Threads.cpp revision 49d00ad9164ea5ce48c85765a2b6460d9b457d38
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 <media/AudioResamplerPublic.h> 30#include <utils/Log.h> 31#include <utils/Trace.h> 32 33#include <private/media/AudioTrackShared.h> 34#include <hardware/audio.h> 35#include <audio_effects/effect_ns.h> 36#include <audio_effects/effect_aec.h> 37#include <audio_utils/primitives.h> 38#include <audio_utils/format.h> 39#include <audio_utils/minifloat.h> 40 41// NBAIO implementations 42#include <media/nbaio/AudioStreamInSource.h> 43#include <media/nbaio/AudioStreamOutSink.h> 44#include <media/nbaio/MonoPipe.h> 45#include <media/nbaio/MonoPipeReader.h> 46#include <media/nbaio/Pipe.h> 47#include <media/nbaio/PipeReader.h> 48#include <media/nbaio/SourceAudioBufferProvider.h> 49 50#include <powermanager/PowerManager.h> 51 52#include <common_time/cc_helper.h> 53#include <common_time/local_clock.h> 54 55#include "AudioFlinger.h" 56#include "AudioMixer.h" 57#include "FastMixer.h" 58#include "FastCapture.h" 59#include "ServiceUtilities.h" 60#include "SchedulingPolicyService.h" 61 62#ifdef ADD_BATTERY_DATA 63#include <media/IMediaPlayerService.h> 64#include <media/IMediaDeathNotifier.h> 65#endif 66 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72// ---------------------------------------------------------------------------- 73 74// Note: the following macro is used for extremely verbose logging message. In 75// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 76// 0; but one side effect of this is to turn all LOGV's as well. Some messages 77// are so verbose that we want to suppress them even when we have ALOG_ASSERT 78// turned on. Do not uncomment the #def below unless you really know what you 79// are doing and want to see all of the extremely verbose messages. 80//#define VERY_VERY_VERBOSE_LOGGING 81#ifdef VERY_VERY_VERBOSE_LOGGING 82#define ALOGVV ALOGV 83#else 84#define ALOGVV(a...) do { } while(0) 85#endif 86 87#define max(a, b) ((a) > (b) ? (a) : (b)) 88 89namespace android { 90 91// retry counts for buffer fill timeout 92// 50 * ~20msecs = 1 second 93static const int8_t kMaxTrackRetries = 50; 94static const int8_t kMaxTrackStartupRetries = 50; 95// allow less retry attempts on direct output thread. 96// direct outputs can be a scarce resource in audio hardware and should 97// be released as quickly as possible. 98static const int8_t kMaxTrackRetriesDirect = 2; 99 100// don't warn about blocked writes or record buffer overflows more often than this 101static const nsecs_t kWarningThrottleNs = seconds(5); 102 103// RecordThread loop sleep time upon application overrun or audio HAL read error 104static const int kRecordThreadSleepUs = 5000; 105 106// maximum time to wait in sendConfigEvent_l() for a status to be received 107static const nsecs_t kConfigEventTimeoutNs = seconds(2); 108 109// minimum sleep time for the mixer thread loop when tracks are active but in underrun 110static const uint32_t kMinThreadSleepTimeUs = 5000; 111// maximum divider applied to the active sleep time in the mixer thread loop 112static const uint32_t kMaxThreadSleepTimeShift = 2; 113 114// minimum normal sink buffer size, expressed in milliseconds rather than frames 115static const uint32_t kMinNormalSinkBufferSizeMs = 20; 116// maximum normal sink buffer size 117static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 118 119// Offloaded output thread standby delay: allows track transition without going to standby 120static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 121 122// Whether to use fast mixer 123static const enum { 124 FastMixer_Never, // never initialize or use: for debugging only 125 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 126 // normal mixer multiplier is 1 127 FastMixer_Static, // initialize if needed, then use all the time if initialized, 128 // multiplier is calculated based on min & max normal mixer buffer size 129 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 130 // multiplier is calculated based on min & max normal mixer buffer size 131 // FIXME for FastMixer_Dynamic: 132 // Supporting this option will require fixing HALs that can't handle large writes. 133 // For example, one HAL implementation returns an error from a large write, 134 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 135 // We could either fix the HAL implementations, or provide a wrapper that breaks 136 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 137} kUseFastMixer = FastMixer_Static; 138 139// Whether to use fast capture 140static const enum { 141 FastCapture_Never, // never initialize or use: for debugging only 142 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 143 FastCapture_Static, // initialize if needed, then use all the time if initialized 144} kUseFastCapture = FastCapture_Static; 145 146// Priorities for requestPriority 147static const int kPriorityAudioApp = 2; 148static const int kPriorityFastMixer = 3; 149static const int kPriorityFastCapture = 3; 150 151// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 152// for the track. The client then sub-divides this into smaller buffers for its use. 153// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 154// So for now we just assume that client is double-buffered for fast tracks. 155// FIXME It would be better for client to tell AudioFlinger the value of N, 156// so AudioFlinger could allocate the right amount of memory. 157// See the client's minBufCount and mNotificationFramesAct calculations for details. 158 159// This is the default value, if not specified by property. 160static const int kFastTrackMultiplier = 2; 161 162// The minimum and maximum allowed values 163static const int kFastTrackMultiplierMin = 1; 164static const int kFastTrackMultiplierMax = 2; 165 166// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 167static int sFastTrackMultiplier = kFastTrackMultiplier; 168 169// See Thread::readOnlyHeap(). 170// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 171// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 172// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 173static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 174 175// ---------------------------------------------------------------------------- 176 177static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 178 179static void sFastTrackMultiplierInit() 180{ 181 char value[PROPERTY_VALUE_MAX]; 182 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 183 char *endptr; 184 unsigned long ul = strtoul(value, &endptr, 0); 185 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 186 sFastTrackMultiplier = (int) ul; 187 } 188 } 189} 190 191// ---------------------------------------------------------------------------- 192 193#ifdef ADD_BATTERY_DATA 194// To collect the amplifier usage 195static void addBatteryData(uint32_t params) { 196 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 197 if (service == NULL) { 198 // it already logged 199 return; 200 } 201 202 service->addBatteryData(params); 203} 204#endif 205 206 207// ---------------------------------------------------------------------------- 208// CPU Stats 209// ---------------------------------------------------------------------------- 210 211class CpuStats { 212public: 213 CpuStats(); 214 void sample(const String8 &title); 215#ifdef DEBUG_CPU_USAGE 216private: 217 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 218 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 219 220 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 221 222 int mCpuNum; // thread's current CPU number 223 int mCpukHz; // frequency of thread's current CPU in kHz 224#endif 225}; 226 227CpuStats::CpuStats() 228#ifdef DEBUG_CPU_USAGE 229 : mCpuNum(-1), mCpukHz(-1) 230#endif 231{ 232} 233 234void CpuStats::sample(const String8 &title 235#ifndef DEBUG_CPU_USAGE 236 __unused 237#endif 238 ) { 239#ifdef DEBUG_CPU_USAGE 240 // get current thread's delta CPU time in wall clock ns 241 double wcNs; 242 bool valid = mCpuUsage.sampleAndEnable(wcNs); 243 244 // record sample for wall clock statistics 245 if (valid) { 246 mWcStats.sample(wcNs); 247 } 248 249 // get the current CPU number 250 int cpuNum = sched_getcpu(); 251 252 // get the current CPU frequency in kHz 253 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 254 255 // check if either CPU number or frequency changed 256 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 257 mCpuNum = cpuNum; 258 mCpukHz = cpukHz; 259 // ignore sample for purposes of cycles 260 valid = false; 261 } 262 263 // if no change in CPU number or frequency, then record sample for cycle statistics 264 if (valid && mCpukHz > 0) { 265 double cycles = wcNs * cpukHz * 0.000001; 266 mHzStats.sample(cycles); 267 } 268 269 unsigned n = mWcStats.n(); 270 // mCpuUsage.elapsed() is expensive, so don't call it every loop 271 if ((n & 127) == 1) { 272 long long elapsed = mCpuUsage.elapsed(); 273 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 274 double perLoop = elapsed / (double) n; 275 double perLoop100 = perLoop * 0.01; 276 double perLoop1k = perLoop * 0.001; 277 double mean = mWcStats.mean(); 278 double stddev = mWcStats.stddev(); 279 double minimum = mWcStats.minimum(); 280 double maximum = mWcStats.maximum(); 281 double meanCycles = mHzStats.mean(); 282 double stddevCycles = mHzStats.stddev(); 283 double minCycles = mHzStats.minimum(); 284 double maxCycles = mHzStats.maximum(); 285 mCpuUsage.resetElapsed(); 286 mWcStats.reset(); 287 mHzStats.reset(); 288 ALOGD("CPU usage for %s over past %.1f secs\n" 289 " (%u mixer loops at %.1f mean ms per loop):\n" 290 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 291 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 292 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 293 title.string(), 294 elapsed * .000000001, n, perLoop * .000001, 295 mean * .001, 296 stddev * .001, 297 minimum * .001, 298 maximum * .001, 299 mean / perLoop100, 300 stddev / perLoop100, 301 minimum / perLoop100, 302 maximum / perLoop100, 303 meanCycles / perLoop1k, 304 stddevCycles / perLoop1k, 305 minCycles / perLoop1k, 306 maxCycles / perLoop1k); 307 308 } 309 } 310#endif 311}; 312 313// ---------------------------------------------------------------------------- 314// ThreadBase 315// ---------------------------------------------------------------------------- 316 317AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 318 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 319 : Thread(false /*canCallJava*/), 320 mType(type), 321 mAudioFlinger(audioFlinger), 322 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 323 // are set by PlaybackThread::readOutputParameters_l() or 324 // RecordThread::readInputParameters_l() 325 //FIXME: mStandby should be true here. Is this some kind of hack? 326 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 327 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 328 // mName will be set by concrete (non-virtual) subclass 329 mDeathRecipient(new PMDeathRecipient(this)) 330{ 331} 332 333AudioFlinger::ThreadBase::~ThreadBase() 334{ 335 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 336 mConfigEvents.clear(); 337 338 // do not lock the mutex in destructor 339 releaseWakeLock_l(); 340 if (mPowerManager != 0) { 341 sp<IBinder> binder = mPowerManager->asBinder(); 342 binder->unlinkToDeath(mDeathRecipient); 343 } 344} 345 346status_t AudioFlinger::ThreadBase::readyToRun() 347{ 348 status_t status = initCheck(); 349 if (status == NO_ERROR) { 350 ALOGI("AudioFlinger's thread %p ready to run", this); 351 } else { 352 ALOGE("No working audio driver found."); 353 } 354 return status; 355} 356 357void AudioFlinger::ThreadBase::exit() 358{ 359 ALOGV("ThreadBase::exit"); 360 // do any cleanup required for exit to succeed 361 preExit(); 362 { 363 // This lock prevents the following race in thread (uniprocessor for illustration): 364 // if (!exitPending()) { 365 // // context switch from here to exit() 366 // // exit() calls requestExit(), what exitPending() observes 367 // // exit() calls signal(), which is dropped since no waiters 368 // // context switch back from exit() to here 369 // mWaitWorkCV.wait(...); 370 // // now thread is hung 371 // } 372 AutoMutex lock(mLock); 373 requestExit(); 374 mWaitWorkCV.broadcast(); 375 } 376 // When Thread::requestExitAndWait is made virtual and this method is renamed to 377 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 378 requestExitAndWait(); 379} 380 381status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 382{ 383 status_t status; 384 385 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 386 Mutex::Autolock _l(mLock); 387 388 return sendSetParameterConfigEvent_l(keyValuePairs); 389} 390 391// sendConfigEvent_l() must be called with ThreadBase::mLock held 392// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 393status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 394{ 395 status_t status = NO_ERROR; 396 397 mConfigEvents.add(event); 398 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 399 mWaitWorkCV.signal(); 400 mLock.unlock(); 401 { 402 Mutex::Autolock _l(event->mLock); 403 while (event->mWaitStatus) { 404 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 405 event->mStatus = TIMED_OUT; 406 event->mWaitStatus = false; 407 } 408 } 409 status = event->mStatus; 410 } 411 mLock.lock(); 412 return status; 413} 414 415void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 416{ 417 Mutex::Autolock _l(mLock); 418 sendIoConfigEvent_l(event, param); 419} 420 421// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 422void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 423{ 424 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 425 sendConfigEvent_l(configEvent); 426} 427 428// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 429void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 430{ 431 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 432 sendConfigEvent_l(configEvent); 433} 434 435// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 436status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 437{ 438 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 439 return sendConfigEvent_l(configEvent); 440} 441 442status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 443 const struct audio_patch *patch, 444 audio_patch_handle_t *handle) 445{ 446 Mutex::Autolock _l(mLock); 447 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 448 status_t status = sendConfigEvent_l(configEvent); 449 if (status == NO_ERROR) { 450 CreateAudioPatchConfigEventData *data = 451 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 452 *handle = data->mHandle; 453 } 454 return status; 455} 456 457status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 458 const audio_patch_handle_t handle) 459{ 460 Mutex::Autolock _l(mLock); 461 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 462 return sendConfigEvent_l(configEvent); 463} 464 465 466// post condition: mConfigEvents.isEmpty() 467void AudioFlinger::ThreadBase::processConfigEvents_l() 468{ 469 bool configChanged = false; 470 471 while (!mConfigEvents.isEmpty()) { 472 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 473 sp<ConfigEvent> event = mConfigEvents[0]; 474 mConfigEvents.removeAt(0); 475 switch (event->mType) { 476 case CFG_EVENT_PRIO: { 477 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 478 // FIXME Need to understand why this has to be done asynchronously 479 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 480 true /*asynchronous*/); 481 if (err != 0) { 482 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 483 data->mPrio, data->mPid, data->mTid, err); 484 } 485 } break; 486 case CFG_EVENT_IO: { 487 