Threads.cpp revision 83b8808faad1e91690c64d7007348be8d9ebde73
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38#include <audio_utils/minifloat.h> 39 40// NBAIO implementations 41#include <media/nbaio/AudioStreamInSource.h> 42#include <media/nbaio/AudioStreamOutSink.h> 43#include <media/nbaio/MonoPipe.h> 44#include <media/nbaio/MonoPipeReader.h> 45#include <media/nbaio/Pipe.h> 46#include <media/nbaio/PipeReader.h> 47#include <media/nbaio/SourceAudioBufferProvider.h> 48 49#include <powermanager/PowerManager.h> 50 51#include <common_time/cc_helper.h> 52#include <common_time/local_clock.h> 53 54#include "AudioFlinger.h" 55#include "AudioMixer.h" 56#include "FastMixer.h" 57#include "FastCapture.h" 58#include "ServiceUtilities.h" 59#include "SchedulingPolicyService.h" 60 61#ifdef ADD_BATTERY_DATA 62#include <media/IMediaPlayerService.h> 63#include <media/IMediaDeathNotifier.h> 64#endif 65 66#ifdef DEBUG_CPU_USAGE 67#include <cpustats/CentralTendencyStatistics.h> 68#include <cpustats/ThreadCpuUsage.h> 69#endif 70 71// ---------------------------------------------------------------------------- 72 73// Note: the following macro is used for extremely verbose logging message. In 74// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 75// 0; but one side effect of this is to turn all LOGV's as well. Some messages 76// are so verbose that we want to suppress them even when we have ALOG_ASSERT 77// turned on. Do not uncomment the #def below unless you really know what you 78// are doing and want to see all of the extremely verbose messages. 79//#define VERY_VERY_VERBOSE_LOGGING 80#ifdef VERY_VERY_VERBOSE_LOGGING 81#define ALOGVV ALOGV 82#else 83#define ALOGVV(a...) do { } while(0) 84#endif 85 86namespace android { 87 88// retry counts for buffer fill timeout 89// 50 * ~20msecs = 1 second 90static const int8_t kMaxTrackRetries = 50; 91static const int8_t kMaxTrackStartupRetries = 50; 92// allow less retry attempts on direct output thread. 93// direct outputs can be a scarce resource in audio hardware and should 94// be released as quickly as possible. 95static const int8_t kMaxTrackRetriesDirect = 2; 96 97// don't warn about blocked writes or record buffer overflows more often than this 98static const nsecs_t kWarningThrottleNs = seconds(5); 99 100// RecordThread loop sleep time upon application overrun or audio HAL read error 101static const int kRecordThreadSleepUs = 5000; 102 103// maximum time to wait in sendConfigEvent_l() for a status to be received 104static const nsecs_t kConfigEventTimeoutNs = seconds(2); 105 106// minimum sleep time for the mixer thread loop when tracks are active but in underrun 107static const uint32_t kMinThreadSleepTimeUs = 5000; 108// maximum divider applied to the active sleep time in the mixer thread loop 109static const uint32_t kMaxThreadSleepTimeShift = 2; 110 111// minimum normal sink buffer size, expressed in milliseconds rather than frames 112static const uint32_t kMinNormalSinkBufferSizeMs = 20; 113// maximum normal sink buffer size 114static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 115 116// Offloaded output thread standby delay: allows track transition without going to standby 117static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 118 119// Whether to use fast mixer 120static const enum { 121 FastMixer_Never, // never initialize or use: for debugging only 122 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 123 // normal mixer multiplier is 1 124 FastMixer_Static, // initialize if needed, then use all the time if initialized, 125 // multiplier is calculated based on min & max normal mixer buffer size 126 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 127 // multiplier is calculated based on min & max normal mixer buffer size 128 // FIXME for FastMixer_Dynamic: 129 // Supporting this option will require fixing HALs that can't handle large writes. 130 // For example, one HAL implementation returns an error from a large write, 131 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 132 // We could either fix the HAL implementations, or provide a wrapper that breaks 133 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 134} kUseFastMixer = FastMixer_Static; 135 136// Whether to use fast capture 137static const enum { 138 FastCapture_Never, // never initialize or use: for debugging only 139 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 140 FastCapture_Static, // initialize if needed, then use all the time if initialized 141} kUseFastCapture = FastCapture_Static; 142 143// Priorities for requestPriority 144static const int kPriorityAudioApp = 2; 145static const int kPriorityFastMixer = 3; 146static const int kPriorityFastCapture = 3; 147 148// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 149// for the track. The client then sub-divides this into smaller buffers for its use. 150// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 151// So for now we just assume that client is double-buffered for fast tracks. 152// FIXME It would be better for client to tell AudioFlinger the value of N, 153// so AudioFlinger could allocate the right amount of memory. 154// See the client's minBufCount and mNotificationFramesAct calculations for details. 155 156// This is the default value, if not specified by property. 157static const int kFastTrackMultiplier = 2; 158 159// The minimum and maximum allowed values 160static const int kFastTrackMultiplierMin = 1; 161static const int kFastTrackMultiplierMax = 2; 162 163// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 164static int sFastTrackMultiplier = kFastTrackMultiplier; 165 166// See Thread::readOnlyHeap(). 167// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 168// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 169// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 170static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 171 172// ---------------------------------------------------------------------------- 173 174static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 175 176static void sFastTrackMultiplierInit() 177{ 178 char value[PROPERTY_VALUE_MAX]; 179 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 180 char *endptr; 181 unsigned long ul = strtoul(value, &endptr, 0); 182 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 183 sFastTrackMultiplier = (int) ul; 184 } 185 } 186} 187 188// ---------------------------------------------------------------------------- 189 190#ifdef ADD_BATTERY_DATA 191// To collect the amplifier usage 192static void addBatteryData(uint32_t params) { 193 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 194 if (service == NULL) { 195 // it already logged 196 return; 197 } 198 199 service->addBatteryData(params); 200} 201#endif 202 203 204// ---------------------------------------------------------------------------- 205// CPU Stats 206// ---------------------------------------------------------------------------- 207 208class CpuStats { 209public: 210 CpuStats(); 211 void sample(const String8 &title); 212#ifdef DEBUG_CPU_USAGE 213private: 214 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 215 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 216 217 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 218 219 int mCpuNum; // thread's current CPU number 220 int mCpukHz; // frequency of thread's current CPU in kHz 221#endif 222}; 223 224CpuStats::CpuStats() 225#ifdef DEBUG_CPU_USAGE 226 : mCpuNum(-1), mCpukHz(-1) 227#endif 228{ 229} 230 231void CpuStats::sample(const String8 &title 232#ifndef DEBUG_CPU_USAGE 233 __unused 234#endif 235 ) { 236#ifdef DEBUG_CPU_USAGE 237 // get current thread's delta CPU time in wall clock ns 238 double wcNs; 239 bool valid = mCpuUsage.sampleAndEnable(wcNs); 240 241 // record sample for wall clock statistics 242 if (valid) { 243 mWcStats.sample(wcNs); 244 } 245 246 // get the current CPU number 247 int cpuNum = sched_getcpu(); 248 249 // get the current CPU frequency in kHz 250 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 251 252 // check if either CPU number or frequency changed 253 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 254 mCpuNum = cpuNum; 255 mCpukHz = cpukHz; 256 // ignore sample for purposes of cycles 257 valid = false; 258 } 259 260 // if no change in CPU number or frequency, then record sample for cycle statistics 261 if (valid && mCpukHz > 0) { 262 double cycles = wcNs * cpukHz * 0.000001; 263 mHzStats.sample(cycles); 264 } 265 266 unsigned n = mWcStats.n(); 267 // mCpuUsage.elapsed() is expensive, so don't call it every loop 268 if ((n & 127) == 1) { 269 long long elapsed = mCpuUsage.elapsed(); 270 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 271 double perLoop = elapsed / (double) n; 272 double perLoop100 = perLoop * 0.01; 273 double perLoop1k = perLoop * 0.001; 274 double mean = mWcStats.mean(); 275 double stddev = mWcStats.stddev(); 276 double minimum = mWcStats.minimum(); 277 double maximum = mWcStats.maximum(); 278 double meanCycles = mHzStats.mean(); 279 double stddevCycles = mHzStats.stddev(); 280 double minCycles = mHzStats.minimum(); 281 double maxCycles = mHzStats.maximum(); 282 mCpuUsage.resetElapsed(); 283 mWcStats.reset(); 284 mHzStats.reset(); 285 ALOGD("CPU usage for %s over past %.1f secs\n" 286 " (%u mixer loops at %.1f mean ms per loop):\n" 287 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 288 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 289 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 290 title.string(), 291 elapsed * .000000001, n, perLoop * .000001, 292 mean * .001, 293 stddev * .001, 294 minimum * .001, 295 maximum * .001, 296 mean / perLoop100, 297 stddev / perLoop100, 298 minimum / perLoop100, 299 maximum / perLoop100, 300 meanCycles / perLoop1k, 301 stddevCycles / perLoop1k, 302 minCycles / perLoop1k, 303 maxCycles / perLoop1k); 304 305 } 306 } 307#endif 308}; 309 310// ---------------------------------------------------------------------------- 311// ThreadBase 312// ---------------------------------------------------------------------------- 313 314AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 315 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 316 : Thread(false /*canCallJava*/), 317 mType(type), 318 mAudioFlinger(audioFlinger), 319 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 320 // are set by PlaybackThread::readOutputParameters_l() or 321 // RecordThread::readInputParameters_l() 322 //FIXME: mStandby should be true here. Is this some kind of hack? 323 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 324 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 325 // mName will be set by concrete (non-virtual) subclass 326 mDeathRecipient(new PMDeathRecipient(this)) 327{ 328} 329 330AudioFlinger::ThreadBase::~ThreadBase() 331{ 332 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 333 mConfigEvents.clear(); 334 335 // do not lock the mutex in destructor 336 releaseWakeLock_l(); 337 if (mPowerManager != 0) { 338 sp<IBinder> binder = mPowerManager->asBinder(); 339 binder->unlinkToDeath(mDeathRecipient); 340 } 341} 342 343status_t AudioFlinger::ThreadBase::readyToRun() 344{ 345 status_t status = initCheck(); 346 if (status == NO_ERROR) { 347 ALOGI("AudioFlinger's thread %p ready to run", this); 348 } else { 349 ALOGE("No working audio driver found."); 350 } 351 return status; 352} 353 354void AudioFlinger::ThreadBase::exit() 355{ 356 ALOGV("ThreadBase::exit"); 357 // do any cleanup required for exit to succeed 358 preExit(); 359 { 360 // This lock prevents the following race in thread (uniprocessor for illustration): 361 // if (!exitPending()) { 362 // // context switch from here to exit() 363 // // exit() calls requestExit(), what exitPending() observes 364 // // exit() calls signal(), which is dropped since no waiters 365 // // context switch back from exit() to here 366 // mWaitWorkCV.wait(...); 367 // // now thread is hung 368 // } 369 AutoMutex lock(mLock); 370 requestExit(); 371 mWaitWorkCV.broadcast(); 372 } 373 // When Thread::requestExitAndWait is made virtual and this method is renamed to 374 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 375 requestExitAndWait(); 376} 377 378status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 379{ 380 status_t status; 381 382 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 383 Mutex::Autolock _l(mLock); 384 385 return sendSetParameterConfigEvent_l(keyValuePairs); 386} 387 388// sendConfigEvent_l() must be called with ThreadBase::mLock held 389// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 390status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 391{ 392 status_t status = NO_ERROR; 393 394 mConfigEvents.add(event); 395 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 396 mWaitWorkCV.signal(); 397 mLock.unlock(); 398 { 399 Mutex::Autolock _l(event->mLock); 400 while (event->mWaitStatus) { 401 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 402 event->mStatus = TIMED_OUT; 403 event->mWaitStatus = false; 404 } 405 } 406 status = event->mStatus; 407 } 408 mLock.lock(); 409 return status; 410} 411 412void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 413{ 414 Mutex::Autolock _l(mLock); 415 sendIoConfigEvent_l(event, param); 416} 417 418// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 419void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 420{ 421 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 422 sendConfigEvent_l(configEvent); 423} 424 425// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 426void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 427{ 428 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 429 sendConfigEvent_l(configEvent); 430} 431 432// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 433status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 434{ 435 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 436 return sendConfigEvent_l(configEvent); 437} 438 439status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 440 const struct audio_patch *patch, 441 audio_patch_handle_t *handle) 442{ 443 Mutex::Autolock _l(mLock); 444 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 445 status_t status = sendConfigEvent_l(configEvent); 446 if (status == NO_ERROR) { 447 CreateAudioPatchConfigEventData *data = 448 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 449 *handle = data->mHandle; 450 } 451 return status; 452} 453 454status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 455 const audio_patch_handle_t handle) 456{ 457 Mutex::Autolock _l(mLock); 458 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 459 return sendConfigEvent_l(configEvent); 460} 461 462 463// post condition: mConfigEvents.isEmpty() 464void AudioFlinger::ThreadBase::processConfigEvents_l() 465{ 466 bool configChanged = false; 467 468 while (!mConfigEvents.isEmpty()) { 469 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 470 sp<ConfigEvent> event = mConfigEvents[0]; 471 mConfigEvents.removeAt(0); 472 switch (event->mType) { 473 case CFG_EVENT_PRIO: { 474 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 475 // FIXME Need to understand why this has to be done asynchronously 476 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 477 true /*asynchronous*/); 478 if (err != 0) { 479 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 480 data->mPrio, data->mPid, data->mTid, err); 481 } 482 } break; 483 case CFG_EVENT_IO: { 484 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 485 audioConfigChanged(data->mEvent, data->mParam); 486 } break; 487 case CFG_EVENT_SET_PARAMETER: { 488 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 489 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 490 configChanged = true; 491 } 492 } break; 493 case CFG_EVENT_CREATE_AUDIO_PATCH: { 494 CreateAudioPatchConfigEventData *data = 495 (CreateAudioPatchConfigEventData *)event->mData.get(); 496 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 497 } break; 498 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 499 ReleaseAudioPatchConfigEventData *data = 500 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 501 event->mStatus = releaseAudioPatch_l(data->mHandle); 502 } break; 503 default: 504 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 505 break; 506 } 507 { 508 Mutex::Autolock _l(event->mLock); 509 if (event->mWaitStatus) { 510 event->mWaitStatus = false; 511 event->mCond.signal(); 512 } 513 } 514 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 515 } 516 517 if (configChanged) { 518 cacheParameters_l(); 519 } 520} 521 522String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 523 String8 s; 524 if (output) { 525 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 526 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 527 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 528 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 529 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 530 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 531 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 532 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 533 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 534 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 535 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 536 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 537 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 538 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 539 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 540 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 541 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 542 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 543 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 544 } else { 545 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 546 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 547 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 548 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 549 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 550 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 551 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 552 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 553 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 554 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 555 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 556 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 557 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 558 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 559 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 560 } 561 int len = s.length(); 562 if (s.length() > 2) { 563 char *str = s.lockBuffer(len); 564 s.unlockBuffer(len - 2); 565 } 566 return s; 567} 568 569void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 570{ 571 const size_t SIZE = 256; 572 char buffer[SIZE]; 573 String8 result; 574 575 bool locked = AudioFlinger::dumpTryLock(mLock); 576 if (!locked) { 577 dprintf(fd, "thread %p maybe dead locked\n", this); 578 } 579 580 dprintf(fd, " I/O handle: %d\n", mId); 581 dprintf(fd, " TID: %d\n", getTid()); 582 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 583 dprintf(fd, " Sample rate: %u\n", mSampleRate); 584 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 585 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 586 dprintf(fd, " Channel Count: %u\n", mChannelCount); 587 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 588 channelMaskToString(mChannelMask, mType != RECORD).string()); 589 dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 590 dprintf(fd, " Frame size: %zu\n", mFrameSize); 591 dprintf(fd, " Pending config events:"); 592 size_t numConfig = mConfigEvents.size(); 593 if (numConfig) { 594 for (size_t i = 0; i < numConfig; i++) { 595 mConfigEvents[i]->dump(buffer, SIZE); 596 dprintf(fd, "\n %s", buffer); 597 } 598 dprintf(fd, "\n"); 599 } else { 600 dprintf(fd, " none\n"); 601 } 602 603 if (locked) { 604 mLock.unlock(); 605 } 606} 607 608void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 609{ 610 const size_t SIZE = 256; 611 char buffer[SIZE]; 612 String8 result; 613 614 size_t numEffectChains = mEffectChains.size(); 615 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 616 write(fd, buffer, strlen(buffer)); 617 618 for (size_t i = 0; i < numEffectChains; ++i) { 619 sp<EffectChain> chain = mEffectChains[i]; 620 if (chain != 0) { 621 chain->dump(fd, args); 622 } 623 } 624} 625 626void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 627{ 628 Mutex::Autolock _l(mLock); 629 acquireWakeLock_l(uid); 630} 631 632String16 AudioFlinger::ThreadBase::getWakeLockTag() 633{ 634 switch (mType) { 635 case MIXER: 636 return String16("AudioMix"); 637 case DIRECT: 638 return String16("AudioDirectOut"); 639 case DUPLICATING: 640 return String16("AudioDup"); 641 case RECORD: 642 return String16("AudioIn"); 643 case OFFLOAD: 644 return String16("AudioOffload"); 645 default: 646 ALOG_ASSERT(false); 647 return String16("AudioUnknown"); 648 } 649} 650 651void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 652{ 653 getPowerManager_l(); 654 if (mPowerManager != 0) { 655 sp<IBinder> binder = new BBinder(); 656 status_t status; 657 if (uid >= 0) { 658 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 659 binder, 660 getWakeLockTag(), 661 String16("media"), 662 uid); 663 } else { 664 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 665 binder, 666 getWakeLockTag(), 667 String16("media")); 668 } 669 if (status == NO_ERROR) { 670 mWakeLockToken = binder; 671 } 672 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 673 } 674} 675 676void AudioFlinger::ThreadBase::releaseWakeLock() 677{ 678 Mutex::Autolock _l(mLock); 679 releaseWakeLock_l(); 680} 681 682void AudioFlinger::ThreadBase::releaseWakeLock_l() 683{ 684 if (mWakeLockToken != 0) { 685 ALOGV("releaseWakeLock_l() %s", mName); 686 if (mPowerManager != 0) { 687 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 688 } 689 mWakeLockToken.clear(); 690 } 691} 692 693void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 694 Mutex::Autolock _l(mLock); 695 updateWakeLockUids_l(uids); 696} 697 698void AudioFlinger::ThreadBase::getPowerManager_l() { 699 700 if (mPowerManager == 0) { 701 // use checkService() to avoid blocking if power service is not up yet 702 sp<IBinder> binder = 703 defaultServiceManager()->checkService(String16("power")); 704 if (binder == 0) { 705 ALOGW("Thread %s cannot connect to the power manager service", mName); 706 } else { 707 mPowerManager = interface_cast<IPowerManager>(binder); 708 binder->linkToDeath(mDeathRecipient); 709 } 710 } 711} 712 713void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 714 715 getPowerManager_l(); 716 if (mWakeLockToken == NULL) { 717 ALOGE("no wake lock to update!"); 718 return; 719 } 720 if (mPowerManager != 0) { 721 sp<IBinder> binder = new BBinder(); 722 status_t status; 723 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 724 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 725 } 726} 727 728void AudioFlinger::ThreadBase::clearPowerManager() 729{ 730 Mutex::Autolock _l(mLock); 731 releaseWakeLock_l(); 732 mPowerManager.clear(); 733} 734 735void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 736{ 737 sp<ThreadBase> thread = mThread.promote(); 738 if (thread != 0) { 739 thread->clearPowerManager(); 740 } 741 ALOGW("power manager service died !!!"); 742} 743 744void AudioFlinger::ThreadBase::setEffectSuspended( 745 const effect_uuid_t *type, bool suspend, int sessionId) 746{ 747 Mutex::Autolock _l(mLock); 748 setEffectSuspended_l(type, suspend, sessionId); 749} 750 751void AudioFlinger::ThreadBase::setEffectSuspended_l( 752 const effect_uuid_t *type, bool suspend, int sessionId) 753{ 754 sp<EffectChain> chain = getEffectChain_l(sessionId); 755 if (chain != 0) { 756 if (type != NULL) { 757 chain->setEffectSuspended_l(type, suspend); 758 } else { 759 chain->setEffectSuspendedAll_l(suspend); 760 } 761 } 762 763 updateSuspendedSessions_l(type, suspend, sessionId); 764} 765 766void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 767{ 768 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 769 if (index < 0) { 770 return; 771 } 772 773 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 774 mSuspendedSessions.valueAt(index); 775 776 for (size_t i = 0; i < sessionEffects.size(); i++) { 777 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 778 for (int j = 0; j < desc->mRefCount; j++) { 779 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 780 chain->setEffectSuspendedAll_l(true); 781 } else { 782 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 783 desc->mType.timeLow); 784 chain->setEffectSuspended_l(&desc->mType, true); 785 } 786 } 787 } 788} 789 790void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 791 bool suspend, 792 int sessionId) 793{ 794 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 795 796 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 797 798 if (suspend) { 799 if (index >= 0) { 800 sessionEffects = mSuspendedSessions.valueAt(index); 801 } else { 802 mSuspendedSessions.add(sessionId, sessionEffects); 803 } 804 } else { 805 if (index < 0) { 806 return; 807 } 808 sessionEffects = mSuspendedSessions.valueAt(index); 809 } 810 811 812 int key = EffectChain::kKeyForSuspendAll; 813 if (type != NULL) { 814 key = type->timeLow; 815 } 816 index = sessionEffects.indexOfKey(key); 817 818 sp<SuspendedSessionDesc> desc; 819 if (suspend) { 820 if (index >= 0) { 821 desc = sessionEffects.valueAt(index); 822 } else { 823 desc = new SuspendedSessionDesc(); 824 if (type != NULL) { 825 desc->mType = *type; 826 } 827 sessionEffects.add(key, desc); 828 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 829 } 830 desc->mRefCount++; 831 } else { 832 if (index < 0) { 833 return; 834 } 835 desc = sessionEffects.valueAt(index); 836 if (--desc->mRefCount == 0) { 837 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 838 sessionEffects.removeItemsAt(index); 839 if (sessionEffects.isEmpty()) { 840 ALOGV("updateSuspendedSessions_l() restore removing session %d", 841 sessionId); 842 mSuspendedSessions.removeItem(sessionId); 843 } 844 } 845 } 846 if (!sessionEffects.isEmpty()) { 847 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 848 } 849} 850 851void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 852 bool enabled, 853 int sessionId) 854{ 855 Mutex::Autolock _l(mLock); 856 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 857} 858 859void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 860 bool enabled, 861 int sessionId) 862{ 863 if (mType != RECORD) { 864 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 865 // another session. This gives the priority to well behaved effect control panels 866 // and applications not using global effects. 