Threads.cpp revision c263ca0ad8b6bdf5b0693996bc5f2f5916e0cd49
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38#include <audio_utils/minifloat.h> 39 40// NBAIO implementations 41#include <media/nbaio/AudioStreamInSource.h> 42#include <media/nbaio/AudioStreamOutSink.h> 43#include <media/nbaio/MonoPipe.h> 44#include <media/nbaio/MonoPipeReader.h> 45#include <media/nbaio/Pipe.h> 46#include <media/nbaio/PipeReader.h> 47#include <media/nbaio/SourceAudioBufferProvider.h> 48 49#include <powermanager/PowerManager.h> 50 51#include <common_time/cc_helper.h> 52#include <common_time/local_clock.h> 53 54#include "AudioFlinger.h" 55#include "AudioMixer.h" 56#include "FastMixer.h" 57#include "FastCapture.h" 58#include "ServiceUtilities.h" 59#include "SchedulingPolicyService.h" 60 61#ifdef ADD_BATTERY_DATA 62#include <media/IMediaPlayerService.h> 63#include <media/IMediaDeathNotifier.h> 64#endif 65 66#ifdef DEBUG_CPU_USAGE 67#include <cpustats/CentralTendencyStatistics.h> 68#include <cpustats/ThreadCpuUsage.h> 69#endif 70 71// ---------------------------------------------------------------------------- 72 73// Note: the following macro is used for extremely verbose logging message. In 74// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 75// 0; but one side effect of this is to turn all LOGV's as well. Some messages 76// are so verbose that we want to suppress them even when we have ALOG_ASSERT 77// turned on. Do not uncomment the #def below unless you really know what you 78// are doing and want to see all of the extremely verbose messages. 79//#define VERY_VERY_VERBOSE_LOGGING 80#ifdef VERY_VERY_VERBOSE_LOGGING 81#define ALOGVV ALOGV 82#else 83#define ALOGVV(a...) do { } while(0) 84#endif 85 86namespace android { 87 88// retry counts for buffer fill timeout 89// 50 * ~20msecs = 1 second 90static const int8_t kMaxTrackRetries = 50; 91static const int8_t kMaxTrackStartupRetries = 50; 92// allow less retry attempts on direct output thread. 93// direct outputs can be a scarce resource in audio hardware and should 94// be released as quickly as possible. 95static const int8_t kMaxTrackRetriesDirect = 2; 96 97// don't warn about blocked writes or record buffer overflows more often than this 98static const nsecs_t kWarningThrottleNs = seconds(5); 99 100// RecordThread loop sleep time upon application overrun or audio HAL read error 101static const int kRecordThreadSleepUs = 5000; 102 103// maximum time to wait in sendConfigEvent_l() for a status to be received 104static const nsecs_t kConfigEventTimeoutNs = seconds(2); 105 106// minimum sleep time for the mixer thread loop when tracks are active but in underrun 107static const uint32_t kMinThreadSleepTimeUs = 5000; 108// maximum divider applied to the active sleep time in the mixer thread loop 109static const uint32_t kMaxThreadSleepTimeShift = 2; 110 111// minimum normal sink buffer size, expressed in milliseconds rather than frames 112static const uint32_t kMinNormalSinkBufferSizeMs = 20; 113// maximum normal sink buffer size 114static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 115 116// Offloaded output thread standby delay: allows track transition without going to standby 117static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 118 119// Whether to use fast mixer 120static const enum { 121 FastMixer_Never, // never initialize or use: for debugging only 122 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 123 // normal mixer multiplier is 1 124 FastMixer_Static, // initialize if needed, then use all the time if initialized, 125 // multiplier is calculated based on min & max normal mixer buffer size 126 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 127 // multiplier is calculated based on min & max normal mixer buffer size 128 // FIXME for FastMixer_Dynamic: 129 // Supporting this option will require fixing HALs that can't handle large writes. 130 // For example, one HAL implementation returns an error from a large write, 131 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 132 // We could either fix the HAL implementations, or provide a wrapper that breaks 133 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 134} kUseFastMixer = FastMixer_Static; 135 136// Whether to use fast capture 137static const enum { 138 FastCapture_Never, // never initialize or use: for debugging only 139 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 140 FastCapture_Static, // initialize if needed, then use all the time if initialized 141} kUseFastCapture = FastCapture_Static; 142 143// Priorities for requestPriority 144static const int kPriorityAudioApp = 2; 145static const int kPriorityFastMixer = 3; 146static const int kPriorityFastCapture = 3; 147 148// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 149// for the track. The client then sub-divides this into smaller buffers for its use. 150// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 151// So for now we just assume that client is double-buffered for fast tracks. 152// FIXME It would be better for client to tell AudioFlinger the value of N, 153// so AudioFlinger could allocate the right amount of memory. 154// See the client's minBufCount and mNotificationFramesAct calculations for details. 155 156// This is the default value, if not specified by property. 157static const int kFastTrackMultiplier = 2; 158 159// The minimum and maximum allowed values 160static const int kFastTrackMultiplierMin = 1; 161static const int kFastTrackMultiplierMax = 2; 162 163// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 164static int sFastTrackMultiplier = kFastTrackMultiplier; 165 166// See Thread::readOnlyHeap(). 167// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 168// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 169// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 170static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 171 172// ---------------------------------------------------------------------------- 173 174static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 175 176static void sFastTrackMultiplierInit() 177{ 178 char value[PROPERTY_VALUE_MAX]; 179 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 180 char *endptr; 181 unsigned long ul = strtoul(value, &endptr, 0); 182 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 183 sFastTrackMultiplier = (int) ul; 184 } 185 } 186} 187 188// ---------------------------------------------------------------------------- 189 190#ifdef ADD_BATTERY_DATA 191// To collect the amplifier usage 192static void addBatteryData(uint32_t params) { 193 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 194 if (service == NULL) { 195 // it already logged 196 return; 197 } 198 199 service->addBatteryData(params); 200} 201#endif 202 203 204// ---------------------------------------------------------------------------- 205// CPU Stats 206// ---------------------------------------------------------------------------- 207 208class CpuStats { 209public: 210 CpuStats(); 211 void sample(const String8 &title); 212#ifdef DEBUG_CPU_USAGE 213private: 214 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 215 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 216 217 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 218 219 int mCpuNum; // thread's current CPU number 220 int mCpukHz; // frequency of thread's current CPU in kHz 221#endif 222}; 223 224CpuStats::CpuStats() 225#ifdef DEBUG_CPU_USAGE 226 : mCpuNum(-1), mCpukHz(-1) 227#endif 228{ 229} 230 231void CpuStats::sample(const String8 &title 232#ifndef DEBUG_CPU_USAGE 233 __unused 234#endif 235 ) { 236#ifdef DEBUG_CPU_USAGE 237 // get current thread's delta CPU time in wall clock ns 238 double wcNs; 239 bool valid = mCpuUsage.sampleAndEnable(wcNs); 240 241 // record sample for wall clock statistics 242 if (valid) { 243 mWcStats.sample(wcNs); 244 } 245 246 // get the current CPU number 247 int cpuNum = sched_getcpu(); 248 249 // get the current CPU frequency in kHz 250 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 251 252 // check if either CPU number or frequency changed 253 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 254 mCpuNum = cpuNum; 255 mCpukHz = cpukHz; 256 // ignore sample for purposes of cycles 257 valid = false; 258 } 259 260 // if no change in CPU number or frequency, then record sample for cycle statistics 261 if (valid && mCpukHz > 0) { 262 double cycles = wcNs * cpukHz * 0.000001; 263 mHzStats.sample(cycles); 264 } 265 266 unsigned n = mWcStats.n(); 267 // mCpuUsage.elapsed() is expensive, so don't call it every loop 268 if ((n & 127) == 1) { 269 long long elapsed = mCpuUsage.elapsed(); 270 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 271 double perLoop = elapsed / (double) n; 272 double perLoop100 = perLoop * 0.01; 273 double perLoop1k = perLoop * 0.001; 274 double mean = mWcStats.mean(); 275 double stddev = mWcStats.stddev(); 276 double minimum = mWcStats.minimum(); 277 double maximum = mWcStats.maximum(); 278 double meanCycles = mHzStats.mean(); 279 double stddevCycles = mHzStats.stddev(); 280 double minCycles = mHzStats.minimum(); 281 double maxCycles = mHzStats.maximum(); 282 mCpuUsage.resetElapsed(); 283 mWcStats.reset(); 284 mHzStats.reset(); 285 ALOGD("CPU usage for %s over past %.1f secs\n" 286 " (%u mixer loops at %.1f mean ms per loop):\n" 287 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 288 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 289 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 290 title.string(), 291 elapsed * .000000001, n, perLoop * .000001, 292 mean * .001, 293 stddev * .001, 294 minimum * .001, 295 maximum * .001, 296 mean / perLoop100, 297 stddev / perLoop100, 298 minimum / perLoop100, 299 maximum / perLoop100, 300 meanCycles / perLoop1k, 301 stddevCycles / perLoop1k, 302 minCycles / perLoop1k, 303 maxCycles / perLoop1k); 304 305 } 306 } 307#endif 308}; 309 310// ---------------------------------------------------------------------------- 311// ThreadBase 312// ---------------------------------------------------------------------------- 313 314AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 315 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 316 : Thread(false /*canCallJava*/), 317 mType(type), 318 mAudioFlinger(audioFlinger), 319 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 320 // are set by PlaybackThread::readOutputParameters_l() or 321 // RecordThread::readInputParameters_l() 322 //FIXME: mStandby should be true here. Is this some kind of hack? 323 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 324 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 325 // mName will be set by concrete (non-virtual) subclass 326 mDeathRecipient(new PMDeathRecipient(this)) 327{ 328} 329 330AudioFlinger::ThreadBase::~ThreadBase() 331{ 332 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 333 mConfigEvents.clear(); 334 335 // do not lock the mutex in destructor 336 releaseWakeLock_l(); 337 if (mPowerManager != 0) { 338 sp<IBinder> binder = mPowerManager->asBinder(); 339 binder->unlinkToDeath(mDeathRecipient); 340 } 341} 342 343status_t AudioFlinger::ThreadBase::readyToRun() 344{ 345 status_t status = initCheck(); 346 if (status == NO_ERROR) { 347 ALOGI("AudioFlinger's thread %p ready to run", this); 348 } else { 349 ALOGE("No working audio driver found."); 350 } 351 return status; 352} 353 354void AudioFlinger::ThreadBase::exit() 355{ 356 ALOGV("ThreadBase::exit"); 357 // do any cleanup required for exit to succeed 358 preExit(); 359 { 360 // This lock prevents the following race in thread (uniprocessor for illustration): 361 // if (!exitPending()) { 362 // // context switch from here to exit() 363 // // exit() calls requestExit(), what exitPending() observes 364 // // exit() calls signal(), which is dropped since no waiters 365 // // context switch back from exit() to here 366 // mWaitWorkCV.wait(...); 367 // // now thread is hung 368 // } 369 AutoMutex lock(mLock); 370 requestExit(); 371 mWaitWorkCV.broadcast(); 372 } 373 // When Thread::requestExitAndWait is made virtual and this method is renamed to 374 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 375 requestExitAndWait(); 376} 377 378status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 379{ 380 status_t status; 381 382 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 383 Mutex::Autolock _l(mLock); 384 385 return sendSetParameterConfigEvent_l(keyValuePairs); 386} 387 388// sendConfigEvent_l() must be called with ThreadBase::mLock held 389// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 390status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 391{ 392 status_t status = NO_ERROR; 393 394 mConfigEvents.add(event); 395 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 396 mWaitWorkCV.signal(); 397 mLock.unlock(); 398 { 399 Mutex::Autolock _l(event->mLock); 400 while (event->mWaitStatus) { 401 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 402 event->mStatus = TIMED_OUT; 403 event->mWaitStatus = false; 404 } 405 } 406 status = event->mStatus; 407 } 408 mLock.lock(); 409 return status; 410} 411 412void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 413{ 414 Mutex::Autolock _l(mLock); 415 sendIoConfigEvent_l(event, param); 416} 417 418// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 419void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 420{ 421 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 422 sendConfigEvent_l(configEvent); 423} 424 425// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 426void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 427{ 428 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 429 sendConfigEvent_l(configEvent); 430} 431 432// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 433status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 434{ 435 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 436 return sendConfigEvent_l(configEvent); 437} 438 439status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 440 const struct audio_patch *patch, 441 audio_patch_handle_t *handle) 442{ 443 Mutex::Autolock _l(mLock); 444 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 445 status_t status = sendConfigEvent_l(configEvent); 446 if (status == NO_ERROR) { 447 CreateAudioPatchConfigEventData *data = 448 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 449 *handle = data->mHandle; 450 } 451 return status; 452} 453 454status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 455 const audio_patch_handle_t handle) 456{ 457 Mutex::Autolock _l(mLock); 458 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 459 return sendConfigEvent_l(configEvent); 460} 461 462 463// post condition: mConfigEvents.isEmpty() 464void AudioFlinger::ThreadBase::processConfigEvents_l() 465{ 466 bool configChanged = false; 467 468 while (!mConfigEvents.isEmpty()) { 469 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 470 sp<ConfigEvent> event = mConfigEvents[0]; 471 mConfigEvents.removeAt(0); 472 switch (event->mType) { 473 case CFG_EVENT_PRIO: { 474 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 475 // FIXME Need to understand why this has to be done asynchronously 476 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 477 true /*asynchronous*/); 478 if (err != 0) { 479 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 480 data->mPrio, data->mPid, data->mTid, err); 481 } 482 } break; 483 case CFG_EVENT_IO: { 484 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 485 audioConfigChanged(data->mEvent, data->mParam); 486 } break; 487 case CFG_EVENT_SET_PARAMETER: { 488 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 489 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 490 configChanged = true; 491 } 492 } break; 493 case CFG_EVENT_CREATE_AUDIO_PATCH: { 494 CreateAudioPatchConfigEventData *data = 495 (CreateAudioPatchConfigEventData *)event->mData.get(); 496 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 497 } break; 498 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 499 ReleaseAudioPatchConfigEventData *data = 500 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 501 event->mStatus = releaseAudioPatch_l(data->mHandle); 502 } break; 503 default: 504 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 505 break; 506 } 507 { 508 Mutex::Autolock _l(event->mLock); 509 if (event->mWaitStatus) { 510 event->mWaitStatus = false; 511 event->mCond.signal(); 512 } 513 } 514 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 515 } 516 517 if (configChanged) { 518 cacheParameters_l(); 519 } 520} 521 522String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 523 String8 s; 524 if (output) { 525 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 526 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 527 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 528 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 529 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 530 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 531 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 532 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 533 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 534 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 535 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 536 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 537 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 538 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 539 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 540 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 541 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 542 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 543 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 544 } else { 545 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 546 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 547 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 548 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 549 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 550 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 551 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 552 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 553 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 554 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 555 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 556 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 557 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 558 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 559 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 560 } 561 int len = s.length(); 562 if (s.length() > 2) { 563 char *str = s.lockBuffer(len); 564 s.unlockBuffer(len - 2); 565 } 566 return s; 567} 568 569void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 570{ 571 const size_t SIZE = 256; 572 char buffer[SIZE]; 573 String8 result; 574 575 bool locked = AudioFlinger::dumpTryLock(mLock); 576 if (!locked) { 577 fdprintf(fd, "thread %p maybe dead locked\n", this); 578 } 579 580 fdprintf(fd, " I/O handle: %d\n", mId); 581 fdprintf(fd, " TID: %d\n", getTid()); 582 fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 583 fdprintf(fd, " Sample rate: %u\n", mSampleRate); 584 fdprintf(fd, " HAL frame count: %zu\n", mFrameCount); 585 fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 586 fdprintf(fd, " Channel Count: %u\n", mChannelCount); 587 fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 588 channelMaskToString(mChannelMask, mType != RECORD).string()); 589 fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 590 fdprintf(fd, " Frame size: %zu\n", mFrameSize); 591 fdprintf(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 fdprintf(fd, "\n %s", buffer); 597 } 598 fdprintf(fd, "\n"); 599 } else { 600 fdprintf(fd, " none\n"); 601 } 602 603 if (locked) { 604 mLock.unlock(); 605 } 606} 607 608void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 609{ 610 const size_t SIZE = 256; 611 char buffer[SIZE]; 612 String8 result; 613 614 size_t numEffectChains = mEffectChains.size(); 615 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 616 write(fd, buffer, strlen(buffer)); 617 618 for (size_t i = 0; i < numEffectChains; ++i) { 619 sp<EffectChain> chain = mEffectChains[i]; 620 if (chain != 0) { 621 chain->dump(fd, args); 622 } 623 } 624} 625 626void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 627{ 628 Mutex::Autolock _l(mLock); 629 acquireWakeLock_l(uid); 630} 631 632String16 AudioFlinger::ThreadBase::getWakeLockTag() 633{ 634 switch (mType) { 635 case MIXER: 636 return String16("AudioMix"); 637 case DIRECT: 638 return String16("AudioDirectOut"); 639 case DUPLICATING: 640 return String16("AudioDup"); 641 case RECORD: 642 return String16("AudioIn"); 643 case OFFLOAD: 644 return String16("AudioOffload"); 645 default: 646 ALOG_ASSERT(false); 647 return String16("AudioUnknown"); 648 } 649} 650 651void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 652{ 653 getPowerManager_l(); 654 if (mPowerManager != 0) { 655 sp<IBinder> binder = new BBinder(); 656 status_t status; 657 if (uid >= 0) { 658 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 659 binder, 660 getWakeLockTag(), 661 String16("media"), 662 uid); 663 } else { 664 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 665 binder, 666 getWakeLockTag(), 667 String16("media")); 668 } 669 if (status == NO_ERROR) { 670 mWakeLockToken = binder; 671 } 672 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 673 } 674} 675 676void AudioFlinger::ThreadBase::releaseWakeLock() 677{ 678 Mutex::Autolock _l(mLock); 679 releaseWakeLock_l(); 680} 681 682void AudioFlinger::ThreadBase::releaseWakeLock_l() 683{ 684 if (mWakeLockToken != 0) { 685 ALOGV("releaseWakeLock_l() %s", mName); 686 if (mPowerManager != 0) { 687 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 688 } 689 mWakeLockToken.clear(); 690 } 691} 692 693void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 694 Mutex::Autolock _l(mLock); 695 updateWakeLockUids_l(uids); 696} 697 698void AudioFlinger::ThreadBase::getPowerManager_l() { 699 700 if (mPowerManager == 0) { 701 // use checkService() to avoid blocking if power service is not up yet 702 sp<IBinder> binder = 703 defaultServiceManager()->checkService(String16("power")); 704 if (binder == 0) { 705 ALOGW("Thread %s cannot connect to the power manager service", mName); 706 } else { 707 mPowerManager = interface_cast<IPowerManager>(binder); 708 binder->linkToDeath(mDeathRecipient); 709 } 710 } 711} 712 713void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 714 715 getPowerManager_l(); 716 if (mWakeLockToken == NULL) { 717 ALOGE("no wake lock to update!"); 718 return; 719 } 720 if (mPowerManager != 0) { 721 sp<IBinder> binder = new BBinder(); 722 status_t status; 723 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 724 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 725 } 726} 727 728void AudioFlinger::ThreadBase::clearPowerManager() 729{ 730 Mutex::Autolock _l(mLock); 731 releaseWakeLock_l(); 732 mPowerManager.clear(); 733} 734 735void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 736{ 737 sp<ThreadBase> thread = mThread.promote(); 738 if (thread != 0) { 739 thread->clearPowerManager(); 740 } 741 ALOGW("power manager service died !!!"); 742} 743 744void AudioFlinger::ThreadBase::setEffectSuspended( 745 const effect_uuid_t *type, bool suspend, int sessionId) 746{ 747 Mutex::Autolock _l(mLock); 748 setEffectSuspended_l(type, suspend, sessionId); 749} 750 751void AudioFlinger::ThreadBase::setEffectSuspended_l( 752 const effect_uuid_t *type, bool suspend, int sessionId) 753{ 754 sp<EffectChain> chain = getEffectChain_l(sessionId); 755 if (chain != 0) { 756 if (type != NULL) { 757 chain->setEffectSuspended_l(type, suspend); 758 } else { 759 chain->setEffectSuspendedAll_l(suspend); 760 } 761 } 762 763 updateSuspendedSessions_l(type, suspend, sessionId); 764} 765 766void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 767{ 768 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 769 if (index < 0) { 770 return; 771 } 772 773 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 774 mSuspendedSessions.valueAt(index); 775 776 for (size_t i = 0; i < sessionEffects.size(); i++) { 777 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 778 for (int j = 0; j < desc->mRefCount; j++) { 779 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 780 chain->setEffectSuspendedAll_l(true); 781 } else { 782 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 783 desc->mType.timeLow); 784 chain->setEffectSuspended_l(&desc->mType, true); 785 } 786 } 787 } 788} 789 790void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 791 bool suspend, 792 int sessionId) 793{ 794 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 795 796 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 797 798 if (suspend) { 799 if (index >= 0) { 800 sessionEffects = mSuspendedSessions.valueAt(index); 801 } else { 802 mSuspendedSessions.add(sessionId, sessionEffects); 803 } 804 } else { 805 if (index < 0) { 806 return; 807 } 808 sessionEffects = mSuspendedSessions.valueAt(index); 809 } 810 811 812 int key = EffectChain::kKeyForSuspendAll; 813 if (type != NULL) { 814 key = type->timeLow; 815 } 816 index = sessionEffects.indexOfKey(key); 817 818 sp<SuspendedSessionDesc> desc; 819 if (suspend) { 820 if (index >= 0) { 821 desc = sessionEffects.valueAt(index); 822 } else { 823 desc = new SuspendedSessionDesc(); 824 if (type != NULL) { 825 desc->mType = *type; 826 } 827 sessionEffects.add(key, desc); 828 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 829 } 830 desc->mRefCount++; 831 } else { 832 if (index < 0) { 833 return; 834 } 835 desc = sessionEffects.valueAt(index); 836 if (--desc->mRefCount == 0) { 837 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 838 sessionEffects.removeItemsAt(index); 839 if (sessionEffects.isEmpty()) { 840 ALOGV("updateSuspendedSessions_l() restore removing session %d", 841 sessionId); 842 mSuspendedSessions.removeItem(sessionId); 843 } 844 } 845 } 846 if (!sessionEffects.isEmpty()) { 847 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 848 } 849} 850 851void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 852 bool enabled, 853 int sessionId) 854{ 855 Mutex::Autolock _l(mLock); 856 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 857} 858 859void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 860 bool enabled, 861 int sessionId) 862{ 863 if (mType != RECORD) { 864 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 865 // another session. This gives the priority to well behaved effect control panels 866 // and applications not using global effects. 