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 488 audioConfigChanged(data->mEvent, data->mParam); 489 } break; 490 case CFG_EVENT_SET_PARAMETER: { 491 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 492 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 493 configChanged = true; 494 } 495 } break; 496 case CFG_EVENT_CREATE_AUDIO_PATCH: { 497 CreateAudioPatchConfigEventData *data = 498 (CreateAudioPatchConfigEventData *)event->mData.get(); 499 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 500 } break; 501 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 502 ReleaseAudioPatchConfigEventData *data = 503 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 504 event->mStatus = releaseAudioPatch_l(data->mHandle); 505 } break; 506 default: 507 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 508 break; 509 } 510 { 511 Mutex::Autolock _l(event->mLock); 512 if (event->mWaitStatus) { 513 event->mWaitStatus = false; 514 event->mCond.signal(); 515 } 516 } 517 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 518 } 519 520 if (configChanged) { 521 cacheParameters_l(); 522 } 523} 524 525String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 526 String8 s; 527 if (output) { 528 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 529 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 530 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 531 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 532 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 533 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 534 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 535 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 536 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 537 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 538 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 539 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 540 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 541 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 542 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 543 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 544 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 545 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 546 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 547 } else { 548 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 549 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 550 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 551 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 552 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 553 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 554 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 555 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 556 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 557 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 558 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 559 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 560 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 561 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 562 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 563 } 564 int len = s.length(); 565 if (s.length() > 2) { 566 char *str = s.lockBuffer(len); 567 s.unlockBuffer(len - 2); 568 } 569 return s; 570} 571 572void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 573{ 574 const size_t SIZE = 256; 575 char buffer[SIZE]; 576 String8 result; 577 578 bool locked = AudioFlinger::dumpTryLock(mLock); 579 if (!locked) { 580 dprintf(fd, "thread %p maybe dead locked\n", this); 581 } 582 583 dprintf(fd, " I/O handle: %d\n", mId); 584 dprintf(fd, " TID: %d\n", getTid()); 585 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 586 dprintf(fd, " Sample rate: %u\n", mSampleRate); 587 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 588 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 589 dprintf(fd, " Channel Count: %u\n", mChannelCount); 590 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 591 channelMaskToString(mChannelMask, mType != RECORD).string()); 592 dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 593 dprintf(fd, " Frame size: %zu\n", mFrameSize); 594 dprintf(fd, " Pending config events:"); 595 size_t numConfig = mConfigEvents.size(); 596 if (numConfig) { 597 for (size_t i = 0; i < numConfig; i++) { 598 mConfigEvents[i]->dump(buffer, SIZE); 599 dprintf(fd, "\n %s", buffer); 600 } 601 dprintf(fd, "\n"); 602 } else { 603 dprintf(fd, " none\n"); 604 } 605 606 if (locked) { 607 mLock.unlock(); 608 } 609} 610 611void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 612{ 613 const size_t SIZE = 256; 614 char buffer[SIZE]; 615 String8 result; 616 617 size_t numEffectChains = mEffectChains.size(); 618 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 619 write(fd, buffer, strlen(buffer)); 620 621 for (size_t i = 0; i < numEffectChains; ++i) { 622 sp<EffectChain> chain = mEffectChains[i]; 623 if (chain != 0) { 624 chain->dump(fd, args); 625 } 626 } 627} 628 629void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 630{ 631 Mutex::Autolock _l(mLock); 632 acquireWakeLock_l(uid); 633} 634 635String16 AudioFlinger::ThreadBase::getWakeLockTag() 636{ 637 switch (mType) { 638 case MIXER: 639 return String16("AudioMix"); 640 case DIRECT: 641 return String16("AudioDirectOut"); 642 case DUPLICATING: 643 return String16("AudioDup"); 644 case RECORD: 645 return String16("AudioIn"); 646 case OFFLOAD: 647 return String16("AudioOffload"); 648 default: 649 ALOG_ASSERT(false); 650 return String16("AudioUnknown"); 651 } 652} 653 654void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 655{ 656 getPowerManager_l(); 657 if (mPowerManager != 0) { 658 sp<IBinder> binder = new BBinder(); 659 status_t status; 660 if (uid >= 0) { 661 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 662 binder, 663 getWakeLockTag(), 664 String16("media"), 665 uid); 666 } else { 667 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 668 binder, 669 getWakeLockTag(), 670 String16("media")); 671 } 672 if (status == NO_ERROR) { 673 mWakeLockToken = binder; 674 } 675 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 676 } 677} 678 679void AudioFlinger::ThreadBase::releaseWakeLock() 680{ 681 Mutex::Autolock _l(mLock); 682 releaseWakeLock_l(); 683} 684 685void AudioFlinger::ThreadBase::releaseWakeLock_l() 686{ 687 if (mWakeLockToken != 0) { 688 ALOGV("releaseWakeLock_l() %s", mName); 689 if (mPowerManager != 0) { 690 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 691 } 692 mWakeLockToken.clear(); 693 } 694} 695 696void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 697 Mutex::Autolock _l(mLock); 698 updateWakeLockUids_l(uids); 699} 700 701void AudioFlinger::ThreadBase::getPowerManager_l() { 702 703 if (mPowerManager == 0) { 704 // use checkService() to avoid blocking if power service is not up yet 705 sp<IBinder> binder = 706 defaultServiceManager()->checkService(String16("power")); 707 if (binder == 0) { 708 ALOGW("Thread %s cannot connect to the power manager service", mName); 709 } else { 710 mPowerManager = interface_cast<IPowerManager>(binder); 711 binder->linkToDeath(mDeathRecipient); 712 } 713 } 714} 715 716void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 717 718 getPowerManager_l(); 719 if (mWakeLockToken == NULL) { 720 ALOGE("no wake lock to update!"); 721 return; 722 } 723 if (mPowerManager != 0) { 724 sp<IBinder> binder = new BBinder(); 725 status_t status; 726 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 727 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 728 } 729} 730 731void AudioFlinger::ThreadBase::clearPowerManager() 732{ 733 Mutex::Autolock _l(mLock); 734 releaseWakeLock_l(); 735 mPowerManager.clear(); 736} 737 738void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 739{ 740 sp<ThreadBase> thread = mThread.promote(); 741 if (thread != 0) { 742 thread->clearPowerManager(); 743 } 744 ALOGW("power manager service died !!!"); 745} 746 747void AudioFlinger::ThreadBase::setEffectSuspended( 748 const effect_uuid_t *type, bool suspend, int sessionId) 749{ 750 Mutex::Autolock _l(mLock); 751 setEffectSuspended_l(type, suspend, sessionId); 752} 753 754void AudioFlinger::ThreadBase::setEffectSuspended_l( 755 const effect_uuid_t *type, bool suspend, int sessionId) 756{ 757 sp<EffectChain> chain = getEffectChain_l(sessionId); 758 if (chain != 0) { 759 if (type != NULL) { 760 chain->setEffectSuspended_l(type, suspend); 761 } else { 762 chain->setEffectSuspendedAll_l(suspend); 763 } 764 } 765 766 updateSuspendedSessions_l(type, suspend, sessionId); 767} 768 769void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 770{ 771 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 772 if (index < 0) { 773 return; 774 } 775 776 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 777 mSuspendedSessions.valueAt(index); 778 779 for (size_t i = 0; i < sessionEffects.size(); i++) { 780 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 781 for (int j = 0; j < desc->mRefCount; j++) { 782 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 783 chain->setEffectSuspendedAll_l(true); 784 } else { 785 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 786 desc->mType.timeLow); 787 chain->setEffectSuspended_l(&desc->mType, true); 788 } 789 } 790 } 791} 792 793void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 794 bool suspend, 795 int sessionId) 796{ 797 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 798 799 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 800 801 if (suspend) { 802 if (index >= 0) { 803 sessionEffects = mSuspendedSessions.valueAt(index); 804 } else { 805 mSuspendedSessions.add(sessionId, sessionEffects); 806 } 807 } else { 808 if (index < 0) { 809 return; 810 } 811 sessionEffects = mSuspendedSessions.valueAt(index); 812 } 813 814 815 int key = EffectChain::kKeyForSuspendAll; 816 if (type != NULL) { 817 key = type->timeLow; 818 } 819 index = sessionEffects.indexOfKey(key); 820 821 sp<SuspendedSessionDesc> desc; 822 if (suspend) { 823 if (index >= 0) { 824 desc = sessionEffects.valueAt(index); 825 } else { 826 desc = new SuspendedSessionDesc(); 827 if (type != NULL) { 828 desc->mType = *type; 829 } 830 sessionEffects.add(key, desc); 831 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 832 } 833 desc->mRefCount++; 834 } else { 835 if (index < 0) { 836 return; 837 } 838 desc = sessionEffects.valueAt(index); 839 if (--desc->mRefCount == 0) { 840 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 841 sessionEffects.removeItemsAt(index); 842 if (sessionEffects.isEmpty()) { 843 ALOGV("updateSuspendedSessions_l() restore removing session %d", 844 sessionId); 845 mSuspendedSessions.removeItem(sessionId); 846 } 847 } 848 } 849 if (!sessionEffects.isEmpty()) { 850 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 851 } 852} 853 854void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 855 bool enabled, 856 int sessionId) 857{ 858 Mutex::Autolock _l(mLock); 859 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 860} 861 862void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 863 bool enabled, 864 int sessionId) 865{ 866 if (mType != RECORD) { 867 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 868 // another session. This gives the priority to well behaved effect control panels 869 // and applications not using global effects. 870 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 871 // global effects 872 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 873 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 874 } 875 } 876 877 sp<EffectChain> chain = getEffectChain_l(sessionId); 878 if (chain != 0) { 879 chain->checkSuspendOnEffectEnabled(effect, enabled); 880 } 881} 882 883// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 884sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 885 const sp<AudioFlinger::Client>& client, 886 const sp<IEffectClient>& effectClient, 887 int32_t priority, 888 int sessionId, 889 effect_descriptor_t *desc, 890 int *enabled, 891 status_t *status) 892{ 893 sp<EffectModule> effect; 894 sp<EffectHandle> handle; 895 status_t lStatus; 896 sp<EffectChain> chain; 897 bool chainCreated = false; 898 bool effectCreated = false; 899 bool effectRegistered = false; 900 901 lStatus = initCheck(); 902 if (lStatus != NO_ERROR) { 903 ALOGW("createEffect_l() Audio driver not initialized."); 904 goto Exit; 905 } 906 907 // Reject any effect on Direct output threads for now, since the format of 908 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 909 if (mType == DIRECT) { 910 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 911 desc->name, mName); 912 lStatus = BAD_VALUE; 913 goto Exit; 914 } 915 916 // Reject any effect on mixer or duplicating multichannel sinks. 917 // TODO: fix both format and multichannel issues with effects. 918 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 919 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 920 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 921 lStatus = BAD_VALUE; 922 goto Exit; 923 } 924 925 // Allow global effects only on offloaded and mixer threads 926 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 927 switch (mType) { 928 case MIXER: 929 case OFFLOAD: 930 break; 931 case DIRECT: 932 case DUPLICATING: 933 case RECORD: 934 default: 935 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 936 lStatus = BAD_VALUE; 937 goto Exit; 938 } 939 } 940 941 // Only Pre processor effects are allowed on input threads and only on input threads 942 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 943 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 944 desc->name, desc->flags, mType); 945 lStatus = BAD_VALUE; 946 goto Exit; 947 } 948 949 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 950 951 { // scope for mLock 952 Mutex::Autolock _l(mLock); 953 954 // check for existing effect chain with the requested audio session 955 chain = getEffectChain_l(sessionId); 956 if (chain == 0) { 957 // create a new chain for this session 958 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 959 chain = new EffectChain(this, sessionId); 960 addEffectChain_l(chain); 961 chain->setStrategy(getStrategyForSession_l(sessionId)); 962 chainCreated = true; 963 } else { 964 effect = chain->getEffectFromDesc_l(desc); 965 } 966 967 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 968 969 if (effect == 0) { 970 int id = mAudioFlinger->nextUniqueId(); 971 // Check CPU and memory usage 972 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 973 if (lStatus != NO_ERROR) { 974 goto Exit; 975 } 976 effectRegistered = true; 977 // create a new effect module if none present in the chain 978 effect = new EffectModule(this, chain, desc, id, sessionId); 979 lStatus = effect->status(); 980 if (lStatus != NO_ERROR) { 981 goto Exit; 982 } 983 effect->setOffloaded(mType == OFFLOAD, mId); 984 985 lStatus = chain->addEffect_l(effect); 986 if (lStatus != NO_ERROR) { 987 goto Exit; 988 } 989 effectCreated = true; 990 991 effect->setDevice(mOutDevice); 992 effect->setDevice(mInDevice); 993 effect->setMode(mAudioFlinger->getMode()); 994 effect->setAudioSource(mAudioSource); 995 } 996 // create effect handle and connect it to effect module 997 handle = new EffectHandle(effect, client, effectClient, priority); 998 lStatus = handle->initCheck(); 999 if (lStatus == OK) { 1000 lStatus = effect->addHandle(handle.get()); 1001 } 1002 if (enabled != NULL) { 1003 *enabled = (int)effect->isEnabled(); 1004 } 1005 } 1006 1007Exit: 1008 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1009 Mutex::Autolock _l(mLock); 1010 if (effectCreated) { 1011 chain->removeEffect_l(effect); 