Threads.cpp revision 9f81de3452dfb2385bd57dc05456a045174a1ab1
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 = 0x2000; 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", mHALFormat, formatToString(mHALFormat)); 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 // Reject any effect on multichannel sinks. 914 // TODO: fix both format and multichannel issues with effects. 915 if (mChannelCount != FCC_2) { 916 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) thread", 917 desc->name, mChannelCount); 918 lStatus = BAD_VALUE; 919 goto Exit; 920 } 921 922 // Allow global effects only on offloaded and mixer threads 923 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 924 switch (mType) { 925 case MIXER: 926 case OFFLOAD: 927 break; 928 case DIRECT: 929 case DUPLICATING: 930 case RECORD: 931 default: 932 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 933 lStatus = BAD_VALUE; 934 goto Exit; 935 } 936 } 937 938 // Only Pre processor effects are allowed on input threads and only on input threads 939 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 940 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 941 desc->name, desc->flags, mType); 942 lStatus = BAD_VALUE; 943 goto Exit; 944 } 945 946 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 947 948 { // scope for mLock 949 Mutex::Autolock _l(mLock); 950 951 // check for existing effect chain with the requested audio session 952 chain = getEffectChain_l(sessionId); 953 if (chain == 0) { 954 // create a new chain for this session 955 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 956 chain = new EffectChain(this, sessionId); 957 addEffectChain_l(chain); 958 chain->setStrategy(getStrategyForSession_l(sessionId)); 959 chainCreated = true; 960 } else { 961 effect = chain->getEffectFromDesc_l(desc); 962 } 963 964 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 965 966 if (effect == 0) { 967 int id = mAudioFlinger->nextUniqueId(); 968 // Check CPU and memory usage 969 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 970 if (lStatus != NO_ERROR) { 971 goto Exit; 972 } 973 effectRegistered = true; 974 // create a new effect module if none present in the chain 975 effect = new EffectModule(this, chain, desc, id, sessionId); 976 lStatus = effect->status(); 977 if (lStatus != NO_ERROR) { 978 goto Exit; 979 } 980 effect->setOffloaded(mType == OFFLOAD, mId); 981 982 lStatus = chain->addEffect_l(effect); 983 if (lStatus != NO_ERROR) { 984 goto Exit; 985 } 986 effectCreated = true; 987 988 effect->setDevice(mOutDevice); 989 effect->setDevice(mInDevice); 990 effect->setMode(mAudioFlinger->getMode()); 991 effect->setAudioSource(mAudioSource); 992 } 993 // create effect handle and connect it to effect module 994 handle = new EffectHandle(effect, client, effectClient, priority); 995 lStatus = handle->initCheck(); 996 if (lStatus == OK) { 997 lStatus = effect->addHandle(handle.get()); 998 } 999 if (enabled != NULL) { 1000 *enabled = (int)effect->isEnabled(); 1001 } 1002 } 1003 1004Exit: 1005 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1006 Mutex::Autolock _l(mLock); 1007 if (effectCreated) { 1008 chain->removeEffect_l(effect); 1009 } 1010 if (effectRegistered) { 1011 AudioSystem::unregisterEffect(effect->id()); 1012 } 1013 if (chainCreated) { 1014 removeEffectChain_l(chain); 1015 } 1016 handle.clear(); 1017 } 1018 1019 *status = lStatus; 1020 return handle; 1021} 1022 1023sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1024{ 1025 Mutex::Autolock _l(mLock); 1026 return getEffect_l(sessionId, effectId); 1027} 1028 1029sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1030{ 1031 sp<EffectChain> chain = getEffectChain_l(sessionId); 1032 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1033} 1034 1035// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1036// PlaybackThread::mLock held 1037status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1038{ 1039 // check for existing effect chain with the requested audio session 1040 int sessionId = effect->sessionId(); 1041 sp<EffectChain> chain = getEffectChain_l(sessionId); 1042 bool chainCreated = false; 1043 1044 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1045 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1046 this, effect->desc().name, effect->desc().flags); 1047 1048 if (chain == 0) { 1049 // create a new chain for this session 1050 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1051 chain = new EffectChain(this, sessionId); 1052 addEffectChain_l(chain); 1053 chain->setStrategy(getStrategyForSession_l(sessionId)); 1054 chainCreated = true; 1055 } 1056 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1057 1058 if (chain->getEffectFromId_l(effect->id()) != 0) { 1059 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1060 this, effect->desc().name, chain.get()); 1061 return BAD_VALUE; 1062 } 1063 1064 effect->setOffloaded(mType == OFFLOAD, mId); 1065 1066 status_t status = chain->addEffect_l(effect); 1067 if (status != NO_ERROR) { 1068 if (chainCreated) { 1069 removeEffectChain_l(chain); 1070 } 1071 return status; 1072 } 1073 1074 effect->setDevice(mOutDevice); 1075 effect->setDevice(mInDevice); 1076 effect->setMode(mAudioFlinger->getMode()); 1077 effect->setAudioSource(mAudioSource); 1078 return NO_ERROR; 1079} 1080 1081void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1082 1083 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1084 effect_descriptor_t desc = effect->desc(); 1085 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1086 detachAuxEffect_l(effect->id()); 1087 } 1088 1089 sp<EffectChain> chain = effect->chain().promote(); 1090 if (chain != 0) { 1091 // remove effect chain if removing last effect 1092 if (chain->removeEffect_l(effect) == 0) { 1093 removeEffectChain_l(chain); 1094 } 1095 } else { 1096 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1097 } 1098} 1099 1100void AudioFlinger::ThreadBase::lockEffectChains_l( 1101 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1102{ 1103 effectChains = mEffectChains; 1104 for (size_t i = 0; i < mEffectChains.size(); i++) { 1105 mEffectChains[i]->lock(); 1106 } 1107} 1108 1109void AudioFlinger::ThreadBase::unlockEffectChains( 1110 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1111{ 1112 for (size_t i = 0; i < effectChains.size(); i++) { 1113 effectChains[i]->unlock(); 1114 } 1115} 1116 1117sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1118{ 1119 Mutex::Autolock _l(mLock); 1120 return getEffectChain_l(sessionId); 1121} 1122 1123sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1124{ 1125 size_t size = mEffectChains.size(); 1126 for (size_t i = 0; i < size; i++) { 1127 if (mEffectChains[i]->sessionId() == sessionId) { 1128 return mEffectChains[i]; 1129 } 1130 } 1131 return 0; 1132} 1133 1134void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1135{ 1136 Mutex::Autolock _l(mLock); 1137 size_t size = mEffectChains.size(); 1138 for (size_t i = 0; i < size; i++) { 1139 mEffectChains[i]->setMode_l(mode); 1140 } 1141} 1142 1143void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1144 EffectHandle *handle, 1145 bool unpinIfLast) { 1146 1147 Mutex::Autolock _l(mLock); 1148 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1149 // delete the effect module if removing last handle on it 1150 if (effect->removeHandle(handle) == 0) { 1151 if (!effect->isPinned() || unpinIfLast) { 1152 removeEffect_l(effect); 1153 AudioSystem::unregisterEffect(effect->id()); 1154 } 1155 } 1156} 1157 1158void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1159{ 1160 config->type = AUDIO_PORT_TYPE_MIX; 1161 config->ext.mix.handle = mId; 1162 config->sample_rate = mSampleRate; 1163 config->format = mFormat; 1164 config->channel_mask = mChannelMask; 1165 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1166 AUDIO_PORT_CONFIG_FORMAT; 1167} 1168 1169 1170// ---------------------------------------------------------------------------- 1171// Playback 1172// ---------------------------------------------------------------------------- 1173 1174AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1175 AudioStreamOut* output, 1176 audio_io_handle_t id, 1177 audio_devices_t device, 1178 type_t type) 1179 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1180 mNormalFrameCount(0), mSinkBuffer(NULL), 1181 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1182 mMixerBuffer(NULL), 1183 mMixerBufferSize(0), 1184 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1185 mMixerBufferValid(false), 1186 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1187 mEffectBuffer(NULL), 1188 mEffectBufferSize(0), 1189 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1190 mEffectBufferValid(false), 1191 mSuspended(0), mBytesWritten(0), 1192 mActiveTracksGeneration(0), 1193 // mStreamTypes[] initialized in constructor body 1194 mOutput(output), 1195 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1196 mMixerStatus(MIXER_IDLE), 1197 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1198 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1199 mBytesRemaining(0), 1200 mCurrentWriteLength(0), 1201 mUseAsyncWrite(false), 1202 mWriteAckSequence(0), 1203 mDrainSequence(0), 1204 mSignalPending(false), 1205 mScreenState(AudioFlinger::mScreenState), 1206 // index 0 is reserved for normal mixer's submix 1207 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1208 // mLatchD, mLatchQ, 1209 mLatchDValid(false), mLatchQValid(false) 1210{ 1211 snprintf(mName, kNameLength, "AudioOut_%X", id); 1212 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1213 1214 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1215 // it would be safer to explicitly pass initial masterVolume/masterMute as 1216 // parameter. 1217 // 1218 // If the HAL we are using has support for master volume or master mute, 1219 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1220 // and the mute set to false). 1221 mMasterVolume = audioFlinger->masterVolume_l(); 1222 mMasterMute = audioFlinger->masterMute_l(); 1223 if (mOutput && mOutput->audioHwDev) { 1224 if (mOutput->audioHwDev->canSetMasterVolume()) { 1225 mMasterVolume = 1.0; 1226 } 1227 1228 if (mOutput->audioHwDev->canSetMasterMute()) { 1229 mMasterMute = false; 1230 } 1231 } 1232 1233 readOutputParameters_l(); 1234 1235 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1236 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1237 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1238 stream = (audio_stream_type_t) (stream + 1)) { 1239 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1240 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1241 } 1242 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1243 // because mAudioFlinger doesn't have one to copy from 1244} 1245 1246AudioFlinger::PlaybackThread::~PlaybackThread() 1247{ 1248 mAudioFlinger->unregisterWriter(mNBLogWriter); 1249 free(mSinkBuffer); 1250 free(mMixerBuffer); 1251 free(mEffectBuffer); 1252} 1253 1254void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1255{ 1256 dumpInternals(fd, args); 1257 dumpTracks(fd, args); 1258 dumpEffectChains(fd, args); 1259} 1260 1261void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1262{ 1263 const size_t SIZE = 256; 1264 char buffer[SIZE]; 1265 String8 result; 1266 1267 result.appendFormat(" Stream volumes in dB: "); 1268 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1269 const stream_type_t *st = &mStreamTypes[i]; 1270 if (i > 0) { 1271 result.appendFormat(", "); 1272 } 1273 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1274 if (st->mute) { 1275 result.append("M"); 1276 } 1277 } 1278 result.append("\n"); 1279 write(fd, result.string(), result.length()); 1280 result.clear(); 1281 1282 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1283 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1284 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1285 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1286 1287 size_t numtracks = mTracks.size(); 1288 size_t numactive = mActiveTracks.size(); 1289 dprintf(fd, " %d Tracks", numtracks); 1290 size_t numactiveseen = 0; 1291 if (numtracks) { 1292 dprintf(fd, " of which %d are active\n", numactive); 1293 Track::appendDumpHeader(result); 1294 for (size_t i = 0; i < numtracks; ++i) { 1295 sp<Track> track = mTracks[i]; 1296 if (track != 0) { 1297 bool active = mActiveTracks.indexOf(track) >= 0; 1298 if (active) { 1299 numactiveseen++; 1300 } 1301 track->dump(buffer, SIZE, active); 1302 result.append(buffer); 1303 } 1304 } 1305 } else { 1306 result.append("\n"); 1307 } 1308 if (numactiveseen != numactive) { 1309 // some tracks in the active list were not in the tracks list 1310 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1311 " not in the track list\n"); 1312 result.append(buffer); 1313 Track::appendDumpHeader(result); 1314 for (size_t i = 0; i < numactive; ++i) { 1315 sp<Track> track = mActiveTracks[i].promote(); 1316 if (track != 0 && mTracks.indexOf(track) < 0) { 1317 track->dump(buffer, SIZE, true); 1318 result.append(buffer); 1319 } 1320 } 1321 } 1322 1323 write(fd, result.string(), result.size()); 1324} 1325 1326void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1327{ 1328 dprintf(fd, "\nOutput thread %p:\n", this); 1329 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1330 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1331 dprintf(fd, " Total writes: %d\n", mNumWrites); 1332 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1333 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1334 dprintf(fd, " Suspend count: %d\n", mSuspended); 1335 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1336 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1337 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1338 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1339 1340 dumpBase(fd, args); 1341} 1342 1343// Thread virtuals 1344 1345void AudioFlinger::PlaybackThread::onFirstRef() 1346{ 1347 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1348} 1349 1350// ThreadBase virtuals 1351void AudioFlinger::PlaybackThread::preExit() 1352{ 1353 ALOGV(" preExit()"); 1354 // FIXME this is using hard-coded strings but in the future, this functionality will be 1355 // converted to use audio HAL extensions required to support tunneling 1356 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1357} 1358 1359// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1360sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1361 const sp<AudioFlinger::Client>& client, 1362 audio_stream_type_t streamType, 1363 uint32_t sampleRate, 1364 audio_format_t format, 1365 audio_channel_mask_t channelMask, 1366 size_t *pFrameCount, 1367 const sp<IMemory>& sharedBuffer, 1368 int sessionId, 1369 IAudioFlinger::track_flags_t *flags, 1370 pid_t tid, 1371 int uid, 1372 status_t *status) 1373{ 1374 size_t frameCount = *pFrameCount; 1375 sp<Track> track; 1376 status_t lStatus; 1377 1378 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1379 1380 // client expresses a preference for FAST, but we get the final say 1381 if (*flags & IAudioFlinger::TRACK_FAST) { 1382 if ( 1383 // not timed 1384 (!isTimed) && 1385 // either of these use cases: 1386 ( 1387 // use case 1: shared buffer with any frame count 1388 ( 1389 (sharedBuffer != 0) 1390 ) || 1391 // use case 2: callback handler and frame count is default or at least as large as HAL 1392 ( 1393 (tid != -1) && 1394 ((frameCount == 0) || 1395 (frameCount >= mFrameCount)) 1396 ) 1397 ) && 1398 // PCM data 1399 audio_is_linear_pcm(format) && 1400 // identical channel mask to sink, or mono in and stereo sink 1401 (channelMask == mChannelMask || 1402 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1403 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1404 // hardware sample rate 1405 (sampleRate == mSampleRate) && 1406 // normal mixer has an associated fast mixer 1407 hasFastMixer() && 1408 // there are sufficient fast track slots available 1409 (mFastTrackAvailMask != 0) 1410 // FIXME test that MixerThread for this fast track has a capable output HAL 1411 // FIXME add a permission test also? 1412 ) { 1413 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1414 if (frameCount == 0) { 1415 // read the fast track multiplier property the first time it is needed 1416 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1417 if (ok != 0) { 1418 ALOGE("%s pthread_once failed: %d", __func__, ok); 1419 } 1420 frameCount = mFrameCount * sFastTrackMultiplier; 1421 } 1422 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1423 frameCount, mFrameCount); 1424 } else { 1425 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1426 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1427 "sampleRate=%u mSampleRate=%u " 1428 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1429 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1430 audio_is_linear_pcm(format), 1431 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1432 *flags &= ~IAudioFlinger::TRACK_FAST; 1433 // For compatibility with AudioTrack calculation, buffer depth is forced 1434 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1435 // This is probably too conservative, but legacy application code may depend on it. 1436 // If you change this calculation, also review the start threshold which is related. 1437 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1438 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1439 if (minBufCount < 2) { 1440 minBufCount = 2; 1441 } 1442 size_t minFrameCount = mNormalFrameCount * minBufCount; 1443 if (frameCount < minFrameCount) { 1444 frameCount = minFrameCount; 1445 } 1446 } 1447 } 1448 *pFrameCount = frameCount; 1449 1450 switch (mType) { 1451 1452 case DIRECT: 1453 if (audio_is_linear_pcm(format)) { 1454 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1455 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1456 "for output %p with format %#x", 1457 sampleRate, format, channelMask, mOutput, mFormat); 1458 lStatus = BAD_VALUE; 1459 goto Exit; 1460 } 1461 } 1462 break; 1463 1464 case OFFLOAD: 1465 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1466 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1467 "for output %p with format %#x", 1468 sampleRate, format, channelMask, mOutput, mFormat); 1469 lStatus = BAD_VALUE; 1470 goto Exit; 1471 } 1472 break; 1473 1474 default: 1475 if (!audio_is_linear_pcm(format)) { 1476 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1477 "for output %p with format %#x", 1478 format, mOutput, mFormat); 1479 lStatus = BAD_VALUE; 1480 goto Exit; 1481 } 1482 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1483 if (sampleRate > mSampleRate*2) { 1484 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1485 lStatus = BAD_VALUE; 1486 goto Exit; 1487 } 1488 break; 1489 1490 } 1491 1492 lStatus = initCheck(); 1493 if (lStatus != NO_ERROR) { 1494 ALOGE("createTrack_l() audio driver not initialized"); 1495 goto Exit; 1496 } 1497 1498 { // scope for mLock 1499 Mutex::Autolock _l(mLock); 1500 1501 // all tracks in same audio session must share the same routing strategy otherwise 1502 // conflicts will happen when tracks are moved from one output to another by audio policy 1503 // manager 1504 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1505 for (size_t i = 0; i < mTracks.size(); ++i) { 1506 sp<Track> t = mTracks[i]; 1507 if (t != 0 && t->isExternalTrack()) { 1508 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1509 if (sessionId == t->sessionId() && strategy != actual) { 1510 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1511 strategy, actual); 1512 lStatus = BAD_VALUE; 1513 goto Exit; 1514 } 1515 } 1516 } 1517 1518 if (!isTimed) { 1519 track = new Track(this, client, streamType, sampleRate, format, 1520 channelMask, frameCount, NULL, sharedBuffer, 1521 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1522 } else { 1523 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1524 channelMask, frameCount, sharedBuffer, sessionId, uid); 1525 } 1526 1527 // new Track always returns non-NULL, 1528 // but TimedTrack::create() is a factory that could fail by returning NULL 1529 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1530 if (lStatus != NO_ERROR) { 1531 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1532 // track must be cleared from the caller as the caller has the AF lock 1533 goto Exit; 1534 } 1535 mTracks.add(track); 1536 1537 sp<EffectChain> chain = getEffectChain_l(sessionId); 1538 if (chain != 0) { 1539 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1540 track->setMainBuffer(chain->inBuffer()); 1541 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1542 chain->incTrackCnt(); 1543 } 1544 1545 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1546 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1547 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1548 // so ask activity manager to do this on our behalf 1549 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1550 } 1551 } 1552 1553 lStatus = NO_ERROR; 1554 1555Exit: 1556 *status = lStatus; 1557 return track; 1558} 1559 1560uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1561{ 1562 return latency; 1563} 1564 1565uint32_t AudioFlinger::PlaybackThread::latency() const 1566{ 1567 Mutex::Autolock _l(mLock); 1568 return latency_l(); 1569} 1570uint32_t AudioFlinger::PlaybackThread::latency_l() const 1571{ 1572 if (initCheck() == NO_ERROR) { 1573 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1574 } else { 1575 return 0; 1576 } 1577} 1578 1579void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1580{ 1581 Mutex::Autolock _l(mLock); 1582 // Don't apply master volume in SW if our HAL can do it for us. 1583 if (mOutput && mOutput->audioHwDev && 1584 mOutput->audioHwDev->canSetMasterVolume()) { 1585 mMasterVolume = 1.0; 1586 } else { 1587 mMasterVolume = value; 1588 } 1589} 1590 1591void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1592{ 1593 Mutex::Autolock _l(mLock); 1594 // Don't apply master mute in SW if our HAL can do it for us. 1595 if (mOutput && mOutput->audioHwDev && 1596 mOutput->audioHwDev->canSetMasterMute()) { 1597 mMasterMute = false; 1598 } else { 1599 mMasterMute = muted; 1600 } 1601} 1602 1603void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1604{ 1605 Mutex::Autolock _l(mLock); 1606 mStreamTypes[stream].volume = value; 1607 broadcast_l(); 1608} 1609 1610void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1611{ 1612 Mutex::Autolock _l(mLock); 1613 mStreamTypes[stream].mute = muted; 1614 broadcast_l(); 1615} 1616 1617float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1618{ 1619 Mutex::Autolock _l(mLock); 1620 return mStreamTypes[stream].volume; 1621} 1622 1623// addTrack_l() must be called with ThreadBase::mLock held 1624status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1625{ 1626 status_t status = ALREADY_EXISTS; 1627 1628 // set retry count for buffer fill 1629 track->mRetryCount = kMaxTrackStartupRetries; 1630 if (mActiveTracks.indexOf(track) < 0) { 1631 // the track is newly added, make sure it fills up all its 1632 // buffers before playing. This is to ensure the client will 1633 // effectively get the latency it requested. 