = mBufferSize / mFrameSize; 5118 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5119 framesToRead, AudioBufferProvider::kInvalidPTS); 5120 if (framesRead == 0) { 5121 // since pipe is non-blocking, simulate blocking input 5122 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5123 } 5124 // otherwise use the HAL / AudioStreamIn directly 5125 } else { 5126 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5127 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5128 if (bytesRead < 0) { 5129 framesRead = bytesRead; 5130 } else { 5131 framesRead = bytesRead / mFrameSize; 5132 } 5133 } 5134 5135 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5136 ALOGE("read failed: framesRead=%d", framesRead); 5137 // Force input into standby so that it tries to recover at next read attempt 5138 inputStandBy(); 5139 sleepUs = kRecordThreadSleepUs; 5140 } 5141 if (framesRead <= 0) { 5142 continue; 5143 } 5144 ALOG_ASSERT(framesRead > 0); 5145 5146 if (mTeeSink != 0) { 5147 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5148 } 5149 // If destination is non-contiguous, we now correct for reading past end of buffer. 5150 size_t part1 = mRsmpInFramesP2 - rear; 5151 if ((size_t) framesRead > part1) { 5152 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5153 (framesRead - part1) * mFrameSize); 5154 } 5155 rear = mRsmpInRear += framesRead; 5156 5157 size = activeTracks.size(); 5158 // loop over each active track 5159 for (size_t i = 0; i < size; i++) { 5160 activeTrack = activeTracks[i]; 5161 5162 // skip fast tracks, as those are handled directly by FastCapture 5163 if (activeTrack->isFastTrack()) { 5164 continue; 5165 } 5166 5167 enum { 5168 OVERRUN_UNKNOWN, 5169 OVERRUN_TRUE, 5170 OVERRUN_FALSE 5171 } overrun = OVERRUN_UNKNOWN; 5172 5173 // loop over getNextBuffer to handle circular sink 5174 for (;;) { 5175 5176 activeTrack->mSink.frameCount = ~0; 5177 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5178 size_t framesOut = activeTrack->mSink.frameCount; 5179 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5180 5181 int32_t front = activeTrack->mRsmpInFront; 5182 ssize_t filled = rear - front; 5183 size_t framesIn; 5184 5185 if (filled < 0) { 5186 // should not happen, but treat like a massive overrun and re-sync 5187 framesIn = 0; 5188 activeTrack->mRsmpInFront = rear; 5189 overrun = OVERRUN_TRUE; 5190 } else if ((size_t) filled <= mRsmpInFrames) { 5191 framesIn = (size_t) filled; 5192 } else { 5193 // client is not keeping up with server, but give it latest data 5194 framesIn = mRsmpInFrames; 5195 activeTrack->mRsmpInFront = front = rear - framesIn; 5196 overrun = OVERRUN_TRUE; 5197 } 5198 5199 if (framesOut == 0 || framesIn == 0) { 5200 break; 5201 } 5202 5203 if (activeTrack->mResampler == NULL) { 5204 // no resampling 5205 if (framesIn > framesOut) { 5206 framesIn = framesOut; 5207 } else { 5208 framesOut = framesIn; 5209 } 5210 int8_t *dst = activeTrack->mSink.i8; 5211 while (framesIn > 0) { 5212 front &= mRsmpInFramesP2 - 1; 5213 size_t part1 = mRsmpInFramesP2 - front; 5214 if (part1 > framesIn) { 5215 part1 = framesIn; 5216 } 5217 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5218 if (mChannelCount == activeTrack->mChannelCount) { 5219 memcpy(dst, src, part1 * mFrameSize); 5220 } else if (mChannelCount == 1) { 5221 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 5222 part1); 5223 } else { 5224 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 5225 part1); 5226 } 5227 dst += part1 * activeTrack->mFrameSize; 5228 front += part1; 5229 framesIn -= part1; 5230 } 5231 activeTrack->mRsmpInFront += framesOut; 5232 5233 } else { 5234 // resampling 5235 // FIXME framesInNeeded should really be part of resampler API, and should 5236 // depend on the SRC ratio 5237 // to keep mRsmpInBuffer full so resampler always has sufficient input 5238 size_t framesInNeeded; 5239 // FIXME only re-calculate when it changes, and optimize for common ratios 5240 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 5241 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 5242 framesInNeeded = ceil(framesOut * inOverOut) + 1; 5243 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5244 framesInNeeded, framesOut, inOverOut); 5245 // Although we theoretically have framesIn in circular buffer, some of those are 5246 // unreleased frames, and thus must be discounted for purpose of budgeting. 