1012 } 1013 if (effectRegistered) { 1014 AudioSystem::unregisterEffect(effect->id()); 1015 } 1016 if (chainCreated) { 1017 removeEffectChain_l(chain); 1018 } 1019 handle.clear(); 1020 } 1021 1022 *status = lStatus; 1023 return handle; 1024} 1025 1026sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1027{ 1028 Mutex::Autolock _l(mLock); 1029 return getEffect_l(sessionId, effectId); 1030} 1031 1032sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1033{ 1034 sp<EffectChain> chain = getEffectChain_l(sessionId); 1035 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1036} 1037 1038// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1039// PlaybackThread::mLock held 1040status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1041{ 1042 // check for existing effect chain with the requested audio session 1043 int sessionId = effect->sessionId(); 1044 sp<EffectChain> chain = getEffectChain_l(sessionId); 1045 bool chainCreated = false; 1046 1047 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1048 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1049 this, effect->desc().name, effect->desc().flags); 1050 1051 if (chain == 0) { 1052 // create a new chain for this session 1053 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1054 chain = new EffectChain(this, sessionId); 1055 addEffectChain_l(chain); 1056 chain->setStrategy(getStrategyForSession_l(sessionId)); 1057 chainCreated = true; 1058 } 1059 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1060 1061 if (chain->getEffectFromId_l(effect->id()) != 0) { 1062 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1063 this, effect->desc().name, chain.get()); 1064 return BAD_VALUE; 1065 } 1066 1067 effect->setOffloaded(mType == OFFLOAD, mId); 1068 1069 status_t status = chain->addEffect_l(effect); 1070 if (status != NO_ERROR) { 1071 if (chainCreated) { 1072 removeEffectChain_l(chain); 1073 } 1074 return status; 1075 } 1076 1077 effect->setDevice(mOutDevice); 1078 effect->setDevice(mInDevice); 1079 effect->setMode(mAudioFlinger->getMode()); 1080 effect->setAudioSource(mAudioSource); 1081 return NO_ERROR; 1082} 1083 1084void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1085 1086 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1087 effect_descriptor_t desc = effect->desc(); 1088 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1089 detachAuxEffect_l(effect->id()); 1090 } 1091 1092 sp<EffectChain> chain = effect->chain().promote(); 1093 if (chain != 0) { 1094 // remove effect chain if removing last effect 1095 if (chain->removeEffect_l(effect) == 0) { 1096 removeEffectChain_l(chain); 1097 } 1098 } else { 1099 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1100 } 1101} 1102 1103void AudioFlinger::ThreadBase::lockEffectChains_l( 1104 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1105{ 1106 effectChains = mEffectChains; 1107 for (size_t i = 0; i < mEffectChains.size(); i++) { 1108 mEffectChains[i]->lock(); 1109 } 1110} 1111 1112void AudioFlinger::ThreadBase::unlockEffectChains( 1113 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1114{ 1115 for (size_t i = 0; i < effectChains.size(); i++) { 1116 effectChains[i]->unlock(); 1117 } 1118} 1119 1120sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1121{ 1122 Mutex::Autolock _l(mLock); 1123 return getEffectChain_l(sessionId); 1124} 1125 1126sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1127{ 1128 size_t size = mEffectChains.size(); 1129 for (size_t i = 0; i < size; i++) { 1130 if (mEffectChains[i]->sessionId() == sessionId) { 1131 return mEffectChains[i]; 1132 } 1133 } 1134 return 0; 1135} 1136 1137void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1138{ 1139 Mutex::Autolock _l(mLock); 1140 size_t size = mEffectChains.size(); 1141 for (size_t i = 0; i < size; i++) { 1142 mEffectChains[i]->setMode_l(mode); 1143 } 1144} 1145 1146void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1147 EffectHandle *handle, 1148 bool unpinIfLast) { 1149 1150 Mutex::Autolock _l(mLock); 1151 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1152 // delete the effect module if removing last handle on it 1153 if (effect->removeHandle(handle) == 0) { 1154 if (!effect->isPinned() || unpinIfLast) { 1155 removeEffect_l(effect); 1156 AudioSystem::unregisterEffect(effect->id()); 1157 } 1158 } 1159} 1160 1161void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1162{ 1163 config->type = AUDIO_PORT_TYPE_MIX; 1164 config->ext.mix.handle = mId; 1165 config->sample_rate = mSampleRate; 1166 config->format = mFormat; 1167 config->channel_mask = mChannelMask; 1168 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1169 AUDIO_PORT_CONFIG_FORMAT; 1170} 1171 1172 1173// ---------------------------------------------------------------------------- 1174// Playback 1175// ---------------------------------------------------------------------------- 1176 1177AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1178 AudioStreamOut* output, 1179 audio_io_handle_t id, 1180 audio_devices_t device, 1181 type_t type) 1182 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1183 mNormalFrameCount(0), mSinkBuffer(NULL), 1184 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1185 mMixerBuffer(NULL), 1186 mMixerBufferSize(0), 1187 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1188 mMixerBufferValid(false), 1189 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1190 mEffectBuffer(NULL), 1191 mEffectBufferSize(0), 1192 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1193 mEffectBufferValid(false), 1194 mSuspended(0), mBytesWritten(0), 1195 mActiveTracksGeneration(0), 1196 // mStreamTypes[] initialized in constructor body 1197 mOutput(output), 1198 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1199 mMixerStatus(MIXER_IDLE), 1200 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1201 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1202 mBytesRemaining(0), 1203 mCurrentWriteLength(0), 1204 mUseAsyncWrite(false), 1205 mWriteAckSequence(0), 1206 mDrainSequence(0), 1207 mSignalPending(false), 1208 mScreenState(AudioFlinger::mScreenState), 1209 // index 0 is reserved for normal mixer's submix 1210 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1211 // mLatchD, mLatchQ, 1212 mLatchDValid(false), mLatchQValid(false) 1213{ 1214 snprintf(mName, kNameLength, "AudioOut_%X", id); 1215 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1216 1217 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1218 // it would be safer to explicitly pass initial masterVolume/masterMute as 1219 // parameter. 1220 // 1221 // If the HAL we are using has support for master volume or master mute, 1222 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1223 // and the mute set to false). 1224 mMasterVolume = audioFlinger->masterVolume_l(); 1225 mMasterMute = audioFlinger->masterMute_l(); 1226 if (mOutput && mOutput->audioHwDev) { 1227 if (mOutput->audioHwDev->canSetMasterVolume()) { 1228 mMasterVolume = 1.0; 1229 } 1230 1231 if (mOutput->audioHwDev->canSetMasterMute()) { 1232 mMasterMute = false; 1233 } 1234 } 1235 1236 readOutputParameters_l(); 1237 1238 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1239 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1240 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1241 stream = (audio_stream_type_t) (stream + 1)) { 1242 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1243 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1244 } 1245 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1246 // because mAudioFlinger doesn't have one to copy from 1247} 1248 1249AudioFlinger::PlaybackThread::~PlaybackThread() 1250{ 1251 mAudioFlinger->unregisterWriter(mNBLogWriter); 1252 free(mSinkBuffer); 1253 free(mMixerBuffer); 1254 free(mEffectBuffer); 1255} 1256 1257void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1258{ 1259 dumpInternals(fd, args); 1260 dumpTracks(fd, args); 1261 dumpEffectChains(fd, args); 1262} 1263 1264void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1265{ 1266 const size_t SIZE = 256; 1267 char buffer[SIZE]; 1268 String8 result; 1269 1270 result.appendFormat(" Stream volumes in dB: "); 1271 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1272 const stream_type_t *st = &mStreamTypes[i]; 1273 if (i > 0) { 1274 result.appendFormat(", "); 1275 } 1276 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1277 if (st->mute) { 1278 result.append("M"); 1279 } 1280 } 1281 result.append("\n"); 1282 write(fd, result.string(), result.length()); 1283 result.clear(); 1284 1285 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1286 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1287 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1288 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1289 1290 size_t numtracks = mTracks.size(); 1291 size_t numactive = mActiveTracks.size(); 1292 dprintf(fd, " %d Tracks", numtracks); 1293 size_t numactiveseen = 0; 1294 if (numtracks) { 1295 dprintf(fd, " of which %d are active\n", numactive); 1296 Track::appendDumpHeader(result); 1297 for (size_t i = 0; i < numtracks; ++i) { 1298 sp<Track> track = mTracks[i]; 1299 if (track != 0) { 1300 bool active = mActiveTracks.indexOf(track) >= 0; 1301 if (active) { 1302 numactiveseen++; 1303 } 1304 track->dump(buffer, SIZE, active); 1305 result.append(buffer); 1306 } 1307 } 1308 } else { 1309 result.append("\n"); 1310 } 1311 if (numactiveseen != numactive) { 1312 // some tracks in the active list were not in the tracks list 1313 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1314 " not in the track list\n"); 1315 result.append(buffer); 1316 Track::appendDumpHeader(result); 1317 for (size_t i = 0; i < numactive; ++i) { 1318 sp<Track> track = mActiveTracks[i].promote(); 1319 if (track != 0 && mTracks.indexOf(track) < 0) { 1320 track->dump(buffer, SIZE, true); 1321 result.append(buffer); 1322 } 1323 } 1324 } 1325 1326 write(fd, result.string(), result.size()); 1327} 1328 1329void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1330{ 1331 dprintf(fd, "\nOutput thread %p:\n", this); 1332 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1333 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1334 dprintf(fd, " Total writes: %d\n", mNumWrites); 1335 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1336 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1337 dprintf(fd, " Suspend count: %d\n", mSuspended); 1338 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1339 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1340 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1341 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1342 1343 dumpBase(fd, args); 1344} 1345 1346// Thread virtuals 1347 1348void AudioFlinger::PlaybackThread::onFirstRef() 1349{ 1350 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1351} 1352 1353// ThreadBase virtuals 1354void AudioFlinger::PlaybackThread::preExit() 1355{ 1356 ALOGV(" preExit()"); 1357 // FIXME this is using hard-coded strings but in the future, this functionality will be 1358 // converted to use audio HAL extensions required to support tunneling 1359 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1360} 1361 1362// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1363sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1364 const sp<AudioFlinger::Client>& client, 1365 audio_stream_type_t streamType, 1366 uint32_t sampleRate, 1367 audio_format_t format, 1368 audio_channel_mask_t channelMask, 1369 size_t *pFrameCount, 1370 const sp<IMemory>& sharedBuffer, 1371 int sessionId, 1372 IAudioFlinger::track_flags_t *flags, 1373 pid_t tid, 1374 int uid, 1375 status_t *status) 1376{ 1377 size_t frameCount = *pFrameCount; 1378 sp<Track> track; 1379 status_t lStatus; 1380 1381 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1382 1383 // client expresses a preference for FAST, but we get the final say 1384 if (*flags & IAudioFlinger::TRACK_FAST) { 1385 if ( 1386 // not timed 1387 (!isTimed) && 1388 // either of these use cases: 1389 ( 1390 // use case 1: shared buffer with any frame count 1391 ( 1392 (sharedBuffer != 0) 1393 ) || 1394 // use case 2: callback handler and frame count is default or at least as large as HAL 1395 ( 1396 (tid != -1) && 1397 ((frameCount == 0) || 1398 (frameCount >= mFrameCount)) 1399 ) 1400 ) && 1401 // PCM data 1402 audio_is_linear_pcm(format) && 1403 // identical channel mask to sink, or mono in and stereo sink 1404 (channelMask == mChannelMask || 1405 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1406 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1407 // hardware sample rate 1408 (sampleRate == mSampleRate) && 1409 // normal mixer has an associated fast mixer 1410 hasFastMixer() && 1411 // there are sufficient fast track slots available 1412 (mFastTrackAvailMask != 0) 1413 // FIXME test that MixerThread for this fast track has a capable output HAL 1414 // FIXME add a permission test also? 1415 ) { 1416 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1417 if (frameCount == 0) { 1418 // read the fast track multiplier property the first time it is needed 1419 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1420 if (ok != 0) { 1421 ALOGE("%s pthread_once failed: %d", __func__, ok); 1422 } 1423 frameCount = mFrameCount * sFastTrackMultiplier; 1424 } 1425 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1426 frameCount, mFrameCount); 1427 } else { 1428 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1429 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1430 "sampleRate=%u mSampleRate=%u " 1431 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1432 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1433 audio_is_linear_pcm(format), 1434 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1435 *flags &= ~IAudioFlinger::TRACK_FAST; 1436 // For compatibility with AudioTrack calculation, buffer depth is forced 1437 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1438 // This is probably too conservative, but legacy application code may depend on it. 1439 // If you change this calculation, also review the start threshold which is related. 