867 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 868 // global effects 869 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 870 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 871 } 872 } 873 874 sp<EffectChain> chain = getEffectChain_l(sessionId); 875 if (chain != 0) { 876 chain->checkSuspendOnEffectEnabled(effect, enabled); 877 } 878} 879 880// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 881sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 882 const sp<AudioFlinger::Client>& client, 883 const sp<IEffectClient>& effectClient, 884 int32_t priority, 885 int sessionId, 886 effect_descriptor_t *desc, 887 int *enabled, 888 status_t *status) 889{ 890 sp<EffectModule> effect; 891 sp<EffectHandle> handle; 892 status_t lStatus; 893 sp<EffectChain> chain; 894 bool chainCreated = false; 895 bool effectCreated = false; 896 bool effectRegistered = false; 897 898 lStatus = initCheck(); 899 if (lStatus != NO_ERROR) { 900 ALOGW("createEffect_l() Audio driver not initialized."); 901 goto Exit; 902 } 903 904 // Reject any effect on Direct output threads for now, since the format of 905 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 906 if (mType == DIRECT) { 907 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 908 desc->name, mName); 909 lStatus = BAD_VALUE; 910 goto Exit; 911 } 912 913 // Allow global effects only on offloaded and mixer threads 914 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 915 switch (mType) { 916 case MIXER: 917 case OFFLOAD: 918 break; 919 case DIRECT: 920 case DUPLICATING: 921 case RECORD: 922 default: 923 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 924 lStatus = BAD_VALUE; 925 goto Exit; 926 } 927 } 928 929 // Only Pre processor effects are allowed on input threads and only on input threads 930 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 931 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 932 desc->name, desc->flags, mType); 933 lStatus = BAD_VALUE; 934 goto Exit; 935 } 936 937 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 938 939 { // scope for mLock 940 Mutex::Autolock _l(mLock); 941 942 // check for existing effect chain with the requested audio session 943 chain = getEffectChain_l(sessionId); 944 if (chain == 0) { 945 // create a new chain for this session 946 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 947 chain = new EffectChain(this, sessionId); 948 addEffectChain_l(chain); 949 chain->setStrategy(getStrategyForSession_l(sessionId)); 950 chainCreated = true; 951 } else { 952 effect = chain->getEffectFromDesc_l(desc); 953 } 954 955 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 956 957 if (effect == 0) { 958 int id = mAudioFlinger->nextUniqueId(); 959 // Check CPU and memory usage 960 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 961 if (lStatus != NO_ERROR) { 962 goto Exit; 963 } 964 effectRegistered = true; 965 // create a new effect module if none present in the chain 966 effect = new EffectModule(this, chain, desc, id, sessionId); 967 lStatus = effect->status(); 968 if (lStatus != NO_ERROR) { 969 goto Exit; 970 } 971 effect->setOffloaded(mType == OFFLOAD, mId); 972 973 lStatus = chain->addEffect_l(effect); 974 if (lStatus != NO_ERROR) { 975 goto Exit; 976 } 977 effectCreated = true; 978 979 effect->setDevice(mOutDevice); 980 effect->setDevice(mInDevice); 981 effect->setMode(mAudioFlinger->getMode()); 982 effect->setAudioSource(mAudioSource); 983 } 984 // create effect handle and connect it to effect module 985 handle = new EffectHandle(effect, client, effectClient, priority); 986 lStatus = handle->initCheck(); 987 if (lStatus == OK) { 988 lStatus = effect->addHandle(handle.get()); 989 } 990 if (enabled != NULL) { 991 *enabled = (int)effect->isEnabled(); 992 } 993 } 994 995Exit: 996 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 997 Mutex::Autolock _l(mLock); 998 if (effectCreated) { 999 chain->removeEffect_l(effect); 1000 } 1001 if (effectRegistered) { 1002 AudioSystem::unregisterEffect(effect->id()); 1003 } 1004 if (chainCreated) { 1005 removeEffectChain_l(chain); 1006 } 1007 handle.clear(); 1008 } 1009 1010 *status = lStatus; 1011 return handle; 1012} 1013 1014sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1015{ 1016 Mutex::Autolock _l(mLock); 1017 return getEffect_l(sessionId, effectId); 1018} 1019 1020sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1021{ 1022 sp<EffectChain> chain = getEffectChain_l(sessionId); 1023 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1024} 1025 1026// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1027// PlaybackThread::mLock held 1028status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1029{ 1030 // check for existing effect chain with the requested audio session 1031 int sessionId = effect->sessionId(); 1032 sp<EffectChain> chain = getEffectChain_l(sessionId); 1033 bool chainCreated = false; 1034 1035 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1036 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1037 this, effect->desc().name, effect->desc().flags); 1038 1039 if (chain == 0) { 1040 // create a new chain for this session 1041 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1042 chain = new EffectChain(this, sessionId); 1043 addEffectChain_l(chain); 1044 chain->setStrategy(getStrategyForSession_l(sessionId)); 1045 chainCreated = true; 1046 } 1047 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1048 1049 if (chain->getEffectFromId_l(effect->id()) != 0) { 1050 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1051 this, effect->desc().name, chain.get()); 1052 return BAD_VALUE; 1053 } 1054 1055 effect->setOffloaded(mType == OFFLOAD, mId); 1056 1057 status_t status = chain->addEffect_l(effect); 1058 if (status != NO_ERROR) { 1059 if (chainCreated) { 1060 removeEffectChain_l(chain); 1061 } 1062 return status; 1063 } 1064 1065 effect->setDevice(mOutDevice); 1066 effect->setDevice(mInDevice); 1067 effect->setMode(mAudioFlinger->getMode()); 1068 effect->setAudioSource(mAudioSource); 1069 return NO_ERROR; 1070} 1071 1072void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1073 1074 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1075 effect_descriptor_t desc = effect->desc(); 1076 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1077 detachAuxEffect_l(effect->id()); 1078 } 1079 1080 sp<EffectChain> chain = effect->chain().promote(); 1081 if (chain != 0) { 1082 // remove effect chain if removing last effect 1083 if (chain->removeEffect_l(effect) == 0) { 1084 removeEffectChain_l(chain); 1085 } 1086 } else { 1087 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1088 } 1089} 1090 1091void AudioFlinger::ThreadBase::lockEffectChains_l( 1092 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1093{ 1094 effectChains = mEffectChains; 1095 for (size_t i = 0; i < mEffectChains.size(); i++) { 1096 mEffectChains[i]->lock(); 1097 } 1098} 1099 1100void AudioFlinger::ThreadBase::unlockEffectChains( 1101 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1102{ 1103 for (size_t i = 0; i < effectChains.size(); i++) { 1104 effectChains[i]->unlock(); 1105 } 1106} 1107 1108sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1109{ 1110 Mutex::Autolock _l(mLock); 1111 return getEffectChain_l(sessionId); 1112} 1113 1114sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1115{ 1116 size_t size = mEffectChains.size(); 1117 for (size_t i = 0; i < size; i++) { 1118 if (mEffectChains[i]->sessionId() == sessionId) { 1119 return mEffectChains[i]; 1120 } 1121 } 1122 return 0; 1123} 1124 1125void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1126{ 1127 Mutex::Autolock _l(mLock); 1128 size_t size = mEffectChains.size(); 1129 for (size_t i = 0; i < size; i++) { 1130 mEffectChains[i]->setMode_l(mode); 1131 } 1132} 1133 1134void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1135 EffectHandle *handle, 1136 bool unpinIfLast) { 1137 1138 Mutex::Autolock _l(mLock); 1139 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1140 // delete the effect module if removing last handle on it 1141 if (effect->removeHandle(handle) == 0) { 1142 if (!effect->isPinned() || unpinIfLast) { 1143 removeEffect_l(effect); 1144 AudioSystem::unregisterEffect(effect->id()); 1145 } 1146 } 1147} 1148 1149// ---------------------------------------------------------------------------- 1150// Playback 1151// ---------------------------------------------------------------------------- 1152 1153AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1154 AudioStreamOut* output, 1155 audio_io_handle_t id, 1156 audio_devices_t device, 1157 type_t type) 1158 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1159 mNormalFrameCount(0), mSinkBuffer(NULL), 1160 mMixerBufferEnabled(false), 1161 mMixerBuffer(NULL), 1162 mMixerBufferSize(0), 1163 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1164 mMixerBufferValid(false), 1165 mEffectBufferEnabled(false), 1166 mEffectBuffer(NULL), 1167 mEffectBufferSize(0), 1168 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1169 mEffectBufferValid(false), 1170 mSuspended(0), mBytesWritten(0), 1171 mActiveTracksGeneration(0), 1172 // mStreamTypes[] initialized in constructor body 1173 mOutput(output), 1174 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1175 mMixerStatus(MIXER_IDLE), 1176 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1177 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1178 mBytesRemaining(0), 1179 mCurrentWriteLength(0), 1180 mUseAsyncWrite(false), 1181 mWriteAckSequence(0), 1182 mDrainSequence(0), 1183 mSignalPending(false), 1184 mScreenState(AudioFlinger::mScreenState), 1185 // index 0 is reserved for normal mixer's submix 1186 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1187 // mLatchD, mLatchQ, 1188 mLatchDValid(false), mLatchQValid(false) 1189{ 1190 snprintf(mName, kNameLength, "AudioOut_%X", id); 1191 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1192 1193 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1194 // it would be safer to explicitly pass initial masterVolume/masterMute as 1195 // parameter. 1196 // 1197 // If the HAL we are using has support for master volume or master mute, 1198 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1199 // and the mute set to false). 1200 mMasterVolume = audioFlinger->masterVolume_l(); 1201 mMasterMute = audioFlinger->masterMute_l(); 1202 if (mOutput && mOutput->audioHwDev) { 1203 if (mOutput->audioHwDev->canSetMasterVolume()) { 1204 mMasterVolume = 1.0; 1205 } 1206 1207 if (mOutput->audioHwDev->canSetMasterMute()) { 1208 mMasterMute = false; 1209 } 1210 } 1211 1212 readOutputParameters_l(); 1213 1214 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1215 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1216 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1217 stream = (audio_stream_type_t) (stream + 1)) { 1218 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1219 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1220 } 1221 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1222 // because mAudioFlinger doesn't have one to copy from 1223} 1224 1225AudioFlinger::PlaybackThread::~PlaybackThread() 1226{ 1227 mAudioFlinger->unregisterWriter(mNBLogWriter); 1228 free(mSinkBuffer); 1229 free(mMixerBuffer); 1230 free(mEffectBuffer); 1231} 1232 1233void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1234{ 1235 dumpInternals(fd, args); 1236 dumpTracks(fd, args); 1237 dumpEffectChains(fd, args); 1238} 1239 1240void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1241{ 1242 const size_t SIZE = 256; 1243 char buffer[SIZE]; 1244 String8 result; 1245 1246 result.appendFormat(" Stream volumes in dB: "); 1247 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1248 const stream_type_t *st = &mStreamTypes[i]; 1249 if (i > 0) { 1250 result.appendFormat(", "); 1251 } 1252 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1253 if (st->mute) { 1254 result.append("M"); 1255 } 1256 } 1257 result.append("\n"); 1258 write(fd, result.string(), result.length()); 1259 result.clear(); 1260 1261 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1262 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1263 fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1264 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1265 1266 size_t numtracks = mTracks.size(); 1267 size_t numactive = mActiveTracks.size(); 1268 fdprintf(fd, " %d Tracks", numtracks); 1269 size_t numactiveseen = 0; 1270 if (numtracks) { 1271 fdprintf(fd, " of which %d are active\n", numactive); 1272 Track::appendDumpHeader(result); 1273 for (size_t i = 0; i < numtracks; ++i) { 1274 sp<Track> track = mTracks[i]; 1275 if (track != 0) { 1276 bool active = mActiveTracks.indexOf(track) >= 0; 1277 if (active) { 1278 numactiveseen++; 1279 } 1280 track->dump(buffer, SIZE, active); 1281 result.append(buffer); 1282 } 1283 } 1284 } else { 1285 result.append("\n"); 1286 } 1287 if (numactiveseen != numactive) { 1288 // some tracks in the active list were not in the tracks list 1289 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1290 " not in the track list\n"); 1291 result.append(buffer); 1292 Track::appendDumpHeader(result); 1293 for (size_t i = 0; i < numactive; ++i) { 1294 sp<Track> track = mActiveTracks[i].promote(); 1295 if (track != 0 && mTracks.indexOf(track) < 0) { 1296 track->dump(buffer, SIZE, true); 1297 result.append(buffer); 1298 } 1299 } 1300 } 1301 1302 write(fd, result.string(), result.size()); 1303 1304} 1305 1306void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1307{ 1308 fdprintf(fd, "\nOutput thread %p:\n", this); 1309 fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1310 fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1311 fdprintf(fd, " Total writes: %d\n", mNumWrites); 1312 fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1313 fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1314 fdprintf(fd, " Suspend count: %d\n", mSuspended); 1315 fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1316 fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1317 fdprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1318 fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1319 1320 dumpBase(fd, args); 1321} 1322 1323// Thread virtuals 1324 1325void AudioFlinger::PlaybackThread::onFirstRef() 1326{ 1327 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1328} 1329 1330// ThreadBase virtuals 1331void AudioFlinger::PlaybackThread::preExit() 1332{ 1333 ALOGV(" preExit()"); 1334 // FIXME this is using hard-coded strings but in the future, this functionality will be 1335 // converted to use audio HAL extensions required to support tunneling 1336 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1337} 1338 1339// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1340sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1341 const sp<AudioFlinger::Client>& client, 1342 audio_stream_type_t streamType, 1343 uint32_t sampleRate, 1344 audio_format_t format, 1345 audio_channel_mask_t channelMask, 1346 size_t *pFrameCount, 1347 const sp<IMemory>& sharedBuffer, 1348 int sessionId, 1349 IAudioFlinger::track_flags_t *flags, 1350 pid_t tid, 1351 int uid, 1352 status_t *status) 1353{ 1354 size_t frameCount = *pFrameCount; 1355 sp<Track> track; 1356 status_t lStatus; 1357 1358 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1359 1360 // client expresses a preference for FAST, but we get the final say 1361 if (*flags & IAudioFlinger::TRACK_FAST) { 1362 if ( 1363 // not timed 1364 (!isTimed) && 1365 // either of these use cases: 1366 ( 1367 // use case 1: shared buffer with any frame count 1368 ( 1369 (sharedBuffer != 0) 1370 ) || 1371 // use case 2: callback handler and frame count is default or at least as large as HAL 1372 ( 1373 (tid != -1) && 1374 ((frameCount == 0) || 1375 (frameCount >= mFrameCount)) 1376 ) 1377 ) && 1378 // PCM data 1379 audio_is_linear_pcm(format) && 1380 // mono or stereo 1381 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1382 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1383 // hardware sample rate 1384 (sampleRate == mSampleRate) && 1385 // normal mixer has an associated fast mixer 1386 hasFastMixer() && 1387 // there are sufficient fast track slots available 1388 (mFastTrackAvailMask != 0) 1389 // FIXME test that MixerThread for this fast track has a capable output HAL 1390 // FIXME add a permission test also? 1391 ) { 1392 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1393 if (frameCount == 0) { 1394 // read the fast track multiplier property the first time it is needed 1395 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1396 if (ok != 0) { 1397 ALOGE("%s pthread_once failed: %d", __func__, ok); 1398 } 1399 frameCount = mFrameCount * sFastTrackMultiplier; 1400 } 1401 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1402 frameCount, mFrameCount); 1403 } else { 1404 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1405 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1406 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1407 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1408 audio_is_linear_pcm(format), 1409 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1410 *flags &= ~IAudioFlinger::TRACK_FAST; 1411 // For compatibility with AudioTrack calculation, buffer depth is forced 1412 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1413 // This is probably too conservative, but legacy application code may depend on it. 1414 // If you change this calculation, also review the start threshold which is related. 1415 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1416 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1417 if (minBufCount < 2) { 1418 minBufCount = 2; 1419 } 1420 size_t minFrameCount = mNormalFrameCount * minBufCount; 1421 if (frameCount < minFrameCount) { 1422 frameCount = minFrameCount; 1423 } 1424 } 1425 } 1426 *pFrameCount = frameCount; 1427 1428 switch (mType) { 1429 1430 case DIRECT: 1431 if (audio_is_linear_pcm(format)) { 1432 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1433 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1434 "for output %p with format %#x", 1435 sampleRate, format, channelMask, mOutput, mFormat); 1436 lStatus = BAD_VALUE; 1437 goto Exit; 1438 } 1439 } 1440 break; 1441 1442 case OFFLOAD: 1443 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1444 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1445 "for output %p with format %#x", 1446 sampleRate, format, channelMask, mOutput, mFormat); 1447 lStatus = BAD_VALUE; 1448 goto Exit; 1449 } 1450 break; 1451 1452 default: 1453 if (!audio_is_linear_pcm(format)) { 1454 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1455 "for output %p with format %#x", 1456 format, mOutput, mFormat); 1457 lStatus = BAD_VALUE; 1458 goto Exit; 1459 } 1460 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1461 if (sampleRate > mSampleRate*2) { 1462 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1463 lStatus = BAD_VALUE; 1464 goto Exit; 1465 } 1466 break; 1467 1468 } 1469 1470 lStatus = initCheck(); 1471 if (lStatus != NO_ERROR) { 1472 ALOGE("createTrack_l() audio driver not initialized"); 1473 goto Exit; 1474 } 1475 1476 { // scope for mLock 1477 Mutex::Autolock _l(mLock); 1478 1479 // all tracks in same audio session must share the same routing strategy otherwise 1480 // conflicts will happen when tracks are moved from one output to another by audio policy 1481 // manager 1482 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1483 for (size_t i = 0; i < mTracks.size(); ++i) { 1484 sp<Track> t = mTracks[i]; 1485 if (t != 0 && !t->isOutputTrack()) { 1486 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1487 if (sessionId == t->sessionId() && strategy != actual) { 1488 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1489 strategy, actual); 1490 lStatus = BAD_VALUE; 1491 goto Exit; 1492 } 1493 } 1494 } 1495 1496 if (!isTimed) { 1497 track = new Track(this, client, streamType, sampleRate, format, 1498 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1499 } else { 1500 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1501 channelMask, frameCount, sharedBuffer, sessionId, uid); 1502 } 1503 1504 // new Track always returns non-NULL, 1505 // but TimedTrack::create() is a factory that could fail by returning NULL 1506 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1507 if (lStatus != NO_ERROR) { 1508 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1509 // track must be cleared from the caller as the caller has the AF lock 1510 goto Exit; 1511 } 1512 mTracks.add(track); 1513 1514 sp<EffectChain> chain = getEffectChain_l(sessionId); 1515 if (chain != 0) { 1516 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1517 track->setMainBuffer(chain->inBuffer()); 1518 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1519 chain->incTrackCnt(); 1520 } 1521 1522 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1523 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1524 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1525 // so ask activity manager to do this on our behalf 1526 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1527 } 1528 } 1529 1530 lStatus = NO_ERROR; 1531 1532Exit: 1533 *status = lStatus; 1534 return track; 1535} 1536 1537uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1538{ 1539 return latency; 1540} 1541 1542uint32_t AudioFlinger::PlaybackThread::latency() const 1543{ 1544 Mutex::Autolock _l(mLock); 1545 return latency_l(); 1546} 1547uint32_t AudioFlinger::PlaybackThread::latency_l() const 1548{ 1549 if (initCheck() == NO_ERROR) { 1550 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1551 } else { 1552 return 0; 1553 } 1554} 1555 1556void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1557{ 1558 Mutex::Autolock _l(mLock); 1559 // Don't apply master volume in SW if our HAL can do it for us. 1560 if (mOutput && mOutput->audioHwDev && 1561 mOutput->audioHwDev->canSetMasterVolume()) { 1562 mMasterVolume = 1.0; 1563 } else { 1564 mMasterVolume = value; 1565 } 1566} 1567 1568void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1569{ 1570 Mutex::Autolock _l(mLock); 1571 // Don't apply master mute in SW if our HAL can do it for us. 1572 if (mOutput && mOutput->audioHwDev && 1573 mOutput->audioHwDev->canSetMasterMute()) { 1574 mMasterMute = false; 1575 } else { 1576 mMasterMute = muted; 1577 } 1578} 1579 1580void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1581{ 1582 Mutex::Autolock _l(mLock); 1583 mStreamTypes[stream].volume = value; 1584 broadcast_l(); 1585} 1586 1587void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1588{ 1589 Mutex::Autolock _l(mLock); 1590 mStreamTypes[stream].mute = muted; 1591 broadcast_l(); 1592} 1593 1594float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1595{ 1596 Mutex::Autolock _l(mLock); 1597 return mStreamTypes[stream].volume; 1598} 1599 1600// addTrack_l() must be called with ThreadBase::mLock held 1601status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1602{ 1603 status_t status = ALREADY_EXISTS; 1604 1605 // set retry count for buffer fill 1606 track->mRetryCount = kMaxTrackStartupRetries; 1607 if (mActiveTracks.indexOf(track) < 0) { 1608 // the track is newly added, make sure it fills up all its 1609 // buffers before playing. This is to ensure the client will 1610 // effectively get the latency it requested. 