Threads.cpp revision 2e8186a258c934798129847f66171df36e95d23e
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 1149// ---------------------------------------------------------------------------- 1150// Playback 1151// ---------------------------------------------------------------------------- 1152 1153AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1154 AudioStreamOut* output, 1155 audio_io_handle_t id, 1156 audio_devices_t device, 1157 type_t type) 1158 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1159 mNormalFrameCount(0), mSinkBuffer(NULL), 1160 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1161 mMixerBuffer(NULL), 1162 mMixerBufferSize(0), 1163 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1164 mMixerBufferValid(false), 1165 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1166 mEffectBuffer(NULL), 1167 mEffectBufferSize(0), 1168 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1169 mEffectBufferValid(false), 1170 mSuspended(0), mBytesWritten(0), 1171 mActiveTracksGeneration(0), 1172 // mStreamTypes[] initialized in constructor body 1173 mOutput(output), 1174 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1175 mMixerStatus(MIXER_IDLE), 1176 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1177 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1178 mBytesRemaining(0), 1179 mCurrentWriteLength(0), 1180 mUseAsyncWrite(false), 1181 mWriteAckSequence(0), 1182 mDrainSequence(0), 1183 mSignalPending(false), 1184 mScreenState(AudioFlinger::mScreenState), 1185 // index 0 is reserved for normal mixer's submix 1186 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1187 // mLatchD, mLatchQ, 1188 mLatchDValid(false), mLatchQValid(false) 1189{ 1190 snprintf(mName, kNameLength, "AudioOut_%X", id); 1191 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1192 1193 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1194 // it would be safer to explicitly pass initial masterVolume/masterMute as 1195 // parameter. 1196 // 1197 // If the HAL we are using has support for master volume or master mute, 1198 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1199 // and the mute set to false). 1200 mMasterVolume = audioFlinger->masterVolume_l(); 1201 mMasterMute = audioFlinger->masterMute_l(); 1202 if (mOutput && mOutput->audioHwDev) { 1203 if (mOutput->audioHwDev->canSetMasterVolume()) { 1204 mMasterVolume = 1.0; 1205 } 1206 1207 if (mOutput->audioHwDev->canSetMasterMute()) { 1208 mMasterMute = false; 1209 } 1210 } 1211 1212 readOutputParameters_l(); 1213 1214 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1215 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1216 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1217 stream = (audio_stream_type_t) (stream + 1)) { 1218 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1219 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1220 } 1221 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1222 // because mAudioFlinger doesn't have one to copy from 1223} 1224 1225AudioFlinger::PlaybackThread::~PlaybackThread() 1226{ 1227 mAudioFlinger->unregisterWriter(mNBLogWriter); 1228 free(mSinkBuffer); 1229 free(mMixerBuffer); 1230 free(mEffectBuffer); 1231} 1232 1233void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1234{ 1235 dumpInternals(fd, args); 1236 dumpTracks(fd, args); 1237 dumpEffectChains(fd, args); 1238} 1239 1240void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1241{ 1242 const size_t SIZE = 256; 1243 char buffer[SIZE]; 1244 String8 result; 1245 1246 result.appendFormat(" Stream volumes in dB: "); 1247 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1248 const stream_type_t *st = &mStreamTypes[i]; 1249 if (i > 0) { 1250 result.appendFormat(", "); 1251 } 1252 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1253 if (st->mute) { 1254 result.append("M"); 1255 } 1256 } 1257 result.append("\n"); 1258 write(fd, result.string(), result.length()); 1259 result.clear(); 1260 1261 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1262 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1263 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1264 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1265 1266 size_t numtracks = mTracks.size(); 1267 size_t numactive = mActiveTracks.size(); 1268 dprintf(fd, " %d Tracks", numtracks); 1269 size_t numactiveseen = 0; 1270 if (numtracks) { 1271 dprintf(fd, " of which %d are active\n", numactive); 1272 Track::appendDumpHeader(result); 1273 for (size_t i = 0; i < numtracks; ++i) { 1274 sp<Track> track = mTracks[i]; 1275 if (track != 0) { 1276 bool active = mActiveTracks.indexOf(track) >= 0; 1277 if (active) { 1278 numactiveseen++; 1279 } 1280 track->dump(buffer, SIZE, active); 1281 result.append(buffer); 1282 } 1283 } 1284 } else { 1285 result.append("\n"); 1286 } 1287 if (numactiveseen != numactive) { 1288 // some tracks in the active list were not in the tracks list 1289 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1290 " not in the track list\n"); 1291 result.append(buffer); 1292 Track::appendDumpHeader(result); 1293 for (size_t i = 0; i < numactive; ++i) { 1294 sp<Track> track = mActiveTracks[i].promote(); 1295 if (track != 0 && mTracks.indexOf(track) < 0) { 1296 track->dump(buffer, SIZE, true); 1297 result.append(buffer); 1298 } 1299 } 1300 } 1301 1302 write(fd, result.string(), result.size()); 1303} 1304 1305void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1306{ 1307 dprintf(fd, "\nOutput thread %p:\n", this); 1308 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1309 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1310 dprintf(fd, " Total writes: %d\n", mNumWrites); 1311 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1312 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1313 dprintf(fd, " Suspend count: %d\n", mSuspended); 1314 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1315 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1316 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1317 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1318 1319 dumpBase(fd, args); 1320} 1321 1322// Thread virtuals 1323 1324void AudioFlinger::PlaybackThread::onFirstRef() 1325{ 1326 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1327} 1328 1329// ThreadBase virtuals 1330void AudioFlinger::PlaybackThread::preExit() 1331{ 1332 ALOGV(" preExit()"); 1333 // FIXME this is using hard-coded strings but in the future, this functionality will be 1334 // converted to use audio HAL extensions required to support tunneling 1335 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1336} 1337 1338// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1339sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1340 const sp<AudioFlinger::Client>& client, 1341 audio_stream_type_t streamType, 1342 uint32_t sampleRate, 1343 audio_format_t format, 1344 audio_channel_mask_t channelMask, 1345 size_t *pFrameCount, 1346 const sp<IMemory>& sharedBuffer, 1347 int sessionId, 1348 IAudioFlinger::track_flags_t *flags, 1349 pid_t tid, 1350 int uid, 1351 status_t *status) 1352{ 1353 size_t frameCount = *pFrameCount; 1354 sp<Track> track; 1355 status_t lStatus; 1356 1357 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1358 1359 // client expresses a preference for FAST, but we get the final say 1360 if (*flags & IAudioFlinger::TRACK_FAST) { 1361 if ( 1362 // not timed 1363 (!isTimed) && 1364 // either of these use cases: 1365 ( 1366 // use case 1: shared buffer with any frame count 1367 ( 1368 (sharedBuffer != 0) 1369 ) || 1370 // use case 2: callback handler and frame count is default or at least as large as HAL 1371 ( 1372 (tid != -1) && 1373 ((frameCount == 0) || 1374 (frameCount >= mFrameCount)) 1375 ) 1376 ) && 1377 // PCM data 1378 audio_is_linear_pcm(format) && 1379 // mono or stereo 1380 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1381 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1382 // hardware sample rate 1383 (sampleRate == mSampleRate) && 1384 // normal mixer has an associated fast mixer 1385 hasFastMixer() && 1386 // there are sufficient fast track slots available 1387 (mFastTrackAvailMask != 0) 1388 // FIXME test that MixerThread for this fast track has a capable output HAL 1389 // FIXME add a permission test also? 1390 ) { 1391 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1392 if (frameCount == 0) { 1393 // read the fast track multiplier property the first time it is needed 1394 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1395 if (ok != 0) { 1396 ALOGE("%s pthread_once failed: %d", __func__, ok); 1397 } 1398 frameCount = mFrameCount * sFastTrackMultiplier; 1399 } 1400 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1401 frameCount, mFrameCount); 1402 } else { 1403 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1404 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1405 "sampleRate=%u mSampleRate=%u " 1406 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1407 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1408 audio_is_linear_pcm(format), 1409 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1410 *flags &= ~IAudioFlinger::TRACK_FAST; 1411 // For compatibility with AudioTrack calculation, buffer depth is forced 1412 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1413 // This is probably too conservative, but legacy application code may depend on it. 1414 // If you change this calculation, also review the start threshold which is related. 1415 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1416 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1417 if (minBufCount < 2) { 1418 minBufCount = 2; 1419 } 1420 size_t minFrameCount = mNormalFrameCount * minBufCount; 1421 if (frameCount < minFrameCount) { 1422 frameCount = minFrameCount; 1423 } 1424 } 1425 } 1426 *pFrameCount = frameCount; 1427 1428 switch (mType) { 1429 1430 case DIRECT: 1431 if (audio_is_linear_pcm(format)) { 1432 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1433 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1434 "for output %p with format %#x", 1435 sampleRate, format, channelMask, mOutput, mFormat); 1436 lStatus = BAD_VALUE; 1437 goto Exit; 1438 } 1439 } 1440 break; 1441 1442 case OFFLOAD: 1443 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1444 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1445 "for output %p with format %#x", 1446 sampleRate, format, channelMask, mOutput, mFormat); 1447 lStatus = BAD_VALUE; 1448 goto Exit; 1449 } 1450 break; 1451 1452 default: 1453 if (!audio_is_linear_pcm(format)) { 1454 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1455 "for output %p with format %#x", 1456 format, mOutput, mFormat); 1457 lStatus = BAD_VALUE; 1458 goto Exit; 1459 } 1460 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1461 if (sampleRate > mSampleRate*2) { 1462 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1463 lStatus = BAD_VALUE; 1464 goto Exit; 1465 } 1466 break; 1467 1468 } 1469 1470 lStatus = initCheck(); 1471 if (lStatus != NO_ERROR) { 1472 ALOGE("createTrack_l() audio driver not initialized"); 1473 goto Exit; 1474 } 1475 1476 { // scope for mLock 1477 Mutex::Autolock _l(mLock); 1478 1479 // all tracks in same audio session must share the same routing strategy otherwise 1480 // conflicts will happen when tracks are moved from one output to another by audio policy 1481 // manager 1482 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1483 for (size_t i = 0; i < mTracks.size(); ++i) { 1484 sp<Track> t = mTracks[i]; 1485 if (t != 0 && !t->isOutputTrack()) { 1486 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1487 if (sessionId == t->sessionId() && strategy != actual) { 1488 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1489 strategy, actual); 1490 lStatus = BAD_VALUE; 1491 goto Exit; 1492 } 1493 } 1494 } 1495 1496 if (!isTimed) { 1497 track = new Track(this, client, streamType, sampleRate, format, 1498 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1499 } else { 1500 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1501 channelMask, frameCount, sharedBuffer, sessionId, uid); 1502 } 1503 1504 // new Track always returns non-NULL, 1505 // but TimedTrack::create() is a factory that could fail by returning NULL 1506 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1507 if (lStatus != NO_ERROR) { 1508 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1509 // track must be cleared from the caller as the caller has the AF lock 1510 goto Exit; 1511 } 1512 mTracks.add(track); 1513 1514 sp<EffectChain> chain = getEffectChain_l(sessionId); 1515 if (chain != 0) { 1516 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1517 track->setMainBuffer(chain->inBuffer()); 1518 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1519 chain->incTrackCnt(); 1520 } 1521 1522 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1523 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1524 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1525 // so ask activity manager to do this on our behalf 1526 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1527 } 1528 } 1529 1530 lStatus = NO_ERROR; 1531 1532Exit: 1533 *status = lStatus; 1534 return track; 1535} 1536 1537uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1538{ 1539 return latency; 1540} 1541 1542uint32_t AudioFlinger::PlaybackThread::latency() const 1543{ 1544 Mutex::Autolock _l(mLock); 1545 return latency_l(); 1546} 1547uint32_t AudioFlinger::PlaybackThread::latency_l() const 1548{ 1549 if (initCheck() == NO_ERROR) { 1550 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1551 } else { 1552 return 0; 1553 } 1554} 1555 1556void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1557{ 1558 Mutex::Autolock _l(mLock); 1559 // Don't apply master volume in SW if our HAL can do it for us. 1560 if (mOutput && mOutput->audioHwDev && 1561 mOutput->audioHwDev->canSetMasterVolume()) { 1562 mMasterVolume = 1.0; 1563 } else { 1564 mMasterVolume = value; 1565 } 1566} 1567 1568void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1569{ 1570 Mutex::Autolock _l(mLock); 1571 // Don't apply master mute in SW if our HAL can do it for us. 1572 if (mOutput && mOutput->audioHwDev && 1573 mOutput->audioHwDev->canSetMasterMute()) { 1574 mMasterMute = false; 1575 } else { 1576 mMasterMute = muted; 1577 } 1578} 1579 1580void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1581{ 1582 Mutex::Autolock _l(mLock); 1583 mStreamTypes[stream].volume = value; 1584 broadcast_l(); 1585} 1586 1587void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1588{ 1589 Mutex::Autolock _l(mLock); 1590 mStreamTypes[stream].mute = muted; 1591 broadcast_l(); 1592} 1593 1594float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1595{ 1596 Mutex::Autolock _l(mLock); 1597 return mStreamTypes[stream].volume; 1598} 1599 1600// addTrack_l() must be called with ThreadBase::mLock held 1601status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1602{ 1603 status_t status = ALREADY_EXISTS; 1604 1605 // set retry count for buffer fill 1606 track->mRetryCount = kMaxTrackStartupRetries; 1607 if (mActiveTracks.indexOf(track) < 0) { 1608 // the track is newly added, make sure it fills up all its 1609 // buffers before playing. This is to ensure the client will 1610 // effectively get the latency it requested. 1611 if (!track->isOutputTrack()) { 1612 TrackBase::track_state state = track->mState; 1613 mLock.unlock(); 1614 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1615 mLock.lock(); 1616 // abort track was stopped/paused while we released the lock 1617 if (state != track->mState) { 1618 if (status == NO_ERROR) { 1619 mLock.unlock(); 1620 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1621 mLock.lock(); 1622 } 1623 return INVALID_OPERATION; 1624 } 1625 // abort if start is rejected by audio policy manager 1626 if (status != NO_ERROR) { 1627 return PERMISSION_DENIED; 1628 } 1629#ifdef ADD_BATTERY_DATA 1630 // to track the speaker usage 1631 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1632#endif 1633 } 1634 1635 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1636 track->mResetDone = false; 1637 track->mPresentationCompleteFrames = 0; 1638 mActiveTracks.add(track); 1639 mWakeLockUids.add(track->uid()); 1640 mActiveTracksGeneration++; 1641 mLatestActiveTrack = track; 1642 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1643 if (chain != 0) { 1644 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1645 track->sessionId()); 1646 chain->incActiveTrackCnt(); 1647 } 1648 1649 status = NO_ERROR; 1650 } 1651 1652 onAddNewTrack_l(); 1653 return status; 1654} 1655 1656bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1657{ 1658 track->terminate(); 1659 // active tracks are removed by threadLoop() 1660 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1661 track->mState = TrackBase::STOPPED; 1662 if (!trackActive) { 1663 removeTrack_l(track); 1664 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1665 track->mState = TrackBase::STOPPING_1; 1666 } 1667 1668 return trackActive; 1669} 1670 1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1672{ 1673 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1674 mTracks.remove(track); 1675 deleteTrackName_l(track->name()); 1676 // redundant as track is about to be destroyed, for dumpsys only 1677 track->mName = -1; 1678 if (track->isFastTrack()) { 1679 int index = track->mFastIndex; 1680 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1681 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1682 mFastTrackAvailMask |= 1 << index; 1683 // redundant as track is about to be destroyed, for dumpsys only 1684 track->mFastIndex = -1; 1685 } 1686 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1687 if (chain != 0) { 1688 chain->decTrackCnt(); 1689 } 1690} 1691 1692void AudioFlinger::PlaybackThread::broadcast_l() 1693{ 1694 // Thread could be blocked waiting for async 1695 // so signal it to handle state changes immediately 1696 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1697 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1698 mSignalPending = true; 1699 mWaitWorkCV.broadcast(); 1700} 1701 1702String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1703{ 1704 Mutex::Autolock _l(mLock); 1705 if (initCheck() != NO_ERROR) { 1706 return String8(); 1707 } 1708 1709 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1710 const String8 out_s8(s); 1711 free(s); 1712 return out_s8; 1713} 1714 1715void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1716 AudioSystem::OutputDescriptor desc; 1717 void *param2 = NULL; 1718 1719 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1720 param); 1721 1722 switch (event) { 1723 case AudioSystem::OUTPUT_OPENED: 1724 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1725 desc.channelMask = mChannelMask; 1726 desc.samplingRate = mSampleRate; 1727 desc.format = mFormat; 1728 desc.frameCount = mNormalFrameCount; // FIXME see 1729 // AudioFlinger::frameCount(audio_io_handle_t) 1730 desc.latency = latency_l(); 1731 param2 = &desc; 1732 break; 1733 1734 case AudioSystem::STREAM_CONFIG_CHANGED: 1735 param2 = ¶m; 1736 case AudioSystem::OUTPUT_CLOSED: 1737 default: 1738 break; 1739 } 1740 mAudioFlinger->audioConfigChanged(event, mId, param2); 1741} 1742 1743void AudioFlinger::PlaybackThread::writeCallback() 1744{ 1745 ALOG_ASSERT(mCallbackThread != 0); 1746 mCallbackThread->resetWriteBlocked(); 1747} 1748 1749void AudioFlinger::PlaybackThread::drainCallback() 1750{ 1751 ALOG_ASSERT(mCallbackThread != 0); 1752 mCallbackThread->resetDraining(); 1753} 1754 1755void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1756{ 1757 Mutex::Autolock _l(mLock); 1758 // reject out of sequence requests 1759 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1760 mWriteAckSequence &= ~1; 1761 mWaitWorkCV.signal(); 1762 } 1763} 1764 1765void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1766{ 1767 Mutex::Autolock _l(mLock); 1768 // reject out of sequence requests 1769 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1770 mDrainSequence &= ~1; 1771 mWaitWorkCV.signal(); 1772 } 1773} 1774 1775// static 1776int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1777 void *param __unused, 1778 void *cookie) 1779{ 1780 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1781 ALOGV("asyncCallback() event %d", event); 1782 switch (event) { 1783 case STREAM_CBK_EVENT_WRITE_READY: 1784 me->writeCallback(); 1785 break; 1786 case STREAM_CBK_EVENT_DRAIN_READY: 1787 me->drainCallback(); 1788 break; 1789 default: 1790 ALOGW("asyncCallback() unknown