867 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 868 // global effects 869 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 870 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 871 } 872 } 873 874 sp<EffectChain> chain = getEffectChain_l(sessionId); 875 if (chain != 0) { 876 chain->checkSuspendOnEffectEnabled(effect, enabled); 877 } 878} 879 880// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 881sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 882 const sp<AudioFlinger::Client>& client, 883 const sp<IEffectClient>& effectClient, 884 int32_t priority, 885 int sessionId, 886 effect_descriptor_t *desc, 887 int *enabled, 888 status_t *status) 889{ 890 sp<EffectModule> effect; 891 sp<EffectHandle> handle; 892 status_t lStatus; 893 sp<EffectChain> chain; 894 bool chainCreated = false; 895 bool effectCreated = false; 896 bool effectRegistered = false; 897 898 lStatus = initCheck(); 899 if (lStatus != NO_ERROR) { 900 ALOGW("createEffect_l() Audio driver not initialized."); 901 goto Exit; 902 } 903 904 // Reject any effect on Direct output threads for now, since the format of 905 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 906 if (mType == DIRECT) { 907 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 908 desc->name, mName); 909 lStatus = BAD_VALUE; 910 goto Exit; 911 } 912 913 // 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 1149void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1150{ 1151 config->type = AUDIO_PORT_TYPE_MIX; 1152 config->ext.mix.handle = mId; 1153 config->sample_rate = mSampleRate; 1154 config->format = mFormat; 1155 config->channel_mask = mChannelMask; 1156 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1157 AUDIO_PORT_CONFIG_FORMAT; 1158} 1159 1160 1161// ---------------------------------------------------------------------------- 1162// Playback 1163// ---------------------------------------------------------------------------- 1164 1165AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1166 AudioStreamOut* output, 1167 audio_io_handle_t id, 1168 audio_devices_t device, 1169 type_t type) 1170 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1171 mNormalFrameCount(0), mSinkBuffer(NULL), 1172 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1173 mMixerBuffer(NULL), 1174 mMixerBufferSize(0), 1175 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1176 mMixerBufferValid(false), 1177 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1178 mEffectBuffer(NULL), 1179 mEffectBufferSize(0), 1180 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1181 mEffectBufferValid(false), 1182 mSuspended(0), mBytesWritten(0), 1183 mActiveTracksGeneration(0), 1184 // mStreamTypes[] initialized in constructor body 1185 mOutput(output), 1186 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1187 mMixerStatus(MIXER_IDLE), 1188 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1189 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1190 mBytesRemaining(0), 1191 mCurrentWriteLength(0), 1192 mUseAsyncWrite(false), 1193 mWriteAckSequence(0), 1194 mDrainSequence(0), 1195 mSignalPending(false), 1196 mScreenState(AudioFlinger::mScreenState), 1197 // index 0 is reserved for normal mixer's submix 1198 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1199 // mLatchD, mLatchQ, 1200 mLatchDValid(false), mLatchQValid(false) 1201{ 1202 snprintf(mName, kNameLength, "AudioOut_%X", id); 1203 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1204 1205 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1206 // it would be safer to explicitly pass initial masterVolume/masterMute as 1207 // parameter. 1208 // 1209 // If the HAL we are using has support for master volume or master mute, 1210 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1211 // and the mute set to false). 1212 mMasterVolume = audioFlinger->masterVolume_l(); 1213 mMasterMute = audioFlinger->masterMute_l(); 1214 if (mOutput && mOutput->audioHwDev) { 1215 if (mOutput->audioHwDev->canSetMasterVolume()) { 1216 mMasterVolume = 1.0; 1217 } 1218 1219 if (mOutput->audioHwDev->canSetMasterMute()) { 1220 mMasterMute = false; 1221 } 1222 } 1223 1224 readOutputParameters_l(); 1225 1226 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1227 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1228 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1229 stream = (audio_stream_type_t) (stream + 1)) { 1230 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1231 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1232 } 1233 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1234 // because mAudioFlinger doesn't have one to copy from 1235} 1236 1237AudioFlinger::PlaybackThread::~PlaybackThread() 1238{ 1239 mAudioFlinger->unregisterWriter(mNBLogWriter); 1240 free(mSinkBuffer); 1241 free(mMixerBuffer); 1242 free(mEffectBuffer); 1243} 1244 1245void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1246{ 1247 dumpInternals(fd, args); 1248 dumpTracks(fd, args); 1249 dumpEffectChains(fd, args); 1250} 1251 1252void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1253{ 1254 const size_t SIZE = 256; 1255 char buffer[SIZE]; 1256 String8 result; 1257 1258 result.appendFormat(" Stream volumes in dB: "); 1259 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1260 const stream_type_t *st = &mStreamTypes[i]; 1261 if (i > 0) { 1262 result.appendFormat(", "); 1263 } 1264 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1265 if (st->mute) { 1266 result.append("M"); 1267 } 1268 } 1269 result.append("\n"); 1270 write(fd, result.string(), result.length()); 1271 result.clear(); 1272 1273 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1274 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1275 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1276 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1277 1278 size_t numtracks = mTracks.size(); 1279 size_t numactive = mActiveTracks.size(); 1280 dprintf(fd, " %d Tracks", numtracks); 1281 size_t numactiveseen = 0; 1282 if (numtracks) { 1283 dprintf(fd, " of which %d are active\n", numactive); 1284 Track::appendDumpHeader(result); 1285 for (size_t i = 0; i < numtracks; ++i) { 1286 sp<Track> track = mTracks[i]; 1287 if (track != 0) { 1288 bool active = mActiveTracks.indexOf(track) >= 0; 1289 if (active) { 1290 numactiveseen++; 1291 } 1292 track->dump(buffer, SIZE, active); 1293 result.append(buffer); 1294 } 1295 } 1296 } else { 1297 result.append("\n"); 1298 } 1299 if (numactiveseen != numactive) { 1300 // some tracks in the active list were not in the tracks list 1301 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1302 " not in the track list\n"); 1303 result.append(buffer); 1304 Track::appendDumpHeader(result); 1305 for (size_t i = 0; i < numactive; ++i) { 1306 sp<Track> track = mActiveTracks[i].promote(); 1307 if (track != 0 && mTracks.indexOf(track) < 0) { 1308 track->dump(buffer, SIZE, true); 1309 result.append(buffer); 1310 } 1311 } 1312 } 1313 1314 write(fd, result.string(), result.size()); 1315} 1316 1317void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1318{ 1319 dprintf(fd, "\nOutput thread %p:\n", this); 1320 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1321 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1322 dprintf(fd, " Total writes: %d\n", mNumWrites); 1323 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1324 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1325 dprintf(fd, " Suspend count: %d\n", mSuspended); 1326 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1327 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1328 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1329 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1330 1331 dumpBase(fd, args); 1332} 1333 1334// Thread virtuals 1335 1336void AudioFlinger::PlaybackThread::onFirstRef() 1337{ 1338 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1339} 1340 1341// ThreadBase virtuals 1342void AudioFlinger::PlaybackThread::preExit() 1343{ 1344 ALOGV(" preExit()"); 1345 // FIXME this is using hard-coded strings but in the future, this functionality will be 1346 // converted to use audio HAL extensions required to support tunneling 1347 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1348} 1349 1350// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1351sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1352 const sp<AudioFlinger::Client>& client, 1353 audio_stream_type_t streamType, 1354 uint32_t sampleRate, 1355 audio_format_t format, 1356 audio_channel_mask_t channelMask, 1357 size_t *pFrameCount, 1358 const sp<IMemory>& sharedBuffer, 1359 int sessionId, 1360 IAudioFlinger::track_flags_t *flags, 1361 pid_t tid, 1362 int uid, 1363 status_t *status) 1364{ 1365 size_t frameCount = *pFrameCount; 1366 sp<Track> track; 1367 status_t lStatus; 1368 1369 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1370 1371 // client expresses a preference for FAST, but we get the final say 1372 if (*flags & IAudioFlinger::TRACK_FAST) { 1373 if ( 1374 // not timed 1375 (!isTimed) && 1376 // either of these use cases: 1377 ( 1378 // use case 1: shared buffer with any frame count 1379 ( 1380 (sharedBuffer != 0) 1381 ) || 1382 // use case 2: callback handler and frame count is default or at least as large as HAL 1383 ( 1384 (tid != -1) && 1385 ((frameCount == 0) || 1386 (frameCount >= mFrameCount)) 1387 ) 1388 ) && 1389 // PCM data 1390 audio_is_linear_pcm(format) && 1391 // mono or stereo 1392 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1393 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1394 // hardware sample rate 1395 (sampleRate == mSampleRate) && 1396 // normal mixer has an associated fast mixer 1397 hasFastMixer() && 1398 // there are sufficient fast track slots available 1399 (mFastTrackAvailMask != 0) 1400 // FIXME test that MixerThread for this fast track has a capable output HAL 1401 // FIXME add a permission test also? 1402 ) { 1403 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1404 if (frameCount == 0) { 1405 // read the fast track multiplier property the first time it is needed 1406 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1407 if (ok != 0) { 1408 ALOGE("%s pthread_once failed: %d", __func__, ok); 1409 } 1410 frameCount = mFrameCount * sFastTrackMultiplier; 1411 } 1412 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1413 frameCount, mFrameCount); 1414 } else { 1415 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1416 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1417 "sampleRate=%u mSampleRate=%u " 1418 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1419 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1420 audio_is_linear_pcm(format), 1421 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1422 *flags &= ~IAudioFlinger::TRACK_FAST; 1423 // For compatibility with AudioTrack calculation, buffer depth is forced 1424 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1425 // This is probably too conservative, but legacy application code may depend on it. 1426 // If you change this calculation, also review the start threshold which is related. 1427 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1428 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1429 if (minBufCount < 2) { 1430 minBufCount = 2; 1431 } 1432 size_t minFrameCount = mNormalFrameCount * minBufCount; 1433 if (frameCount < minFrameCount) { 1434 frameCount = minFrameCount; 1435 } 1436 } 1437 } 1438 *pFrameCount = frameCount; 1439 1440 switch (mType) { 1441 1442 case DIRECT: 1443 if (audio_is_linear_pcm(format)) { 1444 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1445 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1446 "for output %p with format %#x", 1447 sampleRate, format, channelMask, mOutput, mFormat); 1448 lStatus = BAD_VALUE; 1449 goto Exit; 1450 } 1451 } 1452 break; 1453 1454 case OFFLOAD: 1455 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1456 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1457 "for output %p with format %#x", 1458 sampleRate, format, channelMask, mOutput, mFormat); 1459 lStatus = BAD_VALUE; 1460 goto Exit; 1461 } 1462 break; 1463 1464 default: 1465 if (!audio_is_linear_pcm(format)) { 1466 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1467 "for output %p with format %#x", 1468 format, mOutput, mFormat); 1469 lStatus = BAD_VALUE; 1470 goto Exit; 1471 } 1472 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1473 if (sampleRate > mSampleRate*2) { 1474 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1475 lStatus = BAD_VALUE; 1476 goto Exit; 1477 } 1478 break; 1479 1480 } 1481 1482 lStatus = initCheck(); 1483 if (lStatus != NO_ERROR) { 1484 ALOGE("createTrack_l() audio driver not initialized"); 1485 goto Exit; 1486 } 1487 1488 { // scope for mLock 1489 Mutex::Autolock _l(mLock); 1490 1491 // all tracks in same audio session must share the same routing strategy otherwise 1492 // conflicts will happen when tracks are moved from one output to another by audio policy 1493 // manager 1494 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1495 for (size_t i = 0; i < mTracks.size(); ++i) { 1496 sp<Track> t = mTracks[i]; 1497 if (t != 0 && t->isExternalTrack()) { 1498 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1499 if (sessionId == t->sessionId() && strategy != actual) { 1500 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1501 strategy, actual); 1502 lStatus = BAD_VALUE; 1503 goto Exit; 1504 } 1505 } 1506 } 1507 1508 if (!isTimed) { 1509 track = new Track(this, client, streamType, sampleRate, format, 1510 channelMask, frameCount, NULL, sharedBuffer, 1511 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1512 } else { 1513 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1514 channelMask, frameCount, sharedBuffer, sessionId, uid); 1515 } 1516 1517 // new Track always returns non-NULL, 1518 // but TimedTrack::create() is a factory that could fail by returning NULL 1519 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1520 if (lStatus != NO_ERROR) { 1521 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1522 // track must be cleared from the caller as the caller has the AF lock 1523 goto Exit; 1524 } 1525 mTracks.add(track); 1526 1527 sp<EffectChain> chain = getEffectChain_l(sessionId); 1528 if (chain != 0) { 1529 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1530 track->setMainBuffer(chain->inBuffer()); 1531 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1532 chain->incTrackCnt(); 1533 } 1534 1535 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1536 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1537 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1538 // so ask activity manager to do this on our behalf 1539 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1540 } 1541 } 1542 1543 lStatus = NO_ERROR; 1544 1545Exit: 1546 *status = lStatus; 1547 return track; 1548} 1549 1550uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1551{ 1552 return latency; 1553} 1554 1555uint32_t AudioFlinger::PlaybackThread::latency() const 1556{ 1557 Mutex::Autolock _l(mLock); 1558 return latency_l(); 1559} 1560uint32_t AudioFlinger::PlaybackThread::latency_l() const 1561{ 1562 if (initCheck() == NO_ERROR) { 1563 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1564 } else { 1565 return 0; 1566 } 1567} 1568 1569void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1570{ 1571 Mutex::Autolock _l(mLock); 1572 // Don't apply master volume in SW if our HAL can do it for us. 1573 if (mOutput && mOutput->audioHwDev && 1574 mOutput->audioHwDev->canSetMasterVolume()) { 1575 mMasterVolume = 1.0; 1576 } else { 1577 mMasterVolume = value; 1578 } 1579} 1580 1581void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1582{ 1583 Mutex::Autolock _l(mLock); 1584 // Don't apply master mute in SW if our HAL can do it for us. 1585 if (mOutput && mOutput->audioHwDev && 1586 mOutput->audioHwDev->canSetMasterMute()) { 1587 mMasterMute = false; 1588 } else { 1589 mMasterMute = muted; 1590 } 1591} 1592 1593void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1594{ 1595 Mutex::Autolock _l(mLock); 1596 mStreamTypes[stream].volume = value; 1597 broadcast_l(); 1598} 1599 1600void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1601{ 1602 Mutex::Autolock _l(mLock); 1603 mStreamTypes[stream].mute = muted; 1604 broadcast_l(); 1605} 1606 1607float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1608{ 1609 Mutex::Autolock _l(mLock); 1610 return mStreamTypes[stream].volume; 1611} 1612 1613// addTrack_l() must be called with ThreadBase::mLock held 1614status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1615{ 1616 status_t status = ALREADY_EXISTS; 1617 1618 // set retry count for buffer fill 1619 track->mRetryCount = kMaxTrackStartupRetries; 1620 if (mActiveTracks.indexOf(track) < 0) { 1621 // the track is newly added, make sure it fills up all its 1622 // buffers before playing. This is to ensure the client will 1623 // effectively get the latency it requested. 1624 if (track->isExternalTrack()) { 1625 TrackBase::track_state state = track->mState; 1626 mLock.unlock(); 1627 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1628 mLock.lock(); 1629 // abort track was stopped/paused while we released the lock 1630 if (state != track->mState) { 1631 if (status == NO_ERROR) { 1632 mLock.unlock(); 1633 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1634 mLock.lock(); 1635 } 1636 return INVALID_OPERATION; 1637 } 1638 // abort if start is rejected by audio policy manager 1639 if (status != NO_ERROR) { 1640 return PERMISSION_DENIED; 1641 } 1642#ifdef ADD_BATTERY_DATA 1643 // to track the speaker usage 1644 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1645#endif 1646 } 1647 1648 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1649 track->mResetDone = false; 1650 track->mPresentationCompleteFrames = 0; 1651 mActiveTracks.add(track); 1652 mWakeLockUids.add(track->uid()); 1653 mActiveTracksGeneration++; 1654 mLatestActiveTrack = track; 1655 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1656 if (chain != 0) { 1657 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1658 track->sessionId()); 1659 chain->incActiveTrackCnt(); 1660 } 1661 1662 status = NO_ERROR; 1663 } 1664 1665 onAddNewTrack_l(); 1666 return status; 1667} 1668 1669bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1670{ 1671 track->terminate(); 1672 // active tracks are removed by threadLoop() 1673 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1674 track->mState = TrackBase::STOPPED; 1675 if (!trackActive) { 1676 removeTrack_l(track); 1677 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1678 track->mState = TrackBase::STOPPING_1; 1679 } 1680 1681 return trackActive; 1682} 1683 1684void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1685{ 1686 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1687 mTracks.remove(track); 1688 deleteTrackName_l(track->name()); 1689 // redundant as track is about to be destroyed, for dumpsys only 1690 track->mName = -1; 1691 if (track->isFastTrack()) { 1692 int index = track->mFastIndex; 1693 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1694 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1695 mFastTrackAvailMask |= 1 << index; 1696 // redundant as track is about to be destroyed, for dumpsys only 1697 track->mFastIndex = -1; 1698 } 1699 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1700 if (chain != 0) { 1701 chain->decTrackCnt(); 1702 } 1703} 1704 1705void AudioFlinger::PlaybackThread::broadcast_l() 1706{ 1707 // Thread could be blocked waiting for async 1708 // so signal it to handle state changes immediately 1709 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1710 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1711 mSignalPending = true; 1712 mWaitWorkCV.broadcast(); 1713} 1714 1715String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1716{ 1717 Mutex::Autolock _l(mLock); 1718 if (initCheck() != NO_ERROR) { 1719 return String8(); 1720 } 1721 1722 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1723 const String8 out_s8(s); 1724 free(s); 1725 return out_s8; 1726} 1727 1728void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1729 AudioSystem::OutputDescriptor desc; 1730 void *param2 = NULL; 1731 1732 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1733 param); 1734 1735 switch (event) { 1736 case AudioSystem::OUTPUT_OPENED: 1737 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1738 desc.channelMask = mChannelMask; 1739 desc.samplingRate = mSampleRate; 1740 desc.format = mFormat; 1741 desc.frameCount = mNormalFrameCount; // FIXME see 1742 // AudioFlinger::frameCount(audio_io_handle_t) 1743 desc.latency = latency_l(); 1744 param2 = &desc; 1745 break; 1746 1747 case AudioSystem::STREAM_CONFIG_CHANGED: 1748 param2 = ¶m; 1749 case AudioSystem::OUTPUT_CLOSED: 1750 default: 1751 break; 1752 } 1753 mAudioFlinger->audioConfigChanged(event, mId, param2); 1754} 1755 1756void AudioFlinger::PlaybackThread::writeCallback() 1757{ 1758 ALOG_ASSERT(mCallbackThread != 0); 1759 mCallbackThread->resetWriteBlocked(); 1760} 1761 1762void AudioFlinger::PlaybackThread::drainCallback() 1763{ 1764 ALOG_ASSERT(mCallbackThread != 0); 1765 mCallbackThread->resetDraining(); 1766} 1767 1768void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1769{ 1770 Mutex::Autolock _l(mLock); 1771 // reject out of sequence requests 1772 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1773 mWriteAckSequence &= ~1; 1774 mWaitWorkCV.signal(); 1775 } 1776} 1777 1778void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1779{ 1780 Mutex::Autolock _l(mLock); 1781 // reject out of sequence