5247 size_t unreleased = activeTrack->mRsmpInUnrel; 5248 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5249 if (framesIn < framesInNeeded) { 5250 ALOGV("not enough to resample: have %u frames in but need %u in to " 5251 "produce %u out given in/out ratio of %.4g", 5252 framesIn, framesInNeeded, framesOut, inOverOut); 5253 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 5254 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5255 if (newFramesOut == 0) { 5256 break; 5257 } 5258 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 5259 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5260 framesInNeeded, newFramesOut, outOverIn); 5261 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5262 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5263 "given in/out ratio of %.4g", 5264 framesIn, framesInNeeded, newFramesOut, inOverOut); 5265 framesOut = newFramesOut; 5266 } else { 5267 ALOGV("success 1: have %u in and need %u in to produce %u out " 5268 "given in/out ratio of %.4g", 5269 framesIn, framesInNeeded, framesOut, inOverOut); 5270 } 5271 5272 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5273 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5274 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5275 delete[] activeTrack->mRsmpOutBuffer; 5276 // resampler always outputs stereo 5277 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5278 activeTrack->mRsmpOutFrameCount = framesOut; 5279 } 5280 5281 // resampler accumulates, but we only have one source track 5282 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5283 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5284 // FIXME how about having activeTrack implement this interface itself? 5285 activeTrack->mResamplerBufferProvider 5286 /*this*/ /* AudioBufferProvider* */); 5287 // ditherAndClamp() works as long as all buffers returned by 5288 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5289 if (activeTrack->mChannelCount == 1) { 5290 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5291 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5292 framesOut); 5293 // the resampler always outputs stereo samples: 5294 // do post stereo to mono conversion 5295 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5296 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5297 } else { 5298 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5299 activeTrack->mRsmpOutBuffer, framesOut); 5300 } 5301 // now done with mRsmpOutBuffer 5302 5303 } 5304 5305 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5306 overrun = OVERRUN_FALSE; 5307 } 5308 5309 if (activeTrack->mFramesToDrop == 0) { 5310 if (framesOut > 0) { 5311 activeTrack->mSink.frameCount = framesOut; 5312 activeTrack->releaseBuffer(&activeTrack->mSink); 5313 } 5314 } else { 5315 // FIXME could do a partial drop of framesOut 5316 if (activeTrack->mFramesToDrop > 0) { 5317 activeTrack->mFramesToDrop -= framesOut; 5318 if (activeTrack->mFramesToDrop <= 0) { 5319 activeTrack->clearSyncStartEvent(); 5320 } 5321 } else { 5322 activeTrack->mFramesToDrop += framesOut; 5323 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5324 activeTrack->mSyncStartEvent->isCancelled()) { 5325 ALOGW("Synced record %s, session %d, trigger session %d", 5326 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5327 activeTrack->sessionId(), 5328 (activeTrack->mSyncStartEvent != 0) ? 5329 activeTrack->mSyncStartEvent->triggerSession() : 0); 5330 activeTrack->clearSyncStartEvent(); 5331 } 5332 } 5333 } 5334 5335 if (framesOut == 0) { 5336 break; 5337 } 5338 } 5339 5340 switch (overrun) { 5341 case OVERRUN_TRUE: 5342 // client isn't retrieving buffers fast enough 5343 if (!activeTrack->setOverflow()) { 5344 nsecs_t now = systemTime(); 5345 // FIXME should lastWarning per track? 