event %d", event); 1791 break; 1792 } 1793 return 0; 1794} 1795 1796void AudioFlinger::PlaybackThread::readOutputParameters_l() 1797{ 1798 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1799 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1800 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1801 if (!audio_is_output_channel(mChannelMask)) { 1802 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1803 } 1804 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1805 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1806 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1807 } 1808 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1809 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1810 mFormat = mHALFormat; 1811 if (!audio_is_valid_format(mFormat)) { 1812 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1813 } 1814 if ((mType == MIXER || mType == DUPLICATING) 1815 && !isValidPcmSinkFormat(mFormat)) { 1816 LOG_FATAL("HAL format %#x not supported for mixed output", 1817 mFormat); 1818 } 1819 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1820 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1821 mFrameCount = mBufferSize / mFrameSize; 1822 if (mFrameCount & 15) { 1823 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1824 mFrameCount); 1825 } 1826 1827 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1828 (mOutput->stream->set_callback != NULL)) { 1829 if (mOutput->stream->set_callback(mOutput->stream, 1830 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1831 mUseAsyncWrite = true; 1832 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1833 } 1834 } 1835 1836 // Calculate size of normal sink buffer relative to the HAL output buffer size 1837 double multiplier = 1.0; 1838 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1839 kUseFastMixer == FastMixer_Dynamic)) { 1840 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1841 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1842 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1843 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1844 maxNormalFrameCount = maxNormalFrameCount & ~15; 1845 if (maxNormalFrameCount < minNormalFrameCount) { 1846 maxNormalFrameCount = minNormalFrameCount; 1847 } 1848 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1849 if (multiplier <= 1.0) { 1850 multiplier = 1.0; 1851 } else if (multiplier <= 2.0) { 1852 if (2 * mFrameCount <= maxNormalFrameCount) { 1853 multiplier = 2.0; 1854 } else { 1855 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1856 } 1857 } else { 1858 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1859 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1860 // track, but we sometimes have to do this to satisfy the maximum frame count 1861 // constraint) 1862 // FIXME this rounding up should not be done if no HAL SRC 1863 uint32_t truncMult = (uint32_t) multiplier; 1864 if ((truncMult & 1)) { 1865 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1866 ++truncMult; 1867 } 1868 } 1869 multiplier = (double) truncMult; 1870 } 1871 } 1872 mNormalFrameCount = multiplier * mFrameCount; 1873 // round up to nearest 16 frames to satisfy AudioMixer 1874 if (mType == MIXER || mType == DUPLICATING) { 1875 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1876 } 1877 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1878 mNormalFrameCount); 1879 1880 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1881 // Originally this was int16_t[] array, need to remove legacy implications. 1882 free(mSinkBuffer); 1883 mSinkBuffer = NULL; 1884 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1885 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1886 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1887 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1888 1889 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1890 // drives the output. 1891 free(mMixerBuffer); 1892 mMixerBuffer = NULL; 1893 if (mMixerBufferEnabled) { 1894 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1895 mMixerBufferSize = mNormalFrameCount * mChannelCount 1896 * audio_bytes_per_sample(mMixerBufferFormat); 1897 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1898 } 1899 free(mEffectBuffer); 1900 mEffectBuffer = NULL; 1901 if (mEffectBufferEnabled) { 1902 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1903 mEffectBufferSize = mNormalFrameCount * mChannelCount 1904 * audio_bytes_per_sample(mEffectBufferFormat); 1905 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1906 } 1907 1908 // force reconfiguration of effect chains and engines to take new buffer size and audio 1909 // parameters into account 1910 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1911 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1912 // matter. 1913 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1914 Vector< sp<EffectChain> > effectChains = mEffectChains; 1915 for (size_t i = 0; i < effectChains.size(); i ++) { 1916 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1917 } 1918} 1919 1920 1921status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1922{ 1923 if (halFrames == NULL || dspFrames == NULL) { 1924 return BAD_VALUE; 1925 } 1926 Mutex::Autolock _l(mLock); 1927 if (initCheck() != NO_ERROR) { 1928 return INVALID_OPERATION; 1929 } 1930 size_t framesWritten = mBytesWritten / mFrameSize; 1931 *halFrames = framesWritten; 1932 1933 if (isSuspended()) { 1934 // return an estimation of rendered frames when the output is suspended 1935 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1936 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1937 return NO_ERROR; 1938 } else { 1939 status_t status; 1940 uint32_t frames; 1941 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1942 *dspFrames = (size_t)frames; 1943 return status; 1944 } 1945} 1946 1947uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1948{ 1949 Mutex::Autolock _l(mLock); 1950 uint32_t result = 0; 1951 if (getEffectChain_l(sessionId) != 0) { 1952 result = EFFECT_SESSION; 1953 } 1954 1955 for (size_t i = 0; i < mTracks.size(); ++i) { 1956 sp<Track> track = mTracks[i]; 1957 if (sessionId == track->sessionId() && !track->isInvalid()) { 1958 result |= TRACK_SESSION; 1959 break; 1960 } 1961 } 1962 1963 return result; 1964} 1965 1966uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1967{ 1968 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1969 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1970 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1971 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1972 } 1973 for (size_t i = 0; i < mTracks.size(); i++) { 1974 sp<Track> track = mTracks[i]; 1975 if (sessionId == track->sessionId() && !track->isInvalid()) { 1976 return AudioSystem::getStrategyForStream(track->streamType()); 1977 } 1978 } 1979 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1980} 1981 1982 1983AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1984{ 1985 Mutex::Autolock _l(mLock); 1986 return mOutput; 1987} 1988 1989AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1990{ 1991 Mutex::Autolock _l(mLock); 1992 AudioStreamOut *output = mOutput; 1993 mOutput = NULL; 1994 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1995 // must push a NULL and wait for ack 1996 mOutputSink.clear(); 1997 mPipeSink.clear(); 1998 mNormalSink.clear(); 1999 return output; 2000} 2001 2002// this method must always be called either with ThreadBase mLock held or inside the thread loop 2003audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2004{ 2005 if (mOutput == NULL) { 2006 return NULL; 2007 } 2008 return &mOutput->stream->common; 2009} 2010 2011uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2012{ 2013 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2014} 2015 2016status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2017{ 2018 if (!isValidSyncEvent(event)) { 2019 return BAD_VALUE; 2020 } 2021 2022 Mutex::Autolock _l(mLock); 2023 2024 for (size_t i = 0; i < mTracks.size(); ++i) { 2025 sp<Track> track = mTracks[i]; 2026 if (event->triggerSession() == track->sessionId()) { 2027 (void) track->setSyncEvent(event); 2028 return NO_ERROR; 2029 } 2030 } 2031 2032 return NAME_NOT_FOUND; 2033} 2034 2035bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2036{ 2037 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2038} 2039 2040void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2041 const Vector< sp<Track> >& tracksToRemove) 2042{ 2043 size_t count = tracksToRemove.size(); 2044 if (count > 0) { 2045 for (size_t i = 0 ; i < count ; i++) { 2046 const sp<Track>& track = tracksToRemove.itemAt(i); 2047 if (!track->isOutputTrack()) { 2048 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2049#ifdef ADD_BATTERY_DATA 2050 // to track the speaker usage 2051 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2052#endif 2053 if (track->isTerminated()) { 2054 AudioSystem::releaseOutput(mId); 2055 } 2056 } 2057 } 2058 } 2059} 2060 2061void AudioFlinger::PlaybackThread::checkSilentMode_l() 2062{ 2063 if (!mMasterMute) { 2064 char value[PROPERTY_VALUE_MAX]; 2065 if (property_get("ro.audio.silent", value, "0") > 0) { 2066 char *endptr; 2067 unsigned long ul = strtoul(value, &endptr, 0); 2068 if (*endptr == '\0' && ul != 0) { 2069 ALOGD("Silence is golden"); 2070 // The setprop command will not allow a property to be changed after 2071 // the first time it is set, so we don't have to worry about un-muting. 2072 setMasterMute_l(true); 2073 } 2074 } 2075 } 2076} 2077 2078// shared by MIXER and DIRECT, overridden by DUPLICATING 2079ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2080{ 2081 // FIXME rewrite to reduce number of system calls 2082 mLastWriteTime = systemTime(); 2083 mInWrite = true; 2084 ssize_t bytesWritten; 2085 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2086 2087 // If an NBAIO sink is present, use it to write the normal mixer's submix 2088 if (mNormalSink != 0) { 2089 const size_t count = mBytesRemaining / mFrameSize; 2090 2091 ATRACE_BEGIN("write"); 2092 // update the setpoint when AudioFlinger::mScreenState changes 2093 uint32_t screenState = AudioFlinger::mScreenState; 2094 if (screenState != mScreenState) { 2095 mScreenState = screenState; 2096 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2097 if (pipe != NULL) { 2098 pipe->setAvgFrames((mScreenState & 1) ? 2099 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2100 } 2101 } 2102 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2103 ATRACE_END(); 2104 if (framesWritten > 0) { 2105 bytesWritten = framesWritten * mFrameSize; 2106 } else { 2107 bytesWritten = framesWritten; 2108 } 2109 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2110 if (status == NO_ERROR) { 2111 size_t totalFramesWritten = mNormalSink->framesWritten(); 2112 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2113 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2114 mLatchDValid = true; 2115 } 2116 } 2117 // otherwise use the HAL / AudioStreamOut directly 2118 } else { 2119 // Direct output and offload threads 2120 2121 if (mUseAsyncWrite) { 2122 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2123 mWriteAckSequence += 2; 2124 mWriteAckSequence |= 1; 2125 ALOG_ASSERT(mCallbackThread != 0); 2126 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2127 } 2128 // FIXME We should have an implementation of timestamps for direct output threads. 2129 // They are used e.g for multichannel PCM playback over HDMI. 2130 bytesWritten = mOutput->stream->write(mOutput->stream, 2131 (char *)mSinkBuffer + offset, mBytesRemaining); 2132 if (mUseAsyncWrite && 2133 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2134 // do not wait for async callback in case of error of full write 2135 mWriteAckSequence &= ~1; 2136 ALOG_ASSERT(mCallbackThread != 0); 2137 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2138 } 2139 } 2140 2141 mNumWrites++; 2142 mInWrite = false; 2143 mStandby = false; 2144 return bytesWritten; 2145} 2146 2147void AudioFlinger::PlaybackThread::threadLoop_drain() 2148{ 2149 if (mOutput->stream->drain) { 2150 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2151 if (mUseAsyncWrite) { 2152 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2153 mDrainSequence |= 1; 2154 ALOG_ASSERT(mCallbackThread != 0); 2155 mCallbackThread->setDraining(mDrainSequence); 2156 } 2157 mOutput->stream->drain(mOutput->stream, 2158 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2159 : AUDIO_DRAIN_ALL); 2160 } 2161} 2162 2163void AudioFlinger::PlaybackThread::threadLoop_exit() 2164{ 2165 // Default implementation has nothing to do 2166} 2167 2168/* 2169The derived values that are cached: 2170 - mSinkBufferSize from frame count * frame size 2171 - activeSleepTime from activeSleepTimeUs() 2172 - idleSleepTime from idleSleepTimeUs() 2173 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2174 - maxPeriod from frame count and sample rate (MIXER only) 2175 2176The parameters that affect these derived values are: 2177 - frame count 2178 - frame size 2179 - sample rate 2180 - device type: A2DP or not 2181 - device latency 2182 - format: PCM or not 2183 - active sleep time 2184 - idle sleep time 2185*/ 2186 2187void AudioFlinger::PlaybackThread::cacheParameters_l() 2188{ 2189 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2190 activeSleepTime = activeSleepTimeUs(); 2191 idleSleepTime = idleSleepTimeUs(); 2192} 2193 2194void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2195{ 2196 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2197 this, streamType, mTracks.size()); 2198 Mutex::Autolock _l(mLock); 2199 2200 size_t size = mTracks.size(); 2201 for (size_t i = 0; i < size; i++) { 2202 sp<Track> t = mTracks[i]; 2203 if (t->streamType() == streamType) { 2204 t->invalidate(); 2205 } 2206 } 2207} 2208 2209status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2210{ 2211 int session = chain->sessionId(); 2212 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2213 ? mEffectBuffer : mSinkBuffer); 2214 bool ownsBuffer = false; 2215 2216 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2217 if (session > 0) { 2218 // Only one effect chain can be present in direct output thread and it uses 2219 // the sink buffer as input 2220 if (mType != DIRECT) { 2221 size_t numSamples = mNormalFrameCount * mChannelCount; 2222 buffer = new int16_t[numSamples]; 2223 memset(buffer, 0, numSamples * sizeof(int16_t)); 2224 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2225 ownsBuffer = true; 2226 } 2227 2228 // Attach all tracks with same session ID to this chain. 2229 for (size_t i = 0; i < mTracks.size(); ++i) { 2230 sp<Track> track = mTracks[i]; 2231 if (session == track->sessionId()) { 2232 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2233 buffer); 2234 track->setMainBuffer(buffer); 2235 chain->incTrackCnt(); 2236 } 2237 } 2238 2239 // indicate all active tracks in the chain 2240 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2241 sp<Track> track = mActiveTracks[i].promote(); 2242 if (track == 0) { 2243 continue; 2244 } 2245 if (session == track->sessionId()) { 2246 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2247 chain->incActiveTrackCnt(); 2248 } 2249 } 2250 } 2251 2252 chain->setInBuffer(buffer, ownsBuffer); 2253 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2254 ? mEffectBuffer : mSinkBuffer)); 2255 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2256 // chains list in order to be processed last as it contains output stage effects 2257 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2258 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2259 // after track specific effects and before output stage 2260 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2261 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2262 // Effect chain for other sessions are inserted at beginning of effect 2263 // chains list to be processed before output mix effects. Relative order between other 2264 // sessions is not important 2265 size_t size = mEffectChains.size(); 2266 size_t i = 0; 2267 for (i = 0; i < size; i++) { 2268 if (mEffectChains[i]->sessionId() < session) { 2269 break; 2270 } 2271 } 2272 mEffectChains.insertAt(chain, i); 2273 checkSuspendOnAddEffectChain_l(chain); 2274 2275 return NO_ERROR; 2276} 2277 2278size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2279{ 2280 int session = chain->sessionId(); 2281 2282 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2283 2284 for (size_t i = 0; i < mEffectChains.size(); i++) { 2285 if (chain == mEffectChains[i]) { 2286 mEffectChains.removeAt(i); 2287 // detach all active tracks from the chain 2288 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2289 sp<Track> track = mActiveTracks[i].promote(); 2290 if (track == 0) { 2291 continue; 2292 } 2293 if (session == track->sessionId()) { 2294 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2295 chain.get(), session); 2296 chain->decActiveTrackCnt(); 2297 } 2298 } 2299 2300 // detach all tracks with same session ID from this chain 2301 for (size_t i = 0; i < mTracks.size(); ++i) { 2302 sp<Track> track = mTracks[i]; 2303 if (session == track->sessionId()) { 2304 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2305 chain->decTrackCnt(); 2306 } 2307 } 2308 break; 2309 } 2310 } 2311 return mEffectChains.size(); 2312} 2313 2314status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2315 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2316{ 2317 Mutex::Autolock _l(mLock); 2318 return attachAuxEffect_l(track, EffectId); 2319} 2320 2321status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2322 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2323{ 2324 status_t status = NO_ERROR; 2325 2326 if (EffectId == 0) { 2327 track->setAuxBuffer(0, NULL); 2328 } else { 2329 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2330 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2331 if (effect != 0) { 2332 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2333 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2334 } else { 2335 status = INVALID_OPERATION; 2336 } 2337 } else { 2338 status = BAD_VALUE; 2339 } 2340 } 2341 return status; 2342} 2343 2344void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2345{ 2346 for (size_t i = 0; i < mTracks.size(); ++i) { 2347 sp<Track> track = mTracks[i]; 2348 if (track->auxEffectId() == effectId) { 2349 attachAuxEffect_l(track, 0); 2350 } 2351 } 2352} 2353 2354bool AudioFlinger::PlaybackThread::threadLoop() 2355{ 2356 Vector< sp<Track> > tracksToRemove; 2357 2358 standbyTime = systemTime(); 2359 2360 // MIXER 2361 nsecs_t lastWarning = 0; 2362 2363 // DUPLICATING 2364 // FIXME could this be made local to while loop? 2365 writeFrames = 0; 2366 2367 int lastGeneration = 0; 2368 2369 cacheParameters_l(); 2370 sleepTime = idleSleepTime; 2371 2372 if (mType == MIXER) { 2373 sleepTimeShift = 0; 2374 } 2375 2376 CpuStats cpuStats; 2377 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2378 2379 acquireWakeLock(); 2380 2381 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2382 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2383 // and then that string will be logged at the next convenient opportunity. 