requests 1782 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1783 mDrainSequence &= ~1; 1784 mWaitWorkCV.signal(); 1785 } 1786} 1787 1788// static 1789int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1790 void *param __unused, 1791 void *cookie) 1792{ 1793 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1794 ALOGV("asyncCallback() event %d", event); 1795 switch (event) { 1796 case STREAM_CBK_EVENT_WRITE_READY: 1797 me->writeCallback(); 1798 break; 1799 case STREAM_CBK_EVENT_DRAIN_READY: 1800 me->drainCallback(); 1801 break; 1802 default: 1803 ALOGW("asyncCallback() unknown event %d", event); 1804 break; 1805 } 1806 return 0; 1807} 1808 1809void AudioFlinger::PlaybackThread::readOutputParameters_l() 1810{ 1811 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1812 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1813 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1814 if (!audio_is_output_channel(mChannelMask)) { 1815 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1816 } 1817 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1818 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1819 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1820 } 1821 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1822 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1823 mFormat = mHALFormat; 1824 if (!audio_is_valid_format(mFormat)) { 1825 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1826 } 1827 if ((mType == MIXER || mType == DUPLICATING) 1828 && !isValidPcmSinkFormat(mFormat)) { 1829 LOG_FATAL("HAL format %#x not supported for mixed output", 1830 mFormat); 1831 } 1832 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1833 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1834 mFrameCount = mBufferSize / mFrameSize; 1835 if (mFrameCount & 15) { 1836 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1837 mFrameCount); 1838 } 1839 1840 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1841 (mOutput->stream->set_callback != NULL)) { 1842 if (mOutput->stream->set_callback(mOutput->stream, 1843 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1844 mUseAsyncWrite = true; 1845 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1846 } 1847 } 1848 1849 // Calculate size of normal sink buffer relative to the HAL output buffer size 1850 double multiplier = 1.0; 1851 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1852 kUseFastMixer == FastMixer_Dynamic)) { 1853 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1854 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1855 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1856 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1857 maxNormalFrameCount = maxNormalFrameCount & ~15; 1858 if (maxNormalFrameCount < minNormalFrameCount) { 1859 maxNormalFrameCount = minNormalFrameCount; 1860 } 1861 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1862 if (multiplier <= 1.0) { 1863 multiplier = 1.0; 1864 } else if (multiplier <= 2.0) { 1865 if (2 * mFrameCount <= maxNormalFrameCount) { 1866 multiplier = 2.0; 1867 } else { 1868 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1869 } 1870 } else { 1871 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1872 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1873 // track, but we sometimes have to do this to satisfy the maximum frame count 1874 // constraint) 1875 // FIXME this rounding up should not be done if no HAL SRC 1876 uint32_t truncMult = (uint32_t) multiplier; 1877 if ((truncMult & 1)) { 1878 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1879 ++truncMult; 1880 } 1881 } 1882 multiplier = (double) truncMult; 1883 } 1884 } 1885 mNormalFrameCount = multiplier * mFrameCount; 1886 // round up to nearest 16 frames to satisfy AudioMixer 1887 if (mType == MIXER || mType == DUPLICATING) { 1888 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1889 } 1890 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1891 mNormalFrameCount); 1892 1893 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1894 // Originally this was int16_t[] array, need to remove legacy implications. 1895 free(mSinkBuffer); 1896 mSinkBuffer = NULL; 1897 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1898 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1899 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1900 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1901 1902 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1903 // drives the output. 1904 free(mMixerBuffer); 1905 mMixerBuffer = NULL; 1906 if (mMixerBufferEnabled) { 1907 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1908 mMixerBufferSize = mNormalFrameCount * mChannelCount 1909 * audio_bytes_per_sample(mMixerBufferFormat); 1910 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1911 } 1912 free(mEffectBuffer); 1913 mEffectBuffer = NULL; 1914 if (mEffectBufferEnabled) { 1915 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1916 mEffectBufferSize = mNormalFrameCount * mChannelCount 1917 * audio_bytes_per_sample(mEffectBufferFormat); 1918 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1919 } 1920 1921 // force reconfiguration of effect chains and engines to take new buffer size and audio 1922 // parameters into account 1923 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1924 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1925 // matter. 1926 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1927 Vector< sp<EffectChain> > effectChains = mEffectChains; 1928 for (size_t i = 0; i < effectChains.size(); i ++) { 1929 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1930 } 1931} 1932 1933 1934status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1935{ 1936 if (halFrames == NULL || dspFrames == NULL) { 1937 return BAD_VALUE; 1938 } 1939 Mutex::Autolock _l(mLock); 1940 if (initCheck() != NO_ERROR) { 1941 return INVALID_OPERATION; 1942 } 1943 size_t framesWritten = mBytesWritten / mFrameSize; 1944 *halFrames = framesWritten; 1945 1946 if (isSuspended()) { 1947 // return an estimation of rendered frames when the output is suspended 1948 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1949 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1950 return NO_ERROR; 1951 } else { 1952 status_t status; 1953 uint32_t frames; 1954 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1955 *dspFrames = (size_t)frames; 1956 return status; 1957 } 1958} 1959 1960uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1961{ 1962 Mutex::Autolock _l(mLock); 1963 uint32_t result = 0; 1964 if (getEffectChain_l(sessionId) != 0) { 1965 result = EFFECT_SESSION; 1966 } 1967 1968 for (size_t i = 0; i < mTracks.size(); ++i) { 1969 sp<Track> track = mTracks[i]; 1970 if (sessionId == track->sessionId() && !track->isInvalid()) { 1971 result |= TRACK_SESSION; 1972 break; 1973 } 1974 } 1975 1976 return result; 1977} 1978 1979uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1980{ 1981 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1982 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1983 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1984 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1985 } 1986 for (size_t i = 0; i < mTracks.size(); i++) { 1987 sp<Track> track = mTracks[i]; 1988 if (sessionId == track->sessionId() && !track->isInvalid()) { 1989 return AudioSystem::getStrategyForStream(track->streamType()); 1990 } 1991 } 1992 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1993} 1994 1995 1996AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1997{ 1998 Mutex::Autolock _l(mLock); 1999 return mOutput; 2000} 2001 2002AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2003{ 2004 Mutex::Autolock _l(mLock); 2005 AudioStreamOut *output = mOutput; 2006 mOutput = NULL; 2007 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2008 // must push a NULL and wait for ack 2009 mOutputSink.clear(); 2010 mPipeSink.clear(); 2011 mNormalSink.clear(); 2012 return output; 2013} 2014 2015// this method must always be called either with ThreadBase mLock held or inside the thread loop 2016audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2017{ 2018 if (mOutput == NULL) { 2019 return NULL; 2020 } 2021 return &mOutput->stream->common; 2022} 2023 2024uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2025{ 2026 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2027} 2028 2029status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2030{ 2031 if (!isValidSyncEvent(event)) { 2032 return BAD_VALUE; 2033 } 2034 2035 Mutex::Autolock _l(mLock); 2036 2037 for (size_t i = 0; i < mTracks.size(); ++i) { 2038 sp<Track> track = mTracks[i]; 2039 if (event->triggerSession() == track->sessionId()) { 2040 (void) track->setSyncEvent(event); 2041 return NO_ERROR; 2042 } 2043 } 2044 2045 return NAME_NOT_FOUND; 2046} 2047 2048bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2049{ 2050 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2051} 2052 2053void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2054 const Vector< sp<Track> >& tracksToRemove) 2055{ 2056 size_t count = tracksToRemove.size(); 2057 if (count > 0) { 2058 for (size_t i = 0 ; i < count ; i++) { 2059 const sp<Track>& track = tracksToRemove.itemAt(i); 2060 if (track->isExternalTrack()) { 2061 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2062#ifdef ADD_BATTERY_DATA 2063 // to track the speaker usage 2064 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2065#endif 2066 if (track->isTerminated()) { 2067 AudioSystem::releaseOutput(mId); 2068 } 2069 } 2070 } 2071 } 2072} 2073 2074void AudioFlinger::PlaybackThread::checkSilentMode_l() 2075{ 2076 if (!mMasterMute) { 2077 char value[PROPERTY_VALUE_MAX]; 2078 if (property_get("ro.audio.silent", value, "0") > 0) { 2079 char *endptr; 2080 unsigned long ul = strtoul(value, &endptr, 0); 2081 if (*endptr == '\0' && ul != 0) { 2082 ALOGD("Silence is golden"); 2083 // The setprop command will not allow a property to be changed after 2084 // the first time it is set, so we don't have to worry about un-muting. 2085 setMasterMute_l(true); 2086 } 2087 } 2088 } 2089} 2090 2091// shared by MIXER and DIRECT, overridden by DUPLICATING 2092ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2093{ 2094 // FIXME rewrite to reduce number of system calls 2095 mLastWriteTime = systemTime(); 2096 mInWrite = true; 2097 ssize_t bytesWritten; 2098 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2099 2100 // If an NBAIO sink is present, use it to write the normal mixer's submix 2101 if (mNormalSink != 0) { 2102 const size_t count = mBytesRemaining / mFrameSize; 2103 2104 ATRACE_BEGIN("write"); 2105 // update the setpoint when AudioFlinger::mScreenState changes 2106 uint32_t screenState = AudioFlinger::mScreenState; 2107 if (screenState != mScreenState) { 2108 mScreenState = screenState; 2109 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2110 if (pipe != NULL) { 2111 pipe->setAvgFrames((mScreenState & 1) ? 2112 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2113 } 2114 } 2115 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2116 ATRACE_END(); 2117 if (framesWritten > 0) { 2118 bytesWritten = framesWritten * mFrameSize; 2119 } else { 2120 bytesWritten = framesWritten; 2121 } 2122 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2123 if (status == NO_ERROR) { 2124 size_t totalFramesWritten = mNormalSink->framesWritten(); 2125 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2126 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2127 mLatchDValid = true; 2128 } 2129 } 2130 // otherwise use the HAL / AudioStreamOut directly 2131 } else { 2132 // Direct output and offload threads 2133 2134 if (mUseAsyncWrite) { 2135 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2136 mWriteAckSequence += 2; 2137 mWriteAckSequence |= 1; 2138 ALOG_ASSERT(mCallbackThread != 0); 2139 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2140 } 2141 // FIXME We should have an implementation of timestamps for direct output threads. 2142 // They are used e.g for multichannel PCM playback over HDMI. 2143 bytesWritten = mOutput->stream->write(mOutput->stream, 2144 (char *)mSinkBuffer + offset, mBytesRemaining); 2145 if (mUseAsyncWrite && 2146 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2147 // do not wait for async callback in case of error of full write 2148 mWriteAckSequence &= ~1; 2149 ALOG_ASSERT(mCallbackThread != 0); 2150 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2151 } 2152 } 2153 2154 mNumWrites++; 2155 mInWrite = false; 2156 mStandby = false; 2157 return bytesWritten; 2158} 2159 2160void AudioFlinger::PlaybackThread::threadLoop_drain() 2161{ 2162 if (mOutput->stream->drain) { 2163 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2164 if (mUseAsyncWrite) { 2165 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2166 mDrainSequence |= 1; 2167 ALOG_ASSERT(mCallbackThread != 0); 2168 mCallbackThread->setDraining(mDrainSequence); 2169 } 2170 mOutput->stream->drain(mOutput->stream, 2171 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2172 : AUDIO_DRAIN_ALL); 2173 } 2174} 2175 2176void AudioFlinger::PlaybackThread::threadLoop_exit() 2177{ 2178 // Default implementation has nothing to do 2179} 2180 2181/* 2182The derived values that are cached: 2183 - mSinkBufferSize from frame count * frame size 2184 - activeSleepTime from activeSleepTimeUs() 2185 - idleSleepTime from idleSleepTimeUs() 2186 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2187 - maxPeriod from frame count and sample rate (MIXER only) 2188 2189The parameters that affect these derived values are: 2190 - frame count 2191 - frame size 2192 - sample rate 2193 - device type: A2DP or not 2194 - device latency 2195 - format: PCM or not 2196 - active sleep time 2197 - idle sleep time 2198*/ 2199 2200void AudioFlinger::PlaybackThread::cacheParameters_l() 2201{ 2202 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2203 activeSleepTime = activeSleepTimeUs(); 2204 idleSleepTime = idleSleepTimeUs(); 2205} 2206 2207void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2208{ 2209 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2210 this, streamType, mTracks.size()); 2211 Mutex::Autolock _l(mLock); 2212 2213 size_t size = mTracks.size(); 2214 for (size_t i = 0; i < size; i++) { 2215 sp<Track> t = mTracks[i]; 2216 if (t->streamType() == streamType) { 2217 t->invalidate(); 2218 } 2219 } 2220} 2221 2222status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2223{ 2224 int session = chain->sessionId(); 2225 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2226 ? mEffectBuffer : mSinkBuffer); 2227 bool ownsBuffer = false; 2228 2229 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2230 if (session > 0) { 2231 // Only one effect chain can be present in direct output thread and it uses 2232 // the sink buffer as input 2233 if (mType != DIRECT) { 2234 size_t numSamples = mNormalFrameCount * mChannelCount; 2235 buffer = new int16_t[numSamples]; 2236 memset(buffer, 0, numSamples * sizeof(int16_t)); 2237 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2238 ownsBuffer = true; 2239 } 2240 2241 // Attach all tracks with same session ID to this chain. 2242 for (size_t i = 0; i < mTracks.size(); ++i) { 2243 sp<Track> track = mTracks[i]; 2244 if (session == track->sessionId()) { 2245 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2246 buffer); 2247 track->setMainBuffer(buffer); 2248 chain->incTrackCnt(); 2249 } 2250 } 2251 2252 // indicate all active tracks in the chain 2253 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2254 sp<Track> track = mActiveTracks[i].promote(); 2255 if (track == 0) { 2256 continue; 2257 } 2258 if (session == track->sessionId()) { 2259 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2260 chain->incActiveTrackCnt(); 2261 } 2262 } 2263 } 2264 2265 chain->setInBuffer(buffer, ownsBuffer); 2266 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2267 ? mEffectBuffer : mSinkBuffer)); 2268 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2269 // chains list in order to be processed last as it contains output stage effects 2270 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2271 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2272 // after track specific effects and before output stage 2273 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2274 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2275 // Effect chain for other sessions are inserted at beginning of effect 2276 // chains list to be processed before output mix effects. Relative order between other 2277 // sessions is not important 2278 size_t size = mEffectChains.size(); 2279 size_t i = 0; 2280 for (i = 0; i < size; i++) { 2281 if (mEffectChains[i]->sessionId() < session) { 2282 break; 2283 } 2284 } 2285 mEffectChains.insertAt(chain, i); 2286 checkSuspendOnAddEffectChain_l(chain); 2287 2288 return NO_ERROR; 2289} 2290 2291size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2292{ 2293 int session = chain->sessionId(); 2294 2295 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2296 2297 for (size_t i = 0; i < mEffectChains.size(); i++) { 2298 if (chain == mEffectChains[i]) { 2299 mEffectChains.removeAt(i); 2300 // detach all active tracks from the chain 2301 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2302 sp<Track> track = mActiveTracks[i].promote(); 2303 if (track == 0) { 2304 continue; 2305 } 2306 if (session == track->sessionId()) { 2307 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2308 chain.get(), session); 2309 chain->decActiveTrackCnt(); 2310 } 2311 } 2312 2313 // detach all tracks with same session ID from this chain 2314 for (size_t i = 0; i < mTracks.size(); ++i) { 2315 sp<Track> track = mTracks[i]; 2316 if (session == track->sessionId()) { 2317 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2318 chain->decTrackCnt(); 2319 } 2320 } 2321 break; 2322 } 2323 } 2324 return mEffectChains.size(); 2325} 2326 2327status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2328 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2329{ 2330 Mutex::Autolock _l(mLock); 2331 return attachAuxEffect_l(track, EffectId); 2332} 2333 2334status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2335 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2336{ 2337 status_t status = NO_ERROR; 2338 2339 if (EffectId == 0) { 2340 track->setAuxBuffer(0, NULL); 2341 } else { 2342 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2343 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2344 if (effect != 0) { 2345 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2346 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2347 } else { 2348 status = INVALID_OPERATION; 2349 } 2350 } else { 2351 status = BAD_VALUE; 2352 } 2353 } 2354 return status; 2355} 2356 2357void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2358{ 2359 for (size_t i = 0; i < mTracks.size(); ++i) { 2360 sp<Track> track = mTracks[i]; 2361 if (track->auxEffectId() == effectId) { 2362 attachAuxEffect_l(track, 0); 2363 } 2364 } 2365} 2366 2367bool AudioFlinger::PlaybackThread::threadLoop() 2368{ 2369 Vector< sp<Track> > tracksToRemove; 2370 2371 standbyTime = systemTime(); 2372 2373 // MIXER 2374 nsecs_t lastWarning = 0; 2375 2376 // DUPLICATING 2377 // FIXME could this be made local to while loop? 2378 writeFrames = 0; 2379 2380 int lastGeneration = 0; 2381 2382 cacheParameters_l(); 2383 sleepTime = idleSleepTime; 2384 2385 if (mType == MIXER) { 2386 sleepTimeShift = 0; 2387 } 2388 2389 CpuStats cpuStats; 2390 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2391 2392 acquireWakeLock(); 2393 2394 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2395 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2396 // and then that string will be logged at the next convenient opportunity. 2397 const char *logString = NULL; 2398 2399 checkSilentMode_l(); 2400 2401 while (!exitPending()) 2402 { 2403 cpuStats.sample(myName); 2404 2405 Vector< sp<EffectChain> > effectChains; 2406 2407 { // scope for mLock 2408 2409 Mutex::Autolock _l(mLock); 2410 2411 processConfigEvents_l(); 2412 2413 if (logString != NULL) { 2414 mNBLogWriter->logTimestamp(); 2415 mNBLogWriter->log(logString); 2416 logString = NULL; 2417 } 2418 2419 if (mLatchDValid) { 2420 mLatchQ = mLatchD; 2421 mLatchDValid = false; 2422 mLatchQValid = true; 2423 } 2424 2425 saveOutputTracks(); 2426 if (mSignalPending) { 2427 // A signal was raised while we were unlocked 2428 mSignalPending = false; 2429 } else if (waitingAsyncCallback_l()) { 2430 if (exitPending()) { 2431 break; 2432 } 2433 releaseWakeLock_l(); 2434 mWakeLockUids.clear(); 2435 mActiveTracksGeneration++; 2436 ALOGV("wait async completion"); 2437 mWaitWorkCV.wait(mLock); 2438 ALOGV("async completion/wake"); 2439 acquireWakeLock_l(); 2440 standbyTime = systemTime() + standbyDelay; 2441 sleepTime = 0; 2442 2443 continue; 2444 } 2445 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2446 isSuspended()) { 2447 // put audio hardware into standby after short delay 2448 if (shouldStandby_l()) { 2449 2450 threadLoop_standby(); 2451 2452 mStandby = true; 2453 } 2454 2455 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2456 // we're about to wait, flush the binder command buffer 2457 IPCThreadState::self()->flushCommands(); 2458 2459 clearOutputTracks(); 2460 2461 if (exitPending()) { 2462 break; 2463 } 2464 2465 releaseWakeLock_l(); 2466 mWakeLockUids.clear(); 2467 mActiveTracksGeneration++; 2468 // wait until we have something to do... 2469 ALOGV("%s going to sleep", myName.string()); 2470 mWaitWorkCV.wait(mLock); 2471 ALOGV("%s waking up", myName.string()); 2472 acquireWakeLock_l(); 2473 2474 mMixerStatus = MIXER_IDLE; 2475 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2476 mBytesWritten = 0; 2477 mBytesRemaining = 0; 2478 checkSilentMode_l(); 2479 2480 standbyTime = systemTime() + standbyDelay; 2481 sleepTime = idleSleepTime; 2482 if (mType == MIXER) { 2483 sleepTimeShift = 0; 2484 } 2485 2486 continue; 2487 } 2488 } 2489 // mMixerStatusIgnoringFastTracks is also updated internally 2490 mMixerStatus = prepareTracks_l(&tracksToRemove); 2491 2492 // compare with previously applied list 2493 if (lastGeneration != mActiveTracksGeneration) { 2494 // update wakelock 2495 updateWakeLockUids_l(mWakeLockUids); 2496 lastGeneration = mActiveTracksGeneration; 2497 } 2498 2499 // prevent any changes in effect chain list and in each effect chain 2500 // during mixing and effect process as the audio buffers could be deleted 2501 // or modified if an effect is created or deleted 2502 lockEffectChains_l(effectChains); 2503 } // mLock scope ends 2504 2505 if (mBytesRemaining == 0) { 2506 mCurrentWriteLength = 0; 2507 if (mMixerStatus == MIXER_TRACKS_READY) { 2508 // threadLoop_mix() sets mCurrentWriteLength 2509 threadLoop_mix(); 2510 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2511 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2512 // threadLoop_sleepTime sets sleepTime to 0 if data 2513 // must be written to HAL 2514 threadLoop_sleepTime(); 2515 if (sleepTime == 0) { 2516 mCurrentWriteLength = mSinkBufferSize; 2517 } 2518 } 2519 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2520 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2521 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2522 // or mSinkBuffer (if there are no effects). 2523 // 2524 // This is done pre-effects computation; if effects change to 2525 // support higher precision, this needs to move. 2526 // 2527 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2528 // TODO use sleepTime == 0 as an additional condition. 