1634 if (track->isExternalTrack()) { 1635 TrackBase::track_state state = track->mState; 1636 mLock.unlock(); 1637 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1638 mLock.lock(); 1639 // abort track was stopped/paused while we released the lock 1640 if (state != track->mState) { 1641 if (status == NO_ERROR) { 1642 mLock.unlock(); 1643 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1644 mLock.lock(); 1645 } 1646 return INVALID_OPERATION; 1647 } 1648 // abort if start is rejected by audio policy manager 1649 if (status != NO_ERROR) { 1650 return PERMISSION_DENIED; 1651 } 1652#ifdef ADD_BATTERY_DATA 1653 // to track the speaker usage 1654 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1655#endif 1656 } 1657 1658 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1659 track->mResetDone = false; 1660 track->mPresentationCompleteFrames = 0; 1661 mActiveTracks.add(track); 1662 mWakeLockUids.add(track->uid()); 1663 mActiveTracksGeneration++; 1664 mLatestActiveTrack = track; 1665 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1666 if (chain != 0) { 1667 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1668 track->sessionId()); 1669 chain->incActiveTrackCnt(); 1670 } 1671 1672 status = NO_ERROR; 1673 } 1674 1675 onAddNewTrack_l(); 1676 return status; 1677} 1678 1679bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1680{ 1681 track->terminate(); 1682 // active tracks are removed by threadLoop() 1683 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1684 track->mState = TrackBase::STOPPED; 1685 if (!trackActive) { 1686 removeTrack_l(track); 1687 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1688 track->mState = TrackBase::STOPPING_1; 1689 } 1690 1691 return trackActive; 1692} 1693 1694void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1695{ 1696 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1697 mTracks.remove(track); 1698 deleteTrackName_l(track->name()); 1699 // redundant as track is about to be destroyed, for dumpsys only 1700 track->mName = -1; 1701 if (track->isFastTrack()) { 1702 int index = track->mFastIndex; 1703 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1704 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1705 mFastTrackAvailMask |= 1 << index; 1706 // redundant as track is about to be destroyed, for dumpsys only 1707 track->mFastIndex = -1; 1708 } 1709 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1710 if (chain != 0) { 1711 chain->decTrackCnt(); 1712 } 1713} 1714 1715void AudioFlinger::PlaybackThread::broadcast_l() 1716{ 1717 // Thread could be blocked waiting for async 1718 // so signal it to handle state changes immediately 1719 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1720 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1721 mSignalPending = true; 1722 mWaitWorkCV.broadcast(); 1723} 1724 1725String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1726{ 1727 Mutex::Autolock _l(mLock); 1728 if (initCheck() != NO_ERROR) { 1729 return String8(); 1730 } 1731 1732 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1733 const String8 out_s8(s); 1734 free(s); 1735 return out_s8; 1736} 1737 1738void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1739 AudioSystem::OutputDescriptor desc; 1740 void *param2 = NULL; 1741 1742 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1743 param); 1744 1745 switch (event) { 1746 case AudioSystem::OUTPUT_OPENED: 1747 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1748 desc.channelMask = mChannelMask; 1749 desc.samplingRate = mSampleRate; 1750 desc.format = mFormat; 1751 desc.frameCount = mNormalFrameCount; // FIXME see 1752 // AudioFlinger::frameCount(audio_io_handle_t) 1753 desc.latency = latency_l(); 1754 param2 = &desc; 1755 break; 1756 1757 case AudioSystem::STREAM_CONFIG_CHANGED: 1758 param2 = ¶m; 1759 case AudioSystem::OUTPUT_CLOSED: 1760 default: 1761 break; 1762 } 1763 mAudioFlinger->audioConfigChanged(event, mId, param2); 1764} 1765 1766void AudioFlinger::PlaybackThread::writeCallback() 1767{ 1768 ALOG_ASSERT(mCallbackThread != 0); 1769 mCallbackThread->resetWriteBlocked(); 1770} 1771 1772void AudioFlinger::PlaybackThread::drainCallback() 1773{ 1774 ALOG_ASSERT(mCallbackThread != 0); 1775 mCallbackThread->resetDraining(); 1776} 1777 1778void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1779{ 1780 Mutex::Autolock _l(mLock); 1781 // reject out of sequence requests 1782 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1783 mWriteAckSequence &= ~1; 1784 mWaitWorkCV.signal(); 1785 } 1786} 1787 1788void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1789{ 1790 Mutex::Autolock _l(mLock); 1791 // reject out of sequence requests 1792 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1793 mDrainSequence &= ~1; 1794 mWaitWorkCV.signal(); 1795 } 1796} 1797 1798// static 1799int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1800 void *param __unused, 1801 void *cookie) 1802{ 1803 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1804 ALOGV("asyncCallback() event %d", event); 1805 switch (event) { 1806 case STREAM_CBK_EVENT_WRITE_READY: 1807 me->writeCallback(); 1808 break; 1809 case STREAM_CBK_EVENT_DRAIN_READY: 1810 me->drainCallback(); 1811 break; 1812 default: 1813 ALOGW("asyncCallback() unknown event %d", event); 1814 break; 1815 } 1816 return 0; 1817} 1818 1819void AudioFlinger::PlaybackThread::readOutputParameters_l() 1820{ 1821 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1822 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1823 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1824 if (!audio_is_output_channel(mChannelMask)) { 1825 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1826 } 1827 if ((mType == MIXER || mType == DUPLICATING) 1828 && !isValidPcmSinkChannelMask(mChannelMask)) { 1829 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1830 mChannelMask); 1831 } 1832 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1833 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1834 mFormat = mHALFormat; 1835 if (!audio_is_valid_format(mFormat)) { 1836 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1837 } 1838 if ((mType == MIXER || mType == DUPLICATING) 1839 && !isValidPcmSinkFormat(mFormat)) { 1840 LOG_FATAL("HAL format %#x not supported for mixed output", 1841 mFormat); 1842 } 1843 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1844 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1845 mFrameCount = mBufferSize / mFrameSize; 1846 if (mFrameCount & 15) { 1847 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1848 mFrameCount); 1849 } 1850 1851 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1852 (mOutput->stream->set_callback != NULL)) { 1853 if (mOutput->stream->set_callback(mOutput->stream, 1854 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1855 mUseAsyncWrite = true; 1856 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1857 } 1858 } 1859 1860 // Calculate size of normal sink buffer relative to the HAL output buffer size 1861 double multiplier = 1.0; 1862 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1863 kUseFastMixer == FastMixer_Dynamic)) { 1864 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1865 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1866 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1867 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1868 maxNormalFrameCount = maxNormalFrameCount & ~15; 1869 if (maxNormalFrameCount < minNormalFrameCount) { 1870 maxNormalFrameCount = minNormalFrameCount; 1871 } 1872 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1873 if (multiplier <= 1.0) { 1874 multiplier = 1.0; 1875 } else if (multiplier <= 2.0) { 1876 if (2 * mFrameCount <= maxNormalFrameCount) { 1877 multiplier = 2.0; 1878 } else { 1879 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1880 } 1881 } else { 1882 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1883 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1884 // track, but we sometimes have to do this to satisfy the maximum frame count 1885 // constraint) 1886 // FIXME this rounding up should not be done if no HAL SRC 1887 uint32_t truncMult = (uint32_t) multiplier; 1888 if ((truncMult & 1)) { 1889 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1890 ++truncMult; 1891 } 1892 } 1893 multiplier = (double) truncMult; 1894 } 1895 } 1896 mNormalFrameCount = multiplier * mFrameCount; 1897 // round up to nearest 16 frames to satisfy AudioMixer 1898 if (mType == MIXER || mType == DUPLICATING) { 1899 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1900 } 1901 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1902 mNormalFrameCount); 1903 1904 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1905 // Originally this was int16_t[] array, need to remove legacy implications. 1906 free(mSinkBuffer); 1907 mSinkBuffer = NULL; 1908 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1909 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1910 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1911 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1912 1913 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1914 // drives the output. 1915 free(mMixerBuffer); 1916 mMixerBuffer = NULL; 1917 if (mMixerBufferEnabled) { 1918 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1919 mMixerBufferSize = mNormalFrameCount * mChannelCount 1920 * audio_bytes_per_sample(mMixerBufferFormat); 1921 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1922 } 1923 free(mEffectBuffer); 1924 mEffectBuffer = NULL; 1925 if (mEffectBufferEnabled) { 1926 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1927 mEffectBufferSize = mNormalFrameCount * mChannelCount 1928 * audio_bytes_per_sample(mEffectBufferFormat); 1929 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1930 } 1931 1932 // force reconfiguration of effect chains and engines to take new buffer size and audio 1933 // parameters into account 1934 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1935 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1936 // matter. 1937 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1938 Vector< sp<EffectChain> > effectChains = mEffectChains; 1939 for (size_t i = 0; i < effectChains.size(); i ++) { 1940 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1941 } 1942} 1943 1944 1945status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1946{ 1947 if (halFrames == NULL || dspFrames == NULL) { 1948 return BAD_VALUE; 1949 } 1950 Mutex::Autolock _l(mLock); 1951 if (initCheck() != NO_ERROR) { 1952 return INVALID_OPERATION; 1953 } 1954 size_t framesWritten = mBytesWritten / mFrameSize; 1955 *halFrames = framesWritten; 1956 1957 if (isSuspended()) { 1958 // return an estimation of rendered frames when the output is suspended 1959 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1960 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1961 return NO_ERROR; 1962 } else { 1963 status_t status; 1964 uint32_t frames; 1965 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1966 *dspFrames = (size_t)frames; 1967 return status; 1968 } 1969} 1970 1971uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1972{ 1973 Mutex::Autolock _l(mLock); 1974 uint32_t result = 0; 1975 if (getEffectChain_l(sessionId) != 0) { 1976 result = EFFECT_SESSION; 1977 } 1978 1979 for (size_t i = 0; i < mTracks.size(); ++i) { 1980 sp<Track> track = mTracks[i]; 1981 if (sessionId == track->sessionId() && !track->isInvalid()) { 1982 result |= TRACK_SESSION; 1983 break; 1984 } 1985 } 1986 1987 return result; 1988} 1989 1990uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1991{ 1992 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1993 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1994 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1995 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1996 } 1997 for (size_t i = 0; i < mTracks.size(); i++) { 1998 sp<Track> track = mTracks[i]; 1999 if (sessionId == track->sessionId() && !track->isInvalid()) { 2000 return AudioSystem::getStrategyForStream(track->streamType()); 2001 } 2002 } 2003 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2004} 2005 2006 2007AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2008{ 2009 Mutex::Autolock _l(mLock); 2010 return mOutput; 2011} 2012 2013AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2014{ 2015 Mutex::Autolock _l(mLock); 2016 AudioStreamOut *output = mOutput; 2017 mOutput = NULL; 2018 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2019 // must push a NULL and wait for ack 2020 mOutputSink.clear(); 2021 mPipeSink.clear(); 2022 mNormalSink.clear(); 2023 return output; 2024} 2025 2026// this method must always be called either with ThreadBase mLock held or inside the thread loop 2027audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2028{ 2029 if (mOutput == NULL) { 2030 return NULL; 2031 } 2032 return &mOutput->stream->common; 2033} 2034 2035uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2036{ 2037 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2038} 2039 2040status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2041{ 2042 if (!isValidSyncEvent(event)) { 2043 return BAD_VALUE; 2044 } 2045 2046 Mutex::Autolock _l(mLock); 2047 2048 for (size_t i = 0; i < mTracks.size(); ++i) { 2049 sp<Track> track = mTracks[i]; 2050 if (event->triggerSession() == track->sessionId()) { 2051 (void) track->setSyncEvent(event); 2052 return NO_ERROR; 2053 } 2054 } 2055 2056 return NAME_NOT_FOUND; 2057} 2058 2059bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2060{ 2061 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2062} 2063 2064void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2065 const Vector< sp<Track> >& tracksToRemove) 2066{ 2067 size_t count = tracksToRemove.size(); 2068 if (count > 0) { 2069 for (size_t i = 0 ; i < count ; i++) { 2070 const sp<Track>& track = tracksToRemove.itemAt(i); 2071 if (track->isExternalTrack()) { 2072 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2073#ifdef ADD_BATTERY_DATA 2074 // to track the speaker usage 2075 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2076#endif 2077 if (track->isTerminated()) { 2078 AudioSystem::releaseOutput(mId); 2079 } 2080 } 2081 } 2082 } 2083} 2084 2085void AudioFlinger::PlaybackThread::checkSilentMode_l() 2086{ 2087 if (!mMasterMute) { 2088 char value[PROPERTY_VALUE_MAX]; 2089 if (property_get("ro.audio.silent", value, "0") > 0) { 2090 char *endptr; 2091 unsigned long ul = strtoul(value, &endptr, 0); 2092 if (*endptr == '\0' && ul != 0) { 2093 ALOGD("Silence is golden"); 2094 // The setprop command will not allow a property to be changed after 2095 // the first time it is set, so we don't have to worry about un-muting. 2096 setMasterMute_l(true); 2097 } 2098 } 2099 } 2100} 2101 2102// shared by MIXER and DIRECT, overridden by DUPLICATING 2103ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2104{ 2105 // FIXME rewrite to reduce number of system calls 2106 mLastWriteTime = systemTime(); 2107 mInWrite = true; 2108 ssize_t bytesWritten; 2109 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2110 2111 // If an NBAIO sink is present, use it to write the normal mixer's submix 2112 if (mNormalSink != 0) { 2113 const size_t count = mBytesRemaining / mFrameSize; 2114 2115 ATRACE_BEGIN("write"); 2116 // update the setpoint when AudioFlinger::mScreenState changes 2117 uint32_t screenState = AudioFlinger::mScreenState; 2118 if (screenState != mScreenState) { 2119 mScreenState = screenState; 2120 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2121 if (pipe != NULL) { 2122 pipe->setAvgFrames((mScreenState & 1) ? 2123 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2124 } 2125 } 2126 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2127 ATRACE_END(); 2128 if (framesWritten > 0) { 2129 bytesWritten = framesWritten * mFrameSize; 2130 } else { 2131 bytesWritten = framesWritten; 2132 } 2133 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2134 if (status == NO_ERROR) { 2135 size_t totalFramesWritten = mNormalSink->framesWritten(); 2136 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2137 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2138 mLatchDValid = true; 2139 } 2140 } 2141 // otherwise use the HAL / AudioStreamOut directly 2142 } else { 2143 // Direct output and offload threads 2144 2145 if (mUseAsyncWrite) { 2146 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2147 mWriteAckSequence += 2; 2148 mWriteAckSequence |= 1; 2149 ALOG_ASSERT(mCallbackThread != 0); 2150 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2151 } 2152 // FIXME We should have an implementation of timestamps for direct output threads. 2153 // They are used e.g for multichannel PCM playback over HDMI. 2154 bytesWritten = mOutput->stream->write(mOutput->stream, 2155 (char *)mSinkBuffer + offset, mBytesRemaining); 2156 if (mUseAsyncWrite && 2157 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2158 // do not wait for async callback in case of error of full write 2159 mWriteAckSequence &= ~1; 2160 ALOG_ASSERT(mCallbackThread != 0); 2161 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2162 } 2163 } 2164 2165 mNumWrites++; 2166 mInWrite = false; 2167 mStandby = false; 2168 return bytesWritten; 2169} 2170 2171void AudioFlinger::PlaybackThread::threadLoop_drain() 2172{ 2173 if (mOutput->stream->drain) { 2174 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2175 if (mUseAsyncWrite) { 2176 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2177 mDrainSequence |= 1; 2178 ALOG_ASSERT(mCallbackThread != 0); 2179 mCallbackThread->setDraining(mDrainSequence); 2180 } 2181 mOutput->stream->drain(mOutput->stream, 2182 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2183 : AUDIO_DRAIN_ALL); 2184 } 2185} 2186 2187void AudioFlinger::PlaybackThread::threadLoop_exit() 2188{ 2189 // Default implementation has nothing to do 2190} 2191 2192/* 2193The derived values that are cached: 2194 - mSinkBufferSize from frame count * frame size 2195 - activeSleepTime from activeSleepTimeUs() 2196 - idleSleepTime from idleSleepTimeUs() 2197 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2198 - maxPeriod from frame count and sample rate (MIXER only) 2199 2200The parameters that affect these derived values are: 2201 - frame count 2202 - frame size 2203 - sample rate 2204 - device type: A2DP or not 2205 - device latency 2206 - format: PCM or not 2207 - active sleep time 2208 - idle sleep time 2209*/ 2210 2211void AudioFlinger::PlaybackThread::cacheParameters_l() 2212{ 2213 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2214 activeSleepTime = activeSleepTimeUs(); 2215 idleSleepTime = idleSleepTimeUs(); 2216} 2217 2218void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2219{ 2220 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2221 this, streamType, mTracks.size()); 2222 Mutex::Autolock _l(mLock); 2223 2224 size_t size = mTracks.size(); 2225 for (size_t i = 0; i < size; i++) { 2226 sp<Track> t = mTracks[i]; 2227 if (t->streamType() == streamType) { 2228 t->invalidate(); 2229 } 2230 } 2231} 2232 2233status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2234{ 2235 int session = chain->sessionId(); 2236 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2237 ? mEffectBuffer : mSinkBuffer); 2238 bool ownsBuffer = false; 2239 2240 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2241 if (session > 0) { 2242 // Only one effect chain can be present in direct output thread and it uses 2243 // the sink buffer as input 2244 if (mType != DIRECT) { 2245 size_t numSamples = mNormalFrameCount * mChannelCount; 2246 buffer = new int16_t[numSamples]; 2247 memset(buffer, 0, numSamples * sizeof(int16_t)); 2248 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2249 ownsBuffer = true; 2250 } 2251 2252 // Attach all tracks with same session ID to this chain. 