1440 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1441 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1442 if (minBufCount < 2) { 1443 minBufCount = 2; 1444 } 1445 size_t minFrameCount = mNormalFrameCount * minBufCount; 1446 if (frameCount < minFrameCount) { 1447 frameCount = minFrameCount; 1448 } 1449 } 1450 } 1451 *pFrameCount = frameCount; 1452 1453 switch (mType) { 1454 1455 case DIRECT: 1456 if (audio_is_linear_pcm(format)) { 1457 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1458 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1459 "for output %p with format %#x", 1460 sampleRate, format, channelMask, mOutput, mFormat); 1461 lStatus = BAD_VALUE; 1462 goto Exit; 1463 } 1464 } 1465 break; 1466 1467 case OFFLOAD: 1468 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1469 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1470 "for output %p with format %#x", 1471 sampleRate, format, channelMask, mOutput, mFormat); 1472 lStatus = BAD_VALUE; 1473 goto Exit; 1474 } 1475 break; 1476 1477 default: 1478 if (!audio_is_linear_pcm(format)) { 1479 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1480 "for output %p with format %#x", 1481 format, mOutput, mFormat); 1482 lStatus = BAD_VALUE; 1483 goto Exit; 1484 } 1485 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1486 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1487 lStatus = BAD_VALUE; 1488 goto Exit; 1489 } 1490 break; 1491 1492 } 1493 1494 lStatus = initCheck(); 1495 if (lStatus != NO_ERROR) { 1496 ALOGE("createTrack_l() audio driver not initialized"); 1497 goto Exit; 1498 } 1499 1500 { // scope for mLock 1501 Mutex::Autolock _l(mLock); 1502 1503 // all tracks in same audio session must share the same routing strategy otherwise 1504 // conflicts will happen when tracks are moved from one output to another by audio policy 1505 // manager 1506 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1507 for (size_t i = 0; i < mTracks.size(); ++i) { 1508 sp<Track> t = mTracks[i]; 1509 if (t != 0 && t->isExternalTrack()) { 1510 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1511 if (sessionId == t->sessionId() && strategy != actual) { 1512 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1513 strategy, actual); 1514 lStatus = BAD_VALUE; 1515 goto Exit; 1516 } 1517 } 1518 } 1519 1520 if (!isTimed) { 1521 track = new Track(this, client, streamType, sampleRate, format, 1522 channelMask, frameCount, NULL, sharedBuffer, 1523 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1524 } else { 1525 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1526 channelMask, frameCount, sharedBuffer, sessionId, uid); 1527 } 1528 1529 // new Track always returns non-NULL, 1530 // but TimedTrack::create() is a factory that could fail by returning NULL 1531 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1532 if (lStatus != NO_ERROR) { 1533 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1534 // track must be cleared from the caller as the caller has the AF lock 1535 goto Exit; 1536 } 1537 mTracks.add(track); 1538 1539 sp<EffectChain> chain = getEffectChain_l(sessionId); 1540 if (chain != 0) { 1541 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1542 track->setMainBuffer(chain->inBuffer()); 1543 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1544 chain->incTrackCnt(); 1545 } 1546 1547 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1548 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1549 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1550 // so ask activity manager to do this on our behalf 1551 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1552 } 1553 } 1554 1555 lStatus = NO_ERROR; 1556 1557Exit: 1558 *status = lStatus; 1559 return track; 1560} 1561 1562uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1563{ 1564 return latency; 1565} 1566 1567uint32_t AudioFlinger::PlaybackThread::latency() const 1568{ 1569 Mutex::Autolock _l(mLock); 1570 return latency_l(); 1571} 1572uint32_t AudioFlinger::PlaybackThread::latency_l() const 1573{ 1574 if (initCheck() == NO_ERROR) { 1575 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1576 } else { 1577 return 0; 1578 } 1579} 1580 1581void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1582{ 1583 Mutex::Autolock _l(mLock); 1584 // Don't apply master volume in SW if our HAL can do it for us. 1585 if (mOutput && mOutput->audioHwDev && 1586 mOutput->audioHwDev->canSetMasterVolume()) { 1587 mMasterVolume = 1.0; 1588 } else { 1589 mMasterVolume = value; 1590 } 1591} 1592 1593void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1594{ 1595 Mutex::Autolock _l(mLock); 1596 // Don't apply master mute in SW if our HAL can do it for us. 1597 if (mOutput && mOutput->audioHwDev && 1598 mOutput->audioHwDev->canSetMasterMute()) { 1599 mMasterMute = false; 1600 } else { 1601 mMasterMute = muted; 1602 } 1603} 1604 1605void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1606{ 1607 Mutex::Autolock _l(mLock); 1608 mStreamTypes[stream].volume = value; 1609 broadcast_l(); 1610} 1611 1612void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1613{ 1614 Mutex::Autolock _l(mLock); 1615 mStreamTypes[stream].mute = muted; 1616 broadcast_l(); 1617} 1618 1619float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1620{ 1621 Mutex::Autolock _l(mLock); 1622 return mStreamTypes[stream].volume; 1623} 1624 1625// addTrack_l() must be called with ThreadBase::mLock held 1626status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1627{ 1628 status_t status = ALREADY_EXISTS; 1629 1630 // set retry count for buffer fill 1631 track->mRetryCount = kMaxTrackStartupRetries; 1632 if (mActiveTracks.indexOf(track) < 0) { 1633 // the track is newly added, make sure it fills up all its 1634 // buffers before playing. This is to ensure the client will 1635 // effectively get the latency it requested. 1636 if (track->isExternalTrack()) { 1637 TrackBase::track_state state = track->mState; 1638 mLock.unlock(); 1639 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1640 mLock.lock(); 1641 // abort track was stopped/paused while we released the lock 1642 if (state != track->mState) { 1643 if (status == NO_ERROR) { 1644 mLock.unlock(); 1645 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1646 mLock.lock(); 1647 } 1648 return INVALID_OPERATION; 1649 } 1650 // abort if start is rejected by audio policy manager 1651 if (status != NO_ERROR) { 1652 return PERMISSION_DENIED; 1653 } 1654#ifdef ADD_BATTERY_DATA 1655 // to track the speaker usage 1656 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1657#endif 1658 } 1659 1660 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1661 track->mResetDone = false; 1662 track->mPresentationCompleteFrames = 0; 1663 mActiveTracks.add(track); 1664 mWakeLockUids.add(track->uid()); 1665 mActiveTracksGeneration++; 1666 mLatestActiveTrack = track; 1667 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1668 if (chain != 0) { 1669 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1670 track->sessionId()); 1671 chain->incActiveTrackCnt(); 1672 } 1673 1674 status = NO_ERROR; 1675 } 1676 1677 onAddNewTrack_l(); 1678 return status; 1679} 1680 1681bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1682{ 1683 track->terminate(); 1684 // active tracks are removed by threadLoop() 1685 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1686 track->mState = TrackBase::STOPPED; 1687 if (!trackActive) { 1688 removeTrack_l(track); 1689 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1690 track->mState = TrackBase::STOPPING_1; 1691 } 1692 1693 return trackActive; 1694} 1695 1696void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1697{ 1698 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1699 mTracks.remove(track); 1700 deleteTrackName_l(track->name()); 1701 // redundant as track is about to be destroyed, for dumpsys only 1702 track->mName = -1; 1703 if (track->isFastTrack()) { 1704 int index = track->mFastIndex; 1705 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1706 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1707 mFastTrackAvailMask |= 1 << index; 1708 // redundant as track is about to be destroyed, for dumpsys only 1709 track->mFastIndex = -1; 1710 } 1711 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1712 if (chain != 0) { 1713 chain->decTrackCnt(); 1714 } 1715} 1716 1717void AudioFlinger::PlaybackThread::broadcast_l() 1718{ 1719 // Thread could be blocked waiting for async 1720 // so signal it to handle state changes immediately 1721 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1722 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1723 mSignalPending = true; 1724 mWaitWorkCV.broadcast(); 1725} 1726 1727String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1728{ 1729 Mutex::Autolock _l(mLock); 1730 if (initCheck() != NO_ERROR) { 1731 return String8(); 1732 } 1733 1734 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1735 const String8 out_s8(s); 1736 free(s); 1737 return out_s8; 1738} 1739 1740void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1741 AudioSystem::OutputDescriptor desc; 1742 void *param2 = NULL; 1743 1744 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1745 param); 1746 1747 switch (event) { 1748 case AudioSystem::OUTPUT_OPENED: 1749 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1750 desc.channelMask = mChannelMask; 1751 desc.samplingRate = mSampleRate; 1752 desc.format = mFormat; 1753 desc.frameCount = mNormalFrameCount; // FIXME see 1754 // AudioFlinger::frameCount(audio_io_handle_t) 1755 desc.latency = latency_l(); 1756 param2 = &desc; 1757 break; 1758 1759 case AudioSystem::STREAM_CONFIG_CHANGED: 1760 param2 = ¶m; 1761 case AudioSystem::OUTPUT_CLOSED: 1762 default: 1763 break; 1764 } 1765 mAudioFlinger->audioConfigChanged(event, mId, param2); 1766} 1767 1768void AudioFlinger::PlaybackThread::writeCallback() 1769{ 1770 ALOG_ASSERT(mCallbackThread != 0); 1771 mCallbackThread->resetWriteBlocked(); 1772} 1773 1774void AudioFlinger::PlaybackThread::drainCallback() 1775{ 1776 ALOG_ASSERT(mCallbackThread != 0); 1777 mCallbackThread->resetDraining(); 1778} 1779 1780void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1781{ 1782 Mutex::Autolock _l(mLock); 1783 // reject out of sequence requests 1784 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1785 mWriteAckSequence &= ~1; 1786 mWaitWorkCV.signal(); 1787 } 1788} 1789 1790void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1791{ 1792 Mutex::Autolock _l(mLock); 1793 // reject out of sequence requests 1794 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1795 mDrainSequence &= ~1; 1796 mWaitWorkCV.signal(); 1797 } 1798} 1799 1800// static 1801int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1802 void *param __unused, 1803 void *cookie) 1804{ 1805 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1806 ALOGV("asyncCallback() event %d", event); 1807 switch (event) { 1808 case STREAM_CBK_EVENT_WRITE_READY: 1809 me->writeCallback(); 1810 break; 1811 case STREAM_CBK_EVENT_DRAIN_READY: 1812 me->drainCallback(); 1813 break; 1814 default: 1815 ALOGW("asyncCallback() unknown event %d", event); 1816 break; 1817 } 1818 return 0; 1819} 1820 1821void AudioFlinger::PlaybackThread::readOutputParameters_l() 1822{ 1823 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1824 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1825 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1826 if (!audio_is_output_channel(mChannelMask)) { 1827 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1828 } 1829 if ((mType == MIXER || mType == DUPLICATING) 1830 && !isValidPcmSinkChannelMask(mChannelMask)) { 1831 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1832 mChannelMask); 1833 } 1834 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1835 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1836 mFormat = mHALFormat; 1837 if (!audio_is_valid_format(mFormat)) { 1838 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1839 } 1840 if ((mType == MIXER || mType == DUPLICATING) 1841 && !isValidPcmSinkFormat(mFormat)) { 1842 LOG_FATAL("HAL format %#x not supported for mixed output", 1843 mFormat); 1844 } 1845 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1846 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1847 mFrameCount = mBufferSize / mFrameSize; 1848 if (mFrameCount & 15) { 1849 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1850 mFrameCount); 1851 } 1852 1853 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1854 (mOutput->stream->set_callback != NULL)) { 1855 if (mOutput->stream->set_callback(mOutput->stream, 1856 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1857 mUseAsyncWrite = true; 1858 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1859 } 1860 } 1861 1862 // Calculate size of normal sink buffer relative to the HAL output buffer size 1863 double multiplier = 1.0; 1864 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1865 kUseFastMixer == FastMixer_Dynamic)) { 1866 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1867 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1868 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1869 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1870 maxNormalFrameCount = maxNormalFrameCount & ~15; 1871 if (maxNormalFrameCount < minNormalFrameCount) { 1872 maxNormalFrameCount = minNormalFrameCount; 1873 } 1874 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1875 if (multiplier <= 1.0) { 1876 multiplier = 1.0; 1877 } else if (multiplier <= 2.0) { 1878 if (2 * mFrameCount <= maxNormalFrameCount) { 1879 multiplier = 2.0; 1880 } else { 1881 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1882 } 1883 } else { 1884 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1885 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1886 // track, but we sometimes have to do this to satisfy the maximum frame count 1887 // constraint) 1888 // FIXME this rounding up should not be done if no HAL SRC 1889 uint32_t truncMult = (uint32_t) multiplier; 1890 if ((truncMult & 1)) { 1891 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1892 ++truncMult; 1893 } 1894 } 1895 multiplier = (double) truncMult; 1896 } 1897 } 1898 mNormalFrameCount = multiplier * mFrameCount; 1899 // round up to nearest 16 frames to satisfy AudioMixer 1900 if (mType == MIXER || mType == DUPLICATING) { 1901 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1902 } 1903 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1904 mNormalFrameCount); 1905 1906 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1907 // Originally this was int16_t[] array, need to remove legacy implications. 1908 free(mSinkBuffer); 1909 mSinkBuffer = NULL; 1910 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1911 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1912 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1913 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1914 1915 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1916 // drives the output. 1917 free(mMixerBuffer); 1918 mMixerBuffer = NULL; 1919 if (mMixerBufferEnabled) { 1920 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1921 mMixerBufferSize = mNormalFrameCount * mChannelCount 1922 * audio_bytes_per_sample(mMixerBufferFormat); 1923 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1924 } 1925 free(mEffectBuffer); 1926 mEffectBuffer = NULL; 1927 if (mEffectBufferEnabled) { 1928 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1929 mEffectBufferSize = mNormalFrameCount * mChannelCount 1930 * audio_bytes_per_sample(mEffectBufferFormat); 1931 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1932 } 1933 1934 // force reconfiguration of effect chains and engines to take new buffer size and audio 1935 // parameters into account 1936 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1937 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1938 // matter. 