1611 if (!track->isOutputTrack()) { 1612 TrackBase::track_state state = track->mState; 1613 mLock.unlock(); 1614 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1615 mLock.lock(); 1616 // abort track was stopped/paused while we released the lock 1617 if (state != track->mState) { 1618 if (status == NO_ERROR) { 1619 mLock.unlock(); 1620 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1621 mLock.lock(); 1622 } 1623 return INVALID_OPERATION; 1624 } 1625 // abort if start is rejected by audio policy manager 1626 if (status != NO_ERROR) { 1627 return PERMISSION_DENIED; 1628 } 1629#ifdef ADD_BATTERY_DATA 1630 // to track the speaker usage 1631 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1632#endif 1633 } 1634 1635 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1636 track->mResetDone = false; 1637 track->mPresentationCompleteFrames = 0; 1638 mActiveTracks.add(track); 1639 mWakeLockUids.add(track->uid()); 1640 mActiveTracksGeneration++; 1641 mLatestActiveTrack = track; 1642 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1643 if (chain != 0) { 1644 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1645 track->sessionId()); 1646 chain->incActiveTrackCnt(); 1647 } 1648 1649 status = NO_ERROR; 1650 } 1651 1652 onAddNewTrack_l(); 1653 return status; 1654} 1655 1656bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1657{ 1658 track->terminate(); 1659 // active tracks are removed by threadLoop() 1660 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1661 track->mState = TrackBase::STOPPED; 1662 if (!trackActive) { 1663 removeTrack_l(track); 1664 } else if (track->isFastTrack() || track->isOffloaded()) { 1665 track->mState = TrackBase::STOPPING_1; 1666 } 1667 1668 return trackActive; 1669} 1670 1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1672{ 1673 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1674 mTracks.remove(track); 1675 deleteTrackName_l(track->name()); 1676 // redundant as track is about to be destroyed, for dumpsys only 1677 track->mName = -1; 1678 if (track->isFastTrack()) { 1679 int index = track->mFastIndex; 1680 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1681 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1682 mFastTrackAvailMask |= 1 << index; 1683 // redundant as track is about to be destroyed, for dumpsys only 1684 track->mFastIndex = -1; 1685 } 1686 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1687 if (chain != 0) { 1688 chain->decTrackCnt(); 1689 } 1690} 1691 1692void AudioFlinger::PlaybackThread::broadcast_l() 1693{ 1694 // Thread could be blocked waiting for async 1695 // so signal it to handle state changes immediately 1696 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1697 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1698 mSignalPending = true; 1699 mWaitWorkCV.broadcast(); 1700} 1701 1702String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1703{ 1704 Mutex::Autolock _l(mLock); 1705 if (initCheck() != NO_ERROR) { 1706 return String8(); 1707 } 1708 1709 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1710 const String8 out_s8(s); 1711 free(s); 1712 return out_s8; 1713} 1714 1715void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1716 AudioSystem::OutputDescriptor desc; 1717 void *param2 = NULL; 1718 1719 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1720 param); 1721 1722 switch (event) { 1723 case AudioSystem::OUTPUT_OPENED: 1724 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1725 desc.channelMask = mChannelMask; 1726 desc.samplingRate = mSampleRate; 1727 desc.format = mFormat; 1728 desc.frameCount = mNormalFrameCount; // FIXME see 1729 // AudioFlinger::frameCount(audio_io_handle_t) 1730 desc.latency = latency_l(); 1731 param2 = &desc; 1732 break; 1733 1734 case AudioSystem::STREAM_CONFIG_CHANGED: 1735 param2 = ¶m; 1736 case AudioSystem::OUTPUT_CLOSED: 1737 default: 1738 break; 1739 } 1740 mAudioFlinger->audioConfigChanged(event, mId, param2); 1741} 1742 1743void AudioFlinger::PlaybackThread::writeCallback() 1744{ 1745 ALOG_ASSERT(mCallbackThread != 0); 1746 mCallbackThread->resetWriteBlocked(); 1747} 1748 1749void AudioFlinger::PlaybackThread::drainCallback() 1750{ 1751 ALOG_ASSERT(mCallbackThread != 0); 1752 mCallbackThread->resetDraining(); 1753} 1754 1755void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1756{ 1757 Mutex::Autolock _l(mLock); 1758 // reject out of sequence requests 1759 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1760 mWriteAckSequence &= ~1; 1761 mWaitWorkCV.signal(); 1762 } 1763} 1764 1765void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1766{ 1767 Mutex::Autolock _l(mLock); 1768 // reject out of sequence requests 1769 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1770 mDrainSequence &= ~1; 1771 mWaitWorkCV.signal(); 1772 } 1773} 1774 1775// static 1776int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1777 void *param __unused, 1778 void *cookie) 1779{ 1780 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1781 ALOGV("asyncCallback() event %d", event); 1782 switch (event) { 1783 case STREAM_CBK_EVENT_WRITE_READY: 1784 me->writeCallback(); 1785 break; 1786 case STREAM_CBK_EVENT_DRAIN_READY: 1787 me->drainCallback(); 1788 break; 1789 default: 1790 ALOGW("asyncCallback() unknown event %d", event); 1791 break; 1792 } 1793 return 0; 1794} 1795 1796void AudioFlinger::PlaybackThread::readOutputParameters_l() 1797{ 1798 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1799 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1800 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1801 if (!audio_is_output_channel(mChannelMask)) { 1802 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1803 } 1804 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1805 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1806 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1807 } 1808 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1809 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1810 if (!audio_is_valid_format(mFormat)) { 1811 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1812 } 1813 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1814 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1815 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1816 } 1817 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1818 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1819 mFrameCount = mBufferSize / mFrameSize; 1820 if (mFrameCount & 15) { 1821 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1822 mFrameCount); 1823 } 1824 1825 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1826 (mOutput->stream->set_callback != NULL)) { 1827 if (mOutput->stream->set_callback(mOutput->stream, 1828 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1829 mUseAsyncWrite = true; 1830 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1831 } 1832 } 1833 1834 // Calculate size of normal sink buffer relative to the HAL output buffer size 1835 double multiplier = 1.0; 1836 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1837 kUseFastMixer == FastMixer_Dynamic)) { 1838 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1839 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1840 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1841 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1842 maxNormalFrameCount = maxNormalFrameCount & ~15; 1843 if (maxNormalFrameCount < minNormalFrameCount) { 1844 maxNormalFrameCount = minNormalFrameCount; 1845 } 1846 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1847 if (multiplier <= 1.0) { 1848 multiplier = 1.0; 1849 } else if (multiplier <= 2.0) { 1850 if (2 * mFrameCount <= maxNormalFrameCount) { 1851 multiplier = 2.0; 1852 } else { 1853 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1854 } 1855 } else { 1856 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1857 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1858 // track, but we sometimes have to do this to satisfy the maximum frame count 1859 // constraint) 1860 // FIXME this rounding up should not be done if no HAL SRC 1861 uint32_t truncMult = (uint32_t) multiplier; 1862 if ((truncMult & 1)) { 1863 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1864 ++truncMult; 1865 } 1866 } 1867 multiplier = (double) truncMult; 1868 } 1869 } 1870 mNormalFrameCount = multiplier * mFrameCount; 1871 // round up to nearest 16 frames to satisfy AudioMixer 1872 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1873 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1874 mNormalFrameCount); 1875 1876 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1877 // Originally this was int16_t[] array, need to remove legacy implications. 1878 free(mSinkBuffer); 1879 mSinkBuffer = NULL; 1880 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1881 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1882 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1883 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1884 1885 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1886 // drives the output. 1887 free(mMixerBuffer); 1888 mMixerBuffer = NULL; 1889 if (mMixerBufferEnabled) { 1890 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1891 mMixerBufferSize = mNormalFrameCount * mChannelCount 1892 * audio_bytes_per_sample(mMixerBufferFormat); 1893 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1894 } 1895 free(mEffectBuffer); 1896 mEffectBuffer = NULL; 1897 if (mEffectBufferEnabled) { 1898 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1899 mEffectBufferSize = mNormalFrameCount * mChannelCount 1900 * audio_bytes_per_sample(mEffectBufferFormat); 1901 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1902 } 1903 1904 // force reconfiguration of effect chains and engines to take new buffer size and audio 1905 // parameters into account 1906 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1907 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1908 // matter. 1909 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1910 Vector< sp<EffectChain> > effectChains = mEffectChains; 1911 for (size_t i = 0; i < effectChains.size(); i ++) { 1912 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1913 } 1914} 1915 1916 1917status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1918{ 1919 if (halFrames == NULL || dspFrames == NULL) { 1920 return BAD_VALUE; 1921 } 1922 Mutex::Autolock _l(mLock); 1923 if (initCheck() != NO_ERROR) { 1924 return INVALID_OPERATION; 1925 } 1926 size_t framesWritten = mBytesWritten / mFrameSize; 1927 *halFrames = framesWritten; 1928 1929 if (isSuspended()) { 1930 // return an estimation of rendered frames when the output is suspended 1931 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1932 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1933 return NO_ERROR; 1934 } else { 1935 status_t status; 1936 uint32_t frames; 1937 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1938 *dspFrames = (size_t)frames; 1939 return status; 1940 } 1941} 1942 1943uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1944{ 1945 Mutex::Autolock _l(mLock); 1946 uint32_t result = 0; 1947 if (getEffectChain_l(sessionId) != 0) { 1948 result = EFFECT_SESSION; 1949 } 1950 1951 for (size_t i = 0; i < mTracks.size(); ++i) { 1952 sp<Track> track = mTracks[i]; 1953 if (sessionId == track->sessionId() && !track->isInvalid()) { 1954 result |= TRACK_SESSION; 1955 break; 1956 } 1957 } 1958 1959 return result; 1960} 1961 1962uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1963{ 1964 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1965 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1966 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1967 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1968 } 1969 for (size_t i = 0; i < mTracks.size(); i++) { 1970 sp<Track> track = mTracks[i]; 1971 if (sessionId == track->sessionId() && !track->isInvalid()) { 1972 return AudioSystem::getStrategyForStream(track->streamType()); 1973 } 1974 } 1975 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1976} 1977 1978 1979AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1980{ 1981 Mutex::Autolock _l(mLock); 1982 return mOutput; 1983} 1984 1985AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1986{ 1987 Mutex::Autolock _l(mLock); 1988 AudioStreamOut *output = mOutput; 1989 mOutput = NULL; 1990 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1991 // must push a NULL and wait for ack 1992 mOutputSink.clear(); 1993 mPipeSink.clear(); 1994 mNormalSink.clear(); 1995 return output; 1996} 1997 1998// this method must always be called either with ThreadBase mLock held or inside the thread loop 1999audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2000{ 2001 if (mOutput == NULL) { 2002 return NULL; 2003 } 2004 return &mOutput->stream->common; 2005} 2006 2007uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2008{ 2009 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2010} 2011 2012status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2013{ 2014 if (!isValidSyncEvent(event)) { 2015 return BAD_VALUE; 2016 } 2017 2018 Mutex::Autolock _l(mLock); 2019 2020 for (size_t i = 0; i < mTracks.size(); ++i) { 2021 sp<Track> track = mTracks[i]; 2022 if (event->triggerSession() == track->sessionId()) { 2023 (void) track->setSyncEvent(event); 2024 return NO_ERROR; 2025 } 2026 } 2027 2028 return NAME_NOT_FOUND; 2029} 2030 2031bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2032{ 2033 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2034} 2035 2036void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2037 const Vector< sp<Track> >& tracksToRemove) 2038{ 2039 size_t count = tracksToRemove.size(); 2040 if (count > 0) { 2041 for (size_t i = 0 ; i < count ; i++) { 2042 const sp<Track>& track = tracksToRemove.itemAt(i); 2043 if (!track->isOutputTrack()) { 2044 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2045#ifdef ADD_BATTERY_DATA 2046 // to track the speaker usage 2047 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2048#endif 2049 if (track->isTerminated()) { 2050 AudioSystem::releaseOutput(mId); 2051 } 2052 } 2053 } 2054 } 2055} 2056 2057void AudioFlinger::PlaybackThread::checkSilentMode_l() 2058{ 2059 if (!mMasterMute) { 2060 char value[PROPERTY_VALUE_MAX]; 2061 if (property_get("ro.audio.silent", value, "0") > 0) { 2062 char *endptr; 2063 unsigned long ul = strtoul(value, &endptr, 0); 2064 if (*endptr == '\0' && ul != 0) { 2065 ALOGD("Silence is golden"); 2066 // The setprop command will not allow a property to be changed after 2067 // the first time it is set, so we don't have to worry about un-muting. 2068 setMasterMute_l(true); 2069 } 2070 } 2071 } 2072} 2073 2074// shared by MIXER and DIRECT, overridden by DUPLICATING 2075ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2076{ 2077 // FIXME rewrite to reduce number of system calls 2078 mLastWriteTime = systemTime(); 2079 mInWrite = true; 2080 ssize_t bytesWritten; 2081 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2082 2083 // If an NBAIO sink is present, use it to write the normal mixer's submix 2084 if (mNormalSink != 0) { 2085 const size_t count = mBytesRemaining / mFrameSize; 2086 2087 ATRACE_BEGIN("write"); 2088 // update the setpoint when AudioFlinger::mScreenState changes 2089 uint32_t screenState = AudioFlinger::mScreenState; 2090 if (screenState != mScreenState) { 2091 mScreenState = screenState; 2092 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2093 if (pipe != NULL) { 2094 pipe->setAvgFrames((mScreenState & 1) ? 2095 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2096 } 2097 } 2098 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2099 ATRACE_END(); 2100 if (framesWritten > 0) { 2101 bytesWritten = framesWritten * mFrameSize; 2102 } else { 2103 bytesWritten = framesWritten; 2104 } 2105 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2106 if (status == NO_ERROR) { 2107 size_t totalFramesWritten = mNormalSink->framesWritten(); 2108 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2109 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2110 mLatchDValid = true; 2111 } 2112 } 2113 // otherwise use the HAL / AudioStreamOut directly 2114 } else { 2115 // Direct output and offload threads 2116 2117 if (mUseAsyncWrite) { 2118 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2119 mWriteAckSequence += 2; 2120 mWriteAckSequence |= 1; 2121 ALOG_ASSERT(mCallbackThread != 0); 2122 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2123 } 2124 // FIXME We should have an implementation of timestamps for direct output threads. 2125 // They are used e.g for multichannel PCM playback over HDMI. 2126 bytesWritten = mOutput->stream->write(mOutput->stream, 2127 (char *)mSinkBuffer + offset, mBytesRemaining); 2128 if (mUseAsyncWrite && 2129 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2130 // do not wait for async callback in case of error of full write 2131 mWriteAckSequence &= ~1; 2132 ALOG_ASSERT(mCallbackThread != 0); 2133 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2134 } 2135 } 2136 2137 mNumWrites++; 2138 mInWrite = false; 2139 mStandby = false; 2140 return bytesWritten; 2141} 2142 2143void AudioFlinger::PlaybackThread::threadLoop_drain() 2144{ 2145 if (mOutput->stream->drain) { 2146 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2147 if (mUseAsyncWrite) { 2148 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2149 mDrainSequence |= 1; 2150 ALOG_ASSERT(mCallbackThread != 0); 2151 mCallbackThread->setDraining(mDrainSequence); 2152 } 2153 mOutput->stream->drain(mOutput->stream, 2154 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2155 : AUDIO_DRAIN_ALL); 2156 } 2157} 2158 2159void AudioFlinger::PlaybackThread::threadLoop_exit() 2160{ 2161 // Default implementation has nothing to do 2162} 2163 2164/* 2165The derived values that are cached: 2166 - mSinkBufferSize from frame count * frame size 2167 - activeSleepTime from activeSleepTimeUs() 2168 - idleSleepTime from idleSleepTimeUs() 2169 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2170 - maxPeriod from frame count and sample rate (MIXER only) 2171 2172The parameters that affect these derived values are: 2173 - frame count 2174 - frame size 2175 - sample rate 2176 - device type: A2DP or not 2177 - device latency 2178 - format: PCM or not 2179 - active sleep time 2180 - idle sleep time 2181*/ 2182 2183void AudioFlinger::PlaybackThread::cacheParameters_l() 2184{ 2185 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2186 activeSleepTime = activeSleepTimeUs(); 2187 idleSleepTime = idleSleepTimeUs(); 2188} 2189 2190void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2191{ 2192 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2193 this, streamType, mTracks.size()); 2194 Mutex::Autolock _l(mLock); 2195 2196 size_t size = mTracks.size(); 2197 for (size_t i = 0; i < size; i++) { 2198 sp<Track> t = mTracks[i]; 2199 if (t->streamType() == streamType) { 2200 t->invalidate(); 2201 } 2202 } 2203} 2204 2205status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2206{ 2207 int session = chain->sessionId(); 2208 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2209 ? mEffectBuffer : mSinkBuffer); 2210 bool ownsBuffer = false; 2211 2212 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2213 if (session > 0) { 2214 // Only one effect chain can be present in direct output thread and it uses 2215 // the sink buffer as input 2216 if (mType != DIRECT) { 2217 size_t numSamples = mNormalFrameCount * mChannelCount; 2218 buffer = new int16_t[numSamples]; 2219 memset(buffer, 0, numSamples * sizeof(int16_t)); 2220 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2221 ownsBuffer = true; 2222 } 2223 2224 // Attach all tracks with same session ID to this chain. 2225 for (size_t i = 0; i < mTracks.size(); ++i) { 2226 sp<Track> track = mTracks[i]; 2227 if (session == track->sessionId()) { 2228 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2229 buffer); 2230 track->setMainBuffer(buffer); 2231 chain->incTrackCnt(); 2232 } 2233 } 2234 2235 // indicate all active tracks in the chain 2236 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2237 sp<Track> track = mActiveTracks[i].promote(); 2238 if (track == 0) { 2239 continue; 2240 } 2241 if (session == track->sessionId()) { 2242 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2243 chain->incActiveTrackCnt(); 2244 } 2245 } 2246 } 2247 2248 chain->setInBuffer(buffer, ownsBuffer); 2249 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2250 ? mEffectBuffer : mSinkBuffer)); 2251 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2252 // chains list in order to be processed last as it contains output stage effects 2253 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2254 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2255 // after track specific effects and before output stage 2256 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2257 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2258 // Effect chain for other sessions are inserted at beginning of effect 2259 // chains list to be processed before output mix effects. Relative order between other 2260 // sessions is not important 2261 size_t size = mEffectChains.size(); 2262 size_t i = 0; 2263 for (i = 0; i < size; i++) { 2264 if (mEffectChains[i]->sessionId() < session) { 2265 break; 2266 } 2267 } 2268 mEffectChains.insertAt(chain, i); 2269 checkSuspendOnAddEffectChain_l(chain); 2270 2271 return NO_ERROR; 2272} 2273 2274size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2275{ 2276 int session = chain->sessionId(); 2277 2278 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2279 2280 for (size_t i = 0; i < mEffectChains.size(); i++) { 2281 if (chain == mEffectChains[i]) { 2282 mEffectChains.removeAt(i); 2283 // detach all active tracks from the chain 2284 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2285 sp<Track> track = mActiveTracks[i].promote(); 2286 if (track == 0) { 2287 continue; 2288 } 2289 if (session == track->sessionId()) { 2290 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2291 chain.get(), session); 2292 chain->decActiveTrackCnt(); 2293 } 2294 } 2295 2296 // detach all tracks with same session ID from this chain 2297 for (size_t i = 0; i < mTracks.size(); ++i) { 2298 sp<Track> track = mTracks[i]; 2299 if (session == track->sessionId()) { 2300 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2301 chain->decTrackCnt(); 2302 } 2303 } 2304 break; 2305 } 2306 } 2307 return mEffectChains.size(); 2308} 2309 2310status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2311 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2312{ 2313 Mutex::Autolock _l(mLock); 2314 return attachAuxEffect_l(track, EffectId); 2315} 2316 2317status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2318 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2319{ 2320 status_t status = NO_ERROR; 2321 2322 if (EffectId == 0) { 2323 track->setAuxBuffer(0, NULL); 2324 } else { 2325 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2326 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2327 if (effect != 0) { 2328 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2329 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2330 } else { 2331 status = INVALID_OPERATION; 2332 } 2333 } else { 2334 status = BAD_VALUE; 2335 } 2336 } 2337 return status; 2338} 2339 2340void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2341{ 2342 for (size_t i = 0; i < mTracks.size(); ++i) { 2343 sp<Track> track = mTracks[i]; 2344 if (track->auxEffectId() == effectId) { 2345 attachAuxEffect_l(track, 0); 2346 } 2347 } 2348} 2349 2350bool AudioFlinger::PlaybackThread::threadLoop() 2351{ 2352 Vector< sp<Track> > tracksToRemove; 2353 2354 standbyTime = systemTime(); 2355 2356 // MIXER 2357 nsecs_t lastWarning = 0; 2358 2359 // DUPLICATING 2360 // FIXME could this be made local to while loop? 2361 writeFrames = 0; 2362 2363 int lastGeneration = 0; 2364 2365 cacheParameters_l(); 2366 sleepTime = idleSleepTime; 2367 2368 if (mType == MIXER) { 2369 sleepTimeShift = 0; 2370 } 2371 2372 CpuStats cpuStats; 2373 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2374 2375 acquireWakeLock(); 2376 2377 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2378 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2379 // and then that string will be logged at the next convenient opportunity. 