5346 if ((now - lastWarning) > kWarningThrottleNs) { 5347 ALOGW("RecordThread: buffer overflow"); 5348 lastWarning = now; 5349 } 5350 } 5351 break; 5352 case OVERRUN_FALSE: 5353 activeTrack->clearOverflow(); 5354 break; 5355 case OVERRUN_UNKNOWN: 5356 break; 5357 } 5358 5359 } 5360 5361 // enable changes in effect chain 5362 unlockEffectChains(effectChains); 5363 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5364 } 5365 5366 standbyIfNotAlreadyInStandby(); 5367 5368 { 5369 Mutex::Autolock _l(mLock); 5370 for (size_t i = 0; i < mTracks.size(); i++) { 5371 sp<RecordTrack> track = mTracks[i]; 5372 track->invalidate(); 5373 } 5374 mActiveTracks.clear(); 5375 mActiveTracksGen++; 5376 mStartStopCond.broadcast(); 5377 } 5378 5379 releaseWakeLock(); 5380 5381 ALOGV("RecordThread %p exiting", this); 5382 return false; 5383} 5384 5385void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5386{ 5387 if (!mStandby) { 5388 inputStandBy(); 5389 mStandby = true; 5390 } 5391} 5392 5393void AudioFlinger::RecordThread::inputStandBy() 5394{ 5395 // Idle the fast capture if it's currently running 5396 if (mFastCapture != 0) { 5397 FastCaptureStateQueue *sq = mFastCapture->sq(); 5398 FastCaptureState *state = sq->begin(); 5399 if (!(state->mCommand & FastCaptureState::IDLE)) { 5400 state->mCommand = FastCaptureState::COLD_IDLE; 5401 state->mColdFutexAddr = &mFastCaptureFutex; 5402 state->mColdGen++; 5403 mFastCaptureFutex = 0; 5404 sq->end(); 5405 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5406 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5407#if 0 5408 if (kUseFastCapture == FastCapture_Dynamic) { 5409 // FIXME 5410 } 5411#endif 5412#ifdef AUDIO_WATCHDOG 5413 // FIXME 5414#endif 5415 } else { 5416 sq->end(false /*didModify*/); 5417 } 5418 } 5419 mInput->stream->common.standby(&mInput->stream->common); 5420} 5421 5422// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5423sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5424 const sp<AudioFlinger::Client>& client, 5425 uint32_t sampleRate, 5426 audio_format_t format, 5427 audio_channel_mask_t channelMask, 5428 size_t *pFrameCount, 5429 int sessionId, 5430 int uid, 5431 IAudioFlinger::track_flags_t *flags, 5432 pid_t tid, 5433 status_t *status) 5434{ 5435 size_t frameCount = *pFrameCount; 5436 sp<RecordTrack> track; 5437 status_t lStatus; 5438 5439 // client expresses a preference for FAST, but we get the final say 5440 if (*flags & IAudioFlinger::TRACK_FAST) { 5441 if ( 5442 // use case: callback handler and frame count is default or at least as large as HAL 5443 ( 5444 (tid != -1) && 5445 ((frameCount == 0) /*|| 5446 // FIXME must be equal to pipe depth, so don't allow it to be specified by client 5447 // FIXME not necessarily true, should be native frame count for native SR! 5448 (frameCount >= mFrameCount)*/) 5449 ) && 5450 // PCM data 5451 audio_is_linear_pcm(format) && 5452 // native format 5453 (format == mFormat) && 5454 // mono or stereo 5455 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5456 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5457 // native channel mask 5458 (channelMask == mChannelMask) && 5459 // native hardware sample rate 5460 (sampleRate == mSampleRate) && 5461 // record thread has an associated fast capture 5462 hasFastCapture() && 5463 // there are sufficient fast track slots available 5464 mFastTrackAvail 5465 ) { 5466 // if frameCount not specified, then it defaults to pipe frame count 5467 if (frameCount == 0) { 5468 frameCount = mPipeFramesP2; 5469 } 5470 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5471 frameCount, mFrameCount); 5472 } else { 5473 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5474 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5475 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5476 frameCount, mFrameCount, format, 5477 audio_is_linear_pcm(format), 5478 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid, mFastTrackAvail); 5479 *flags &= ~IAudioFlinger::TRACK_FAST; 5480 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5481 // For compatibility with AudioRecord calculation, buffer depth is forced 5482 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5483 // This is probably too conservative, but legacy application code may depend on it. 5484 // If you change this calculation, also review the start threshold which is related. 5485 // FIXME It's not clear how input latency actually matters. Perhaps this should be 0. 