2529 if (mMixerBufferValid) { 2530 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2531 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2532 2533 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2534 mNormalFrameCount * mChannelCount); 2535 } 2536 2537 mBytesRemaining = mCurrentWriteLength; 2538 if (isSuspended()) { 2539 sleepTime = suspendSleepTimeUs(); 2540 // simulate write to HAL when suspended 2541 mBytesWritten += mSinkBufferSize; 2542 mBytesRemaining = 0; 2543 } 2544 2545 // only process effects if we're going to write 2546 if (sleepTime == 0 && mType != OFFLOAD) { 2547 for (size_t i = 0; i < effectChains.size(); i ++) { 2548 effectChains[i]->process_l(); 2549 } 2550 } 2551 } 2552 // Process effect chains for offloaded thread even if no audio 2553 // was read from audio track: process only updates effect state 2554 // and thus does have to be synchronized with audio writes but may have 2555 // to be called while waiting for async write callback 2556 if (mType == OFFLOAD) { 2557 for (size_t i = 0; i < effectChains.size(); i ++) { 2558 effectChains[i]->process_l(); 2559 } 2560 } 2561 2562 // Only if the Effects buffer is enabled and there is data in the 2563 // Effects buffer (buffer valid), we need to 2564 // copy into the sink buffer. 2565 // TODO use sleepTime == 0 as an additional condition. 2566 if (mEffectBufferValid) { 2567 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2568 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2569 mNormalFrameCount * mChannelCount); 2570 } 2571 2572 // enable changes in effect chain 2573 unlockEffectChains(effectChains); 2574 2575 if (!waitingAsyncCallback()) { 2576 // sleepTime == 0 means we must write to audio hardware 2577 if (sleepTime == 0) { 2578 if (mBytesRemaining) { 2579 ssize_t ret = threadLoop_write(); 2580 if (ret < 0) { 2581 mBytesRemaining = 0; 2582 } else { 2583 mBytesWritten += ret; 2584 mBytesRemaining -= ret; 2585 } 2586 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2587 (mMixerStatus == MIXER_DRAIN_ALL)) { 2588 threadLoop_drain(); 2589 } 2590 if (mType == MIXER) { 2591 // write blocked detection 2592 nsecs_t now = systemTime(); 2593 nsecs_t delta = now - mLastWriteTime; 2594 if (!mStandby && delta > maxPeriod) { 2595 mNumDelayedWrites++; 2596 if ((now - lastWarning) > kWarningThrottleNs) { 2597 ATRACE_NAME("underrun"); 2598 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2599 ns2ms(delta), mNumDelayedWrites, this); 2600 lastWarning = now; 2601 } 2602 } 2603 } 2604 2605 } else { 2606 usleep(sleepTime); 2607 } 2608 } 2609 2610 // Finally let go of removed track(s), without the lock held 2611 // since we can't guarantee the destructors won't acquire that 2612 // same lock. This will also mutate and push a new fast mixer state. 2613 threadLoop_removeTracks(tracksToRemove); 2614 tracksToRemove.clear(); 2615 2616 // FIXME I don't understand the need for this here; 2617 // it was in the original code but maybe the 2618 // assignment in saveOutputTracks() makes this unnecessary? 2619 clearOutputTracks(); 2620 2621 // Effect chains will be actually deleted here if they were removed from 2622 // mEffectChains list during mixing or effects processing 2623 effectChains.clear(); 2624 2625 // FIXME Note that the above .clear() is no longer necessary since effectChains 2626 // is now local to this block, but will keep it for now (at least until merge done). 2627 } 2628 2629 threadLoop_exit(); 2630 2631 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2632 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2633 // put output stream into standby mode 2634 if (!mStandby) { 2635 mOutput->stream->common.standby(&mOutput->stream->common); 2636 } 2637 } 2638 2639 releaseWakeLock(); 2640 mWakeLockUids.clear(); 2641 mActiveTracksGeneration++; 2642 2643 ALOGV("Thread %p type %d exiting", this, mType); 2644 return false; 2645} 2646 2647// removeTracks_l() must be called with ThreadBase::mLock held 2648void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2649{ 2650 size_t count = tracksToRemove.size(); 2651 if (count > 0) { 2652 for (size_t i=0 ; i<count ; i++) { 2653 const sp<Track>& track = tracksToRemove.itemAt(i); 2654 mActiveTracks.remove(track); 2655 mWakeLockUids.remove(track->uid()); 2656 mActiveTracksGeneration++; 2657 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2658 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2659 if (chain != 0) { 2660 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2661 track->sessionId()); 2662 chain->decActiveTrackCnt(); 2663 } 2664 if (track->isTerminated()) { 2665 removeTrack_l(track); 2666 } 2667 } 2668 } 2669 2670} 2671 2672status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2673{ 2674 if (mNormalSink != 0) { 2675 return mNormalSink->getTimestamp(timestamp); 2676 } 2677 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2678 uint64_t position64; 2679 int ret = mOutput->stream->get_presentation_position( 2680 mOutput->stream, &position64, ×tamp.mTime); 2681 if (ret == 0) { 2682 timestamp.mPosition = (uint32_t)position64; 2683 return NO_ERROR; 2684 } 2685 } 2686 return INVALID_OPERATION; 2687} 2688 2689status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2690 audio_patch_handle_t *handle) 2691{ 2692 status_t status = NO_ERROR; 2693 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2694 // store new device and send to effects 2695 audio_devices_t type = AUDIO_DEVICE_NONE; 2696 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2697 type |= patch->sinks[i].ext.device.type; 2698 } 2699 mOutDevice = type; 2700 for (size_t i = 0; i < mEffectChains.size(); i++) { 2701 mEffectChains[i]->setDevice_l(mOutDevice); 2702 } 2703 2704 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2705 status = hwDevice->create_audio_patch(hwDevice, 2706 patch->num_sources, 2707 patch->sources, 2708 patch->num_sinks, 2709 patch->sinks, 2710 handle); 2711 } else { 2712 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2713 } 2714 return status; 2715} 2716 2717status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2718{ 2719 status_t status = NO_ERROR; 2720 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2721 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2722 status = hwDevice->release_audio_patch(hwDevice, handle); 2723 } else { 2724 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2725 } 2726 return status; 2727} 2728 2729void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2730{ 2731 Mutex::Autolock _l(mLock); 2732 mTracks.add(track); 2733} 2734 2735void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2736{ 2737 Mutex::Autolock _l(mLock); 2738 destroyTrack_l(track); 2739} 2740 2741void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2742{ 2743 ThreadBase::getAudioPortConfig(config); 2744 config->role = AUDIO_PORT_ROLE_SOURCE; 2745 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2746 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2747} 2748 2749// ---------------------------------------------------------------------------- 2750 2751AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2752 audio_io_handle_t id, audio_devices_t device, type_t type) 2753 : PlaybackThread(audioFlinger, output, id, device, type), 2754 // mAudioMixer below 2755 // mFastMixer below 2756 mFastMixerFutex(0) 2757 // mOutputSink below 2758 // mPipeSink below 2759 // mNormalSink below 2760{ 2761 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2762 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2763 "mFrameCount=%d, mNormalFrameCount=%d", 2764 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2765 mNormalFrameCount); 2766 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2767 2768 // FIXME - Current mixer implementation only supports stereo output 2769 if (mChannelCount != FCC_2) { 2770 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2771 } 2772 2773 // create an NBAIO sink for the HAL output stream, and negotiate 2774 mOutputSink = new AudioStreamOutSink(output->stream); 2775 size_t numCounterOffers = 0; 2776 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2777 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2778 ALOG_ASSERT(index == 0); 2779 2780 // initialize fast mixer depending on configuration 2781 bool initFastMixer; 2782 switch (kUseFastMixer) { 2783 case FastMixer_Never: 2784 initFastMixer = false; 2785 break; 2786 case FastMixer_Always: 2787 initFastMixer = true; 2788 break; 2789 case FastMixer_Static: 2790 case FastMixer_Dynamic: 2791 initFastMixer = mFrameCount < mNormalFrameCount; 2792 break; 2793 } 2794 if (initFastMixer) { 2795 audio_format_t fastMixerFormat; 2796 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2797 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2798 } else { 2799 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2800 } 2801 if (mFormat != fastMixerFormat) { 2802 // change our Sink format to accept our intermediate precision 2803 mFormat = fastMixerFormat; 2804 free(mSinkBuffer); 2805 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2806 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2807 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2808 } 2809 2810 // create a MonoPipe to connect our submix to FastMixer 2811 NBAIO_Format format = mOutputSink->format(); 2812 // adjust format to match that of the Fast Mixer 2813 format.mFormat = fastMixerFormat; 2814 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2815 2816 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2817 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2818 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2819 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2820 const NBAIO_Format offers[1] = {format}; 2821 size_t numCounterOffers = 0; 2822 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2823 ALOG_ASSERT(index == 0); 2824 monoPipe->setAvgFrames((mScreenState & 1) ? 2825 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2826 mPipeSink = monoPipe; 2827 2828#ifdef TEE_SINK 2829 if (mTeeSinkOutputEnabled) { 2830 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2831 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2832 numCounterOffers = 0; 2833 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2834 ALOG_ASSERT(index == 0); 2835 mTeeSink = teeSink; 2836 PipeReader *teeSource = new PipeReader(*teeSink); 2837 numCounterOffers = 0; 2838 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2839 ALOG_ASSERT(index == 0); 2840 mTeeSource = teeSource; 2841 } 2842#endif 2843 2844 // create fast mixer and configure it initially with just one fast track for our submix 2845 mFastMixer = new FastMixer(); 2846 FastMixerStateQueue *sq = mFastMixer->sq(); 2847#ifdef STATE_QUEUE_DUMP 2848 sq->setObserverDump(&mStateQueueObserverDump); 2849 sq->setMutatorDump(&mStateQueueMutatorDump); 2850#endif 2851 FastMixerState *state = sq->begin(); 2852 FastTrack *fastTrack = &state->mFastTracks[0]; 2853 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2854 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2855 fastTrack->mVolumeProvider = NULL; 2856 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2857 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2858 fastTrack->mGeneration++; 2859 state->mFastTracksGen++; 2860 state->mTrackMask = 1; 2861 // fast mixer will use the HAL output sink 2862 state->mOutputSink = mOutputSink.get(); 2863 state->mOutputSinkGen++; 2864 state->mFrameCount = mFrameCount; 2865 state->mCommand = FastMixerState::COLD_IDLE; 2866 // already done in constructor initialization list 2867 //mFastMixerFutex = 0; 2868 state->mColdFutexAddr = &mFastMixerFutex; 2869 state->mColdGen++; 2870 state->mDumpState = &mFastMixerDumpState; 2871#ifdef TEE_SINK 2872 state->mTeeSink = mTeeSink.get(); 2873#endif 2874 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2875 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2876 sq->end(); 2877 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2878 2879 // start the fast mixer 2880 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2881 pid_t tid = mFastMixer->getTid(); 2882 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2883 if (err != 0) { 2884 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2885 kPriorityFastMixer, getpid_cached, tid, err); 2886 } 2887 2888#ifdef AUDIO_WATCHDOG 2889 // create and start the watchdog 2890 mAudioWatchdog = new AudioWatchdog(); 2891 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2892 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2893 tid = mAudioWatchdog->getTid(); 2894 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2895 if (err != 0) { 2896 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2897 kPriorityFastMixer, getpid_cached, tid, err); 2898 } 2899#endif 2900 2901 } 2902 2903 switch (kUseFastMixer) { 2904 case FastMixer_Never: 2905 case FastMixer_Dynamic: 2906 mNormalSink = mOutputSink; 2907 break; 2908 case FastMixer_Always: 2909 mNormalSink = mPipeSink; 2910 break; 2911 case FastMixer_Static: 2912 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2913 break; 2914 } 2915} 2916 2917AudioFlinger::MixerThread::~MixerThread() 2918{ 2919 if (mFastMixer != 0) { 2920 FastMixerStateQueue *sq = mFastMixer->sq(); 2921 FastMixerState *state = sq->begin(); 2922 if (state->mCommand == FastMixerState::COLD_IDLE) { 2923 int32_t old = android_atomic_inc(&mFastMixerFutex); 2924 if (old == -1) { 2925 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2926 } 2927 } 2928 state->mCommand = FastMixerState::EXIT; 2929 sq->end(); 2930 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2931 mFastMixer->join(); 2932 // Though the fast mixer thread has exited, it's state queue is still valid. 2933 // We'll use that extract the final state which contains one remaining fast track 2934 // corresponding to our sub-mix. 2935 state = sq->begin(); 2936 ALOG_ASSERT(state->mTrackMask == 1); 2937 FastTrack *fastTrack = &state->mFastTracks[0]; 2938 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2939 delete fastTrack->mBufferProvider; 2940 sq->end(false /*didModify*/); 2941 mFastMixer.clear(); 2942#ifdef AUDIO_WATCHDOG 2943 if (mAudioWatchdog != 0) { 2944 mAudioWatchdog->requestExit(); 2945 mAudioWatchdog->requestExitAndWait(); 2946 mAudioWatchdog.clear(); 2947 } 2948#endif 2949 } 2950 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2951 delete mAudioMixer; 2952} 2953 2954 2955uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2956{ 2957 if (mFastMixer != 0) { 2958 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2959 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2960 } 2961 return latency; 2962} 2963 2964 2965void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2966{ 2967 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2968} 2969 2970ssize_t AudioFlinger::MixerThread::threadLoop_write() 2971{ 2972 // FIXME we should only do one push per cycle; confirm this is true 2973 // Start the fast mixer if it's not already running 2974 if (mFastMixer != 0) { 2975 FastMixerStateQueue *sq = mFastMixer->sq(); 2976 FastMixerState *state = sq->begin(); 2977 if (state->mCommand != FastMixerState::MIX_WRITE && 2978 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2979 if (state->mCommand == FastMixerState::COLD_IDLE) { 2980 int32_t old = android_atomic_inc(&mFastMixerFutex); 2981 if (old == -1) { 2982 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2983 } 2984#ifdef AUDIO_WATCHDOG 2985 if (mAudioWatchdog != 0) { 2986 mAudioWatchdog->resume(); 2987 } 2988#endif 2989 } 2990 state->mCommand = FastMixerState::MIX_WRITE; 2991 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2992 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2993 sq->end(); 2994 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2995 if (kUseFastMixer == FastMixer_Dynamic) { 2996 mNormalSink = mPipeSink; 2997 } 2998 } else { 2999 sq->end(false /*didModify*/); 3000 } 3001 } 3002 return PlaybackThread::threadLoop_write(); 3003} 3004 3005void AudioFlinger::MixerThread::threadLoop_standby() 3006{ 3007 // Idle the fast mixer if it's currently running 3008 if (mFastMixer != 0) { 3009 FastMixerStateQueue *sq = mFastMixer->sq(); 3010 FastMixerState *state = sq->begin(); 3011 if (!(state->mCommand & FastMixerState::IDLE)) { 3012 state->mCommand = FastMixerState::COLD_IDLE; 3013 state->mColdFutexAddr = &mFastMixerFutex; 3014 state->mColdGen++; 3015 mFastMixerFutex = 0; 3016 sq->end(); 3017 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3018 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3019 if (kUseFastMixer == FastMixer_Dynamic) { 3020 mNormalSink = mOutputSink; 3021 } 3022#ifdef AUDIO_WATCHDOG 3023 if (mAudioWatchdog != 0) { 3024 mAudioWatchdog->pause(); 3025 } 3026#endif 3027 } else { 3028 sq->end(false /*didModify*/); 3029 } 3030 } 3031 PlaybackThread::threadLoop_standby(); 3032} 3033 3034bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3035{ 3036 return false; 3037} 3038 3039bool AudioFlinger::PlaybackThread::shouldStandby_l() 3040{ 3041 return !mStandby; 3042} 3043 3044bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3045{ 3046 Mutex::Autolock _l(mLock); 3047 return waitingAsyncCallback_l(); 3048} 3049 3050// shared by MIXER and DIRECT, overridden by DUPLICATING 3051void AudioFlinger::PlaybackThread::threadLoop_standby() 3052{ 3053 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3054 mOutput->stream->common.standby(&mOutput->stream->common); 3055 if (mUseAsyncWrite != 0) { 3056 // discard any pending drain or write ack by incrementing sequence 3057 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3058 mDrainSequence = (mDrainSequence + 2) & ~1; 3059 ALOG_ASSERT(mCallbackThread != 0); 3060 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3061 mCallbackThread->setDraining(mDrainSequence); 3062 } 3063} 3064 3065void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3066{ 3067 ALOGV("signal playback thread"); 3068 broadcast_l(); 3069} 3070 3071void AudioFlinger::MixerThread::threadLoop_mix() 3072{ 3073 // obtain the presentation timestamp of the next output buffer 3074 int64_t pts; 3075 status_t status = INVALID_OPERATION; 3076 3077 if (mNormalSink != 0) { 3078 status = mNormalSink->getNextWriteTimestamp(&pts); 3079 } else { 3080 status = mOutputSink->getNextWriteTimestamp(&pts); 3081 } 3082 3083 if (status != NO_ERROR) { 3084 pts = AudioBufferProvider::kInvalidPTS; 3085 } 3086 3087 // mix buffers... 3088 mAudioMixer->process(pts); 3089 mCurrentWriteLength = mSinkBufferSize; 3090 // increase sleep time progressively when application underrun condition clears. 3091 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3092 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3093 // such that we would underrun the audio HAL. 3094 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3095 sleepTimeShift--; 3096 } 3097 sleepTime = 0; 3098 standbyTime = systemTime() + standbyDelay; 3099 //TODO: delay standby when effects have a tail 3100} 3101 3102void AudioFlinger::MixerThread::threadLoop_sleepTime() 3103{ 3104 // If no tracks are ready, sleep once for the duration of an output 3105 // buffer size, then write 0s to the output 3106 if (sleepTime == 0) { 3107 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3108 sleepTime = activeSleepTime >> sleepTimeShift; 3109 if (sleepTime < kMinThreadSleepTimeUs) { 3110 sleepTime = kMinThreadSleepTimeUs; 3111 } 3112 // reduce sleep time in case of consecutive application underruns to avoid 3113 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3114 // duration we would end up writing less data than needed by the audio HAL if 3115 // the condition persists. 3116 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3117 sleepTimeShift++; 3118 } 3119 } else { 3120 sleepTime = idleSleepTime; 3121 } 3122 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3123 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3124 // before effects processing or output. 3125 if (mMixerBufferValid) { 3126 memset(mMixerBuffer, 0, mMixerBufferSize); 3127 } else { 3128 memset(mSinkBuffer, 0, mSinkBufferSize); 3129 } 3130 sleepTime = 0; 3131 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3132 "anticipated start"); 3133 } 3134 // TODO add standby time extension fct of effect tail 3135} 3136 3137// prepareTracks_l() must be called with ThreadBase::mLock held 3138AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3139 Vector< sp<Track> > *tracksToRemove) 3140{ 3141 3142 mixer_state mixerStatus = MIXER_IDLE; 3143 // find out which tracks need to be processed 3144 size_t count = mActiveTracks.size(); 3145 size_t mixedTracks = 0; 3146 size_t tracksWithEffect = 0; 3147 // counts only _active_ fast tracks 3148 size_t fastTracks = 0; 3149 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3150 3151 float masterVolume = mMasterVolume; 3152 bool masterMute = mMasterMute; 3153 3154 if (masterMute) { 3155 masterVolume = 0; 3156 } 3157 // Delegate master volume control to effect in output mix effect chain if needed 3158 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3159 if (chain != 0) { 3160 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3161 chain->setVolume_l(&v, &v); 3162 masterVolume = (float)((v + (1 << 23)) >> 24); 3163 chain.clear(); 3164 } 3165 3166 // prepare a new state to push 3167 FastMixerStateQueue *sq = NULL; 3168 FastMixerState *state = NULL; 3169 bool didModify = false; 3170 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3171 if (mFastMixer != 0) { 3172 sq = mFastMixer->sq(); 3173 state = sq->begin(); 3174 } 3175 3176 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3177 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3178 3179 for (size_t i=0 ; i<count ; i++) { 3180 const sp<Track> t = mActiveTracks[i].promote(); 3181 if (t == 0) { 3182 continue; 3183 } 3184 3185 // this const just means the local variable doesn't change 3186 Track* const track = t.get(); 3187 3188 // process fast tracks 3189 if (track->isFastTrack()) { 3190 3191 // It's theoretically possible (though unlikely) for a fast track to be created 3192 // and then removed within the same normal mix cycle. This is not a problem, as 3193 // the track never becomes active so it's fast mixer slot is never touched. 3194 // The converse, of removing an (active) track and then creating a new track 3195 // at the identical fast mixer slot within the same normal mix cycle, 3196 // is impossible because the slot isn't marked available until the end of each cycle. 3197 int j = track->mFastIndex; 3198 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3199 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3200 FastTrack *fastTrack = &state->mFastTracks[j]; 3201 3202 // Determine whether the track is currently in underrun condition, 3203 // and whether it had a recent underrun. 3204 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3205 FastTrackUnderruns underruns = ftDump->mUnderruns; 3206 uint32_t recentFull = (underruns.mBitFields.mFull - 3207 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3208 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3209 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3210 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3211 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3212 uint32_t recentUnderruns = recentPartial + recentEmpty; 3213 track->mObservedUnderruns = underruns; 3214 // don't count underruns that occur while stopping or pausing 3215 // or stopped which can occur when flush() is called while active 3216 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3217 recentUnderruns > 0) { 3218 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3219 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3220 } 3221 3222 // This is similar to the state machine for normal tracks, 3223 // with a few modifications for fast tracks. 3224 bool isActive = true; 3225 switch (track->mState) { 3226 case TrackBase::STOPPING_1: 3227 // track stays active in STOPPING_1 state until first underrun 3228 if (recentUnderruns > 0 || track->isTerminated()) { 3229 track->mState = TrackBase::STOPPING_2; 3230 } 3231 break; 3232 case TrackBase::PAUSING: 3233 // ramp down is not yet implemented 3234 track->setPaused(); 3235 break; 3236 case TrackBase::RESUMING: 3237 // ramp up is not yet implemented 3238 track->mState = TrackBase::ACTIVE; 3239 break; 3240 case TrackBase::ACTIVE: 3241 if (recentFull > 0 || recentPartial > 0) { 3242 // track has provided at least some frames recently: reset retry count 3243 track->mRetryCount = kMaxTrackRetries; 3244 } 3245 if (recentUnderruns == 0) { 3246 // no recent underruns: stay active 3247 break; 3248 } 3249 // there has recently been an underrun of some kind 3250 if (track->sharedBuffer() == 0) { 3251 // were any of the recent underruns "empty" (no frames available)? 