2380 const char *logString = NULL; 2381 2382 checkSilentMode_l(); 2383 2384 while (!exitPending()) 2385 { 2386 cpuStats.sample(myName); 2387 2388 Vector< sp<EffectChain> > effectChains; 2389 2390 { // scope for mLock 2391 2392 Mutex::Autolock _l(mLock); 2393 2394 processConfigEvents_l(); 2395 2396 if (logString != NULL) { 2397 mNBLogWriter->logTimestamp(); 2398 mNBLogWriter->log(logString); 2399 logString = NULL; 2400 } 2401 2402 if (mLatchDValid) { 2403 mLatchQ = mLatchD; 2404 mLatchDValid = false; 2405 mLatchQValid = true; 2406 } 2407 2408 saveOutputTracks(); 2409 if (mSignalPending) { 2410 // A signal was raised while we were unlocked 2411 mSignalPending = false; 2412 } else if (waitingAsyncCallback_l()) { 2413 if (exitPending()) { 2414 break; 2415 } 2416 releaseWakeLock_l(); 2417 mWakeLockUids.clear(); 2418 mActiveTracksGeneration++; 2419 ALOGV("wait async completion"); 2420 mWaitWorkCV.wait(mLock); 2421 ALOGV("async completion/wake"); 2422 acquireWakeLock_l(); 2423 standbyTime = systemTime() + standbyDelay; 2424 sleepTime = 0; 2425 2426 continue; 2427 } 2428 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2429 isSuspended()) { 2430 // put audio hardware into standby after short delay 2431 if (shouldStandby_l()) { 2432 2433 threadLoop_standby(); 2434 2435 mStandby = true; 2436 } 2437 2438 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2439 // we're about to wait, flush the binder command buffer 2440 IPCThreadState::self()->flushCommands(); 2441 2442 clearOutputTracks(); 2443 2444 if (exitPending()) { 2445 break; 2446 } 2447 2448 releaseWakeLock_l(); 2449 mWakeLockUids.clear(); 2450 mActiveTracksGeneration++; 2451 // wait until we have something to do... 2452 ALOGV("%s going to sleep", myName.string()); 2453 mWaitWorkCV.wait(mLock); 2454 ALOGV("%s waking up", myName.string()); 2455 acquireWakeLock_l(); 2456 2457 mMixerStatus = MIXER_IDLE; 2458 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2459 mBytesWritten = 0; 2460 mBytesRemaining = 0; 2461 checkSilentMode_l(); 2462 2463 standbyTime = systemTime() + standbyDelay; 2464 sleepTime = idleSleepTime; 2465 if (mType == MIXER) { 2466 sleepTimeShift = 0; 2467 } 2468 2469 continue; 2470 } 2471 } 2472 // mMixerStatusIgnoringFastTracks is also updated internally 2473 mMixerStatus = prepareTracks_l(&tracksToRemove); 2474 2475 // compare with previously applied list 2476 if (lastGeneration != mActiveTracksGeneration) { 2477 // update wakelock 2478 updateWakeLockUids_l(mWakeLockUids); 2479 lastGeneration = mActiveTracksGeneration; 2480 } 2481 2482 // prevent any changes in effect chain list and in each effect chain 2483 // during mixing and effect process as the audio buffers could be deleted 2484 // or modified if an effect is created or deleted 2485 lockEffectChains_l(effectChains); 2486 } // mLock scope ends 2487 2488 if (mBytesRemaining == 0) { 2489 mCurrentWriteLength = 0; 2490 if (mMixerStatus == MIXER_TRACKS_READY) { 2491 // threadLoop_mix() sets mCurrentWriteLength 2492 threadLoop_mix(); 2493 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2494 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2495 // threadLoop_sleepTime sets sleepTime to 0 if data 2496 // must be written to HAL 2497 threadLoop_sleepTime(); 2498 if (sleepTime == 0) { 2499 mCurrentWriteLength = mSinkBufferSize; 2500 } 2501 } 2502 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2503 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2504 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2505 // or mSinkBuffer (if there are no effects). 2506 // 2507 // This is done pre-effects computation; if effects change to 2508 // support higher precision, this needs to move. 2509 // 2510 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2511 // TODO use sleepTime == 0 as an additional condition. 2512 if (mMixerBufferValid) { 2513 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2514 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2515 2516 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2517 mNormalFrameCount * mChannelCount); 2518 } 2519 2520 mBytesRemaining = mCurrentWriteLength; 2521 if (isSuspended()) { 2522 sleepTime = suspendSleepTimeUs(); 2523 // simulate write to HAL when suspended 2524 mBytesWritten += mSinkBufferSize; 2525 mBytesRemaining = 0; 2526 } 2527 2528 // only process effects if we're going to write 2529 if (sleepTime == 0 && mType != OFFLOAD) { 2530 for (size_t i = 0; i < effectChains.size(); i ++) { 2531 effectChains[i]->process_l(); 2532 } 2533 } 2534 } 2535 // Process effect chains for offloaded thread even if no audio 2536 // was read from audio track: process only updates effect state 2537 // and thus does have to be synchronized with audio writes but may have 2538 // to be called while waiting for async write callback 2539 if (mType == OFFLOAD) { 2540 for (size_t i = 0; i < effectChains.size(); i ++) { 2541 effectChains[i]->process_l(); 2542 } 2543 } 2544 2545 // Only if the Effects buffer is enabled and there is data in the 2546 // Effects buffer (buffer valid), we need to 2547 // copy into the sink buffer. 2548 // TODO use sleepTime == 0 as an additional condition. 2549 if (mEffectBufferValid) { 2550 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2551 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2552 mNormalFrameCount * mChannelCount); 2553 } 2554 2555 // enable changes in effect chain 2556 unlockEffectChains(effectChains); 2557 2558 if (!waitingAsyncCallback()) { 2559 // sleepTime == 0 means we must write to audio hardware 2560 if (sleepTime == 0) { 2561 if (mBytesRemaining) { 2562 ssize_t ret = threadLoop_write(); 2563 if (ret < 0) { 2564 mBytesRemaining = 0; 2565 } else { 2566 mBytesWritten += ret; 2567 mBytesRemaining -= ret; 2568 } 2569 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2570 (mMixerStatus == MIXER_DRAIN_ALL)) { 2571 threadLoop_drain(); 2572 } 2573 if (mType == MIXER) { 2574 // write blocked detection 2575 nsecs_t now = systemTime(); 2576 nsecs_t delta = now - mLastWriteTime; 2577 if (!mStandby && delta > maxPeriod) { 2578 mNumDelayedWrites++; 2579 if ((now - lastWarning) > kWarningThrottleNs) { 2580 ATRACE_NAME("underrun"); 2581 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2582 ns2ms(delta), mNumDelayedWrites, this); 2583 lastWarning = now; 2584 } 2585 } 2586 } 2587 2588 } else { 2589 usleep(sleepTime); 2590 } 2591 } 2592 2593 // Finally let go of removed track(s), without the lock held 2594 // since we can't guarantee the destructors won't acquire that 2595 // same lock. This will also mutate and push a new fast mixer state. 2596 threadLoop_removeTracks(tracksToRemove); 2597 tracksToRemove.clear(); 2598 2599 // FIXME I don't understand the need for this here; 2600 // it was in the original code but maybe the 2601 // assignment in saveOutputTracks() makes this unnecessary? 2602 clearOutputTracks(); 2603 2604 // Effect chains will be actually deleted here if they were removed from 2605 // mEffectChains list during mixing or effects processing 2606 effectChains.clear(); 2607 2608 // FIXME Note that the above .clear() is no longer necessary since effectChains 2609 // is now local to this block, but will keep it for now (at least until merge done). 2610 } 2611 2612 threadLoop_exit(); 2613 2614 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2615 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2616 // put output stream into standby mode 2617 if (!mStandby) { 2618 mOutput->stream->common.standby(&mOutput->stream->common); 2619 } 2620 } 2621 2622 releaseWakeLock(); 2623 mWakeLockUids.clear(); 2624 mActiveTracksGeneration++; 2625 2626 ALOGV("Thread %p type %d exiting", this, mType); 2627 return false; 2628} 2629 2630// removeTracks_l() must be called with ThreadBase::mLock held 2631void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2632{ 2633 size_t count = tracksToRemove.size(); 2634 if (count > 0) { 2635 for (size_t i=0 ; i<count ; i++) { 2636 const sp<Track>& track = tracksToRemove.itemAt(i); 2637 mActiveTracks.remove(track); 2638 mWakeLockUids.remove(track->uid()); 2639 mActiveTracksGeneration++; 2640 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2641 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2642 if (chain != 0) { 2643 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2644 track->sessionId()); 2645 chain->decActiveTrackCnt(); 2646 } 2647 if (track->isTerminated()) { 2648 removeTrack_l(track); 2649 } 2650 } 2651 } 2652 2653} 2654 2655status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2656{ 2657 if (mNormalSink != 0) { 2658 return mNormalSink->getTimestamp(timestamp); 2659 } 2660 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2661 uint64_t position64; 2662 int ret = mOutput->stream->get_presentation_position( 2663 mOutput->stream, &position64, ×tamp.mTime); 2664 if (ret == 0) { 2665 timestamp.mPosition = (uint32_t)position64; 2666 return NO_ERROR; 2667 } 2668 } 2669 return INVALID_OPERATION; 2670} 2671 2672status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2673 audio_patch_handle_t *handle) 2674{ 2675 status_t status = NO_ERROR; 2676 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2677 // store new device and send to effects 2678 audio_devices_t type = AUDIO_DEVICE_NONE; 2679 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2680 type |= patch->sinks[i].ext.device.type; 2681 } 2682 mOutDevice = type; 2683 for (size_t i = 0; i < mEffectChains.size(); i++) { 2684 mEffectChains[i]->setDevice_l(mOutDevice); 2685 } 2686 2687 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2688 status = hwDevice->create_audio_patch(hwDevice, 2689 patch->num_sources, 2690 patch->sources, 2691 patch->num_sinks, 2692 patch->sinks, 2693 handle); 2694 } else { 2695 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2696 } 2697 return status; 2698} 2699 2700status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2701{ 2702 status_t status = NO_ERROR; 2703 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2704 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2705 status = hwDevice->release_audio_patch(hwDevice, handle); 2706 } else { 2707 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2708 } 2709 return status; 2710} 2711 2712// ---------------------------------------------------------------------------- 2713 2714AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2715 audio_io_handle_t id, audio_devices_t device, type_t type) 2716 : PlaybackThread(audioFlinger, output, id, device, type), 2717 // mAudioMixer below 2718 // mFastMixer below 2719 mFastMixerFutex(0) 2720 // mOutputSink below 2721 // mPipeSink below 2722 // mNormalSink below 2723{ 2724 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2725 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2726 "mFrameCount=%d, mNormalFrameCount=%d", 2727 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2728 mNormalFrameCount); 2729 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2730 2731 // FIXME - Current mixer implementation only supports stereo output 2732 if (mChannelCount != FCC_2) { 2733 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2734 } 2735 2736 // create an NBAIO sink for the HAL output stream, and negotiate 2737 mOutputSink = new AudioStreamOutSink(output->stream); 2738 size_t numCounterOffers = 0; 2739 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2740 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2741 ALOG_ASSERT(index == 0); 2742 2743 // initialize fast mixer depending on configuration 2744 bool initFastMixer; 2745 switch (kUseFastMixer) { 2746 case FastMixer_Never: 2747 initFastMixer = false; 2748 break; 2749 case FastMixer_Always: 2750 initFastMixer = true; 2751 break; 2752 case FastMixer_Static: 2753 case FastMixer_Dynamic: 2754 initFastMixer = mFrameCount < mNormalFrameCount; 2755 break; 2756 } 2757 if (initFastMixer) { 2758 audio_format_t fastMixerFormat; 2759 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2760 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2761 } else { 2762 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2763 } 2764 if (mFormat != fastMixerFormat) { 2765 // change our Sink format to accept our intermediate precision 2766 mFormat = fastMixerFormat; 2767 free(mSinkBuffer); 2768 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2769 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2770 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2771 } 2772 2773 // create a MonoPipe to connect our submix to FastMixer 2774 NBAIO_Format format = mOutputSink->format(); 2775 // adjust format to match that of the Fast Mixer 2776 format.mFormat = fastMixerFormat; 2777 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2778 2779 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2780 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2781 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2782 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2783 const NBAIO_Format offers[1] = {format}; 2784 size_t numCounterOffers = 0; 2785 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2786 ALOG_ASSERT(index == 0); 2787 monoPipe->setAvgFrames((mScreenState & 1) ? 2788 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2789 mPipeSink = monoPipe; 2790 2791#ifdef TEE_SINK 2792 if (mTeeSinkOutputEnabled) { 2793 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2794 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2795 numCounterOffers = 0; 2796 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2797 ALOG_ASSERT(index == 0); 2798 mTeeSink = teeSink; 2799 PipeReader *teeSource = new PipeReader(*teeSink); 2800 numCounterOffers = 0; 2801 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2802 ALOG_ASSERT(index == 0); 2803 mTeeSource = teeSource; 2804 } 2805#endif 2806 2807 // create fast mixer and configure it initially with just one fast track for our submix 2808 mFastMixer = new FastMixer(); 2809 FastMixerStateQueue *sq = mFastMixer->sq(); 2810#ifdef STATE_QUEUE_DUMP 2811 sq->setObserverDump(&mStateQueueObserverDump); 2812 sq->setMutatorDump(&mStateQueueMutatorDump); 2813#endif 2814 FastMixerState *state = sq->begin(); 2815 FastTrack *fastTrack = &state->mFastTracks[0]; 2816 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2817 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2818 fastTrack->mVolumeProvider = NULL; 2819 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2820 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2821 fastTrack->mGeneration++; 2822 state->mFastTracksGen++; 2823 state->mTrackMask = 1; 2824 // fast mixer will use the HAL output sink 2825 state->mOutputSink = mOutputSink.get(); 2826 state->mOutputSinkGen++; 2827 state->mFrameCount = mFrameCount; 2828 state->mCommand = FastMixerState::COLD_IDLE; 2829 // already done in constructor initialization list 2830 //mFastMixerFutex = 0; 2831 state->mColdFutexAddr = &mFastMixerFutex; 2832 state->mColdGen++; 2833 state->mDumpState = &mFastMixerDumpState; 2834#ifdef TEE_SINK 2835 state->mTeeSink = mTeeSink.get(); 2836#endif 2837 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2838 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2839 sq->end(); 2840 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2841 2842 // start the fast mixer 2843 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2844 pid_t tid = mFastMixer->getTid(); 2845 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2846 if (err != 0) { 2847 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2848 kPriorityFastMixer, getpid_cached, tid, err); 2849 } 2850 2851#ifdef AUDIO_WATCHDOG 2852 // create and start the watchdog 2853 mAudioWatchdog = new AudioWatchdog(); 2854 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2855 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2856 tid = mAudioWatchdog->getTid(); 2857 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2858 if (err != 0) { 2859 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2860 kPriorityFastMixer, getpid_cached, tid, err); 2861 } 2862#endif 2863 2864 } else { 2865 mFastMixer = NULL; 2866 } 2867 2868 switch (kUseFastMixer) { 2869 case FastMixer_Never: 2870 case FastMixer_Dynamic: 2871 mNormalSink = mOutputSink; 2872 break; 2873 case FastMixer_Always: 2874 mNormalSink = mPipeSink; 2875 break; 2876 case FastMixer_Static: 2877 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2878 break; 2879 } 2880} 2881 2882AudioFlinger::MixerThread::~MixerThread() 2883{ 2884 if (mFastMixer != NULL) { 2885 FastMixerStateQueue *sq = mFastMixer->sq(); 2886 FastMixerState *state = sq->begin(); 2887 if (state->mCommand == FastMixerState::COLD_IDLE) { 2888 int32_t old = android_atomic_inc(&mFastMixerFutex); 2889 if (old == -1) { 2890 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2891 } 2892 } 2893 state->mCommand = FastMixerState::EXIT; 2894 sq->end(); 2895 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2896 mFastMixer->join(); 2897 // Though the fast mixer thread has exited, it's state queue is still valid. 2898 // We'll use that extract the final state which contains one remaining fast track 2899 // corresponding to our sub-mix. 2900 state = sq->begin(); 2901 ALOG_ASSERT(state->mTrackMask == 1); 2902 FastTrack *fastTrack = &state->mFastTracks[0]; 2903 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2904 delete fastTrack->mBufferProvider; 2905 sq->end(false /*didModify*/); 2906 delete mFastMixer; 2907#ifdef AUDIO_WATCHDOG 2908 if (mAudioWatchdog != 0) { 2909 mAudioWatchdog->requestExit(); 2910 mAudioWatchdog->requestExitAndWait(); 2911 mAudioWatchdog.clear(); 2912 } 2913#endif 2914 } 2915 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2916 delete mAudioMixer; 2917} 2918 2919 2920uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2921{ 2922 if (mFastMixer != NULL) { 2923 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2924 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2925 } 2926 return latency; 2927} 2928 2929 2930void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2931{ 2932 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2933} 2934 2935ssize_t AudioFlinger::MixerThread::threadLoop_write() 2936{ 2937 // FIXME we should only do one push per cycle; confirm this is true 2938 // Start the fast mixer if it's not already running 2939 if (mFastMixer != NULL) { 2940 FastMixerStateQueue *sq = mFastMixer->sq(); 2941 FastMixerState *state = sq->begin(); 2942 if (state->mCommand != FastMixerState::MIX_WRITE && 2943 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2944 if (state->mCommand == FastMixerState::COLD_IDLE) { 2945 int32_t old = android_atomic_inc(&mFastMixerFutex); 2946 if (old == -1) { 2947 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2948 } 2949#ifdef AUDIO_WATCHDOG 2950 if (mAudioWatchdog != 0) { 2951 mAudioWatchdog->resume(); 2952 } 2953#endif 2954 } 2955 state->mCommand = FastMixerState::MIX_WRITE; 2956 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2957 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2958 sq->end(); 2959 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2960 if (kUseFastMixer == FastMixer_Dynamic) { 2961 mNormalSink = mPipeSink; 2962 } 2963 } else { 2964 sq->end(false /*didModify*/); 2965 } 2966 } 2967 return PlaybackThread::threadLoop_write(); 2968} 2969 2970void AudioFlinger::MixerThread::threadLoop_standby() 2971{ 2972 // Idle the fast mixer if it's currently running 2973 if (mFastMixer != NULL) { 2974 FastMixerStateQueue *sq = mFastMixer->sq(); 2975 FastMixerState *state = sq->begin(); 2976 if (!(state->mCommand & FastMixerState::IDLE)) { 2977 state->mCommand = FastMixerState::COLD_IDLE; 2978 state->mColdFutexAddr = &mFastMixerFutex; 2979 state->mColdGen++; 2980 mFastMixerFutex = 0; 2981 sq->end(); 2982 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2983 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2984 if (kUseFastMixer == FastMixer_Dynamic) { 2985 mNormalSink = mOutputSink; 2986 } 2987#ifdef AUDIO_WATCHDOG 2988 if (mAudioWatchdog != 0) { 2989 mAudioWatchdog->pause(); 2990 } 2991#endif 2992 } else { 2993 sq->end(false /*didModify*/); 2994 } 2995 } 2996 PlaybackThread::threadLoop_standby(); 2997} 2998 2999bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3000{ 3001 return false; 3002} 3003 3004bool AudioFlinger::PlaybackThread::shouldStandby_l() 3005{ 3006 return !mStandby; 3007} 3008 3009bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3010{ 3011 Mutex::Autolock _l(mLock); 3012 return waitingAsyncCallback_l(); 3013} 3014 3015// shared by MIXER and DIRECT, overridden by DUPLICATING 3016void AudioFlinger::PlaybackThread::threadLoop_standby() 3017{ 3018 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3019 mOutput->stream->common.standby(&mOutput->stream->common); 3020 if (mUseAsyncWrite != 0) { 3021 // discard any pending drain or write ack by incrementing sequence 3022 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3023 mDrainSequence = (mDrainSequence + 2) & ~1; 3024 ALOG_ASSERT(mCallbackThread != 0); 3025 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3026 mCallbackThread->setDraining(mDrainSequence); 3027 } 3028} 3029 3030void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3031{ 3032 ALOGV("signal playback thread"); 3033 broadcast_l(); 3034} 3035 3036void AudioFlinger::MixerThread::threadLoop_mix() 3037{ 3038 // obtain the presentation timestamp of the next output buffer 3039 int64_t pts; 3040 status_t status = INVALID_OPERATION; 3041 3042 if (mNormalSink != 0) { 3043 status = mNormalSink->getNextWriteTimestamp(&pts); 3044 } else { 3045 status = mOutputSink->getNextWriteTimestamp(&pts); 3046 } 3047 3048 if (status != NO_ERROR) { 3049 pts = AudioBufferProvider::kInvalidPTS; 3050 } 3051 3052 // mix buffers... 3053 mAudioMixer->process(pts); 3054 mCurrentWriteLength = mSinkBufferSize; 3055 // increase sleep time progressively when application underrun condition clears. 3056 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3057 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3058 // such that we would underrun the audio HAL. 3059 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3060 sleepTimeShift--; 3061 } 3062 sleepTime = 0; 3063 standbyTime = systemTime() + standbyDelay; 3064 //TODO: delay standby when effects have a tail 3065} 3066 3067void AudioFlinger::MixerThread::threadLoop_sleepTime() 3068{ 3069 // If no tracks are ready, sleep once for the duration of an output 3070 // buffer size, then write 0s to the output 3071 if (sleepTime == 0) { 3072 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3073 sleepTime = activeSleepTime >> sleepTimeShift; 3074 if (sleepTime < kMinThreadSleepTimeUs) { 3075 sleepTime = kMinThreadSleepTimeUs; 3076 } 3077 // reduce sleep time in case of consecutive application underruns to avoid 3078 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3079 // duration we would end up writing less data than needed by the audio HAL if 3080 // the condition persists. 3081 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3082 sleepTimeShift++; 3083 } 3084 } else { 3085 sleepTime = idleSleepTime; 3086 } 3087 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3088 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3089 // before effects processing or output. 3090 if (mMixerBufferValid) { 3091 memset(mMixerBuffer, 0, mMixerBufferSize); 3092 } else { 3093 memset(mSinkBuffer, 0, mSinkBufferSize); 3094 } 3095 sleepTime = 0; 3096 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3097 "anticipated start"); 3098 } 3099 // TODO add standby time extension fct of effect tail 3100} 3101 3102// prepareTracks_l() must be called with ThreadBase::mLock held 3103AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3104 Vector< sp<Track> > *tracksToRemove) 3105{ 3106 3107 mixer_state mixerStatus = MIXER_IDLE; 3108 // find out which tracks need to be processed 3109 size_t count = mActiveTracks.size(); 3110 size_t mixedTracks = 0; 3111 size_t tracksWithEffect = 0; 3112 // counts only _active_ fast tracks 3113 size_t fastTracks = 0; 3114 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3115 3116 float masterVolume = mMasterVolume; 3117 bool masterMute = mMasterMute; 3118 3119 if (masterMute) { 3120 masterVolume = 0; 3121 } 3122 // Delegate master volume control to effect in output mix effect chain if needed 3123 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3124 if (chain != 0) { 3125 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3126 chain->setVolume_l(&v, &v); 3127 masterVolume = (float)((v + (1 << 23)) >> 24); 3128 chain.clear(); 3129 } 3130 3131 // prepare a new state to push 3132 FastMixerStateQueue *sq = NULL; 3133 FastMixerState *state = NULL; 3134 bool didModify = false; 3135 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3136 if (mFastMixer != NULL) { 3137 sq = mFastMixer->sq(); 3138 state = sq->begin(); 3139 } 3140 3141 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3142 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3143 3144 for (size_t i=0 ; i<count ; i++) { 3145 const sp<Track> t = mActiveTracks[i].promote(); 3146 if (t == 0) { 3147 continue; 3148 } 3149 3150 // this const just means the local variable doesn't change 3151 Track* const track = t.get(); 3152 3153 // process fast tracks 3154 if (track->isFastTrack()) { 3155 3156 // It's theoretically possible (though unlikely) for a fast track to be created 3157 // and then removed within the same normal mix cycle. This is not a problem, as 3158 // the track never becomes active so it's fast mixer slot is never touched. 