5486 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5487 size_t mNormalFrameCount = 2048; // FIXME 5488 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5489 if (minBufCount < 2) { 5490 minBufCount = 2; 5491 } 5492 size_t minFrameCount = mNormalFrameCount * minBufCount; 5493 if (frameCount < minFrameCount) { 5494 frameCount = minFrameCount; 5495 } 5496 } 5497 } 5498 *pFrameCount = frameCount; 5499 5500 lStatus = initCheck(); 5501 if (lStatus != NO_ERROR) { 5502 ALOGE("createRecordTrack_l() audio driver not initialized"); 5503 goto Exit; 5504 } 5505 5506 { // scope for mLock 5507 Mutex::Autolock _l(mLock); 5508 5509 track = new RecordTrack(this, client, sampleRate, 5510 format, channelMask, frameCount, sessionId, uid, 5511 *flags); 5512 5513 lStatus = track->initCheck(); 5514 if (lStatus != NO_ERROR) { 5515 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5516 // track must be cleared from the caller as the caller has the AF lock 5517 goto Exit; 5518 } 5519 mTracks.add(track); 5520 5521 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5522 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5523 mAudioFlinger->btNrecIsOff(); 5524 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5525 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5526 5527 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5528 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5529 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5530 // so ask activity manager to do this on our behalf 5531 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5532 } 5533 } 5534 5535 lStatus = NO_ERROR; 5536 5537Exit: 5538 *status = lStatus; 5539 return track; 5540} 5541 5542status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5543 AudioSystem::sync_event_t event, 5544 int triggerSession) 5545{ 5546 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5547 sp<ThreadBase> strongMe = this; 5548 status_t status = NO_ERROR; 5549 5550 if (event == AudioSystem::SYNC_EVENT_NONE) { 5551 recordTrack->clearSyncStartEvent(); 5552 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5553 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5554 triggerSession, 5555 recordTrack->sessionId(), 5556 syncStartEventCallback, 5557 recordTrack); 5558 // Sync event can be cancelled by the trigger session if the track is not in a 5559 // compatible state in which case we start record immediately 5560 if (recordTrack->mSyncStartEvent->isCancelled()) { 5561 recordTrack->clearSyncStartEvent(); 5562 } else { 5563 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5564 recordTrack->mFramesToDrop = - 5565 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5566 } 5567 } 5568 5569 { 5570 // This section is a rendezvous between binder thread executing start() and RecordThread 5571 AutoMutex lock(mLock); 5572 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5573 if (recordTrack->mState == TrackBase::PAUSING) { 5574 ALOGV("active record track PAUSING -> ACTIVE"); 5575 recordTrack->mState = TrackBase::ACTIVE; 5576 } else { 5577 ALOGV("active record track state %d", recordTrack->mState); 5578 } 5579 return status; 5580 } 5581 5582 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5583 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5584 // or using a separate command thread 5585 recordTrack->mState = TrackBase::STARTING_1; 5586 mActiveTracks.add(recordTrack); 5587 mActiveTracksGen++; 5588 mLock.unlock(); 5589 status_t status = AudioSystem::startInput(mId); 5590 mLock.lock(); 5591 // FIXME should verify that recordTrack is still in mActiveTracks 5592 if (status != NO_ERROR) { 5593 mActiveTracks.remove(recordTrack); 5594 mActiveTracksGen++; 5595 recordTrack->clearSyncStartEvent(); 5596 return status; 5597 } 5598 // Catch up with current buffer indices if thread is already running. 5599 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5600 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5601 // see previously buffered data before it called start(), but with greater risk of overrun. 5602 5603 recordTrack->mRsmpInFront = mRsmpInRear; 5604 recordTrack->mRsmpInUnrel = 0; 5605 // FIXME why reset? 