3252 if (recentEmpty == 0) { 3253 // no, then ignore the partial underruns as they are allowed indefinitely 3254 break; 3255 } 3256 // there has recently been an "empty" underrun: decrement the retry counter 3257 if (--(track->mRetryCount) > 0) { 3258 break; 3259 } 3260 // indicate to client process that the track was disabled because of underrun; 3261 // it will then automatically call start() when data is available 3262 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3263 // remove from active list, but state remains ACTIVE [confusing but true] 3264 isActive = false; 3265 break; 3266 } 3267 // fall through 3268 case TrackBase::STOPPING_2: 3269 case TrackBase::PAUSED: 3270 case TrackBase::STOPPED: 3271 case TrackBase::FLUSHED: // flush() while active 3272 // Check for presentation complete if track is inactive 3273 // We have consumed all the buffers of this track. 3274 // This would be incomplete if we auto-paused on underrun 3275 { 3276 size_t audioHALFrames = 3277 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3278 size_t framesWritten = mBytesWritten / mFrameSize; 3279 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3280 // track stays in active list until presentation is complete 3281 break; 3282 } 3283 } 3284 if (track->isStopping_2()) { 3285 track->mState = TrackBase::STOPPED; 3286 } 3287 if (track->isStopped()) { 3288 // Can't reset directly, as fast mixer is still polling this track 3289 // track->reset(); 3290 // So instead mark this track as needing to be reset after push with ack 3291 resetMask |= 1 << i; 3292 } 3293 isActive = false; 3294 break; 3295 case TrackBase::IDLE: 3296 default: 3297 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3298 } 3299 3300 if (isActive) { 3301 // was it previously inactive? 3302 if (!(state->mTrackMask & (1 << j))) { 3303 ExtendedAudioBufferProvider *eabp = track; 3304 VolumeProvider *vp = track; 3305 fastTrack->mBufferProvider = eabp; 3306 fastTrack->mVolumeProvider = vp; 3307 fastTrack->mChannelMask = track->mChannelMask; 3308 fastTrack->mFormat = track->mFormat; 3309 fastTrack->mGeneration++; 3310 state->mTrackMask |= 1 << j; 3311 didModify = true; 3312 // no acknowledgement required for newly active tracks 3313 } 3314 // cache the combined master volume and stream type volume for fast mixer; this 3315 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3316 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3317 ++fastTracks; 3318 } else { 3319 // was it previously active? 3320 if (state->mTrackMask & (1 << j)) { 3321 fastTrack->mBufferProvider = NULL; 3322 fastTrack->mGeneration++; 3323 state->mTrackMask &= ~(1 << j); 3324 didModify = true; 3325 // If any fast tracks were removed, we must wait for acknowledgement 3326 // because we're about to decrement the last sp<> on those tracks. 3327 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3328 } else { 3329 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3330 } 3331 tracksToRemove->add(track); 3332 // Avoids a misleading display in dumpsys 3333 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3334 } 3335 continue; 3336 } 3337 3338 { // local variable scope to avoid goto warning 3339 3340 audio_track_cblk_t* cblk = track->cblk(); 3341 3342 // The first time a track is added we wait 3343 // for all its buffers to be filled before processing it 3344 int name = track->name(); 3345 // make sure that we have enough frames to mix one full buffer. 3346 // enforce this condition only once to enable draining the buffer in case the client 3347 // app does not call stop() and relies on underrun to stop: 3348 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3349 // during last round 3350 size_t desiredFrames; 3351 uint32_t sr = track->sampleRate(); 3352 if (sr == mSampleRate) { 3353 desiredFrames = mNormalFrameCount; 3354 } else { 3355 // +1 for rounding and +1 for additional sample needed for interpolation 3356 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3357 // add frames already consumed but not yet released by the resampler 3358 // because mAudioTrackServerProxy->framesReady() will include these frames 3359 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3360#if 0 3361 // the minimum track buffer size is normally twice the number of frames necessary 3362 // to fill one buffer and the resampler should not leave more than one buffer worth 3363 // of unreleased frames after each pass, but just in case... 3364 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3365#endif 3366 } 3367 uint32_t minFrames = 1; 3368 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3369 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3370 minFrames = desiredFrames; 3371 } 3372 3373 size_t framesReady = track->framesReady(); 3374 if ((framesReady >= minFrames) && track->isReady() && 3375 !track->isPaused() && !track->isTerminated()) 3376 { 3377 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3378 3379 mixedTracks++; 3380 3381 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3382 // there is an effect chain connected to the track 3383 chain.clear(); 3384 if (track->mainBuffer() != mSinkBuffer && 3385 track->mainBuffer() != mMixerBuffer) { 3386 if (mEffectBufferEnabled) { 3387 mEffectBufferValid = true; // Later can set directly. 3388 } 3389 chain = getEffectChain_l(track->sessionId()); 3390 // Delegate volume control to effect in track effect chain if needed 3391 if (chain != 0) { 3392 tracksWithEffect++; 3393 } else { 3394 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3395 "session %d", 3396 name, track->sessionId()); 3397 } 3398 } 3399 3400 3401 int param = AudioMixer::VOLUME; 3402 if (track->mFillingUpStatus == Track::FS_FILLED) { 3403 // no ramp for the first volume setting 3404 track->mFillingUpStatus = Track::FS_ACTIVE; 3405 if (track->mState == TrackBase::RESUMING) { 3406 track->mState = TrackBase::ACTIVE; 3407 param = AudioMixer::RAMP_VOLUME; 3408 } 3409 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3410 // FIXME should not make a decision based on mServer 3411 } else if (cblk->mServer != 0) { 3412 // If the track is stopped before the first frame was mixed, 3413 // do not apply ramp 3414 param = AudioMixer::RAMP_VOLUME; 3415 } 3416 3417 // compute volume for this track 3418 uint32_t vl, vr; // in U8.24 integer format 3419 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3420 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3421 vl = vr = 0; 3422 vlf = vrf = vaf = 0.; 3423 if (track->isPausing()) { 3424 track->setPaused(); 3425 } 3426 } else { 3427 3428 // read original volumes with volume control 3429 float typeVolume = mStreamTypes[track->streamType()].volume; 3430 float v = masterVolume * typeVolume; 3431 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3432 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3433 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3434 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3435 // track volumes come from shared memory, so can't be trusted and must be clamped 3436 if (vlf > GAIN_FLOAT_UNITY) { 3437 ALOGV("Track left volume out of range: %.3g", vlf); 3438 vlf = GAIN_FLOAT_UNITY; 3439 } 3440 if (vrf > GAIN_FLOAT_UNITY) { 3441 ALOGV("Track right volume out of range: %.3g", vrf); 3442 vrf = GAIN_FLOAT_UNITY; 3443 } 3444 // now apply the master volume and stream type volume 3445 vlf *= v; 3446 vrf *= v; 3447 // assuming master volume and stream type volume each go up to 1.0, 3448 // then derive vl and vr as U8.24 versions for the effect chain 3449 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3450 vl = (uint32_t) (scaleto8_24 * vlf); 3451 vr = (uint32_t) (scaleto8_24 * vrf); 3452 // vl and vr are now in U8.24 format 3453 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3454 // send level comes from shared memory and so may be corrupt 3455 if (sendLevel > MAX_GAIN_INT) { 3456 ALOGV("Track send level out of range: %04X", sendLevel); 3457 sendLevel = MAX_GAIN_INT; 3458 } 3459 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3460 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3461 } 3462 3463 // Delegate volume control to effect in track effect chain if needed 3464 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3465 // Do not ramp volume if volume is controlled by effect 3466 param = AudioMixer::VOLUME; 3467 // Update remaining floating point volume levels 3468 vlf = (float)vl / (1 << 24); 3469 vrf = (float)vr / (1 << 24); 3470 track->mHasVolumeController = true; 3471 } else { 3472 // force no volume ramp when volume controller was just disabled or removed 3473 // from effect chain to avoid volume spike 3474 if (track->mHasVolumeController) { 3475 param = AudioMixer::VOLUME; 3476 } 3477 track->mHasVolumeController = false; 3478 } 3479 3480 // XXX: these things DON'T need to be done each time 3481 mAudioMixer->setBufferProvider(name, track); 3482 mAudioMixer->enable(name); 3483 3484 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3485 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3486 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3487 mAudioMixer->setParameter( 3488 name, 3489 AudioMixer::TRACK, 3490 AudioMixer::FORMAT, (void *)track->format()); 3491 mAudioMixer->setParameter( 3492 name, 3493 AudioMixer::TRACK, 3494 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3495 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3496 uint32_t maxSampleRate = mSampleRate * 2; 3497 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3498 if (reqSampleRate == 0) { 3499 reqSampleRate = mSampleRate; 3500 } else if (reqSampleRate > maxSampleRate) { 3501 reqSampleRate = maxSampleRate; 3502 } 3503 mAudioMixer->setParameter( 3504 name, 3505 AudioMixer::RESAMPLE, 3506 AudioMixer::SAMPLE_RATE, 3507 (void *)(uintptr_t)reqSampleRate); 3508 /* 3509 * Select the appropriate output buffer for the track. 3510 * 3511 * Tracks with effects go into their own effects chain buffer 3512 * and from there into either mEffectBuffer or mSinkBuffer. 3513 * 3514 * Other tracks can use mMixerBuffer for higher precision 3515 * channel accumulation. If this buffer is enabled 3516 * (mMixerBufferEnabled true), then selected tracks will accumulate 3517 * into it. 3518 * 3519 */ 3520 if (mMixerBufferEnabled 3521 && (track->mainBuffer() == mSinkBuffer 3522 || track->mainBuffer() == mMixerBuffer)) { 3523 mAudioMixer->setParameter( 3524 name, 3525 AudioMixer::TRACK, 3526 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3527 mAudioMixer->setParameter( 3528 name, 3529 AudioMixer::TRACK, 3530 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3531 // TODO: override track->mainBuffer()? 3532 mMixerBufferValid = true; 3533 } else { 3534 mAudioMixer->setParameter( 3535 name, 3536 AudioMixer::TRACK, 3537 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3538 mAudioMixer->setParameter( 3539 name, 3540 AudioMixer::TRACK, 3541 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3542 } 3543 mAudioMixer->setParameter( 3544 name, 3545 AudioMixer::TRACK, 3546 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3547 3548 // reset retry count 3549 track->mRetryCount = kMaxTrackRetries; 3550 3551 // If one track is ready, set the mixer ready if: 3552 // - the mixer was not ready during previous round OR 3553 // - no other track is not ready 3554 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3555 mixerStatus != MIXER_TRACKS_ENABLED) { 3556 mixerStatus = MIXER_TRACKS_READY; 3557 } 3558 } else { 3559 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3560 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3561 } 3562 // clear effect chain input buffer if an active track underruns to avoid sending 3563 // previous audio buffer again to effects 3564 chain = getEffectChain_l(track->sessionId()); 3565 if (chain != 0) { 3566 chain->clearInputBuffer(); 3567 } 3568 3569 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3570 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3571 track->isStopped() || track->isPaused()) { 3572 // We have consumed all the buffers of this track. 3573 // Remove it from the list of active tracks. 3574 // TODO: use actual buffer filling status instead of latency when available from 3575 // audio HAL 3576 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3577 size_t framesWritten = mBytesWritten / mFrameSize; 3578 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3579 if (track->isStopped()) { 3580 track->reset(); 3581 } 3582 tracksToRemove->add(track); 3583 } 3584 } else { 3585 // No buffers for this track. Give it a few chances to 3586 // fill a buffer, then remove it from active list. 3587 if (--(track->mRetryCount) <= 0) { 3588 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3589 tracksToRemove->add(track); 3590 // indicate to client process that the track was disabled because of underrun; 3591 // it will then automatically call start() when data is available 3592 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3593 // If one track is not ready, mark the mixer also not ready if: 3594 // - the mixer was ready during previous round OR 3595 // - no other track is ready 3596 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3597 mixerStatus != MIXER_TRACKS_READY) { 3598 mixerStatus = MIXER_TRACKS_ENABLED; 3599 } 3600 } 3601 mAudioMixer->disable(name); 3602 } 3603 3604 } // local variable scope to avoid goto warning 3605track_is_ready: ; 3606 3607 } 3608 3609 // Push the new FastMixer state if necessary 3610 bool pauseAudioWatchdog = false; 3611 if (didModify) { 3612 state->mFastTracksGen++; 3613 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3614 if (kUseFastMixer == FastMixer_Dynamic && 3615 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3616 state->mCommand = FastMixerState::COLD_IDLE; 3617 state->mColdFutexAddr = &mFastMixerFutex; 3618 state->mColdGen++; 3619 mFastMixerFutex = 0; 3620 if (kUseFastMixer == FastMixer_Dynamic) { 3621 mNormalSink = mOutputSink; 3622 } 3623 // If we go into cold idle, need to wait for acknowledgement 3624 // so that fast mixer stops doing I/O. 3625 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3626 pauseAudioWatchdog = true; 3627 } 3628 } 3629 if (sq != NULL) { 3630 sq->end(didModify); 3631 sq->push(block); 3632 } 3633#ifdef AUDIO_WATCHDOG 3634 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3635 mAudioWatchdog->pause(); 3636 } 3637#endif 3638 3639 // Now perform the deferred reset on fast tracks that have stopped 3640 while (resetMask != 0) { 3641 size_t i = __builtin_ctz(resetMask); 3642 ALOG_ASSERT(i < count); 3643 resetMask &= ~(1 << i); 3644 sp<Track> t = mActiveTracks[i].promote(); 3645 if (t == 0) { 3646 continue; 3647 } 3648 Track* track = t.get(); 3649 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3650 track->reset(); 3651 } 3652 3653 // remove all the tracks that need to be... 3654 removeTracks_l(*tracksToRemove); 3655 3656 // sink or mix buffer must be cleared if all tracks are connected to an 3657 // effect chain as in this case the mixer will not write to the sink or mix buffer 3658 // and track effects will accumulate into it 3659 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3660 (mixedTracks == 0 && fastTracks > 0))) { 3661 // FIXME as a performance optimization, should remember previous zero status 3662 if (mMixerBufferValid) { 3663 memset(mMixerBuffer, 0, mMixerBufferSize); 3664 // TODO: In testing, mSinkBuffer below need not be cleared because 3665 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3666 // after mixing. 3667 // 3668 // To enforce this guarantee: 3669 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3670 // (mixedTracks == 0 && fastTracks > 0)) 3671 // must imply MIXER_TRACKS_READY. 3672 // Later, we may clear buffers regardless, and skip much of this logic. 3673 } 3674 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3675 if (mEffectBufferValid) { 3676 memset(mEffectBuffer, 0, mEffectBufferSize); 3677 } 3678 // FIXME as a performance optimization, should remember previous zero status 3679 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3680 } 3681 3682 // if any fast tracks, then status is ready 3683 mMixerStatusIgnoringFastTracks = mixerStatus; 3684 if (fastTracks > 0) { 3685 mixerStatus = MIXER_TRACKS_READY; 3686 } 3687 return mixerStatus; 3688} 3689 3690// getTrackName_l() must be called with ThreadBase::mLock held 3691int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3692 audio_format_t format, int sessionId) 3693{ 3694 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3695} 3696 3697// deleteTrackName_l() must be called with ThreadBase::mLock held 3698void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3699{ 3700 ALOGV("remove track (%d) and delete from mixer", name); 3701 mAudioMixer->deleteTrackName(name); 3702} 3703 3704// checkForNewParameter_l() must be called with ThreadBase::mLock held 3705bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3706 status_t& status) 3707{ 3708 bool reconfig = false; 3709 3710 status = NO_ERROR; 3711 3712 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3713 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3714 if (mFastMixer != 0) { 3715 FastMixerStateQueue *sq = mFastMixer->sq(); 3716 FastMixerState *state = sq->begin(); 3717 if (!(state->mCommand & FastMixerState::IDLE)) { 3718 previousCommand = state->mCommand; 3719 state->mCommand = FastMixerState::HOT_IDLE; 3720 sq->end(); 3721 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3722 } else { 3723 sq->end(false /*didModify*/); 3724 } 3725 } 3726 3727 AudioParameter param = AudioParameter(keyValuePair); 3728 int value; 3729 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3730 reconfig = true; 3731 } 3732 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3733 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3734 status = BAD_VALUE; 3735 } else { 3736 // no need to save value, since it's constant 3737 reconfig = true; 3738 } 3739 } 3740 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3741 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3742 status = BAD_VALUE; 3743 } else { 3744 // no need to save value, since it's constant 3745 reconfig = true; 3746 } 3747 } 3748 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3749 // do not accept frame count changes if tracks are open as the track buffer 3750 // size depends on frame count and correct behavior would not be guaranteed 3751 // if frame count is changed after track creation 3752 if (!mTracks.isEmpty()) { 3753 status = INVALID_OPERATION; 3754 } else { 3755 reconfig = true; 3756 } 3757 } 3758 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3759#ifdef ADD_BATTERY_DATA 3760 // when changing the audio output device, call addBatteryData to notify 3761 // the change 3762 if (mOutDevice != value) { 3763 uint32_t params = 0; 3764 // check whether speaker is on 3765 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3766 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3767 } 3768 3769 audio_devices_t deviceWithoutSpeaker 3770 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3771 // check if any other device (except speaker) is on 3772 if (value & deviceWithoutSpeaker ) { 3773 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3774 } 3775 3776 if (params != 0) { 3777 addBatteryData(params); 3778 } 3779 } 3780#endif 3781 3782 // forward device change to effects that have requested to be 3783 // aware of attached audio device. 3784 if (value != AUDIO_DEVICE_NONE) { 3785 mOutDevice = value; 3786 for (size_t i = 0; i < mEffectChains.size(); i++) { 3787 mEffectChains[i]->setDevice_l(mOutDevice); 3788 } 3789 } 3790 } 3791 3792 if (status == NO_ERROR) { 3793 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3794 keyValuePair.string()); 3795 if (!mStandby && status == INVALID_OPERATION) { 3796 mOutput->stream->common.standby(&mOutput->stream->common); 3797 mStandby = true; 3798 mBytesWritten = 0; 3799 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3800 keyValuePair.string()); 3801 } 3802 if (status == NO_ERROR && reconfig) { 3803 readOutputParameters_l(); 3804 delete mAudioMixer; 3805 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3806 for (size_t i = 0; i < mTracks.size() ; i++) { 3807 int name = getTrackName_l(mTracks[i]->mChannelMask, 3808 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3809 if (name < 0) { 3810 break; 3811 } 3812 mTracks[i]->mName = name; 3813 } 3814 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3815 } 3816 } 3817 3818 if (!(previousCommand & FastMixerState::IDLE)) { 3819 ALOG_ASSERT(mFastMixer != 0); 3820 FastMixerStateQueue *sq = mFastMixer->sq(); 3821 FastMixerState *state = sq->begin(); 3822 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3823 state->mCommand = previousCommand; 3824 sq->end(); 3825 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3826 } 3827 3828 return reconfig; 3829} 3830 3831 3832void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3833{ 3834 const size_t SIZE = 256; 3835 char buffer[SIZE]; 3836 String8 result; 3837 3838 PlaybackThread::dumpInternals(fd, args); 3839 3840 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3841 3842 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3843 const FastMixerDumpState copy(mFastMixerDumpState); 3844 copy.dump(fd); 3845 3846#ifdef STATE_QUEUE_DUMP 3847 // Similar for state queue 3848 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3849 observerCopy.dump(fd); 3850 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3851 mutatorCopy.dump(fd); 3852#endif 3853 3854#ifdef TEE_SINK 3855 // Write the tee output to a .wav file 3856 dumpTee(fd, mTeeSource, mId); 3857#endif 3858 3859#ifdef AUDIO_WATCHDOG 3860 if (mAudioWatchdog != 0) { 3861 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3862 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3863 wdCopy.dump(fd); 3864 } 3865#endif 3866} 3867 3868uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3869{ 3870 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3871} 3872 3873uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3874{ 3875 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3876} 3877 3878void AudioFlinger::MixerThread::cacheParameters_l() 3879{ 3880 PlaybackThread::cacheParameters_l(); 3881 3882 // FIXME: Relaxed timing because of a certain device that can't meet latency 3883 // Should be reduced to 2x after the vendor fixes the driver issue 3884 // increase threshold again due to low power audio mode. The way this warning 3885 // threshold is calculated and its usefulness should be reconsidered anyway. 3886 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3887} 3888 3889// ---------------------------------------------------------------------------- 3890 3891AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3892 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3893 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3894 // mLeftVolFloat, mRightVolFloat 3895{ 3896} 3897 3898AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3899 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3900 ThreadBase::type_t type) 3901 : PlaybackThread(audioFlinger, output, id, device, type) 3902 // mLeftVolFloat, mRightVolFloat 3903{ 3904} 3905 3906AudioFlinger::DirectOutputThread::~DirectOutputThread() 3907{ 3908} 3909 3910void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3911{ 3912 audio_track_cblk_t* cblk = track->cblk(); 3913 float left, right; 3914 3915 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3916 left = right = 0; 3917 } else { 3918 float typeVolume = mStreamTypes[track->streamType()].volume; 3919 float v = mMasterVolume * typeVolume; 3920 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3921 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3922 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3923 if (left > GAIN_FLOAT_UNITY) { 3924 left = GAIN_FLOAT_UNITY; 3925 } 3926 left *= v; 3927 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3928 if (right > GAIN_FLOAT_UNITY) { 3929 right = GAIN_FLOAT_UNITY; 3930 } 3931 right *= v; 3932 } 3933 3934 if (lastTrack) { 3935 if (left != mLeftVolFloat || right != mRightVolFloat) { 3936 mLeftVolFloat = left; 3937 mRightVolFloat = right; 3938 3939 // Convert volumes from float to 8.24 3940 uint32_t vl = (uint32_t)(left * (1 << 24)); 3941 uint32_t vr = (uint32_t)(right * (1 << 24)); 3942 3943 // Delegate volume control to effect in track effect chain if needed 3944 // only one effect chain can be present on DirectOutputThread, so if 3945 // there is one, the track is connected to it 3946 if (!mEffectChains.isEmpty()) { 3947 mEffectChains[0]->setVolume_l(&vl, &vr); 3948 left = (float)vl / (1 << 24); 3949 right = (float)vr / (1 << 24); 3950 } 3951 if (mOutput->stream->set_volume) { 3952 mOutput->stream->set_volume(mOutput->stream, left, right); 3953 } 3954 } 3955 } 3956} 3957 3958 3959AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3960 Vector< sp<Track> > *tracksToRemove 3961) 3962{ 3963 size_t count = mActiveTracks.size(); 3964 mixer_state mixerStatus = MIXER_IDLE; 3965 3966 // find out which tracks need to be processed 3967 for (size_t i = 0; i < count; i++) { 3968 sp<Track> t = mActiveTracks[i].promote(); 3969 // The track died recently 3970 if (t == 0) { 3971 continue; 3972 } 3973 3974 Track* const track = t.get(); 3975 audio_track_cblk_t* cblk = track->cblk(); 3976 // Only consider last track started for volume and mixer state control. 