3159 // The converse, of removing an (active) track and then creating a new track 3160 // at the identical fast mixer slot within the same normal mix cycle, 3161 // is impossible because the slot isn't marked available until the end of each cycle. 3162 int j = track->mFastIndex; 3163 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3164 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3165 FastTrack *fastTrack = &state->mFastTracks[j]; 3166 3167 // Determine whether the track is currently in underrun condition, 3168 // and whether it had a recent underrun. 3169 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3170 FastTrackUnderruns underruns = ftDump->mUnderruns; 3171 uint32_t recentFull = (underruns.mBitFields.mFull - 3172 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3173 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3174 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3175 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3176 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3177 uint32_t recentUnderruns = recentPartial + recentEmpty; 3178 track->mObservedUnderruns = underruns; 3179 // don't count underruns that occur while stopping or pausing 3180 // or stopped which can occur when flush() is called while active 3181 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3182 recentUnderruns > 0) { 3183 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3184 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3185 } 3186 3187 // This is similar to the state machine for normal tracks, 3188 // with a few modifications for fast tracks. 3189 bool isActive = true; 3190 switch (track->mState) { 3191 case TrackBase::STOPPING_1: 3192 // track stays active in STOPPING_1 state until first underrun 3193 if (recentUnderruns > 0 || track->isTerminated()) { 3194 track->mState = TrackBase::STOPPING_2; 3195 } 3196 break; 3197 case TrackBase::PAUSING: 3198 // ramp down is not yet implemented 3199 track->setPaused(); 3200 break; 3201 case TrackBase::RESUMING: 3202 // ramp up is not yet implemented 3203 track->mState = TrackBase::ACTIVE; 3204 break; 3205 case TrackBase::ACTIVE: 3206 if (recentFull > 0 || recentPartial > 0) { 3207 // track has provided at least some frames recently: reset retry count 3208 track->mRetryCount = kMaxTrackRetries; 3209 } 3210 if (recentUnderruns == 0) { 3211 // no recent underruns: stay active 3212 break; 3213 } 3214 // there has recently been an underrun of some kind 3215 if (track->sharedBuffer() == 0) { 3216 // were any of the recent underruns "empty" (no frames available)? 3217 if (recentEmpty == 0) { 3218 // no, then ignore the partial underruns as they are allowed indefinitely 3219 break; 3220 } 3221 // there has recently been an "empty" underrun: decrement the retry counter 3222 if (--(track->mRetryCount) > 0) { 3223 break; 3224 } 3225 // indicate to client process that the track was disabled because of underrun; 3226 // it will then automatically call start() when data is available 3227 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3228 // remove from active list, but state remains ACTIVE [confusing but true] 3229 isActive = false; 3230 break; 3231 } 3232 // fall through 3233 case TrackBase::STOPPING_2: 3234 case TrackBase::PAUSED: 3235 case TrackBase::STOPPED: 3236 case TrackBase::FLUSHED: // flush() while active 3237 // Check for presentation complete if track is inactive 3238 // We have consumed all the buffers of this track. 3239 // This would be incomplete if we auto-paused on underrun 3240 { 3241 size_t audioHALFrames = 3242 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3243 size_t framesWritten = mBytesWritten / mFrameSize; 3244 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3245 // track stays in active list until presentation is complete 3246 break; 3247 } 3248 } 3249 if (track->isStopping_2()) { 3250 track->mState = TrackBase::STOPPED; 3251 } 3252 if (track->isStopped()) { 3253 // Can't reset directly, as fast mixer is still polling this track 3254 // track->reset(); 3255 // So instead mark this track as needing to be reset after push with ack 3256 resetMask |= 1 << i; 3257 } 3258 isActive = false; 3259 break; 3260 case TrackBase::IDLE: 3261 default: 3262 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3263 } 3264 3265 if (isActive) { 3266 // was it previously inactive? 3267 if (!(state->mTrackMask & (1 << j))) { 3268 ExtendedAudioBufferProvider *eabp = track; 3269 VolumeProvider *vp = track; 3270 fastTrack->mBufferProvider = eabp; 3271 fastTrack->mVolumeProvider = vp; 3272 fastTrack->mChannelMask = track->mChannelMask; 3273 fastTrack->mFormat = track->mFormat; 3274 fastTrack->mGeneration++; 3275 state->mTrackMask |= 1 << j; 3276 didModify = true; 3277 // no acknowledgement required for newly active tracks 3278 } 3279 // cache the combined master volume and stream type volume for fast mixer; this 3280 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3281 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3282 ++fastTracks; 3283 } else { 3284 // was it previously active? 3285 if (state->mTrackMask & (1 << j)) { 3286 fastTrack->mBufferProvider = NULL; 3287 fastTrack->mGeneration++; 3288 state->mTrackMask &= ~(1 << j); 3289 didModify = true; 3290 // If any fast tracks were removed, we must wait for acknowledgement 3291 // because we're about to decrement the last sp<> on those tracks. 3292 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3293 } else { 3294 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3295 } 3296 tracksToRemove->add(track); 3297 // Avoids a misleading display in dumpsys 3298 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3299 } 3300 continue; 3301 } 3302 3303 { // local variable scope to avoid goto warning 3304 3305 audio_track_cblk_t* cblk = track->cblk(); 3306 3307 // The first time a track is added we wait 3308 // for all its buffers to be filled before processing it 3309 int name = track->name(); 3310 // make sure that we have enough frames to mix one full buffer. 3311 // enforce this condition only once to enable draining the buffer in case the client 3312 // app does not call stop() and relies on underrun to stop: 3313 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3314 // during last round 3315 size_t desiredFrames; 3316 uint32_t sr = track->sampleRate(); 3317 if (sr == mSampleRate) { 3318 desiredFrames = mNormalFrameCount; 3319 } else { 3320 // +1 for rounding and +1 for additional sample needed for interpolation 3321 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3322 // add frames already consumed but not yet released by the resampler 3323 // because mAudioTrackServerProxy->framesReady() will include these frames 3324 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3325#if 0 3326 // the minimum track buffer size is normally twice the number of frames necessary 3327 // to fill one buffer and the resampler should not leave more than one buffer worth 3328 // of unreleased frames after each pass, but just in case... 3329 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3330#endif 3331 } 3332 uint32_t minFrames = 1; 3333 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3334 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3335 minFrames = desiredFrames; 3336 } 3337 3338 size_t framesReady = track->framesReady(); 3339 if ((framesReady >= minFrames) && track->isReady() && 3340 !track->isPaused() && !track->isTerminated()) 3341 { 3342 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3343 3344 mixedTracks++; 3345 3346 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3347 // there is an effect chain connected to the track 3348 chain.clear(); 3349 if (track->mainBuffer() != mSinkBuffer && 3350 track->mainBuffer() != mMixerBuffer) { 3351 if (mEffectBufferEnabled) { 3352 mEffectBufferValid = true; // Later can set directly. 3353 } 3354 chain = getEffectChain_l(track->sessionId()); 3355 // Delegate volume control to effect in track effect chain if needed 3356 if (chain != 0) { 3357 tracksWithEffect++; 3358 } else { 3359 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3360 "session %d", 3361 name, track->sessionId()); 3362 } 3363 } 3364 3365 3366 int param = AudioMixer::VOLUME; 3367 if (track->mFillingUpStatus == Track::FS_FILLED) { 3368 // no ramp for the first volume setting 3369 track->mFillingUpStatus = Track::FS_ACTIVE; 3370 if (track->mState == TrackBase::RESUMING) { 3371 track->mState = TrackBase::ACTIVE; 3372 param = AudioMixer::RAMP_VOLUME; 3373 } 3374 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3375 // FIXME should not make a decision based on mServer 3376 } else if (cblk->mServer != 0) { 3377 // If the track is stopped before the first frame was mixed, 3378 // do not apply ramp 3379 param = AudioMixer::RAMP_VOLUME; 3380 } 3381 3382 // compute volume for this track 3383 uint32_t vl, vr; // in U8.24 integer format 3384 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3385 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3386 vl = vr = 0; 3387 vlf = vrf = vaf = 0.; 3388 if (track->isPausing()) { 3389 track->setPaused(); 3390 } 3391 } else { 3392 3393 // read original volumes with volume control 3394 float typeVolume = mStreamTypes[track->streamType()].volume; 3395 float v = masterVolume * typeVolume; 3396 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3397 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3398 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3399 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3400 // track volumes come from shared memory, so can't be trusted and must be clamped 3401 if (vlf > GAIN_FLOAT_UNITY) { 3402 ALOGV("Track left volume out of range: %.3g", vlf); 3403 vlf = GAIN_FLOAT_UNITY; 3404 } 3405 if (vrf > GAIN_FLOAT_UNITY) { 3406 ALOGV("Track right volume out of range: %.3g", vrf); 3407 vrf = GAIN_FLOAT_UNITY; 3408 } 3409 // now apply the master volume and stream type volume 3410 vlf *= v; 3411 vrf *= v; 3412 // assuming master volume and stream type volume each go up to 1.0, 3413 // then derive vl and vr as U8.24 versions for the effect chain 3414 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3415 vl = (uint32_t) (scaleto8_24 * vlf); 3416 vr = (uint32_t) (scaleto8_24 * vrf); 3417 // vl and vr are now in U8.24 format 3418 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3419 // send level comes from shared memory and so may be corrupt 3420 if (sendLevel > MAX_GAIN_INT) { 3421 ALOGV("Track send level out of range: %04X", sendLevel); 3422 sendLevel = MAX_GAIN_INT; 3423 } 3424 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3425 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3426 } 3427 3428 // Delegate volume control to effect in track effect chain if needed 3429 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3430 // Do not ramp volume if volume is controlled by effect 3431 param = AudioMixer::VOLUME; 3432 track->mHasVolumeController = true; 3433 } else { 3434 // force no volume ramp when volume controller was just disabled or removed 3435 // from effect chain to avoid volume spike 3436 if (track->mHasVolumeController) { 3437 param = AudioMixer::VOLUME; 3438 } 3439 track->mHasVolumeController = false; 3440 } 3441 3442 // XXX: these things DON'T need to be done each time 3443 mAudioMixer->setBufferProvider(name, track); 3444 mAudioMixer->enable(name); 3445 3446 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3447 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3448 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3449 mAudioMixer->setParameter( 3450 name, 3451 AudioMixer::TRACK, 3452 AudioMixer::FORMAT, (void *)track->format()); 3453 mAudioMixer->setParameter( 3454 name, 3455 AudioMixer::TRACK, 3456 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3457 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3458 uint32_t maxSampleRate = mSampleRate * 2; 3459 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3460 if (reqSampleRate == 0) { 3461 reqSampleRate = mSampleRate; 3462 } else if (reqSampleRate > maxSampleRate) { 3463 reqSampleRate = maxSampleRate; 3464 } 3465 mAudioMixer->setParameter( 3466 name, 3467 AudioMixer::RESAMPLE, 3468 AudioMixer::SAMPLE_RATE, 3469 (void *)(uintptr_t)reqSampleRate); 3470 /* 3471 * Select the appropriate output buffer for the track. 3472 * 3473 * Tracks with effects go into their own effects chain buffer 3474 * and from there into either mEffectBuffer or mSinkBuffer. 3475 * 3476 * Other tracks can use mMixerBuffer for higher precision 3477 * channel accumulation. If this buffer is enabled 3478 * (mMixerBufferEnabled true), then selected tracks will accumulate 3479 * into it. 3480 * 3481 */ 3482 if (mMixerBufferEnabled 3483 && (track->mainBuffer() == mSinkBuffer 3484 || track->mainBuffer() == mMixerBuffer)) { 3485 mAudioMixer->setParameter( 3486 name, 3487 AudioMixer::TRACK, 3488 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3489 mAudioMixer->setParameter( 3490 name, 3491 AudioMixer::TRACK, 3492 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3493 // TODO: override track->mainBuffer()? 3494 mMixerBufferValid = true; 3495 } else { 3496 mAudioMixer->setParameter( 3497 name, 3498 AudioMixer::TRACK, 3499 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3500 mAudioMixer->setParameter( 3501 name, 3502 AudioMixer::TRACK, 3503 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3504 } 3505 mAudioMixer->setParameter( 3506 name, 3507 AudioMixer::TRACK, 3508 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3509 3510 // reset retry count 3511 track->mRetryCount = kMaxTrackRetries; 3512 3513 // If one track is ready, set the mixer ready if: 3514 // - the mixer was not ready during previous round OR 3515 // - no other track is not ready 3516 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3517 mixerStatus != MIXER_TRACKS_ENABLED) { 3518 mixerStatus = MIXER_TRACKS_READY; 3519 } 3520 } else { 3521 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3522 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3523 } 3524 // clear effect chain input buffer if an active track underruns to avoid sending 3525 // previous audio buffer again to effects 3526 chain = getEffectChain_l(track->sessionId()); 3527 if (chain != 0) { 3528 chain->clearInputBuffer(); 3529 } 3530 3531 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3532 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3533 track->isStopped() || track->isPaused()) { 3534 // We have consumed all the buffers of this track. 3535 // Remove it from the list of active tracks. 3536 // TODO: use actual buffer filling status instead of latency when available from 3537 // audio HAL 3538 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3539 size_t framesWritten = mBytesWritten / mFrameSize; 3540 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3541 if (track->isStopped()) { 3542 track->reset(); 3543 } 3544 tracksToRemove->add(track); 3545 } 3546 } else { 3547 // No buffers for this track. Give it a few chances to 3548 // fill a buffer, then remove it from active list. 3549 if (--(track->mRetryCount) <= 0) { 3550 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3551 tracksToRemove->add(track); 3552 // indicate to client process that the track was disabled because of underrun; 3553 // it will then automatically call start() when data is available 3554 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3555 // If one track is not ready, mark the mixer also not ready if: 3556 // - the mixer was ready during previous round OR 3557 // - no other track is ready 3558 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3559 mixerStatus != MIXER_TRACKS_READY) { 3560 mixerStatus = MIXER_TRACKS_ENABLED; 3561 } 3562 } 3563 mAudioMixer->disable(name); 3564 } 3565 3566 } // local variable scope to avoid goto warning 3567track_is_ready: ; 3568 3569 } 3570 3571 // Push the new FastMixer state if necessary 3572 bool pauseAudioWatchdog = false; 3573 if (didModify) { 3574 state->mFastTracksGen++; 3575 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3576 if (kUseFastMixer == FastMixer_Dynamic && 3577 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3578 state->mCommand = FastMixerState::COLD_IDLE; 3579 state->mColdFutexAddr = &mFastMixerFutex; 3580 state->mColdGen++; 3581 mFastMixerFutex = 0; 3582 if (kUseFastMixer == FastMixer_Dynamic) { 3583 mNormalSink = mOutputSink; 3584 } 3585 // If we go into cold idle, need to wait for acknowledgement 3586 // so that fast mixer stops doing I/O. 3587 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3588 pauseAudioWatchdog = true; 3589 } 3590 } 3591 if (sq != NULL) { 3592 sq->end(didModify); 3593 sq->push(block); 3594 } 3595#ifdef AUDIO_WATCHDOG 3596 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3597 mAudioWatchdog->pause(); 3598 } 3599#endif 3600 3601 // Now perform the deferred reset on fast tracks that have stopped 3602 while (resetMask != 0) { 3603 size_t i = __builtin_ctz(resetMask); 3604 ALOG_ASSERT(i < count); 3605 resetMask &= ~(1 << i); 3606 sp<Track> t = mActiveTracks[i].promote(); 3607 if (t == 0) { 3608 continue; 3609 } 3610 Track* track = t.get(); 3611 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3612 track->reset(); 3613 } 3614 3615 // remove all the tracks that need to be... 3616 removeTracks_l(*tracksToRemove); 3617 3618 // sink or mix buffer must be cleared if all tracks are connected to an 3619 // effect chain as in this case the mixer will not write to the sink or mix buffer 3620 // and track effects will accumulate into it 3621 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3622 (mixedTracks == 0 && fastTracks > 0))) { 3623 // FIXME as a performance optimization, should remember previous zero status 3624 if (mMixerBufferValid) { 3625 memset(mMixerBuffer, 0, mMixerBufferSize); 3626 // TODO: In testing, mSinkBuffer below need not be cleared because 3627 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3628 // after mixing. 3629 // 3630 // To enforce this guarantee: 3631 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3632 // (mixedTracks == 0 && fastTracks > 0)) 3633 // must imply MIXER_TRACKS_READY. 3634 // Later, we may clear buffers regardless, and skip much of this logic. 3635 } 3636 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3637 if (mEffectBufferValid) { 3638 memset(mEffectBuffer, 0, mEffectBufferSize); 3639 } 3640 // FIXME as a performance optimization, should remember previous zero status 3641 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3642 } 3643 3644 // if any fast tracks, then status is ready 3645 mMixerStatusIgnoringFastTracks = mixerStatus; 3646 if (fastTracks > 0) { 3647 mixerStatus = MIXER_TRACKS_READY; 3648 } 3649 return mixerStatus; 3650} 3651 3652// getTrackName_l() must be called with ThreadBase::mLock held 3653int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3654 audio_format_t format, int sessionId) 3655{ 3656 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3657} 3658 3659// deleteTrackName_l() must be called with ThreadBase::mLock held 3660void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3661{ 3662 ALOGV("remove track (%d) and delete from mixer", name); 3663 mAudioMixer->deleteTrackName(name); 3664} 3665 3666// checkForNewParameter_l() must be called with ThreadBase::mLock held 3667bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3668 status_t& status) 3669{ 3670 bool reconfig = false; 3671 3672 status = NO_ERROR; 3673 3674 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3675 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3676 if (mFastMixer != NULL) { 3677 FastMixerStateQueue *sq = mFastMixer->sq(); 3678 FastMixerState *state = sq->begin(); 3679 if (!(state->mCommand & FastMixerState::IDLE)) { 3680 previousCommand = state->mCommand; 3681 state->mCommand = FastMixerState::HOT_IDLE; 3682 sq->end(); 3683 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3684 } else { 3685 sq->end(false /*didModify*/); 3686 } 3687 } 3688 3689 AudioParameter param = AudioParameter(keyValuePair); 3690 int value; 3691 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3692 reconfig = true; 3693 } 3694 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3695 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3696 status = BAD_VALUE; 3697 } else { 3698 // no need to save value, since it's constant 3699 reconfig = true; 3700 } 3701 } 3702 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3703 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3704 status = BAD_VALUE; 3705 } else { 3706 // no need to save value, since it's constant 3707 reconfig = true; 3708 } 3709 } 3710 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3711 // do not accept frame count changes if tracks are open as the track buffer 3712 // size depends on frame count and correct behavior would not be guaranteed 3713 // if frame count is changed after track creation 3714 if (!mTracks.isEmpty()) { 3715 status = INVALID_OPERATION; 3716 } else { 3717 reconfig = true; 3718 } 3719 } 3720 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3721#ifdef ADD_BATTERY_DATA 3722 // when changing the audio output device, call addBatteryData to notify 3723 // the change 3724 if (mOutDevice != value) { 3725 uint32_t params = 0; 3726 // check whether speaker is on 3727 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3728 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3729 } 3730 3731 audio_devices_t deviceWithoutSpeaker 3732 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3733 // check if any other device (except speaker) is on 3734 if (value & deviceWithoutSpeaker ) { 3735 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3736 } 3737 3738 if (params != 0) { 3739 addBatteryData(params); 3740 } 3741 } 3742#endif 3743 3744 // forward device change to effects that have requested to be 3745 // aware of attached audio device. 3746 if (value != AUDIO_DEVICE_NONE) { 3747 mOutDevice = value; 3748 for (size_t i = 0; i < mEffectChains.size(); i++) { 3749 mEffectChains[i]->setDevice_l(mOutDevice); 3750 } 3751 } 3752 } 3753 3754 if (status == NO_ERROR) { 3755 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3756 keyValuePair.string()); 3757 if (!mStandby && status == INVALID_OPERATION) { 3758 mOutput->stream->common.standby(&mOutput->stream->common); 3759 mStandby = true; 3760 mBytesWritten = 0; 3761 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3762 keyValuePair.string()); 3763 } 3764 if (status == NO_ERROR && reconfig) { 3765 readOutputParameters_l(); 3766 delete mAudioMixer; 3767 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3768 for (size_t i = 0; i < mTracks.size() ; i++) { 3769 int name = getTrackName_l(mTracks[i]->mChannelMask, 3770 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3771 if (name < 0) { 3772 break; 3773 } 3774 mTracks[i]->mName = name; 3775 } 3776 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3777 } 3778 } 3779 3780 if (!(previousCommand & FastMixerState::IDLE)) { 3781 ALOG_ASSERT(mFastMixer != NULL); 3782 FastMixerStateQueue *sq = mFastMixer->sq(); 3783 FastMixerState *state = sq->begin(); 3784 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3785 state->mCommand = previousCommand; 3786 sq->end(); 3787 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3788 } 3789 3790 return reconfig; 3791} 3792 3793 3794void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3795{ 3796 const size_t SIZE = 256; 3797 char buffer[SIZE]; 3798 String8 result; 3799 3800 PlaybackThread::dumpInternals(fd, args); 3801 3802 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3803 3804 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3805 const FastMixerDumpState copy(mFastMixerDumpState); 3806 copy.dump(fd); 3807 3808#ifdef STATE_QUEUE_DUMP 3809 // Similar for state queue 3810 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3811 observerCopy.dump(fd); 3812 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3813 mutatorCopy.dump(fd); 3814#endif 3815 3816#ifdef TEE_SINK 3817 // Write the tee output to a .wav file 3818 dumpTee(fd, mTeeSource, mId); 3819#endif 3820 3821#ifdef AUDIO_WATCHDOG 3822 if (mAudioWatchdog != 0) { 3823 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3824 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3825 wdCopy.dump(fd); 3826 } 3827#endif 3828} 3829 3830uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3831{ 3832 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3833} 3834 3835uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3836{ 3837 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3838} 3839 3840void AudioFlinger::MixerThread::cacheParameters_l() 3841{ 3842 PlaybackThread::cacheParameters_l(); 3843 3844 // FIXME: Relaxed timing because of a certain device that can't meet latency 3845 // Should be reduced to 2x after the vendor fixes the driver issue 3846 // increase threshold again due to low power audio mode. The way this warning 3847 // threshold is calculated and its usefulness should be reconsidered anyway. 