5606 if (recordTrack->mResampler != NULL) { 5607 recordTrack->mResampler->reset(); 5608 } 5609 recordTrack->mState = TrackBase::STARTING_2; 5610 // signal thread to start 5611 mWaitWorkCV.broadcast(); 5612 if (mActiveTracks.indexOf(recordTrack) < 0) { 5613 ALOGV("Record failed to start"); 5614 status = BAD_VALUE; 5615 goto startError; 5616 } 5617 return status; 5618 } 5619 5620startError: 5621 AudioSystem::stopInput(mId); 5622 recordTrack->clearSyncStartEvent(); 5623 // FIXME I wonder why we do not reset the state here? 5624 return status; 5625} 5626 5627void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5628{ 5629 sp<SyncEvent> strongEvent = event.promote(); 5630 5631 if (strongEvent != 0) { 5632 sp<RefBase> ptr = strongEvent->cookie().promote(); 5633 if (ptr != 0) { 5634 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5635 recordTrack->handleSyncStartEvent(strongEvent); 5636 } 5637 } 5638} 5639 5640bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5641 ALOGV("RecordThread::stop"); 5642 AutoMutex _l(mLock); 5643 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5644 return false; 5645 } 5646 // note that threadLoop may still be processing the track at this point [without lock] 5647 recordTrack->mState = TrackBase::PAUSING; 5648 // do not wait for mStartStopCond if exiting 5649 if (exitPending()) { 5650 return true; 5651 } 5652 // FIXME incorrect usage of wait: no explicit predicate or loop 5653 mStartStopCond.wait(mLock); 5654 // if we have been restarted, recordTrack is in mActiveTracks here 5655 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5656 ALOGV("Record stopped OK"); 5657 return true; 5658 } 5659 return false; 5660} 5661 5662bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5663{ 5664 return false; 5665} 5666 5667status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5668{ 5669#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5670 if (!isValidSyncEvent(event)) { 5671 return BAD_VALUE; 5672 } 5673 5674 int eventSession = event->triggerSession(); 5675 status_t ret = NAME_NOT_FOUND; 5676 5677 Mutex::Autolock _l(mLock); 5678 5679 for (size_t i = 0; i < mTracks.size(); i++) { 5680 sp<RecordTrack> track = mTracks[i]; 5681 if (eventSession == track->sessionId()) { 5682 (void) track->setSyncEvent(event); 5683 ret = NO_ERROR; 5684 } 5685 } 5686 return ret; 5687#else 5688 return BAD_VALUE; 5689#endif 5690} 5691 5692// destroyTrack_l() must be called with ThreadBase::mLock held 5693void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5694{ 5695 track->terminate(); 5696 track->mState = TrackBase::STOPPED; 5697 // active tracks are removed by threadLoop() 5698 if (mActiveTracks.indexOf(track) < 0) { 5699 removeTrack_l(track); 5700 } 5701} 5702 5703void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5704{ 5705 mTracks.remove(track); 5706 // need anything related to effects here? 5707 if (track->isFastTrack()) { 5708 ALOG_ASSERT(!mFastTrackAvail); 5709 mFastTrackAvail = true; 5710 } 5711} 5712 5713void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5714{ 5715 dumpInternals(fd, args); 5716 dumpTracks(fd, args); 5717 dumpEffectChains(fd, args); 5718} 5719 5720void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5721{ 5722 dprintf(fd, "\nInput thread %p:\n", this); 5723 5724 if (mActiveTracks.size() > 0) { 5725 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5726 } else { 5727 dprintf(fd, " No active record clients\n"); 5728 } 5729 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5730 5731 dumpBase(fd, args); 5732} 5733 5734void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5735{ 5736 const size_t SIZE = 256; 5737 char buffer[SIZE]; 5738 String8 result; 5739 5740 size_t numtracks = mTracks.size(); 5741 size_t numactive = mActiveTracks.size(); 5742 size_t numactiveseen = 0; 5743 dprintf(fd, " %d Tracks", numtracks); 5744 if (numtracks) { 5745 dprintf(fd, " of which %d are active\n", numactive); 5746 RecordTrack::appendDumpHeader(result); 5747 for (size_t i = 0; i < numtracks ; ++i) { 5748 sp<RecordTrack> track = mTracks[i]; 5749 if (track != 0) { 5750 bool active = mActiveTracks.indexOf(track) >= 0; 5751 if (active) { 5752 numactiveseen++; 5753 } 5754 track->dump(buffer, SIZE, active); 5755 result.append(buffer); 5756 } 5757 } 5758 } else { 5759 dprintf(fd, "\n"); 5760 } 5761 5762 if (numactiveseen != numactive) { 5763 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5764 " not in the track list\n"); 5765 result.append(buffer); 5766 RecordTrack::appendDumpHeader(result); 