2253 for (size_t i = 0; i < mTracks.size(); ++i) { 2254 sp<Track> track = mTracks[i]; 2255 if (session == track->sessionId()) { 2256 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2257 buffer); 2258 track->setMainBuffer(buffer); 2259 chain->incTrackCnt(); 2260 } 2261 } 2262 2263 // indicate all active tracks in the chain 2264 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2265 sp<Track> track = mActiveTracks[i].promote(); 2266 if (track == 0) { 2267 continue; 2268 } 2269 if (session == track->sessionId()) { 2270 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2271 chain->incActiveTrackCnt(); 2272 } 2273 } 2274 } 2275 2276 chain->setInBuffer(buffer, ownsBuffer); 2277 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2278 ? mEffectBuffer : mSinkBuffer)); 2279 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2280 // chains list in order to be processed last as it contains output stage effects 2281 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2282 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2283 // after track specific effects and before output stage 2284 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2285 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2286 // Effect chain for other sessions are inserted at beginning of effect 2287 // chains list to be processed before output mix effects. Relative order between other 2288 // sessions is not important 2289 size_t size = mEffectChains.size(); 2290 size_t i = 0; 2291 for (i = 0; i < size; i++) { 2292 if (mEffectChains[i]->sessionId() < session) { 2293 break; 2294 } 2295 } 2296 mEffectChains.insertAt(chain, i); 2297 checkSuspendOnAddEffectChain_l(chain); 2298 2299 return NO_ERROR; 2300} 2301 2302size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2303{ 2304 int session = chain->sessionId(); 2305 2306 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2307 2308 for (size_t i = 0; i < mEffectChains.size(); i++) { 2309 if (chain == mEffectChains[i]) { 2310 mEffectChains.removeAt(i); 2311 // detach all active tracks from the chain 2312 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2313 sp<Track> track = mActiveTracks[i].promote(); 2314 if (track == 0) { 2315 continue; 2316 } 2317 if (session == track->sessionId()) { 2318 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2319 chain.get(), session); 2320 chain->decActiveTrackCnt(); 2321 } 2322 } 2323 2324 // detach all tracks with same session ID from this chain 2325 for (size_t i = 0; i < mTracks.size(); ++i) { 2326 sp<Track> track = mTracks[i]; 2327 if (session == track->sessionId()) { 2328 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2329 chain->decTrackCnt(); 2330 } 2331 } 2332 break; 2333 } 2334 } 2335 return mEffectChains.size(); 2336} 2337 2338status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2339 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2340{ 2341 Mutex::Autolock _l(mLock); 2342 return attachAuxEffect_l(track, EffectId); 2343} 2344 2345status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2346 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2347{ 2348 status_t status = NO_ERROR; 2349 2350 if (EffectId == 0) { 2351 track->setAuxBuffer(0, NULL); 2352 } else { 2353 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2354 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2355 if (effect != 0) { 2356 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2357 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2358 } else { 2359 status = INVALID_OPERATION; 2360 } 2361 } else { 2362 status = BAD_VALUE; 2363 } 2364 } 2365 return status; 2366} 2367 2368void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2369{ 2370 for (size_t i = 0; i < mTracks.size(); ++i) { 2371 sp<Track> track = mTracks[i]; 2372 if (track->auxEffectId() == effectId) { 2373 attachAuxEffect_l(track, 0); 2374 } 2375 } 2376} 2377 2378bool AudioFlinger::PlaybackThread::threadLoop() 2379{ 2380 Vector< sp<Track> > tracksToRemove; 2381 2382 standbyTime = systemTime(); 2383 2384 // MIXER 2385 nsecs_t lastWarning = 0; 2386 2387 // DUPLICATING 2388 // FIXME could this be made local to while loop? 2389 writeFrames = 0; 2390 2391 int lastGeneration = 0; 2392 2393 cacheParameters_l(); 2394 sleepTime = idleSleepTime; 2395 2396 if (mType == MIXER) { 2397 sleepTimeShift = 0; 2398 } 2399 2400 CpuStats cpuStats; 2401 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2402 2403 acquireWakeLock(); 2404 2405 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2406 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2407 // and then that string will be logged at the next convenient opportunity. 2408 const char *logString = NULL; 2409 2410 checkSilentMode_l(); 2411 2412 while (!exitPending()) 2413 { 2414 cpuStats.sample(myName); 2415 2416 Vector< sp<EffectChain> > effectChains; 2417 2418 { // scope for mLock 2419 2420 Mutex::Autolock _l(mLock); 2421 2422 processConfigEvents_l(); 2423 2424 if (logString != NULL) { 2425 mNBLogWriter->logTimestamp(); 2426 mNBLogWriter->log(logString); 2427 logString = NULL; 2428 } 2429 2430 if (mLatchDValid) { 2431 mLatchQ = mLatchD; 2432 mLatchDValid = false; 2433 mLatchQValid = true; 2434 } 2435 2436 saveOutputTracks(); 2437 if (mSignalPending) { 2438 // A signal was raised while we were unlocked 2439 mSignalPending = false; 2440 } else if (waitingAsyncCallback_l()) { 2441 if (exitPending()) { 2442 break; 2443 } 2444 releaseWakeLock_l(); 2445 mWakeLockUids.clear(); 2446 mActiveTracksGeneration++; 2447 ALOGV("wait async completion"); 2448 mWaitWorkCV.wait(mLock); 2449 ALOGV("async completion/wake"); 2450 acquireWakeLock_l(); 2451 standbyTime = systemTime() + standbyDelay; 2452 sleepTime = 0; 2453 2454 continue; 2455 } 2456 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2457 isSuspended()) { 2458 // put audio hardware into standby after short delay 2459 if (shouldStandby_l()) { 2460 2461 threadLoop_standby(); 2462 2463 mStandby = true; 2464 } 2465 2466 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2467 // we're about to wait, flush the binder command buffer 2468 IPCThreadState::self()->flushCommands(); 2469 2470 clearOutputTracks(); 2471 2472 if (exitPending()) { 2473 break; 2474 } 2475 2476 releaseWakeLock_l(); 2477 mWakeLockUids.clear(); 2478 mActiveTracksGeneration++; 2479 // wait until we have something to do... 2480 ALOGV("%s going to sleep", myName.string()); 2481 mWaitWorkCV.wait(mLock); 2482 ALOGV("%s waking up", myName.string()); 2483 acquireWakeLock_l(); 2484 2485 mMixerStatus = MIXER_IDLE; 2486 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2487 mBytesWritten = 0; 2488 mBytesRemaining = 0; 2489 checkSilentMode_l(); 2490 2491 standbyTime = systemTime() + standbyDelay; 2492 sleepTime = idleSleepTime; 2493 if (mType == MIXER) { 2494 sleepTimeShift = 0; 2495 } 2496 2497 continue; 2498 } 2499 } 2500 // mMixerStatusIgnoringFastTracks is also updated internally 2501 mMixerStatus = prepareTracks_l(&tracksToRemove); 2502 2503 // compare with previously applied list 2504 if (lastGeneration != mActiveTracksGeneration) { 2505 // update wakelock 2506 updateWakeLockUids_l(mWakeLockUids); 2507 lastGeneration = mActiveTracksGeneration; 2508 } 2509 2510 // prevent any changes in effect chain list and in each effect chain 2511 // during mixing and effect process as the audio buffers could be deleted 2512 // or modified if an effect is created or deleted 2513 lockEffectChains_l(effectChains); 2514 } // mLock scope ends 2515 2516 if (mBytesRemaining == 0) { 2517 mCurrentWriteLength = 0; 2518 if (mMixerStatus == MIXER_TRACKS_READY) { 2519 // threadLoop_mix() sets mCurrentWriteLength 2520 threadLoop_mix(); 2521 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2522 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2523 // threadLoop_sleepTime sets sleepTime to 0 if data 2524 // must be written to HAL 2525 threadLoop_sleepTime(); 2526 if (sleepTime == 0) { 2527 mCurrentWriteLength = mSinkBufferSize; 2528 } 2529 } 2530 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2531 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2532 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2533 // or mSinkBuffer (if there are no effects). 2534 // 2535 // This is done pre-effects computation; if effects change to 2536 // support higher precision, this needs to move. 2537 // 2538 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2539 // TODO use sleepTime == 0 as an additional condition. 2540 if (mMixerBufferValid) { 2541 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2542 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2543 2544 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2545 mNormalFrameCount * mChannelCount); 2546 } 2547 2548 mBytesRemaining = mCurrentWriteLength; 2549 if (isSuspended()) { 2550 sleepTime = suspendSleepTimeUs(); 2551 // simulate write to HAL when suspended 2552 mBytesWritten += mSinkBufferSize; 2553 mBytesRemaining = 0; 2554 } 2555 2556 // only process effects if we're going to write 2557 if (sleepTime == 0 && mType != OFFLOAD) { 2558 for (size_t i = 0; i < effectChains.size(); i ++) { 2559 effectChains[i]->process_l(); 2560 } 2561 } 2562 } 2563 // Process effect chains for offloaded thread even if no audio 2564 // was read from audio track: process only updates effect state 2565 // and thus does have to be synchronized with audio writes but may have 2566 // to be called while waiting for async write callback 2567 if (mType == OFFLOAD) { 2568 for (size_t i = 0; i < effectChains.size(); i ++) { 2569 effectChains[i]->process_l(); 2570 } 2571 } 2572 2573 // Only if the Effects buffer is enabled and there is data in the 2574 // Effects buffer (buffer valid), we need to 2575 // copy into the sink buffer. 2576 // TODO use sleepTime == 0 as an additional condition. 2577 if (mEffectBufferValid) { 2578 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2579 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2580 mNormalFrameCount * mChannelCount); 2581 } 2582 2583 // enable changes in effect chain 2584 unlockEffectChains(effectChains); 2585 2586 if (!waitingAsyncCallback()) { 2587 // sleepTime == 0 means we must write to audio hardware 2588 if (sleepTime == 0) { 2589 if (mBytesRemaining) { 2590 ssize_t ret = threadLoop_write(); 2591 if (ret < 0) { 2592 mBytesRemaining = 0; 2593 } else { 2594 mBytesWritten += ret; 2595 mBytesRemaining -= ret; 2596 } 2597 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2598 (mMixerStatus == MIXER_DRAIN_ALL)) { 2599 threadLoop_drain(); 2600 } 2601 if (mType == MIXER) { 2602 // write blocked detection 2603 nsecs_t now = systemTime(); 2604 nsecs_t delta = now - mLastWriteTime; 2605 if (!mStandby && delta > maxPeriod) { 2606 mNumDelayedWrites++; 2607 if ((now - lastWarning) > kWarningThrottleNs) { 2608 ATRACE_NAME("underrun"); 2609 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2610 ns2ms(delta), mNumDelayedWrites, this); 2611 lastWarning = now; 2612 } 2613 } 2614 } 2615 2616 } else { 2617 usleep(sleepTime); 2618 } 2619 } 2620 2621 // Finally let go of removed track(s), without the lock held 2622 // since we can't guarantee the destructors won't acquire that 2623 // same lock. This will also mutate and push a new fast mixer state. 2624 threadLoop_removeTracks(tracksToRemove); 2625 tracksToRemove.clear(); 2626 2627 // FIXME I don't understand the need for this here; 2628 // it was in the original code but maybe the 2629 // assignment in saveOutputTracks() makes this unnecessary? 2630 clearOutputTracks(); 2631 2632 // Effect chains will be actually deleted here if they were removed from 2633 // mEffectChains list during mixing or effects processing 2634 effectChains.clear(); 2635 2636 // FIXME Note that the above .clear() is no longer necessary since effectChains 2637 // is now local to this block, but will keep it for now (at least until merge done). 2638 } 2639 2640 threadLoop_exit(); 2641 2642 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2643 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2644 // put output stream into standby mode 2645 if (!mStandby) { 2646 mOutput->stream->common.standby(&mOutput->stream->common); 2647 } 2648 } 2649 2650 releaseWakeLock(); 2651 mWakeLockUids.clear(); 2652 mActiveTracksGeneration++; 2653 2654 ALOGV("Thread %p type %d exiting", this, mType); 2655 return false; 2656} 2657 2658// removeTracks_l() must be called with ThreadBase::mLock held 2659void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2660{ 2661 size_t count = tracksToRemove.size(); 2662 if (count > 0) { 2663 for (size_t i=0 ; i<count ; i++) { 2664 const sp<Track>& track = tracksToRemove.itemAt(i); 2665 mActiveTracks.remove(track); 2666 mWakeLockUids.remove(track->uid()); 2667 mActiveTracksGeneration++; 2668 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2669 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2670 if (chain != 0) { 2671 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2672 track->sessionId()); 2673 chain->decActiveTrackCnt(); 2674 } 2675 if (track->isTerminated()) { 2676 removeTrack_l(track); 2677 } 2678 } 2679 } 2680 2681} 2682 2683status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2684{ 2685 if (mNormalSink != 0) { 2686 return mNormalSink->getTimestamp(timestamp); 2687 } 2688 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2689 uint64_t position64; 2690 int ret = mOutput->stream->get_presentation_position( 2691 mOutput->stream, &position64, ×tamp.mTime); 2692 if (ret == 0) { 2693 timestamp.mPosition = (uint32_t)position64; 2694 return NO_ERROR; 2695 } 2696 } 2697 return INVALID_OPERATION; 2698} 2699 2700status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2701 audio_patch_handle_t *handle) 2702{ 2703 status_t status = NO_ERROR; 2704 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2705 // store new device and send to effects 2706 audio_devices_t type = AUDIO_DEVICE_NONE; 2707 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2708 type |= patch->sinks[i].ext.device.type; 2709 } 2710 mOutDevice = type; 2711 for (size_t i = 0; i < mEffectChains.size(); i++) { 2712 mEffectChains[i]->setDevice_l(mOutDevice); 2713 } 2714 2715 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2716 status = hwDevice->create_audio_patch(hwDevice, 2717 patch->num_sources, 2718 patch->sources, 2719 patch->num_sinks, 2720 patch->sinks, 2721 handle); 2722 } else { 2723 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2724 } 2725 return status; 2726} 2727 2728status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2729{ 2730 status_t status = NO_ERROR; 2731 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2732 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2733 status = hwDevice->release_audio_patch(hwDevice, handle); 2734 } else { 2735 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2736 } 2737 return status; 2738} 2739 2740void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2741{ 2742 Mutex::Autolock _l(mLock); 2743 mTracks.add(track); 2744} 2745 2746void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2747{ 2748 Mutex::Autolock _l(mLock); 2749 destroyTrack_l(track); 2750} 2751 2752void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2753{ 2754 ThreadBase::getAudioPortConfig(config); 2755 config->role = AUDIO_PORT_ROLE_SOURCE; 2756 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2757 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2758} 2759 2760// ---------------------------------------------------------------------------- 2761 2762AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2763 audio_io_handle_t id, audio_devices_t device, type_t type) 2764 : PlaybackThread(audioFlinger, output, id, device, type), 2765 // mAudioMixer below 2766 // mFastMixer below 2767 mFastMixerFutex(0) 2768 // mOutputSink below 2769 // mPipeSink below 2770 // mNormalSink below 2771{ 2772 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2773 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2774 "mFrameCount=%d, mNormalFrameCount=%d", 2775 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2776 mNormalFrameCount); 2777 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2778 2779 // create an NBAIO sink for the HAL output stream, and negotiate 2780 mOutputSink = new AudioStreamOutSink(output->stream); 2781 size_t numCounterOffers = 0; 2782 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2783 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2784 ALOG_ASSERT(index == 0); 2785 2786 // initialize fast mixer depending on configuration 2787 bool initFastMixer; 2788 switch (kUseFastMixer) { 2789 case FastMixer_Never: 2790 initFastMixer = false; 2791 break; 2792 case FastMixer_Always: 2793 initFastMixer = true; 2794 break; 2795 case FastMixer_Static: 2796 case FastMixer_Dynamic: 2797 initFastMixer = mFrameCount < mNormalFrameCount; 2798 break; 2799 } 2800 if (initFastMixer) { 2801 audio_format_t fastMixerFormat; 2802 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2803 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2804 } else { 2805 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2806 } 2807 if (mFormat != fastMixerFormat) { 2808 // change our Sink format to accept our intermediate precision 2809 mFormat = fastMixerFormat; 2810 free(mSinkBuffer); 2811 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2812 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2813 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2814 } 2815 2816 // create a MonoPipe to connect our submix to FastMixer 2817 NBAIO_Format format = mOutputSink->format(); 2818 // adjust format to match that of the Fast Mixer 2819 format.mFormat = fastMixerFormat; 2820 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2821 2822 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2823 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2824 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2825 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2826 const NBAIO_Format offers[1] = {format}; 2827 size_t numCounterOffers = 0; 2828 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2829 ALOG_ASSERT(index == 0); 2830 monoPipe->setAvgFrames((mScreenState & 1) ? 2831 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2832 mPipeSink = monoPipe; 2833 2834#ifdef TEE_SINK 2835 if (mTeeSinkOutputEnabled) { 2836 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2837 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2838 numCounterOffers = 0; 2839 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2840 ALOG_ASSERT(index == 0); 2841 mTeeSink = teeSink; 2842 PipeReader *teeSource = new PipeReader(*teeSink); 2843 numCounterOffers = 0; 2844 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2845 ALOG_ASSERT(index == 0); 2846 mTeeSource = teeSource; 2847 } 2848#endif 2849 2850 // create fast mixer and configure it initially with just one fast track for our submix 2851 mFastMixer = new FastMixer(); 2852 FastMixerStateQueue *sq = mFastMixer->sq(); 2853#ifdef STATE_QUEUE_DUMP 2854 sq->setObserverDump(&mStateQueueObserverDump); 2855 sq->setMutatorDump(&mStateQueueMutatorDump); 2856#endif 2857 FastMixerState *state = sq->begin(); 2858 FastTrack *fastTrack = &state->mFastTracks[0]; 2859 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2860 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2861 fastTrack->mVolumeProvider = NULL; 2862 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2863 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2864 fastTrack->mGeneration++; 2865 state->mFastTracksGen++; 2866 state->mTrackMask = 1; 2867 // fast mixer will use the HAL output sink 2868 state->mOutputSink = mOutputSink.get(); 2869 state->mOutputSinkGen++; 2870 state->mFrameCount = mFrameCount; 2871 state->mCommand = FastMixerState::COLD_IDLE; 2872 // already done in constructor initialization list 2873 //mFastMixerFutex = 0; 2874 state->mColdFutexAddr = &mFastMixerFutex; 2875 state->mColdGen++; 2876 state->mDumpState = &mFastMixerDumpState; 2877#ifdef TEE_SINK 2878 state->mTeeSink = mTeeSink.get(); 2879#endif 2880 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2881 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2882 sq->end(); 2883 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2884 2885 // start the fast mixer 2886 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2887 pid_t tid = mFastMixer->getTid(); 2888 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2889 if (err != 0) { 2890 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2891 kPriorityFastMixer, getpid_cached, tid, err); 2892 } 2893 2894#ifdef AUDIO_WATCHDOG 2895 // create and start the watchdog 2896 mAudioWatchdog = new AudioWatchdog(); 2897 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2898 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2899 tid = mAudioWatchdog->getTid(); 2900 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2901 if (err != 0) { 2902 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2903 kPriorityFastMixer, getpid_cached, tid, err); 2904 } 2905#endif 2906 2907 } 2908 2909 switch (kUseFastMixer) { 2910 case FastMixer_Never: 2911 case FastMixer_Dynamic: 2912 mNormalSink = mOutputSink; 2913 break; 2914 case FastMixer_Always: 2915 mNormalSink = mPipeSink; 2916 break; 2917 case FastMixer_Static: 2918 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2919 break; 2920 } 2921} 2922 2923AudioFlinger::MixerThread::~MixerThread() 2924{ 2925 if (mFastMixer != 0) { 2926 FastMixerStateQueue *sq = mFastMixer->sq(); 2927 FastMixerState *state = sq->begin(); 2928 if (state->mCommand == FastMixerState::COLD_IDLE) { 2929 int32_t old = android_atomic_inc(&mFastMixerFutex); 2930 if (old == -1) { 2931 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2932 } 2933 } 2934 state->mCommand = FastMixerState::EXIT; 2935 sq->end(); 2936 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2937 mFastMixer->join(); 2938 // Though the fast mixer thread has exited, it's state queue is still valid. 