3977 // In theory an older track could underrun and restart after the new one starts 3978 // but as we only care about the transition phase between two tracks on a 3979 // direct output, it is not a problem to ignore the underrun case. 3980 sp<Track> l = mLatestActiveTrack.promote(); 3981 bool last = l.get() == track; 3982 3983 // The first time a track is added we wait 3984 // for all its buffers to be filled before processing it 3985 uint32_t minFrames; 3986 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 3987 minFrames = mNormalFrameCount; 3988 } else { 3989 minFrames = 1; 3990 } 3991 3992 ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ", 3993 minFrames, track->mState, track->framesReady()); 3994 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 3995 !track->isStopping_2() && !track->isStopped()) 3996 { 3997 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3998 3999 if (track->mFillingUpStatus == Track::FS_FILLED) { 4000 track->mFillingUpStatus = Track::FS_ACTIVE; 4001 // make sure processVolume_l() will apply new volume even if 0 4002 mLeftVolFloat = mRightVolFloat = -1.0; 4003 if (track->mState == TrackBase::RESUMING) { 4004 track->mState = TrackBase::ACTIVE; 4005 } 4006 } 4007 4008 // compute volume for this track 4009 processVolume_l(track, last); 4010 if (last) { 4011 // reset retry count 4012 track->mRetryCount = kMaxTrackRetriesDirect; 4013 mActiveTrack = t; 4014 mixerStatus = MIXER_TRACKS_READY; 4015 } 4016 } else { 4017 // clear effect chain input buffer if the last active track started underruns 4018 // to avoid sending previous audio buffer again to effects 4019 if (!mEffectChains.isEmpty() && last) { 4020 mEffectChains[0]->clearInputBuffer(); 4021 } 4022 if (track->isStopping_1()) { 4023 track->mState = TrackBase::STOPPING_2; 4024 } 4025 if ((track->sharedBuffer() != 0) || track->isStopped() || 4026 track->isStopping_2() || track->isPaused()) { 4027 // We have consumed all the buffers of this track. 4028 // Remove it from the list of active tracks. 4029 size_t audioHALFrames; 4030 if (audio_is_linear_pcm(mFormat)) { 4031 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4032 } else { 4033 audioHALFrames = 0; 4034 } 4035 4036 size_t framesWritten = mBytesWritten / mFrameSize; 4037 if (mStandby || !last || 4038 track->presentationComplete(framesWritten, audioHALFrames)) { 4039 if (track->isStopping_2()) { 4040 track->mState = TrackBase::STOPPED; 4041 } 4042 if (track->isStopped()) { 4043 track->reset(); 4044 } 4045 tracksToRemove->add(track); 4046 } 4047 } else { 4048 // No buffers for this track. Give it a few chances to 4049 // fill a buffer, then remove it from active list. 4050 // Only consider last track started for mixer state control 4051 if (--(track->mRetryCount) <= 0) { 4052 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4053 tracksToRemove->add(track); 4054 // indicate to client process that the track was disabled because of underrun; 4055 // it will then automatically call start() when data is available 4056 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4057 } else if (last) { 4058 mixerStatus = MIXER_TRACKS_ENABLED; 4059 } 4060 } 4061 } 4062 } 4063 4064 // remove all the tracks that need to be... 4065 removeTracks_l(*tracksToRemove); 4066 4067 return mixerStatus; 4068} 4069 4070void AudioFlinger::DirectOutputThread::threadLoop_mix() 4071{ 4072 size_t frameCount = mFrameCount; 4073 int8_t *curBuf = (int8_t *)mSinkBuffer; 4074 // output audio to hardware 4075 while (frameCount) { 4076 AudioBufferProvider::Buffer buffer; 4077 buffer.frameCount = frameCount; 4078 mActiveTrack->getNextBuffer(&buffer); 4079 if (buffer.raw == NULL) { 4080 memset(curBuf, 0, frameCount * mFrameSize); 4081 break; 4082 } 4083 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4084 frameCount -= buffer.frameCount; 4085 curBuf += buffer.frameCount * mFrameSize; 4086 mActiveTrack->releaseBuffer(&buffer); 4087 } 4088 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4089 sleepTime = 0; 4090 standbyTime = systemTime() + standbyDelay; 4091 mActiveTrack.clear(); 4092} 4093 4094void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4095{ 4096 if (sleepTime == 0) { 4097 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4098 sleepTime = activeSleepTime; 4099 } else { 4100 sleepTime = idleSleepTime; 4101 } 4102 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4103 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4104 sleepTime = 0; 4105 } 4106} 4107 4108// getTrackName_l() must be called with ThreadBase::mLock held 4109int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4110 audio_format_t format __unused, int sessionId __unused) 4111{ 4112 return 0; 4113} 4114 4115// deleteTrackName_l() must be called with ThreadBase::mLock held 4116void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4117{ 4118} 4119 4120// checkForNewParameter_l() must be called with ThreadBase::mLock held 4121bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4122 status_t& status) 4123{ 4124 bool reconfig = false; 4125 4126 status = NO_ERROR; 4127 4128 AudioParameter param = AudioParameter(keyValuePair); 4129 int value; 4130 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4131 // forward device change to effects that have requested to be 4132 // aware of attached audio device. 4133 if (value != AUDIO_DEVICE_NONE) { 4134 mOutDevice = value; 4135 for (size_t i = 0; i < mEffectChains.size(); i++) { 4136 mEffectChains[i]->setDevice_l(mOutDevice); 4137 } 4138 } 4139 } 4140 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4141 // do not accept frame count changes if tracks are open as the track buffer 4142 // size depends on frame count and correct behavior would not be garantied 4143 // if frame count is changed after track creation 4144 if (!mTracks.isEmpty()) { 4145 status = INVALID_OPERATION; 4146 } else { 4147 reconfig = true; 4148 } 4149 } 4150 if (status == NO_ERROR) { 4151 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4152 keyValuePair.string()); 4153 if (!mStandby && status == INVALID_OPERATION) { 4154 mOutput->stream->common.standby(&mOutput->stream->common); 4155 mStandby = true; 4156 mBytesWritten = 0; 4157 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4158 keyValuePair.string()); 4159 } 4160 if (status == NO_ERROR && reconfig) { 4161 readOutputParameters_l(); 4162 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4163 } 4164 } 4165 4166 return reconfig; 4167} 4168 4169uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4170{ 4171 uint32_t time; 4172 if (audio_is_linear_pcm(mFormat)) { 4173 time = PlaybackThread::activeSleepTimeUs(); 4174 } else { 4175 time = 10000; 4176 } 4177 return time; 4178} 4179 4180uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4181{ 4182 uint32_t time; 4183 if (audio_is_linear_pcm(mFormat)) { 4184 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4185 } else { 4186 time = 10000; 4187 } 4188 return time; 4189} 4190 4191uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4192{ 4193 uint32_t time; 4194 if (audio_is_linear_pcm(mFormat)) { 4195 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4196 } else { 4197 time = 10000; 4198 } 4199 return time; 4200} 4201 4202void AudioFlinger::DirectOutputThread::cacheParameters_l() 4203{ 4204 PlaybackThread::cacheParameters_l(); 4205 4206 // use shorter standby delay as on normal output to release 4207 // hardware resources as soon as possible 4208 if (audio_is_linear_pcm(mFormat)) { 4209 standbyDelay = microseconds(activeSleepTime*2); 4210 } else { 4211 standbyDelay = kOffloadStandbyDelayNs; 4212 } 4213} 4214 4215// ---------------------------------------------------------------------------- 4216 4217AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4218 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4219 : Thread(false /*canCallJava*/), 4220 mPlaybackThread(playbackThread), 4221 mWriteAckSequence(0), 4222 mDrainSequence(0) 4223{ 4224} 4225 4226AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4227{ 4228} 4229 4230void AudioFlinger::AsyncCallbackThread::onFirstRef() 4231{ 4232 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4233} 4234 4235bool AudioFlinger::AsyncCallbackThread::threadLoop() 4236{ 4237 while (!exitPending()) { 4238 uint32_t writeAckSequence; 4239 uint32_t drainSequence; 4240 4241 { 4242 Mutex::Autolock _l(mLock); 4243 while (!((mWriteAckSequence & 1) || 4244 (mDrainSequence & 1) || 4245 exitPending())) { 4246 mWaitWorkCV.wait(mLock); 4247 } 4248 4249 if (exitPending()) { 4250 break; 4251 } 4252 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4253 mWriteAckSequence, mDrainSequence); 4254 writeAckSequence = mWriteAckSequence; 4255 mWriteAckSequence &= ~1; 4256 drainSequence = mDrainSequence; 4257 mDrainSequence &= ~1; 4258 } 4259 { 4260 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4261 if (playbackThread != 0) { 4262 if (writeAckSequence & 1) { 4263 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4264 } 4265 if (drainSequence & 1) { 4266 playbackThread->resetDraining(drainSequence >> 1); 4267 } 4268 } 4269 } 4270 } 4271 return false; 4272} 4273 4274void AudioFlinger::AsyncCallbackThread::exit() 4275{ 4276 ALOGV("AsyncCallbackThread::exit"); 4277 Mutex::Autolock _l(mLock); 4278 requestExit(); 4279 mWaitWorkCV.broadcast(); 4280} 4281 4282void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4283{ 4284 Mutex::Autolock _l(mLock); 4285 // bit 0 is cleared 4286 mWriteAckSequence = sequence << 1; 4287} 4288 4289void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4290{ 4291 Mutex::Autolock _l(mLock); 4292 // ignore unexpected callbacks 4293 if (mWriteAckSequence & 2) { 4294 mWriteAckSequence |= 1; 4295 mWaitWorkCV.signal(); 4296 } 4297} 4298 4299void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4300{ 4301 Mutex::Autolock _l(mLock); 4302 // bit 0 is cleared 4303 mDrainSequence = sequence << 1; 4304} 4305 4306void AudioFlinger::AsyncCallbackThread::resetDraining() 4307{ 4308 Mutex::Autolock _l(mLock); 4309 // ignore unexpected callbacks 4310 if (mDrainSequence & 2) { 4311 mDrainSequence |= 1; 4312 mWaitWorkCV.signal(); 4313 } 4314} 4315 4316 4317// ---------------------------------------------------------------------------- 4318AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4319 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4320 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4321 mHwPaused(false), 4322 mFlushPending(false), 4323 mPausedBytesRemaining(0) 4324{ 4325 //FIXME: mStandby should be set to true by ThreadBase constructor 4326 mStandby = true; 4327} 4328 4329void AudioFlinger::OffloadThread::threadLoop_exit() 4330{ 4331 if (mFlushPending || mHwPaused) { 4332 // If a flush is pending or track was paused, just discard buffered data 4333 flushHw_l(); 4334 } else { 4335 mMixerStatus = MIXER_DRAIN_ALL; 4336 threadLoop_drain(); 4337 } 4338 if (mUseAsyncWrite) { 4339 ALOG_ASSERT(mCallbackThread != 0); 4340 mCallbackThread->exit(); 4341 } 4342 PlaybackThread::threadLoop_exit(); 4343} 4344 4345AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4346 Vector< sp<Track> > *tracksToRemove 4347) 4348{ 4349 size_t count = mActiveTracks.size(); 4350 4351 mixer_state mixerStatus = MIXER_IDLE; 4352 bool doHwPause = false; 4353 bool doHwResume = false; 4354 4355 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4356 4357 // find out which tracks need to be processed 4358 for (size_t i = 0; i < count; i++) { 4359 sp<Track> t = mActiveTracks[i].promote(); 4360 // The track died recently 4361 if (t == 0) { 4362 continue; 4363 } 4364 Track* const track = t.get(); 4365 audio_track_cblk_t* cblk = track->cblk(); 4366 // Only consider last track started for volume and mixer state control. 4367 // In theory an older track could underrun and restart after the new one starts 4368 // but as we only care about the transition phase between two tracks on a 4369 // direct output, it is not a problem to ignore the underrun case. 4370 sp<Track> l = mLatestActiveTrack.promote(); 4371 bool last = l.get() == track; 4372 4373 if (track->isInvalid()) { 4374 ALOGW("An invalidated track shouldn't be in active list"); 4375 tracksToRemove->add(track); 4376 continue; 4377 } 4378 4379 if (track->mState == TrackBase::IDLE) { 4380 ALOGW("An idle track shouldn't be in active list"); 4381 continue; 4382 } 4383 4384 if (track->isPausing()) { 4385 track->setPaused(); 4386 if (last) { 4387 if (!mHwPaused) { 4388 doHwPause = true; 4389 mHwPaused = true; 4390 } 4391 // If we were part way through writing the mixbuffer to 4392 // the HAL we must save this until we resume 4393 // BUG - this will be wrong if a different track is made active, 4394 // in that case we want to discard the pending data in the 4395 // mixbuffer and tell the client to present it again when the 4396 // track is resumed 4397 mPausedWriteLength = mCurrentWriteLength; 4398 mPausedBytesRemaining = mBytesRemaining; 4399 mBytesRemaining = 0; // stop writing 4400 } 4401 tracksToRemove->add(track); 4402 } else if (track->isFlushPending()) { 4403 track->flushAck(); 4404 if (last) { 4405 mFlushPending = true; 4406 } 4407 } else if (track->isResumePending()){ 4408 track->resumeAck(); 4409 if (last) { 4410 if (mPausedBytesRemaining) { 4411 // Need to continue write that was interrupted 4412 mCurrentWriteLength = mPausedWriteLength; 4413 mBytesRemaining = mPausedBytesRemaining; 4414 mPausedBytesRemaining = 0; 4415 } 4416 if (mHwPaused) { 4417 doHwResume = true; 4418 mHwPaused = false; 4419 // threadLoop_mix() will handle the case that we need to 4420 // resume an interrupted write 4421 } 4422 // enable write to audio HAL 4423 sleepTime = 0; 4424 4425 // Do not handle new data in this iteration even if track->framesReady() 4426 mixerStatus = MIXER_TRACKS_ENABLED; 4427 } 4428 } else if (track->framesReady() && track->isReady() && 4429 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4430 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4431 if (track->mFillingUpStatus == Track::FS_FILLED) { 4432 track->mFillingUpStatus = Track::FS_ACTIVE; 4433 // make sure processVolume_l() will apply new volume even if 0 4434 mLeftVolFloat = mRightVolFloat = -1.0; 4435 } 4436 4437 if (last) { 4438 sp<Track> previousTrack = mPreviousTrack.promote(); 4439 if (previousTrack != 0) { 4440 if (track != previousTrack.get()) { 4441 // Flush any data still being written from last track 4442 mBytesRemaining = 0; 4443 if (mPausedBytesRemaining) { 4444 // Last track was paused so we also need to flush saved 4445 // mixbuffer state and invalidate track so that it will 4446 // re-submit that unwritten data when it is next resumed 4447 mPausedBytesRemaining = 0; 4448 // Invalidate is a bit drastic - would be more efficient 4449 // to have a flag to tell client that some of the 4450 // previously written data was lost 4451 previousTrack->invalidate(); 4452 } 4453 // flush data already sent to the DSP if changing audio session as audio 4454 // comes from a different source. Also invalidate previous track to force a 4455 // seek when resuming. 4456 if (previousTrack->sessionId() != track->sessionId()) { 4457 previousTrack->invalidate(); 4458 } 4459 } 4460 } 4461 mPreviousTrack = track; 4462 // reset retry count 4463 track->mRetryCount = kMaxTrackRetriesOffload; 4464 mActiveTrack = t; 4465 mixerStatus = MIXER_TRACKS_READY; 4466 } 4467 } else { 4468 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4469 if (track->isStopping_1()) { 4470 // Hardware buffer can hold a large amount of audio so we must 4471 // wait for all current track's data to drain before we say 4472 // that the track is stopped. 4473 if (mBytesRemaining == 0) { 4474 // Only start draining when all data in mixbuffer 4475 // has been written 4476 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4477 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4478 // do not drain if no data was ever sent to HAL (mStandby == true) 4479 if (last && !mStandby) { 4480 // do not modify drain sequence if we are already draining. This happens 4481 // when resuming from pause after drain. 4482 if ((mDrainSequence & 1) == 0) { 4483 sleepTime = 0; 4484 standbyTime = systemTime() + standbyDelay; 4485 mixerStatus = MIXER_DRAIN_TRACK; 4486 mDrainSequence += 2; 4487 } 4488 if (mHwPaused) { 4489 // It is possible to move from PAUSED to STOPPING_1 without 4490 // a resume so we must ensure hardware is running 4491 doHwResume = true; 4492 mHwPaused = false; 4493 } 4494 } 4495 } 4496 } else if (track->isStopping_2()) { 4497 // Drain has completed or we are in standby, signal presentation complete 4498 if (!(mDrainSequence & 1) || !last || mStandby) { 4499 track->mState = TrackBase::STOPPED; 4500 size_t audioHALFrames = 4501 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4502 size_t framesWritten = 4503 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4504 track->presentationComplete(framesWritten, audioHALFrames); 4505 track->reset(); 4506 tracksToRemove->add(track); 4507 } 4508 } else { 4509 // No buffers for this track. Give it a few chances to 4510 // fill a buffer, then remove it from active list. 4511 if (--(track->mRetryCount) <= 0) { 4512 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4513 track->name()); 4514 tracksToRemove->add(track); 4515 // indicate to client process that the track was disabled because of underrun; 4516 // it will then automatically call start() when data is available 4517 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4518 } else if (last){ 4519 mixerStatus = MIXER_TRACKS_ENABLED; 4520 } 4521 } 4522 } 4523 // compute volume for this track 4524 processVolume_l(track, last); 4525 } 4526 4527 // make sure the pause/flush/resume sequence is executed in the right order. 4528 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4529 // before flush and then resume HW. This can happen in case of pause/flush/resume 4530 // if resume is received before pause is executed. 4531 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4532 mOutput->stream->pause(mOutput->stream); 4533 } 4534 if (mFlushPending) { 4535 flushHw_l(); 4536 mFlushPending = false; 4537 } 4538 if (!mStandby && doHwResume) { 4539 mOutput->stream->resume(mOutput->stream); 4540 } 4541 4542 // remove all the tracks that need to be... 4543 removeTracks_l(*tracksToRemove); 4544 4545 return mixerStatus; 4546} 4547 4548// must be called with thread mutex locked 4549bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4550{ 4551 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4552 mWriteAckSequence, mDrainSequence); 4553 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4554 return true; 4555 } 4556 return false; 4557} 4558 4559// must be called with thread mutex locked 4560bool AudioFlinger::OffloadThread::shouldStandby_l() 4561{ 4562 bool trackPaused = false; 4563 4564 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4565 // after a timeout and we will enter standby then. 4566 if (mTracks.size() > 0) { 4567 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4568 } 4569 4570 return !mStandby && !trackPaused; 4571} 4572 4573 4574bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4575{ 4576 Mutex::Autolock _l(mLock); 4577 return waitingAsyncCallback_l(); 4578} 4579 4580void AudioFlinger::OffloadThread::flushHw_l() 4581{ 4582 mOutput->stream->flush(mOutput->stream); 4583 // Flush anything still waiting in the mixbuffer 4584 mCurrentWriteLength = 0; 4585 mBytesRemaining = 0; 4586 mPausedWriteLength = 0; 4587 mPausedBytesRemaining = 0; 4588 mHwPaused = false; 4589 4590 if (mUseAsyncWrite) { 4591 // discard any pending drain or write ack by incrementing sequence 4592 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4593 mDrainSequence = (mDrainSequence + 2) & ~1; 4594 ALOG_ASSERT(mCallbackThread != 0); 4595 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4596 mCallbackThread->setDraining(mDrainSequence); 4597 } 4598} 4599 4600void AudioFlinger::OffloadThread::onAddNewTrack_l() 4601{ 4602 sp<Track> previousTrack = mPreviousTrack.promote(); 4603 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4604 4605 if (previousTrack != 0 && latestTrack != 0 && 4606 (previousTrack->sessionId() != latestTrack->sessionId())) { 4607 mFlushPending = true; 4608 } 4609 PlaybackThread::onAddNewTrack_l(); 4610} 4611 4612// ---------------------------------------------------------------------------- 4613 4614AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4615 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4616 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4617 DUPLICATING), 4618 mWaitTimeMs(UINT_MAX) 4619{ 4620 addOutputTrack(mainThread); 4621} 4622 4623AudioFlinger::DuplicatingThread::~DuplicatingThread() 4624{ 4625 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4626 mOutputTracks[i]->destroy(); 4627 } 4628} 4629 4630void AudioFlinger::DuplicatingThread::threadLoop_mix() 4631{ 4632 // mix buffers... 4633 if (outputsReady(outputTracks)) { 4634 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4635 } else { 4636 memset(mSinkBuffer, 0, mSinkBufferSize); 4637 } 4638 sleepTime = 0; 4639 writeFrames = mNormalFrameCount; 4640 mCurrentWriteLength = mSinkBufferSize; 4641 standbyTime = systemTime() + standbyDelay; 4642} 4643 4644void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4645{ 4646 if (sleepTime == 0) { 4647 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4648 sleepTime = activeSleepTime; 4649 } else { 4650 sleepTime = idleSleepTime; 4651 } 4652 } else if (mBytesWritten != 0) { 4653 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4654 writeFrames = mNormalFrameCount; 4655 memset(mSinkBuffer, 0, mSinkBufferSize); 4656 } else { 4657 // flush remaining overflow buffers in output tracks 4658 writeFrames = 0; 4659 } 4660 sleepTime = 0; 4661 } 4662} 4663 4664ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4665{ 4666 for (size_t i = 0; i < outputTracks.size(); i++) { 4667 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4668 // for delivery downstream as needed. This in-place conversion is safe as 4669 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4670 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4671 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4672 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4673 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4674 } 4675 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4676 } 4677 mStandby = false; 4678 return (ssize_t)mSinkBufferSize; 4679} 4680 4681void AudioFlinger::DuplicatingThread::threadLoop_standby() 4682{ 4683 // DuplicatingThread implements standby by stopping all tracks 4684 for (size_t i = 0; i < outputTracks.size(); i++) { 4685 outputTracks[i]->stop(); 4686 } 4687} 4688 4689void AudioFlinger::DuplicatingThread::saveOutputTracks() 4690{ 4691 outputTracks = mOutputTracks; 4692} 4693 4694void AudioFlinger::DuplicatingThread::clearOutputTracks() 4695{ 4696 outputTracks.clear(); 4697} 4698 4699void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4700{ 4701 Mutex::Autolock _l(mLock); 4702 // FIXME explain this formula 4703 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4704 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4705 // due to current usage case and restrictions on the AudioBufferProvider. 