5767 for (size_t i = 0; i < numactive; ++i) { 5768 sp<RecordTrack> track = mActiveTracks[i]; 5769 if (mTracks.indexOf(track) < 0) { 5770 track->dump(buffer, SIZE, true); 5771 result.append(buffer); 5772 } 5773 } 5774 5775 } 5776 write(fd, result.string(), result.size()); 5777} 5778 5779// AudioBufferProvider interface 5780status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5781 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5782{ 5783 RecordTrack *activeTrack = mRecordTrack; 5784 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5785 if (threadBase == 0) { 5786 buffer->frameCount = 0; 5787 buffer->raw = NULL; 5788 return NOT_ENOUGH_DATA; 5789 } 5790 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5791 int32_t rear = recordThread->mRsmpInRear; 5792 int32_t front = activeTrack->mRsmpInFront; 5793 ssize_t filled = rear - front; 5794 // FIXME should not be P2 (don't want to increase latency) 5795 // FIXME if client not keeping up, discard 5796 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5797 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5798 front &= recordThread->mRsmpInFramesP2 - 1; 5799 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5800 if (part1 > (size_t) filled) { 5801 part1 = filled; 5802 } 5803 size_t ask = buffer->frameCount; 5804 ALOG_ASSERT(ask > 0); 5805 if (part1 > ask) { 5806 part1 = ask; 5807 } 5808 if (part1 == 0) { 5809 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5810 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5811 buffer->raw = NULL; 5812 buffer->frameCount = 0; 5813 activeTrack->mRsmpInUnrel = 0; 5814 return NOT_ENOUGH_DATA; 5815 } 5816 5817 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5818 buffer->frameCount = part1; 5819 activeTrack->mRsmpInUnrel = part1; 5820 return NO_ERROR; 5821} 5822 5823// AudioBufferProvider interface 5824void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5825 AudioBufferProvider::Buffer* buffer) 5826{ 5827 RecordTrack *activeTrack = mRecordTrack; 5828 size_t stepCount = buffer->frameCount; 5829 if (stepCount == 0) { 5830 return; 5831 } 5832 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5833 activeTrack->mRsmpInUnrel -= stepCount; 5834 activeTrack->mRsmpInFront += stepCount; 5835 buffer->raw = NULL; 5836 buffer->frameCount = 0; 5837} 5838 5839bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5840 status_t& status) 5841{ 5842 bool reconfig = false; 5843 5844 status = NO_ERROR; 5845 5846 audio_format_t reqFormat = mFormat; 5847 uint32_t samplingRate = mSampleRate; 5848 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5849 5850 AudioParameter param = AudioParameter(keyValuePair); 5851 int value; 5852 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5853 // channel count change can be requested. Do we mandate the first client defines the 5854 // HAL sampling rate and channel count or do we allow changes on the fly? 5855 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5856 samplingRate = value; 5857 reconfig = true; 5858 } 5859 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5860 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5861 status = BAD_VALUE; 5862 } else { 5863 reqFormat = (audio_format_t) value; 5864 reconfig = true; 5865 } 5866 } 5867 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5868 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5869 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5870 status = BAD_VALUE; 5871 } else { 5872 channelMask = mask; 5873 reconfig = true; 5874 } 5875 } 5876 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5877 // do not accept frame count changes if tracks are open as the track buffer 5878 // size depends on frame count and correct behavior would not be guaranteed 5879 // if frame count is changed after track creation 5880 if (mActiveTracks.size() > 0) { 5881 status = INVALID_OPERATION; 5882 } else { 5883 reconfig = true; 5884 } 5885 } 5886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5887 // forward device change to effects that have requested to be 5888 // aware of attached audio device. 