3848 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3849} 3850 3851// ---------------------------------------------------------------------------- 3852 3853AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3854 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3855 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3856 // mLeftVolFloat, mRightVolFloat 3857{ 3858} 3859 3860AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3861 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3862 ThreadBase::type_t type) 3863 : PlaybackThread(audioFlinger, output, id, device, type) 3864 // mLeftVolFloat, mRightVolFloat 3865{ 3866} 3867 3868AudioFlinger::DirectOutputThread::~DirectOutputThread() 3869{ 3870} 3871 3872void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3873{ 3874 audio_track_cblk_t* cblk = track->cblk(); 3875 float left, right; 3876 3877 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3878 left = right = 0; 3879 } else { 3880 float typeVolume = mStreamTypes[track->streamType()].volume; 3881 float v = mMasterVolume * typeVolume; 3882 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3883 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3884 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3885 if (left > GAIN_FLOAT_UNITY) { 3886 left = GAIN_FLOAT_UNITY; 3887 } 3888 left *= v; 3889 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3890 if (right > GAIN_FLOAT_UNITY) { 3891 right = GAIN_FLOAT_UNITY; 3892 } 3893 right *= v; 3894 } 3895 3896 if (lastTrack) { 3897 if (left != mLeftVolFloat || right != mRightVolFloat) { 3898 mLeftVolFloat = left; 3899 mRightVolFloat = right; 3900 3901 // Convert volumes from float to 8.24 3902 uint32_t vl = (uint32_t)(left * (1 << 24)); 3903 uint32_t vr = (uint32_t)(right * (1 << 24)); 3904 3905 // Delegate volume control to effect in track effect chain if needed 3906 // only one effect chain can be present on DirectOutputThread, so if 3907 // there is one, the track is connected to it 3908 if (!mEffectChains.isEmpty()) { 3909 mEffectChains[0]->setVolume_l(&vl, &vr); 3910 left = (float)vl / (1 << 24); 3911 right = (float)vr / (1 << 24); 3912 } 3913 if (mOutput->stream->set_volume) { 3914 mOutput->stream->set_volume(mOutput->stream, left, right); 3915 } 3916 } 3917 } 3918} 3919 3920 3921AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3922 Vector< sp<Track> > *tracksToRemove 3923) 3924{ 3925 size_t count = mActiveTracks.size(); 3926 mixer_state mixerStatus = MIXER_IDLE; 3927 3928 // find out which tracks need to be processed 3929 for (size_t i = 0; i < count; i++) { 3930 sp<Track> t = mActiveTracks[i].promote(); 3931 // The track died recently 3932 if (t == 0) { 3933 continue; 3934 } 3935 3936 Track* const track = t.get(); 3937 audio_track_cblk_t* cblk = track->cblk(); 3938 // Only consider last track started for volume and mixer state control. 3939 // In theory an older track could underrun and restart after the new one starts 3940 // but as we only care about the transition phase between two tracks on a 3941 // direct output, it is not a problem to ignore the underrun case. 3942 sp<Track> l = mLatestActiveTrack.promote(); 3943 bool last = l.get() == track; 3944 3945 // The first time a track is added we wait 3946 // for all its buffers to be filled before processing it 3947 uint32_t minFrames; 3948 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3949 minFrames = mNormalFrameCount; 3950 } else { 3951 minFrames = 1; 3952 } 3953 3954 if ((track->framesReady() >= minFrames) && track->isReady() && 3955 !track->isPaused() && !track->isTerminated()) 3956 { 3957 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3958 3959 if (track->mFillingUpStatus == Track::FS_FILLED) { 3960 track->mFillingUpStatus = Track::FS_ACTIVE; 3961 // make sure processVolume_l() will apply new volume even if 0 3962 mLeftVolFloat = mRightVolFloat = -1.0; 3963 if (track->mState == TrackBase::RESUMING) { 3964 track->mState = TrackBase::ACTIVE; 3965 } 3966 } 3967 3968 // compute volume for this track 3969 processVolume_l(track, last); 3970 if (last) { 3971 // reset retry count 3972 track->mRetryCount = kMaxTrackRetriesDirect; 3973 mActiveTrack = t; 3974 mixerStatus = MIXER_TRACKS_READY; 3975 } 3976 } else { 3977 // clear effect chain input buffer if the last active track started underruns 3978 // to avoid sending previous audio buffer again to effects 3979 if (!mEffectChains.isEmpty() && last) { 3980 mEffectChains[0]->clearInputBuffer(); 3981 } 3982 3983 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3984 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3985 track->isStopped() || track->isPaused()) { 3986 // We have consumed all the buffers of this track. 3987 // Remove it from the list of active tracks. 3988 // TODO: implement behavior for compressed audio 3989 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3990 size_t framesWritten = mBytesWritten / mFrameSize; 3991 if (mStandby || !last || 3992 track->presentationComplete(framesWritten, audioHALFrames)) { 3993 if (track->isStopped()) { 3994 track->reset(); 3995 } 3996 tracksToRemove->add(track); 3997 } 3998 } else { 3999 // No buffers for this track. Give it a few chances to 4000 // fill a buffer, then remove it from active list. 4001 // Only consider last track started for mixer state control 4002 if (--(track->mRetryCount) <= 0) { 4003 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4004 tracksToRemove->add(track); 4005 // indicate to client process that the track was disabled because of underrun; 4006 // it will then automatically call start() when data is available 4007 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4008 } else if (last) { 4009 mixerStatus = MIXER_TRACKS_ENABLED; 4010 } 4011 } 4012 } 4013 } 4014 4015 // remove all the tracks that need to be... 4016 removeTracks_l(*tracksToRemove); 4017 4018 return mixerStatus; 4019} 4020 4021void AudioFlinger::DirectOutputThread::threadLoop_mix() 4022{ 4023 size_t frameCount = mFrameCount; 4024 int8_t *curBuf = (int8_t *)mSinkBuffer; 4025 // output audio to hardware 4026 while (frameCount) { 4027 AudioBufferProvider::Buffer buffer; 4028 buffer.frameCount = frameCount; 4029 mActiveTrack->getNextBuffer(&buffer); 4030 if (buffer.raw == NULL) { 4031 memset(curBuf, 0, frameCount * mFrameSize); 4032 break; 4033 } 4034 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4035 frameCount -= buffer.frameCount; 4036 curBuf += buffer.frameCount * mFrameSize; 4037 mActiveTrack->releaseBuffer(&buffer); 4038 } 4039 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4040 sleepTime = 0; 4041 standbyTime = systemTime() + standbyDelay; 4042 mActiveTrack.clear(); 4043} 4044 4045void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4046{ 4047 if (sleepTime == 0) { 4048 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4049 sleepTime = activeSleepTime; 4050 } else { 4051 sleepTime = idleSleepTime; 4052 } 4053 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4054 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4055 sleepTime = 0; 4056 } 4057} 4058 4059// getTrackName_l() must be called with ThreadBase::mLock held 4060int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4061 audio_format_t format __unused, int sessionId __unused) 4062{ 4063 return 0; 4064} 4065 4066// deleteTrackName_l() must be called with ThreadBase::mLock held 4067void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4068{ 4069} 4070 4071// checkForNewParameter_l() must be called with ThreadBase::mLock held 4072bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4073 status_t& status) 4074{ 4075 bool reconfig = false; 4076 4077 status = NO_ERROR; 4078 4079 AudioParameter param = AudioParameter(keyValuePair); 4080 int value; 4081 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4082 // forward device change to effects that have requested to be 4083 // aware of attached audio device. 4084 if (value != AUDIO_DEVICE_NONE) { 4085 mOutDevice = value; 4086 for (size_t i = 0; i < mEffectChains.size(); i++) { 4087 mEffectChains[i]->setDevice_l(mOutDevice); 4088 } 4089 } 4090 } 4091 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4092 // do not accept frame count changes if tracks are open as the track buffer 4093 // size depends on frame count and correct behavior would not be garantied 4094 // if frame count is changed after track creation 4095 if (!mTracks.isEmpty()) { 4096 status = INVALID_OPERATION; 4097 } else { 4098 reconfig = true; 4099 } 4100 } 4101 if (status == NO_ERROR) { 4102 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4103 keyValuePair.string()); 4104 if (!mStandby && status == INVALID_OPERATION) { 4105 mOutput->stream->common.standby(&mOutput->stream->common); 4106 mStandby = true; 4107 mBytesWritten = 0; 4108 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4109 keyValuePair.string()); 4110 } 4111 if (status == NO_ERROR && reconfig) { 4112 readOutputParameters_l(); 4113 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4114 } 4115 } 4116 4117 return reconfig; 4118} 4119 4120uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4121{ 4122 uint32_t time; 4123 if (audio_is_linear_pcm(mFormat)) { 4124 time = PlaybackThread::activeSleepTimeUs(); 4125 } else { 4126 time = 10000; 4127 } 4128 return time; 4129} 4130 4131uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4132{ 4133 uint32_t time; 4134 if (audio_is_linear_pcm(mFormat)) { 4135 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4136 } else { 4137 time = 10000; 4138 } 4139 return time; 4140} 4141 4142uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4143{ 4144 uint32_t time; 4145 if (audio_is_linear_pcm(mFormat)) { 4146 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4147 } else { 4148 time = 10000; 4149 } 4150 return time; 4151} 4152 4153void AudioFlinger::DirectOutputThread::cacheParameters_l() 4154{ 4155 PlaybackThread::cacheParameters_l(); 4156 4157 // use shorter standby delay as on normal output to release 4158 // hardware resources as soon as possible 4159 if (audio_is_linear_pcm(mFormat)) { 4160 standbyDelay = microseconds(activeSleepTime*2); 4161 } else { 4162 standbyDelay = kOffloadStandbyDelayNs; 4163 } 4164} 4165 4166// ---------------------------------------------------------------------------- 4167 4168AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4169 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4170 : Thread(false /*canCallJava*/), 4171 mPlaybackThread(playbackThread), 4172 mWriteAckSequence(0), 4173 mDrainSequence(0) 4174{ 4175} 4176 4177AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4178{ 4179} 4180 4181void AudioFlinger::AsyncCallbackThread::onFirstRef() 4182{ 4183 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4184} 4185 4186bool AudioFlinger::AsyncCallbackThread::threadLoop() 4187{ 4188 while (!exitPending()) { 4189 uint32_t writeAckSequence; 4190 uint32_t drainSequence; 4191 4192 { 4193 Mutex::Autolock _l(mLock); 4194 while (!((mWriteAckSequence & 1) || 4195 (mDrainSequence & 1) || 4196 exitPending())) { 4197 mWaitWorkCV.wait(mLock); 4198 } 4199 4200 if (exitPending()) { 4201 break; 4202 } 4203 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4204 mWriteAckSequence, mDrainSequence); 4205 writeAckSequence = mWriteAckSequence; 4206 mWriteAckSequence &= ~1; 4207 drainSequence = mDrainSequence; 4208 mDrainSequence &= ~1; 4209 } 4210 { 4211 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4212 if (playbackThread != 0) { 4213 if (writeAckSequence & 1) { 4214 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4215 } 4216 if (drainSequence & 1) { 4217 playbackThread->resetDraining(drainSequence >> 1); 4218 } 4219 } 4220 } 4221 } 4222 return false; 4223} 4224 4225void AudioFlinger::AsyncCallbackThread::exit() 4226{ 4227 ALOGV("AsyncCallbackThread::exit"); 4228 Mutex::Autolock _l(mLock); 4229 requestExit(); 4230 mWaitWorkCV.broadcast(); 4231} 4232 4233void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4234{ 4235 Mutex::Autolock _l(mLock); 4236 // bit 0 is cleared 4237 mWriteAckSequence = sequence << 1; 4238} 4239 4240void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4241{ 4242 Mutex::Autolock _l(mLock); 4243 // ignore unexpected callbacks 4244 if (mWriteAckSequence & 2) { 4245 mWriteAckSequence |= 1; 4246 mWaitWorkCV.signal(); 4247 } 4248} 4249 4250void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4251{ 4252 Mutex::Autolock _l(mLock); 4253 // bit 0 is cleared 4254 mDrainSequence = sequence << 1; 4255} 4256 4257void AudioFlinger::AsyncCallbackThread::resetDraining() 4258{ 4259 Mutex::Autolock _l(mLock); 4260 // ignore unexpected callbacks 4261 if (mDrainSequence & 2) { 4262 mDrainSequence |= 1; 4263 mWaitWorkCV.signal(); 4264 } 4265} 4266 4267 4268// ---------------------------------------------------------------------------- 4269AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4270 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4271 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4272 mHwPaused(false), 4273 mFlushPending(false), 4274 mPausedBytesRemaining(0) 4275{ 4276 //FIXME: mStandby should be set to true by ThreadBase constructor 4277 mStandby = true; 4278} 4279 4280void AudioFlinger::OffloadThread::threadLoop_exit() 4281{ 4282 if (mFlushPending || mHwPaused) { 4283 // If a flush is pending or track was paused, just discard buffered data 4284 flushHw_l(); 4285 } else { 4286 mMixerStatus = MIXER_DRAIN_ALL; 4287 threadLoop_drain(); 4288 } 4289 if (mUseAsyncWrite) { 4290 ALOG_ASSERT(mCallbackThread != 0); 4291 mCallbackThread->exit(); 4292 } 4293 PlaybackThread::threadLoop_exit(); 4294} 4295 4296AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4297 Vector< sp<Track> > *tracksToRemove 4298) 4299{ 4300 size_t count = mActiveTracks.size(); 4301 4302 mixer_state mixerStatus = MIXER_IDLE; 4303 bool doHwPause = false; 4304 bool doHwResume = false; 4305 4306 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4307 4308 // find out which tracks need to be processed 4309 for (size_t i = 0; i < count; i++) { 4310 sp<Track> t = mActiveTracks[i].promote(); 4311 // The track died recently 4312 if (t == 0) { 4313 continue; 4314 } 4315 Track* const track = t.get(); 4316 audio_track_cblk_t* cblk = track->cblk(); 4317 // Only consider last track started for volume and mixer state control. 4318 // In theory an older track could underrun and restart after the new one starts 4319 // but as we only care about the transition phase between two tracks on a 4320 // direct output, it is not a problem to ignore the underrun case. 4321 sp<Track> l = mLatestActiveTrack.promote(); 4322 bool last = l.get() == track; 4323 4324 if (track->isInvalid()) { 4325 ALOGW("An invalidated track shouldn't be in active list"); 4326 tracksToRemove->add(track); 4327 continue; 4328 } 4329 4330 if (track->mState == TrackBase::IDLE) { 4331 ALOGW("An idle track shouldn't be in active list"); 4332 continue; 4333 } 4334 4335 if (track->isPausing()) { 4336 track->setPaused(); 4337 if (last) { 4338 if (!mHwPaused) { 4339 doHwPause = true; 4340 mHwPaused = true; 4341 } 4342 // If we were part way through writing the mixbuffer to 4343 // the HAL we must save this until we resume 4344 // BUG - this will be wrong if a different track is made active, 4345 // in that case we want to discard the pending data in the 4346 // mixbuffer and tell the client to present it again when the 4347 // track is resumed 4348 mPausedWriteLength = mCurrentWriteLength; 4349 mPausedBytesRemaining = mBytesRemaining; 4350 mBytesRemaining = 0; // stop writing 4351 } 4352 tracksToRemove->add(track); 4353 } else if (track->isFlushPending()) { 4354 track->flushAck(); 4355 if (last) { 4356 mFlushPending = true; 4357 } 4358 } else if (track->isResumePending()){ 4359 track->resumeAck(); 4360 if (last) { 4361 if (mPausedBytesRemaining) { 4362 // Need to continue write that was interrupted 4363 mCurrentWriteLength = mPausedWriteLength; 4364 mBytesRemaining = mPausedBytesRemaining; 4365 mPausedBytesRemaining = 0; 4366 } 4367 if (mHwPaused) { 4368 doHwResume = true; 4369 mHwPaused = false; 4370 // threadLoop_mix() will handle the case that we need to 4371 // resume an interrupted write 4372 } 4373 // enable write to audio HAL 4374 sleepTime = 0; 4375 4376 // Do not handle new data in this iteration even if track->framesReady() 4377 mixerStatus = MIXER_TRACKS_ENABLED; 4378 } 4379 } else if (track->framesReady() && track->isReady() && 4380 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4381 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4382 if (track->mFillingUpStatus == Track::FS_FILLED) { 4383 track->mFillingUpStatus = Track::FS_ACTIVE; 4384 // make sure processVolume_l() will apply new volume even if 0 4385 mLeftVolFloat = mRightVolFloat = -1.0; 4386 } 4387 4388 if (last) { 4389 sp<Track> previousTrack = mPreviousTrack.promote(); 4390 if (previousTrack != 0) { 4391 if (track != previousTrack.get()) { 4392 // Flush any data still being written from last track 4393 mBytesRemaining = 0; 4394 if (mPausedBytesRemaining) { 4395 // Last track was paused so we also need to flush saved 4396 // mixbuffer state and invalidate track so that it will 4397 // re-submit that unwritten data when it is next resumed 4398 mPausedBytesRemaining = 0; 4399 // Invalidate is a bit drastic - would be more efficient 4400 // to have a flag to tell client that some of the 4401 // previously written data was lost 4402 previousTrack->invalidate(); 4403 } 4404 // flush data already sent to the DSP if changing audio session as audio 4405 // comes from a different source. Also invalidate previous track to force a 4406 // seek when resuming. 4407 if (previousTrack->sessionId() != track->sessionId()) { 4408 previousTrack->invalidate(); 4409 } 4410 } 4411 } 4412 mPreviousTrack = track; 4413 // reset retry count 4414 track->mRetryCount = kMaxTrackRetriesOffload; 4415 mActiveTrack = t; 4416 mixerStatus = MIXER_TRACKS_READY; 4417 } 4418 } else { 4419 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4420 if (track->isStopping_1()) { 4421 // Hardware buffer can hold a large amount of audio so we must 4422 // wait for all current track's data to drain before we say 4423 // that the track is stopped. 4424 if (mBytesRemaining == 0) { 4425 // Only start draining when all data in mixbuffer 4426 // has been written 4427 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4428 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4429 // do not drain if no data was ever sent to HAL (mStandby == true) 4430 if (last && !mStandby) { 4431 // do not modify drain sequence if we are already draining. This happens 4432 // when resuming from pause after drain. 4433 if ((mDrainSequence & 1) == 0) { 4434 sleepTime = 0; 4435 standbyTime = systemTime() + standbyDelay; 4436 mixerStatus = MIXER_DRAIN_TRACK; 4437 mDrainSequence += 2; 4438 } 4439 if (mHwPaused) { 4440 // It is possible to move from PAUSED to STOPPING_1 without 4441 // a resume so we must ensure hardware is running 4442 doHwResume = true; 4443 mHwPaused = false; 4444 } 4445 } 4446 } 4447 } else if (track->isStopping_2()) { 4448 // Drain has completed or we are in standby, signal presentation complete 4449 if (!(mDrainSequence & 1) || !last || mStandby) { 4450 track->mState = TrackBase::STOPPED; 4451 size_t audioHALFrames = 4452 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4453 size_t framesWritten = 4454 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4455 track->presentationComplete(framesWritten, audioHALFrames); 4456 track->reset(); 4457 tracksToRemove->add(track); 4458 } 4459 } else { 4460 // No buffers for this track. Give it a few chances to 4461 // fill a buffer, then remove it from active list. 4462 if (--(track->mRetryCount) <= 0) { 4463 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4464 track->name()); 4465 tracksToRemove->add(track); 4466 // indicate to client process that the track was disabled because of underrun; 4467 // it will then automatically call start() when data is available 4468 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4469 } else if (last){ 4470 mixerStatus = MIXER_TRACKS_ENABLED; 4471 } 4472 } 4473 } 4474 // compute volume for this track 4475 processVolume_l(track, last); 4476 } 4477 4478 // make sure the pause/flush/resume sequence is executed in the right order. 4479 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4480 // before flush and then resume HW. This can happen in case of pause/flush/resume 4481 // if resume is received before pause is executed. 4482 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4483 mOutput->stream->pause(mOutput->stream); 4484 } 4485 if (mFlushPending) { 4486 flushHw_l(); 4487 mFlushPending = false; 4488 } 4489 if (!mStandby && doHwResume) { 4490 mOutput->stream->resume(mOutput->stream); 4491 } 4492 4493 // remove all the tracks that need to be... 4494 removeTracks_l(*tracksToRemove); 4495 4496 return mixerStatus; 4497} 4498 4499// must be called with thread mutex locked 4500bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4501{ 4502 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4503 mWriteAckSequence, mDrainSequence); 4504 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4505 return true; 4506 } 4507 return false; 4508} 4509 4510// must be called with thread mutex locked 4511bool AudioFlinger::OffloadThread::shouldStandby_l() 4512{ 4513 bool trackPaused = false; 4514 4515 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4516 // after a timeout and we will enter standby then. 4517 if (mTracks.size() > 0) { 4518 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4519 } 4520 4521 return !mStandby && !trackPaused; 4522} 4523 4524 4525bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4526{ 4527 Mutex::Autolock _l(mLock); 4528 return waitingAsyncCallback_l(); 4529} 4530 4531void AudioFlinger::OffloadThread::flushHw_l() 4532{ 4533 mOutput->stream->flush(mOutput->stream); 4534 // Flush anything still waiting in the mixbuffer 4535 mCurrentWriteLength = 0; 4536 mBytesRemaining = 0; 4537 mPausedWriteLength = 0; 4538 mPausedBytesRemaining = 0; 4539 mHwPaused = false; 4540 4541 if (mUseAsyncWrite) { 4542 // discard any pending drain or write ack by incrementing sequence 4543 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4544 mDrainSequence = (mDrainSequence + 2) & ~1; 4545 ALOG_ASSERT(mCallbackThread != 0); 4546 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4547 mCallbackThread->setDraining(mDrainSequence); 4548 } 4549} 4550 4551void AudioFlinger::OffloadThread::onAddNewTrack_l() 4552{ 4553 sp<Track> previousTrack = mPreviousTrack.promote(); 4554 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4555 4556 if (previousTrack != 0 && latestTrack != 0 && 4557 (previousTrack->sessionId() != latestTrack->sessionId())) { 4558 mFlushPending = true; 4559 } 4560 PlaybackThread::onAddNewTrack_l(); 4561} 4562 4563// ---------------------------------------------------------------------------- 4564 4565AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4566 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4567 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4568 DUPLICATING), 4569 mWaitTimeMs(UINT_MAX) 4570{ 4571 addOutputTrack(mainThread); 4572} 4573 4574AudioFlinger::DuplicatingThread::~DuplicatingThread() 4575{ 4576 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4577 mOutputTracks[i]->destroy(); 4578 } 4579} 4580 4581void AudioFlinger::DuplicatingThread::threadLoop_mix() 4582{ 4583 // mix buffers... 