2939 // We'll use that extract the final state which contains one remaining fast track 2940 // corresponding to our sub-mix. 2941 state = sq->begin(); 2942 ALOG_ASSERT(state->mTrackMask == 1); 2943 FastTrack *fastTrack = &state->mFastTracks[0]; 2944 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2945 delete fastTrack->mBufferProvider; 2946 sq->end(false /*didModify*/); 2947 mFastMixer.clear(); 2948#ifdef AUDIO_WATCHDOG 2949 if (mAudioWatchdog != 0) { 2950 mAudioWatchdog->requestExit(); 2951 mAudioWatchdog->requestExitAndWait(); 2952 mAudioWatchdog.clear(); 2953 } 2954#endif 2955 } 2956 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2957 delete mAudioMixer; 2958} 2959 2960 2961uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2962{ 2963 if (mFastMixer != 0) { 2964 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2965 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2966 } 2967 return latency; 2968} 2969 2970 2971void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2972{ 2973 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2974} 2975 2976ssize_t AudioFlinger::MixerThread::threadLoop_write() 2977{ 2978 // FIXME we should only do one push per cycle; confirm this is true 2979 // Start the fast mixer if it's not already running 2980 if (mFastMixer != 0) { 2981 FastMixerStateQueue *sq = mFastMixer->sq(); 2982 FastMixerState *state = sq->begin(); 2983 if (state->mCommand != FastMixerState::MIX_WRITE && 2984 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2985 if (state->mCommand == FastMixerState::COLD_IDLE) { 2986 int32_t old = android_atomic_inc(&mFastMixerFutex); 2987 if (old == -1) { 2988 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2989 } 2990#ifdef AUDIO_WATCHDOG 2991 if (mAudioWatchdog != 0) { 2992 mAudioWatchdog->resume(); 2993 } 2994#endif 2995 } 2996 state->mCommand = FastMixerState::MIX_WRITE; 2997 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2998 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2999 sq->end(); 3000 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3001 if (kUseFastMixer == FastMixer_Dynamic) { 3002 mNormalSink = mPipeSink; 3003 } 3004 } else { 3005 sq->end(false /*didModify*/); 3006 } 3007 } 3008 return PlaybackThread::threadLoop_write(); 3009} 3010 3011void AudioFlinger::MixerThread::threadLoop_standby() 3012{ 3013 // Idle the fast mixer if it's currently running 3014 if (mFastMixer != 0) { 3015 FastMixerStateQueue *sq = mFastMixer->sq(); 3016 FastMixerState *state = sq->begin(); 3017 if (!(state->mCommand & FastMixerState::IDLE)) { 3018 state->mCommand = FastMixerState::COLD_IDLE; 3019 state->mColdFutexAddr = &mFastMixerFutex; 3020 state->mColdGen++; 3021 mFastMixerFutex = 0; 3022 sq->end(); 3023 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3024 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3025 if (kUseFastMixer == FastMixer_Dynamic) { 3026 mNormalSink = mOutputSink; 3027 } 3028#ifdef AUDIO_WATCHDOG 3029 if (mAudioWatchdog != 0) { 3030 mAudioWatchdog->pause(); 3031 } 3032#endif 3033 } else { 3034 sq->end(false /*didModify*/); 3035 } 3036 } 3037 PlaybackThread::threadLoop_standby(); 3038} 3039 3040bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3041{ 3042 return false; 3043} 3044 3045bool AudioFlinger::PlaybackThread::shouldStandby_l() 3046{ 3047 return !mStandby; 3048} 3049 3050bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3051{ 3052 Mutex::Autolock _l(mLock); 3053 return waitingAsyncCallback_l(); 3054} 3055 3056// shared by MIXER and DIRECT, overridden by DUPLICATING 3057void AudioFlinger::PlaybackThread::threadLoop_standby() 3058{ 3059 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3060 mOutput->stream->common.standby(&mOutput->stream->common); 3061 if (mUseAsyncWrite != 0) { 3062 // discard any pending drain or write ack by incrementing sequence 3063 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3064 mDrainSequence = (mDrainSequence + 2) & ~1; 3065 ALOG_ASSERT(mCallbackThread != 0); 3066 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3067 mCallbackThread->setDraining(mDrainSequence); 3068 } 3069} 3070 3071void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3072{ 3073 ALOGV("signal playback thread"); 3074 broadcast_l(); 3075} 3076 3077void AudioFlinger::MixerThread::threadLoop_mix() 3078{ 3079 // obtain the presentation timestamp of the next output buffer 3080 int64_t pts; 3081 status_t status = INVALID_OPERATION; 3082 3083 if (mNormalSink != 0) { 3084 status = mNormalSink->getNextWriteTimestamp(&pts); 3085 } else { 3086 status = mOutputSink->getNextWriteTimestamp(&pts); 3087 } 3088 3089 if (status != NO_ERROR) { 3090 pts = AudioBufferProvider::kInvalidPTS; 3091 } 3092 3093 // mix buffers... 3094 mAudioMixer->process(pts); 3095 mCurrentWriteLength = mSinkBufferSize; 3096 // increase sleep time progressively when application underrun condition clears. 3097 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3098 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3099 // such that we would underrun the audio HAL. 3100 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3101 sleepTimeShift--; 3102 } 3103 sleepTime = 0; 3104 standbyTime = systemTime() + standbyDelay; 3105 //TODO: delay standby when effects have a tail 3106} 3107 3108void AudioFlinger::MixerThread::threadLoop_sleepTime() 3109{ 3110 // If no tracks are ready, sleep once for the duration of an output 3111 // buffer size, then write 0s to the output 3112 if (sleepTime == 0) { 3113 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3114 sleepTime = activeSleepTime >> sleepTimeShift; 3115 if (sleepTime < kMinThreadSleepTimeUs) { 3116 sleepTime = kMinThreadSleepTimeUs; 3117 } 3118 // reduce sleep time in case of consecutive application underruns to avoid 3119 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3120 // duration we would end up writing less data than needed by the audio HAL if 3121 // the condition persists. 3122 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3123 sleepTimeShift++; 3124 } 3125 } else { 3126 sleepTime = idleSleepTime; 3127 } 3128 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3129 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3130 // before effects processing or output. 3131 if (mMixerBufferValid) { 3132 memset(mMixerBuffer, 0, mMixerBufferSize); 3133 } else { 3134 memset(mSinkBuffer, 0, mSinkBufferSize); 3135 } 3136 sleepTime = 0; 3137 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3138 "anticipated start"); 3139 } 3140 // TODO add standby time extension fct of effect tail 3141} 3142 3143// prepareTracks_l() must be called with ThreadBase::mLock held 3144AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3145 Vector< sp<Track> > *tracksToRemove) 3146{ 3147 3148 mixer_state mixerStatus = MIXER_IDLE; 3149 // find out which tracks need to be processed 3150 size_t count = mActiveTracks.size(); 3151 size_t mixedTracks = 0; 3152 size_t tracksWithEffect = 0; 3153 // counts only _active_ fast tracks 3154 size_t fastTracks = 0; 3155 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3156 3157 float masterVolume = mMasterVolume; 3158 bool masterMute = mMasterMute; 3159 3160 if (masterMute) { 3161 masterVolume = 0; 3162 } 3163 // Delegate master volume control to effect in output mix effect chain if needed 3164 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3165 if (chain != 0) { 3166 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3167 chain->setVolume_l(&v, &v); 3168 masterVolume = (float)((v + (1 << 23)) >> 24); 3169 chain.clear(); 3170 } 3171 3172 // prepare a new state to push 3173 FastMixerStateQueue *sq = NULL; 3174 FastMixerState *state = NULL; 3175 bool didModify = false; 3176 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3177 if (mFastMixer != 0) { 3178 sq = mFastMixer->sq(); 3179 state = sq->begin(); 3180 } 3181 3182 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3183 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3184 3185 for (size_t i=0 ; i<count ; i++) { 3186 const sp<Track> t = mActiveTracks[i].promote(); 3187 if (t == 0) { 3188 continue; 3189 } 3190 3191 // this const just means the local variable doesn't change 3192 Track* const track = t.get(); 3193 3194 // process fast tracks 3195 if (track->isFastTrack()) { 3196 3197 // It's theoretically possible (though unlikely) for a fast track to be created 3198 // and then removed within the same normal mix cycle. This is not a problem, as 3199 // the track never becomes active so it's fast mixer slot is never touched. 3200 // The converse, of removing an (active) track and then creating a new track 3201 // at the identical fast mixer slot within the same normal mix cycle, 3202 // is impossible because the slot isn't marked available until the end of each cycle. 3203 int j = track->mFastIndex; 3204 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3205 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3206 FastTrack *fastTrack = &state->mFastTracks[j]; 3207 3208 // Determine whether the track is currently in underrun condition, 3209 // and whether it had a recent underrun. 3210 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3211 FastTrackUnderruns underruns = ftDump->mUnderruns; 3212 uint32_t recentFull = (underruns.mBitFields.mFull - 3213 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3214 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3215 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3216 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3217 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3218 uint32_t recentUnderruns = recentPartial + recentEmpty; 3219 track->mObservedUnderruns = underruns; 3220 // don't count underruns that occur while stopping or pausing 3221 // or stopped which can occur when flush() is called while active 3222 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3223 recentUnderruns > 0) { 3224 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3225 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3226 } 3227 3228 // This is similar to the state machine for normal tracks, 3229 // with a few modifications for fast tracks. 3230 bool isActive = true; 3231 switch (track->mState) { 3232 case TrackBase::STOPPING_1: 3233 // track stays active in STOPPING_1 state until first underrun 3234 if (recentUnderruns > 0 || track->isTerminated()) { 3235 track->mState = TrackBase::STOPPING_2; 3236 } 3237 break; 3238 case TrackBase::PAUSING: 3239 // ramp down is not yet implemented 3240 track->setPaused(); 3241 break; 3242 case TrackBase::RESUMING: 3243 // ramp up is not yet implemented 3244 track->mState = TrackBase::ACTIVE; 3245 break; 3246 case TrackBase::ACTIVE: 3247 if (recentFull > 0 || recentPartial > 0) { 3248 // track has provided at least some frames recently: reset retry count 3249 track->mRetryCount = kMaxTrackRetries; 3250 } 3251 if (recentUnderruns == 0) { 3252 // no recent underruns: stay active 3253 break; 3254 } 3255 // there has recently been an underrun of some kind 3256 if (track->sharedBuffer() == 0) { 3257 // were any of the recent underruns "empty" (no frames available)? 3258 if (recentEmpty == 0) { 3259 // no, then ignore the partial underruns as they are allowed indefinitely 3260 break; 3261 } 3262 // there has recently been an "empty" underrun: decrement the retry counter 3263 if (--(track->mRetryCount) > 0) { 3264 break; 3265 } 3266 // indicate to client process that the track was disabled because of underrun; 3267 // it will then automatically call start() when data is available 3268 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3269 // remove from active list, but state remains ACTIVE [confusing but true] 3270 isActive = false; 3271 break; 3272 } 3273 // fall through 3274 case TrackBase::STOPPING_2: 3275 case TrackBase::PAUSED: 3276 case TrackBase::STOPPED: 3277 case TrackBase::FLUSHED: // flush() while active 3278 // Check for presentation complete if track is inactive 3279 // We have consumed all the buffers of this track. 3280 // This would be incomplete if we auto-paused on underrun 3281 { 3282 size_t audioHALFrames = 3283 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3284 size_t framesWritten = mBytesWritten / mFrameSize; 3285 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3286 // track stays in active list until presentation is complete 3287 break; 3288 } 3289 } 3290 if (track->isStopping_2()) { 3291 track->mState = TrackBase::STOPPED; 3292 } 3293 if (track->isStopped()) { 3294 // Can't reset directly, as fast mixer is still polling this track 3295 // track->reset(); 3296 // So instead mark this track as needing to be reset after push with ack 3297 resetMask |= 1 << i; 3298 } 3299 isActive = false; 3300 break; 3301 case TrackBase::IDLE: 3302 default: 3303 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3304 } 3305 3306 if (isActive) { 3307 // was it previously inactive? 3308 if (!(state->mTrackMask & (1 << j))) { 3309 ExtendedAudioBufferProvider *eabp = track; 3310 VolumeProvider *vp = track; 3311 fastTrack->mBufferProvider = eabp; 3312 fastTrack->mVolumeProvider = vp; 3313 fastTrack->mChannelMask = track->mChannelMask; 3314 fastTrack->mFormat = track->mFormat; 3315 fastTrack->mGeneration++; 3316 state->mTrackMask |= 1 << j; 3317 didModify = true; 3318 // no acknowledgement required for newly active tracks 3319 } 3320 // cache the combined master volume and stream type volume for fast mixer; this 3321 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3322 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3323 ++fastTracks; 3324 } else { 3325 // was it previously active? 3326 if (state->mTrackMask & (1 << j)) { 3327 fastTrack->mBufferProvider = NULL; 3328 fastTrack->mGeneration++; 3329 state->mTrackMask &= ~(1 << j); 3330 didModify = true; 3331 // If any fast tracks were removed, we must wait for acknowledgement 3332 // because we're about to decrement the last sp<> on those tracks. 3333 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3334 } else { 3335 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3336 } 3337 tracksToRemove->add(track); 3338 // Avoids a misleading display in dumpsys 3339 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3340 } 3341 continue; 3342 } 3343 3344 { // local variable scope to avoid goto warning 3345 3346 audio_track_cblk_t* cblk = track->cblk(); 3347 3348 // The first time a track is added we wait 3349 // for all its buffers to be filled before processing it 3350 int name = track->name(); 3351 // make sure that we have enough frames to mix one full buffer. 3352 // enforce this condition only once to enable draining the buffer in case the client 3353 // app does not call stop() and relies on underrun to stop: 3354 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3355 // during last round 3356 size_t desiredFrames; 3357 uint32_t sr = track->sampleRate(); 3358 if (sr == mSampleRate) { 3359 desiredFrames = mNormalFrameCount; 3360 } else { 3361 // +1 for rounding and +1 for additional sample needed for interpolation 3362 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3363 // add frames already consumed but not yet released by the resampler 3364 // because mAudioTrackServerProxy->framesReady() will include these frames 3365 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3366#if 0 3367 // the minimum track buffer size is normally twice the number of frames necessary 3368 // to fill one buffer and the resampler should not leave more than one buffer worth 3369 // of unreleased frames after each pass, but just in case... 3370 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3371#endif 3372 } 3373 uint32_t minFrames = 1; 3374 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3375 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3376 minFrames = desiredFrames; 3377 } 3378 3379 size_t framesReady = track->framesReady(); 3380 if ((framesReady >= minFrames) && track->isReady() && 3381 !track->isPaused() && !track->isTerminated()) 3382 { 3383 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3384 3385 mixedTracks++; 3386 3387 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3388 // there is an effect chain connected to the track 3389 chain.clear(); 3390 if (track->mainBuffer() != mSinkBuffer && 3391 track->mainBuffer() != mMixerBuffer) { 3392 if (mEffectBufferEnabled) { 3393 mEffectBufferValid = true; // Later can set directly. 3394 } 3395 chain = getEffectChain_l(track->sessionId()); 3396 // Delegate volume control to effect in track effect chain if needed 3397 if (chain != 0) { 3398 tracksWithEffect++; 3399 } else { 3400 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3401 "session %d", 3402 name, track->sessionId()); 3403 } 3404 } 3405 3406 3407 int param = AudioMixer::VOLUME; 3408 if (track->mFillingUpStatus == Track::FS_FILLED) { 3409 // no ramp for the first volume setting 3410 track->mFillingUpStatus = Track::FS_ACTIVE; 3411 if (track->mState == TrackBase::RESUMING) { 3412 track->mState = TrackBase::ACTIVE; 3413 param = AudioMixer::RAMP_VOLUME; 3414 } 3415 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3416 // FIXME should not make a decision based on mServer 3417 } else if (cblk->mServer != 0) { 3418 // If the track is stopped before the first frame was mixed, 3419 // do not apply ramp 3420 param = AudioMixer::RAMP_VOLUME; 3421 } 3422 3423 // compute volume for this track 3424 uint32_t vl, vr; // in U8.24 integer format 3425 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3426 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3427 vl = vr = 0; 3428 vlf = vrf = vaf = 0.; 3429 if (track->isPausing()) { 3430 track->setPaused(); 3431 } 3432 } else { 3433 3434 // read original volumes with volume control 3435 float typeVolume = mStreamTypes[track->streamType()].volume; 3436 float v = masterVolume * typeVolume; 3437 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3438 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3439 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3440 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3441 // track volumes come from shared memory, so can't be trusted and must be clamped 3442 if (vlf > GAIN_FLOAT_UNITY) { 3443 ALOGV("Track left volume out of range: %.3g", vlf); 3444 vlf = GAIN_FLOAT_UNITY; 3445 } 3446 if (vrf > GAIN_FLOAT_UNITY) { 3447 ALOGV("Track right volume out of range: %.3g", vrf); 3448 vrf = GAIN_FLOAT_UNITY; 3449 } 3450 // now apply the master volume and stream type volume 3451 vlf *= v; 3452 vrf *= v; 3453 // assuming master volume and stream type volume each go up to 1.0, 3454 // then derive vl and vr as U8.24 versions for the effect chain 3455 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3456 vl = (uint32_t) (scaleto8_24 * vlf); 3457 vr = (uint32_t) (scaleto8_24 * vrf); 3458 // vl and vr are now in U8.24 format 3459 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3460 // send level comes from shared memory and so may be corrupt 3461 if (sendLevel > MAX_GAIN_INT) { 3462 ALOGV("Track send level out of range: %04X", sendLevel); 3463 sendLevel = MAX_GAIN_INT; 3464 } 3465 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3466 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3467 } 3468 3469 // Delegate volume control to effect in track effect chain if needed 3470 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3471 // Do not ramp volume if volume is controlled by effect 3472 param = AudioMixer::VOLUME; 3473 // Update remaining floating point volume levels 3474 vlf = (float)vl / (1 << 24); 3475 vrf = (float)vr / (1 << 24); 3476 track->mHasVolumeController = true; 3477 } else { 3478 // force no volume ramp when volume controller was just disabled or removed 3479 // from effect chain to avoid volume spike 3480 if (track->mHasVolumeController) { 3481 param = AudioMixer::VOLUME; 3482 } 3483 track->mHasVolumeController = false; 3484 } 3485 3486 // XXX: these things DON'T need to be done each time 3487 mAudioMixer->setBufferProvider(name, track); 3488 mAudioMixer->enable(name); 3489 3490 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3491 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3492 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3493 mAudioMixer->setParameter( 3494 name, 3495 AudioMixer::TRACK, 3496 AudioMixer::FORMAT, (void *)track->format()); 3497 mAudioMixer->setParameter( 3498 name, 3499 AudioMixer::TRACK, 3500 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3501 mAudioMixer->setParameter( 3502 name, 3503 AudioMixer::TRACK, 3504 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3505 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3506 uint32_t maxSampleRate = mSampleRate * 2; 3507 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3508 if (reqSampleRate == 0) { 3509 reqSampleRate = mSampleRate; 3510 } else if (reqSampleRate > maxSampleRate) { 3511 reqSampleRate = maxSampleRate; 3512 } 3513 mAudioMixer->setParameter( 3514 name, 3515 AudioMixer::RESAMPLE, 3516 AudioMixer::SAMPLE_RATE, 3517 (void *)(uintptr_t)reqSampleRate); 3518 /* 3519 * Select the appropriate output buffer for the track. 3520 * 3521 * Tracks with effects go into their own effects chain buffer 3522 * and from there into either mEffectBuffer or mSinkBuffer. 3523 * 3524 * Other tracks can use mMixerBuffer for higher precision 3525 * channel accumulation. If this buffer is enabled 3526 * (mMixerBufferEnabled true), then selected tracks will accumulate 3527 * into it. 3528 * 3529 */ 3530 if (mMixerBufferEnabled 3531 && (track->mainBuffer() == mSinkBuffer 3532 || track->mainBuffer() == mMixerBuffer)) { 3533 mAudioMixer->setParameter( 3534 name, 3535 AudioMixer::TRACK, 3536 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3537 mAudioMixer->setParameter( 3538 name, 3539 AudioMixer::TRACK, 3540 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3541 // TODO: override track->mainBuffer()? 