1939 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1940 Vector< sp<EffectChain> > effectChains = mEffectChains; 1941 for (size_t i = 0; i < effectChains.size(); i ++) { 1942 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1943 } 1944} 1945 1946 1947status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1948{ 1949 if (halFrames == NULL || dspFrames == NULL) { 1950 return BAD_VALUE; 1951 } 1952 Mutex::Autolock _l(mLock); 1953 if (initCheck() != NO_ERROR) { 1954 return INVALID_OPERATION; 1955 } 1956 size_t framesWritten = mBytesWritten / mFrameSize; 1957 *halFrames = framesWritten; 1958 1959 if (isSuspended()) { 1960 // return an estimation of rendered frames when the output is suspended 1961 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1962 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1963 return NO_ERROR; 1964 } else { 1965 status_t status; 1966 uint32_t frames; 1967 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1968 *dspFrames = (size_t)frames; 1969 return status; 1970 } 1971} 1972 1973uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1974{ 1975 Mutex::Autolock _l(mLock); 1976 uint32_t result = 0; 1977 if (getEffectChain_l(sessionId) != 0) { 1978 result = EFFECT_SESSION; 1979 } 1980 1981 for (size_t i = 0; i < mTracks.size(); ++i) { 1982 sp<Track> track = mTracks[i]; 1983 if (sessionId == track->sessionId() && !track->isInvalid()) { 1984 result |= TRACK_SESSION; 1985 break; 1986 } 1987 } 1988 1989 return result; 1990} 1991 1992uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1993{ 1994 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1995 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1996 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1997 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1998 } 1999 for (size_t i = 0; i < mTracks.size(); i++) { 2000 sp<Track> track = mTracks[i]; 2001 if (sessionId == track->sessionId() && !track->isInvalid()) { 2002 return AudioSystem::getStrategyForStream(track->streamType()); 2003 } 2004 } 2005 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2006} 2007 2008 2009AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2010{ 2011 Mutex::Autolock _l(mLock); 2012 return mOutput; 2013} 2014 2015AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2016{ 2017 Mutex::Autolock _l(mLock); 2018 AudioStreamOut *output = mOutput; 2019 mOutput = NULL; 2020 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2021 // must push a NULL and wait for ack 2022 mOutputSink.clear(); 2023 mPipeSink.clear(); 2024 mNormalSink.clear(); 2025 return output; 2026} 2027 2028// this method must always be called either with ThreadBase mLock held or inside the thread loop 2029audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2030{ 2031 if (mOutput == NULL) { 2032 return NULL; 2033 } 2034 return &mOutput->stream->common; 2035} 2036 2037uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2038{ 2039 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2040} 2041 2042status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2043{ 2044 if (!isValidSyncEvent(event)) { 2045 return BAD_VALUE; 2046 } 2047 2048 Mutex::Autolock _l(mLock); 2049 2050 for (size_t i = 0; i < mTracks.size(); ++i) { 2051 sp<Track> track = mTracks[i]; 2052 if (event->triggerSession() == track->sessionId()) { 2053 (void) track->setSyncEvent(event); 2054 return NO_ERROR; 2055 } 2056 } 2057 2058 return NAME_NOT_FOUND; 2059} 2060 2061bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2062{ 2063 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2064} 2065 2066void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2067 const Vector< sp<Track> >& tracksToRemove) 2068{ 2069 size_t count = tracksToRemove.size(); 2070 if (count > 0) { 2071 for (size_t i = 0 ; i < count ; i++) { 2072 const sp<Track>& track = tracksToRemove.itemAt(i); 2073 if (track->isExternalTrack()) { 2074 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2075#ifdef ADD_BATTERY_DATA 2076 // to track the speaker usage 2077 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2078#endif 2079 if (track->isTerminated()) { 2080 AudioSystem::releaseOutput(mId); 2081 } 2082 } 2083 } 2084 } 2085} 2086 2087void AudioFlinger::PlaybackThread::checkSilentMode_l() 2088{ 2089 if (!mMasterMute) { 2090 char value[PROPERTY_VALUE_MAX]; 2091 if (property_get("ro.audio.silent", value, "0") > 0) { 2092 char *endptr; 2093 unsigned long ul = strtoul(value, &endptr, 0); 2094 if (*endptr == '\0' && ul != 0) { 2095 ALOGD("Silence is golden"); 2096 // The setprop command will not allow a property to be changed after 2097 // the first time it is set, so we don't have to worry about un-muting. 2098 setMasterMute_l(true); 2099 } 2100 } 2101 } 2102} 2103 2104// shared by MIXER and DIRECT, overridden by DUPLICATING 2105ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2106{ 2107 // FIXME rewrite to reduce number of system calls 2108 mLastWriteTime = systemTime(); 2109 mInWrite = true; 2110 ssize_t bytesWritten; 2111 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2112 2113 // If an NBAIO sink is present, use it to write the normal mixer's submix 2114 if (mNormalSink != 0) { 2115 const size_t count = mBytesRemaining / mFrameSize; 2116 2117 ATRACE_BEGIN("write"); 2118 // update the setpoint when AudioFlinger::mScreenState changes 2119 uint32_t screenState = AudioFlinger::mScreenState; 2120 if (screenState != mScreenState) { 2121 mScreenState = screenState; 2122 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2123 if (pipe != NULL) { 2124 pipe->setAvgFrames((mScreenState & 1) ? 2125 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2126 } 2127 } 2128 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2129 ATRACE_END(); 2130 if (framesWritten > 0) { 2131 bytesWritten = framesWritten * mFrameSize; 2132 } else { 2133 bytesWritten = framesWritten; 2134 } 2135 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2136 if (status == NO_ERROR) { 2137 size_t totalFramesWritten = mNormalSink->framesWritten(); 2138 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2139 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2140 mLatchDValid = true; 2141 } 2142 } 2143 // otherwise use the HAL / AudioStreamOut directly 2144 } else { 2145 // Direct output and offload threads 2146 2147 if (mUseAsyncWrite) { 2148 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2149 mWriteAckSequence += 2; 2150 mWriteAckSequence |= 1; 2151 ALOG_ASSERT(mCallbackThread != 0); 2152 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2153 } 2154 // FIXME We should have an implementation of timestamps for direct output threads. 2155 // They are used e.g for multichannel PCM playback over HDMI. 2156 bytesWritten = mOutput->stream->write(mOutput->stream, 2157 (char *)mSinkBuffer + offset, mBytesRemaining); 2158 if (mUseAsyncWrite && 2159 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2160 // do not wait for async callback in case of error of full write 2161 mWriteAckSequence &= ~1; 2162 ALOG_ASSERT(mCallbackThread != 0); 2163 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2164 } 2165 } 2166 2167 mNumWrites++; 2168 mInWrite = false; 2169 mStandby = false; 2170 return bytesWritten; 2171} 2172 2173void AudioFlinger::PlaybackThread::threadLoop_drain() 2174{ 2175 if (mOutput->stream->drain) { 2176 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2177 if (mUseAsyncWrite) { 2178 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2179 mDrainSequence |= 1; 2180 ALOG_ASSERT(mCallbackThread != 0); 2181 mCallbackThread->setDraining(mDrainSequence); 2182 } 2183 mOutput->stream->drain(mOutput->stream, 2184 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2185 : AUDIO_DRAIN_ALL); 2186 } 2187} 2188 2189void AudioFlinger::PlaybackThread::threadLoop_exit() 2190{ 2191 // Default implementation has nothing to do 2192} 2193 2194/* 2195The derived values that are cached: 2196 - mSinkBufferSize from frame count * frame size 2197 - activeSleepTime from activeSleepTimeUs() 2198 - idleSleepTime from idleSleepTimeUs() 2199 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2200 - maxPeriod from frame count and sample rate (MIXER only) 2201 2202The parameters that affect these derived values are: 2203 - frame count 2204 - frame size 2205 - sample rate 2206 - device type: A2DP or not 2207 - device latency 2208 - format: PCM or not 2209 - active sleep time 2210 - idle sleep time 2211*/ 2212 2213void AudioFlinger::PlaybackThread::cacheParameters_l() 2214{ 2215 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2216 activeSleepTime = activeSleepTimeUs(); 2217 idleSleepTime = idleSleepTimeUs(); 2218} 2219 2220void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2221{ 2222 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2223 this, streamType, mTracks.size()); 2224 Mutex::Autolock _l(mLock); 2225 2226 size_t size = mTracks.size(); 2227 for (size_t i = 0; i < size; i++) { 2228 sp<Track> t = mTracks[i]; 2229 if (t->streamType() == streamType) { 2230 t->invalidate(); 2231 } 2232 } 2233} 2234 2235status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2236{ 2237 int session = chain->sessionId(); 2238 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2239 ? mEffectBuffer : mSinkBuffer); 2240 bool ownsBuffer = false; 2241 2242 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2243 if (session > 0) { 2244 // Only one effect chain can be present in direct output thread and it uses 2245 // the sink buffer as input 2246 if (mType != DIRECT) { 2247 size_t numSamples = mNormalFrameCount * mChannelCount; 2248 buffer = new int16_t[numSamples]; 2249 memset(buffer, 0, numSamples * sizeof(int16_t)); 2250 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2251 ownsBuffer = true; 2252 } 2253 2254 // Attach all tracks with same session ID to this chain. 2255 for (size_t i = 0; i < mTracks.size(); ++i) { 2256 sp<Track> track = mTracks[i]; 2257 if (session == track->sessionId()) { 2258 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2259 buffer); 2260 track->setMainBuffer(buffer); 2261 chain->incTrackCnt(); 2262 } 2263 } 2264 2265 // indicate all active tracks in the chain 2266 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2267 sp<Track> track = mActiveTracks[i].promote(); 2268 if (track == 0) { 2269 continue; 2270 } 2271 if (session == track->sessionId()) { 2272 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2273 chain->incActiveTrackCnt(); 2274 } 2275 } 2276 } 2277 2278 chain->setInBuffer(buffer, ownsBuffer); 2279 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2280 ? mEffectBuffer : mSinkBuffer)); 2281 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2282 // chains list in order to be processed last as it contains output stage effects 2283 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2284 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2285 // after track specific effects and before output stage 2286 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2287 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2288 // Effect chain for other sessions are inserted at beginning of effect 2289 // chains list to be processed before output mix effects. Relative order between other 2290 // sessions is not important 2291 size_t size = mEffectChains.size(); 2292 size_t i = 0; 2293 for (i = 0; i < size; i++) { 2294 if (mEffectChains[i]->sessionId() < session) { 2295 break; 2296 } 2297 } 2298 mEffectChains.insertAt(chain, i); 2299 checkSuspendOnAddEffectChain_l(chain); 2300 2301 return NO_ERROR; 2302} 2303 2304size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2305{ 2306 int session = chain->sessionId(); 2307 2308 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2309 2310 for (size_t i = 0; i < mEffectChains.size(); i++) { 2311 if (chain == mEffectChains[i]) { 2312 mEffectChains.removeAt(i); 2313 // detach all active tracks from the chain 2314 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2315 sp<Track> track = mActiveTracks[i].promote(); 2316 if (track == 0) { 2317 continue; 2318 } 2319 if (session == track->sessionId()) { 2320 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2321 chain.get(), session); 2322 chain->decActiveTrackCnt(); 2323 } 2324 } 2325 2326 // detach all tracks with same session ID from this chain 2327 for (size_t i = 0; i < mTracks.size(); ++i) { 2328 sp<Track> track = mTracks[i]; 2329 if (session == track->sessionId()) { 2330 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2331 chain->decTrackCnt(); 2332 } 2333 } 2334 break; 2335 } 2336 } 2337 return mEffectChains.size(); 2338} 2339 2340status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2341 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2342{ 2343 Mutex::Autolock _l(mLock); 2344 return attachAuxEffect_l(track, EffectId); 2345} 2346 2347status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2348 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2349{ 2350 status_t status = NO_ERROR; 2351 2352 if (EffectId == 0) { 2353 track->setAuxBuffer(0, NULL); 2354 } else { 2355 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2356 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2357 if (effect != 0) { 2358 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2359 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2360 } else { 2361 status = INVALID_OPERATION; 2362 } 2363 } else { 2364 status = BAD_VALUE; 2365 } 2366 } 2367 return status; 2368} 2369 2370void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2371{ 2372 for (size_t i = 0; i < mTracks.size(); ++i) { 2373 sp<Track> track = mTracks[i]; 2374 if (track->auxEffectId() == effectId) { 2375 attachAuxEffect_l(track, 0); 2376 } 2377 } 2378} 2379 2380bool AudioFlinger::PlaybackThread::threadLoop() 2381{ 2382 Vector< sp<Track> > tracksToRemove; 2383 2384 standbyTime = systemTime(); 2385 2386 // MIXER 2387 nsecs_t lastWarning = 0; 2388 2389 // DUPLICATING 2390 // FIXME could this be made local to while loop? 2391 writeFrames = 0; 2392 2393 int lastGeneration = 0; 2394 2395 cacheParameters_l(); 2396 sleepTime = idleSleepTime; 2397 2398 if (mType == MIXER) { 2399 sleepTimeShift = 0; 2400 } 2401 2402 CpuStats cpuStats; 2403 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2404 2405 acquireWakeLock(); 2406 2407 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2408 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2409 // and then that string will be logged at the next convenient opportunity. 2410 const char *logString = NULL; 2411 2412 checkSilentMode_l(); 2413 2414 while (!exitPending()) 2415 { 2416 cpuStats.sample(myName); 2417 2418 Vector< sp<EffectChain> > effectChains; 2419 2420 { // scope for mLock 2421 2422 Mutex::Autolock _l(mLock); 2423 2424 processConfigEvents_l(); 2425 2426 if (logString != NULL) { 2427 mNBLogWriter->logTimestamp(); 2428 mNBLogWriter->log(logString); 2429 logString = NULL; 2430 } 2431 2432 if (mLatchDValid) { 2433 mLatchQ = mLatchD; 2434 mLatchDValid = false; 2435 mLatchQValid = true; 2436 } 2437 2438 saveOutputTracks(); 2439 if (mSignalPending) { 2440 // A signal was raised while we were unlocked 2441 mSignalPending = false; 2442 } else if (waitingAsyncCallback_l()) { 2443 if (exitPending()) { 2444 break; 2445 } 2446 releaseWakeLock_l(); 2447 mWakeLockUids.clear(); 2448 mActiveTracksGeneration++; 2449 ALOGV("wait async completion"); 2450 mWaitWorkCV.wait(mLock); 2451 ALOGV("async completion/wake"); 2452 acquireWakeLock_l(); 2453 standbyTime = systemTime() + standbyDelay; 2454 sleepTime = 0; 2455 2456 continue; 2457 } 2458 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2459 isSuspended()) { 2460 // put audio hardware into standby after short delay 2461 if (shouldStandby_l()) { 2462 2463 threadLoop_standby(); 2464 2465 mStandby = true; 2466 } 2467 2468 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2469 // we're about to wait, flush the binder command buffer 2470 IPCThreadState::self()->flushCommands(); 2471 2472 clearOutputTracks(); 2473 2474 if (exitPending()) { 2475 break; 2476 } 2477 2478 releaseWakeLock_l(); 2479 mWakeLockUids.clear(); 2480 mActiveTracksGeneration++; 2481 // wait until we have something to do... 2482 ALOGV("%s going to sleep", myName.string()); 2483 mWaitWorkCV.wait(mLock); 2484 ALOGV("%s waking up", myName.string()); 2485 acquireWakeLock_l(); 2486 2487 mMixerStatus = MIXER_IDLE; 2488 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2489 mBytesWritten = 0; 2490 mBytesRemaining = 0; 2491 checkSilentMode_l(); 2492 2493 standbyTime = systemTime() + standbyDelay; 2494 sleepTime = idleSleepTime; 2495 if (mType == MIXER) { 2496 sleepTimeShift = 0; 2497 } 2498 2499 continue; 2500 } 2501 } 2502 // mMixerStatusIgnoringFastTracks is also updated internally 2503 mMixerStatus = prepareTracks_l(&tracksToRemove); 2504 2505 // compare with previously applied list 2506 if (lastGeneration != mActiveTracksGeneration) { 2507 // update wakelock 2508 updateWakeLockUids_l(mWakeLockUids); 2509 lastGeneration = mActiveTracksGeneration; 2510 } 2511 2512 // prevent any changes in effect chain list and in each effect chain 2513 // during mixing and effect process as the audio buffers could be deleted 2514 // or modified if an effect is created or deleted 2515 lockEffectChains_l(effectChains); 2516 } // mLock scope ends 2517 2518 if (mBytesRemaining == 0) { 2519 mCurrentWriteLength = 0; 2520 if (mMixerStatus == MIXER_TRACKS_READY) { 2521 // threadLoop_mix() sets mCurrentWriteLength 2522 threadLoop_mix(); 2523 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2524 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2525 // threadLoop_sleepTime sets sleepTime to 0 if data 2526 // must be written to HAL 2527 threadLoop_sleepTime(); 2528 if (sleepTime == 0) { 2529 mCurrentWriteLength = mSinkBufferSize; 2530 } 2531 } 2532 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2533 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2534 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2535 // or mSinkBuffer (if there are no effects). 