2384 const char *logString = NULL; 2385 2386 checkSilentMode_l(); 2387 2388 while (!exitPending()) 2389 { 2390 cpuStats.sample(myName); 2391 2392 Vector< sp<EffectChain> > effectChains; 2393 2394 { // scope for mLock 2395 2396 Mutex::Autolock _l(mLock); 2397 2398 processConfigEvents_l(); 2399 2400 if (logString != NULL) { 2401 mNBLogWriter->logTimestamp(); 2402 mNBLogWriter->log(logString); 2403 logString = NULL; 2404 } 2405 2406 if (mLatchDValid) { 2407 mLatchQ = mLatchD; 2408 mLatchDValid = false; 2409 mLatchQValid = true; 2410 } 2411 2412 saveOutputTracks(); 2413 if (mSignalPending) { 2414 // A signal was raised while we were unlocked 2415 mSignalPending = false; 2416 } else if (waitingAsyncCallback_l()) { 2417 if (exitPending()) { 2418 break; 2419 } 2420 releaseWakeLock_l(); 2421 mWakeLockUids.clear(); 2422 mActiveTracksGeneration++; 2423 ALOGV("wait async completion"); 2424 mWaitWorkCV.wait(mLock); 2425 ALOGV("async completion/wake"); 2426 acquireWakeLock_l(); 2427 standbyTime = systemTime() + standbyDelay; 2428 sleepTime = 0; 2429 2430 continue; 2431 } 2432 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2433 isSuspended()) { 2434 // put audio hardware into standby after short delay 2435 if (shouldStandby_l()) { 2436 2437 threadLoop_standby(); 2438 2439 mStandby = true; 2440 } 2441 2442 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2443 // we're about to wait, flush the binder command buffer 2444 IPCThreadState::self()->flushCommands(); 2445 2446 clearOutputTracks(); 2447 2448 if (exitPending()) { 2449 break; 2450 } 2451 2452 releaseWakeLock_l(); 2453 mWakeLockUids.clear(); 2454 mActiveTracksGeneration++; 2455 // wait until we have something to do... 2456 ALOGV("%s going to sleep", myName.string()); 2457 mWaitWorkCV.wait(mLock); 2458 ALOGV("%s waking up", myName.string()); 2459 acquireWakeLock_l(); 2460 2461 mMixerStatus = MIXER_IDLE; 2462 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2463 mBytesWritten = 0; 2464 mBytesRemaining = 0; 2465 checkSilentMode_l(); 2466 2467 standbyTime = systemTime() + standbyDelay; 2468 sleepTime = idleSleepTime; 2469 if (mType == MIXER) { 2470 sleepTimeShift = 0; 2471 } 2472 2473 continue; 2474 } 2475 } 2476 // mMixerStatusIgnoringFastTracks is also updated internally 2477 mMixerStatus = prepareTracks_l(&tracksToRemove); 2478 2479 // compare with previously applied list 2480 if (lastGeneration != mActiveTracksGeneration) { 2481 // update wakelock 2482 updateWakeLockUids_l(mWakeLockUids); 2483 lastGeneration = mActiveTracksGeneration; 2484 } 2485 2486 // prevent any changes in effect chain list and in each effect chain 2487 // during mixing and effect process as the audio buffers could be deleted 2488 // or modified if an effect is created or deleted 2489 lockEffectChains_l(effectChains); 2490 } // mLock scope ends 2491 2492 if (mBytesRemaining == 0) { 2493 mCurrentWriteLength = 0; 2494 if (mMixerStatus == MIXER_TRACKS_READY) { 2495 // threadLoop_mix() sets mCurrentWriteLength 2496 threadLoop_mix(); 2497 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2498 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2499 // threadLoop_sleepTime sets sleepTime to 0 if data 2500 // must be written to HAL 2501 threadLoop_sleepTime(); 2502 if (sleepTime == 0) { 2503 mCurrentWriteLength = mSinkBufferSize; 2504 } 2505 } 2506 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2507 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2508 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2509 // or mSinkBuffer (if there are no effects). 2510 // 2511 // This is done pre-effects computation; if effects change to 2512 // support higher precision, this needs to move. 2513 // 2514 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2515 // TODO use sleepTime == 0 as an additional condition. 2516 if (mMixerBufferValid) { 2517 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2518 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2519 2520 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2521 mNormalFrameCount * mChannelCount); 2522 } 2523 2524 mBytesRemaining = mCurrentWriteLength; 2525 if (isSuspended()) { 2526 sleepTime = suspendSleepTimeUs(); 2527 // simulate write to HAL when suspended 2528 mBytesWritten += mSinkBufferSize; 2529 mBytesRemaining = 0; 2530 } 2531 2532 // only process effects if we're going to write 2533 if (sleepTime == 0 && mType != OFFLOAD) { 2534 for (size_t i = 0; i < effectChains.size(); i ++) { 2535 effectChains[i]->process_l(); 2536 } 2537 } 2538 } 2539 // Process effect chains for offloaded thread even if no audio 2540 // was read from audio track: process only updates effect state 2541 // and thus does have to be synchronized with audio writes but may have 2542 // to be called while waiting for async write callback 2543 if (mType == OFFLOAD) { 2544 for (size_t i = 0; i < effectChains.size(); i ++) { 2545 effectChains[i]->process_l(); 2546 } 2547 } 2548 2549 // Only if the Effects buffer is enabled and there is data in the 2550 // Effects buffer (buffer valid), we need to 2551 // copy into the sink buffer. 2552 // TODO use sleepTime == 0 as an additional condition. 2553 if (mEffectBufferValid) { 2554 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2555 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2556 mNormalFrameCount * mChannelCount); 2557 } 2558 2559 // enable changes in effect chain 2560 unlockEffectChains(effectChains); 2561 2562 if (!waitingAsyncCallback()) { 2563 // sleepTime == 0 means we must write to audio hardware 2564 if (sleepTime == 0) { 2565 if (mBytesRemaining) { 2566 ssize_t ret = threadLoop_write(); 2567 if (ret < 0) { 2568 mBytesRemaining = 0; 2569 } else { 2570 mBytesWritten += ret; 2571 mBytesRemaining -= ret; 2572 } 2573 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2574 (mMixerStatus == MIXER_DRAIN_ALL)) { 2575 threadLoop_drain(); 2576 } 2577 if (mType == MIXER) { 2578 // write blocked detection 2579 nsecs_t now = systemTime(); 2580 nsecs_t delta = now - mLastWriteTime; 2581 if (!mStandby && delta > maxPeriod) { 2582 mNumDelayedWrites++; 2583 if ((now - lastWarning) > kWarningThrottleNs) { 2584 ATRACE_NAME("underrun"); 2585 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2586 ns2ms(delta), mNumDelayedWrites, this); 2587 lastWarning = now; 2588 } 2589 } 2590 } 2591 2592 } else { 2593 usleep(sleepTime); 2594 } 2595 } 2596 2597 // Finally let go of removed track(s), without the lock held 2598 // since we can't guarantee the destructors won't acquire that 2599 // same lock. This will also mutate and push a new fast mixer state. 2600 threadLoop_removeTracks(tracksToRemove); 2601 tracksToRemove.clear(); 2602 2603 // FIXME I don't understand the need for this here; 2604 // it was in the original code but maybe the 2605 // assignment in saveOutputTracks() makes this unnecessary? 2606 clearOutputTracks(); 2607 2608 // Effect chains will be actually deleted here if they were removed from 2609 // mEffectChains list during mixing or effects processing 2610 effectChains.clear(); 2611 2612 // FIXME Note that the above .clear() is no longer necessary since effectChains 2613 // is now local to this block, but will keep it for now (at least until merge done). 2614 } 2615 2616 threadLoop_exit(); 2617 2618 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2619 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2620 // put output stream into standby mode 2621 if (!mStandby) { 2622 mOutput->stream->common.standby(&mOutput->stream->common); 2623 } 2624 } 2625 2626 releaseWakeLock(); 2627 mWakeLockUids.clear(); 2628 mActiveTracksGeneration++; 2629 2630 ALOGV("Thread %p type %d exiting", this, mType); 2631 return false; 2632} 2633 2634// removeTracks_l() must be called with ThreadBase::mLock held 2635void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2636{ 2637 size_t count = tracksToRemove.size(); 2638 if (count > 0) { 2639 for (size_t i=0 ; i<count ; i++) { 2640 const sp<Track>& track = tracksToRemove.itemAt(i); 2641 mActiveTracks.remove(track); 2642 mWakeLockUids.remove(track->uid()); 2643 mActiveTracksGeneration++; 2644 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2645 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2646 if (chain != 0) { 2647 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2648 track->sessionId()); 2649 chain->decActiveTrackCnt(); 2650 } 2651 if (track->isTerminated()) { 2652 removeTrack_l(track); 2653 } 2654 } 2655 } 2656 2657} 2658 2659status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2660{ 2661 if (mNormalSink != 0) { 2662 return mNormalSink->getTimestamp(timestamp); 2663 } 2664 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2665 uint64_t position64; 2666 int ret = mOutput->stream->get_presentation_position( 2667 mOutput->stream, &position64, ×tamp.mTime); 2668 if (ret == 0) { 2669 timestamp.mPosition = (uint32_t)position64; 2670 return NO_ERROR; 2671 } 2672 } 2673 return INVALID_OPERATION; 2674} 2675 2676status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2677 audio_patch_handle_t *handle) 2678{ 2679 status_t status = NO_ERROR; 2680 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2681 // store new device and send to effects 2682 audio_devices_t type = AUDIO_DEVICE_NONE; 2683 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2684 type |= patch->sinks[i].ext.device.type; 2685 } 2686 mOutDevice = type; 2687 for (size_t i = 0; i < mEffectChains.size(); i++) { 2688 mEffectChains[i]->setDevice_l(mOutDevice); 2689 } 2690 2691 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2692 status = hwDevice->create_audio_patch(hwDevice, 2693 patch->num_sources, 2694 patch->sources, 2695 patch->num_sinks, 2696 patch->sinks, 2697 handle); 2698 } else { 2699 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2700 } 2701 return status; 2702} 2703 2704status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2705{ 2706 status_t status = NO_ERROR; 2707 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2708 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2709 status = hwDevice->release_audio_patch(hwDevice, handle); 2710 } else { 2711 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2712 } 2713 return status; 2714} 2715 2716// ---------------------------------------------------------------------------- 2717 2718AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2719 audio_io_handle_t id, audio_devices_t device, type_t type) 2720 : PlaybackThread(audioFlinger, output, id, device, type), 2721 // mAudioMixer below 2722 // mFastMixer below 2723 mFastMixerFutex(0) 2724 // mOutputSink below 2725 // mPipeSink below 2726 // mNormalSink below 2727{ 2728 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2729 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2730 "mFrameCount=%d, mNormalFrameCount=%d", 2731 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2732 mNormalFrameCount); 2733 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2734 2735 // FIXME - Current mixer implementation only supports stereo output 2736 if (mChannelCount != FCC_2) { 2737 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2738 } 2739 2740 // create an NBAIO sink for the HAL output stream, and negotiate 2741 mOutputSink = new AudioStreamOutSink(output->stream); 2742 size_t numCounterOffers = 0; 2743 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2744 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2745 ALOG_ASSERT(index == 0); 2746 2747 // initialize fast mixer depending on configuration 2748 bool initFastMixer; 2749 switch (kUseFastMixer) { 2750 case FastMixer_Never: 2751 initFastMixer = false; 2752 break; 2753 case FastMixer_Always: 2754 initFastMixer = true; 2755 break; 2756 case FastMixer_Static: 2757 case FastMixer_Dynamic: 2758 initFastMixer = mFrameCount < mNormalFrameCount; 2759 break; 2760 } 2761 if (initFastMixer) { 2762 audio_format_t fastMixerFormat; 2763 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2764 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2765 } else { 2766 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2767 } 2768 if (mFormat != fastMixerFormat) { 2769 // change our Sink format to accept our intermediate precision 2770 mFormat = fastMixerFormat; 2771 free(mSinkBuffer); 2772 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2773 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2774 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2775 } 2776 2777 // create a MonoPipe to connect our submix to FastMixer 2778 NBAIO_Format format = mOutputSink->format(); 2779 // adjust format to match that of the Fast Mixer 2780 format.mFormat = fastMixerFormat; 2781 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2782 2783 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2784 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2785 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2786 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2787 const NBAIO_Format offers[1] = {format}; 2788 size_t numCounterOffers = 0; 2789 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2790 ALOG_ASSERT(index == 0); 2791 monoPipe->setAvgFrames((mScreenState & 1) ? 2792 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2793 mPipeSink = monoPipe; 2794 2795#ifdef TEE_SINK 2796 if (mTeeSinkOutputEnabled) { 2797 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2798 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2799 numCounterOffers = 0; 2800 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2801 ALOG_ASSERT(index == 0); 2802 mTeeSink = teeSink; 2803 PipeReader *teeSource = new PipeReader(*teeSink); 2804 numCounterOffers = 0; 2805 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2806 ALOG_ASSERT(index == 0); 2807 mTeeSource = teeSource; 2808 } 2809#endif 2810 2811 // create fast mixer and configure it initially with just one fast track for our submix 2812 mFastMixer = new FastMixer(); 2813 FastMixerStateQueue *sq = mFastMixer->sq(); 2814#ifdef STATE_QUEUE_DUMP 2815 sq->setObserverDump(&mStateQueueObserverDump); 2816 sq->setMutatorDump(&mStateQueueMutatorDump); 2817#endif 2818 FastMixerState *state = sq->begin(); 2819 FastTrack *fastTrack = &state->mFastTracks[0]; 2820 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2821 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2822 fastTrack->mVolumeProvider = NULL; 2823 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2824 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2825 fastTrack->mGeneration++; 2826 state->mFastTracksGen++; 2827 state->mTrackMask = 1; 2828 // fast mixer will use the HAL output sink 2829 state->mOutputSink = mOutputSink.get(); 2830 state->mOutputSinkGen++; 2831 state->mFrameCount = mFrameCount; 2832 state->mCommand = FastMixerState::COLD_IDLE; 2833 // already done in constructor initialization list 2834 //mFastMixerFutex = 0; 2835 state->mColdFutexAddr = &mFastMixerFutex; 2836 state->mColdGen++; 2837 state->mDumpState = &mFastMixerDumpState; 2838#ifdef TEE_SINK 2839 state->mTeeSink = mTeeSink.get(); 2840#endif 2841 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2842 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2843 sq->end(); 2844 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2845 2846 // start the fast mixer 2847 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2848 pid_t tid = mFastMixer->getTid(); 2849 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2850 if (err != 0) { 2851 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2852 kPriorityFastMixer, getpid_cached, tid, err); 2853 } 2854 2855#ifdef AUDIO_WATCHDOG 2856 // create and start the watchdog 2857 mAudioWatchdog = new AudioWatchdog(); 2858 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2859 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2860 tid = mAudioWatchdog->getTid(); 2861 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2862 if (err != 0) { 2863 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2864 kPriorityFastMixer, getpid_cached, tid, err); 2865 } 2866#endif 2867 2868 } 2869 2870 switch (kUseFastMixer) { 2871 case FastMixer_Never: 2872 case FastMixer_Dynamic: 2873 mNormalSink = mOutputSink; 2874 break; 2875 case FastMixer_Always: 2876 mNormalSink = mPipeSink; 2877 break; 2878 case FastMixer_Static: 2879 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2880 break; 2881 } 2882} 2883 2884AudioFlinger::MixerThread::~MixerThread() 2885{ 2886 if (mFastMixer != 0) { 2887 FastMixerStateQueue *sq = mFastMixer->sq(); 2888 FastMixerState *state = sq->begin(); 2889 if (state->mCommand == FastMixerState::COLD_IDLE) { 2890 int32_t old = android_atomic_inc(&mFastMixerFutex); 2891 if (old == -1) { 2892 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2893 } 2894 } 2895 state->mCommand = FastMixerState::EXIT; 2896 sq->end(); 2897 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2898 mFastMixer->join(); 2899 // Though the fast mixer thread has exited, it's state queue is still valid. 2900 // We'll use that extract the final state which contains one remaining fast track 2901 // corresponding to our sub-mix. 2902 state = sq->begin(); 2903 ALOG_ASSERT(state->mTrackMask == 1); 2904 FastTrack *fastTrack = &state->mFastTracks[0]; 2905 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2906 delete fastTrack->mBufferProvider; 2907 sq->end(false /*didModify*/); 2908 mFastMixer.clear(); 2909#ifdef AUDIO_WATCHDOG 2910 if (mAudioWatchdog != 0) { 2911 mAudioWatchdog->requestExit(); 2912 mAudioWatchdog->requestExitAndWait(); 2913 mAudioWatchdog.clear(); 2914 } 2915#endif 2916 } 2917 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2918 delete mAudioMixer; 2919} 2920 2921 2922uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2923{ 2924 if (mFastMixer != 0) { 2925 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2926 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2927 } 2928 return latency; 2929} 2930 2931 2932void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2933{ 2934 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2935} 2936 2937ssize_t AudioFlinger::MixerThread::threadLoop_write() 2938{ 2939 // FIXME we should only do one push per cycle; confirm this is true 2940 // Start the fast mixer if it's not already running 2941 if (mFastMixer != 0) { 2942 FastMixerStateQueue *sq = mFastMixer->sq(); 2943 FastMixerState *state = sq->begin(); 2944 if (state->mCommand != FastMixerState::MIX_WRITE && 2945 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2946 if (state->mCommand == FastMixerState::COLD_IDLE) { 2947 int32_t old = android_atomic_inc(&mFastMixerFutex); 2948 if (old == -1) { 2949 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2950 } 2951#ifdef AUDIO_WATCHDOG 2952 if (mAudioWatchdog != 0) { 2953 mAudioWatchdog->resume(); 2954 } 2955#endif 2956 } 2957 state->mCommand = FastMixerState::MIX_WRITE; 2958 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2959 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2960 sq->end(); 2961 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2962 if (kUseFastMixer == FastMixer_Dynamic) { 2963 mNormalSink = mPipeSink; 2964 } 2965 } else { 2966 sq->end(false /*didModify*/); 2967 } 2968 } 2969 return PlaybackThread::threadLoop_write(); 2970} 2971 2972void AudioFlinger::MixerThread::threadLoop_standby() 2973{ 2974 // Idle the fast mixer if it's currently running 2975 if (mFastMixer != 0) { 2976 FastMixerStateQueue *sq = mFastMixer->sq(); 2977 FastMixerState *state = sq->begin(); 2978 if (!(state->mCommand & FastMixerState::IDLE)) { 2979 state->mCommand = FastMixerState::COLD_IDLE; 2980 state->mColdFutexAddr = &mFastMixerFutex; 2981 state->mColdGen++; 2982 mFastMixerFutex = 0; 2983 sq->end(); 2984 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2985 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2986 if (kUseFastMixer == FastMixer_Dynamic) { 2987 mNormalSink = mOutputSink; 2988 } 2989#ifdef AUDIO_WATCHDOG 2990 if (mAudioWatchdog != 0) { 2991 mAudioWatchdog->pause(); 2992 } 2993#endif 2994 } else { 2995 sq->end(false /*didModify*/); 2996 } 2997 } 2998 PlaybackThread::threadLoop_standby(); 2999} 3000 3001bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3002{ 3003 return false; 3004} 3005 3006bool AudioFlinger::PlaybackThread::shouldStandby_l() 3007{ 3008 return !mStandby; 3009} 3010 3011bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3012{ 3013 Mutex::Autolock _l(mLock); 3014 return waitingAsyncCallback_l(); 3015} 3016 3017// shared by MIXER and DIRECT, overridden by DUPLICATING 3018void AudioFlinger::PlaybackThread::threadLoop_standby() 3019{ 3020 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3021 mOutput->stream->common.standby(&mOutput->stream->common); 3022 if (mUseAsyncWrite != 0) { 3023 // discard any pending drain or write ack by incrementing sequence 3024 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3025 mDrainSequence = (mDrainSequence + 2) & ~1; 3026 ALOG_ASSERT(mCallbackThread != 0); 3027 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3028 mCallbackThread->setDraining(mDrainSequence); 3029 } 3030} 3031 3032void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3033{ 3034 ALOGV("signal playback thread"); 3035 broadcast_l(); 3036} 3037 3038void AudioFlinger::MixerThread::threadLoop_mix() 3039{ 3040 // obtain the presentation timestamp of the next output buffer 3041 int64_t pts; 3042 status_t status = INVALID_OPERATION; 3043 3044 if (mNormalSink != 0) { 3045 status = mNormalSink->getNextWriteTimestamp(&pts); 3046 } else { 3047 status = mOutputSink->getNextWriteTimestamp(&pts); 3048 } 3049 3050 if (status != NO_ERROR) { 3051 pts = AudioBufferProvider::kInvalidPTS; 3052 } 3053 3054 // mix buffers... 3055 mAudioMixer->process(pts); 3056 mCurrentWriteLength = mSinkBufferSize; 3057 // increase sleep time progressively when application underrun condition clears. 