3542 mMixerBufferValid = true; 3543 } else { 3544 mAudioMixer->setParameter( 3545 name, 3546 AudioMixer::TRACK, 3547 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3548 mAudioMixer->setParameter( 3549 name, 3550 AudioMixer::TRACK, 3551 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3552 } 3553 mAudioMixer->setParameter( 3554 name, 3555 AudioMixer::TRACK, 3556 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3557 3558 // reset retry count 3559 track->mRetryCount = kMaxTrackRetries; 3560 3561 // If one track is ready, set the mixer ready if: 3562 // - the mixer was not ready during previous round OR 3563 // - no other track is not ready 3564 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3565 mixerStatus != MIXER_TRACKS_ENABLED) { 3566 mixerStatus = MIXER_TRACKS_READY; 3567 } 3568 } else { 3569 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3570 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3571 } 3572 // clear effect chain input buffer if an active track underruns to avoid sending 3573 // previous audio buffer again to effects 3574 chain = getEffectChain_l(track->sessionId()); 3575 if (chain != 0) { 3576 chain->clearInputBuffer(); 3577 } 3578 3579 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3580 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3581 track->isStopped() || track->isPaused()) { 3582 // We have consumed all the buffers of this track. 3583 // Remove it from the list of active tracks. 3584 // TODO: use actual buffer filling status instead of latency when available from 3585 // audio HAL 3586 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3587 size_t framesWritten = mBytesWritten / mFrameSize; 3588 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3589 if (track->isStopped()) { 3590 track->reset(); 3591 } 3592 tracksToRemove->add(track); 3593 } 3594 } else { 3595 // No buffers for this track. Give it a few chances to 3596 // fill a buffer, then remove it from active list. 3597 if (--(track->mRetryCount) <= 0) { 3598 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3599 tracksToRemove->add(track); 3600 // indicate to client process that the track was disabled because of underrun; 3601 // it will then automatically call start() when data is available 3602 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3603 // If one track is not ready, mark the mixer also not ready if: 3604 // - the mixer was ready during previous round OR 3605 // - no other track is ready 3606 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3607 mixerStatus != MIXER_TRACKS_READY) { 3608 mixerStatus = MIXER_TRACKS_ENABLED; 3609 } 3610 } 3611 mAudioMixer->disable(name); 3612 } 3613 3614 } // local variable scope to avoid goto warning 3615track_is_ready: ; 3616 3617 } 3618 3619 // Push the new FastMixer state if necessary 3620 bool pauseAudioWatchdog = false; 3621 if (didModify) { 3622 state->mFastTracksGen++; 3623 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3624 if (kUseFastMixer == FastMixer_Dynamic && 3625 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3626 state->mCommand = FastMixerState::COLD_IDLE; 3627 state->mColdFutexAddr = &mFastMixerFutex; 3628 state->mColdGen++; 3629 mFastMixerFutex = 0; 3630 if (kUseFastMixer == FastMixer_Dynamic) { 3631 mNormalSink = mOutputSink; 3632 } 3633 // If we go into cold idle, need to wait for acknowledgement 3634 // so that fast mixer stops doing I/O. 3635 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3636 pauseAudioWatchdog = true; 3637 } 3638 } 3639 if (sq != NULL) { 3640 sq->end(didModify); 3641 sq->push(block); 3642 } 3643#ifdef AUDIO_WATCHDOG 3644 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3645 mAudioWatchdog->pause(); 3646 } 3647#endif 3648 3649 // Now perform the deferred reset on fast tracks that have stopped 3650 while (resetMask != 0) { 3651 size_t i = __builtin_ctz(resetMask); 3652 ALOG_ASSERT(i < count); 3653 resetMask &= ~(1 << i); 3654 sp<Track> t = mActiveTracks[i].promote(); 3655 if (t == 0) { 3656 continue; 3657 } 3658 Track* track = t.get(); 3659 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3660 track->reset(); 3661 } 3662 3663 // remove all the tracks that need to be... 3664 removeTracks_l(*tracksToRemove); 3665 3666 // sink or mix buffer must be cleared if all tracks are connected to an 3667 // effect chain as in this case the mixer will not write to the sink or mix buffer 3668 // and track effects will accumulate into it 3669 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3670 (mixedTracks == 0 && fastTracks > 0))) { 3671 // FIXME as a performance optimization, should remember previous zero status 3672 if (mMixerBufferValid) { 3673 memset(mMixerBuffer, 0, mMixerBufferSize); 3674 // TODO: In testing, mSinkBuffer below need not be cleared because 3675 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3676 // after mixing. 3677 // 3678 // To enforce this guarantee: 3679 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3680 // (mixedTracks == 0 && fastTracks > 0)) 3681 // must imply MIXER_TRACKS_READY. 3682 // Later, we may clear buffers regardless, and skip much of this logic. 3683 } 3684 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3685 if (mEffectBufferValid) { 3686 memset(mEffectBuffer, 0, mEffectBufferSize); 3687 } 3688 // FIXME as a performance optimization, should remember previous zero status 3689 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3690 } 3691 3692 // if any fast tracks, then status is ready 3693 mMixerStatusIgnoringFastTracks = mixerStatus; 3694 if (fastTracks > 0) { 3695 mixerStatus = MIXER_TRACKS_READY; 3696 } 3697 return mixerStatus; 3698} 3699 3700// getTrackName_l() must be called with ThreadBase::mLock held 3701int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3702 audio_format_t format, int sessionId) 3703{ 3704 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3705} 3706 3707// deleteTrackName_l() must be called with ThreadBase::mLock held 3708void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3709{ 3710 ALOGV("remove track (%d) and delete from mixer", name); 3711 mAudioMixer->deleteTrackName(name); 3712} 3713 3714// checkForNewParameter_l() must be called with ThreadBase::mLock held 3715bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3716 status_t& status) 3717{ 3718 bool reconfig = false; 3719 3720 status = NO_ERROR; 3721 3722 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3723 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3724 if (mFastMixer != 0) { 3725 FastMixerStateQueue *sq = mFastMixer->sq(); 3726 FastMixerState *state = sq->begin(); 3727 if (!(state->mCommand & FastMixerState::IDLE)) { 3728 previousCommand = state->mCommand; 3729 state->mCommand = FastMixerState::HOT_IDLE; 3730 sq->end(); 3731 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3732 } else { 3733 sq->end(false /*didModify*/); 3734 } 3735 } 3736 3737 AudioParameter param = AudioParameter(keyValuePair); 3738 int value; 3739 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3740 reconfig = true; 3741 } 3742 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3743 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3744 status = BAD_VALUE; 3745 } else { 3746 // no need to save value, since it's constant 3747 reconfig = true; 3748 } 3749 } 3750 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3751 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3752 status = BAD_VALUE; 3753 } else { 3754 // no need to save value, since it's constant 3755 reconfig = true; 3756 } 3757 } 3758 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3759 // do not accept frame count changes if tracks are open as the track buffer 3760 // size depends on frame count and correct behavior would not be guaranteed 3761 // if frame count is changed after track creation 3762 if (!mTracks.isEmpty()) { 3763 status = INVALID_OPERATION; 3764 } else { 3765 reconfig = true; 3766 } 3767 } 3768 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3769#ifdef ADD_BATTERY_DATA 3770 // when changing the audio output device, call addBatteryData to notify 3771 // the change 3772 if (mOutDevice != value) { 3773 uint32_t params = 0; 3774 // check whether speaker is on 3775 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3776 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3777 } 3778 3779 audio_devices_t deviceWithoutSpeaker 3780 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3781 // check if any other device (except speaker) is on 3782 if (value & deviceWithoutSpeaker ) { 3783 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3784 } 3785 3786 if (params != 0) { 3787 addBatteryData(params); 3788 } 3789 } 3790#endif 3791 3792 // forward device change to effects that have requested to be 3793 // aware of attached audio device. 3794 if (value != AUDIO_DEVICE_NONE) { 3795 mOutDevice = value; 3796 for (size_t i = 0; i < mEffectChains.size(); i++) { 3797 mEffectChains[i]->setDevice_l(mOutDevice); 3798 } 3799 } 3800 } 3801 3802 if (status == NO_ERROR) { 3803 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3804 keyValuePair.string()); 3805 if (!mStandby && status == INVALID_OPERATION) { 3806 mOutput->stream->common.standby(&mOutput->stream->common); 3807 mStandby = true; 3808 mBytesWritten = 0; 3809 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3810 keyValuePair.string()); 3811 } 3812 if (status == NO_ERROR && reconfig) { 3813 readOutputParameters_l(); 3814 delete mAudioMixer; 3815 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3816 for (size_t i = 0; i < mTracks.size() ; i++) { 3817 int name = getTrackName_l(mTracks[i]->mChannelMask, 3818 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3819 if (name < 0) { 3820 break; 3821 } 3822 mTracks[i]->mName = name; 3823 } 3824 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3825 } 3826 } 3827 3828 if (!(previousCommand & FastMixerState::IDLE)) { 3829 ALOG_ASSERT(mFastMixer != 0); 3830 FastMixerStateQueue *sq = mFastMixer->sq(); 3831 FastMixerState *state = sq->begin(); 3832 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3833 state->mCommand = previousCommand; 3834 sq->end(); 3835 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3836 } 3837 3838 return reconfig; 3839} 3840 3841 3842void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3843{ 3844 const size_t SIZE = 256; 3845 char buffer[SIZE]; 3846 String8 result; 3847 3848 PlaybackThread::dumpInternals(fd, args); 3849 3850 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3851 3852 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3853 const FastMixerDumpState copy(mFastMixerDumpState); 3854 copy.dump(fd); 3855 3856#ifdef STATE_QUEUE_DUMP 3857 // Similar for state queue 3858 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3859 observerCopy.dump(fd); 3860 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3861 mutatorCopy.dump(fd); 3862#endif 3863 3864#ifdef TEE_SINK 3865 // Write the tee output to a .wav file 3866 dumpTee(fd, mTeeSource, mId); 3867#endif 3868 3869#ifdef AUDIO_WATCHDOG 3870 if (mAudioWatchdog != 0) { 3871 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3872 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3873 wdCopy.dump(fd); 3874 } 3875#endif 3876} 3877 3878uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3879{ 3880 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3881} 3882 3883uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3884{ 3885 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3886} 3887 3888void AudioFlinger::MixerThread::cacheParameters_l() 3889{ 3890 PlaybackThread::cacheParameters_l(); 3891 3892 // FIXME: Relaxed timing because of a certain device that can't meet latency 3893 // Should be reduced to 2x after the vendor fixes the driver issue 3894 // increase threshold again due to low power audio mode. The way this warning 3895 // threshold is calculated and its usefulness should be reconsidered anyway. 3896 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3897} 3898 3899// ---------------------------------------------------------------------------- 3900 3901AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3902 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3903 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3904 // mLeftVolFloat, mRightVolFloat 3905{ 3906} 3907 3908AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3909 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3910 ThreadBase::type_t type) 3911 : PlaybackThread(audioFlinger, output, id, device, type) 3912 // mLeftVolFloat, mRightVolFloat 3913{ 3914} 3915 3916AudioFlinger::DirectOutputThread::~DirectOutputThread() 3917{ 3918} 3919 3920void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3921{ 3922 audio_track_cblk_t* cblk = track->cblk(); 3923 float left, right; 3924 3925 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3926 left = right = 0; 3927 } else { 3928 float typeVolume = mStreamTypes[track->streamType()].volume; 3929 float v = mMasterVolume * typeVolume; 3930 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3931 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3932 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3933 if (left > GAIN_FLOAT_UNITY) { 3934 left = GAIN_FLOAT_UNITY; 3935 } 3936 left *= v; 3937 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3938 if (right > GAIN_FLOAT_UNITY) { 3939 right = GAIN_FLOAT_UNITY; 3940 } 3941 right *= v; 3942 } 3943 3944 if (lastTrack) { 3945 if (left != mLeftVolFloat || right != mRightVolFloat) { 3946 mLeftVolFloat = left; 3947 mRightVolFloat = right; 3948 3949 // Convert volumes from float to 8.24 3950 uint32_t vl = (uint32_t)(left * (1 << 24)); 3951 uint32_t vr = (uint32_t)(right * (1 << 24)); 3952 3953 // Delegate volume control to effect in track effect chain if needed 3954 // only one effect chain can be present on DirectOutputThread, so if 3955 // there is one, the track is connected to it 3956 if (!mEffectChains.isEmpty()) { 3957 mEffectChains[0]->setVolume_l(&vl, &vr); 3958 left = (float)vl / (1 << 24); 3959 right = (float)vr / (1 << 24); 3960 } 3961 if (mOutput->stream->set_volume) { 3962 mOutput->stream->set_volume(mOutput->stream, left, right); 3963 } 3964 } 3965 } 3966} 3967 3968 3969AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3970 Vector< sp<Track> > *tracksToRemove 3971) 3972{ 3973 size_t count = mActiveTracks.size(); 3974 mixer_state mixerStatus = MIXER_IDLE; 3975 3976 // find out which tracks need to be processed 3977 for (size_t i = 0; i < count; i++) { 3978 sp<Track> t = mActiveTracks[i].promote(); 3979 // The track died recently 3980 if (t == 0) { 3981 continue; 3982 } 3983 3984 Track* const track = t.get(); 3985 audio_track_cblk_t* cblk = track->cblk(); 3986 // Only consider last track started for volume and mixer state control. 3987 // In theory an older track could underrun and restart after the new one starts 3988 // but as we only care about the transition phase between two tracks on a 3989 // direct output, it is not a problem to ignore the underrun case. 3990 sp<Track> l = mLatestActiveTrack.promote(); 3991 bool last = l.get() == track; 3992 3993 // The first time a track is added we wait 3994 // for all its buffers to be filled before processing it 3995 uint32_t minFrames; 3996 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 3997 minFrames = mNormalFrameCount; 3998 } else { 3999 minFrames = 1; 4000 } 4001 4002 ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ", 4003 minFrames, track->mState, track->framesReady()); 4004 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4005 !track->isStopping_2() && !track->isStopped()) 4006 { 4007 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4008 4009 if (track->mFillingUpStatus == Track::FS_FILLED) { 4010 track->mFillingUpStatus = Track::FS_ACTIVE; 4011 // make sure processVolume_l() will apply new volume even if 0 4012 mLeftVolFloat = mRightVolFloat = -1.0; 4013 if (track->mState == TrackBase::RESUMING) { 4014 track->mState = TrackBase::ACTIVE; 4015 } 4016 } 4017 4018 // compute volume for this track 4019 processVolume_l(track, last); 4020 if (last) { 4021 // reset retry count 4022 track->mRetryCount = kMaxTrackRetriesDirect; 4023 mActiveTrack = t; 4024 mixerStatus = MIXER_TRACKS_READY; 4025 } 4026 } else { 4027 // clear effect chain input buffer if the last active track started underruns 4028 // to avoid sending previous audio buffer again to effects 4029 if (!mEffectChains.isEmpty() && last) { 4030 mEffectChains[0]->clearInputBuffer(); 4031 } 4032 if (track->isStopping_1()) { 4033 track->mState = TrackBase::STOPPING_2; 4034 } 4035 if ((track->sharedBuffer() != 0) || track->isStopped() || 4036 track->isStopping_2() || track->isPaused()) { 4037 // We have consumed all the buffers of this track. 4038 // Remove it from the list of active tracks. 4039 size_t audioHALFrames; 4040 if (audio_is_linear_pcm(mFormat)) { 4041 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4042 } else { 4043 audioHALFrames = 0; 4044 } 4045 4046 size_t framesWritten = mBytesWritten / mFrameSize; 4047 if (mStandby || !last || 4048 track->presentationComplete(framesWritten, audioHALFrames)) { 4049 if (track->isStopping_2()) { 4050 track->mState = TrackBase::STOPPED; 4051 } 4052 if (track->isStopped()) { 4053 track->reset(); 4054 } 4055 tracksToRemove->add(track); 4056 } 4057 } else { 4058 // No buffers for this track. Give it a few chances to 4059 // fill a buffer, then remove it from active list. 4060 // Only consider last track started for mixer state control 4061 if (--(track->mRetryCount) <= 0) { 4062 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4063 tracksToRemove->add(track); 4064 // indicate to client process that the track was disabled because of underrun; 4065 // it will then automatically call start() when data is available 4066 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4067 } else if (last) { 4068 mixerStatus = MIXER_TRACKS_ENABLED; 4069 } 4070 } 4071 } 4072 } 4073 4074 // remove all the tracks that need to be... 4075 removeTracks_l(*tracksToRemove); 4076 4077 return mixerStatus; 4078} 4079 4080void AudioFlinger::DirectOutputThread::threadLoop_mix() 4081{ 4082 size_t frameCount = mFrameCount; 4083 int8_t *curBuf = (int8_t *)mSinkBuffer; 4084 // output audio to hardware 4085 while (frameCount) { 4086 AudioBufferProvider::Buffer buffer; 4087 buffer.frameCount = frameCount; 4088 mActiveTrack->getNextBuffer(&buffer); 4089 if (buffer.raw == NULL) { 4090 memset(curBuf, 0, frameCount * mFrameSize); 4091 break; 4092 } 4093 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4094 frameCount -= buffer.frameCount; 4095 curBuf += buffer.frameCount * mFrameSize; 4096 mActiveTrack->releaseBuffer(&buffer); 4097 } 4098 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4099 sleepTime = 0; 4100 standbyTime = systemTime() + standbyDelay; 4101 mActiveTrack.clear(); 4102} 4103 4104void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4105{ 4106 if (sleepTime == 0) { 4107 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4108 sleepTime = activeSleepTime; 4109 } else { 4110 sleepTime = idleSleepTime; 4111 } 4112 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4113 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4114 sleepTime = 0; 4115 } 4116} 4117 4118// getTrackName_l() must be called with ThreadBase::mLock held 4119int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4120 audio_format_t format __unused, int sessionId __unused) 4121{ 4122 return 0; 4123} 4124 4125// deleteTrackName_l() must be called with ThreadBase::mLock held 4126void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4127{ 4128} 4129 4130// checkForNewParameter_l() must be called with ThreadBase::mLock held 4131bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4132 status_t& status) 4133{ 4134 bool reconfig = false; 4135 4136 status = NO_ERROR; 4137 4138 AudioParameter param = AudioParameter(keyValuePair); 4139 int value; 4140 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4141 // forward device change to effects that have requested to be 4142 // aware of attached audio device. 