2536 // 2537 // This is done pre-effects computation; if effects change to 2538 // support higher precision, this needs to move. 2539 // 2540 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2541 // TODO use sleepTime == 0 as an additional condition. 2542 if (mMixerBufferValid) { 2543 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2544 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2545 2546 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2547 mNormalFrameCount * mChannelCount); 2548 } 2549 2550 mBytesRemaining = mCurrentWriteLength; 2551 if (isSuspended()) { 2552 sleepTime = suspendSleepTimeUs(); 2553 // simulate write to HAL when suspended 2554 mBytesWritten += mSinkBufferSize; 2555 mBytesRemaining = 0; 2556 } 2557 2558 // only process effects if we're going to write 2559 if (sleepTime == 0 && mType != OFFLOAD) { 2560 for (size_t i = 0; i < effectChains.size(); i ++) { 2561 effectChains[i]->process_l(); 2562 } 2563 } 2564 } 2565 // Process effect chains for offloaded thread even if no audio 2566 // was read from audio track: process only updates effect state 2567 // and thus does have to be synchronized with audio writes but may have 2568 // to be called while waiting for async write callback 2569 if (mType == OFFLOAD) { 2570 for (size_t i = 0; i < effectChains.size(); i ++) { 2571 effectChains[i]->process_l(); 2572 } 2573 } 2574 2575 // Only if the Effects buffer is enabled and there is data in the 2576 // Effects buffer (buffer valid), we need to 2577 // copy into the sink buffer. 2578 // TODO use sleepTime == 0 as an additional condition. 2579 if (mEffectBufferValid) { 2580 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2581 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2582 mNormalFrameCount * mChannelCount); 2583 } 2584 2585 // enable changes in effect chain 2586 unlockEffectChains(effectChains); 2587 2588 if (!waitingAsyncCallback()) { 2589 // sleepTime == 0 means we must write to audio hardware 2590 if (sleepTime == 0) { 2591 if (mBytesRemaining) { 2592 ssize_t ret = threadLoop_write(); 2593 if (ret < 0) { 2594 mBytesRemaining = 0; 2595 } else { 2596 mBytesWritten += ret; 2597 mBytesRemaining -= ret; 2598 } 2599 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2600 (mMixerStatus == MIXER_DRAIN_ALL)) { 2601 threadLoop_drain(); 2602 } 2603 if (mType == MIXER) { 2604 // write blocked detection 2605 nsecs_t now = systemTime(); 2606 nsecs_t delta = now - mLastWriteTime; 2607 if (!mStandby && delta > maxPeriod) { 2608 mNumDelayedWrites++; 2609 if ((now - lastWarning) > kWarningThrottleNs) { 2610 ATRACE_NAME("underrun"); 2611 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2612 ns2ms(delta), mNumDelayedWrites, this); 2613 lastWarning = now; 2614 } 2615 } 2616 } 2617 2618 } else { 2619 usleep(sleepTime); 2620 } 2621 } 2622 2623 // Finally let go of removed track(s), without the lock held 2624 // since we can't guarantee the destructors won't acquire that 2625 // same lock. This will also mutate and push a new fast mixer state. 2626 threadLoop_removeTracks(tracksToRemove); 2627 tracksToRemove.clear(); 2628 2629 // FIXME I don't understand the need for this here; 2630 // it was in the original code but maybe the 2631 // assignment in saveOutputTracks() makes this unnecessary? 2632 clearOutputTracks(); 2633 2634 // Effect chains will be actually deleted here if they were removed from 2635 // mEffectChains list during mixing or effects processing 2636 effectChains.clear(); 2637 2638 // FIXME Note that the above .clear() is no longer necessary since effectChains 2639 // is now local to this block, but will keep it for now (at least until merge done). 2640 } 2641 2642 threadLoop_exit(); 2643 2644 if (!mStandby) { 2645 threadLoop_standby(); 2646 mStandby = true; 2647 } 2648 2649 releaseWakeLock(); 2650 mWakeLockUids.clear(); 2651 mActiveTracksGeneration++; 2652 2653 ALOGV("Thread %p type %d exiting", this, mType); 2654 return false; 2655} 2656 2657// removeTracks_l() must be called with ThreadBase::mLock held 2658void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2659{ 2660 size_t count = tracksToRemove.size(); 2661 if (count > 0) { 2662 for (size_t i=0 ; i<count ; i++) { 2663 const sp<Track>& track = tracksToRemove.itemAt(i); 2664 mActiveTracks.remove(track); 2665 mWakeLockUids.remove(track->uid()); 2666 mActiveTracksGeneration++; 2667 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2668 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2669 if (chain != 0) { 2670 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2671 track->sessionId()); 2672 chain->decActiveTrackCnt(); 2673 } 2674 if (track->isTerminated()) { 2675 removeTrack_l(track); 2676 } 2677 } 2678 } 2679 2680} 2681 2682status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2683{ 2684 if (mNormalSink != 0) { 2685 return mNormalSink->getTimestamp(timestamp); 2686 } 2687 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2688 uint64_t position64; 2689 int ret = mOutput->stream->get_presentation_position( 2690 mOutput->stream, &position64, ×tamp.mTime); 2691 if (ret == 0) { 2692 timestamp.mPosition = (uint32_t)position64; 2693 return NO_ERROR; 2694 } 2695 } 2696 return INVALID_OPERATION; 2697} 2698 2699status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2700 audio_patch_handle_t *handle) 2701{ 2702 status_t status = NO_ERROR; 2703 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2704 // store new device and send to effects 2705 audio_devices_t type = AUDIO_DEVICE_NONE; 2706 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2707 type |= patch->sinks[i].ext.device.type; 2708 } 2709 mOutDevice = type; 2710 for (size_t i = 0; i < mEffectChains.size(); i++) { 2711 mEffectChains[i]->setDevice_l(mOutDevice); 2712 } 2713 2714 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2715 status = hwDevice->create_audio_patch(hwDevice, 2716 patch->num_sources, 2717 patch->sources, 2718 patch->num_sinks, 2719 patch->sinks, 2720 handle); 2721 } else { 2722 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2723 } 2724 return status; 2725} 2726 2727status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2728{ 2729 status_t status = NO_ERROR; 2730 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2731 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2732 status = hwDevice->release_audio_patch(hwDevice, handle); 2733 } else { 2734 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2735 } 2736 return status; 2737} 2738 2739void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2740{ 2741 Mutex::Autolock _l(mLock); 2742 mTracks.add(track); 2743} 2744 2745void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2746{ 2747 Mutex::Autolock _l(mLock); 2748 destroyTrack_l(track); 2749} 2750 2751void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2752{ 2753 ThreadBase::getAudioPortConfig(config); 2754 config->role = AUDIO_PORT_ROLE_SOURCE; 2755 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2756 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2757} 2758 2759// ---------------------------------------------------------------------------- 2760 2761AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2762 audio_io_handle_t id, audio_devices_t device, type_t type) 2763 : PlaybackThread(audioFlinger, output, id, device, type), 2764 // mAudioMixer below 2765 // mFastMixer below 2766 mFastMixerFutex(0) 2767 // mOutputSink below 2768 // mPipeSink below 2769 // mNormalSink below 2770{ 2771 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2772 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2773 "mFrameCount=%d, mNormalFrameCount=%d", 2774 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2775 mNormalFrameCount); 2776 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2777 2778 // create an NBAIO sink for the HAL output stream, and negotiate 2779 mOutputSink = new AudioStreamOutSink(output->stream); 2780 size_t numCounterOffers = 0; 2781 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2782 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2783 ALOG_ASSERT(index == 0); 2784 2785 // initialize fast mixer depending on configuration 2786 bool initFastMixer; 2787 switch (kUseFastMixer) { 2788 case FastMixer_Never: 2789 initFastMixer = false; 2790 break; 2791 case FastMixer_Always: 2792 initFastMixer = true; 2793 break; 2794 case FastMixer_Static: 2795 case FastMixer_Dynamic: 2796 initFastMixer = mFrameCount < mNormalFrameCount; 2797 break; 2798 } 2799 if (initFastMixer) { 2800 audio_format_t fastMixerFormat; 2801 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2802 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2803 } else { 2804 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2805 } 2806 if (mFormat != fastMixerFormat) { 2807 // change our Sink format to accept our intermediate precision 2808 mFormat = fastMixerFormat; 2809 free(mSinkBuffer); 2810 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2811 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2812 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2813 } 2814 2815 // create a MonoPipe to connect our submix to FastMixer 2816 NBAIO_Format format = mOutputSink->format(); 2817 // adjust format to match that of the Fast Mixer 2818 format.mFormat = fastMixerFormat; 2819 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2820 2821 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2822 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2823 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2824 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2825 const NBAIO_Format offers[1] = {format}; 2826 size_t numCounterOffers = 0; 2827 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2828 ALOG_ASSERT(index == 0); 2829 monoPipe->setAvgFrames((mScreenState & 1) ? 2830 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2831 mPipeSink = monoPipe; 2832 2833#ifdef TEE_SINK 2834 if (mTeeSinkOutputEnabled) { 2835 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2836 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2837 numCounterOffers = 0; 2838 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2839 ALOG_ASSERT(index == 0); 2840 mTeeSink = teeSink; 2841 PipeReader *teeSource = new PipeReader(*teeSink); 2842 numCounterOffers = 0; 2843 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2844 ALOG_ASSERT(index == 0); 2845 mTeeSource = teeSource; 2846 } 2847#endif 2848 2849 // create fast mixer and configure it initially with just one fast track for our submix 2850 mFastMixer = new FastMixer(); 2851 FastMixerStateQueue *sq = mFastMixer->sq(); 2852#ifdef STATE_QUEUE_DUMP 2853 sq->setObserverDump(&mStateQueueObserverDump); 2854 sq->setMutatorDump(&mStateQueueMutatorDump); 2855#endif 2856 FastMixerState *state = sq->begin(); 2857 FastTrack *fastTrack = &state->mFastTracks[0]; 2858 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2859 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2860 fastTrack->mVolumeProvider = NULL; 2861 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2862 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2863 fastTrack->mGeneration++; 2864 state->mFastTracksGen++; 2865 state->mTrackMask = 1; 2866 // fast mixer will use the HAL output sink 2867 state->mOutputSink = mOutputSink.get(); 2868 state->mOutputSinkGen++; 2869 state->mFrameCount = mFrameCount; 2870 state->mCommand = FastMixerState::COLD_IDLE; 2871 // already done in constructor initialization list 2872 //mFastMixerFutex = 0; 2873 state->mColdFutexAddr = &mFastMixerFutex; 2874 state->mColdGen++; 2875 state->mDumpState = &mFastMixerDumpState; 2876#ifdef TEE_SINK 2877 state->mTeeSink = mTeeSink.get(); 2878#endif 2879 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2880 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2881 sq->end(); 2882 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2883 2884 // start the fast mixer 2885 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2886 pid_t tid = mFastMixer->getTid(); 2887 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2888 if (err != 0) { 2889 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2890 kPriorityFastMixer, getpid_cached, tid, err); 2891 } 2892 2893#ifdef AUDIO_WATCHDOG 2894 // create and start the watchdog 2895 mAudioWatchdog = new AudioWatchdog(); 2896 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2897 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2898 tid = mAudioWatchdog->getTid(); 2899 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2900 if (err != 0) { 2901 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2902 kPriorityFastMixer, getpid_cached, tid, err); 2903 } 2904#endif 2905 2906 } 2907 2908 switch (kUseFastMixer) { 2909 case FastMixer_Never: 2910 case FastMixer_Dynamic: 2911 mNormalSink = mOutputSink; 2912 break; 2913 case FastMixer_Always: 2914 mNormalSink = mPipeSink; 2915 break; 2916 case FastMixer_Static: 2917 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2918 break; 2919 } 2920} 2921 2922AudioFlinger::MixerThread::~MixerThread() 2923{ 2924 if (mFastMixer != 0) { 2925 FastMixerStateQueue *sq = mFastMixer->sq(); 2926 FastMixerState *state = sq->begin(); 2927 if (state->mCommand == FastMixerState::COLD_IDLE) { 2928 int32_t old = android_atomic_inc(&mFastMixerFutex); 2929 if (old == -1) { 2930 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2931 } 2932 } 2933 state->mCommand = FastMixerState::EXIT; 2934 sq->end(); 2935 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2936 mFastMixer->join(); 2937 // Though the fast mixer thread has exited, it's state queue is still valid. 2938 // We'll use that extract the final state which contains one remaining fast track 2939 // corresponding to our sub-mix. 2940 state = sq->begin(); 2941 ALOG_ASSERT(state->mTrackMask == 1); 2942 FastTrack *fastTrack = &state->mFastTracks[0]; 2943 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2944 delete fastTrack->mBufferProvider; 2945 sq->end(false /*didModify*/); 2946 mFastMixer.clear(); 2947#ifdef AUDIO_WATCHDOG 2948 if (mAudioWatchdog != 0) { 2949 mAudioWatchdog->requestExit(); 2950 mAudioWatchdog->requestExitAndWait(); 2951 mAudioWatchdog.clear(); 2952 } 2953#endif 2954 } 2955 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2956 delete mAudioMixer; 2957} 2958 2959 2960uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2961{ 2962 if (mFastMixer != 0) { 2963 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2964 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2965 } 2966 return latency; 2967} 2968 2969 2970void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2971{ 2972 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2973} 2974 2975ssize_t AudioFlinger::MixerThread::threadLoop_write() 2976{ 2977 // FIXME we should only do one push per cycle; confirm this is true 2978 // Start the fast mixer if it's not already running 2979 if (mFastMixer != 0) { 2980 FastMixerStateQueue *sq = mFastMixer->sq(); 2981 FastMixerState *state = sq->begin(); 2982 if (state->mCommand != FastMixerState::MIX_WRITE && 2983 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2984 if (state->mCommand == FastMixerState::COLD_IDLE) { 2985 int32_t old = android_atomic_inc(&mFastMixerFutex); 2986 if (old == -1) { 2987 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2988 } 2989#ifdef AUDIO_WATCHDOG 2990 if (mAudioWatchdog != 0) { 2991 mAudioWatchdog->resume(); 2992 } 2993#endif 2994 } 2995 state->mCommand = FastMixerState::MIX_WRITE; 2996 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2997 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2998 sq->end(); 2999 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3000 if (kUseFastMixer == FastMixer_Dynamic) { 3001 mNormalSink = mPipeSink; 3002 } 3003 } else { 3004 sq->end(false /*didModify*/); 3005 } 3006 } 3007 return PlaybackThread::threadLoop_write(); 3008} 3009 3010void AudioFlinger::MixerThread::threadLoop_standby() 3011{ 3012 // Idle the fast mixer if it's currently running 3013 if (mFastMixer != 0) { 3014 FastMixerStateQueue *sq = mFastMixer->sq(); 3015 FastMixerState *state = sq->begin(); 3016 if (!