3058 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3059 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3060 // such that we would underrun the audio HAL. 3061 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3062 sleepTimeShift--; 3063 } 3064 sleepTime = 0; 3065 standbyTime = systemTime() + standbyDelay; 3066 //TODO: delay standby when effects have a tail 3067} 3068 3069void AudioFlinger::MixerThread::threadLoop_sleepTime() 3070{ 3071 // If no tracks are ready, sleep once for the duration of an output 3072 // buffer size, then write 0s to the output 3073 if (sleepTime == 0) { 3074 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3075 sleepTime = activeSleepTime >> sleepTimeShift; 3076 if (sleepTime < kMinThreadSleepTimeUs) { 3077 sleepTime = kMinThreadSleepTimeUs; 3078 } 3079 // reduce sleep time in case of consecutive application underruns to avoid 3080 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3081 // duration we would end up writing less data than needed by the audio HAL if 3082 // the condition persists. 3083 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3084 sleepTimeShift++; 3085 } 3086 } else { 3087 sleepTime = idleSleepTime; 3088 } 3089 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3090 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3091 // before effects processing or output. 3092 if (mMixerBufferValid) { 3093 memset(mMixerBuffer, 0, mMixerBufferSize); 3094 } else { 3095 memset(mSinkBuffer, 0, mSinkBufferSize); 3096 } 3097 sleepTime = 0; 3098 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3099 "anticipated start"); 3100 } 3101 // TODO add standby time extension fct of effect tail 3102} 3103 3104// prepareTracks_l() must be called with ThreadBase::mLock held 3105AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3106 Vector< sp<Track> > *tracksToRemove) 3107{ 3108 3109 mixer_state mixerStatus = MIXER_IDLE; 3110 // find out which tracks need to be processed 3111 size_t count = mActiveTracks.size(); 3112 size_t mixedTracks = 0; 3113 size_t tracksWithEffect = 0; 3114 // counts only _active_ fast tracks 3115 size_t fastTracks = 0; 3116 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3117 3118 float masterVolume = mMasterVolume; 3119 bool masterMute = mMasterMute; 3120 3121 if (masterMute) { 3122 masterVolume = 0; 3123 } 3124 // Delegate master volume control to effect in output mix effect chain if needed 3125 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3126 if (chain != 0) { 3127 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3128 chain->setVolume_l(&v, &v); 3129 masterVolume = (float)((v + (1 << 23)) >> 24); 3130 chain.clear(); 3131 } 3132 3133 // prepare a new state to push 3134 FastMixerStateQueue *sq = NULL; 3135 FastMixerState *state = NULL; 3136 bool didModify = false; 3137 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3138 if (mFastMixer != 0) { 3139 sq = mFastMixer->sq(); 3140 state = sq->begin(); 3141 } 3142 3143 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3144 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3145 3146 for (size_t i=0 ; i<count ; i++) { 3147 const sp<Track> t = mActiveTracks[i].promote(); 3148 if (t == 0) { 3149 continue; 3150 } 3151 3152 // this const just means the local variable doesn't change 3153 Track* const track = t.get(); 3154 3155 // process fast tracks 3156 if (track->isFastTrack()) { 3157 3158 // It's theoretically possible (though unlikely) for a fast track to be created 3159 // and then removed within the same normal mix cycle. This is not a problem, as 3160 // the track never becomes active so it's fast mixer slot is never touched. 3161 // The converse, of removing an (active) track and then creating a new track 3162 // at the identical fast mixer slot within the same normal mix cycle, 3163 // is impossible because the slot isn't marked available until the end of each cycle. 3164 int j = track->mFastIndex; 3165 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3166 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3167 FastTrack *fastTrack = &state->mFastTracks[j]; 3168 3169 // Determine whether the track is currently in underrun condition, 3170 // and whether it had a recent underrun. 3171 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3172 FastTrackUnderruns underruns = ftDump->mUnderruns; 3173 uint32_t recentFull = (underruns.mBitFields.mFull - 3174 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3175 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3176 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3177 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3178 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3179 uint32_t recentUnderruns = recentPartial + recentEmpty; 3180 track->mObservedUnderruns = underruns; 3181 // don't count underruns that occur while stopping or pausing 3182 // or stopped which can occur when flush() is called while active 3183 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3184 recentUnderruns > 0) { 3185 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3186 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3187 } 3188 3189 // This is similar to the state machine for normal tracks, 3190 // with a few modifications for fast tracks. 3191 bool isActive = true; 3192 switch (track->mState) { 3193 case TrackBase::STOPPING_1: 3194 // track stays active in STOPPING_1 state until first underrun 3195 if (recentUnderruns > 0 || track->isTerminated()) { 3196 track->mState = TrackBase::STOPPING_2; 3197 } 3198 break; 3199 case TrackBase::PAUSING: 3200 // ramp down is not yet implemented 3201 track->setPaused(); 3202 break; 3203 case TrackBase::RESUMING: 3204 // ramp up is not yet implemented 3205 track->mState = TrackBase::ACTIVE; 3206 break; 3207 case TrackBase::ACTIVE: 3208 if (recentFull > 0 || recentPartial > 0) { 3209 // track has provided at least some frames recently: reset retry count 3210 track->mRetryCount = kMaxTrackRetries; 3211 } 3212 if (recentUnderruns == 0) { 3213 // no recent underruns: stay active 3214 break; 3215 } 3216 // there has recently been an underrun of some kind 3217 if (track->sharedBuffer() == 0) { 3218 // were any of the recent underruns "empty" (no frames available)? 3219 if (recentEmpty == 0) { 3220 // no, then ignore the partial underruns as they are allowed indefinitely 3221 break; 3222 } 3223 // there has recently been an "empty" underrun: decrement the retry counter 3224 if (--(track->mRetryCount) > 0) { 3225 break; 3226 } 3227 // indicate to client process that the track was disabled because of underrun; 3228 // it will then automatically call start() when data is available 3229 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3230 // remove from active list, but state remains ACTIVE [confusing but true] 3231 isActive = false; 3232 break; 3233 } 3234 // fall through 3235 case TrackBase::STOPPING_2: 3236 case TrackBase::PAUSED: 3237 case TrackBase::STOPPED: 3238 case TrackBase::FLUSHED: // flush() while active 3239 // Check for presentation complete if track is inactive 3240 // We have consumed all the buffers of this track. 3241 // This would be incomplete if we auto-paused on underrun 3242 { 3243 size_t audioHALFrames = 3244 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3245 size_t framesWritten = mBytesWritten / mFrameSize; 3246 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3247 // track stays in active list until presentation is complete 3248 break; 3249 } 3250 } 3251 if (track->isStopping_2()) { 3252 track->mState = TrackBase::STOPPED; 3253 } 3254 if (track->isStopped()) { 3255 // Can't reset directly, as fast mixer is still polling this track 3256 // track->reset(); 3257 // So instead mark this track as needing to be reset after push with ack 3258 resetMask |= 1 << i; 3259 } 3260 isActive = false; 3261 break; 3262 case TrackBase::IDLE: 3263 default: 3264 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3265 } 3266 3267 if (isActive) { 3268 // was it previously inactive? 3269 if (!(state->mTrackMask & (1 << j))) { 3270 ExtendedAudioBufferProvider *eabp = track; 3271 VolumeProvider *vp = track; 3272 fastTrack->mBufferProvider = eabp; 3273 fastTrack->mVolumeProvider = vp; 3274 fastTrack->mChannelMask = track->mChannelMask; 3275 fastTrack->mFormat = track->mFormat; 3276 fastTrack->mGeneration++; 3277 state->mTrackMask |= 1 << j; 3278 didModify = true; 3279 // no acknowledgement required for newly active tracks 3280 } 3281 // cache the combined master volume and stream type volume for fast mixer; this 3282 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3283 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3284 ++fastTracks; 3285 } else { 3286 // was it previously active? 3287 if (state->mTrackMask & (1 << j)) { 3288 fastTrack->mBufferProvider = NULL; 3289 fastTrack->mGeneration++; 3290 state->mTrackMask &= ~(1 << j); 3291 didModify = true; 3292 // If any fast tracks were removed, we must wait for acknowledgement 3293 // because we're about to decrement the last sp<> on those tracks. 3294 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3295 } else { 3296 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3297 } 3298 tracksToRemove->add(track); 3299 // Avoids a misleading display in dumpsys 3300 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3301 } 3302 continue; 3303 } 3304 3305 { // local variable scope to avoid goto warning 3306 3307 audio_track_cblk_t* cblk = track->cblk(); 3308 3309 // The first time a track is added we wait 3310 // for all its buffers to be filled before processing it 3311 int name = track->name(); 3312 // make sure that we have enough frames to mix one full buffer. 3313 // enforce this condition only once to enable draining the buffer in case the client 3314 // app does not call stop() and relies on underrun to stop: 3315 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3316 // during last round 3317 size_t desiredFrames; 3318 uint32_t sr = track->sampleRate(); 3319 if (sr == mSampleRate) { 3320 desiredFrames = mNormalFrameCount; 3321 } else { 3322 // +1 for rounding and +1 for additional sample needed for interpolation 3323 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3324 // add frames already consumed but not yet released by the resampler 3325 // because mAudioTrackServerProxy->framesReady() will include these frames 3326 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3327#if 0 3328 // the minimum track buffer size is normally twice the number of frames necessary 3329 // to fill one buffer and the resampler should not leave more than one buffer worth 3330 // of unreleased frames after each pass, but just in case... 3331 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3332#endif 3333 } 3334 uint32_t minFrames = 1; 3335 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3336 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3337 minFrames = desiredFrames; 3338 } 3339 3340 size_t framesReady = track->framesReady(); 3341 if ((framesReady >= minFrames) && track->isReady() && 3342 !track->isPaused() && !track->isTerminated()) 3343 { 3344 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3345 3346 mixedTracks++; 3347 3348 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3349 // there is an effect chain connected to the track 3350 chain.clear(); 3351 if (track->mainBuffer() != mSinkBuffer && 3352 track->mainBuffer() != mMixerBuffer) { 3353 if (mEffectBufferEnabled) { 3354 mEffectBufferValid = true; // Later can set directly. 3355 } 3356 chain = getEffectChain_l(track->sessionId()); 3357 // Delegate volume control to effect in track effect chain if needed 3358 if (chain != 0) { 3359 tracksWithEffect++; 3360 } else { 3361 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3362 "session %d", 3363 name, track->sessionId()); 3364 } 3365 } 3366 3367 3368 int param = AudioMixer::VOLUME; 3369 if (track->mFillingUpStatus == Track::FS_FILLED) { 3370 // no ramp for the first volume setting 3371 track->mFillingUpStatus = Track::FS_ACTIVE; 3372 if (track->mState == TrackBase::RESUMING) { 3373 track->mState = TrackBase::ACTIVE; 3374 param = AudioMixer::RAMP_VOLUME; 3375 } 3376 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3377 // FIXME should not make a decision based on mServer 3378 } else if (cblk->mServer != 0) { 3379 // If the track is stopped before the first frame was mixed, 3380 // do not apply ramp 3381 param = AudioMixer::RAMP_VOLUME; 3382 } 3383 3384 // compute volume for this track 3385 uint32_t vl, vr; // in U8.24 integer format 3386 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3387 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3388 vl = vr = 0; 3389 vlf = vrf = vaf = 0.; 3390 if (track->isPausing()) { 3391 track->setPaused(); 3392 } 3393 } else { 3394 3395 // read original volumes with volume control 3396 float typeVolume = mStreamTypes[track->streamType()].volume; 3397 float v = masterVolume * typeVolume; 3398 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3399 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3400 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3401 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3402 // track volumes come from shared memory, so can't be trusted and must be clamped 3403 if (vlf > GAIN_FLOAT_UNITY) { 3404 ALOGV("Track left volume out of range: %.3g", vlf); 3405 vlf = GAIN_FLOAT_UNITY; 3406 } 3407 if (vrf > GAIN_FLOAT_UNITY) { 3408 ALOGV("Track right volume out of range: %.3g", vrf); 3409 vrf = GAIN_FLOAT_UNITY; 3410 } 3411 // now apply the master volume and stream type volume 3412 vlf *= v; 3413 vrf *= v; 3414 // assuming master volume and stream type volume each go up to 1.0, 3415 // then derive vl and vr as U8.24 versions for the effect chain 3416 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3417 vl = (uint32_t) (scaleto8_24 * vlf); 3418 vr = (uint32_t) (scaleto8_24 * vrf); 3419 // vl and vr are now in U8.24 format 3420 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3421 // send level comes from shared memory and so may be corrupt 3422 if (sendLevel > MAX_GAIN_INT) { 3423 ALOGV("Track send level out of range: %04X", sendLevel); 3424 sendLevel = MAX_GAIN_INT; 3425 } 3426 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3427 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3428 } 3429 3430 // Delegate volume control to effect in track effect chain if needed 3431 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3432 // Do not ramp volume if volume is controlled by effect 3433 param = AudioMixer::VOLUME; 3434 // Update remaining floating point volume levels 3435 vlf = (float)vl / (1 << 24); 3436 vrf = (float)vr / (1 << 24); 3437 track->mHasVolumeController = true; 3438 } else { 3439 // force no volume ramp when volume controller was just disabled or removed 3440 // from effect chain to avoid volume spike 3441 if (track->mHasVolumeController) { 3442 param = AudioMixer::VOLUME; 3443 } 3444 track->mHasVolumeController = false; 3445 } 3446 3447 // XXX: these things DON'T need to be done each time 3448 mAudioMixer->setBufferProvider(name, track); 3449 mAudioMixer->enable(name); 3450 3451 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3452 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3453 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3454 mAudioMixer->setParameter( 3455 name, 3456 AudioMixer::TRACK, 3457 AudioMixer::FORMAT, (void *)track->format()); 3458 mAudioMixer->setParameter( 3459 name, 3460 AudioMixer::TRACK, 3461 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3462 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3463 uint32_t maxSampleRate = mSampleRate * 2; 3464 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3465 if (reqSampleRate == 0) { 3466 reqSampleRate = mSampleRate; 3467 } else if (reqSampleRate > maxSampleRate) { 3468 reqSampleRate = maxSampleRate; 3469 } 3470 mAudioMixer->setParameter( 3471 name, 3472 AudioMixer::RESAMPLE, 3473 AudioMixer::SAMPLE_RATE, 3474 (void *)(uintptr_t)reqSampleRate); 3475 /* 3476 * Select the appropriate output buffer for the track. 3477 * 3478 * Tracks with effects go into their own effects chain buffer 3479 * and from there into either mEffectBuffer or mSinkBuffer. 3480 * 3481 * Other tracks can use mMixerBuffer for higher precision 3482 * channel accumulation. If this buffer is enabled 3483 * (mMixerBufferEnabled true), then selected tracks will accumulate 3484 * into it. 3485 * 3486 */ 3487 if (mMixerBufferEnabled 3488 && (track->mainBuffer() == mSinkBuffer 3489 || track->mainBuffer() == mMixerBuffer)) { 3490 mAudioMixer->setParameter( 3491 name, 3492 AudioMixer::TRACK, 3493 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3494 mAudioMixer->setParameter( 3495 name, 3496 AudioMixer::TRACK, 3497 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3498 // TODO: override track->mainBuffer()? 3499 mMixerBufferValid = true; 3500 } else { 3501 mAudioMixer->setParameter( 3502 name, 3503 AudioMixer::TRACK, 3504 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3505 mAudioMixer->setParameter( 3506 name, 3507 AudioMixer::TRACK, 3508 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3509 } 3510 mAudioMixer->setParameter( 3511 name, 3512 AudioMixer::TRACK, 3513 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3514 3515 // reset retry count 3516 track->mRetryCount = kMaxTrackRetries; 3517 3518 // If one track is ready, set the mixer ready if: 3519 // - the mixer was not ready during previous round OR 3520 // - no other track is not ready 3521 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3522 mixerStatus != MIXER_TRACKS_ENABLED) { 3523 mixerStatus = MIXER_TRACKS_READY; 3524 } 3525 } else { 3526 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3527 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3528 } 3529 // clear effect chain input buffer if an active track underruns to avoid sending 3530 // previous audio buffer again to effects 3531 chain = getEffectChain_l(track->sessionId()); 3532 if (chain != 0) { 3533 chain->clearInputBuffer(); 3534 } 3535 3536 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3537 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3538 track->isStopped() || track->isPaused()) { 3539 // We have consumed all the buffers of this track. 3540 // Remove it from the list of active tracks. 3541 // TODO: use actual buffer filling status instead of latency when available from 3542 // audio HAL 3543 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3544 size_t framesWritten = mBytesWritten / mFrameSize; 3545 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3546 if (track->isStopped()) { 3547 track->reset(); 3548 } 3549 tracksToRemove->add(track); 3550 } 3551 } else { 3552 // No buffers for this track. Give it a few chances to 3553 // fill a buffer, then remove it from active list. 3554 if (--(track->mRetryCount) <= 0) { 3555 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3556 tracksToRemove->add(track); 3557 // indicate to client process that the track was disabled because of underrun; 3558 // it will then automatically call start() when data is available 3559 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3560 // If one track is not ready, mark the mixer also not ready if: 3561 // - the mixer was ready during previous round OR 3562 // - no other track is ready 3563 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3564 mixerStatus != MIXER_TRACKS_READY) { 3565 mixerStatus = MIXER_TRACKS_ENABLED; 3566 } 3567 } 3568 mAudioMixer->disable(name); 3569 } 3570 3571 } // local variable scope to avoid goto warning 3572track_is_ready: ; 3573 3574 } 3575 3576 // Push the new FastMixer state if necessary 3577 bool pauseAudioWatchdog = false; 3578 if (didModify) { 3579 state->mFastTracksGen++; 3580 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3581 if (kUseFastMixer == FastMixer_Dynamic && 3582 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3583 state->mCommand = FastMixerState::COLD_IDLE; 3584 state->mColdFutexAddr = &mFastMixerFutex; 3585 state->mColdGen++; 3586 mFastMixerFutex = 0; 3587 if (kUseFastMixer == FastMixer_Dynamic) { 3588 mNormalSink = mOutputSink; 3589 } 3590 // If we go into cold idle, need to wait for acknowledgement 3591 // so that fast mixer stops doing I/O. 3592 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3593 pauseAudioWatchdog = true; 3594 } 3595 } 3596 if (sq != NULL) { 3597 sq->end(didModify); 3598 sq->push(block); 3599 } 3600#ifdef AUDIO_WATCHDOG 3601 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3602 mAudioWatchdog->pause(); 3603 } 3604#endif 3605 3606 // Now perform the deferred reset on fast tracks that have stopped 3607 while (resetMask != 0) { 3608 size_t i = __builtin_ctz(resetMask); 3609 ALOG_ASSERT(i < count); 3610 resetMask &= ~(1 << i); 3611 sp<Track> t = mActiveTracks[i].promote(); 3612 if (t == 0) { 3613 continue; 3614 } 3615 Track* track = t.get(); 3616 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3617 track->reset(); 3618 } 3619 3620 // remove all the tracks that need to be... 3621 removeTracks_l(*tracksToRemove); 3622 3623 // sink or mix buffer must be cleared if all tracks are connected to an 3624 // effect chain as in this case the mixer will not write to the sink or mix buffer 3625 // and track effects will accumulate into it 3626 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3627 (mixedTracks == 0 && fastTracks > 0))) { 3628 // FIXME as a performance optimization, should remember previous zero status 3629 if (mMixerBufferValid) { 3630 memset(mMixerBuffer, 0, mMixerBufferSize); 3631 // TODO: In testing, mSinkBuffer below need not be cleared because 3632 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3633 // after mixing. 