4143 if (value != AUDIO_DEVICE_NONE) { 4144 mOutDevice = value; 4145 for (size_t i = 0; i < mEffectChains.size(); i++) { 4146 mEffectChains[i]->setDevice_l(mOutDevice); 4147 } 4148 } 4149 } 4150 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4151 // do not accept frame count changes if tracks are open as the track buffer 4152 // size depends on frame count and correct behavior would not be garantied 4153 // if frame count is changed after track creation 4154 if (!mTracks.isEmpty()) { 4155 status = INVALID_OPERATION; 4156 } else { 4157 reconfig = true; 4158 } 4159 } 4160 if (status == NO_ERROR) { 4161 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4162 keyValuePair.string()); 4163 if (!mStandby && status == INVALID_OPERATION) { 4164 mOutput->stream->common.standby(&mOutput->stream->common); 4165 mStandby = true; 4166 mBytesWritten = 0; 4167 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4168 keyValuePair.string()); 4169 } 4170 if (status == NO_ERROR && reconfig) { 4171 readOutputParameters_l(); 4172 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4173 } 4174 } 4175 4176 return reconfig; 4177} 4178 4179uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4180{ 4181 uint32_t time; 4182 if (audio_is_linear_pcm(mFormat)) { 4183 time = PlaybackThread::activeSleepTimeUs(); 4184 } else { 4185 time = 10000; 4186 } 4187 return time; 4188} 4189 4190uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4191{ 4192 uint32_t time; 4193 if (audio_is_linear_pcm(mFormat)) { 4194 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4195 } else { 4196 time = 10000; 4197 } 4198 return time; 4199} 4200 4201uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4202{ 4203 uint32_t time; 4204 if (audio_is_linear_pcm(mFormat)) { 4205 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4206 } else { 4207 time = 10000; 4208 } 4209 return time; 4210} 4211 4212void AudioFlinger::DirectOutputThread::cacheParameters_l() 4213{ 4214 PlaybackThread::cacheParameters_l(); 4215 4216 // use shorter standby delay as on normal output to release 4217 // hardware resources as soon as possible 4218 if (audio_is_linear_pcm(mFormat)) { 4219 standbyDelay = microseconds(activeSleepTime*2); 4220 } else { 4221 standbyDelay = kOffloadStandbyDelayNs; 4222 } 4223} 4224 4225// ---------------------------------------------------------------------------- 4226 4227AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4228 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4229 : Thread(false /*canCallJava*/), 4230 mPlaybackThread(playbackThread), 4231 mWriteAckSequence(0), 4232 mDrainSequence(0) 4233{ 4234} 4235 4236AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4237{ 4238} 4239 4240void AudioFlinger::AsyncCallbackThread::onFirstRef() 4241{ 4242 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4243} 4244 4245bool AudioFlinger::AsyncCallbackThread::threadLoop() 4246{ 4247 while (!exitPending()) { 4248 uint32_t writeAckSequence; 4249 uint32_t drainSequence; 4250 4251 { 4252 Mutex::Autolock _l(mLock); 4253 while (!((mWriteAckSequence & 1) || 4254 (mDrainSequence & 1) || 4255 exitPending())) { 4256 mWaitWorkCV.wait(mLock); 4257 } 4258 4259 if (exitPending()) { 4260 break; 4261 } 4262 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4263 mWriteAckSequence, mDrainSequence); 4264 writeAckSequence = mWriteAckSequence; 4265 mWriteAckSequence &= ~1; 4266 drainSequence = mDrainSequence; 4267 mDrainSequence &= ~1; 4268 } 4269 { 4270 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4271 if (playbackThread != 0) { 4272 if (writeAckSequence & 1) { 4273 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4274 } 4275 if (drainSequence & 1) { 4276 playbackThread->resetDraining(drainSequence >> 1); 4277 } 4278 } 4279 } 4280 } 4281 return false; 4282} 4283 4284void AudioFlinger::AsyncCallbackThread::exit() 4285{ 4286 ALOGV("AsyncCallbackThread::exit"); 4287 Mutex::Autolock _l(mLock); 4288 requestExit(); 4289 mWaitWorkCV.broadcast(); 4290} 4291 4292void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4293{ 4294 Mutex::Autolock _l(mLock); 4295 // bit 0 is cleared 4296 mWriteAckSequence = sequence << 1; 4297} 4298 4299void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4300{ 4301 Mutex::Autolock _l(mLock); 4302 // ignore unexpected callbacks 4303 if (mWriteAckSequence & 2) { 4304 mWriteAckSequence |= 1; 4305 mWaitWorkCV.signal(); 4306 } 4307} 4308 4309void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4310{ 4311 Mutex::Autolock _l(mLock); 4312 // bit 0 is cleared 4313 mDrainSequence = sequence << 1; 4314} 4315 4316void AudioFlinger::AsyncCallbackThread::resetDraining() 4317{ 4318 Mutex::Autolock _l(mLock); 4319 // ignore unexpected callbacks 4320 if (mDrainSequence & 2) { 4321 mDrainSequence |= 1; 4322 mWaitWorkCV.signal(); 4323 } 4324} 4325 4326 4327// ---------------------------------------------------------------------------- 4328AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4329 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4330 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4331 mHwPaused(false), 4332 mFlushPending(false), 4333 mPausedBytesRemaining(0) 4334{ 4335 //FIXME: mStandby should be set to true by ThreadBase constructor 4336 mStandby = true; 4337} 4338 4339void AudioFlinger::OffloadThread::threadLoop_exit() 4340{ 4341 if (mFlushPending || mHwPaused) { 4342 // If a flush is pending or track was paused, just discard buffered data 4343 flushHw_l(); 4344 } else { 4345 mMixerStatus = MIXER_DRAIN_ALL; 4346 threadLoop_drain(); 4347 } 4348 if (mUseAsyncWrite) { 4349 ALOG_ASSERT(mCallbackThread != 0); 4350 mCallbackThread->exit(); 4351 } 4352 PlaybackThread::threadLoop_exit(); 4353} 4354 4355AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4356 Vector< sp<Track> > *tracksToRemove 4357) 4358{ 4359 size_t count = mActiveTracks.size(); 4360 4361 mixer_state mixerStatus = MIXER_IDLE; 4362 bool doHwPause = false; 4363 bool doHwResume = false; 4364 4365 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4366 4367 // find out which tracks need to be processed 4368 for (size_t i = 0; i < count; i++) { 4369 sp<Track> t = mActiveTracks[i].promote(); 4370 // The track died recently 4371 if (t == 0) { 4372 continue; 4373 } 4374 Track* const track = t.get(); 4375 audio_track_cblk_t* cblk = track->cblk(); 4376 // Only consider last track started for volume and mixer state control. 4377 // In theory an older track could underrun and restart after the new one starts 4378 // but as we only care about the transition phase between two tracks on a 4379 // direct output, it is not a problem to ignore the underrun case. 4380 sp<Track> l = mLatestActiveTrack.promote(); 4381 bool last = l.get() == track; 4382 4383 if (track->isInvalid()) { 4384 ALOGW("An invalidated track shouldn't be in active list"); 4385 tracksToRemove->add(track); 4386 continue; 4387 } 4388 4389 if (track->mState == TrackBase::IDLE) { 4390 ALOGW("An idle track shouldn't be in active list"); 4391 continue; 4392 } 4393 4394 if (track->isPausing()) { 4395 track->setPaused(); 4396 if (last) { 4397 if (!mHwPaused) { 4398 doHwPause = true; 4399 mHwPaused = true; 4400 } 4401 // If we were part way through writing the mixbuffer to 4402 // the HAL we must save this until we resume 4403 // BUG - this will be wrong if a different track is made active, 4404 // in that case we want to discard the pending data in the 4405 // mixbuffer and tell the client to present it again when the 4406 // track is resumed 4407 mPausedWriteLength = mCurrentWriteLength; 4408 mPausedBytesRemaining = mBytesRemaining; 4409 mBytesRemaining = 0; // stop writing 4410 } 4411 tracksToRemove->add(track); 4412 } else if (track->isFlushPending()) { 4413 track->flushAck(); 4414 if (last) { 4415 mFlushPending = true; 4416 } 4417 } else if (track->isResumePending()){ 4418 track->resumeAck(); 4419 if (last) { 4420 if (mPausedBytesRemaining) { 4421 // Need to continue write that was interrupted 4422 mCurrentWriteLength = mPausedWriteLength; 4423 mBytesRemaining = mPausedBytesRemaining; 4424 mPausedBytesRemaining = 0; 4425 } 4426 if (mHwPaused) { 4427 doHwResume = true; 4428 mHwPaused = false; 4429 // threadLoop_mix() will handle the case that we need to 4430 // resume an interrupted write 4431 } 4432 // enable write to audio HAL 4433 sleepTime = 0; 4434 4435 // Do not handle new data in this iteration even if track->framesReady() 4436 mixerStatus = MIXER_TRACKS_ENABLED; 4437 } 4438 } else if (track->framesReady() && track->isReady() && 4439 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4440 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4441 if (track->mFillingUpStatus == Track::FS_FILLED) { 4442 track->mFillingUpStatus = Track::FS_ACTIVE; 4443 // make sure processVolume_l() will apply new volume even if 0 4444 mLeftVolFloat = mRightVolFloat = -1.0; 4445 } 4446 4447 if (last) { 4448 sp<Track> previousTrack = mPreviousTrack.promote(); 4449 if (previousTrack != 0) { 4450 if (track != previousTrack.get()) { 4451 // Flush any data still being written from last track 4452 mBytesRemaining = 0; 4453 if (mPausedBytesRemaining) { 4454 // Last track was paused so we also need to flush saved 4455 // mixbuffer state and invalidate track so that it will 4456 // re-submit that unwritten data when it is next resumed 4457 mPausedBytesRemaining = 0; 4458 // Invalidate is a bit drastic - would be more efficient 4459 // to have a flag to tell client that some of the 4460 // previously written data was lost 4461 previousTrack->invalidate(); 4462 } 4463 // flush data already sent to the DSP if changing audio session as audio 4464 // comes from a different source. Also invalidate previous track to force a 4465 // seek when resuming. 4466 if (previousTrack->sessionId() != track->sessionId()) { 4467 previousTrack->invalidate(); 4468 } 4469 } 4470 } 4471 mPreviousTrack = track; 4472 // reset retry count 4473 track->mRetryCount = kMaxTrackRetriesOffload; 4474 mActiveTrack = t; 4475 mixerStatus = MIXER_TRACKS_READY; 4476 } 4477 } else { 4478 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4479 if (track->isStopping_1()) { 4480 // Hardware buffer can hold a large amount of audio so we must 4481 // wait for all current track's data to drain before we say 4482 // that the track is stopped. 4483 if (mBytesRemaining == 0) { 4484 // Only start draining when all data in mixbuffer 4485 // has been written 4486 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4487 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4488 // do not drain if no data was ever sent to HAL (mStandby == true) 4489 if (last && !mStandby) { 4490 // do not modify drain sequence if we are already draining. This happens 4491 // when resuming from pause after drain. 4492 if ((mDrainSequence & 1) == 0) { 4493 sleepTime = 0; 4494 standbyTime = systemTime() + standbyDelay; 4495 mixerStatus = MIXER_DRAIN_TRACK; 4496 mDrainSequence += 2; 4497 } 4498 if (mHwPaused) { 4499 // It is possible to move from PAUSED to STOPPING_1 without 4500 // a resume so we must ensure hardware is running 4501 doHwResume = true; 4502 mHwPaused = false; 4503 } 4504 } 4505 } 4506 } else if (track->isStopping_2()) { 4507 // Drain has completed or we are in standby, signal presentation complete 4508 if (!(mDrainSequence & 1) || !last || mStandby) { 4509 track->mState = TrackBase::STOPPED; 4510 size_t audioHALFrames = 4511 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4512 size_t framesWritten = 4513 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4514 track->presentationComplete(framesWritten, audioHALFrames); 4515 track->reset(); 4516 tracksToRemove->add(track); 4517 } 4518 } else { 4519 // No buffers for this track. Give it a few chances to 4520 // fill a buffer, then remove it from active list. 4521 if (--(track->mRetryCount) <= 0) { 4522 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4523 track->name()); 4524 tracksToRemove->add(track); 4525 // indicate to client process that the track was disabled because of underrun; 4526 // it will then automatically call start() when data is available 4527 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4528 } else if (last){ 4529 mixerStatus = MIXER_TRACKS_ENABLED; 4530 } 4531 } 4532 } 4533 // compute volume for this track 4534 processVolume_l(track, last); 4535 } 4536 4537 // make sure the pause/flush/resume sequence is executed in the right order. 4538 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4539 // before flush and then resume HW. This can happen in case of pause/flush/resume 4540 // if resume is received before pause is executed. 4541 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4542 mOutput->stream->pause(mOutput->stream); 4543 } 4544 if (mFlushPending) { 4545 flushHw_l(); 4546 mFlushPending = false; 4547 } 4548 if (!mStandby && doHwResume) { 4549 mOutput->stream->resume(mOutput->stream); 4550 } 4551 4552 // remove all the tracks that need to be... 4553 removeTracks_l(*tracksToRemove); 4554 4555 return mixerStatus; 4556} 4557 4558// must be called with thread mutex locked 4559bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4560{ 4561 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4562 mWriteAckSequence, mDrainSequence); 4563 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4564 return true; 4565 } 4566 return false; 4567} 4568 4569// must be called with thread mutex locked 4570bool AudioFlinger::OffloadThread::shouldStandby_l() 4571{ 4572 bool trackPaused = false; 4573 4574 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4575 // after a timeout and we will enter standby then. 4576 if (mTracks.size() > 0) { 4577 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4578 } 4579 4580 return !mStandby && !trackPaused; 4581} 4582 4583 4584bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4585{ 4586 Mutex::Autolock _l(mLock); 4587 return waitingAsyncCallback_l(); 4588} 4589 4590void AudioFlinger::OffloadThread::flushHw_l() 4591{ 4592 mOutput->stream->flush(mOutput->stream); 4593 // Flush anything still waiting in the mixbuffer 4594 mCurrentWriteLength = 0; 4595 mBytesRemaining = 0; 4596 mPausedWriteLength = 0; 4597 mPausedBytesRemaining = 0; 4598 mHwPaused = false; 4599 4600 if (mUseAsyncWrite) { 4601 // discard any pending drain or write ack by incrementing sequence 4602 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4603 mDrainSequence = (mDrainSequence + 2) & ~1; 4604 ALOG_ASSERT(mCallbackThread != 0); 4605 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4606 mCallbackThread->setDraining(mDrainSequence); 4607 } 4608} 4609 4610void AudioFlinger::OffloadThread::onAddNewTrack_l() 4611{ 4612 sp<Track> previousTrack = mPreviousTrack.promote(); 4613 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4614 4615 if (previousTrack != 0 && latestTrack != 0 && 4616 (previousTrack->sessionId() != latestTrack->sessionId())) { 4617 mFlushPending = true; 4618 } 4619 PlaybackThread::onAddNewTrack_l(); 4620} 4621 4622// ---------------------------------------------------------------------------- 4623 4624AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4625 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4626 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4627 DUPLICATING), 4628 mWaitTimeMs(UINT_MAX) 4629{ 4630 addOutputTrack(mainThread); 4631} 4632 4633AudioFlinger::DuplicatingThread::~DuplicatingThread() 4634{ 4635 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4636 mOutputTracks[i]->destroy(); 4637 } 4638} 4639 4640void AudioFlinger::DuplicatingThread::threadLoop_mix() 4641{ 4642 // mix buffers... 4643 if (outputsReady(outputTracks)) { 4644 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4645 } else { 4646 memset(mSinkBuffer, 0, mSinkBufferSize); 4647 } 4648 sleepTime = 0; 4649 writeFrames = mNormalFrameCount; 4650 mCurrentWriteLength = mSinkBufferSize; 4651 standbyTime = systemTime() + standbyDelay; 4652} 4653 4654void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4655{ 4656 if (sleepTime == 0) { 4657 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4658 sleepTime = activeSleepTime; 4659 } else { 4660 sleepTime = idleSleepTime; 4661 } 4662 } else if (mBytesWritten != 0) { 4663 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4664 writeFrames = mNormalFrameCount; 4665 memset(mSinkBuffer, 0, mSinkBufferSize); 4666 } else { 4667 // flush remaining overflow buffers in output tracks 4668 writeFrames = 0; 4669 } 4670 sleepTime = 0; 4671 } 4672} 4673 4674ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4675{ 4676 for (size_t i = 0; i < outputTracks.size(); i++) { 4677 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4678 // for delivery downstream as needed. This in-place conversion is safe as 4679 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4680 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4681 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4682 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4683 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4684 } 4685 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4686 } 4687 mStandby = false; 4688 return (ssize_t)mSinkBufferSize; 4689} 4690 4691void AudioFlinger::DuplicatingThread::threadLoop_standby() 4692{ 4693 // DuplicatingThread implements standby by stopping all tracks 4694 for (size_t i = 0; i < outputTracks.size(); i++) { 4695 outputTracks[i]->stop(); 4696 } 4697} 4698 4699void AudioFlinger::DuplicatingThread::saveOutputTracks() 4700{ 4701 outputTracks = mOutputTracks; 4702} 4703 4704void AudioFlinger::DuplicatingThread::clearOutputTracks() 4705{ 4706 outputTracks.clear(); 4707} 4708 4709void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4710{ 4711 Mutex::Autolock _l(mLock); 4712 // FIXME explain this formula 4713 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4714 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4715 // due to current usage case and restrictions on the AudioBufferProvider. 4716 // Actual buffer conversion is done in threadLoop_write(). 4717 // 4718 // TODO: This may change in the future, depending on multichannel 4719 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4720 OutputTrack *outputTrack = new OutputTrack(thread, 4721 this, 4722 mSampleRate, 4723 AUDIO_FORMAT_PCM_16_BIT, 4724 mChannelMask, 4725 frameCount, 4726 IPCThreadState::self()->getCallingUid()); 4727 if (outputTrack->cblk() != NULL) { 4728 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4729 mOutputTracks.add(outputTrack); 4730 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4731 updateWaitTime_l(); 4732 } 4733} 4734 4735void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4736{ 4737 Mutex::Autolock _l(mLock); 4738 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4739 if (mOutputTracks[i]->thread() == thread) { 4740 mOutputTracks[i]->destroy(); 4741 mOutputTracks.removeAt(i); 4742 updateWaitTime_l(); 4743 return; 4744 } 4745 } 4746 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4747} 4748 4749// caller must hold mLock 4750void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4751{ 4752 mWaitTimeMs = UINT_MAX; 4753 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4754 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4755 if (strong != 0) { 4756 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4757 if (waitTimeMs < mWaitTimeMs) { 4758 mWaitTimeMs = waitTimeMs; 4759 } 4760 } 4761 } 4762} 4763 4764 4765bool AudioFlinger::DuplicatingThread::outputsReady( 4766 const SortedVector< sp<OutputTrack> > &outputTracks) 4767{ 4768 for (size_t i = 0; i < outputTracks.size(); i++) { 4769 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4770 if (thread == 0) { 4771 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4772 outputTracks[i].get()); 4773 return false; 4774 } 4775 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4776 // see note at standby() declaration 4777 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4778 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4779 thread.get()); 4780 return false; 4781 } 4782 } 4783 return true; 4784} 4785 4786uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4787{ 4788 return (mWaitTimeMs * 1000) / 2; 4789} 4790 4791void AudioFlinger::DuplicatingThread::cacheParameters_l() 4792{ 4793 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4794 updateWaitTime_l(); 4795 4796 MixerThread::cacheParameters_l(); 4797} 4798 4799// ---------------------------------------------------------------------------- 4800// Record 4801// ---------------------------------------------------------------------------- 4802 4803AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4804 AudioStreamIn *input, 4805 audio_io_handle_t id, 4806 audio_devices_t outDevice, 4807 audio_devices_t inDevice 4808#ifdef TEE_SINK 4809 , const sp<NBAIO_Sink>& teeSink 4810#endif 4811 ) : 4812 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4813 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4814 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4815 mRsmpInRear(0) 4816#ifdef TEE_SINK 4817 , mTeeSink(teeSink) 4818#endif 4819 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4820 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4821 // mFastCapture below 4822 , mFastCaptureFutex(0) 4823 // mInputSource 4824 // mPipeSink 4825 // mPipeSource 4826 , mPipeFramesP2(0) 4827 // mPipeMemory 4828 // mFastCaptureNBLogWriter 4829 , mFastTrackAvail(false) 4830{ 4831 snprintf(mName, kNameLength, "AudioIn_%X", id); 4832 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4833 4834 readInputParameters_l(); 4835 4836 // create an NBAIO source for the HAL input stream, and negotiate 4837 mInputSource = new AudioStreamInSource(input->stream); 4838 size_t numCounterOffers = 0; 4839 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4840 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4841 ALOG_ASSERT(index == 0); 4842 4843 // initialize fast capture depending on configuration 4844 bool initFastCapture; 4845 switch (kUseFastCapture) { 4846 case FastCapture_Never: 4847 initFastCapture = false; 4848 break; 4849 case FastCapture_Always: 4850 initFastCapture = true; 4851 break; 4852 case FastCapture_Static: 4853 uint32_t primaryOutputSampleRate; 4854 { 4855 AutoMutex _l(audioFlinger->mHardwareLock); 4856 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4857 } 4858 initFastCapture = 4859 // either capture sample rate is same as (a reasonable) primary output sample rate 4860 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4861 (mSampleRate == primaryOutputSampleRate)) || 4862 // or primary output sample rate is unknown, and capture sample rate is reasonable 4863 ((primaryOutputSampleRate == 0) && 4864 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4865 // and the buffer size is < 12 ms 4866 (mFrameCount * 1000) / mSampleRate < 12; 4867 break; 4868 // case FastCapture_Dynamic: 4869 } 4870 4871 if (initFastCapture) { 4872 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4873 NBAIO_Format format = mInputSource->format(); 4874 size_t pipeFramesP2 = roundup(mFrameCount * 8); 4875 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4876 