5889 for (size_t i = 0; i < mEffectChains.size(); i++) { 5890 mEffectChains[i]->setDevice_l(value); 5891 } 5892 5893 // store input device and output device but do not forward output device to audio HAL. 5894 // Note that status is ignored by the caller for output device 5895 // (see AudioFlinger::setParameters() 5896 if (audio_is_output_devices(value)) { 5897 mOutDevice = value; 5898 status = BAD_VALUE; 5899 } else { 5900 mInDevice = value; 5901 // disable AEC and NS if the device is a BT SCO headset supporting those 5902 // pre processings 5903 if (mTracks.size() > 0) { 5904 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5905 mAudioFlinger->btNrecIsOff(); 5906 for (size_t i = 0; i < mTracks.size(); i++) { 5907 sp<RecordTrack> track = mTracks[i]; 5908 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5909 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5910 } 5911 } 5912 } 5913 } 5914 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5915 mAudioSource != (audio_source_t)value) { 5916 // forward device change to effects that have requested to be 5917 // aware of attached audio device. 5918 for (size_t i = 0; i < mEffectChains.size(); i++) { 5919 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5920 } 5921 mAudioSource = (audio_source_t)value; 5922 } 5923 5924 if (status == NO_ERROR) { 5925 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5926 keyValuePair.string()); 5927 if (status == INVALID_OPERATION) { 5928 inputStandBy(); 5929 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5930 keyValuePair.string()); 5931 } 5932 if (reconfig) { 5933 if (status == BAD_VALUE && 5934 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5935 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5936 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5937 <= (2 * samplingRate)) && 5938 audio_channel_count_from_in_mask( 5939 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5940 (channelMask == AUDIO_CHANNEL_IN_MONO || 5941 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5942 status = NO_ERROR; 5943 } 5944 if (status == NO_ERROR) { 5945 readInputParameters_l(); 5946 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5947 } 5948 } 5949 } 5950 5951 return reconfig; 5952} 5953 5954String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5955{ 5956 Mutex::Autolock _l(mLock); 5957 if (initCheck() != NO_ERROR) { 5958 return String8(); 5959 } 5960 5961 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5962 const String8 out_s8(s); 5963 free(s); 5964 return out_s8; 5965} 5966 5967void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5968 AudioSystem::OutputDescriptor desc; 5969 const void *param2 = NULL; 5970 5971 switch (event) { 5972 case AudioSystem::INPUT_OPENED: 5973 case AudioSystem::INPUT_CONFIG_CHANGED: 5974 desc.channelMask = mChannelMask; 5975 desc.samplingRate = mSampleRate; 5976 desc.format = mFormat; 5977 desc.frameCount = mFrameCount; 5978 desc.latency = 0; 5979 param2 = &desc; 5980 break; 5981 5982 case AudioSystem::INPUT_CLOSED: 5983 default: 5984 break; 5985 } 5986 mAudioFlinger->audioConfigChanged(event, mId, param2); 5987} 5988 5989void AudioFlinger::RecordThread::readInputParameters_l() 5990{ 5991 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5992 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5993 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 5994 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5995 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5996 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5997 } 5998 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5999 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6000 mFrameCount = mBufferSize / mFrameSize; 6001 // This is the formula for calculating the temporary buffer size. 6002 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6003 // 1 full output buffer, regardless of the alignment of the available input. 6004 // The value is somewhat arbitrary, and could probably be even larger. 6005 // A larger value should allow more old data to be read after a track calls start(), 6006 // without increasing latency. 