3634 // 3635 // To enforce this guarantee: 3636 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3637 // (mixedTracks == 0 && fastTracks > 0)) 3638 // must imply MIXER_TRACKS_READY. 3639 // Later, we may clear buffers regardless, and skip much of this logic. 3640 } 3641 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3642 if (mEffectBufferValid) { 3643 memset(mEffectBuffer, 0, mEffectBufferSize); 3644 } 3645 // FIXME as a performance optimization, should remember previous zero status 3646 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3647 } 3648 3649 // if any fast tracks, then status is ready 3650 mMixerStatusIgnoringFastTracks = mixerStatus; 3651 if (fastTracks > 0) { 3652 mixerStatus = MIXER_TRACKS_READY; 3653 } 3654 return mixerStatus; 3655} 3656 3657// getTrackName_l() must be called with ThreadBase::mLock held 3658int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3659 audio_format_t format, int sessionId) 3660{ 3661 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3662} 3663 3664// deleteTrackName_l() must be called with ThreadBase::mLock held 3665void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3666{ 3667 ALOGV("remove track (%d) and delete from mixer", name); 3668 mAudioMixer->deleteTrackName(name); 3669} 3670 3671// checkForNewParameter_l() must be called with ThreadBase::mLock held 3672bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3673 status_t& status) 3674{ 3675 bool reconfig = false; 3676 3677 status = NO_ERROR; 3678 3679 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3680 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3681 if (mFastMixer != 0) { 3682 FastMixerStateQueue *sq = mFastMixer->sq(); 3683 FastMixerState *state = sq->begin(); 3684 if (!(state->mCommand & FastMixerState::IDLE)) { 3685 previousCommand = state->mCommand; 3686 state->mCommand = FastMixerState::HOT_IDLE; 3687 sq->end(); 3688 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3689 } else { 3690 sq->end(false /*didModify*/); 3691 } 3692 } 3693 3694 AudioParameter param = AudioParameter(keyValuePair); 3695 int value; 3696 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3697 reconfig = true; 3698 } 3699 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3700 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3701 status = BAD_VALUE; 3702 } else { 3703 // no need to save value, since it's constant 3704 reconfig = true; 3705 } 3706 } 3707 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3708 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3709 status = BAD_VALUE; 3710 } else { 3711 // no need to save value, since it's constant 3712 reconfig = true; 3713 } 3714 } 3715 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3716 // do not accept frame count changes if tracks are open as the track buffer 3717 // size depends on frame count and correct behavior would not be guaranteed 3718 // if frame count is changed after track creation 3719 if (!mTracks.isEmpty()) { 3720 status = INVALID_OPERATION; 3721 } else { 3722 reconfig = true; 3723 } 3724 } 3725 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3726#ifdef ADD_BATTERY_DATA 3727 // when changing the audio output device, call addBatteryData to notify 3728 // the change 3729 if (mOutDevice != value) { 3730 uint32_t params = 0; 3731 // check whether speaker is on 3732 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3733 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3734 } 3735 3736 audio_devices_t deviceWithoutSpeaker 3737 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3738 // check if any other device (except speaker) is on 3739 if (value & deviceWithoutSpeaker ) { 3740 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3741 } 3742 3743 if (params != 0) { 3744 addBatteryData(params); 3745 } 3746 } 3747#endif 3748 3749 // forward device change to effects that have requested to be 3750 // aware of attached audio device. 3751 if (value != AUDIO_DEVICE_NONE) { 3752 mOutDevice = value; 3753 for (size_t i = 0; i < mEffectChains.size(); i++) { 3754 mEffectChains[i]->setDevice_l(mOutDevice); 3755 } 3756 } 3757 } 3758 3759 if (status == NO_ERROR) { 3760 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3761 keyValuePair.string()); 3762 if (!mStandby && status == INVALID_OPERATION) { 3763 mOutput->stream->common.standby(&mOutput->stream->common); 3764 mStandby = true; 3765 mBytesWritten = 0; 3766 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3767 keyValuePair.string()); 3768 } 3769 if (status == NO_ERROR && reconfig) { 3770 readOutputParameters_l(); 3771 delete mAudioMixer; 3772 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3773 for (size_t i = 0; i < mTracks.size() ; i++) { 3774 int name = getTrackName_l(mTracks[i]->mChannelMask, 3775 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3776 if (name < 0) { 3777 break; 3778 } 3779 mTracks[i]->mName = name; 3780 } 3781 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3782 } 3783 } 3784 3785 if (!(previousCommand & FastMixerState::IDLE)) { 3786 ALOG_ASSERT(mFastMixer != 0); 3787 FastMixerStateQueue *sq = mFastMixer->sq(); 3788 FastMixerState *state = sq->begin(); 3789 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3790 state->mCommand = previousCommand; 3791 sq->end(); 3792 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3793 } 3794 3795 return reconfig; 3796} 3797 3798 3799void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3800{ 3801 const size_t SIZE = 256; 3802 char buffer[SIZE]; 3803 String8 result; 3804 3805 PlaybackThread::dumpInternals(fd, args); 3806 3807 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3808 3809 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3810 const FastMixerDumpState copy(mFastMixerDumpState); 3811 copy.dump(fd); 3812 3813#ifdef STATE_QUEUE_DUMP 3814 // Similar for state queue 3815 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3816 observerCopy.dump(fd); 3817 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3818 mutatorCopy.dump(fd); 3819#endif 3820 3821#ifdef TEE_SINK 3822 // Write the tee output to a .wav file 3823 dumpTee(fd, mTeeSource, mId); 3824#endif 3825 3826#ifdef AUDIO_WATCHDOG 3827 if (mAudioWatchdog != 0) { 3828 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3829 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3830 wdCopy.dump(fd); 3831 } 3832#endif 3833} 3834 3835uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3836{ 3837 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3838} 3839 3840uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3841{ 3842 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3843} 3844 3845void AudioFlinger::MixerThread::cacheParameters_l() 3846{ 3847 PlaybackThread::cacheParameters_l(); 3848 3849 // FIXME: Relaxed timing because of a certain device that can't meet latency 3850 // Should be reduced to 2x after the vendor fixes the driver issue 3851 // increase threshold again due to low power audio mode. The way this warning 3852 // threshold is calculated and its usefulness should be reconsidered anyway. 3853 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3854} 3855 3856// ---------------------------------------------------------------------------- 3857 3858AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3859 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3860 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3861 // mLeftVolFloat, mRightVolFloat 3862{ 3863} 3864 3865AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3866 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3867 ThreadBase::type_t type) 3868 : PlaybackThread(audioFlinger, output, id, device, type) 3869 // mLeftVolFloat, mRightVolFloat 3870{ 3871} 3872 3873AudioFlinger::DirectOutputThread::~DirectOutputThread() 3874{ 3875} 3876 3877void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3878{ 3879 audio_track_cblk_t* cblk = track->cblk(); 3880 float left, right; 3881 3882 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3883 left = right = 0; 3884 } else { 3885 float typeVolume = mStreamTypes[track->streamType()].volume; 3886 float v = mMasterVolume * typeVolume; 3887 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3888 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3889 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3890 if (left > GAIN_FLOAT_UNITY) { 3891 left = GAIN_FLOAT_UNITY; 3892 } 3893 left *= v; 3894 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3895 if (right > GAIN_FLOAT_UNITY) { 3896 right = GAIN_FLOAT_UNITY; 3897 } 3898 right *= v; 3899 } 3900 3901 if (lastTrack) { 3902 if (left != mLeftVolFloat || right != mRightVolFloat) { 3903 mLeftVolFloat = left; 3904 mRightVolFloat = right; 3905 3906 // Convert volumes from float to 8.24 3907 uint32_t vl = (uint32_t)(left * (1 << 24)); 3908 uint32_t vr = (uint32_t)(right * (1 << 24)); 3909 3910 // Delegate volume control to effect in track effect chain if needed 3911 // only one effect chain can be present on DirectOutputThread, so if 3912 // there is one, the track is connected to it 3913 if (!mEffectChains.isEmpty()) { 3914 mEffectChains[0]->setVolume_l(&vl, &vr); 3915 left = (float)vl / (1 << 24); 3916 right = (float)vr / (1 << 24); 3917 } 3918 if (mOutput->stream->set_volume) { 3919 mOutput->stream->set_volume(mOutput->stream, left, right); 3920 } 3921 } 3922 } 3923} 3924 3925 3926AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3927 Vector< sp<Track> > *tracksToRemove 3928) 3929{ 3930 size_t count = mActiveTracks.size(); 3931 mixer_state mixerStatus = MIXER_IDLE; 3932 3933 // find out which tracks need to be processed 3934 for (size_t i = 0; i < count; i++) { 3935 sp<Track> t = mActiveTracks[i].promote(); 3936 // The track died recently 3937 if (t == 0) { 3938 continue; 3939 } 3940 3941 Track* const track = t.get(); 3942 audio_track_cblk_t* cblk = track->cblk(); 3943 // Only consider last track started for volume and mixer state control. 3944 // In theory an older track could underrun and restart after the new one starts 3945 // but as we only care about the transition phase between two tracks on a 3946 // direct output, it is not a problem to ignore the underrun case. 3947 sp<Track> l = mLatestActiveTrack.promote(); 3948 bool last = l.get() == track; 3949 3950 // The first time a track is added we wait 3951 // for all its buffers to be filled before processing it 3952 uint32_t minFrames; 3953 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 3954 minFrames = mNormalFrameCount; 3955 } else { 3956 minFrames = 1; 3957 } 3958 3959 ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ", 3960 minFrames, track->mState, track->framesReady()); 3961 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 3962 !track->isStopping_2() && !track->isStopped()) 3963 { 3964 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3965 3966 if (track->mFillingUpStatus == Track::FS_FILLED) { 3967 track->mFillingUpStatus = Track::FS_ACTIVE; 3968 // make sure processVolume_l() will apply new volume even if 0 3969 mLeftVolFloat = mRightVolFloat = -1.0; 3970 if (track->mState == TrackBase::RESUMING) { 3971 track->mState = TrackBase::ACTIVE; 3972 } 3973 } 3974 3975 // compute volume for this track 3976 processVolume_l(track, last); 3977 if (last) { 3978 // reset retry count 3979 track->mRetryCount = kMaxTrackRetriesDirect; 3980 mActiveTrack = t; 3981 mixerStatus = MIXER_TRACKS_READY; 3982 } 3983 } else { 3984 // clear effect chain input buffer if the last active track started underruns 3985 // to avoid sending previous audio buffer again to effects 3986 if (!mEffectChains.isEmpty() && last) { 3987 mEffectChains[0]->clearInputBuffer(); 3988 } 3989 if (track->isStopping_1()) { 3990 track->mState = TrackBase::STOPPING_2; 3991 } 3992 if ((track->sharedBuffer() != 0) || track->isStopped() || 3993 track->isStopping_2() || track->isPaused()) { 3994 // We have consumed all the buffers of this track. 3995 // Remove it from the list of active tracks. 3996 size_t audioHALFrames; 3997 if (audio_is_linear_pcm(mFormat)) { 3998 audioHALFrames = (latency_l() * mSampleRate) / 1000; 3999 } else { 4000 audioHALFrames = 0; 4001 } 4002 4003 size_t framesWritten = mBytesWritten / mFrameSize; 4004 if (mStandby || !last || 4005 track->presentationComplete(framesWritten, audioHALFrames)) { 4006 if (track->isStopping_2()) { 4007 track->mState = TrackBase::STOPPED; 4008 } 4009 if (track->isStopped()) { 4010 track->reset(); 4011 } 4012 tracksToRemove->add(track); 4013 } 4014 } else { 4015 // No buffers for this track. Give it a few chances to 4016 // fill a buffer, then remove it from active list. 4017 // Only consider last track started for mixer state control 4018 if (--(track->mRetryCount) <= 0) { 4019 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4020 tracksToRemove->add(track); 4021 // indicate to client process that the track was disabled because of underrun; 4022 // it will then automatically call start() when data is available 4023 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4024 } else if (last) { 4025 mixerStatus = MIXER_TRACKS_ENABLED; 4026 } 4027 } 4028 } 4029 } 4030 4031 // remove all the tracks that need to be... 4032 removeTracks_l(*tracksToRemove); 4033 4034 return mixerStatus; 4035} 4036 4037void AudioFlinger::DirectOutputThread::threadLoop_mix() 4038{ 4039 size_t frameCount = mFrameCount; 4040 int8_t *curBuf = (int8_t *)mSinkBuffer; 4041 // output audio to hardware 4042 while (frameCount) { 4043 AudioBufferProvider::Buffer buffer; 4044 buffer.frameCount = frameCount; 4045 mActiveTrack->getNextBuffer(&buffer); 4046 if (buffer.raw == NULL) { 4047 memset(curBuf, 0, frameCount * mFrameSize); 4048 break; 4049 } 4050 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4051 frameCount -= buffer.frameCount; 4052 curBuf += buffer.frameCount * mFrameSize; 4053 mActiveTrack->releaseBuffer(&buffer); 4054 } 4055 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4056 sleepTime = 0; 4057 standbyTime = systemTime() + standbyDelay; 4058 mActiveTrack.clear(); 4059} 4060 4061void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4062{ 4063 if (sleepTime == 0) { 4064 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4065 sleepTime = activeSleepTime; 4066 } else { 4067 sleepTime = idleSleepTime; 4068 } 4069 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4070 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4071 sleepTime = 0; 4072 } 4073} 4074 4075// getTrackName_l() must be called with ThreadBase::mLock held 4076int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4077 audio_format_t format __unused, int sessionId __unused) 4078{ 4079 return 0; 4080} 4081 4082// deleteTrackName_l() must be called with ThreadBase::mLock held 4083void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4084{ 4085} 4086 4087// checkForNewParameter_l() must be called with ThreadBase::mLock held 4088bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4089 status_t& status) 4090{ 4091 bool reconfig = false; 4092 4093 status = NO_ERROR; 4094 4095 AudioParameter param = AudioParameter(keyValuePair); 4096 int value; 4097 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4098 // forward device change to effects that have requested to be 4099 // aware of attached audio device. 4100 if (value != AUDIO_DEVICE_NONE) { 4101 mOutDevice = value; 4102 for (size_t i = 0; i < mEffectChains.size(); i++) { 4103 mEffectChains[i]->setDevice_l(mOutDevice); 4104 } 4105 } 4106 } 4107 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4108 // do not accept frame count changes if tracks are open as the track buffer 4109 // size depends on frame count and correct behavior would not be garantied 4110 // if frame count is changed after track creation 4111 if (!mTracks.isEmpty()) { 4112 status = INVALID_OPERATION; 4113 } else { 4114 reconfig = true; 4115 } 4116 } 4117 if (status == NO_ERROR) { 4118 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4119 keyValuePair.string()); 4120 if (!mStandby && status == INVALID_OPERATION) { 4121 mOutput->stream->common.standby(&mOutput->stream->common); 4122 mStandby = true; 4123 mBytesWritten = 0; 4124 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4125 keyValuePair.string()); 4126 } 4127 if (status == NO_ERROR && reconfig) { 4128 readOutputParameters_l(); 4129 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4130 } 4131 } 4132 4133 return reconfig; 4134} 4135 4136uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4137{ 4138 uint32_t time; 4139 if (audio_is_linear_pcm(mFormat)) { 4140 time = PlaybackThread::activeSleepTimeUs(); 4141 } else { 4142 time = 10000; 4143 } 4144 return time; 4145} 4146 4147uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4148{ 4149 uint32_t time; 4150 if (audio_is_linear_pcm(mFormat)) { 4151 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4152 } else { 4153 time = 10000; 4154 } 4155 return time; 4156} 4157 4158uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4159{ 4160 uint32_t time; 4161 if (audio_is_linear_pcm(mFormat)) { 4162 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4163 } else { 4164 time = 10000; 4165 } 4166 return time; 4167} 4168 4169void AudioFlinger::DirectOutputThread::cacheParameters_l() 4170{ 4171 PlaybackThread::cacheParameters_l(); 4172 4173 // use shorter standby delay as on normal output to release 4174 // hardware resources as soon as possible 4175 if (audio_is_linear_pcm(mFormat)) { 4176 standbyDelay = microseconds(activeSleepTime*2); 4177 } else { 4178 standbyDelay = kOffloadStandbyDelayNs; 4179 } 4180} 4181 4182// ---------------------------------------------------------------------------- 4183 4184AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4185 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4186 : Thread(false /*canCallJava*/), 4187 mPlaybackThread(playbackThread), 4188 mWriteAckSequence(0), 4189 mDrainSequence(0) 4190{ 4191} 4192 4193AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4194{ 4195} 4196 4197void AudioFlinger::AsyncCallbackThread::onFirstRef() 4198{ 4199 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4200} 4201 4202bool AudioFlinger::AsyncCallbackThread::threadLoop() 4203{ 4204 while (!exitPending()) { 4205 uint32_t writeAckSequence; 4206 uint32_t drainSequence; 4207 4208 { 4209 Mutex::Autolock _l(mLock); 4210 while (!