4706 // Actual buffer conversion is done in threadLoop_write(). 4707 // 4708 // TODO: This may change in the future, depending on multichannel 4709 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4710 OutputTrack *outputTrack = new OutputTrack(thread, 4711 this, 4712 mSampleRate, 4713 AUDIO_FORMAT_PCM_16_BIT, 4714 mChannelMask, 4715 frameCount, 4716 IPCThreadState::self()->getCallingUid()); 4717 if (outputTrack->cblk() != NULL) { 4718 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4719 mOutputTracks.add(outputTrack); 4720 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4721 updateWaitTime_l(); 4722 } 4723} 4724 4725void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4726{ 4727 Mutex::Autolock _l(mLock); 4728 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4729 if (mOutputTracks[i]->thread() == thread) { 4730 mOutputTracks[i]->destroy(); 4731 mOutputTracks.removeAt(i); 4732 updateWaitTime_l(); 4733 return; 4734 } 4735 } 4736 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4737} 4738 4739// caller must hold mLock 4740void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4741{ 4742 mWaitTimeMs = UINT_MAX; 4743 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4744 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4745 if (strong != 0) { 4746 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4747 if (waitTimeMs < mWaitTimeMs) { 4748 mWaitTimeMs = waitTimeMs; 4749 } 4750 } 4751 } 4752} 4753 4754 4755bool AudioFlinger::DuplicatingThread::outputsReady( 4756 const SortedVector< sp<OutputTrack> > &outputTracks) 4757{ 4758 for (size_t i = 0; i < outputTracks.size(); i++) { 4759 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4760 if (thread == 0) { 4761 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4762 outputTracks[i].get()); 4763 return false; 4764 } 4765 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4766 // see note at standby() declaration 4767 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4768 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4769 thread.get()); 4770 return false; 4771 } 4772 } 4773 return true; 4774} 4775 4776uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4777{ 4778 return (mWaitTimeMs * 1000) / 2; 4779} 4780 4781void AudioFlinger::DuplicatingThread::cacheParameters_l() 4782{ 4783 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4784 updateWaitTime_l(); 4785 4786 MixerThread::cacheParameters_l(); 4787} 4788 4789// ---------------------------------------------------------------------------- 4790// Record 4791// ---------------------------------------------------------------------------- 4792 4793AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4794 AudioStreamIn *input, 4795 audio_io_handle_t id, 4796 audio_devices_t outDevice, 4797 audio_devices_t inDevice 4798#ifdef TEE_SINK 4799 , const sp<NBAIO_Sink>& teeSink 4800#endif 4801 ) : 4802 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4803 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4804 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4805 mRsmpInRear(0) 4806#ifdef TEE_SINK 4807 , mTeeSink(teeSink) 4808#endif 4809 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4810 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4811 // mFastCapture below 4812 , mFastCaptureFutex(0) 4813 // mInputSource 4814 // mPipeSink 4815 // mPipeSource 4816 , mPipeFramesP2(0) 4817 // mPipeMemory 4818 // mFastCaptureNBLogWriter 4819 , mFastTrackAvail(false) 4820{ 4821 snprintf(mName, kNameLength, "AudioIn_%X", id); 4822 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4823 4824 readInputParameters_l(); 4825 4826 // create an NBAIO source for the HAL input stream, and negotiate 4827 mInputSource = new AudioStreamInSource(input->stream); 4828 size_t numCounterOffers = 0; 4829 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4830 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4831 ALOG_ASSERT(index == 0); 4832 4833 // initialize fast capture depending on configuration 4834 bool initFastCapture; 4835 switch (kUseFastCapture) { 4836 case FastCapture_Never: 4837 initFastCapture = false; 4838 break; 4839 case FastCapture_Always: 4840 initFastCapture = true; 4841 break; 4842 case FastCapture_Static: 4843 uint32_t primaryOutputSampleRate; 4844 { 4845 AutoMutex _l(audioFlinger->mHardwareLock); 4846 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4847 } 4848 initFastCapture = 4849 // either capture sample rate is same as (a reasonable) primary output sample rate 4850 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4851 (mSampleRate == primaryOutputSampleRate)) || 4852 // or primary output sample rate is unknown, and capture sample rate is reasonable 4853 ((primaryOutputSampleRate == 0) && 4854 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4855 // and the buffer size is < 10 ms 4856 (mFrameCount * 1000) / mSampleRate < 10; 4857 break; 4858 // case FastCapture_Dynamic: 4859 } 4860 4861 if (initFastCapture) { 4862 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4863 NBAIO_Format format = mInputSource->format(); 4864 size_t pipeFramesP2 = roundup(mFrameCount * 8); 4865 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4866 void *pipeBuffer; 4867 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4868 sp<IMemory> pipeMemory; 4869 if ((roHeap == 0) || 4870 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4871 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4872 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4873 goto failed; 4874 } 4875 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4876 memset(pipeBuffer, 0, pipeSize); 4877 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4878 const NBAIO_Format offers[1] = {format}; 4879 size_t numCounterOffers = 0; 4880 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4881 ALOG_ASSERT(index == 0); 4882 mPipeSink = pipe; 4883 PipeReader *pipeReader = new PipeReader(*pipe); 4884 numCounterOffers = 0; 4885 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4886 ALOG_ASSERT(index == 0); 4887 mPipeSource = pipeReader; 4888 mPipeFramesP2 = pipeFramesP2; 4889 mPipeMemory = pipeMemory; 4890 4891 // create fast capture 4892 mFastCapture = new FastCapture(); 4893 FastCaptureStateQueue *sq = mFastCapture->sq(); 4894#ifdef STATE_QUEUE_DUMP 4895 // FIXME 4896#endif 4897 FastCaptureState *state = sq->begin(); 4898 state->mCblk = NULL; 4899 state->mInputSource = mInputSource.get(); 4900 state->mInputSourceGen++; 4901 state->mPipeSink = pipe; 4902 state->mPipeSinkGen++; 4903 state->mFrameCount = mFrameCount; 4904 state->mCommand = FastCaptureState::COLD_IDLE; 4905 // already done in constructor initialization list 4906 //mFastCaptureFutex = 0; 4907 state->mColdFutexAddr = &mFastCaptureFutex; 4908 state->mColdGen++; 4909 state->mDumpState = &mFastCaptureDumpState; 4910#ifdef TEE_SINK 4911 // FIXME 4912#endif 4913 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4914 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4915 sq->end(); 4916 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4917 4918 // start the fast capture 4919 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4920 pid_t tid = mFastCapture->getTid(); 4921 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4922 if (err != 0) { 4923 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4924 kPriorityFastCapture, getpid_cached, tid, err); 4925 } 4926 4927#ifdef AUDIO_WATCHDOG 4928 // FIXME 4929#endif 4930 4931 mFastTrackAvail = true; 4932 } 4933failed: ; 4934 4935 // FIXME mNormalSource 4936} 4937 4938 4939AudioFlinger::RecordThread::~RecordThread() 4940{ 4941 if (mFastCapture != 0) { 4942 FastCaptureStateQueue *sq = mFastCapture->sq(); 4943 FastCaptureState *state = sq->begin(); 4944 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4945 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4946 if (old == -1) { 4947 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4948 } 4949 } 4950 state->mCommand = FastCaptureState::EXIT; 4951 sq->end(); 4952 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4953 mFastCapture->join(); 4954 mFastCapture.clear(); 4955 } 4956 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4957 mAudioFlinger->unregisterWriter(mNBLogWriter); 4958 delete[] mRsmpInBuffer; 4959} 4960 4961void AudioFlinger::RecordThread::onFirstRef() 4962{ 4963 run(mName, PRIORITY_URGENT_AUDIO); 4964} 4965 4966bool AudioFlinger::RecordThread::threadLoop() 4967{ 4968 nsecs_t lastWarning = 0; 4969 4970 inputStandBy(); 4971 4972reacquire_wakelock: 4973 sp<RecordTrack> activeTrack; 4974 int activeTracksGen; 4975 { 4976 Mutex::Autolock _l(mLock); 4977 size_t size = mActiveTracks.size(); 4978 activeTracksGen = mActiveTracksGen; 4979 if (size > 0) { 4980 // FIXME an arbitrary choice 4981 activeTrack = mActiveTracks[0]; 4982 acquireWakeLock_l(activeTrack->uid()); 4983 if (size > 1) { 4984 SortedVector<int> tmp; 4985 for (size_t i = 0; i < size; i++) { 4986 tmp.add(mActiveTracks[i]->uid()); 4987 } 4988 updateWakeLockUids_l(tmp); 4989 } 4990 } else { 4991 acquireWakeLock_l(-1); 4992 } 4993 } 4994 4995 // used to request a deferred sleep, to be executed later while mutex is unlocked 4996 uint32_t sleepUs = 0; 4997 4998 // loop while there is work to do 4999 for (;;) { 5000 Vector< sp<EffectChain> > effectChains; 5001 5002 // sleep with mutex unlocked 5003 if (sleepUs > 0) { 5004 usleep(sleepUs); 5005 sleepUs = 0; 5006 } 5007 5008 // activeTracks accumulates a copy of a subset of mActiveTracks 5009 Vector< sp<RecordTrack> > activeTracks; 5010 5011 // reference to the (first and only) fast track 5012 sp<RecordTrack> fastTrack; 5013 5014 { // scope for mLock 5015 Mutex::Autolock _l(mLock); 5016 5017 processConfigEvents_l(); 5018 5019 // check exitPending here because checkForNewParameters_l() and 5020 // checkForNewParameters_l() can temporarily release mLock 5021 if (exitPending()) { 5022 break; 5023 } 5024 5025 // if no active track(s), then standby and release wakelock 5026 size_t size = mActiveTracks.size(); 5027 if (size == 0) { 5028 standbyIfNotAlreadyInStandby(); 5029 // exitPending() can't become true here 5030 releaseWakeLock_l(); 5031 ALOGV("RecordThread: loop stopping"); 5032 // go to sleep 5033 mWaitWorkCV.wait(mLock); 5034 ALOGV("RecordThread: loop starting"); 5035 goto reacquire_wakelock; 5036 } 5037 5038 if (mActiveTracksGen != activeTracksGen) { 5039 activeTracksGen = mActiveTracksGen; 5040 SortedVector<int> tmp; 5041 for (size_t i = 0; i < size; i++) { 5042 tmp.add(mActiveTracks[i]->uid()); 5043 } 5044 updateWakeLockUids_l(tmp); 5045 } 5046 5047 bool doBroadcast = false; 5048 for (size_t i = 0; i < size; ) { 5049 5050 activeTrack = mActiveTracks[i]; 5051 if (activeTrack->isTerminated()) { 5052 removeTrack_l(activeTrack); 5053 mActiveTracks.remove(activeTrack); 5054 mActiveTracksGen++; 5055 size--; 5056 continue; 5057 } 5058 5059 TrackBase::track_state activeTrackState = activeTrack->mState; 5060 switch (activeTrackState) { 5061 5062 case TrackBase::PAUSING: 5063 mActiveTracks.remove(activeTrack); 5064 mActiveTracksGen++; 5065 doBroadcast = true; 5066 size--; 5067 continue; 5068 5069 case TrackBase::STARTING_1: 5070 sleepUs = 10000; 5071 i++; 5072 continue; 5073 5074 case TrackBase::STARTING_2: 5075 doBroadcast = true; 5076 mStandby = false; 5077 activeTrack->mState = TrackBase::ACTIVE; 5078 break; 5079 5080 case TrackBase::ACTIVE: 5081 break; 5082 5083 case TrackBase::IDLE: 5084 i++; 5085 continue; 5086 5087 default: 5088 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5089 } 5090 5091 activeTracks.add(activeTrack); 5092 i++; 5093 5094 if (activeTrack->isFastTrack()) { 5095 ALOG_ASSERT(!mFastTrackAvail); 5096 ALOG_ASSERT(fastTrack == 0); 5097 fastTrack = activeTrack; 5098 } 5099 } 5100 if (doBroadcast) { 5101 mStartStopCond.broadcast(); 5102 } 5103 5104 // sleep if there are no active tracks to process 5105 if (activeTracks.size() == 0) { 5106 if (sleepUs == 0) { 5107 sleepUs = kRecordThreadSleepUs; 5108 } 5109 continue; 5110 } 5111 sleepUs = 0; 5112 5113 lockEffectChains_l(effectChains); 5114 } 5115 5116 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5117 5118 size_t size = effectChains.size(); 5119 for (size_t i = 0; i < size; i++) { 5120 // thread mutex is not locked, but effect chain is locked 5121 effectChains[i]->process_l(); 5122 } 5123 5124 // Start the fast capture if it's not already running 5125 if (mFastCapture != 0) { 5126 FastCaptureStateQueue *sq = mFastCapture->sq(); 5127 FastCaptureState *state = sq->begin(); 5128 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5129 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5130 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5131 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5132 if (old == -1) { 5133 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5134 } 5135 } 5136 state->mCommand = FastCaptureState::READ_WRITE; 5137#if 0 // FIXME 5138 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5139 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5140#endif 5141 state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL; 5142 sq->end(); 5143 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5144#if 0 5145 if (kUseFastCapture == FastCapture_Dynamic) { 5146 mNormalSource = mPipeSource; 5147 } 5148#endif 5149 } else { 5150 sq->end(false /*didModify*/); 5151 } 5152 } 5153 5154 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5155 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5156 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5157 // If destination is non-contiguous, first read past the nominal end of buffer, then 5158 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5159 5160 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5161 ssize_t framesRead; 5162 5163 // If an NBAIO source is present, use it to read the normal capture's data 5164 if (mPipeSource != 0) { 5165 size_t framesToRead = mBufferSize / mFrameSize; 5166 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5167 framesToRead, AudioBufferProvider::kInvalidPTS); 5168 if (framesRead == 0) { 5169 // since pipe is non-blocking, simulate blocking input 5170 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5171 } 5172 // otherwise use the HAL / AudioStreamIn directly 5173 } else { 5174 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5175 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5176 if (bytesRead < 0) { 5177 framesRead = bytesRead; 5178 } else { 5179 framesRead = bytesRead / mFrameSize; 5180 } 5181 } 5182 5183 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5184 ALOGE("read failed: framesRead=%d", framesRead); 5185 // Force input into standby so that it tries to recover at next read attempt 5186 inputStandBy(); 5187 sleepUs = kRecordThreadSleepUs; 5188 } 5189 if (framesRead <= 0) { 5190 goto unlock; 5191 } 5192 ALOG_ASSERT(framesRead > 0); 5193 5194 if (mTeeSink != 0) { 5195 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5196 } 5197 // If destination is non-contiguous, we now correct for reading past end of buffer. 5198 { 5199 size_t part1 = mRsmpInFramesP2 - rear; 5200 if ((size_t) framesRead > part1) { 5201 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5202 (framesRead - part1) * mFrameSize); 5203 } 5204 } 5205 rear = mRsmpInRear += framesRead; 5206 5207 size = activeTracks.size(); 5208 // loop over each active track 5209 for (size_t i = 0; i < size; i++) { 5210 activeTrack = activeTracks[i]; 5211 5212 // skip fast tracks, as those are handled directly by FastCapture 5213 if (activeTrack->isFastTrack()) { 5214 continue; 5215 } 5216 5217 enum { 5218 OVERRUN_UNKNOWN, 5219 OVERRUN_TRUE, 5220 OVERRUN_FALSE 5221 } overrun = OVERRUN_UNKNOWN; 5222 5223 // loop over getNextBuffer to handle circular sink 5224 for (;;) { 5225 5226 activeTrack->mSink.frameCount = ~0; 5227 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5228 size_t framesOut = activeTrack->mSink.frameCount; 5229 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5230 5231 int32_t front = activeTrack->mRsmpInFront; 5232 ssize_t filled = rear - front; 5233 size_t framesIn; 5234 5235 if (filled < 0) { 5236 // should not happen, but treat like a massive overrun and re-sync 5237 framesIn = 0; 5238 activeTrack->mRsmpInFront = rear; 5239 overrun = OVERRUN_TRUE; 5240 } else if ((size_t) filled <= mRsmpInFrames) { 5241 framesIn = (size_t) filled; 5242 } else { 5243 // client is not keeping up with server, but give it latest data 5244 framesIn = mRsmpInFrames; 5245 activeTrack->mRsmpInFront = front = rear - framesIn; 5246 overrun = OVERRUN_TRUE; 5247 } 5248 5249 if (framesOut == 0 || framesIn == 0) { 5250 break; 5251 } 5252 5253 if (activeTrack->mResampler == NULL) { 5254 // no resampling 5255 if (framesIn > framesOut) { 5256 framesIn = framesOut; 5257 } else { 5258 framesOut = framesIn; 5259 } 5260 int8_t *dst = activeTrack->mSink.i8; 5261 while (framesIn > 0) { 5262 front &= mRsmpInFramesP2 - 1; 5263 size_t part1 = mRsmpInFramesP2 - front; 5264 if (part1 > framesIn) { 5265 part1 = framesIn; 5266 } 5267 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5268 if (mChannelCount == activeTrack->mChannelCount) { 5269 memcpy(dst, src, part1 * mFrameSize); 5270 } else if (mChannelCount == 1) { 5271 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5272 part1); 5273 } else { 5274 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5275 part1); 5276 } 5277 dst += part1 * activeTrack->mFrameSize; 5278 front += part1; 5279 framesIn -= part1; 5280 } 5281 activeTrack->mRsmpInFront += framesOut; 5282 5283 } else { 5284 // resampling 5285 // FIXME framesInNeeded should really be part of resampler API, and should 5286 // depend on the SRC ratio 5287 // to keep mRsmpInBuffer full so resampler always has sufficient input 5288 size_t framesInNeeded; 5289 // FIXME only re-calculate when it changes, and optimize for common ratios 5290 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 5291 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 5292 framesInNeeded = ceil(framesOut * inOverOut) + 1; 5293 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5294 framesInNeeded, framesOut, inOverOut); 5295 // Although we theoretically have framesIn in circular buffer, some of those are 5296 // unreleased frames, and thus must be discounted for purpose of budgeting. 5297 size_t unreleased = activeTrack->mRsmpInUnrel; 5298 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5299 if (framesIn < framesInNeeded) { 5300 ALOGV("not enough to resample: have %u frames in but need %u in to " 5301 "produce %u out given in/out ratio of %.4g", 5302 framesIn, framesInNeeded, framesOut, inOverOut); 5303 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 5304 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5305 if (newFramesOut == 0) { 5306 break; 5307 } 5308 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 5309 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5310 framesInNeeded, newFramesOut, outOverIn); 5311 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5312 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5313 "given in/out ratio of %.4g", 5314 framesIn, framesInNeeded, newFramesOut, inOverOut); 5315 framesOut = newFramesOut; 5316 } else { 5317 ALOGV("success 1: have %u in and need %u in to produce %u out " 5318 "given in/out ratio of %.4g", 5319 framesIn, framesInNeeded, framesOut, inOverOut); 5320 } 5321 5322 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5323 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5324 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5325 delete[] activeTrack->mRsmpOutBuffer; 5326 // resampler always outputs stereo 5327 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5328 activeTrack->mRsmpOutFrameCount = framesOut; 5329 } 5330 5331 // resampler accumulates, but we only have one source track 5332 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5333 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5334 // FIXME how about having activeTrack implement this interface itself? 5335 activeTrack->mResamplerBufferProvider 5336 /*this*/ /* AudioBufferProvider* */); 5337 // ditherAndClamp() works as long as all buffers returned by 5338 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5339 if (activeTrack->mChannelCount == 1) { 5340 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5341 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5342 framesOut); 5343 // the resampler always outputs stereo samples: 5344 // do post stereo to mono conversion 5345 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5346 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5347 } else { 5348 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5349 activeTrack->mRsmpOutBuffer, framesOut); 5350 } 5351 // now done with mRsmpOutBuffer 5352 5353 } 5354 5355 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5356 overrun = OVERRUN_FALSE; 5357 } 5358 5359 if (activeTrack->mFramesToDrop == 0) { 5360 if (framesOut > 0) { 5361 activeTrack->mSink.frameCount = framesOut; 5362 activeTrack->releaseBuffer(&activeTrack->mSink); 5363 } 5364 } else { 5365 // FIXME could do a partial drop of framesOut 5366 if (activeTrack->mFramesToDrop > 0) { 5367 activeTrack->mFramesToDrop -= framesOut; 5368 if (activeTrack->mFramesToDrop <= 0) { 5369 activeTrack->clearSyncStartEvent(); 5370 } 5371 } else { 5372 activeTrack->mFramesToDrop += framesOut; 5373 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5374 activeTrack->mSyncStartEvent->isCancelled()) { 5375 ALOGW("Synced record %s, session %d, trigger session %d", 5376 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5377 activeTrack->sessionId(), 5378 (activeTrack->mSyncStartEvent != 0) ? 5379 activeTrack->mSyncStartEvent->triggerSession() : 0); 5380 activeTrack->clearSyncStartEvent(); 5381 } 5382 } 5383 } 5384 5385 if (framesOut == 0) { 5386 break; 5387 } 5388 } 5389 5390 switch (overrun) { 5391 case OVERRUN_TRUE: 5392 // client isn't retrieving buffers fast enough 5393 if (!activeTrack->setOverflow()) { 5394 nsecs_t now = systemTime(); 5395 // FIXME should lastWarning per track? 5396 if ((now - lastWarning) > kWarningThrottleNs) { 5397 ALOGW("RecordThread: buffer overflow"); 5398 lastWarning = now; 5399 } 5400 } 5401 break; 5402 case OVERRUN_FALSE: 5403 activeTrack->clearOverflow(); 5404 break; 5405 case OVERRUN_UNKNOWN: 5406 break; 5407 } 5408 5409 } 5410 5411unlock: 5412 // enable changes in effect chain 5413 unlockEffectChains(effectChains); 5414 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5415 } 5416 5417 standbyIfNotAlreadyInStandby(); 5418 5419 { 5420 Mutex::Autolock _l(mLock); 5421 for (size_t i = 0; i < mTracks.size(); i++) { 5422 sp<RecordTrack> track = mTracks[i]; 5423 track->invalidate(); 5424 } 5425 mActiveTracks.clear(); 5426 mActiveTracksGen++; 5427 mStartStopCond.broadcast(); 5428 } 5429 5430 releaseWakeLock(); 5431 5432 ALOGV("RecordThread %p exiting", this); 5433 return false; 5434} 5435 5436void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5437{ 5438 if (!mStandby) { 5439 inputStandBy(); 5440 mStandby = true; 5441 } 5442} 5443 5444void AudioFlinger::RecordThread::inputStandBy() 5445{ 5446 // Idle the fast capture if it's currently running 5447 if (mFastCapture != 0) { 5448 FastCaptureStateQueue *sq = mFastCapture->sq(); 5449 FastCaptureState *state = sq->begin(); 5450 if (!