6007 mRsmpInFrames = mFrameCount * 7; 6008 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6009 delete[] mRsmpInBuffer; 6010 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6011 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6012 6013 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6014 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6015} 6016 6017uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6018{ 6019 Mutex::Autolock _l(mLock); 6020 if (initCheck() != NO_ERROR) { 6021 return 0; 6022 } 6023 6024 return mInput->stream->get_input_frames_lost(mInput->stream); 6025} 6026 6027uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6028{ 6029 Mutex::Autolock _l(mLock); 6030 uint32_t result = 0; 6031 if (getEffectChain_l(sessionId) != 0) { 6032 result = EFFECT_SESSION; 6033 } 6034 6035 for (size_t i = 0; i < mTracks.size(); ++i) { 6036 if (sessionId == mTracks[i]->sessionId()) { 6037 result |= TRACK_SESSION; 6038 break; 6039 } 6040 } 6041 6042 return result; 6043} 6044 6045KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6046{ 6047 KeyedVector<int, bool> ids; 6048 Mutex::Autolock _l(mLock); 6049 for (size_t j = 0; j < mTracks.size(); ++j) { 6050 sp<RecordThread::RecordTrack> track = mTracks[j]; 6051 int sessionId = track->sessionId(); 6052 if (ids.indexOfKey(sessionId) < 0) { 6053 ids.add(sessionId, true); 6054 } 6055 } 6056 return ids; 6057} 6058 6059AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6060{ 6061 Mutex::Autolock _l(mLock); 6062 AudioStreamIn *input = mInput; 6063 mInput = NULL; 6064 return input; 6065} 6066 6067// this method must always be called either with ThreadBase mLock held or inside the thread loop 6068audio_stream_t* AudioFlinger::RecordThread::stream() const 6069{ 6070 if (mInput == NULL) { 6071 return NULL; 6072 } 6073 return &mInput->stream->common; 6074} 6075 6076status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6077{ 6078 // only one chain per input thread 6079 if (mEffectChains.size() != 0) { 6080 return INVALID_OPERATION; 6081 } 6082 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6083 6084 chain->setInBuffer(NULL); 6085 chain->setOutBuffer(NULL); 6086 6087 checkSuspendOnAddEffectChain_l(chain); 6088 6089 mEffectChains.add(chain); 6090 6091 return NO_ERROR; 6092} 6093 6094size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6095{ 6096 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6097 ALOGW_IF(mEffectChains.size() != 1, 6098 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6099 chain.get(), mEffectChains.size(), this); 6100 if (mEffectChains.size() == 1) { 6101 mEffectChains.removeAt(0); 6102 } 6103 return 0; 6104} 6105 6106status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6107 audio_patch_handle_t *handle) 6108{ 6109 status_t status = NO_ERROR; 6110 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6111 // store new device and send to effects 6112 mInDevice = patch->sources[0].ext.device.type; 6113 for (size_t i = 0; i < mEffectChains.size(); i++) { 6114 mEffectChains[i]->setDevice_l(mInDevice); 6115 } 6116 6117 // disable AEC and NS if the device is a BT SCO headset supporting those 6118 // pre processings 6119 if (mTracks.size() > 0) { 6120 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6121 mAudioFlinger->btNrecIsOff(); 6122 for (size_t i = 0; i < mTracks.size(); i++) { 6123 sp<RecordTrack> track = mTracks[i]; 6124 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6125 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6126 } 6127 } 6128 6129 // store new source and send to effects 6130 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6131 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6132 for (size_t i = 0; i < mEffectChains.size(); i++) { 6133 mEffectChains[i]->setAudioSource_l(mAudioSource); 6134 } 6135 } 6136 6137 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6138 status = hwDevice->create_audio_patch(hwDevice, 6139 patch->num_sources, 6140 patch->sources, 6141 patch->num_sinks, 6142 patch->sinks, 6143 handle); 6144 } else { 6145 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6146 } 6147 return status; 6148} 6149 6150status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6151{ 6152 status_t status = NO_ERROR; 6153 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6154 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6155 status = hwDevice->release_audio_patch(hwDevice, handle); 6156 } else { 6157 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6158 } 6159 return status; 6160} 6161 6162 6163}; // namespace android 6164