((mWriteAckSequence & 1) || 4211 (mDrainSequence & 1) || 4212 exitPending())) { 4213 mWaitWorkCV.wait(mLock); 4214 } 4215 4216 if (exitPending()) { 4217 break; 4218 } 4219 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4220 mWriteAckSequence, mDrainSequence); 4221 writeAckSequence = mWriteAckSequence; 4222 mWriteAckSequence &= ~1; 4223 drainSequence = mDrainSequence; 4224 mDrainSequence &= ~1; 4225 } 4226 { 4227 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4228 if (playbackThread != 0) { 4229 if (writeAckSequence & 1) { 4230 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4231 } 4232 if (drainSequence & 1) { 4233 playbackThread->resetDraining(drainSequence >> 1); 4234 } 4235 } 4236 } 4237 } 4238 return false; 4239} 4240 4241void AudioFlinger::AsyncCallbackThread::exit() 4242{ 4243 ALOGV("AsyncCallbackThread::exit"); 4244 Mutex::Autolock _l(mLock); 4245 requestExit(); 4246 mWaitWorkCV.broadcast(); 4247} 4248 4249void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4250{ 4251 Mutex::Autolock _l(mLock); 4252 // bit 0 is cleared 4253 mWriteAckSequence = sequence << 1; 4254} 4255 4256void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4257{ 4258 Mutex::Autolock _l(mLock); 4259 // ignore unexpected callbacks 4260 if (mWriteAckSequence & 2) { 4261 mWriteAckSequence |= 1; 4262 mWaitWorkCV.signal(); 4263 } 4264} 4265 4266void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4267{ 4268 Mutex::Autolock _l(mLock); 4269 // bit 0 is cleared 4270 mDrainSequence = sequence << 1; 4271} 4272 4273void AudioFlinger::AsyncCallbackThread::resetDraining() 4274{ 4275 Mutex::Autolock _l(mLock); 4276 // ignore unexpected callbacks 4277 if (mDrainSequence & 2) { 4278 mDrainSequence |= 1; 4279 mWaitWorkCV.signal(); 4280 } 4281} 4282 4283 4284// ---------------------------------------------------------------------------- 4285AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4286 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4287 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4288 mHwPaused(false), 4289 mFlushPending(false), 4290 mPausedBytesRemaining(0) 4291{ 4292 //FIXME: mStandby should be set to true by ThreadBase constructor 4293 mStandby = true; 4294} 4295 4296void AudioFlinger::OffloadThread::threadLoop_exit() 4297{ 4298 if (mFlushPending || mHwPaused) { 4299 // If a flush is pending or track was paused, just discard buffered data 4300 flushHw_l(); 4301 } else { 4302 mMixerStatus = MIXER_DRAIN_ALL; 4303 threadLoop_drain(); 4304 } 4305 if (mUseAsyncWrite) { 4306 ALOG_ASSERT(mCallbackThread != 0); 4307 mCallbackThread->exit(); 4308 } 4309 PlaybackThread::threadLoop_exit(); 4310} 4311 4312AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4313 Vector< sp<Track> > *tracksToRemove 4314) 4315{ 4316 size_t count = mActiveTracks.size(); 4317 4318 mixer_state mixerStatus = MIXER_IDLE; 4319 bool doHwPause = false; 4320 bool doHwResume = false; 4321 4322 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4323 4324 // find out which tracks need to be processed 4325 for (size_t i = 0; i < count; i++) { 4326 sp<Track> t = mActiveTracks[i].promote(); 4327 // The track died recently 4328 if (t == 0) { 4329 continue; 4330 } 4331 Track* const track = t.get(); 4332 audio_track_cblk_t* cblk = track->cblk(); 4333 // Only consider last track started for volume and mixer state control. 4334 // In theory an older track could underrun and restart after the new one starts 4335 // but as we only care about the transition phase between two tracks on a 4336 // direct output, it is not a problem to ignore the underrun case. 4337 sp<Track> l = mLatestActiveTrack.promote(); 4338 bool last = l.get() == track; 4339 4340 if (track->isInvalid()) { 4341 ALOGW("An invalidated track shouldn't be in active list"); 4342 tracksToRemove->add(track); 4343 continue; 4344 } 4345 4346 if (track->mState == TrackBase::IDLE) { 4347 ALOGW("An idle track shouldn't be in active list"); 4348 continue; 4349 } 4350 4351 if (track->isPausing()) { 4352 track->setPaused(); 4353 if (last) { 4354 if (!mHwPaused) { 4355 doHwPause = true; 4356 mHwPaused = true; 4357 } 4358 // If we were part way through writing the mixbuffer to 4359 // the HAL we must save this until we resume 4360 // BUG - this will be wrong if a different track is made active, 4361 // in that case we want to discard the pending data in the 4362 // mixbuffer and tell the client to present it again when the 4363 // track is resumed 4364 mPausedWriteLength = mCurrentWriteLength; 4365 mPausedBytesRemaining = mBytesRemaining; 4366 mBytesRemaining = 0; // stop writing 4367 } 4368 tracksToRemove->add(track); 4369 } else if (track->isFlushPending()) { 4370 track->flushAck(); 4371 if (last) { 4372 mFlushPending = true; 4373 } 4374 } else if (track->isResumePending()){ 4375 track->resumeAck(); 4376 if (last) { 4377 if (mPausedBytesRemaining) { 4378 // Need to continue write that was interrupted 4379 mCurrentWriteLength = mPausedWriteLength; 4380 mBytesRemaining = mPausedBytesRemaining; 4381 mPausedBytesRemaining = 0; 4382 } 4383 if (mHwPaused) { 4384 doHwResume = true; 4385 mHwPaused = false; 4386 // threadLoop_mix() will handle the case that we need to 4387 // resume an interrupted write 4388 } 4389 // enable write to audio HAL 4390 sleepTime = 0; 4391 4392 // Do not handle new data in this iteration even if track->framesReady() 4393 mixerStatus = MIXER_TRACKS_ENABLED; 4394 } 4395 } else if (track->framesReady() && track->isReady() && 4396 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4397 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4398 if (track->mFillingUpStatus == Track::FS_FILLED) { 4399 track->mFillingUpStatus = Track::FS_ACTIVE; 4400 // make sure processVolume_l() will apply new volume even if 0 4401 mLeftVolFloat = mRightVolFloat = -1.0; 4402 } 4403 4404 if (last) { 4405 sp<Track> previousTrack = mPreviousTrack.promote(); 4406 if (previousTrack != 0) { 4407 if (track != previousTrack.get()) { 4408 // Flush any data still being written from last track 4409 mBytesRemaining = 0; 4410 if (mPausedBytesRemaining) { 4411 // Last track was paused so we also need to flush saved 4412 // mixbuffer state and invalidate track so that it will 4413 // re-submit that unwritten data when it is next resumed 4414 mPausedBytesRemaining = 0; 4415 // Invalidate is a bit drastic - would be more efficient 4416 // to have a flag to tell client that some of the 4417 // previously written data was lost 4418 previousTrack->invalidate(); 4419 } 4420 // flush data already sent to the DSP if changing audio session as audio 4421 // comes from a different source. Also invalidate previous track to force a 4422 // seek when resuming. 4423 if (previousTrack->sessionId() != track->sessionId()) { 4424 previousTrack->invalidate(); 4425 } 4426 } 4427 } 4428 mPreviousTrack = track; 4429 // reset retry count 4430 track->mRetryCount = kMaxTrackRetriesOffload; 4431 mActiveTrack = t; 4432 mixerStatus = MIXER_TRACKS_READY; 4433 } 4434 } else { 4435 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4436 if (track->isStopping_1()) { 4437 // Hardware buffer can hold a large amount of audio so we must 4438 // wait for all current track's data to drain before we say 4439 // that the track is stopped. 4440 if (mBytesRemaining == 0) { 4441 // Only start draining when all data in mixbuffer 4442 // has been written 4443 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4444 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4445 // do not drain if no data was ever sent to HAL (mStandby == true) 4446 if (last && !mStandby) { 4447 // do not modify drain sequence if we are already draining. This happens 4448 // when resuming from pause after drain. 4449 if ((mDrainSequence & 1) == 0) { 4450 sleepTime = 0; 4451 standbyTime = systemTime() + standbyDelay; 4452 mixerStatus = MIXER_DRAIN_TRACK; 4453 mDrainSequence += 2; 4454 } 4455 if (mHwPaused) { 4456 // It is possible to move from PAUSED to STOPPING_1 without 4457 // a resume so we must ensure hardware is running 4458 doHwResume = true; 4459 mHwPaused = false; 4460 } 4461 } 4462 } 4463 } else if (track->isStopping_2()) { 4464 // Drain has completed or we are in standby, signal presentation complete 4465 if (!(mDrainSequence & 1) || !last || mStandby) { 4466 track->mState = TrackBase::STOPPED; 4467 size_t audioHALFrames = 4468 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4469 size_t framesWritten = 4470 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4471 track->presentationComplete(framesWritten, audioHALFrames); 4472 track->reset(); 4473 tracksToRemove->add(track); 4474 } 4475 } else { 4476 // No buffers for this track. Give it a few chances to 4477 // fill a buffer, then remove it from active list. 4478 if (--(track->mRetryCount) <= 0) { 4479 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4480 track->name()); 4481 tracksToRemove->add(track); 4482 // indicate to client process that the track was disabled because of underrun; 4483 // it will then automatically call start() when data is available 4484 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4485 } else if (last){ 4486 mixerStatus = MIXER_TRACKS_ENABLED; 4487 } 4488 } 4489 } 4490 // compute volume for this track 4491 processVolume_l(track, last); 4492 } 4493 4494 // make sure the pause/flush/resume sequence is executed in the right order. 4495 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4496 // before flush and then resume HW. This can happen in case of pause/flush/resume 4497 // if resume is received before pause is executed. 4498 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4499 mOutput->stream->pause(mOutput->stream); 4500 } 4501 if (mFlushPending) { 4502 flushHw_l(); 4503 mFlushPending = false; 4504 } 4505 if (!mStandby && doHwResume) { 4506 mOutput->stream->resume(mOutput->stream); 4507 } 4508 4509 // remove all the tracks that need to be... 4510 removeTracks_l(*tracksToRemove); 4511 4512 return mixerStatus; 4513} 4514 4515// must be called with thread mutex locked 4516bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4517{ 4518 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4519 mWriteAckSequence, mDrainSequence); 4520 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4521 return true; 4522 } 4523 return false; 4524} 4525 4526// must be called with thread mutex locked 4527bool AudioFlinger::OffloadThread::shouldStandby_l() 4528{ 4529 bool trackPaused = false; 4530 4531 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4532 // after a timeout and we will enter standby then. 4533 if (mTracks.size() > 0) { 4534 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4535 } 4536 4537 return !mStandby && !trackPaused; 4538} 4539 4540 4541bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4542{ 4543 Mutex::Autolock _l(mLock); 4544 return waitingAsyncCallback_l(); 4545} 4546 4547void AudioFlinger::OffloadThread::flushHw_l() 4548{ 4549 mOutput->stream->flush(mOutput->stream); 4550 // Flush anything still waiting in the mixbuffer 4551 mCurrentWriteLength = 0; 4552 mBytesRemaining = 0; 4553 mPausedWriteLength = 0; 4554 mPausedBytesRemaining = 0; 4555 mHwPaused = false; 4556 4557 if (mUseAsyncWrite) { 4558 // discard any pending drain or write ack by incrementing sequence 4559 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4560 mDrainSequence = (mDrainSequence + 2) & ~1; 4561 ALOG_ASSERT(mCallbackThread != 0); 4562 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4563 mCallbackThread->setDraining(mDrainSequence); 4564 } 4565} 4566 4567void AudioFlinger::OffloadThread::onAddNewTrack_l() 4568{ 4569 sp<Track> previousTrack = mPreviousTrack.promote(); 4570 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4571 4572 if (previousTrack != 0 && latestTrack != 0 && 4573 (previousTrack->sessionId() != latestTrack->sessionId())) { 4574 mFlushPending = true; 4575 } 4576 PlaybackThread::onAddNewTrack_l(); 4577} 4578 4579// ---------------------------------------------------------------------------- 4580 4581AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4582 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4583 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4584 DUPLICATING), 4585 mWaitTimeMs(UINT_MAX) 4586{ 4587 addOutputTrack(mainThread); 4588} 4589 4590AudioFlinger::DuplicatingThread::~DuplicatingThread() 4591{ 4592 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4593 mOutputTracks[i]->destroy(); 4594 } 4595} 4596 4597void AudioFlinger::DuplicatingThread::threadLoop_mix() 4598{ 4599 // mix buffers... 4600 if (outputsReady(outputTracks)) { 4601 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4602 } else { 4603 memset(mSinkBuffer, 0, mSinkBufferSize); 4604 } 4605 sleepTime = 0; 4606 writeFrames = mNormalFrameCount; 4607 mCurrentWriteLength = mSinkBufferSize; 4608 standbyTime = systemTime() + standbyDelay; 4609} 4610 4611void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4612{ 4613 if (sleepTime == 0) { 4614 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4615 sleepTime = activeSleepTime; 4616 } else { 4617 sleepTime = idleSleepTime; 4618 } 4619 } else if (mBytesWritten != 0) { 4620 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4621 writeFrames = mNormalFrameCount; 4622 memset(mSinkBuffer, 0, mSinkBufferSize); 4623 } else { 4624 // flush remaining overflow buffers in output tracks 4625 writeFrames = 0; 4626 } 4627 sleepTime = 0; 4628 } 4629} 4630 4631ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4632{ 4633 for (size_t i = 0; i < outputTracks.size(); i++) { 4634 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4635 // for delivery downstream as needed. This in-place conversion is safe as 4636 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4637 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4638 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4639 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4640 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4641 } 4642 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4643 } 4644 mStandby = false; 4645 return (ssize_t)mSinkBufferSize; 4646} 4647 4648void AudioFlinger::DuplicatingThread::threadLoop_standby() 4649{ 4650 // DuplicatingThread implements standby by stopping all tracks 4651 for (size_t i = 0; i < outputTracks.size(); i++) { 4652 outputTracks[i]->stop(); 4653 } 4654} 4655 4656void AudioFlinger::DuplicatingThread::saveOutputTracks() 4657{ 4658 outputTracks = mOutputTracks; 4659} 4660 4661void AudioFlinger::DuplicatingThread::clearOutputTracks() 4662{ 4663 outputTracks.clear(); 4664} 4665 4666void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4667{ 4668 Mutex::Autolock _l(mLock); 4669 // FIXME explain this formula 4670 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4671 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4672 // due to current usage case and restrictions on the AudioBufferProvider. 4673 // Actual buffer conversion is done in threadLoop_write(). 4674 // 4675 // TODO: This may change in the future, depending on multichannel 4676 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4677 OutputTrack *outputTrack = new OutputTrack(thread, 4678 this, 4679 mSampleRate, 4680 AUDIO_FORMAT_PCM_16_BIT, 4681 mChannelMask, 4682 frameCount, 4683 IPCThreadState::self()->getCallingUid()); 4684 if (outputTrack->cblk() != NULL) { 4685 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4686 mOutputTracks.add(outputTrack); 4687 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4688 updateWaitTime_l(); 4689 } 4690} 4691 4692void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4693{ 4694 Mutex::Autolock _l(mLock); 4695 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4696 if (mOutputTracks[i]->thread() == thread) { 4697 mOutputTracks[i]->destroy(); 4698 mOutputTracks.removeAt(i); 4699 updateWaitTime_l(); 4700 return; 4701 } 4702 } 4703 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4704} 4705 4706// caller must hold mLock 4707void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4708{ 4709 mWaitTimeMs = UINT_MAX; 4710 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4711 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4712 if (strong != 0) { 4713 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4714 if (waitTimeMs < mWaitTimeMs) { 4715 mWaitTimeMs = waitTimeMs; 4716 } 4717 } 4718 } 4719} 4720 4721 4722bool AudioFlinger::DuplicatingThread::outputsReady( 4723 const SortedVector< sp<OutputTrack> > &outputTracks) 4724{ 4725 for (size_t i = 0; i < outputTracks.size(); i++) { 4726 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4727 if (thread == 0) { 4728 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4729 outputTracks[i].get()); 4730 return false; 4731 } 4732 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4733 // see note at standby() declaration 4734 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4735 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4736 thread.get()); 4737 return false; 4738 } 4739 } 4740 return true; 4741} 4742 4743uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4744{ 4745 return (mWaitTimeMs * 1000) / 2; 4746} 4747 4748void AudioFlinger::DuplicatingThread::cacheParameters_l() 4749{ 4750 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4751 updateWaitTime_l(); 4752 4753 MixerThread::cacheParameters_l(); 4754} 4755 4756// ---------------------------------------------------------------------------- 4757// Record 4758// ---------------------------------------------------------------------------- 4759 4760AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4761 AudioStreamIn *input, 4762 audio_io_handle_t id, 4763 audio_devices_t outDevice, 4764 audio_devices_t inDevice 4765#ifdef TEE_SINK 4766 , const sp<NBAIO_Sink>& teeSink 4767#endif 4768 ) : 4769 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4770 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4771 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4772 mRsmpInRear(0) 4773#ifdef TEE_SINK 4774 , mTeeSink(teeSink) 4775#endif 4776 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4777 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4778 // mFastCapture below 4779 , mFastCaptureFutex(0) 4780 // mInputSource 4781 // mPipeSink 4782 // mPipeSource 4783 , mPipeFramesP2(0) 4784 // mPipeMemory 4785 // mFastCaptureNBLogWriter 4786 , mFastTrackAvail(false) 4787{ 4788 snprintf(mName, kNameLength, "AudioIn_%X", id); 4789 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4790 4791 readInputParameters_l(); 4792 4793 // create an NBAIO source for the HAL input stream, and negotiate 4794 mInputSource = new AudioStreamInSource(input->stream); 4795 size_t numCounterOffers = 0; 4796 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4797 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4798 ALOG_ASSERT(index == 0); 4799 4800 // initialize fast capture depending on configuration 4801 bool initFastCapture; 4802 switch (kUseFastCapture) { 4803 case FastCapture_Never: 4804 initFastCapture = false; 4805 break; 4806 case FastCapture_Always: 4807 initFastCapture = true; 4808 break; 4809 case FastCapture_Static: 4810 uint32_t primaryOutputSampleRate; 4811 { 4812 AutoMutex _l(audioFlinger->mHardwareLock); 4813 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4814 } 4815 initFastCapture = 4816 // either capture sample rate is same as (a reasonable) primary output sample rate 4817 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4818 (mSampleRate == primaryOutputSampleRate)) || 4819 // or primary output sample rate is unknown, and capture sample rate is reasonable 4820 ((primaryOutputSampleRate == 0) && 4821 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4822 // and the buffer size is < 10 ms 4823 (mFrameCount * 1000) / mSampleRate < 10; 4824 break; 4825 // case FastCapture_Dynamic: 4826 } 4827 4828 if (initFastCapture) { 4829 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4830 NBAIO_Format format = mInputSource->format(); 4831 size_t pipeFramesP2 = roundup(mFrameCount * 8); 4832 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4833 void *pipeBuffer; 4834 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4835 sp<IMemory> pipeMemory; 4836 if ((roHeap == 0) || 4837 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4838 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4839 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4840 goto failed; 4841 } 4842 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4843 memset(pipeBuffer, 0, pipeSize); 4844 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4845 const NBAIO_Format offers[1] = {format}; 4846 size_t numCounterOffers = 0; 4847 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4848 ALOG_ASSERT(index == 0); 4849 mPipeSink = pipe; 4850 PipeReader *pipeReader = new PipeReader(*pipe); 4851 numCounterOffers = 0; 4852 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4853 ALOG_ASSERT(index == 0); 4854 mPipeSource = pipeReader; 4855 mPipeFramesP2 = pipeFramesP2; 4856 mPipeMemory = pipeMemory; 4857 4858 // create fast capture 4859 mFastCapture = new FastCapture(); 4860 FastCaptureStateQueue *sq = mFastCapture->sq(); 4861#ifdef STATE_QUEUE_DUMP 4862 // FIXME 4863#endif 4864 FastCaptureState *state = sq->begin(); 4865 state->mCblk = NULL; 4866 state->mInputSource = mInputSource.get(); 4867 state->mInputSourceGen++; 4868 state->mPipeSink = pipe; 4869 state->mPipeSinkGen++; 4870 state->mFrameCount = mFrameCount; 4871 state->mCommand = FastCaptureState::COLD_IDLE; 4872 // already done in constructor initialization list 4873 //mFastCaptureFutex = 0; 4874 state->mColdFutexAddr = &mFastCaptureFutex; 4875 state->mColdGen++; 4876 state->mDumpState = &mFastCaptureDumpState; 4877#ifdef TEE_SINK 4878 // FIXME 4879#endif 4880 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4881 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4882 