(state->mCommand & FastMixerState::IDLE)) { 3017 state->mCommand = FastMixerState::COLD_IDLE; 3018 state->mColdFutexAddr = &mFastMixerFutex; 3019 state->mColdGen++; 3020 mFastMixerFutex = 0; 3021 sq->end(); 3022 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3023 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3024 if (kUseFastMixer == FastMixer_Dynamic) { 3025 mNormalSink = mOutputSink; 3026 } 3027#ifdef AUDIO_WATCHDOG 3028 if (mAudioWatchdog != 0) { 3029 mAudioWatchdog->pause(); 3030 } 3031#endif 3032 } else { 3033 sq->end(false /*didModify*/); 3034 } 3035 } 3036 PlaybackThread::threadLoop_standby(); 3037} 3038 3039bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3040{ 3041 return false; 3042} 3043 3044bool AudioFlinger::PlaybackThread::shouldStandby_l() 3045{ 3046 return !mStandby; 3047} 3048 3049bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3050{ 3051 Mutex::Autolock _l(mLock); 3052 return waitingAsyncCallback_l(); 3053} 3054 3055// shared by MIXER and DIRECT, overridden by DUPLICATING 3056void AudioFlinger::PlaybackThread::threadLoop_standby() 3057{ 3058 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3059 mOutput->stream->common.standby(&mOutput->stream->common); 3060 if (mUseAsyncWrite != 0) { 3061 // discard any pending drain or write ack by incrementing sequence 3062 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3063 mDrainSequence = (mDrainSequence + 2) & ~1; 3064 ALOG_ASSERT(mCallbackThread != 0); 3065 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3066 mCallbackThread->setDraining(mDrainSequence); 3067 } 3068} 3069 3070void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3071{ 3072 ALOGV("signal playback thread"); 3073 broadcast_l(); 3074} 3075 3076void AudioFlinger::MixerThread::threadLoop_mix() 3077{ 3078 // obtain the presentation timestamp of the next output buffer 3079 int64_t pts; 3080 status_t status = INVALID_OPERATION; 3081 3082 if (mNormalSink != 0) { 3083 status = mNormalSink->getNextWriteTimestamp(&pts); 3084 } else { 3085 status = mOutputSink->getNextWriteTimestamp(&pts); 3086 } 3087 3088 if (status != NO_ERROR) { 3089 pts = AudioBufferProvider::kInvalidPTS; 3090 } 3091 3092 // mix buffers... 3093 mAudioMixer->process(pts); 3094 mCurrentWriteLength = mSinkBufferSize; 3095 // increase sleep time progressively when application underrun condition clears. 3096 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3097 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3098 // such that we would underrun the audio HAL. 3099 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3100 sleepTimeShift--; 3101 } 3102 sleepTime = 0; 3103 standbyTime = systemTime() + standbyDelay; 3104 //TODO: delay standby when effects have a tail 3105} 3106 3107void AudioFlinger::MixerThread::threadLoop_sleepTime() 3108{ 3109 // If no tracks are ready, sleep once for the duration of an output 3110 // buffer size, then write 0s to the output 3111 if (sleepTime == 0) { 3112 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3113 sleepTime = activeSleepTime >> sleepTimeShift; 3114 if (sleepTime < kMinThreadSleepTimeUs) { 3115 sleepTime = kMinThreadSleepTimeUs; 3116 } 3117 // reduce sleep time in case of consecutive application underruns to avoid 3118 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3119 // duration we would end up writing less data than needed by the audio HAL if 3120 // the condition persists. 3121 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3122 sleepTimeShift++; 3123 } 3124 } else { 3125 sleepTime = idleSleepTime; 3126 } 3127 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3128 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3129 // before effects processing or output. 3130 if (mMixerBufferValid) { 3131 memset(mMixerBuffer, 0, mMixerBufferSize); 3132 } else { 3133 memset(mSinkBuffer, 0, mSinkBufferSize); 3134 } 3135 sleepTime = 0; 3136 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3137 "anticipated start"); 3138 } 3139 // TODO add standby time extension fct of effect tail 3140} 3141 3142// prepareTracks_l() must be called with ThreadBase::mLock held 3143AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3144 Vector< sp<Track> > *tracksToRemove) 3145{ 3146 3147 mixer_state mixerStatus = MIXER_IDLE; 3148 // find out which tracks need to be processed 3149 size_t count = mActiveTracks.size(); 3150 size_t mixedTracks = 0; 3151 size_t tracksWithEffect = 0; 3152 // counts only _active_ fast tracks 3153 size_t fastTracks = 0; 3154 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3155 3156 float masterVolume = mMasterVolume; 3157 bool masterMute = mMasterMute; 3158 3159 if (masterMute) { 3160 masterVolume = 0; 3161 } 3162 // Delegate master volume control to effect in output mix effect chain if needed 3163 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3164 if (chain != 0) { 3165 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3166 chain->setVolume_l(&v, &v); 3167 masterVolume = (float)((v + (1 << 23)) >> 24); 3168 chain.clear(); 3169 } 3170 3171 // prepare a new state to push 3172 FastMixerStateQueue *sq = NULL; 3173 FastMixerState *state = NULL; 3174 bool didModify = false; 3175 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3176 if (mFastMixer != 0) { 3177 sq = mFastMixer->sq(); 3178 state = sq->begin(); 3179 } 3180 3181 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3182 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3183 3184 for (size_t i=0 ; i<count ; i++) { 3185 const sp<Track> t = mActiveTracks[i].promote(); 3186 if (t == 0) { 3187 continue; 3188 } 3189 3190 // this const just means the local variable doesn't change 3191 Track* const track = t.get(); 3192 3193 // process fast tracks 3194 if (track->isFastTrack()) { 3195 3196 // It's theoretically possible (though unlikely) for a fast track to be created 3197 // and then removed within the same normal mix cycle. This is not a problem, as 3198 // the track never becomes active so it's fast mixer slot is never touched. 3199 // The converse, of removing an (active) track and then creating a new track 3200 // at the identical fast mixer slot within the same normal mix cycle, 3201 // is impossible because the slot isn't marked available until the end of each cycle. 3202 int j = track->mFastIndex; 3203 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3204 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3205 FastTrack *fastTrack = &state->mFastTracks[j]; 3206 3207 // Determine whether the track is currently in underrun condition, 3208 // and whether it had a recent underrun. 3209 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3210 FastTrackUnderruns underruns = ftDump->mUnderruns; 3211 uint32_t recentFull = (underruns.mBitFields.mFull - 3212 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3213 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3214 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3215 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3216 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3217 uint32_t recentUnderruns = recentPartial + recentEmpty; 3218 track->mObservedUnderruns = underruns; 3219 // don't count underruns that occur while stopping or pausing 3220 // or stopped which can occur when flush() is called while active 3221 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3222 recentUnderruns > 0) { 3223 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3224 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3225 } 3226 3227 // This is similar to the state machine for normal tracks, 3228 // with a few modifications for fast tracks. 3229 bool isActive = true; 3230 switch (track->mState) { 3231 case TrackBase::STOPPING_1: 3232 // track stays active in STOPPING_1 state until first underrun 3233 if (recentUnderruns > 0 || track->isTerminated()) { 3234 track->mState = TrackBase::STOPPING_2; 3235 } 3236 break; 3237 case TrackBase::PAUSING: 3238 // ramp down is not yet implemented 3239 track->setPaused(); 3240 break; 3241 case TrackBase::RESUMING: 3242 // ramp up is not yet implemented 3243 track->mState = TrackBase::ACTIVE; 3244 break; 3245 case TrackBase::ACTIVE: 3246 if (recentFull > 0 || recentPartial > 0) { 3247 // track has provided at least some frames recently: reset retry count 3248 track->mRetryCount = kMaxTrackRetries; 3249 } 3250 if (recentUnderruns == 0) { 3251 // no recent underruns: stay active 3252 break; 3253 } 3254 // there has recently been an underrun of some kind 3255 if (track->sharedBuffer() == 0) { 3256 // were any of the recent underruns "empty" (no frames available)? 3257 if (recentEmpty == 0) { 3258 // no, then ignore the partial underruns as they are allowed indefinitely 3259 break; 3260 } 3261 // there has recently been an "empty" underrun: decrement the retry counter 3262 if (--(track->mRetryCount) > 0) { 3263 break; 3264 } 3265 // indicate to client process that the track was disabled because of underrun; 3266 // it will then automatically call start() when data is available 3267 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3268 // remove from active list, but state remains ACTIVE [confusing but true] 3269 isActive = false; 3270 break; 3271 } 3272 // fall through 3273 case TrackBase::STOPPING_2: 3274 case TrackBase::PAUSED: 3275 case TrackBase::STOPPED: 3276 case TrackBase::FLUSHED: // flush() while active 3277 // Check for presentation complete if track is inactive 3278 // We have consumed all the buffers of this track. 3279 // This would be incomplete if we auto-paused on underrun 3280 { 3281 size_t audioHALFrames = 3282 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3283 size_t framesWritten = mBytesWritten / mFrameSize; 3284 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3285 // track stays in active list until presentation is complete 3286 break; 3287 } 3288 } 3289 if (track->isStopping_2()) { 3290 track->mState = TrackBase::STOPPED; 3291 } 3292 if (track->isStopped()) { 3293 // Can't reset directly, as fast mixer is still polling this track 3294 // track->reset(); 3295 // So instead mark this track as needing to be reset after push with ack 3296 resetMask |= 1 << i; 3297 } 3298 isActive = false; 3299 break; 3300 case TrackBase::IDLE: 3301 default: 3302 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3303 } 3304 3305 if (isActive) { 3306 // was it previously inactive? 3307 if (!(state->mTrackMask & (1 << j))) { 3308 ExtendedAudioBufferProvider *eabp = track; 3309 VolumeProvider *vp = track; 3310 fastTrack->mBufferProvider = eabp; 3311 fastTrack->mVolumeProvider = vp; 3312 fastTrack->mChannelMask = track->mChannelMask; 3313 fastTrack->mFormat = track->mFormat; 3314 fastTrack->mGeneration++; 3315 state->mTrackMask |= 1 << j; 3316 didModify = true; 3317 // no acknowledgement required for newly active tracks 3318 } 3319 // cache the combined master volume and stream type volume for fast mixer; this 3320 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3321 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3322 ++fastTracks; 3323 } else { 3324 // was it previously active? 3325 if (state->mTrackMask & (1 << j)) { 3326 fastTrack->mBufferProvider = NULL; 3327 fastTrack->mGeneration++; 3328 state->mTrackMask &= ~(1 << j); 3329 didModify = true; 3330 // If any fast tracks were removed, we must wait for acknowledgement 3331 // because we're about to decrement the last sp<> on those tracks. 3332 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3333 } else { 3334 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3335 } 3336 tracksToRemove->add(track); 3337 // Avoids a misleading display in dumpsys 3338 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3339 } 3340 continue; 3341 } 3342 3343 { // local variable scope to avoid goto warning 3344 3345 audio_track_cblk_t* cblk = track->cblk(); 3346 3347 // The first time a track is added we wait 3348 // for all its buffers to be filled before processing it 3349 int name = track->name(); 3350 // make sure that we have enough frames to mix one full buffer. 3351 // enforce this condition only once to enable draining the buffer in case the client 3352 // app does not call stop() and relies on underrun to stop: 3353 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3354 // during last round 3355 size_t desiredFrames; 3356 uint32_t sr = track->sampleRate(); 3357 if (sr == mSampleRate) { 3358 desiredFrames = mNormalFrameCount; 3359 } else { 3360 // +1 for rounding and +1 for additional sample needed for interpolation 3361 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3362 // add frames already consumed but not yet released by the resampler 3363 // because mAudioTrackServerProxy->framesReady() will include these frames 3364 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3365#if 0 3366 // the minimum track buffer size is normally twice the number of frames necessary 3367 // to fill one buffer and the resampler should not leave more than one buffer worth 3368 // of unreleased frames after each pass, but just in case... 3369 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3370#endif 3371 } 3372 uint32_t minFrames = 1; 3373 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3374 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3375 minFrames = desiredFrames; 3376 } 3377 3378 size_t framesReady = track->framesReady(); 3379 if ((framesReady >= minFrames) && track->isReady() && 3380 !track->isPaused() && !track->isTerminated()) 3381 { 3382 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3383 3384 mixedTracks++; 3385 3386 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3387 // there is an effect chain connected to the track 3388 chain.clear(); 3389 if (track->mainBuffer() != mSinkBuffer && 3390 track->mainBuffer() != mMixerBuffer) { 3391 if (mEffectBufferEnabled) { 3392 mEffectBufferValid = true; // Later can set directly. 