4584 if (outputsReady(outputTracks)) { 4585 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4586 } else { 4587 memset(mSinkBuffer, 0, mSinkBufferSize); 4588 } 4589 sleepTime = 0; 4590 writeFrames = mNormalFrameCount; 4591 mCurrentWriteLength = mSinkBufferSize; 4592 standbyTime = systemTime() + standbyDelay; 4593} 4594 4595void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4596{ 4597 if (sleepTime == 0) { 4598 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4599 sleepTime = activeSleepTime; 4600 } else { 4601 sleepTime = idleSleepTime; 4602 } 4603 } else if (mBytesWritten != 0) { 4604 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4605 writeFrames = mNormalFrameCount; 4606 memset(mSinkBuffer, 0, mSinkBufferSize); 4607 } else { 4608 // flush remaining overflow buffers in output tracks 4609 writeFrames = 0; 4610 } 4611 sleepTime = 0; 4612 } 4613} 4614 4615ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4616{ 4617 for (size_t i = 0; i < outputTracks.size(); i++) { 4618 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4619 // for delivery downstream as needed. This in-place conversion is safe as 4620 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4621 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4622 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4623 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4624 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4625 } 4626 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4627 } 4628 mStandby = false; 4629 return (ssize_t)mSinkBufferSize; 4630} 4631 4632void AudioFlinger::DuplicatingThread::threadLoop_standby() 4633{ 4634 // DuplicatingThread implements standby by stopping all tracks 4635 for (size_t i = 0; i < outputTracks.size(); i++) { 4636 outputTracks[i]->stop(); 4637 } 4638} 4639 4640void AudioFlinger::DuplicatingThread::saveOutputTracks() 4641{ 4642 outputTracks = mOutputTracks; 4643} 4644 4645void AudioFlinger::DuplicatingThread::clearOutputTracks() 4646{ 4647 outputTracks.clear(); 4648} 4649 4650void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4651{ 4652 Mutex::Autolock _l(mLock); 4653 // FIXME explain this formula 4654 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4655 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4656 // due to current usage case and restrictions on the AudioBufferProvider. 4657 // Actual buffer conversion is done in threadLoop_write(). 4658 // 4659 // TODO: This may change in the future, depending on multichannel 4660 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4661 OutputTrack *outputTrack = new OutputTrack(thread, 4662 this, 4663 mSampleRate, 4664 AUDIO_FORMAT_PCM_16_BIT, 4665 mChannelMask, 4666 frameCount, 4667 IPCThreadState::self()->getCallingUid()); 4668 if (outputTrack->cblk() != NULL) { 4669 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4670 mOutputTracks.add(outputTrack); 4671 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4672 updateWaitTime_l(); 4673 } 4674} 4675 4676void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4677{ 4678 Mutex::Autolock _l(mLock); 4679 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4680 if (mOutputTracks[i]->thread() == thread) { 4681 mOutputTracks[i]->destroy(); 4682 mOutputTracks.removeAt(i); 4683 updateWaitTime_l(); 4684 return; 4685 } 4686 } 4687 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4688} 4689 4690// caller must hold mLock 4691void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4692{ 4693 mWaitTimeMs = UINT_MAX; 4694 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4695 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4696 if (strong != 0) { 4697 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4698 if (waitTimeMs < mWaitTimeMs) { 4699 mWaitTimeMs = waitTimeMs; 4700 } 4701 } 4702 } 4703} 4704 4705 4706bool AudioFlinger::DuplicatingThread::outputsReady( 4707 const SortedVector< sp<OutputTrack> > &outputTracks) 4708{ 4709 for (size_t i = 0; i < outputTracks.size(); i++) { 4710 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4711 if (thread == 0) { 4712 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4713 outputTracks[i].get()); 4714 return false; 4715 } 4716 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4717 // see note at standby() declaration 4718 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4719 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4720 thread.get()); 4721 return false; 4722 } 4723 } 4724 return true; 4725} 4726 4727uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4728{ 4729 return (mWaitTimeMs * 1000) / 2; 4730} 4731 4732void AudioFlinger::DuplicatingThread::cacheParameters_l() 4733{ 4734 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4735 updateWaitTime_l(); 4736 4737 MixerThread::cacheParameters_l(); 4738} 4739 4740// ---------------------------------------------------------------------------- 4741// Record 4742// ---------------------------------------------------------------------------- 4743 4744AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4745 AudioStreamIn *input, 4746 audio_io_handle_t id, 4747 audio_devices_t outDevice, 4748 audio_devices_t inDevice 4749#ifdef TEE_SINK 4750 , const sp<NBAIO_Sink>& teeSink 4751#endif 4752 ) : 4753 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4754 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4755 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4756 mRsmpInRear(0) 4757#ifdef TEE_SINK 4758 , mTeeSink(teeSink) 4759#endif 4760 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4761 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4762 // mFastCapture below 4763 , mFastCaptureFutex(0) 4764 // mInputSource 4765 // mPipeSink 4766 // mPipeSource 4767 , mPipeFramesP2(0) 4768 // mPipeMemory 4769 // mFastCaptureNBLogWriter 4770 , mFastTrackAvail(true) 4771{ 4772 snprintf(mName, kNameLength, "AudioIn_%X", id); 4773 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4774 4775 readInputParameters_l(); 4776 4777 // create an NBAIO source for the HAL input stream, and negotiate 4778 mInputSource = new AudioStreamInSource(input->stream); 4779 size_t numCounterOffers = 0; 4780 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4781 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4782 ALOG_ASSERT(index == 0); 4783 4784 // initialize fast capture depending on configuration 4785 bool initFastCapture; 4786 switch (kUseFastCapture) { 4787 case FastCapture_Never: 4788 initFastCapture = false; 4789 break; 4790 case FastCapture_Always: 4791 initFastCapture = true; 4792 break; 4793 case FastCapture_Static: 4794 uint32_t primaryOutputSampleRate; 4795 { 4796 AutoMutex _l(audioFlinger->mHardwareLock); 4797 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4798 } 4799 initFastCapture = 4800 // either capture sample rate is same as (a reasonable) primary output sample rate 4801 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4802 (mSampleRate == primaryOutputSampleRate)) || 4803 // or primary output sample rate is unknown, and capture sample rate is reasonable 4804 ((primaryOutputSampleRate == 0) && 4805 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4806 // and the buffer size is < 10 ms 4807 (mFrameCount * 1000) / mSampleRate < 10; 4808 break; 4809 // case FastCapture_Dynamic: 4810 } 4811 4812 if (initFastCapture) { 4813 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4814 NBAIO_Format format = mInputSource->format(); 4815 size_t pipeFramesP2 = roundup(mFrameCount * 8); 4816 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4817 void *pipeBuffer; 4818 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4819 sp<IMemory> pipeMemory; 4820 if ((roHeap == 0) || 4821 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4822 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4823 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4824 goto failed; 4825 } 4826 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4827 memset(pipeBuffer, 0, pipeSize); 4828 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4829 const NBAIO_Format offers[1] = {format}; 4830 size_t numCounterOffers = 0; 4831 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4832 ALOG_ASSERT(index == 0); 4833 mPipeSink = pipe; 4834 PipeReader *pipeReader = new PipeReader(*pipe); 4835 numCounterOffers = 0; 4836 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4837 ALOG_ASSERT(index == 0); 4838 mPipeSource = pipeReader; 4839 mPipeFramesP2 = pipeFramesP2; 4840 mPipeMemory = pipeMemory; 4841 4842 // create fast capture 4843 mFastCapture = new FastCapture(); 4844 FastCaptureStateQueue *sq = mFastCapture->sq(); 4845#ifdef STATE_QUEUE_DUMP 4846 // FIXME 4847#endif 4848 FastCaptureState *state = sq->begin(); 4849 state->mCblk = NULL; 4850 state->mInputSource = mInputSource.get(); 4851 state->mInputSourceGen++; 4852 state->mPipeSink = pipe; 4853 state->mPipeSinkGen++; 4854 state->mFrameCount = mFrameCount; 4855 state->mCommand = FastCaptureState::COLD_IDLE; 4856 // already done in constructor initialization list 4857 //mFastCaptureFutex = 0; 4858 state->mColdFutexAddr = &mFastCaptureFutex; 4859 state->mColdGen++; 4860 state->mDumpState = &mFastCaptureDumpState; 4861#ifdef TEE_SINK 4862 // FIXME 4863#endif 4864 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4865 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4866 sq->end(); 4867 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4868 4869 // start the fast capture 4870 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4871 pid_t tid = mFastCapture->getTid(); 4872 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4873 if (err != 0) { 4874 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4875 kPriorityFastCapture, getpid_cached, tid, err); 4876 } 4877 4878#ifdef AUDIO_WATCHDOG 4879 // FIXME 4880#endif 4881 4882 } 4883failed: ; 4884 4885 // FIXME mNormalSource 4886} 4887 4888 4889AudioFlinger::RecordThread::~RecordThread() 4890{ 4891 if (mFastCapture != 0) { 4892 FastCaptureStateQueue *sq = mFastCapture->sq(); 4893 FastCaptureState *state = sq->begin(); 4894 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4895 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4896 if (old == -1) { 4897 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4898 } 4899 } 4900 state->mCommand = FastCaptureState::EXIT; 4901 sq->end(); 4902 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4903 mFastCapture->join(); 4904 mFastCapture.clear(); 4905 } 4906 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4907 mAudioFlinger->unregisterWriter(mNBLogWriter); 4908 delete[] mRsmpInBuffer; 4909} 4910 4911void AudioFlinger::RecordThread::onFirstRef() 4912{ 4913 run(mName, PRIORITY_URGENT_AUDIO); 4914} 4915 4916bool AudioFlinger::RecordThread::threadLoop() 4917{ 4918 nsecs_t lastWarning = 0; 4919 4920 inputStandBy(); 4921 4922reacquire_wakelock: 4923 sp<RecordTrack> activeTrack; 4924 int activeTracksGen; 4925 { 4926 Mutex::Autolock _l(mLock); 4927 size_t size = mActiveTracks.size(); 4928 activeTracksGen = mActiveTracksGen; 4929 if (size > 0) { 4930 // FIXME an arbitrary choice 4931 activeTrack = mActiveTracks[0]; 4932 acquireWakeLock_l(activeTrack->uid()); 4933 if (size > 1) { 4934 SortedVector<int> tmp; 4935 for (size_t i = 0; i < size; i++) { 4936 tmp.add(mActiveTracks[i]->uid()); 4937 } 4938 updateWakeLockUids_l(tmp); 4939 } 4940 } else { 4941 acquireWakeLock_l(-1); 4942 } 4943 } 4944 4945 // used to request a deferred sleep, to be executed later while mutex is unlocked 4946 uint32_t sleepUs = 0; 4947 4948 // loop while there is work to do 4949 for (;;) { 4950 Vector< sp<EffectChain> > effectChains; 4951 4952 // sleep with mutex unlocked 4953 if (sleepUs > 0) { 4954 usleep(sleepUs); 4955 sleepUs = 0; 4956 } 4957 4958 // activeTracks accumulates a copy of a subset of mActiveTracks 4959 Vector< sp<RecordTrack> > activeTracks; 4960 4961 // reference to the (first and only) fast track 4962 sp<RecordTrack> fastTrack; 4963 4964 { // scope for mLock 4965 Mutex::Autolock _l(mLock); 4966 4967 processConfigEvents_l(); 4968 4969 // check exitPending here because checkForNewParameters_l() and 4970 // checkForNewParameters_l() can temporarily release mLock 4971 if (exitPending()) { 4972 break; 4973 } 4974 4975 // if no active track(s), then standby and release wakelock 4976 size_t size = mActiveTracks.size(); 4977 if (size == 0) { 4978 standbyIfNotAlreadyInStandby(); 4979 // exitPending() can't become true here 4980 releaseWakeLock_l(); 4981 ALOGV("RecordThread: loop stopping"); 4982 // go to sleep 4983 mWaitWorkCV.wait(mLock); 4984 ALOGV("RecordThread: loop starting"); 4985 goto reacquire_wakelock; 4986 } 4987 4988 if (mActiveTracksGen != activeTracksGen) { 4989 activeTracksGen = mActiveTracksGen; 4990 SortedVector<int> tmp; 4991 for (size_t i = 0; i < size; i++) { 4992 tmp.add(mActiveTracks[i]->uid()); 4993 } 4994 updateWakeLockUids_l(tmp); 4995 } 4996 4997 bool doBroadcast = false; 4998 for (size_t i = 0; i < size; ) { 4999 5000 activeTrack = mActiveTracks[i]; 5001 if (activeTrack->isTerminated()) { 5002 removeTrack_l(activeTrack); 5003 mActiveTracks.remove(activeTrack); 5004 mActiveTracksGen++; 5005 size--; 5006 continue; 5007 } 5008 5009 TrackBase::track_state activeTrackState = activeTrack->mState; 5010 switch (activeTrackState) { 5011 5012 case TrackBase::PAUSING: 5013 mActiveTracks.remove(activeTrack); 5014 mActiveTracksGen++; 5015 doBroadcast = true; 5016 size--; 5017 continue; 5018 5019 case TrackBase::STARTING_1: 5020 sleepUs = 10000; 5021 i++; 5022 continue; 5023 5024 case TrackBase::STARTING_2: 5025 doBroadcast = true; 5026 mStandby = false; 5027 activeTrack->mState = TrackBase::ACTIVE; 5028 break; 5029 5030 case TrackBase::ACTIVE: 5031 break; 5032 5033 case TrackBase::IDLE: 5034 i++; 5035 continue; 5036 5037 default: 5038 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5039 } 5040 5041 activeTracks.add(activeTrack); 5042 i++; 5043 5044 if (activeTrack->isFastTrack()) { 5045 ALOG_ASSERT(!mFastTrackAvail); 5046 ALOG_ASSERT(fastTrack == 0); 5047 fastTrack = activeTrack; 5048 } 5049 } 5050 if (doBroadcast) { 5051 mStartStopCond.broadcast(); 5052 } 5053 5054 // sleep if there are no active tracks to process 5055 if (activeTracks.size() == 0) { 5056 if (sleepUs == 0) { 5057 sleepUs = kRecordThreadSleepUs; 5058 } 5059 continue; 5060 } 5061 sleepUs = 0; 5062 5063 lockEffectChains_l(effectChains); 5064 } 5065 5066 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5067 5068 size_t size = effectChains.size(); 5069 for (size_t i = 0; i < size; i++) { 5070 // thread mutex is not locked, but effect chain is locked 5071 effectChains[i]->process_l(); 5072 } 5073 5074 // Start the fast capture if it's not already running 5075 if (mFastCapture != 0) { 5076 FastCaptureStateQueue *sq = mFastCapture->sq(); 5077 FastCaptureState *state = sq->begin(); 5078 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5079 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5080 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5081 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5082 if (old == -1) { 5083 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5084 } 5085 } 5086 state->mCommand = FastCaptureState::READ_WRITE; 5087#if 0 // FIXME 5088 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5089 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5090#endif 5091 state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL; 5092 sq->end(); 5093 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5094#if 0 5095 if (kUseFastCapture == FastCapture_Dynamic) { 5096 mNormalSource = mPipeSource; 5097 } 5098#endif 5099 } else { 5100 sq->end(false /*didModify*/); 5101 } 5102 } 5103 5104 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5105 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5106 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5107 // If destination is non-contiguous, first read past the nominal end of buffer, then 5108 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5109 5110 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5111 ssize_t framesRead; 5112 5113 // If an NBAIO source is present, use it to read the normal capture's data 5114 if (mPipeSource != 0) { 5115 size_t framesToRead = mBufferSize / mFrameSize; 5116 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5117 framesToRead, AudioBufferProvider::kInvalidPTS); 5118 if (framesRead == 0) { 5119 // since pipe is non-blocking, simulate blocking input 5120 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5121 } 5122 // otherwise use the HAL / AudioStreamIn directly 5123 } else { 5124 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5125 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5126 if (bytesRead < 0) { 5127 framesRead = bytesRead; 5128 } else { 5129 framesRead = bytesRead / mFrameSize; 5130 } 5131 } 5132 5133 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5134 ALOGE("read failed: framesRead=%d", framesRead); 5135 // Force input into standby so that it tries to recover at next read attempt 5136 inputStandBy(); 5137 sleepUs = kRecordThreadSleepUs; 5138 } 5139 if (framesRead <= 0) { 5140 continue; 5141 } 5142 ALOG_ASSERT(framesRead > 0); 5143 5144 if (mTeeSink != 0) { 5145 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5146 } 5147 // If destination is non-contiguous, we now correct for reading past end of buffer. 5148 size_t part1 = mRsmpInFramesP2 - rear; 5149 if ((size_t) framesRead > part1) { 5150 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5151 (framesRead - part1) * mFrameSize); 5152 } 5153 rear = mRsmpInRear += framesRead; 5154 5155 size = activeTracks.size(); 5156 // loop over each active track 5157 for (size_t i = 0; i < size; i++) { 5158 activeTrack = activeTracks[i]; 5159 5160 // skip fast tracks, as those are handled directly by FastCapture 5161 if (activeTrack->isFastTrack()) { 5162 continue; 5163 } 5164 5165 enum { 5166 OVERRUN_UNKNOWN, 5167 OVERRUN_TRUE, 5168 OVERRUN_FALSE 5169 } overrun = OVERRUN_UNKNOWN; 5170 5171 // loop over getNextBuffer to handle circular sink 5172 for (;;) { 5173 5174 activeTrack->mSink.frameCount = ~0; 5175 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5176 size_t framesOut = activeTrack->mSink.frameCount; 5177 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5178 5179 int32_t front = activeTrack->mRsmpInFront; 5180 ssize_t filled = rear - front; 5181 size_t framesIn; 5182 5183 if (filled < 0) { 5184 // should not happen, but treat like a massive overrun and re-sync 5185 framesIn = 0; 5186 activeTrack->mRsmpInFront = rear; 5187 overrun = OVERRUN_TRUE; 5188 } else if ((size_t) filled <= mRsmpInFrames) { 5189 framesIn = (size_t) filled; 5190 } else { 5191 // client is not keeping up with server, but give it latest data 5192 framesIn = mRsmpInFrames; 5193 activeTrack->mRsmpInFront = front = rear - framesIn; 5194 overrun = OVERRUN_TRUE; 5195 } 5196 5197 if (framesOut == 0 || framesIn == 0) { 5198 break; 5199 } 5200 5201 if (activeTrack->mResampler == NULL) { 5202 // no resampling 5203 if (framesIn > framesOut) { 5204 framesIn = framesOut; 5205 } else { 5206 framesOut = framesIn; 5207 } 5208 int8_t *dst = activeTrack->mSink.i8; 5209 while (framesIn > 0) { 5210 front &= mRsmpInFramesP2 - 1; 5211 size_t part1 = mRsmpInFramesP2 - front; 5212 if (part1 > framesIn) { 5213 part1 = framesIn; 5214 } 5215 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5216 if (mChannelCount == activeTrack->mChannelCount) { 5217 memcpy(dst, src, part1 * mFrameSize); 5218 } else if (mChannelCount == 1) { 5219 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 5220 part1); 5221 } else { 5222 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 5223 part1); 5224 } 5225 dst += part1 * activeTrack->mFrameSize; 5226 front += part1; 5227 framesIn -= part1; 5228 } 5229 activeTrack->mRsmpInFront += framesOut; 5230 5231 } else { 5232 // resampling 5233 // FIXME framesInNeeded should really be part of resampler API, and should 5234 // depend on the SRC ratio 5235 // to keep mRsmpInBuffer full so resampler always has sufficient input 5236 size_t framesInNeeded; 5237 // FIXME only re-calculate when it changes, and optimize for common ratios 5238 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 5239 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 5240 framesInNeeded = ceil(framesOut * inOverOut) + 1; 5241 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5242 framesInNeeded, framesOut, inOverOut); 5243 // Although we theoretically have framesIn in circular buffer, some of those are 5244 // unreleased frames, and thus must be discounted for purpose of budgeting. 5245 size_t unreleased = activeTrack->mRsmpInUnrel; 5246 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5247 if (framesIn < framesInNeeded) { 5248 ALOGV("not enough to resample: have %u frames in but need %u in to " 5249 "produce %u out given in/out ratio of %.4g", 5250 framesIn, framesInNeeded, framesOut, inOverOut); 5251 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 5252 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5253 if (newFramesOut == 0) { 5254 break; 5255 } 5256 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 5257 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5258 framesInNeeded, newFramesOut, outOverIn); 5259 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5260 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5261 "given in/out ratio of %.4g", 5262 framesIn, framesInNeeded, newFramesOut, inOverOut); 5263 framesOut = newFramesOut; 5264 } else { 5265 ALOGV("success 1: have %u in and need %u in to produce %u out " 5266 "given in/out ratio of %.4g", 5267 framesIn, framesInNeeded, framesOut, inOverOut); 5268 } 5269 5270 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5271 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5272 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5273 delete[] activeTrack->mRsmpOutBuffer; 5274 // resampler always outputs stereo 5275 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5276 activeTrack->mRsmpOutFrameCount = framesOut; 5277 } 5278 5279 // resampler accumulates, but we only have one source track 5280 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5281 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5282 // FIXME how about having activeTrack implement this interface itself? 5283 activeTrack->mResamplerBufferProvider 5284 /*this*/ /* AudioBufferProvider* */); 5285 // ditherAndClamp() works as long as all buffers returned by 5286 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5287 if (activeTrack->mChannelCount == 1) { 5288 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5289 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5290 framesOut); 5291 // the resampler always outputs stereo samples: 5292 // do post stereo to mono conversion 5293 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5294 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5295 } else { 5296 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5297 activeTrack->mRsmpOutBuffer, framesOut); 5298 } 5299 // now done with mRsmpOutBuffer 5300 5301 } 5302 5303 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5304 overrun = OVERRUN_FALSE; 5305 } 5306 5307 if (activeTrack->mFramesToDrop == 0) { 5308 if (framesOut > 0) { 5309 activeTrack->mSink.frameCount = framesOut; 5310 activeTrack->releaseBuffer(&activeTrack->mSink); 5311 } 5312 } else { 5313 // FIXME could do a partial drop of framesOut 5314 if (activeTrack->mFramesToDrop > 0) { 5315 activeTrack->mFramesToDrop -= framesOut; 5316 if (activeTrack->mFramesToDrop <= 0) { 5317 activeTrack->clearSyncStartEvent(); 5318 } 5319 } else { 5320 activeTrack->mFramesToDrop += framesOut; 5321 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5322 activeTrack->mSyncStartEvent->isCancelled()) { 5323 ALOGW("Synced record %s, session %d, trigger session %d", 5324 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5325 activeTrack->sessionId(), 5326 (activeTrack->mSyncStartEvent != 0) ? 5327 activeTrack->mSyncStartEvent->triggerSession() : 0); 5328 activeTrack->clearSyncStartEvent(); 5329 } 5330 } 5331 } 5332 5333 if (framesOut == 0) { 5334 break; 5335 } 5336 } 5337 5338 switch (overrun) { 5339 case OVERRUN_TRUE: 5340 // client isn't retrieving buffers fast enough 5341 if (!activeTrack->setOverflow()) { 5342 nsecs_t now = systemTime(); 5343 // FIXME should lastWarning per track? 5344 if ((now - lastWarning) > kWarningThrottleNs) { 5345 ALOGW("RecordThread: buffer overflow"); 5346 lastWarning = now; 5347 } 5348 } 5349 break; 5350 case OVERRUN_FALSE: 5351 activeTrack->clearOverflow(); 5352 break; 5353 case OVERRUN_UNKNOWN: 5354 break; 5355 } 5356 5357 } 5358 5359 // enable changes in effect chain 5360 unlockEffectChains(effectChains); 5361 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5362 } 5363 5364 standbyIfNotAlreadyInStandby(); 5365 5366 { 5367 Mutex::Autolock _l(mLock); 5368 for (size_t i = 0; i < mTracks.size(); i++) { 5369 sp<RecordTrack> track = mTracks[i]; 5370 track->invalidate(); 5371 } 5372 mActiveTracks.clear(); 5373 mActiveTracksGen++; 5374 mStartStopCond.broadcast(); 5375 } 5376 5377 releaseWakeLock(); 5378 5379 ALOGV("RecordThread %p exiting", this); 5380 return false; 5381} 5382 5383void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5384{ 5385 if (!mStandby) { 5386 inputStandBy(); 5387 mStandby = true; 5388 } 5389} 5390 5391void AudioFlinger::RecordThread::inputStandBy() 5392{ 5393 // Idle the fast capture if it's currently running 5394 if (mFastCapture != 0) { 5395 FastCaptureStateQueue *sq = mFastCapture->sq(); 5396 FastCaptureState *state = sq->begin(); 5397 if (!