sq->end(); 4883 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4884 4885 // start the fast capture 4886 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4887 pid_t tid = mFastCapture->getTid(); 4888 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4889 if (err != 0) { 4890 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4891 kPriorityFastCapture, getpid_cached, tid, err); 4892 } 4893 4894#ifdef AUDIO_WATCHDOG 4895 // FIXME 4896#endif 4897 4898 mFastTrackAvail = true; 4899 } 4900failed: ; 4901 4902 // FIXME mNormalSource 4903} 4904 4905 4906AudioFlinger::RecordThread::~RecordThread() 4907{ 4908 if (mFastCapture != 0) { 4909 FastCaptureStateQueue *sq = mFastCapture->sq(); 4910 FastCaptureState *state = sq->begin(); 4911 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4912 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4913 if (old == -1) { 4914 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4915 } 4916 } 4917 state->mCommand = FastCaptureState::EXIT; 4918 sq->end(); 4919 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4920 mFastCapture->join(); 4921 mFastCapture.clear(); 4922 } 4923 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4924 mAudioFlinger->unregisterWriter(mNBLogWriter); 4925 delete[] mRsmpInBuffer; 4926} 4927 4928void AudioFlinger::RecordThread::onFirstRef() 4929{ 4930 run(mName, PRIORITY_URGENT_AUDIO); 4931} 4932 4933bool AudioFlinger::RecordThread::threadLoop() 4934{ 4935 nsecs_t lastWarning = 0; 4936 4937 inputStandBy(); 4938 4939reacquire_wakelock: 4940 sp<RecordTrack> activeTrack; 4941 int activeTracksGen; 4942 { 4943 Mutex::Autolock _l(mLock); 4944 size_t size = mActiveTracks.size(); 4945 activeTracksGen = mActiveTracksGen; 4946 if (size > 0) { 4947 // FIXME an arbitrary choice 4948 activeTrack = mActiveTracks[0]; 4949 acquireWakeLock_l(activeTrack->uid()); 4950 if (size > 1) { 4951 SortedVector<int> tmp; 4952 for (size_t i = 0; i < size; i++) { 4953 tmp.add(mActiveTracks[i]->uid()); 4954 } 4955 updateWakeLockUids_l(tmp); 4956 } 4957 } else { 4958 acquireWakeLock_l(-1); 4959 } 4960 } 4961 4962 // used to request a deferred sleep, to be executed later while mutex is unlocked 4963 uint32_t sleepUs = 0; 4964 4965 // loop while there is work to do 4966 for (;;) { 4967 Vector< sp<EffectChain> > effectChains; 4968 4969 // sleep with mutex unlocked 4970 if (sleepUs > 0) { 4971 usleep(sleepUs); 4972 sleepUs = 0; 4973 } 4974 4975 // activeTracks accumulates a copy of a subset of mActiveTracks 4976 Vector< sp<RecordTrack> > activeTracks; 4977 4978 // reference to the (first and only) fast track 4979 sp<RecordTrack> fastTrack; 4980 4981 { // scope for mLock 4982 Mutex::Autolock _l(mLock); 4983 4984 processConfigEvents_l(); 4985 4986 // check exitPending here because checkForNewParameters_l() and 4987 // checkForNewParameters_l() can temporarily release mLock 4988 if (exitPending()) { 4989 break; 4990 } 4991 4992 // if no active track(s), then standby and release wakelock 4993 size_t size = mActiveTracks.size(); 4994 if (size == 0) { 4995 standbyIfNotAlreadyInStandby(); 4996 // exitPending() can't become true here 4997 releaseWakeLock_l(); 4998 ALOGV("RecordThread: loop stopping"); 4999 // go to sleep 5000 mWaitWorkCV.wait(mLock); 5001 ALOGV("RecordThread: loop starting"); 5002 goto reacquire_wakelock; 5003 } 5004 5005 if (mActiveTracksGen != activeTracksGen) { 5006 activeTracksGen = mActiveTracksGen; 5007 SortedVector<int> tmp; 5008 for (size_t i = 0; i < size; i++) { 5009 tmp.add(mActiveTracks[i]->uid()); 5010 } 5011 updateWakeLockUids_l(tmp); 5012 } 5013 5014 bool doBroadcast = false; 5015 for (size_t i = 0; i < size; ) { 5016 5017 activeTrack = mActiveTracks[i]; 5018 if (activeTrack->isTerminated()) { 5019 removeTrack_l(activeTrack); 5020 mActiveTracks.remove(activeTrack); 5021 mActiveTracksGen++; 5022 size--; 5023 continue; 5024 } 5025 5026 TrackBase::track_state activeTrackState = activeTrack->mState; 5027 switch (activeTrackState) { 5028 5029 case TrackBase::PAUSING: 5030 mActiveTracks.remove(activeTrack); 5031 mActiveTracksGen++; 5032 doBroadcast = true; 5033 size--; 5034 continue; 5035 5036 case TrackBase::STARTING_1: 5037 sleepUs = 10000; 5038 i++; 5039 continue; 5040 5041 case TrackBase::STARTING_2: 5042 doBroadcast = true; 5043 mStandby = false; 5044 activeTrack->mState = TrackBase::ACTIVE; 5045 break; 5046 5047 case TrackBase::ACTIVE: 5048 break; 5049 5050 case TrackBase::IDLE: 5051 i++; 5052 continue; 5053 5054 default: 5055 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5056 } 5057 5058 activeTracks.add(activeTrack); 5059 i++; 5060 5061 if (activeTrack->isFastTrack()) { 5062 ALOG_ASSERT(!mFastTrackAvail); 5063 ALOG_ASSERT(fastTrack == 0); 5064 fastTrack = activeTrack; 5065 } 5066 } 5067 if (doBroadcast) { 5068 mStartStopCond.broadcast(); 5069 } 5070 5071 // sleep if there are no active tracks to process 5072 if (activeTracks.size() == 0) { 5073 if (sleepUs == 0) { 5074 sleepUs = kRecordThreadSleepUs; 5075 } 5076 continue; 5077 } 5078 sleepUs = 0; 5079 5080 lockEffectChains_l(effectChains); 5081 } 5082 5083 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5084 5085 size_t size = effectChains.size(); 5086 for (size_t i = 0; i < size; i++) { 5087 // thread mutex is not locked, but effect chain is locked 5088 effectChains[i]->process_l(); 5089 } 5090 5091 // Start the fast capture if it's not already running 5092 if (mFastCapture != 0) { 5093 FastCaptureStateQueue *sq = mFastCapture->sq(); 5094 FastCaptureState *state = sq->begin(); 5095 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5096 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5097 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5098 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5099 if (old == -1) { 5100 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5101 } 5102 } 5103 state->mCommand = FastCaptureState::READ_WRITE; 5104#if 0 // FIXME 5105 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5106 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5107#endif 5108 state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL; 5109 sq->end(); 5110 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5111#if 0 5112 if (kUseFastCapture == FastCapture_Dynamic) { 5113 mNormalSource = mPipeSource; 5114 } 5115#endif 5116 } else { 5117 sq->end(false /*didModify*/); 5118 } 5119 } 5120 5121 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5122 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5123 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5124 // If destination is non-contiguous, first read past the nominal end of buffer, then 5125 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5126 5127 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5128 ssize_t framesRead; 5129 5130 // If an NBAIO source is present, use it to read the normal capture's data 5131 if (mPipeSource != 0) { 5132 size_t framesToRead = mBufferSize / mFrameSize; 5133 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5134 framesToRead, AudioBufferProvider::kInvalidPTS); 5135 if (framesRead == 0) { 5136 // since pipe is non-blocking, simulate blocking input 5137 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5138 } 5139 // otherwise use the HAL / AudioStreamIn directly 5140 } else { 5141 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5142 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5143 if (bytesRead < 0) { 5144 framesRead = bytesRead; 5145 } else { 5146 framesRead = bytesRead / mFrameSize; 5147 } 5148 } 5149 5150 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5151 ALOGE("read failed: framesRead=%d", framesRead); 5152 // Force input into standby so that it tries to recover at next read attempt 5153 inputStandBy(); 5154 sleepUs = kRecordThreadSleepUs; 5155 } 5156 if (framesRead <= 0) { 5157 goto unlock; 5158 } 5159 ALOG_ASSERT(framesRead > 0); 5160 5161 if (mTeeSink != 0) { 5162 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5163 } 5164 // If destination is non-contiguous, we now correct for reading past end of buffer. 5165 { 5166 size_t part1 = mRsmpInFramesP2 - rear; 5167 if ((size_t) framesRead > part1) { 5168 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5169 (framesRead - part1) * mFrameSize); 5170 } 5171 } 5172 rear = mRsmpInRear += framesRead; 5173 5174 size = activeTracks.size(); 5175 // loop over each active track 5176 for (size_t i = 0; i < size; i++) { 5177 activeTrack = activeTracks[i]; 5178 5179 // skip fast tracks, as those are handled directly by FastCapture 5180 if (activeTrack->isFastTrack()) { 5181 continue; 5182 } 5183 5184 enum { 5185 OVERRUN_UNKNOWN, 5186 OVERRUN_TRUE, 5187 OVERRUN_FALSE 5188 } overrun = OVERRUN_UNKNOWN; 5189 5190 // loop over getNextBuffer to handle circular sink 5191 for (;;) { 5192 5193 activeTrack->mSink.frameCount = ~0; 5194 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5195 size_t framesOut = activeTrack->mSink.frameCount; 5196 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5197 5198 int32_t front = activeTrack->mRsmpInFront; 5199 ssize_t filled = rear - front; 5200 size_t framesIn; 5201 5202 if (filled < 0) { 5203 // should not happen, but treat like a massive overrun and re-sync 5204 framesIn = 0; 5205 activeTrack->mRsmpInFront = rear; 5206 overrun = OVERRUN_TRUE; 5207 } else if ((size_t) filled <= mRsmpInFrames) { 5208 framesIn = (size_t) filled; 5209 } else { 5210 // client is not keeping up with server, but give it latest data 5211 framesIn = mRsmpInFrames; 5212 activeTrack->mRsmpInFront = front = rear - framesIn; 5213 overrun = OVERRUN_TRUE; 5214 } 5215 5216 if (framesOut == 0 || framesIn == 0) { 5217 break; 5218 } 5219 5220 if (activeTrack->mResampler == NULL) { 5221 // no resampling 5222 if (framesIn > framesOut) { 5223 framesIn = framesOut; 5224 } else { 5225 framesOut = framesIn; 5226 } 5227 int8_t *dst = activeTrack->mSink.i8; 5228 while (framesIn > 0) { 5229 front &= mRsmpInFramesP2 - 1; 5230 size_t part1 = mRsmpInFramesP2 - front; 5231 if (part1 > framesIn) { 5232 part1 = framesIn; 5233 } 5234 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5235 if (mChannelCount == activeTrack->mChannelCount) { 5236 memcpy(dst, src, part1 * mFrameSize); 5237 } else if (mChannelCount == 1) { 5238 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5239 part1); 5240 } else { 5241 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5242 part1); 5243 } 5244 dst += part1 * activeTrack->mFrameSize; 5245 front += part1; 5246 framesIn -= part1; 5247 } 5248 activeTrack->mRsmpInFront += framesOut; 5249 5250 } else { 5251 // resampling 5252 // FIXME framesInNeeded should really be part of resampler API, and should 5253 // depend on the SRC ratio 5254 // to keep mRsmpInBuffer full so resampler always has sufficient input 5255 size_t framesInNeeded; 5256 // FIXME only re-calculate when it changes, and optimize for common ratios 5257 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 5258 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 5259 framesInNeeded = ceil(framesOut * inOverOut) + 1; 5260 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5261 framesInNeeded, framesOut, inOverOut); 5262 // Although we theoretically have framesIn in circular buffer, some of those are 5263 // unreleased frames, and thus must be discounted for purpose of budgeting. 5264 size_t unreleased = activeTrack->mRsmpInUnrel; 5265 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5266 if (framesIn < framesInNeeded) { 5267 ALOGV("not enough to resample: have %u frames in but need %u in to " 5268 "produce %u out given in/out ratio of %.4g", 5269 framesIn, framesInNeeded, framesOut, inOverOut); 5270 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 5271 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5272 if (newFramesOut == 0) { 5273 break; 5274 } 5275 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 5276 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5277 framesInNeeded, newFramesOut, outOverIn); 5278 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5279 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5280 "given in/out ratio of %.4g", 5281 framesIn, framesInNeeded, newFramesOut, inOverOut); 5282 framesOut = newFramesOut; 5283 } else { 5284 ALOGV("success 1: have %u in and need %u in to produce %u out " 5285 "given in/out ratio of %.4g", 5286 framesIn, framesInNeeded, framesOut, inOverOut); 5287 } 5288 5289 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5290 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5291 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5292 delete[] activeTrack->mRsmpOutBuffer; 5293 // resampler always outputs stereo 5294 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5295 activeTrack->mRsmpOutFrameCount = framesOut; 5296 } 5297 5298 // resampler accumulates, but we only have one source track 5299 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5300 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5301 // FIXME how about having activeTrack implement this interface itself? 5302 activeTrack->mResamplerBufferProvider 5303 /*this*/ /* AudioBufferProvider* */); 5304 // ditherAndClamp() works as long as all buffers returned by 5305 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5306 if (activeTrack->mChannelCount == 1) { 5307 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5308 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5309 framesOut); 5310 // the resampler always outputs stereo samples: 5311 // do post stereo to mono conversion 5312 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5313 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5314 } else { 5315 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5316 activeTrack->mRsmpOutBuffer, framesOut); 5317 } 5318 // now done with mRsmpOutBuffer 5319 5320 } 5321 5322 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5323 overrun = OVERRUN_FALSE; 5324 } 5325 5326 if (activeTrack->mFramesToDrop == 0) { 5327 if (framesOut > 0) { 5328 activeTrack->mSink.frameCount = framesOut; 5329 activeTrack->releaseBuffer(&activeTrack->mSink); 5330 } 5331 } else { 5332 // FIXME could do a partial drop of framesOut 5333 if (activeTrack->mFramesToDrop > 0) { 5334 activeTrack->mFramesToDrop -= framesOut; 5335 if (activeTrack->mFramesToDrop <= 0) { 5336 activeTrack->clearSyncStartEvent(); 5337 } 5338 } else { 5339 activeTrack->mFramesToDrop += framesOut; 5340 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5341 activeTrack->mSyncStartEvent->isCancelled()) { 5342 ALOGW("Synced record %s, session %d, trigger session %d", 5343 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5344 activeTrack->sessionId(), 5345 (activeTrack->mSyncStartEvent != 0) ? 5346 activeTrack->mSyncStartEvent->triggerSession() : 0); 5347 activeTrack->clearSyncStartEvent(); 5348 } 5349 } 5350 } 5351 5352 if (framesOut == 0) { 5353 break; 5354 } 5355 } 5356 5357 switch (overrun) { 5358 case OVERRUN_TRUE: 5359 // client isn't retrieving buffers fast enough 5360 if (!activeTrack->setOverflow()) { 5361 nsecs_t now = systemTime(); 5362 // FIXME should lastWarning per track? 5363 if ((now - lastWarning) > kWarningThrottleNs) { 5364 ALOGW("RecordThread: buffer overflow"); 5365 lastWarning = now; 5366 } 5367 } 5368 break; 5369 case OVERRUN_FALSE: 5370 activeTrack->clearOverflow(); 5371 break; 5372 case OVERRUN_UNKNOWN: 5373 break; 5374 } 5375 5376 } 5377 5378unlock: 5379 // enable changes in effect chain 5380 unlockEffectChains(effectChains); 5381 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5382 } 5383 5384 standbyIfNotAlreadyInStandby(); 5385 5386 { 5387 Mutex::Autolock _l(mLock); 5388 for (size_t i = 0; i < mTracks.size(); i++) { 5389 sp<RecordTrack> track = mTracks[i]; 5390 track->invalidate(); 5391 } 5392 mActiveTracks.clear(); 5393 mActiveTracksGen++; 5394 mStartStopCond.broadcast(); 5395 } 5396 5397 releaseWakeLock(); 5398 5399 ALOGV("RecordThread %p exiting", this); 5400 return false; 5401} 5402 5403void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5404{ 5405 if (!mStandby) { 5406 inputStandBy(); 5407 mStandby = true; 5408 } 5409} 5410 5411void AudioFlinger::RecordThread::inputStandBy() 5412{ 5413 // Idle the fast capture if it's currently running 5414 if (mFastCapture != 0) { 5415 FastCaptureStateQueue *sq = mFastCapture->sq(); 5416 FastCaptureState *state = sq->begin(); 5417 if (!(state->mCommand & FastCaptureState::IDLE)) { 5418 state->mCommand = FastCaptureState::COLD_IDLE; 5419 state->mColdFutexAddr = &mFastCaptureFutex; 5420 state->mColdGen++; 5421 mFastCaptureFutex = 0; 5422 sq->end(); 5423 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5424 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5425#if 0 5426 if (kUseFastCapture == FastCapture_Dynamic) { 5427 // FIXME 5428 } 5429#endif 5430#ifdef AUDIO_WATCHDOG 5431 // FIXME 5432#endif 5433 } else { 5434 sq->end(false /*didModify*/); 5435 } 5436 } 5437 mInput->stream->common.standby(&mInput->stream->common); 5438} 5439 5440// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5441sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5442 const sp<AudioFlinger::Client>& client, 5443 uint32_t sampleRate, 5444 audio_format_t format, 5445 audio_channel_mask_t channelMask, 5446 size_t *pFrameCount, 5447 int sessionId, 5448 size_t *notificationFrames, 5449 int uid, 5450 IAudioFlinger::track_flags_t *flags, 5451 pid_t tid, 5452 status_t *status) 5453{ 5454 size_t frameCount = *pFrameCount; 5455 sp<RecordTrack> track; 5456 status_t lStatus; 5457 5458 // client expresses a preference for FAST, but we get the final say 5459 if (*flags & IAudioFlinger::TRACK_FAST) { 5460 if ( 5461 // use case: callback handler and frame count is default or at least as large as HAL 5462 ( 5463 (tid != -1) && 5464 ((frameCount == 0) /*|| 5465 // FIXME must be equal to pipe depth, so don't allow it to be specified by client 5466 // FIXME not necessarily true, should be native frame count for native SR! 5467 (frameCount >= mFrameCount)*/) 5468 ) && 5469 // PCM data 5470 audio_is_linear_pcm(format) && 5471 // native format 5472 (format == mFormat) && 5473 // mono or stereo 5474 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5475 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5476 // native channel mask 5477 (channelMask == mChannelMask) && 5478 // native hardware sample rate 5479 (sampleRate == mSampleRate) && 5480 // record thread has an associated fast capture 5481 hasFastCapture() && 5482 // there are sufficient fast track slots available 5483 mFastTrackAvail 5484 ) { 5485 // if frameCount not specified, then it defaults to pipe frame count 5486 if (frameCount == 0) { 5487 frameCount = mPipeFramesP2; 5488 } 5489 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5490 frameCount, mFrameCount); 5491 } else { 5492 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5493 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5494 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5495 frameCount, mFrameCount, format, 5496 audio_is_linear_pcm(format), 5497 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid, mFastTrackAvail); 5498 *flags &= ~IAudioFlinger::TRACK_FAST; 5499 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5500 // For compatibility with AudioRecord calculation, buffer depth is forced 5501 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5502 // This is probably too conservative, but legacy application code may depend on it. 5503 // If you change this calculation, also review the start threshold which is related. 5504 // FIXME It's not clear how input latency actually matters. Perhaps this should be 0. 