void *pipeBuffer; 4877 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4878 sp<IMemory> pipeMemory; 4879 if ((roHeap == 0) || 4880 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4881 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4882 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4883 goto failed; 4884 } 4885 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4886 memset(pipeBuffer, 0, pipeSize); 4887 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4888 const NBAIO_Format offers[1] = {format}; 4889 size_t numCounterOffers = 0; 4890 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4891 ALOG_ASSERT(index == 0); 4892 mPipeSink = pipe; 4893 PipeReader *pipeReader = new PipeReader(*pipe); 4894 numCounterOffers = 0; 4895 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4896 ALOG_ASSERT(index == 0); 4897 mPipeSource = pipeReader; 4898 mPipeFramesP2 = pipeFramesP2; 4899 mPipeMemory = pipeMemory; 4900 4901 // create fast capture 4902 mFastCapture = new FastCapture(); 4903 FastCaptureStateQueue *sq = mFastCapture->sq(); 4904#ifdef STATE_QUEUE_DUMP 4905 // FIXME 4906#endif 4907 FastCaptureState *state = sq->begin(); 4908 state->mCblk = NULL; 4909 state->mInputSource = mInputSource.get(); 4910 state->mInputSourceGen++; 4911 state->mPipeSink = pipe; 4912 state->mPipeSinkGen++; 4913 state->mFrameCount = mFrameCount; 4914 state->mCommand = FastCaptureState::COLD_IDLE; 4915 // already done in constructor initialization list 4916 //mFastCaptureFutex = 0; 4917 state->mColdFutexAddr = &mFastCaptureFutex; 4918 state->mColdGen++; 4919 state->mDumpState = &mFastCaptureDumpState; 4920#ifdef TEE_SINK 4921 // FIXME 4922#endif 4923 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4924 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4925 sq->end(); 4926 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4927 4928 // start the fast capture 4929 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4930 pid_t tid = mFastCapture->getTid(); 4931 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4932 if (err != 0) { 4933 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4934 kPriorityFastCapture, getpid_cached, tid, err); 4935 } 4936 4937#ifdef AUDIO_WATCHDOG 4938 // FIXME 4939#endif 4940 4941 mFastTrackAvail = true; 4942 } 4943failed: ; 4944 4945 // FIXME mNormalSource 4946} 4947 4948 4949AudioFlinger::RecordThread::~RecordThread() 4950{ 4951 if (mFastCapture != 0) { 4952 FastCaptureStateQueue *sq = mFastCapture->sq(); 4953 FastCaptureState *state = sq->begin(); 4954 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4955 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4956 if (old == -1) { 4957 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4958 } 4959 } 4960 state->mCommand = FastCaptureState::EXIT; 4961 sq->end(); 4962 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4963 mFastCapture->join(); 4964 mFastCapture.clear(); 4965 } 4966 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4967 mAudioFlinger->unregisterWriter(mNBLogWriter); 4968 delete[] mRsmpInBuffer; 4969} 4970 4971void AudioFlinger::RecordThread::onFirstRef() 4972{ 4973 run(mName, PRIORITY_URGENT_AUDIO); 4974} 4975 4976bool AudioFlinger::RecordThread::threadLoop() 4977{ 4978 nsecs_t lastWarning = 0; 4979 4980 inputStandBy(); 4981 4982reacquire_wakelock: 4983 sp<RecordTrack> activeTrack; 4984 int activeTracksGen; 4985 { 4986 Mutex::Autolock _l(mLock); 4987 size_t size = mActiveTracks.size(); 4988 activeTracksGen = mActiveTracksGen; 4989 if (size > 0) { 4990 // FIXME an arbitrary choice 4991 activeTrack = mActiveTracks[0]; 4992 acquireWakeLock_l(activeTrack->uid()); 4993 if (size > 1) { 4994 SortedVector<int> tmp; 4995 for (size_t i = 0; i < size; i++) { 4996 tmp.add(mActiveTracks[i]->uid()); 4997 } 4998 updateWakeLockUids_l(tmp); 4999 } 5000 } else { 5001 acquireWakeLock_l(-1); 5002 } 5003 } 5004 5005 // used to request a deferred sleep, to be executed later while mutex is unlocked 5006 uint32_t sleepUs = 0; 5007 5008 // loop while there is work to do 5009 for (;;) { 5010 Vector< sp<EffectChain> > effectChains; 5011 5012 // sleep with mutex unlocked 5013 if (sleepUs > 0) { 5014 usleep(sleepUs); 5015 sleepUs = 0; 5016 } 5017 5018 // activeTracks accumulates a copy of a subset of mActiveTracks 5019 Vector< sp<RecordTrack> > activeTracks; 5020 5021 // reference to the (first and only) fast track 5022 sp<RecordTrack> fastTrack; 5023 5024 { // scope for mLock 5025 Mutex::Autolock _l(mLock); 5026 5027 processConfigEvents_l(); 5028 5029 // check exitPending here because checkForNewParameters_l() and 5030 // checkForNewParameters_l() can temporarily release mLock 5031 if (exitPending()) { 5032 break; 5033 } 5034 5035 // if no active track(s), then standby and release wakelock 5036 size_t size = mActiveTracks.size(); 5037 if (size == 0) { 5038 standbyIfNotAlreadyInStandby(); 5039 // exitPending() can't become true here 5040 releaseWakeLock_l(); 5041 ALOGV("RecordThread: loop stopping"); 5042 // go to sleep 5043 mWaitWorkCV.wait(mLock); 5044 ALOGV("RecordThread: loop starting"); 5045 goto reacquire_wakelock; 5046 } 5047 5048 if (mActiveTracksGen != activeTracksGen) { 5049 activeTracksGen = mActiveTracksGen; 5050 SortedVector<int> tmp; 5051 for (size_t i = 0; i < size; i++) { 5052 tmp.add(mActiveTracks[i]->uid()); 5053 } 5054 updateWakeLockUids_l(tmp); 5055 } 5056 5057 bool doBroadcast = false; 5058 for (size_t i = 0; i < size; ) { 5059 5060 activeTrack = mActiveTracks[i]; 5061 if (activeTrack->isTerminated()) { 5062 removeTrack_l(activeTrack); 5063 mActiveTracks.remove(activeTrack); 5064 mActiveTracksGen++; 5065 size--; 5066 continue; 5067 } 5068 5069 TrackBase::track_state activeTrackState = activeTrack->mState; 5070 switch (activeTrackState) { 5071 5072 case TrackBase::PAUSING: 5073 mActiveTracks.remove(activeTrack); 5074 mActiveTracksGen++; 5075 doBroadcast = true; 5076 size--; 5077 continue; 5078 5079 case TrackBase::STARTING_1: 5080 sleepUs = 10000; 5081 i++; 5082 continue; 5083 5084 case TrackBase::STARTING_2: 5085 doBroadcast = true; 5086 mStandby = false; 5087 activeTrack->mState = TrackBase::ACTIVE; 5088 break; 5089 5090 case TrackBase::ACTIVE: 5091 break; 5092 5093 case TrackBase::IDLE: 5094 i++; 5095 continue; 5096 5097 default: 5098 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5099 } 5100 5101 activeTracks.add(activeTrack); 5102 i++; 5103 5104 if (activeTrack->isFastTrack()) { 5105 ALOG_ASSERT(!mFastTrackAvail); 5106 ALOG_ASSERT(fastTrack == 0); 5107 fastTrack = activeTrack; 5108 } 5109 } 5110 if (doBroadcast) { 5111 mStartStopCond.broadcast(); 5112 } 5113 5114 // sleep if there are no active tracks to process 5115 if (activeTracks.size() == 0) { 5116 if (sleepUs == 0) { 5117 sleepUs = kRecordThreadSleepUs; 5118 } 5119 continue; 5120 } 5121 sleepUs = 0; 5122 5123 lockEffectChains_l(effectChains); 5124 } 5125 5126 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5127 5128 size_t size = effectChains.size(); 5129 for (size_t i = 0; i < size; i++) { 5130 // thread mutex is not locked, but effect chain is locked 5131 effectChains[i]->process_l(); 5132 } 5133 5134 // Start the fast capture if it's not already running 5135 if (mFastCapture != 0) { 5136 FastCaptureStateQueue *sq = mFastCapture->sq(); 5137 FastCaptureState *state = sq->begin(); 5138 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5139 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5140 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5141 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5142 if (old == -1) { 5143 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5144 } 5145 } 5146 state->mCommand = FastCaptureState::READ_WRITE; 5147#if 0 // FIXME 5148 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5149 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5150#endif 5151 state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL; 5152 sq->end(); 5153 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5154#if 0 5155 if (kUseFastCapture == FastCapture_Dynamic) { 5156 mNormalSource = mPipeSource; 5157 } 5158#endif 5159 } else { 5160 sq->end(false /*didModify*/); 5161 } 5162 } 5163 5164 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5165 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5166 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5167 // If destination is non-contiguous, first read past the nominal end of buffer, then 5168 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5169 5170 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5171 ssize_t framesRead; 5172 5173 // If an NBAIO source is present, use it to read the normal capture's data 5174 if (mPipeSource != 0) { 5175 size_t framesToRead = mBufferSize / mFrameSize; 5176 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5177 framesToRead, AudioBufferProvider::kInvalidPTS); 5178 if (framesRead == 0) { 5179 // since pipe is non-blocking, simulate blocking input 5180 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5181 } 5182 // otherwise use the HAL / AudioStreamIn directly 5183 } else { 5184 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5185 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5186 if (bytesRead < 0) { 5187 framesRead = bytesRead; 5188 } else { 5189 framesRead = bytesRead / mFrameSize; 5190 } 5191 } 5192 5193 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5194 ALOGE("read failed: framesRead=%d", framesRead); 5195 // Force input into standby so that it tries to recover at next read attempt 5196 inputStandBy(); 5197 sleepUs = kRecordThreadSleepUs; 5198 } 5199 if (framesRead <= 0) { 5200 goto unlock; 5201 } 5202 ALOG_ASSERT(framesRead > 0); 5203 5204 if (mTeeSink != 0) { 5205 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5206 } 5207 // If destination is non-contiguous, we now correct for reading past end of buffer. 5208 { 5209 size_t part1 = mRsmpInFramesP2 - rear; 5210 if ((size_t) framesRead > part1) { 5211 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5212 (framesRead - part1) * mFrameSize); 5213 } 5214 } 5215 rear = mRsmpInRear += framesRead; 5216 5217 size = activeTracks.size(); 5218 // loop over each active track 5219 for (size_t i = 0; i < size; i++) { 5220 activeTrack = activeTracks[i]; 5221 5222 // skip fast tracks, as those are handled directly by FastCapture 5223 if (activeTrack->isFastTrack()) { 5224 continue; 5225 } 5226 5227 enum { 5228 OVERRUN_UNKNOWN, 5229 OVERRUN_TRUE, 5230 OVERRUN_FALSE 5231 } overrun = OVERRUN_UNKNOWN; 5232 5233 // loop over getNextBuffer to handle circular sink 5234 for (;;) { 5235 5236 activeTrack->mSink.frameCount = ~0; 5237 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5238 size_t framesOut = activeTrack->mSink.frameCount; 5239 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5240 5241 int32_t front = activeTrack->mRsmpInFront; 5242 ssize_t filled = rear - front; 5243 size_t framesIn; 5244 5245 if (filled < 0) { 5246 // should not happen, but treat like a massive overrun and re-sync 5247 framesIn = 0; 5248 activeTrack->mRsmpInFront = rear; 5249 overrun = OVERRUN_TRUE; 5250 } else if ((size_t) filled <= mRsmpInFrames) { 5251 framesIn = (size_t) filled; 5252 } else { 5253 // client is not keeping up with server, but give it latest data 5254 framesIn = mRsmpInFrames; 5255 activeTrack->mRsmpInFront = front = rear - framesIn; 5256 overrun = OVERRUN_TRUE; 5257 } 5258 5259 if (framesOut == 0 || framesIn == 0) { 5260 break; 5261 } 5262 5263 if (activeTrack->mResampler == NULL) { 5264 // no resampling 5265 if (framesIn > framesOut) { 5266 framesIn = framesOut; 5267 } else { 5268 framesOut = framesIn; 5269 } 5270 int8_t *dst = activeTrack->mSink.i8; 5271 while (framesIn > 0) { 5272 front &= mRsmpInFramesP2 - 1; 5273 size_t part1 = mRsmpInFramesP2 - front; 5274 if (part1 > framesIn) { 5275 part1 = framesIn; 5276 } 5277 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5278 if (mChannelCount == activeTrack->mChannelCount) { 5279 memcpy(dst, src, part1 * mFrameSize); 5280 } else if (mChannelCount == 1) { 5281 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5282 part1); 5283 } else { 5284 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5285 part1); 5286 } 5287 dst += part1 * activeTrack->mFrameSize; 5288 front += part1; 5289 framesIn -= part1; 5290 } 5291 activeTrack->mRsmpInFront += framesOut; 5292 5293 } else { 5294 // resampling 5295 // FIXME framesInNeeded should really be part of resampler API, and should 5296 // depend on the SRC ratio 5297 // to keep mRsmpInBuffer full so resampler always has sufficient input 5298 size_t framesInNeeded; 5299 // FIXME only re-calculate when it changes, and optimize for common ratios 5300 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 5301 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 5302 framesInNeeded = ceil(framesOut * inOverOut) + 1; 5303 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5304 framesInNeeded, framesOut, inOverOut); 5305 // Although we theoretically have framesIn in circular buffer, some of those are 5306 // unreleased frames, and thus must be discounted for purpose of budgeting. 5307 size_t unreleased = activeTrack->mRsmpInUnrel; 5308 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5309 if (framesIn < framesInNeeded) { 5310 ALOGV("not enough to resample: have %u frames in but need %u in to " 5311 "produce %u out given in/out ratio of %.4g", 5312 framesIn, framesInNeeded, framesOut, inOverOut); 5313 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 5314 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5315 if (newFramesOut == 0) { 5316 break; 5317 } 5318 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 5319 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5320 framesInNeeded, newFramesOut, outOverIn); 5321 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5322 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5323 "given in/out ratio of %.4g", 5324 framesIn, framesInNeeded, newFramesOut, inOverOut); 5325 framesOut = newFramesOut; 5326 } else { 5327 ALOGV("success 1: have %u in and need %u in to produce %u out " 5328 "given in/out ratio of %.4g", 5329 framesIn, framesInNeeded, framesOut, inOverOut); 5330 } 5331 5332 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5333 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5334 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5335 delete[] activeTrack->mRsmpOutBuffer; 5336 // resampler always outputs stereo 5337 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5338 activeTrack->mRsmpOutFrameCount = framesOut; 5339 } 5340 5341 // resampler accumulates, but we only have one source track 5342 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5343 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5344 // FIXME how about having activeTrack implement this interface itself? 5345 activeTrack->mResamplerBufferProvider 5346 /*this*/ /* AudioBufferProvider* */); 5347 // ditherAndClamp() works as long as all buffers returned by 5348 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5349 if (activeTrack->mChannelCount == 1) { 5350 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5351 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5352 framesOut); 5353 // the resampler always outputs stereo samples: 5354 // do post stereo to mono conversion 5355 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5356 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5357 } else { 5358 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5359 activeTrack->mRsmpOutBuffer, framesOut); 5360 } 5361 // now done with mRsmpOutBuffer 5362 5363 } 5364 5365 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5366 overrun = OVERRUN_FALSE; 5367 } 5368 5369 if (activeTrack->mFramesToDrop == 0) { 5370 if (framesOut > 0) { 5371 activeTrack->mSink.frameCount = framesOut; 5372 activeTrack->releaseBuffer(&activeTrack->mSink); 5373 } 5374 } else { 5375 // FIXME could do a partial drop of framesOut 5376 if (activeTrack->mFramesToDrop > 0) { 5377 activeTrack->mFramesToDrop -= framesOut; 5378 if (activeTrack->mFramesToDrop <= 0) { 5379 activeTrack->clearSyncStartEvent(); 5380 } 5381 } else { 5382 activeTrack->mFramesToDrop += framesOut; 5383 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5384 activeTrack->mSyncStartEvent->isCancelled()) { 5385 ALOGW("Synced record %s, session %d, trigger session %d", 5386 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5387 activeTrack->sessionId(), 5388 (activeTrack->mSyncStartEvent != 0) ? 5389 activeTrack->mSyncStartEvent->triggerSession() : 0); 5390 activeTrack->clearSyncStartEvent(); 5391 } 5392 } 5393 } 5394 5395 if (framesOut == 0) { 5396 break; 5397 } 5398 } 5399 5400 switch (overrun) { 5401 case OVERRUN_TRUE: 5402 // client isn't retrieving buffers fast enough 5403 if (!activeTrack->setOverflow()) { 5404 nsecs_t now = systemTime(); 5405 // FIXME should lastWarning per track? 5406 if ((now - lastWarning) > kWarningThrottleNs) { 5407 ALOGW("RecordThread: buffer overflow"); 5408 lastWarning = now; 5409 } 5410 } 5411 break; 5412 case OVERRUN_FALSE: 5413 activeTrack->clearOverflow(); 5414 break; 5415 case OVERRUN_UNKNOWN: 5416 break; 5417 } 5418 5419 } 5420 5421unlock: 5422 // enable changes in effect chain 5423 unlockEffectChains(effectChains); 5424 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5425 } 5426 5427 standbyIfNotAlreadyInStandby(); 5428 5429 { 5430 Mutex::Autolock _l(mLock); 5431 for (size_t i = 0; i < mTracks.size(); i++) { 5432 sp<RecordTrack> track = mTracks[i]; 5433 track->invalidate(); 5434 } 5435 mActiveTracks.clear(); 5436 mActiveTracksGen++; 5437 mStartStopCond.broadcast(); 5438 } 5439 5440 releaseWakeLock(); 5441 5442 ALOGV("RecordThread %p exiting", this); 5443 return false; 5444} 5445 5446void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5447{ 5448 if (!mStandby) { 5449 inputStandBy(); 5450 mStandby = true; 5451 } 5452} 5453 5454void AudioFlinger::RecordThread::inputStandBy() 5455{ 5456 // Idle the fast capture if it's currently running 5457 if (mFastCapture != 0) { 5458 FastCaptureStateQueue *sq = mFastCapture->sq(); 5459 FastCaptureState *state = sq->begin(); 5460 if (!(state->mCommand & FastCaptureState::IDLE)) { 5461 state->mCommand = FastCaptureState::COLD_IDLE; 5462 state->mColdFutexAddr = &mFastCaptureFutex; 5463 state->mColdGen++; 5464 mFastCaptureFutex = 0; 5465 sq->end(); 5466 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5467 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5468#if 0 5469 if (kUseFastCapture == FastCapture_Dynamic) { 5470 // FIXME 5471 } 5472#endif 5473#ifdef AUDIO_WATCHDOG 5474 // FIXME 5475#endif 5476 } else { 5477 sq->end(false /*didModify*/); 5478 } 5479 } 5480 mInput->stream->common.standby(&mInput->stream->common); 5481} 5482 5483// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5484sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5485 const sp<AudioFlinger::Client>& client, 5486 uint32_t sampleRate, 5487 audio_format_t format, 5488 audio_channel_mask_t channelMask, 5489 size_t *pFrameCount, 5490 int sessionId, 5491 size_t *notificationFrames, 5492 int uid, 5493 IAudioFlinger::track_flags_t *flags, 5494 pid_t tid, 5495 status_t *status) 5496{ 5497 size_t frameCount = *pFrameCount; 5498 sp<RecordTrack> track; 5499 status_t lStatus; 5500 5501 // client expresses a preference for FAST, but we get the final say 5502 if (*flags & IAudioFlinger::TRACK_FAST) { 5503 if ( 5504 // use case: callback handler 5505 (tid != -1) && 5506 // frame count is not specified, or is exactly the pipe depth 5507 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5508 // PCM data 5509 audio_is_linear_pcm(format) && 5510 // native format 5511 (format == mFormat) && 5512 // native channel mask 5513 (channelMask == mChannelMask) && 5514 // native hardware sample rate 5515 (sampleRate == mSampleRate) && 5516 // record thread has an associated fast capture 5517 hasFastCapture() && 5518 // there are sufficient fast track slots available 5519 mFastTrackAvail 5520 ) { 5521 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5522 frameCount, mFrameCount); 5523 } else { 5524 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5525 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5526 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5527 frameCount, mFrameCount, mPipeFramesP2, 5528 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5529 hasFastCapture(), tid, mFastTrackAvail); 5530 *flags &= ~IAudioFlinger::TRACK_FAST; 5531 } 5532 } 5533 5534 // compute track buffer size in frames, and suggest the notification frame count 5535 if (*flags & IAudioFlinger::TRACK_FAST) { 5536 // fast track: frame count is exactly the pipe depth 5537 frameCount = mPipeFramesP2; 5538 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5539 *notificationFrames = mFrameCount; 5540 } else { 5541 // not fast track: frame count is at least 2 HAL buffers and at least 20 ms 5542 size_t minFrameCount = ((int64_t) mFrameCount * 2 * sampleRate + mSampleRate - 1) / 5543 mSampleRate; 5544 if (frameCount < minFrameCount) { 5545 frameCount = minFrameCount; 5546 } 5547 minFrameCount = (sampleRate * 20 / 1000 + 1) & ~1; 5548 if (frameCount < minFrameCount) { 5549 frameCount = minFrameCount; 5550 } 5551 // notification is forced to be at least double-buffering 5552 size_t maxNotification = frameCount / 2; 5553 if (*notificationFrames == 0 || *notificationFrames > maxNotification) { 5554 *notificationFrames = maxNotification; 5555 } 5556 } 5557 *pFrameCount = frameCount; 5558 5559 lStatus = initCheck(); 5560 if (lStatus != NO_ERROR) { 5561 ALOGE("createRecordTrack_l() audio driver not initialized"); 5562 goto Exit; 5563 } 5564 5565 { // scope for mLock 5566 