(state->mCommand & FastCaptureState::IDLE)) { 5398 state->mCommand = FastCaptureState::COLD_IDLE; 5399 state->mColdFutexAddr = &mFastCaptureFutex; 5400 state->mColdGen++; 5401 mFastCaptureFutex = 0; 5402 sq->end(); 5403 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5404 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5405#if 0 5406 if (kUseFastCapture == FastCapture_Dynamic) { 5407 // FIXME 5408 } 5409#endif 5410#ifdef AUDIO_WATCHDOG 5411 // FIXME 5412#endif 5413 } else { 5414 sq->end(false /*didModify*/); 5415 } 5416 } 5417 mInput->stream->common.standby(&mInput->stream->common); 5418} 5419 5420// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5421sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5422 const sp<AudioFlinger::Client>& client, 5423 uint32_t sampleRate, 5424 audio_format_t format, 5425 audio_channel_mask_t channelMask, 5426 size_t *pFrameCount, 5427 int sessionId, 5428 int uid, 5429 IAudioFlinger::track_flags_t *flags, 5430 pid_t tid, 5431 status_t *status) 5432{ 5433 size_t frameCount = *pFrameCount; 5434 sp<RecordTrack> track; 5435 status_t lStatus; 5436 5437 // client expresses a preference for FAST, but we get the final say 5438 if (*flags & IAudioFlinger::TRACK_FAST) { 5439 if ( 5440 // use case: callback handler and frame count is default or at least as large as HAL 5441 ( 5442 (tid != -1) && 5443 ((frameCount == 0) /*|| 5444 // FIXME must be equal to pipe depth, so don't allow it to be specified by client 5445 // FIXME not necessarily true, should be native frame count for native SR! 5446 (frameCount >= mFrameCount)*/) 5447 ) && 5448 // PCM data 5449 audio_is_linear_pcm(format) && 5450 // native format 5451 (format == mFormat) && 5452 // mono or stereo 5453 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5454 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5455 // native channel mask 5456 (channelMask == mChannelMask) && 5457 // native hardware sample rate 5458 (sampleRate == mSampleRate) && 5459 // record thread has an associated fast capture 5460 hasFastCapture() && 5461 // there are sufficient fast track slots available 5462 mFastTrackAvail 5463 ) { 5464 // if frameCount not specified, then it defaults to pipe frame count 5465 if (frameCount == 0) { 5466 frameCount = mPipeFramesP2; 5467 } 5468 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5469 frameCount, mFrameCount); 5470 } else { 5471 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5472 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5473 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5474 frameCount, mFrameCount, format, 5475 audio_is_linear_pcm(format), 5476 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid, mFastTrackAvail); 5477 *flags &= ~IAudioFlinger::TRACK_FAST; 5478 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5479 // For compatibility with AudioRecord calculation, buffer depth is forced 5480 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5481 // This is probably too conservative, but legacy application code may depend on it. 5482 // If you change this calculation, also review the start threshold which is related. 5483 // FIXME It's not clear how input latency actually matters. Perhaps this should be 0. 5484 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5485 size_t mNormalFrameCount = 2048; // FIXME 5486 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5487 if (minBufCount < 2) { 5488 minBufCount = 2; 5489 } 5490 size_t minFrameCount = mNormalFrameCount * minBufCount; 5491 if (frameCount < minFrameCount) { 5492 frameCount = minFrameCount; 5493 } 5494 } 5495 } 5496 *pFrameCount = frameCount; 5497 5498 lStatus = initCheck(); 5499 if (lStatus != NO_ERROR) { 5500 ALOGE("createRecordTrack_l() audio driver not initialized"); 5501 goto Exit; 5502 } 5503 5504 { // scope for mLock 5505 Mutex::Autolock _l(mLock); 5506 5507 track = new RecordTrack(this, client, sampleRate, 5508 format, channelMask, frameCount, sessionId, uid, 5509 *flags); 5510 5511 lStatus = track->initCheck(); 5512 if (lStatus != NO_ERROR) { 5513 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5514 // track must be cleared from the caller as the caller has the AF lock 5515 goto Exit; 5516 } 5517 mTracks.add(track); 5518 5519 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5520 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5521 mAudioFlinger->btNrecIsOff(); 5522 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5523 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5524 5525 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5526 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5527 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5528 // so ask activity manager to do this on our behalf 5529 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5530 } 5531 } 5532 5533 lStatus = NO_ERROR; 5534 5535Exit: 5536 *status = lStatus; 5537 return track; 5538} 5539 5540status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5541 AudioSystem::sync_event_t event, 5542 int triggerSession) 5543{ 5544 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5545 sp<ThreadBase> strongMe = this; 5546 status_t status = NO_ERROR; 5547 5548 if (event == AudioSystem::SYNC_EVENT_NONE) { 5549 recordTrack->clearSyncStartEvent(); 5550 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5551 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5552 triggerSession, 5553 recordTrack->sessionId(), 5554 syncStartEventCallback, 5555 recordTrack); 5556 // Sync event can be cancelled by the trigger session if the track is not in a 5557 // compatible state in which case we start record immediately 5558 if (recordTrack->mSyncStartEvent->isCancelled()) { 5559 recordTrack->clearSyncStartEvent(); 5560 } else { 5561 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5562 recordTrack->mFramesToDrop = - 5563 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5564 } 5565 } 5566 5567 { 5568 // This section is a rendezvous between binder thread executing start() and RecordThread 5569 AutoMutex lock(mLock); 5570 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5571 if (recordTrack->mState == TrackBase::PAUSING) { 5572 ALOGV("active record track PAUSING -> ACTIVE"); 5573 recordTrack->mState = TrackBase::ACTIVE; 5574 } else { 5575 ALOGV("active record track state %d", recordTrack->mState); 5576 } 5577 return status; 5578 } 5579 5580 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5581 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5582 // or using a separate command thread 5583 recordTrack->mState = TrackBase::STARTING_1; 5584 mActiveTracks.add(recordTrack); 5585 mActiveTracksGen++; 5586 mLock.unlock(); 5587 status_t status = AudioSystem::startInput(mId); 5588 mLock.lock(); 5589 // FIXME should verify that recordTrack is still in mActiveTracks 5590 if (status != NO_ERROR) { 5591 mActiveTracks.remove(recordTrack); 5592 mActiveTracksGen++; 5593 recordTrack->clearSyncStartEvent(); 5594 return status; 5595 } 5596 // Catch up with current buffer indices if thread is already running. 5597 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5598 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5599 // see previously buffered data before it called start(), but with greater risk of overrun. 5600 5601 recordTrack->mRsmpInFront = mRsmpInRear; 5602 recordTrack->mRsmpInUnrel = 0; 5603 // FIXME why reset? 5604 if (recordTrack->mResampler != NULL) { 5605 recordTrack->mResampler->reset(); 5606 } 5607 recordTrack->mState = TrackBase::STARTING_2; 5608 // signal thread to start 5609 mWaitWorkCV.broadcast(); 5610 if (mActiveTracks.indexOf(recordTrack) < 0) { 5611 ALOGV("Record failed to start"); 5612 status = BAD_VALUE; 5613 goto startError; 5614 } 5615 return status; 5616 } 5617 5618startError: 5619 AudioSystem::stopInput(mId); 5620 recordTrack->clearSyncStartEvent(); 5621 // FIXME I wonder why we do not reset the state here? 5622 return status; 5623} 5624 5625void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5626{ 5627 sp<SyncEvent> strongEvent = event.promote(); 5628 5629 if (strongEvent != 0) { 5630 sp<RefBase> ptr = strongEvent->cookie().promote(); 5631 if (ptr != 0) { 5632 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5633 recordTrack->handleSyncStartEvent(strongEvent); 5634 } 5635 } 5636} 5637 5638bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5639 ALOGV("RecordThread::stop"); 5640 AutoMutex _l(mLock); 5641 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5642 return false; 5643 } 5644 // note that threadLoop may still be processing the track at this point [without lock] 5645 recordTrack->mState = TrackBase::PAUSING; 5646 // do not wait for mStartStopCond if exiting 5647 if (exitPending()) { 5648 return true; 5649 } 5650 // FIXME incorrect usage of wait: no explicit predicate or loop 5651 mStartStopCond.wait(mLock); 5652 // if we have been restarted, recordTrack is in mActiveTracks here 5653 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5654 ALOGV("Record stopped OK"); 5655 return true; 5656 } 5657 return false; 5658} 5659 5660bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5661{ 5662 return false; 5663} 5664 5665status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5666{ 5667#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5668 if (!isValidSyncEvent(event)) { 5669 return BAD_VALUE; 5670 } 5671 5672 int eventSession = event->triggerSession(); 5673 status_t ret = NAME_NOT_FOUND; 5674 5675 Mutex::Autolock _l(mLock); 5676 5677 for (size_t i = 0; i < mTracks.size(); i++) { 5678 sp<RecordTrack> track = mTracks[i]; 5679 if (eventSession == track->sessionId()) { 5680 (void) track->setSyncEvent(event); 5681 ret = NO_ERROR; 5682 } 5683 } 5684 return ret; 5685#else 5686 return BAD_VALUE; 5687#endif 5688} 5689 5690// destroyTrack_l() must be called with ThreadBase::mLock held 5691void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5692{ 5693 track->terminate(); 5694 track->mState = TrackBase::STOPPED; 5695 // active tracks are removed by threadLoop() 5696 if (mActiveTracks.indexOf(track) < 0) { 5697 removeTrack_l(track); 5698 } 5699} 5700 5701void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5702{ 5703 mTracks.remove(track); 5704 // need anything related to effects here? 5705 if (track->isFastTrack()) { 5706 ALOG_ASSERT(!mFastTrackAvail); 5707 mFastTrackAvail = true; 5708 } 5709} 5710 5711void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5712{ 5713 dumpInternals(fd, args); 5714 dumpTracks(fd, args); 5715 dumpEffectChains(fd, args); 5716} 5717 5718void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5719{ 5720 fdprintf(fd, "\nInput thread %p:\n", this); 5721 5722 if (mActiveTracks.size() > 0) { 5723 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5724 } else { 5725 fdprintf(fd, " No active record clients\n"); 5726 } 5727 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5728 5729 dumpBase(fd, args); 5730} 5731 5732void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5733{ 5734 const size_t SIZE = 256; 5735 char buffer[SIZE]; 5736 String8 result; 5737 5738 size_t numtracks = mTracks.size(); 5739 size_t numactive = mActiveTracks.size(); 5740 size_t numactiveseen = 0; 5741 fdprintf(fd, " %d Tracks", numtracks); 5742 if (numtracks) { 5743 fdprintf(fd, " of which %d are active\n", numactive); 5744 RecordTrack::appendDumpHeader(result); 5745 for (size_t i = 0; i < numtracks ; ++i) { 5746 sp<RecordTrack> track = mTracks[i]; 5747 if (track != 0) { 5748 bool active = mActiveTracks.indexOf(track) >= 0; 5749 if (active) { 5750 numactiveseen++; 5751 } 5752 track->dump(buffer, SIZE, active); 5753 result.append(buffer); 5754 } 5755 } 5756 } else { 5757 fdprintf(fd, "\n"); 5758 } 5759 5760 if (numactiveseen != numactive) { 5761 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5762 " not in the track list\n"); 5763 result.append(buffer); 5764 RecordTrack::appendDumpHeader(result); 5765 for (size_t i = 0; i < numactive; ++i) { 5766 sp<RecordTrack> track = mActiveTracks[i]; 5767 if (mTracks.indexOf(track) < 0) { 5768 track->dump(buffer, SIZE, true); 5769 result.append(buffer); 5770 } 5771 } 5772 5773 } 5774 write(fd, result.string(), result.size()); 5775} 5776 5777// AudioBufferProvider interface 5778status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5779 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5780{ 5781 RecordTrack *activeTrack = mRecordTrack; 5782 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5783 if (threadBase == 0) { 5784 buffer->frameCount = 0; 5785 buffer->raw = NULL; 5786 return NOT_ENOUGH_DATA; 5787 } 5788 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5789 int32_t rear = recordThread->mRsmpInRear; 5790 int32_t front = activeTrack->mRsmpInFront; 5791 ssize_t filled = rear - front; 5792 // FIXME should not be P2 (don't want to increase latency) 5793 // FIXME if client not keeping up, discard 5794 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5795 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5796 front &= recordThread->mRsmpInFramesP2 - 1; 5797 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5798 if (part1 > (size_t) filled) { 5799 part1 = filled; 5800 } 5801 size_t ask = buffer->frameCount; 5802 ALOG_ASSERT(ask > 0); 5803 if (part1 > ask) { 5804 part1 = ask; 5805 } 5806 if (part1 == 0) { 5807 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5808 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5809 buffer->raw = NULL; 5810 buffer->frameCount = 0; 5811 activeTrack->mRsmpInUnrel = 0; 5812 return NOT_ENOUGH_DATA; 5813 } 5814 5815 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5816 buffer->frameCount = part1; 5817 activeTrack->mRsmpInUnrel = part1; 5818 return NO_ERROR; 5819} 5820 5821// AudioBufferProvider interface 5822void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5823 AudioBufferProvider::Buffer* buffer) 5824{ 5825 RecordTrack *activeTrack = mRecordTrack; 5826 size_t stepCount = buffer->frameCount; 5827 if (stepCount == 0) { 5828 return; 5829 } 5830 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5831 activeTrack->mRsmpInUnrel -= stepCount; 5832 activeTrack->mRsmpInFront += stepCount; 5833 buffer->raw = NULL; 5834 buffer->frameCount = 0; 5835} 5836 5837bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5838 status_t& status) 5839{ 5840 bool reconfig = false; 5841 5842 status = NO_ERROR; 5843 5844 audio_format_t reqFormat = mFormat; 5845 uint32_t samplingRate = mSampleRate; 5846 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5847 5848 AudioParameter param = AudioParameter(keyValuePair); 5849 int value; 5850 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5851 // channel count change can be requested. Do we mandate the first client defines the 5852 // HAL sampling rate and channel count or do we allow changes on the fly? 5853 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5854 samplingRate = value; 5855 reconfig = true; 5856 } 5857 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5858 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5859 status = BAD_VALUE; 5860 } else { 5861 reqFormat = (audio_format_t) value; 5862 reconfig = true; 5863 } 5864 } 5865 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5866 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5867 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5868 status = BAD_VALUE; 5869 } else { 5870 channelMask = mask; 5871 reconfig = true; 5872 } 5873 } 5874 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5875 // do not accept frame count changes if tracks are open as the track buffer 5876 // size depends on frame count and correct behavior would not be guaranteed 5877 // if frame count is changed after track creation 5878 if (mActiveTracks.size() > 0) { 5879 status = INVALID_OPERATION; 5880 } else { 5881 reconfig = true; 5882 } 5883 } 5884 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5885 // forward device change to effects that have requested to be 5886 // aware of attached audio device. 5887 for (size_t i = 0; i < mEffectChains.size(); i++) { 5888 mEffectChains[i]->setDevice_l(value); 5889 } 5890 5891 // store input device and output device but do not forward output device to audio HAL. 5892 // Note that status is ignored by the caller for output device 5893 // (see AudioFlinger::setParameters() 5894 if (audio_is_output_devices(value)) { 5895 mOutDevice = value; 5896 status = BAD_VALUE; 5897 } else { 5898 mInDevice = value; 5899 // disable AEC and NS if the device is a BT SCO headset supporting those 5900 // pre processings 5901 if (mTracks.size() > 0) { 5902 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5903 mAudioFlinger->btNrecIsOff(); 5904 for (size_t i = 0; i < mTracks.size(); i++) { 5905 sp<RecordTrack> track = mTracks[i]; 5906 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5907 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5908 } 5909 } 5910 } 5911 } 5912 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5913 mAudioSource != (audio_source_t)value) { 5914 // forward device change to effects that have requested to be 5915 // aware of attached audio device. 5916 for (size_t i = 0; i < mEffectChains.size(); i++) { 5917 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5918 } 5919 mAudioSource = (audio_source_t)value; 5920 } 5921 5922 if (status == NO_ERROR) { 5923 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5924 keyValuePair.string()); 5925 if (status == INVALID_OPERATION) { 5926 inputStandBy(); 5927 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5928 keyValuePair.string()); 5929 } 5930 if (reconfig) { 5931 if (status == BAD_VALUE && 5932 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5933 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5934 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5935 <= (2 * samplingRate)) && 5936 audio_channel_count_from_in_mask( 5937 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5938 (channelMask == AUDIO_CHANNEL_IN_MONO || 5939 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5940 status = NO_ERROR; 5941 } 5942 if (status == NO_ERROR) { 5943 readInputParameters_l(); 5944 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5945 } 5946 } 5947 } 5948 5949 return reconfig; 5950} 5951 5952String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5953{ 5954 Mutex::Autolock _l(mLock); 5955 if (initCheck() != NO_ERROR) { 5956 return String8(); 5957 } 5958 5959 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5960 const String8 out_s8(s); 5961 free(s); 5962 return out_s8; 5963} 5964 5965void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5966 AudioSystem::OutputDescriptor desc; 5967 const void *param2 = NULL; 5968 5969 switch (event) { 5970 case AudioSystem::INPUT_OPENED: 5971 case AudioSystem::INPUT_CONFIG_CHANGED: 5972 desc.channelMask = mChannelMask; 5973 desc.samplingRate = mSampleRate; 5974 desc.format = mFormat; 5975 desc.frameCount = mFrameCount; 5976 desc.latency = 0; 5977 param2 = &desc; 5978 break; 5979 5980 case AudioSystem::INPUT_CLOSED: 5981 default: 5982 break; 5983 } 5984 mAudioFlinger->audioConfigChanged(event, mId, param2); 5985} 5986 5987void AudioFlinger::RecordThread::readInputParameters_l() 5988{ 5989 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5990 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5991 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 5992 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5993 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5994 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5995 } 5996 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5997 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5998 mFrameCount = mBufferSize / mFrameSize; 5999 // This is the formula for calculating the temporary buffer size. 6000 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6001 // 1 full output buffer, regardless of the alignment of the available input. 6002 // The value is somewhat arbitrary, and could probably be even larger. 6003 // A larger value should allow more old data to be read after a track calls start(), 6004 // without increasing latency. 6005 mRsmpInFrames = mFrameCount * 7; 6006 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6007 delete[] mRsmpInBuffer; 6008 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6009 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6010 6011 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6012 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6013} 6014 6015uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6016{ 6017 Mutex::Autolock _l(mLock); 6018 if (initCheck() != NO_ERROR) { 6019 return 0; 6020 } 6021 6022 return mInput->stream->get_input_frames_lost(mInput->stream); 6023} 6024 6025uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6026{ 6027 Mutex::Autolock _l(mLock); 6028 uint32_t result = 0; 6029 if (getEffectChain_l(sessionId) != 0) { 6030 result = EFFECT_SESSION; 6031 } 6032 6033 for (size_t i = 0; i < mTracks.size(); ++i) { 6034 if (sessionId == mTracks[i]->sessionId()) { 6035 result |= TRACK_SESSION; 6036 break; 6037 } 6038 } 6039 6040 return result; 6041} 6042 6043KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6044{ 6045 KeyedVector<int, bool> ids; 6046 Mutex::Autolock _l(mLock); 6047 for (size_t j = 0; j < mTracks.size(); ++j) { 6048 sp<RecordThread::RecordTrack> track = mTracks[j]; 6049 int sessionId = track->sessionId(); 6050 if (ids.indexOfKey(sessionId) < 0) { 6051 ids.add(sessionId, true); 6052 } 6053 } 6054 return ids; 6055} 6056 6057AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6058{ 6059 Mutex::Autolock _l(mLock); 6060 AudioStreamIn *input = mInput; 6061 mInput = NULL; 6062 return input; 6063} 6064 6065// this method must always be called either with ThreadBase mLock held or inside the thread loop 6066audio_stream_t* AudioFlinger::RecordThread::stream() const 6067{ 6068 if (mInput == NULL) { 6069 return NULL; 6070 } 6071 return &mInput->stream->common; 6072} 6073 6074status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6075{ 6076 // only one chain per input thread 6077 if (mEffectChains.size() != 0) { 6078 return INVALID_OPERATION; 6079 } 6080 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6081 6082 chain->setInBuffer(NULL); 6083 chain->setOutBuffer(NULL); 6084 6085 checkSuspendOnAddEffectChain_l(chain); 6086 6087 mEffectChains.add(chain); 6088 6089 return NO_ERROR; 6090} 6091 6092size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6093{ 6094 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6095 ALOGW_IF(mEffectChains.size() != 1, 6096 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6097 chain.get(), mEffectChains.size(), this); 6098 if (mEffectChains.size() == 1) { 6099 mEffectChains.removeAt(0); 6100 } 6101 return 0; 6102} 6103 6104status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6105 audio_patch_handle_t *handle) 6106{ 6107 status_t status = NO_ERROR; 6108 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6109 // store new device and send to effects 6110 mInDevice = patch->sources[0].ext.device.type; 6111 for (size_t i = 0; i < mEffectChains.size(); i++) { 6112 mEffectChains[i]->setDevice_l(mInDevice); 6113 } 6114 6115 // disable AEC and NS if the device is a BT SCO headset supporting those 6116 // pre processings 6117 if (mTracks.size() > 0) { 6118 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6119 mAudioFlinger->btNrecIsOff(); 6120 for (size_t i = 0; i < mTracks.size(); i++) { 6121 sp<RecordTrack> track = mTracks[i]; 6122 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6123 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6124 } 6125 } 6126 6127 // store new source and send to effects 6128 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6129 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6130 for (size_t i = 0; i < mEffectChains.size(); i++) { 6131 mEffectChains[i]->setAudioSource_l(mAudioSource); 6132 } 6133 } 6134 6135 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6136 status = hwDevice->create_audio_patch(hwDevice, 6137 patch->num_sources, 6138 patch->sources, 6139 patch->num_sinks, 6140 patch->sinks, 6141 handle); 6142 } else { 6143 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6144 } 6145 return status; 6146} 6147 6148status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6149{ 6150 status_t status = NO_ERROR; 6151 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6152 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6153 status = hwDevice->release_audio_patch(hwDevice, handle); 6154 } else { 6155 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6156 } 6157 return status; 6158} 6159 6160 6161}; // namespace android 6162