5505 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5506 size_t mNormalFrameCount = 2048; // FIXME 5507 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5508 if (minBufCount < 2) { 5509 minBufCount = 2; 5510 } 5511 size_t minFrameCount = mNormalFrameCount * minBufCount; 5512 if (frameCount < minFrameCount) { 5513 frameCount = minFrameCount; 5514 } 5515 } 5516 } 5517 *pFrameCount = frameCount; 5518 *notificationFrames = 0; // FIXME implement 5519 5520 lStatus = initCheck(); 5521 if (lStatus != NO_ERROR) { 5522 ALOGE("createRecordTrack_l() audio driver not initialized"); 5523 goto Exit; 5524 } 5525 5526 { // scope for mLock 5527 Mutex::Autolock _l(mLock); 5528 5529 track = new RecordTrack(this, client, sampleRate, 5530 format, channelMask, frameCount, sessionId, uid, 5531 *flags); 5532 5533 lStatus = track->initCheck(); 5534 if (lStatus != NO_ERROR) { 5535 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5536 // track must be cleared from the caller as the caller has the AF lock 5537 goto Exit; 5538 } 5539 mTracks.add(track); 5540 5541 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5542 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5543 mAudioFlinger->btNrecIsOff(); 5544 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5545 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5546 5547 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5548 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5549 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5550 // so ask activity manager to do this on our behalf 5551 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5552 } 5553 } 5554 5555 lStatus = NO_ERROR; 5556 5557Exit: 5558 *status = lStatus; 5559 return track; 5560} 5561 5562status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5563 AudioSystem::sync_event_t event, 5564 int triggerSession) 5565{ 5566 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5567 sp<ThreadBase> strongMe = this; 5568 status_t status = NO_ERROR; 5569 5570 if (event == AudioSystem::SYNC_EVENT_NONE) { 5571 recordTrack->clearSyncStartEvent(); 5572 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5573 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5574 triggerSession, 5575 recordTrack->sessionId(), 5576 syncStartEventCallback, 5577 recordTrack); 5578 // Sync event can be cancelled by the trigger session if the track is not in a 5579 // compatible state in which case we start record immediately 5580 if (recordTrack->mSyncStartEvent->isCancelled()) { 5581 recordTrack->clearSyncStartEvent(); 5582 } else { 5583 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5584 recordTrack->mFramesToDrop = - 5585 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5586 } 5587 } 5588 5589 { 5590 // This section is a rendezvous between binder thread executing start() and RecordThread 5591 AutoMutex lock(mLock); 5592 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5593 if (recordTrack->mState == TrackBase::PAUSING) { 5594 ALOGV("active record track PAUSING -> ACTIVE"); 5595 recordTrack->mState = TrackBase::ACTIVE; 5596 } else { 5597 ALOGV("active record track state %d", recordTrack->mState); 5598 } 5599 return status; 5600 } 5601 5602 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5603 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5604 // or using a separate command thread 5605 recordTrack->mState = TrackBase::STARTING_1; 5606 mActiveTracks.add(recordTrack); 5607 mActiveTracksGen++; 5608 mLock.unlock(); 5609 status_t status = AudioSystem::startInput(mId); 5610 mLock.lock(); 5611 // FIXME should verify that recordTrack is still in mActiveTracks 5612 if (status != NO_ERROR) { 5613 mActiveTracks.remove(recordTrack); 5614 mActiveTracksGen++; 5615 recordTrack->clearSyncStartEvent(); 5616 return status; 5617 } 5618 // Catch up with current buffer indices if thread is already running. 5619 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5620 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5621 // see previously buffered data before it called start(), but with greater risk of overrun. 5622 5623 recordTrack->mRsmpInFront = mRsmpInRear; 5624 recordTrack->mRsmpInUnrel = 0; 5625 // FIXME why reset? 5626 if (recordTrack->mResampler != NULL) { 5627 recordTrack->mResampler->reset(); 5628 } 5629 recordTrack->mState = TrackBase::STARTING_2; 5630 // signal thread to start 5631 mWaitWorkCV.broadcast(); 5632 if (mActiveTracks.indexOf(recordTrack) < 0) { 5633 ALOGV("Record failed to start"); 5634 status = BAD_VALUE; 5635 goto startError; 5636 } 5637 return status; 5638 } 5639 5640startError: 5641 AudioSystem::stopInput(mId); 5642 recordTrack->clearSyncStartEvent(); 5643 // FIXME I wonder why we do not reset the state here? 5644 return status; 5645} 5646 5647void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5648{ 5649 sp<SyncEvent> strongEvent = event.promote(); 5650 5651 if (strongEvent != 0) { 5652 sp<RefBase> ptr = strongEvent->cookie().promote(); 5653 if (ptr != 0) { 5654 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5655 recordTrack->handleSyncStartEvent(strongEvent); 5656 } 5657 } 5658} 5659 5660bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5661 ALOGV("RecordThread::stop"); 5662 AutoMutex _l(mLock); 5663 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5664 return false; 5665 } 5666 // note that threadLoop may still be processing the track at this point [without lock] 5667 recordTrack->mState = TrackBase::PAUSING; 5668 // do not wait for mStartStopCond if exiting 5669 if (exitPending()) { 5670 return true; 5671 } 5672 // FIXME incorrect usage of wait: no explicit predicate or loop 5673 mStartStopCond.wait(mLock); 5674 // if we have been restarted, recordTrack is in mActiveTracks here 5675 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5676 ALOGV("Record stopped OK"); 5677 return true; 5678 } 5679 return false; 5680} 5681 5682bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5683{ 5684 return false; 5685} 5686 5687status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5688{ 5689#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5690 if (!isValidSyncEvent(event)) { 5691 return BAD_VALUE; 5692 } 5693 5694 int eventSession = event->triggerSession(); 5695 status_t ret = NAME_NOT_FOUND; 5696 5697 Mutex::Autolock _l(mLock); 5698 5699 for (size_t i = 0; i < mTracks.size(); i++) { 5700 sp<RecordTrack> track = mTracks[i]; 5701 if (eventSession == track->sessionId()) { 5702 (void) track->setSyncEvent(event); 5703 ret = NO_ERROR; 5704 } 5705 } 5706 return ret; 5707#else 5708 return BAD_VALUE; 5709#endif 5710} 5711 5712// destroyTrack_l() must be called with ThreadBase::mLock held 5713void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5714{ 5715 track->terminate(); 5716 track->mState = TrackBase::STOPPED; 5717 // active tracks are removed by threadLoop() 5718 if (mActiveTracks.indexOf(track) < 0) { 5719 removeTrack_l(track); 5720 } 5721} 5722 5723void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5724{ 5725 mTracks.remove(track); 5726 // need anything related to effects here? 5727 if (track->isFastTrack()) { 5728 ALOG_ASSERT(!mFastTrackAvail); 5729 mFastTrackAvail = true; 5730 } 5731} 5732 5733void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5734{ 5735 dumpInternals(fd, args); 5736 dumpTracks(fd, args); 5737 dumpEffectChains(fd, args); 5738} 5739 5740void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5741{ 5742 dprintf(fd, "\nInput thread %p:\n", this); 5743 5744 if (mActiveTracks.size() > 0) { 5745 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5746 } else { 5747 dprintf(fd, " No active record clients\n"); 5748 } 5749 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5750 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5751 5752 dumpBase(fd, args); 5753} 5754 5755void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5756{ 5757 const size_t SIZE = 256; 5758 char buffer[SIZE]; 5759 String8 result; 5760 5761 size_t numtracks = mTracks.size(); 5762 size_t numactive = mActiveTracks.size(); 5763 size_t numactiveseen = 0; 5764 dprintf(fd, " %d Tracks", numtracks); 5765 if (numtracks) { 5766 dprintf(fd, " of which %d are active\n", numactive); 5767 RecordTrack::appendDumpHeader(result); 5768 for (size_t i = 0; i < numtracks ; ++i) { 5769 sp<RecordTrack> track = mTracks[i]; 5770 if (track != 0) { 5771 bool active = mActiveTracks.indexOf(track) >= 0; 5772 if (active) { 5773 numactiveseen++; 5774 } 5775 track->dump(buffer, SIZE, active); 5776 result.append(buffer); 5777 } 5778 } 5779 } else { 5780 dprintf(fd, "\n"); 5781 } 5782 5783 if (numactiveseen != numactive) { 5784 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5785 " not in the track list\n"); 5786 result.append(buffer); 5787 RecordTrack::appendDumpHeader(result); 5788 for (size_t i = 0; i < numactive; ++i) { 5789 sp<RecordTrack> track = mActiveTracks[i]; 5790 if (mTracks.indexOf(track) < 0) { 5791 track->dump(buffer, SIZE, true); 5792 result.append(buffer); 5793 } 5794 } 5795 5796 } 5797 write(fd, result.string(), result.size()); 5798} 5799 5800// AudioBufferProvider interface 5801status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5802 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5803{ 5804 RecordTrack *activeTrack = mRecordTrack; 5805 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5806 if (threadBase == 0) { 5807 buffer->frameCount = 0; 5808 buffer->raw = NULL; 5809 return NOT_ENOUGH_DATA; 5810 } 5811 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5812 int32_t rear = recordThread->mRsmpInRear; 5813 int32_t front = activeTrack->mRsmpInFront; 5814 ssize_t filled = rear - front; 5815 // FIXME should not be P2 (don't want to increase latency) 5816 // FIXME if client not keeping up, discard 5817 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5818 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5819 front &= recordThread->mRsmpInFramesP2 - 1; 5820 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5821 if (part1 > (size_t) filled) { 5822 part1 = filled; 5823 } 5824 size_t ask = buffer->frameCount; 5825 ALOG_ASSERT(ask > 0); 5826 if (part1 > ask) { 5827 part1 = ask; 5828 } 5829 if (part1 == 0) { 5830 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5831 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5832 buffer->raw = NULL; 5833 buffer->frameCount = 0; 5834 activeTrack->mRsmpInUnrel = 0; 5835 return NOT_ENOUGH_DATA; 5836 } 5837 5838 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5839 buffer->frameCount = part1; 5840 activeTrack->mRsmpInUnrel = part1; 5841 return NO_ERROR; 5842} 5843 5844// AudioBufferProvider interface 5845void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5846 AudioBufferProvider::Buffer* buffer) 5847{ 5848 RecordTrack *activeTrack = mRecordTrack; 5849 size_t stepCount = buffer->frameCount; 5850 if (stepCount == 0) { 5851 return; 5852 } 5853 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5854 activeTrack->mRsmpInUnrel -= stepCount; 5855 activeTrack->mRsmpInFront += stepCount; 5856 buffer->raw = NULL; 5857 buffer->frameCount = 0; 5858} 5859 5860bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5861 status_t& status) 5862{ 5863 bool reconfig = false; 5864 5865 status = NO_ERROR; 5866 5867 audio_format_t reqFormat = mFormat; 5868 uint32_t samplingRate = mSampleRate; 5869 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5870 5871 AudioParameter param = AudioParameter(keyValuePair); 5872 int value; 5873 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5874 // channel count change can be requested. Do we mandate the first client defines the 5875 // HAL sampling rate and channel count or do we allow changes on the fly? 5876 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5877 samplingRate = value; 5878 reconfig = true; 5879 } 5880 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5881 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5882 status = BAD_VALUE; 5883 } else { 5884 reqFormat = (audio_format_t) value; 5885 reconfig = true; 5886 } 5887 } 5888 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5889 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5890 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5891 status = BAD_VALUE; 5892 } else { 5893 channelMask = mask; 5894 reconfig = true; 5895 } 5896 } 5897 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5898 // do not accept frame count changes if tracks are open as the track buffer 5899 // size depends on frame count and correct behavior would not be guaranteed 5900 // if frame count is changed after track creation 5901 if (mActiveTracks.size() > 0) { 5902 status = INVALID_OPERATION; 5903 } else { 5904 reconfig = true; 5905 } 5906 } 5907 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5908 // forward device change to effects that have requested to be 5909 // aware of attached audio device. 5910 for (size_t i = 0; i < mEffectChains.size(); i++) { 5911 mEffectChains[i]->setDevice_l(value); 5912 } 5913 5914 // store input device and output device but do not forward output device to audio HAL. 5915 // Note that status is ignored by the caller for output device 5916 // (see AudioFlinger::setParameters() 5917 if (audio_is_output_devices(value)) { 5918 mOutDevice = value; 5919 status = BAD_VALUE; 5920 } else { 5921 mInDevice = value; 5922 // disable AEC and NS if the device is a BT SCO headset supporting those 5923 // pre processings 5924 if (mTracks.size() > 0) { 5925 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5926 mAudioFlinger->btNrecIsOff(); 5927 for (size_t i = 0; i < mTracks.size(); i++) { 5928 sp<RecordTrack> track = mTracks[i]; 5929 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5930 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5931 } 5932 } 5933 } 5934 } 5935 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5936 mAudioSource != (audio_source_t)value) { 5937 // forward device change to effects that have requested to be 5938 // aware of attached audio device. 5939 for (size_t i = 0; i < mEffectChains.size(); i++) { 5940 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5941 } 5942 mAudioSource = (audio_source_t)value; 5943 } 5944 5945 if (status == NO_ERROR) { 5946 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5947 keyValuePair.string()); 5948 if (status == INVALID_OPERATION) { 5949 inputStandBy(); 5950 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5951 keyValuePair.string()); 5952 } 5953 if (reconfig) { 5954 if (status == BAD_VALUE && 5955 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5956 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5957 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5958 <= (2 * samplingRate)) && 5959 audio_channel_count_from_in_mask( 5960 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5961 (channelMask == AUDIO_CHANNEL_IN_MONO || 5962 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5963 status = NO_ERROR; 5964 } 5965 if (status == NO_ERROR) { 5966 readInputParameters_l(); 5967 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5968 } 5969 } 5970 } 5971 5972 return reconfig; 5973} 5974 5975String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5976{ 5977 Mutex::Autolock _l(mLock); 5978 if (initCheck() != NO_ERROR) { 5979 return String8(); 5980 } 5981 5982 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5983 const String8 out_s8(s); 5984 free(s); 5985 return out_s8; 5986} 5987 5988void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5989 AudioSystem::OutputDescriptor desc; 5990 const void *param2 = NULL; 5991 5992 switch (event) { 5993 case AudioSystem::INPUT_OPENED: 5994 case AudioSystem::INPUT_CONFIG_CHANGED: 5995 desc.channelMask = mChannelMask; 5996 desc.samplingRate = mSampleRate; 5997 desc.format = mFormat; 5998 desc.frameCount = mFrameCount; 5999 desc.latency = 0; 6000 param2 = &desc; 6001 break; 6002 6003 case AudioSystem::INPUT_CLOSED: 6004 default: 6005 break; 6006 } 6007 mAudioFlinger->audioConfigChanged(event, mId, param2); 6008} 6009 6010void AudioFlinger::RecordThread::readInputParameters_l() 6011{ 6012 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6013 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6014 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6015 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6016 mFormat = mHALFormat; 6017 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6018 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6019 } 6020 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6021 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6022 mFrameCount = mBufferSize / mFrameSize; 6023 // This is the formula for calculating the temporary buffer size. 6024 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6025 // 1 full output buffer, regardless of the alignment of the available input. 6026 // The value is somewhat arbitrary, and could probably be even larger. 6027 // A larger value should allow more old data to be read after a track calls start(), 6028 // without increasing latency. 6029 mRsmpInFrames = mFrameCount * 7; 6030 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6031 delete[] mRsmpInBuffer; 6032 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6033 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6034 6035 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6036 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6037} 6038 6039uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6040{ 6041 Mutex::Autolock _l(mLock); 6042 if (initCheck() != NO_ERROR) { 6043 return 0; 6044 } 6045 6046 return mInput->stream->get_input_frames_lost(mInput->stream); 6047} 6048 6049uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6050{ 6051 Mutex::Autolock _l(mLock); 6052 uint32_t result = 0; 6053 if (getEffectChain_l(sessionId) != 0) { 6054 result = EFFECT_SESSION; 6055 } 6056 6057 for (size_t i = 0; i < mTracks.size(); ++i) { 6058 if (sessionId == mTracks[i]->sessionId()) { 6059 result |= TRACK_SESSION; 6060 break; 6061 } 6062 } 6063 6064 return result; 6065} 6066 6067KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6068{ 6069 KeyedVector<int, bool> ids; 6070 Mutex::Autolock _l(mLock); 6071 for (size_t j = 0; j < mTracks.size(); ++j) { 6072 sp<RecordThread::RecordTrack> track = mTracks[j]; 6073 int sessionId = track->sessionId(); 6074 if (ids.indexOfKey(sessionId) < 0) { 6075 ids.add(sessionId, true); 6076 } 6077 } 6078 return ids; 6079} 6080 6081AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6082{ 6083 Mutex::Autolock _l(mLock); 6084 AudioStreamIn *input = mInput; 6085 mInput = NULL; 6086 return input; 6087} 6088 6089// this method must always be called either with ThreadBase mLock held or inside the thread loop 6090audio_stream_t* AudioFlinger::RecordThread::stream() const 6091{ 6092 if (mInput == NULL) { 6093 return NULL; 6094 } 6095 return &mInput->stream->common; 6096} 6097 6098status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6099{ 6100 // only one chain per input thread 6101 if (mEffectChains.size() != 0) { 6102 return INVALID_OPERATION; 6103 } 6104 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6105 6106 chain->setInBuffer(NULL); 6107 chain->setOutBuffer(NULL); 6108 6109 checkSuspendOnAddEffectChain_l(chain); 6110 6111 mEffectChains.add(chain); 6112 6113 return NO_ERROR; 6114} 6115 6116size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6117{ 6118 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6119 ALOGW_IF(mEffectChains.size() != 1, 6120 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6121 chain.get(), mEffectChains.size(), this); 6122 if (mEffectChains.size() == 1) { 6123 mEffectChains.removeAt(0); 6124 } 6125 return 0; 6126} 6127 6128status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6129 audio_patch_handle_t *handle) 6130{ 6131 status_t status = NO_ERROR; 6132 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6133 // store new device and send to effects 6134 mInDevice = patch->sources[0].ext.device.type; 6135 for (size_t i = 0; i < mEffectChains.size(); i++) { 6136 mEffectChains[i]->setDevice_l(mInDevice); 6137 } 6138 6139 // disable AEC and NS if the device is a BT SCO headset supporting those 6140 // pre processings 6141 if (mTracks.size() > 0) { 6142 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6143 mAudioFlinger->btNrecIsOff(); 6144 for (size_t i = 0; i < mTracks.size(); i++) { 6145 sp<RecordTrack> track = mTracks[i]; 6146 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6147 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6148 } 6149 } 6150 6151 // store new source and send to effects 6152 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6153 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6154 for (size_t i = 0; i < mEffectChains.size(); i++) { 6155 mEffectChains[i]->setAudioSource_l(mAudioSource); 6156 } 6157 } 6158 6159 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6160 status = hwDevice->create_audio_patch(hwDevice, 6161 patch->num_sources, 6162 patch->sources, 6163 patch->num_sinks, 6164 patch->sinks, 6165 handle); 6166 } else { 6167 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6168 } 6169 return status; 6170} 6171 6172status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6173{ 6174 status_t status = NO_ERROR; 6175 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6176 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6177 status = hwDevice->release_audio_patch(hwDevice, handle); 6178 } else { 6179 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6180 } 6181 return status; 6182} 6183 6184 6185}; // namespace android 6186