Mutex::Autolock _l(mLock); 5567 5568 track = new RecordTrack(this, client, sampleRate, 5569 format, channelMask, frameCount, NULL, sessionId, uid, 5570 *flags, TrackBase::TYPE_DEFAULT); 5571 5572 lStatus = track->initCheck(); 5573 if (lStatus != NO_ERROR) { 5574 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5575 // track must be cleared from the caller as the caller has the AF lock 5576 goto Exit; 5577 } 5578 mTracks.add(track); 5579 5580 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5581 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5582 mAudioFlinger->btNrecIsOff(); 5583 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5584 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5585 5586 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5587 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5588 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5589 // so ask activity manager to do this on our behalf 5590 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5591 } 5592 } 5593 5594 lStatus = NO_ERROR; 5595 5596Exit: 5597 *status = lStatus; 5598 return track; 5599} 5600 5601status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5602 AudioSystem::sync_event_t event, 5603 int triggerSession) 5604{ 5605 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5606 sp<ThreadBase> strongMe = this; 5607 status_t status = NO_ERROR; 5608 5609 if (event == AudioSystem::SYNC_EVENT_NONE) { 5610 recordTrack->clearSyncStartEvent(); 5611 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5612 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5613 triggerSession, 5614 recordTrack->sessionId(), 5615 syncStartEventCallback, 5616 recordTrack); 5617 // Sync event can be cancelled by the trigger session if the track is not in a 5618 // compatible state in which case we start record immediately 5619 if (recordTrack->mSyncStartEvent->isCancelled()) { 5620 recordTrack->clearSyncStartEvent(); 5621 } else { 5622 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5623 recordTrack->mFramesToDrop = - 5624 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5625 } 5626 } 5627 5628 { 5629 // This section is a rendezvous between binder thread executing start() and RecordThread 5630 AutoMutex lock(mLock); 5631 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5632 if (recordTrack->mState == TrackBase::PAUSING) { 5633 ALOGV("active record track PAUSING -> ACTIVE"); 5634 recordTrack->mState = TrackBase::ACTIVE; 5635 } else { 5636 ALOGV("active record track state %d", recordTrack->mState); 5637 } 5638 return status; 5639 } 5640 5641 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5642 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5643 // or using a separate command thread 5644 recordTrack->mState = TrackBase::STARTING_1; 5645 mActiveTracks.add(recordTrack); 5646 mActiveTracksGen++; 5647 status_t status = NO_ERROR; 5648 if (recordTrack->isExternalTrack()) { 5649 mLock.unlock(); 5650 status = AudioSystem::startInput(mId); 5651 mLock.lock(); 5652 // FIXME should verify that recordTrack is still in mActiveTracks 5653 if (status != NO_ERROR) { 5654 mActiveTracks.remove(recordTrack); 5655 mActiveTracksGen++; 5656 recordTrack->clearSyncStartEvent(); 5657 ALOGV("RecordThread::start error %d", status); 5658 return status; 5659 } 5660 } 5661 // Catch up with current buffer indices if thread is already running. 5662 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5663 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5664 // see previously buffered data before it called start(), but with greater risk of overrun. 5665 5666 recordTrack->mRsmpInFront = mRsmpInRear; 5667 recordTrack->mRsmpInUnrel = 0; 5668 // FIXME why reset? 5669 if (recordTrack->mResampler != NULL) { 5670 recordTrack->mResampler->reset(); 5671 } 5672 recordTrack->mState = TrackBase::STARTING_2; 5673 // signal thread to start 5674 mWaitWorkCV.broadcast(); 5675 if (mActiveTracks.indexOf(recordTrack) < 0) { 5676 ALOGV("Record failed to start"); 5677 status = BAD_VALUE; 5678 goto startError; 5679 } 5680 return status; 5681 } 5682 5683startError: 5684 if (recordTrack->isExternalTrack()) { 5685 AudioSystem::stopInput(mId); 5686 } 5687 recordTrack->clearSyncStartEvent(); 5688 // FIXME I wonder why we do not reset the state here? 5689 return status; 5690} 5691 5692void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5693{ 5694 sp<SyncEvent> strongEvent = event.promote(); 5695 5696 if (strongEvent != 0) { 5697 sp<RefBase> ptr = strongEvent->cookie().promote(); 5698 if (ptr != 0) { 5699 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5700 recordTrack->handleSyncStartEvent(strongEvent); 5701 } 5702 } 5703} 5704 5705bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5706 ALOGV("RecordThread::stop"); 5707 AutoMutex _l(mLock); 5708 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5709 return false; 5710 } 5711 // note that threadLoop may still be processing the track at this point [without lock] 5712 recordTrack->mState = TrackBase::PAUSING; 5713 // do not wait for mStartStopCond if exiting 5714 if (exitPending()) { 5715 return true; 5716 } 5717 // FIXME incorrect usage of wait: no explicit predicate or loop 5718 mStartStopCond.wait(mLock); 5719 // if we have been restarted, recordTrack is in mActiveTracks here 5720 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5721 ALOGV("Record stopped OK"); 5722 return true; 5723 } 5724 return false; 5725} 5726 5727bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5728{ 5729 return false; 5730} 5731 5732status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5733{ 5734#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5735 if (!isValidSyncEvent(event)) { 5736 return BAD_VALUE; 5737 } 5738 5739 int eventSession = event->triggerSession(); 5740 status_t ret = NAME_NOT_FOUND; 5741 5742 Mutex::Autolock _l(mLock); 5743 5744 for (size_t i = 0; i < mTracks.size(); i++) { 5745 sp<RecordTrack> track = mTracks[i]; 5746 if (eventSession == track->sessionId()) { 5747 (void) track->setSyncEvent(event); 5748 ret = NO_ERROR; 5749 } 5750 } 5751 return ret; 5752#else 5753 return BAD_VALUE; 5754#endif 5755} 5756 5757// destroyTrack_l() must be called with ThreadBase::mLock held 5758void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5759{ 5760 track->terminate(); 5761 track->mState = TrackBase::STOPPED; 5762 // active tracks are removed by threadLoop() 5763 if (mActiveTracks.indexOf(track) < 0) { 5764 removeTrack_l(track); 5765 } 5766} 5767 5768void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5769{ 5770 mTracks.remove(track); 5771 // need anything related to effects here? 5772 if (track->isFastTrack()) { 5773 ALOG_ASSERT(!mFastTrackAvail); 5774 mFastTrackAvail = true; 5775 } 5776} 5777 5778void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5779{ 5780 dumpInternals(fd, args); 5781 dumpTracks(fd, args); 5782 dumpEffectChains(fd, args); 5783} 5784 5785void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5786{ 5787 dprintf(fd, "\nInput thread %p:\n", this); 5788 5789 if (mActiveTracks.size() > 0) { 5790 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5791 } else { 5792 dprintf(fd, " No active record clients\n"); 5793 } 5794 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5795 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5796 5797 dumpBase(fd, args); 5798} 5799 5800void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5801{ 5802 const size_t SIZE = 256; 5803 char buffer[SIZE]; 5804 String8 result; 5805 5806 size_t numtracks = mTracks.size(); 5807 size_t numactive = mActiveTracks.size(); 5808 size_t numactiveseen = 0; 5809 dprintf(fd, " %d Tracks", numtracks); 5810 if (numtracks) { 5811 dprintf(fd, " of which %d are active\n", numactive); 5812 RecordTrack::appendDumpHeader(result); 5813 for (size_t i = 0; i < numtracks ; ++i) { 5814 sp<RecordTrack> track = mTracks[i]; 5815 if (track != 0) { 5816 bool active = mActiveTracks.indexOf(track) >= 0; 5817 if (active) { 5818 numactiveseen++; 5819 } 5820 track->dump(buffer, SIZE, active); 5821 result.append(buffer); 5822 } 5823 } 5824 } else { 5825 dprintf(fd, "\n"); 5826 } 5827 5828 if (numactiveseen != numactive) { 5829 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5830 " not in the track list\n"); 5831 result.append(buffer); 5832 RecordTrack::appendDumpHeader(result); 5833 for (size_t i = 0; i < numactive; ++i) { 5834 sp<RecordTrack> track = mActiveTracks[i]; 5835 if (mTracks.indexOf(track) < 0) { 5836 track->dump(buffer, SIZE, true); 5837 result.append(buffer); 5838 } 5839 } 5840 5841 } 5842 write(fd, result.string(), result.size()); 5843} 5844 5845// AudioBufferProvider interface 5846status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5847 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5848{ 5849 RecordTrack *activeTrack = mRecordTrack; 5850 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5851 if (threadBase == 0) { 5852 buffer->frameCount = 0; 5853 buffer->raw = NULL; 5854 return NOT_ENOUGH_DATA; 5855 } 5856 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5857 int32_t rear = recordThread->mRsmpInRear; 5858 int32_t front = activeTrack->mRsmpInFront; 5859 ssize_t filled = rear - front; 5860 // FIXME should not be P2 (don't want to increase latency) 5861 // FIXME if client not keeping up, discard 5862 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5863 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5864 front &= recordThread->mRsmpInFramesP2 - 1; 5865 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5866 if (part1 > (size_t) filled) { 5867 part1 = filled; 5868 } 5869 size_t ask = buffer->frameCount; 5870 ALOG_ASSERT(ask > 0); 5871 if (part1 > ask) { 5872 part1 = ask; 5873 } 5874 if (part1 == 0) { 5875 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5876 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5877 buffer->raw = NULL; 5878 buffer->frameCount = 0; 5879 activeTrack->mRsmpInUnrel = 0; 5880 return NOT_ENOUGH_DATA; 5881 } 5882 5883 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5884 buffer->frameCount = part1; 5885 activeTrack->mRsmpInUnrel = part1; 5886 return NO_ERROR; 5887} 5888 5889// AudioBufferProvider interface 5890void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5891 AudioBufferProvider::Buffer* buffer) 5892{ 5893 RecordTrack *activeTrack = mRecordTrack; 5894 size_t stepCount = buffer->frameCount; 5895 if (stepCount == 0) { 5896 return; 5897 } 5898 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5899 activeTrack->mRsmpInUnrel -= stepCount; 5900 activeTrack->mRsmpInFront += stepCount; 5901 buffer->raw = NULL; 5902 buffer->frameCount = 0; 5903} 5904 5905bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5906 status_t& status) 5907{ 5908 bool reconfig = false; 5909 5910 status = NO_ERROR; 5911 5912 audio_format_t reqFormat = mFormat; 5913 uint32_t samplingRate = mSampleRate; 5914 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5915 5916 AudioParameter param = AudioParameter(keyValuePair); 5917 int value; 5918 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5919 // channel count change can be requested. Do we mandate the first client defines the 5920 // HAL sampling rate and channel count or do we allow changes on the fly? 5921 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5922 samplingRate = value; 5923 reconfig = true; 5924 } 5925 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5926 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5927 status = BAD_VALUE; 5928 } else { 5929 reqFormat = (audio_format_t) value; 5930 reconfig = true; 5931 } 5932 } 5933 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5934 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5935 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5936 status = BAD_VALUE; 5937 } else { 5938 channelMask = mask; 5939 reconfig = true; 5940 } 5941 } 5942 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5943 // do not accept frame count changes if tracks are open as the track buffer 5944 // size depends on frame count and correct behavior would not be guaranteed 5945 // if frame count is changed after track creation 5946 if (mActiveTracks.size() > 0) { 5947 status = INVALID_OPERATION; 5948 } else { 5949 reconfig = true; 5950 } 5951 } 5952 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5953 // forward device change to effects that have requested to be 5954 // aware of attached audio device. 5955 for (size_t i = 0; i < mEffectChains.size(); i++) { 5956 mEffectChains[i]->setDevice_l(value); 5957 } 5958 5959 // store input device and output device but do not forward output device to audio HAL. 5960 // Note that status is ignored by the caller for output device 5961 // (see AudioFlinger::setParameters() 5962 if (audio_is_output_devices(value)) { 5963 mOutDevice = value; 5964 status = BAD_VALUE; 5965 } else { 5966 mInDevice = value; 5967 // disable AEC and NS if the device is a BT SCO headset supporting those 5968 // pre processings 5969 if (mTracks.size() > 0) { 5970 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5971 mAudioFlinger->btNrecIsOff(); 5972 for (size_t i = 0; i < mTracks.size(); i++) { 5973 sp<RecordTrack> track = mTracks[i]; 5974 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5975 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5976 } 5977 } 5978 } 5979 } 5980 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5981 mAudioSource != (audio_source_t)value) { 5982 // forward device change to effects that have requested to be 5983 // aware of attached audio device. 5984 for (size_t i = 0; i < mEffectChains.size(); i++) { 5985 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5986 } 5987 mAudioSource = (audio_source_t)value; 5988 } 5989 5990 if (status == NO_ERROR) { 5991 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5992 keyValuePair.string()); 5993 if (status == INVALID_OPERATION) { 5994 inputStandBy(); 5995 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5996 keyValuePair.string()); 5997 } 5998 if (reconfig) { 5999 if (status == BAD_VALUE && 6000 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6001 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6002 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6003 <= (2 * samplingRate)) && 6004 audio_channel_count_from_in_mask( 6005 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6006 (channelMask == AUDIO_CHANNEL_IN_MONO || 6007 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6008 status = NO_ERROR; 6009 } 6010 if (status == NO_ERROR) { 6011 readInputParameters_l(); 6012 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6013 } 6014 } 6015 } 6016 6017 return reconfig; 6018} 6019 6020String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6021{ 6022 Mutex::Autolock _l(mLock); 6023 if (initCheck() != NO_ERROR) { 6024 return String8(); 6025 } 6026 6027 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6028 const String8 out_s8(s); 6029 free(s); 6030 return out_s8; 6031} 6032 6033void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6034 AudioSystem::OutputDescriptor desc; 6035 const void *param2 = NULL; 6036 6037 switch (event) { 6038 case AudioSystem::INPUT_OPENED: 6039 case AudioSystem::INPUT_CONFIG_CHANGED: 6040 desc.channelMask = mChannelMask; 6041 desc.samplingRate = mSampleRate; 6042 desc.format = mFormat; 6043 desc.frameCount = mFrameCount; 6044 desc.latency = 0; 6045 param2 = &desc; 6046 break; 6047 6048 case AudioSystem::INPUT_CLOSED: 6049 default: 6050 break; 6051 } 6052 mAudioFlinger->audioConfigChanged(event, mId, param2); 6053} 6054 6055void AudioFlinger::RecordThread::readInputParameters_l() 6056{ 6057 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6058 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6059 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6060 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6061 mFormat = mHALFormat; 6062 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6063 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6064 } 6065 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6066 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6067 mFrameCount = mBufferSize / mFrameSize; 6068 // This is the formula for calculating the temporary buffer size. 6069 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6070 // 1 full output buffer, regardless of the alignment of the available input. 6071 // The value is somewhat arbitrary, and could probably be even larger. 6072 // A larger value should allow more old data to be read after a track calls start(), 6073 // without increasing latency. 6074 mRsmpInFrames = mFrameCount * 7; 6075 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6076 delete[] mRsmpInBuffer; 6077 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6078 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6079 6080 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6081 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6082} 6083 6084uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6085{ 6086 Mutex::Autolock _l(mLock); 6087 if (initCheck() != NO_ERROR) { 6088 return 0; 6089 } 6090 6091 return mInput->stream->get_input_frames_lost(mInput->stream); 6092} 6093 6094uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6095{ 6096 Mutex::Autolock _l(mLock); 6097 uint32_t result = 0; 6098 if (getEffectChain_l(sessionId) != 0) { 6099 result = EFFECT_SESSION; 6100 } 6101 6102 for (size_t i = 0; i < mTracks.size(); ++i) { 6103 if (sessionId == mTracks[i]->sessionId()) { 6104 result |= TRACK_SESSION; 6105 break; 6106 } 6107 } 6108 6109 return result; 6110} 6111 6112KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6113{ 6114 KeyedVector<int, bool> ids; 6115 Mutex::Autolock _l(mLock); 6116 for (size_t j = 0; j < mTracks.size(); ++j) { 6117 sp<RecordThread::RecordTrack> track = mTracks[j]; 6118 int sessionId = track->sessionId(); 6119 if (ids.indexOfKey(sessionId) < 0) { 6120 ids.add(sessionId, true); 6121 } 6122 } 6123 return ids; 6124} 6125 6126AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6127{ 6128 Mutex::Autolock _l(mLock); 6129 AudioStreamIn *input = mInput; 6130 mInput = NULL; 6131 return input; 6132} 6133 6134// this method must always be called either with ThreadBase mLock held or inside the thread loop 6135audio_stream_t* AudioFlinger::RecordThread::stream() const 6136{ 6137 if (mInput == NULL) { 6138 return NULL; 6139 } 6140 return &mInput->stream->common; 6141} 6142 6143status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6144{ 6145 // only one chain per input thread 6146 if (mEffectChains.size() != 0) { 6147 return INVALID_OPERATION; 6148 } 6149 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6150 6151 chain->setInBuffer(NULL); 6152 chain->setOutBuffer(NULL); 6153 6154 checkSuspendOnAddEffectChain_l(chain); 6155 6156 mEffectChains.add(chain); 6157 6158 return NO_ERROR; 6159} 6160 6161size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6162{ 6163 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6164 ALOGW_IF(mEffectChains.size() != 1, 6165 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6166 chain.get(), mEffectChains.size(), this); 6167 if (mEffectChains.size() == 1) { 6168 mEffectChains.removeAt(0); 6169 } 6170 return 0; 6171} 6172 6173status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6174 audio_patch_handle_t *handle) 6175{ 6176 status_t status = NO_ERROR; 6177 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6178 // store new device and send to effects 6179 mInDevice = patch->sources[0].ext.device.type; 6180 for (size_t i = 0; i < mEffectChains.size(); i++) { 6181 mEffectChains[i]->setDevice_l(mInDevice); 6182 } 6183 6184 // disable AEC and NS if the device is a BT SCO headset supporting those 6185 // pre processings 6186 if (mTracks.size() > 0) { 6187 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6188 mAudioFlinger->btNrecIsOff(); 6189 for (size_t i = 0; i < mTracks.size(); i++) { 6190 sp<RecordTrack> track = mTracks[i]; 6191 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6192 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6193 } 6194 } 6195 6196 // store new source and send to effects 6197 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6198 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6199 for (size_t i = 0; i < mEffectChains.size(); i++) { 6200 mEffectChains[i]->setAudioSource_l(mAudioSource); 6201 } 6202 } 6203 6204 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6205 status = hwDevice->create_audio_patch(hwDevice, 6206 patch->num_sources, 6207 patch->sources, 6208 patch->num_sinks, 6209 patch->sinks, 6210 handle); 6211 } else { 6212 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6213 } 6214 return status; 6215} 6216 6217status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6218{ 6219 status_t status = NO_ERROR; 6220 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6221 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6222 status = hwDevice->release_audio_patch(hwDevice, handle); 6223 } else { 6224 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6225 } 6226 return status; 6227} 6228 6229void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6230{ 6231 Mutex::Autolock _l(mLock); 6232 mTracks.add(record); 6233} 6234 6235void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6236{ 6237 Mutex::Autolock _l(mLock); 6238 destroyTrack_l(record); 6239} 6240 6241void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6242{ 6243 ThreadBase::getAudioPortConfig(config); 6244 config->role = AUDIO_PORT_ROLE_SINK; 6245 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6246 config->ext.mix.usecase.source = mAudioSource; 6247} 6248 6249}; // namespace android 6250