(state->mCommand & FastCaptureState::IDLE)) { 5451 state->mCommand = FastCaptureState::COLD_IDLE; 5452 state->mColdFutexAddr = &mFastCaptureFutex; 5453 state->mColdGen++; 5454 mFastCaptureFutex = 0; 5455 sq->end(); 5456 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5457 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5458#if 0 5459 if (kUseFastCapture == FastCapture_Dynamic) { 5460 // FIXME 5461 } 5462#endif 5463#ifdef AUDIO_WATCHDOG 5464 // FIXME 5465#endif 5466 } else { 5467 sq->end(false /*didModify*/); 5468 } 5469 } 5470 mInput->stream->common.standby(&mInput->stream->common); 5471} 5472 5473// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5474sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5475 const sp<AudioFlinger::Client>& client, 5476 uint32_t sampleRate, 5477 audio_format_t format, 5478 audio_channel_mask_t channelMask, 5479 size_t *pFrameCount, 5480 int sessionId, 5481 size_t *notificationFrames, 5482 int uid, 5483 IAudioFlinger::track_flags_t *flags, 5484 pid_t tid, 5485 status_t *status) 5486{ 5487 size_t frameCount = *pFrameCount; 5488 sp<RecordTrack> track; 5489 status_t lStatus; 5490 5491 // client expresses a preference for FAST, but we get the final say 5492 if (*flags & IAudioFlinger::TRACK_FAST) { 5493 if ( 5494 // use case: callback handler 5495 (tid != -1) && 5496 // frame count is not specified, or is exactly the pipe depth 5497 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5498 // PCM data 5499 audio_is_linear_pcm(format) && 5500 // native format 5501 (format == mFormat) && 5502 // native channel mask 5503 (channelMask == mChannelMask) && 5504 // native hardware sample rate 5505 (sampleRate == mSampleRate) && 5506 // record thread has an associated fast capture 5507 hasFastCapture() && 5508 // there are sufficient fast track slots available 5509 mFastTrackAvail 5510 ) { 5511 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5512 frameCount, mFrameCount); 5513 } else { 5514 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5515 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5516 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5517 frameCount, mFrameCount, mPipeFramesP2, 5518 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5519 hasFastCapture(), tid, mFastTrackAvail); 5520 *flags &= ~IAudioFlinger::TRACK_FAST; 5521 } 5522 } 5523 5524 // compute track buffer size in frames, and suggest the notification frame count 5525 if (*flags & IAudioFlinger::TRACK_FAST) { 5526 // fast track: frame count is exactly the pipe depth 5527 frameCount = mPipeFramesP2; 5528 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5529 *notificationFrames = mFrameCount; 5530 } else { 5531 // not fast track: frame count is at least 2 HAL buffers and at least 20 ms 5532 size_t minFrameCount = ((int64_t) mFrameCount * 2 * sampleRate + mSampleRate - 1) / 5533 mSampleRate; 5534 if (frameCount < minFrameCount) { 5535 frameCount = minFrameCount; 5536 } 5537 minFrameCount = (sampleRate * 20 / 1000 + 1) & ~1; 5538 if (frameCount < minFrameCount) { 5539 frameCount = minFrameCount; 5540 } 5541 // notification is forced to be at least double-buffering 5542 size_t maxNotification = frameCount / 2; 5543 if (*notificationFrames == 0 || *notificationFrames > maxNotification) { 5544 *notificationFrames = maxNotification; 5545 } 5546 } 5547 *pFrameCount = frameCount; 5548 5549 lStatus = initCheck(); 5550 if (lStatus != NO_ERROR) { 5551 ALOGE("createRecordTrack_l() audio driver not initialized"); 5552 goto Exit; 5553 } 5554 5555 { // scope for mLock 5556 Mutex::Autolock _l(mLock); 5557 5558 track = new RecordTrack(this, client, sampleRate, 5559 format, channelMask, frameCount, NULL, sessionId, uid, 5560 *flags, TrackBase::TYPE_DEFAULT); 5561 5562 lStatus = track->initCheck(); 5563 if (lStatus != NO_ERROR) { 5564 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5565 // track must be cleared from the caller as the caller has the AF lock 5566 goto Exit; 5567 } 5568 mTracks.add(track); 5569 5570 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5571 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5572 mAudioFlinger->btNrecIsOff(); 5573 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5574 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5575 5576 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5577 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5578 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5579 // so ask activity manager to do this on our behalf 5580 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5581 } 5582 } 5583 5584 lStatus = NO_ERROR; 5585 5586Exit: 5587 *status = lStatus; 5588 return track; 5589} 5590 5591status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5592 AudioSystem::sync_event_t event, 5593 int triggerSession) 5594{ 5595 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5596 sp<ThreadBase> strongMe = this; 5597 status_t status = NO_ERROR; 5598 5599 if (event == AudioSystem::SYNC_EVENT_NONE) { 5600 recordTrack->clearSyncStartEvent(); 5601 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5602 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5603 triggerSession, 5604 recordTrack->sessionId(), 5605 syncStartEventCallback, 5606 recordTrack); 5607 // Sync event can be cancelled by the trigger session if the track is not in a 5608 // compatible state in which case we start record immediately 5609 if (recordTrack->mSyncStartEvent->isCancelled()) { 5610 recordTrack->clearSyncStartEvent(); 5611 } else { 5612 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5613 recordTrack->mFramesToDrop = - 5614 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5615 } 5616 } 5617 5618 { 5619 // This section is a rendezvous between binder thread executing start() and RecordThread 5620 AutoMutex lock(mLock); 5621 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5622 if (recordTrack->mState == TrackBase::PAUSING) { 5623 ALOGV("active record track PAUSING -> ACTIVE"); 5624 recordTrack->mState = TrackBase::ACTIVE; 5625 } else { 5626 ALOGV("active record track state %d", recordTrack->mState); 5627 } 5628 return status; 5629 } 5630 5631 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5632 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5633 // or using a separate command thread 5634 recordTrack->mState = TrackBase::STARTING_1; 5635 mActiveTracks.add(recordTrack); 5636 mActiveTracksGen++; 5637 status_t status = NO_ERROR; 5638 if (recordTrack->isExternalTrack()) { 5639 mLock.unlock(); 5640 status = AudioSystem::startInput(mId); 5641 mLock.lock(); 5642 // FIXME should verify that recordTrack is still in mActiveTracks 5643 if (status != NO_ERROR) { 5644 mActiveTracks.remove(recordTrack); 5645 mActiveTracksGen++; 5646 recordTrack->clearSyncStartEvent(); 5647 ALOGV("RecordThread::start error %d", status); 5648 return status; 5649 } 5650 } 5651 // Catch up with current buffer indices if thread is already running. 5652 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5653 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5654 // see previously buffered data before it called start(), but with greater risk of overrun. 5655 5656 recordTrack->mRsmpInFront = mRsmpInRear; 5657 recordTrack->mRsmpInUnrel = 0; 5658 // FIXME why reset? 5659 if (recordTrack->mResampler != NULL) { 5660 recordTrack->mResampler->reset(); 5661 } 5662 recordTrack->mState = TrackBase::STARTING_2; 5663 // signal thread to start 5664 mWaitWorkCV.broadcast(); 5665 if (mActiveTracks.indexOf(recordTrack) < 0) { 5666 ALOGV("Record failed to start"); 5667 status = BAD_VALUE; 5668 goto startError; 5669 } 5670 return status; 5671 } 5672 5673startError: 5674 if (recordTrack->isExternalTrack()) { 5675 AudioSystem::stopInput(mId); 5676 } 5677 recordTrack->clearSyncStartEvent(); 5678 // FIXME I wonder why we do not reset the state here? 5679 return status; 5680} 5681 5682void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5683{ 5684 sp<SyncEvent> strongEvent = event.promote(); 5685 5686 if (strongEvent != 0) { 5687 sp<RefBase> ptr = strongEvent->cookie().promote(); 5688 if (ptr != 0) { 5689 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5690 recordTrack->handleSyncStartEvent(strongEvent); 5691 } 5692 } 5693} 5694 5695bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5696 ALOGV("RecordThread::stop"); 5697 AutoMutex _l(mLock); 5698 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5699 return false; 5700 } 5701 // note that threadLoop may still be processing the track at this point [without lock] 5702 recordTrack->mState = TrackBase::PAUSING; 5703 // do not wait for mStartStopCond if exiting 5704 if (exitPending()) { 5705 return true; 5706 } 5707 // FIXME incorrect usage of wait: no explicit predicate or loop 5708 mStartStopCond.wait(mLock); 5709 // if we have been restarted, recordTrack is in mActiveTracks here 5710 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5711 ALOGV("Record stopped OK"); 5712 return true; 5713 } 5714 return false; 5715} 5716 5717bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5718{ 5719 return false; 5720} 5721 5722status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5723{ 5724#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5725 if (!isValidSyncEvent(event)) { 5726 return BAD_VALUE; 5727 } 5728 5729 int eventSession = event->triggerSession(); 5730 status_t ret = NAME_NOT_FOUND; 5731 5732 Mutex::Autolock _l(mLock); 5733 5734 for (size_t i = 0; i < mTracks.size(); i++) { 5735 sp<RecordTrack> track = mTracks[i]; 5736 if (eventSession == track->sessionId()) { 5737 (void) track->setSyncEvent(event); 5738 ret = NO_ERROR; 5739 } 5740 } 5741 return ret; 5742#else 5743 return BAD_VALUE; 5744#endif 5745} 5746 5747// destroyTrack_l() must be called with ThreadBase::mLock held 5748void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5749{ 5750 track->terminate(); 5751 track->mState = TrackBase::STOPPED; 5752 // active tracks are removed by threadLoop() 5753 if (mActiveTracks.indexOf(track) < 0) { 5754 removeTrack_l(track); 5755 } 5756} 5757 5758void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5759{ 5760 mTracks.remove(track); 5761 // need anything related to effects here? 5762 if (track->isFastTrack()) { 5763 ALOG_ASSERT(!mFastTrackAvail); 5764 mFastTrackAvail = true; 5765 } 5766} 5767 5768void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5769{ 5770 dumpInternals(fd, args); 5771 dumpTracks(fd, args); 5772 dumpEffectChains(fd, args); 5773} 5774 5775void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5776{ 5777 dprintf(fd, "\nInput thread %p:\n", this); 5778 5779 if (mActiveTracks.size() > 0) { 5780 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5781 } else { 5782 dprintf(fd, " No active record clients\n"); 5783 } 5784 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5785 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5786 5787 dumpBase(fd, args); 5788} 5789 5790void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5791{ 5792 const size_t SIZE = 256; 5793 char buffer[SIZE]; 5794 String8 result; 5795 5796 size_t numtracks = mTracks.size(); 5797 size_t numactive = mActiveTracks.size(); 5798 size_t numactiveseen = 0; 5799 dprintf(fd, " %d Tracks", numtracks); 5800 if (numtracks) { 5801 dprintf(fd, " of which %d are active\n", numactive); 5802 RecordTrack::appendDumpHeader(result); 5803 for (size_t i = 0; i < numtracks ; ++i) { 5804 sp<RecordTrack> track = mTracks[i]; 5805 if (track != 0) { 5806 bool active = mActiveTracks.indexOf(track) >= 0; 5807 if (active) { 5808 numactiveseen++; 5809 } 5810 track->dump(buffer, SIZE, active); 5811 result.append(buffer); 5812 } 5813 } 5814 } else { 5815 dprintf(fd, "\n"); 5816 } 5817 5818 if (numactiveseen != numactive) { 5819 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5820 " not in the track list\n"); 5821 result.append(buffer); 5822 RecordTrack::appendDumpHeader(result); 5823 for (size_t i = 0; i < numactive; ++i) { 5824 sp<RecordTrack> track = mActiveTracks[i]; 5825 if (mTracks.indexOf(track) < 0) { 5826 track->dump(buffer, SIZE, true); 5827 result.append(buffer); 5828 } 5829 } 5830 5831 } 5832 write(fd, result.string(), result.size()); 5833} 5834 5835// AudioBufferProvider interface 5836status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5837 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5838{ 5839 RecordTrack *activeTrack = mRecordTrack; 5840 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5841 if (threadBase == 0) { 5842 buffer->frameCount = 0; 5843 buffer->raw = NULL; 5844 return NOT_ENOUGH_DATA; 5845 } 5846 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5847 int32_t rear = recordThread->mRsmpInRear; 5848 int32_t front = activeTrack->mRsmpInFront; 5849 ssize_t filled = rear - front; 5850 // FIXME should not be P2 (don't want to increase latency) 5851 // FIXME if client not keeping up, discard 5852 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5853 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5854 front &= recordThread->mRsmpInFramesP2 - 1; 5855 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5856 if (part1 > (size_t) filled) { 5857 part1 = filled; 5858 } 5859 size_t ask = buffer->frameCount; 5860 ALOG_ASSERT(ask > 0); 5861 if (part1 > ask) { 5862 part1 = ask; 5863 } 5864 if (part1 == 0) { 5865 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5866 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5867 buffer->raw = NULL; 5868 buffer->frameCount = 0; 5869 activeTrack->mRsmpInUnrel = 0; 5870 return NOT_ENOUGH_DATA; 5871 } 5872 5873 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5874 buffer->frameCount = part1; 5875 activeTrack->mRsmpInUnrel = part1; 5876 return NO_ERROR; 5877} 5878 5879// AudioBufferProvider interface 5880void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5881 AudioBufferProvider::Buffer* buffer) 5882{ 5883 RecordTrack *activeTrack = mRecordTrack; 5884 size_t stepCount = buffer->frameCount; 5885 if (stepCount == 0) { 5886 return; 5887 } 5888 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5889 activeTrack->mRsmpInUnrel -= stepCount; 5890 activeTrack->mRsmpInFront += stepCount; 5891 buffer->raw = NULL; 5892 buffer->frameCount = 0; 5893} 5894 5895bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5896 status_t& status) 5897{ 5898 bool reconfig = false; 5899 5900 status = NO_ERROR; 5901 5902 audio_format_t reqFormat = mFormat; 5903 uint32_t samplingRate = mSampleRate; 5904 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5905 5906 AudioParameter param = AudioParameter(keyValuePair); 5907 int value; 5908 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5909 // channel count change can be requested. Do we mandate the first client defines the 5910 // HAL sampling rate and channel count or do we allow changes on the fly? 5911 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5912 samplingRate = value; 5913 reconfig = true; 5914 } 5915 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5916 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5917 status = BAD_VALUE; 5918 } else { 5919 reqFormat = (audio_format_t) value; 5920 reconfig = true; 5921 } 5922 } 5923 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5924 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5925 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5926 status = BAD_VALUE; 5927 } else { 5928 channelMask = mask; 5929 reconfig = true; 5930 } 5931 } 5932 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5933 // do not accept frame count changes if tracks are open as the track buffer 5934 // size depends on frame count and correct behavior would not be guaranteed 5935 // if frame count is changed after track creation 5936 if (mActiveTracks.size() > 0) { 5937 status = INVALID_OPERATION; 5938 } else { 5939 reconfig = true; 5940 } 5941 } 5942 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5943 // forward device change to effects that have requested to be 5944 // aware of attached audio device. 5945 for (size_t i = 0; i < mEffectChains.size(); i++) { 5946 mEffectChains[i]->setDevice_l(value); 5947 } 5948 5949 // store input device and output device but do not forward output device to audio HAL. 5950 // Note that status is ignored by the caller for output device 5951 // (see AudioFlinger::setParameters() 5952 if (audio_is_output_devices(value)) { 5953 mOutDevice = value; 5954 status = BAD_VALUE; 5955 } else { 5956 mInDevice = value; 5957 // disable AEC and NS if the device is a BT SCO headset supporting those 5958 // pre processings 5959 if (mTracks.size() > 0) { 5960 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5961 mAudioFlinger->btNrecIsOff(); 5962 for (size_t i = 0; i < mTracks.size(); i++) { 5963 sp<RecordTrack> track = mTracks[i]; 5964 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5965 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5966 } 5967 } 5968 } 5969 } 5970 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5971 mAudioSource != (audio_source_t)value) { 5972 // forward device change to effects that have requested to be 5973 // aware of attached audio device. 5974 for (size_t i = 0; i < mEffectChains.size(); i++) { 5975 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5976 } 5977 mAudioSource = (audio_source_t)value; 5978 } 5979 5980 if (status == NO_ERROR) { 5981 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5982 keyValuePair.string()); 5983 if (status == INVALID_OPERATION) { 5984 inputStandBy(); 5985 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5986 keyValuePair.string()); 5987 } 5988 if (reconfig) { 5989 if (status == BAD_VALUE && 5990 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5991 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5992 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5993 <= (2 * samplingRate)) && 5994 audio_channel_count_from_in_mask( 5995 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5996 (channelMask == AUDIO_CHANNEL_IN_MONO || 5997 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5998 status = NO_ERROR; 5999 } 6000 if (status == NO_ERROR) { 6001 readInputParameters_l(); 6002 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6003 } 6004 } 6005 } 6006 6007 return reconfig; 6008} 6009 6010String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6011{ 6012 Mutex::Autolock _l(mLock); 6013 if (initCheck() != NO_ERROR) { 6014 return String8(); 6015 } 6016 6017 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6018 const String8 out_s8(s); 6019 free(s); 6020 return out_s8; 6021} 6022 6023void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6024 AudioSystem::OutputDescriptor desc; 6025 const void *param2 = NULL; 6026 6027 switch (event) { 6028 case AudioSystem::INPUT_OPENED: 6029 case AudioSystem::INPUT_CONFIG_CHANGED: 6030 desc.channelMask = mChannelMask; 6031 desc.samplingRate = mSampleRate; 6032 desc.format = mFormat; 6033 desc.frameCount = mFrameCount; 6034 desc.latency = 0; 6035 param2 = &desc; 6036 break; 6037 6038 case AudioSystem::INPUT_CLOSED: 6039 default: 6040 break; 6041 } 6042 mAudioFlinger->audioConfigChanged(event, mId, param2); 6043} 6044 6045void AudioFlinger::RecordThread::readInputParameters_l() 6046{ 6047 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6048 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6049 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6050 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6051 mFormat = mHALFormat; 6052 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6053 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6054 } 6055 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6056 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6057 mFrameCount = mBufferSize / mFrameSize; 6058 // This is the formula for calculating the temporary buffer size. 6059 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6060 // 1 full output buffer, regardless of the alignment of the available input. 6061 // The value is somewhat arbitrary, and could probably be even larger. 6062 // A larger value should allow more old data to be read after a track calls start(), 6063 // without increasing latency. 6064 mRsmpInFrames = mFrameCount * 7; 6065 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6066 delete[] mRsmpInBuffer; 6067 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6068 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6069 6070 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6071 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6072} 6073 6074uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6075{ 6076 Mutex::Autolock _l(mLock); 6077 if (initCheck() != NO_ERROR) { 6078 return 0; 6079 } 6080 6081 return mInput->stream->get_input_frames_lost(mInput->stream); 6082} 6083 6084uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6085{ 6086 Mutex::Autolock _l(mLock); 6087 uint32_t result = 0; 6088 if (getEffectChain_l(sessionId) != 0) { 6089 result = EFFECT_SESSION; 6090 } 6091 6092 for (size_t i = 0; i < mTracks.size(); ++i) { 6093 if (sessionId == mTracks[i]->sessionId()) { 6094 result |= TRACK_SESSION; 6095 break; 6096 } 6097 } 6098 6099 return result; 6100} 6101 6102KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6103{ 6104 KeyedVector<int, bool> ids; 6105 Mutex::Autolock _l(mLock); 6106 for (size_t j = 0; j < mTracks.size(); ++j) { 6107 sp<RecordThread::RecordTrack> track = mTracks[j]; 6108 int sessionId = track->sessionId(); 6109 if (ids.indexOfKey(sessionId) < 0) { 6110 ids.add(sessionId, true); 6111 } 6112 } 6113 return ids; 6114} 6115 6116AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6117{ 6118 Mutex::Autolock _l(mLock); 6119 AudioStreamIn *input = mInput; 6120 mInput = NULL; 6121 return input; 6122} 6123 6124// this method must always be called either with ThreadBase mLock held or inside the thread loop 6125audio_stream_t* AudioFlinger::RecordThread::stream() const 6126{ 6127 if (mInput == NULL) { 6128 return NULL; 6129 } 6130 return &mInput->stream->common; 6131} 6132 6133status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6134{ 6135 // only one chain per input thread 6136 if (mEffectChains.size() != 0) { 6137 return INVALID_OPERATION; 6138 } 6139 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6140 6141 chain->setInBuffer(NULL); 6142 chain->setOutBuffer(NULL); 6143 6144 checkSuspendOnAddEffectChain_l(chain); 6145 6146 mEffectChains.add(chain); 6147 6148 return NO_ERROR; 6149} 6150 6151size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6152{ 6153 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6154 ALOGW_IF(mEffectChains.size() != 1, 6155 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6156 chain.get(), mEffectChains.size(), this); 6157 if (mEffectChains.size() == 1) { 6158 mEffectChains.removeAt(0); 6159 } 6160 return 0; 6161} 6162 6163status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6164 audio_patch_handle_t *handle) 6165{ 6166 status_t status = NO_ERROR; 6167 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6168 // store new device and send to effects 6169 mInDevice = patch->sources[0].ext.device.type; 6170 for (size_t i = 0; i < mEffectChains.size(); i++) { 6171 mEffectChains[i]->setDevice_l(mInDevice); 6172 } 6173 6174 // disable AEC and NS if the device is a BT SCO headset supporting those 6175 // pre processings 6176 if (mTracks.size() > 0) { 6177 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6178 mAudioFlinger->btNrecIsOff(); 6179 for (size_t i = 0; i < mTracks.size(); i++) { 6180 sp<RecordTrack> track = mTracks[i]; 6181 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6182 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6183 } 6184 } 6185 6186 // store new source and send to effects 6187 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6188 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6189 for (size_t i = 0; i < mEffectChains.size(); i++) { 6190 mEffectChains[i]->setAudioSource_l(mAudioSource); 6191 } 6192 } 6193 6194 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6195 status = hwDevice->create_audio_patch(hwDevice, 6196 patch->num_sources, 6197 patch->sources, 6198 patch->num_sinks, 6199 patch->sinks, 6200 handle); 6201 } else { 6202 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6203 } 6204 return status; 6205} 6206 6207status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6208{ 6209 status_t status = NO_ERROR; 6210 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6211 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6212 status = hwDevice->release_audio_patch(hwDevice, handle); 6213 } else { 6214 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6215 } 6216 return status; 6217} 6218 6219void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6220{ 6221 Mutex::Autolock _l(mLock); 6222 mTracks.add(record); 6223} 6224 6225void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6226{ 6227 Mutex::Autolock _l(mLock); 6228 destroyTrack_l(record); 6229} 6230 6231void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6232{ 6233 ThreadBase::getAudioPortConfig(config); 6234 config->role = AUDIO_PORT_ROLE_SINK; 6235 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6236 config->ext.mix.usecase.source = mAudioSource; 6237} 6238 6239}; // namespace android 6240