3393 } 3394 chain = getEffectChain_l(track->sessionId()); 3395 // Delegate volume control to effect in track effect chain if needed 3396 if (chain != 0) { 3397 tracksWithEffect++; 3398 } else { 3399 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3400 "session %d", 3401 name, track->sessionId()); 3402 } 3403 } 3404 3405 3406 int param = AudioMixer::VOLUME; 3407 if (track->mFillingUpStatus == Track::FS_FILLED) { 3408 // no ramp for the first volume setting 3409 track->mFillingUpStatus = Track::FS_ACTIVE; 3410 if (track->mState == TrackBase::RESUMING) { 3411 track->mState = TrackBase::ACTIVE; 3412 param = AudioMixer::RAMP_VOLUME; 3413 } 3414 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3415 // FIXME should not make a decision based on mServer 3416 } else if (cblk->mServer != 0) { 3417 // If the track is stopped before the first frame was mixed, 3418 // do not apply ramp 3419 param = AudioMixer::RAMP_VOLUME; 3420 } 3421 3422 // compute volume for this track 3423 uint32_t vl, vr; // in U8.24 integer format 3424 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3425 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3426 vl = vr = 0; 3427 vlf = vrf = vaf = 0.; 3428 if (track->isPausing()) { 3429 track->setPaused(); 3430 } 3431 } else { 3432 3433 // read original volumes with volume control 3434 float typeVolume = mStreamTypes[track->streamType()].volume; 3435 float v = masterVolume * typeVolume; 3436 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3437 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3438 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3439 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3440 // track volumes come from shared memory, so can't be trusted and must be clamped 3441 if (vlf > GAIN_FLOAT_UNITY) { 3442 ALOGV("Track left volume out of range: %.3g", vlf); 3443 vlf = GAIN_FLOAT_UNITY; 3444 } 3445 if (vrf > GAIN_FLOAT_UNITY) { 3446 ALOGV("Track right volume out of range: %.3g", vrf); 3447 vrf = GAIN_FLOAT_UNITY; 3448 } 3449 // now apply the master volume and stream type volume 3450 vlf *= v; 3451 vrf *= v; 3452 // assuming master volume and stream type volume each go up to 1.0, 3453 // then derive vl and vr as U8.24 versions for the effect chain 3454 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3455 vl = (uint32_t) (scaleto8_24 * vlf); 3456 vr = (uint32_t) (scaleto8_24 * vrf); 3457 // vl and vr are now in U8.24 format 3458 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3459 // send level comes from shared memory and so may be corrupt 3460 if (sendLevel > MAX_GAIN_INT) { 3461 ALOGV("Track send level out of range: %04X", sendLevel); 3462 sendLevel = MAX_GAIN_INT; 3463 } 3464 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3465 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3466 } 3467 3468 // Delegate volume control to effect in track effect chain if needed 3469 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3470 // Do not ramp volume if volume is controlled by effect 3471 param = AudioMixer::VOLUME; 3472 // Update remaining floating point volume levels 3473 vlf = (float)vl / (1 << 24); 3474 vrf = (float)vr / (1 << 24); 3475 track->mHasVolumeController = true; 3476 } else { 3477 // force no volume ramp when volume controller was just disabled or removed 3478 // from effect chain to avoid volume spike 3479 if (track->mHasVolumeController) { 3480 param = AudioMixer::VOLUME; 3481 } 3482 track->mHasVolumeController = false; 3483 } 3484 3485 // XXX: these things DON'T need to be done each time 3486 mAudioMixer->setBufferProvider(name, track); 3487 mAudioMixer->enable(name); 3488 3489 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3490 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3491 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3492 mAudioMixer->setParameter( 3493 name, 3494 AudioMixer::TRACK, 3495 AudioMixer::FORMAT, (void *)track->format()); 3496 mAudioMixer->setParameter( 3497 name, 3498 AudioMixer::TRACK, 3499 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3500 mAudioMixer->setParameter( 3501 name, 3502 AudioMixer::TRACK, 3503 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3504 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3505 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3506 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3507 if (reqSampleRate == 0) { 3508 reqSampleRate = mSampleRate; 3509 } else if (reqSampleRate > maxSampleRate) { 3510 reqSampleRate = maxSampleRate; 3511 } 3512 mAudioMixer->setParameter( 3513 name, 3514 AudioMixer::RESAMPLE, 3515 AudioMixer::SAMPLE_RATE, 3516 (void *)(uintptr_t)reqSampleRate); 3517 /* 3518 * Select the appropriate output buffer for the track. 3519 * 3520 * Tracks with effects go into their own effects chain buffer 3521 * and from there into either mEffectBuffer or mSinkBuffer. 3522 * 3523 * Other tracks can use mMixerBuffer for higher precision 3524 * channel accumulation. If this buffer is enabled 3525 * (mMixerBufferEnabled true), then selected tracks will accumulate 3526 * into it. 3527 * 3528 */ 3529 if (mMixerBufferEnabled 3530 && (track->mainBuffer() == mSinkBuffer 3531 || track->mainBuffer() == mMixerBuffer)) { 3532 mAudioMixer->setParameter( 3533 name, 3534 AudioMixer::TRACK, 3535 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3536 mAudioMixer->setParameter( 3537 name, 3538 AudioMixer::TRACK, 3539 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3540 // TODO: override track->mainBuffer()? 3541 mMixerBufferValid = true; 3542 } else { 3543 mAudioMixer->setParameter( 3544 name, 3545 AudioMixer::TRACK, 3546 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3547 mAudioMixer->setParameter( 3548 name, 3549 AudioMixer::TRACK, 3550 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3551 } 3552 mAudioMixer->setParameter( 3553 name, 3554 AudioMixer::TRACK, 3555 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3556 3557 // reset retry count 3558 track->mRetryCount = kMaxTrackRetries; 3559 3560 // If one track is ready, set the mixer ready if: 3561 // - the mixer was not ready during previous round OR 3562 // - no other track is not ready 3563 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3564 mixerStatus != MIXER_TRACKS_ENABLED) { 3565 mixerStatus = MIXER_TRACKS_READY; 3566 } 3567 } else { 3568 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3569 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3570 } 3571 // clear effect chain input buffer if an active track underruns to avoid sending 3572 // previous audio buffer again to effects 3573 chain = getEffectChain_l(track->sessionId()); 3574 if (chain != 0) { 3575 chain->clearInputBuffer(); 3576 } 3577 3578 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3579 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3580 track->isStopped() || track->isPaused()) { 3581 // We have consumed all the buffers of this track. 3582 // Remove it from the list of active tracks. 3583 // TODO: use actual buffer filling status instead of latency when available from 3584 // audio HAL 3585 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3586 size_t framesWritten = mBytesWritten / mFrameSize; 3587 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3588 if (track->isStopped()) { 3589 track->reset(); 3590 } 3591 tracksToRemove->add(track); 3592 } 3593 } else { 3594 // No buffers for this track. Give it a few chances to 3595 // fill a buffer, then remove it from active list. 3596 if (--(track->mRetryCount) <= 0) { 3597 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3598 tracksToRemove->add(track); 3599 // indicate to client process that the track was disabled because of underrun; 3600 // it will then automatically call start() when data is available 3601 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3602 // If one track is not ready, mark the mixer also not ready if: 3603 // - the mixer was ready during previous round OR 3604 // - no other track is ready 3605 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3606 mixerStatus != MIXER_TRACKS_READY) { 3607 mixerStatus = MIXER_TRACKS_ENABLED; 3608 } 3609 } 3610 mAudioMixer->disable(name); 3611 } 3612 3613 } // local variable scope to avoid goto warning 3614track_is_ready: ; 3615 3616 } 3617 3618 // Push the new FastMixer state if necessary 3619 bool pauseAudioWatchdog = false; 3620 if (didModify) { 3621 state->mFastTracksGen++; 3622 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3623 if (kUseFastMixer == FastMixer_Dynamic && 3624 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3625 state->mCommand = FastMixerState::COLD_IDLE; 3626 state->mColdFutexAddr = &mFastMixerFutex; 3627 state->mColdGen++; 3628 mFastMixerFutex = 0; 3629 if (kUseFastMixer == FastMixer_Dynamic) { 3630 mNormalSink = mOutputSink; 3631 } 3632 // If we go into cold idle, need to wait for acknowledgement 3633 // so that fast mixer stops doing I/O. 3634 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3635 pauseAudioWatchdog = true; 3636 } 3637 } 3638 if (sq != NULL) { 3639 sq->end(didModify); 3640 sq->push(block); 3641 } 3642#ifdef AUDIO_WATCHDOG 3643 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3644 mAudioWatchdog->pause(); 3645 } 3646#endif 3647 3648 // Now perform the deferred reset on fast tracks that have stopped 3649 while (resetMask != 0) { 3650 size_t i = __builtin_ctz(resetMask); 3651 ALOG_ASSERT(i < count); 3652 resetMask &= ~(1 << i); 3653 sp<Track> t = mActiveTracks[i].promote(); 3654 if (t == 0) { 3655 continue; 3656 } 3657 Track* track = t.get(); 3658 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3659 track->reset(); 3660 } 3661 3662 // remove all the tracks that need to be... 3663 removeTracks_l(*tracksToRemove); 3664 3665 // sink or mix buffer must be cleared if all tracks are connected to an 3666 // effect chain as in this case the mixer will not write to the sink or mix buffer 3667 // and track effects will accumulate into it 3668 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3669 (mixedTracks == 0 && fastTracks > 0))) { 3670 // FIXME as a performance optimization, should remember previous zero status 3671 if (mMixerBufferValid) { 3672 memset(mMixerBuffer, 0, mMixerBufferSize); 3673 // TODO: In testing, mSinkBuffer below need not be cleared because 3674 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3675 // after mixing. 3676 // 3677 // To enforce this guarantee: 3678 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3679 // (mixedTracks == 0 && fastTracks > 0)) 3680 // must imply MIXER_TRACKS_READY. 3681 // Later, we may clear buffers regardless, and skip much of this logic. 3682 } 3683 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3684 if (mEffectBufferValid) { 3685 memset(mEffectBuffer, 0, mEffectBufferSize); 3686 } 3687 // FIXME as a performance optimization, should remember previous zero status 3688 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3689 } 3690 3691 // if any fast tracks, then status is ready 3692 mMixerStatusIgnoringFastTracks = mixerStatus; 3693 if (fastTracks > 0) { 3694 mixerStatus = MIXER_TRACKS_READY; 3695 } 3696 return mixerStatus; 3697} 3698 3699// getTrackName_l() must be called with ThreadBase::mLock held 3700int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3701 audio_format_t format, int sessionId) 3702{ 3703 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3704} 3705 3706// deleteTrackName_l() must be called with ThreadBase::mLock held 3707void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3708{ 3709 ALOGV("remove track (%d) and delete from mixer", name); 3710 mAudioMixer->deleteTrackName(name); 3711} 3712 3713// checkForNewParameter_l() must be called with ThreadBase::mLock held 3714bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3715 status_t& status) 3716{ 3717 bool reconfig = false; 3718 3719 status = NO_ERROR; 3720 3721 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3722 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3723 if (mFastMixer != 0) { 3724 FastMixerStateQueue *sq = mFastMixer->sq(); 3725 FastMixerState *state = sq->begin(); 3726 if (!(state->mCommand & FastMixerState::IDLE)) { 3727 previousCommand = state->mCommand; 3728 state->mCommand = FastMixerState::HOT_IDLE; 3729 sq->end(); 3730 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3731 } else { 3732 sq->end(false /*didModify*/); 3733 } 3734 } 3735 3736 AudioParameter param = AudioParameter(keyValuePair); 3737 int value; 3738 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3739 reconfig = true; 3740 } 3741 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3742 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3743 status = BAD_VALUE; 3744 } else { 3745 // no need to save value, since it's constant 3746 reconfig = true; 3747 } 3748 } 3749 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3750 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3751 status = BAD_VALUE; 3752 } else { 3753 // no need to save value, since it's constant 3754 reconfig = true; 3755 } 3756 } 3757 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3758 // do not accept frame count changes if tracks are open as the track buffer 3759 // size depends on frame count and correct behavior would not be guaranteed 3760 // if frame count is changed after track creation 3761 if (!mTracks.isEmpty()) { 3762 status = INVALID_OPERATION; 3763 } else { 3764 reconfig = true; 3765 } 3766 } 3767 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3768#ifdef ADD_BATTERY_DATA 3769 // when changing the audio output device, call addBatteryData to notify 3770 // the change 3771 if (mOutDevice != value) { 3772 uint32_t params = 0; 3773 // check whether speaker is on 3774 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3775 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3776 } 3777 3778 audio_devices_t deviceWithoutSpeaker 3779 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3780 // check if any other device (except speaker) is on 3781 if (value & deviceWithoutSpeaker ) { 3782 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3783 } 3784 3785 if (params != 0) { 3786 addBatteryData(params); 3787 } 3788 } 3789#endif 3790 3791 // forward device change to effects that have requested to be 3792 // aware of attached audio device. 3793 if (value != AUDIO_DEVICE_NONE) { 3794 mOutDevice = value; 3795 for (size_t i = 0; i < mEffectChains.size(); i++) { 3796 mEffectChains[i]->setDevice_l(mOutDevice); 3797 } 3798 } 3799 } 3800 3801 if (status == NO_ERROR) { 3802 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3803 keyValuePair.string()); 3804 if (!mStandby && status == INVALID_OPERATION) { 3805 mOutput->stream->common.standby(&mOutput->stream->common); 3806 mStandby = true; 3807 mBytesWritten = 0; 3808 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3809 keyValuePair.string()); 3810 } 3811 if (status == NO_ERROR && reconfig) { 3812 readOutputParameters_l(); 3813 delete mAudioMixer; 3814 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3815 for (size_t i = 0; i < mTracks.size() ; i++) { 3816 int name = getTrackName_l(mTracks[i]->mChannelMask, 3817 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3818 if (name < 0) { 3819 break; 3820 } 3821 mTracks[i]->mName = name; 3822 } 3823 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3824 } 3825 } 3826 3827 if (!(previousCommand & FastMixerState::IDLE)) { 3828 ALOG_ASSERT(mFastMixer != 0); 3829 FastMixerStateQueue *sq = mFastMixer->sq(); 3830 FastMixerState *state = sq->begin(); 3831 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3832 state->mCommand = previousCommand; 3833 sq->end(); 3834 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3835 } 3836 3837 return reconfig; 3838} 3839 3840 3841void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3842{ 3843 const size_t SIZE = 256; 3844 char buffer[SIZE]; 3845 String8 result; 3846 3847 PlaybackThread::dumpInternals(fd, args); 3848 3849 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3850 3851 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3852 const FastMixerDumpState copy(mFastMixerDumpState); 3853 copy.dump(fd); 3854 3855#ifdef STATE_QUEUE_DUMP 3856 // Similar for state queue 3857 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3858 observerCopy.dump(fd); 3859 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3860 mutatorCopy.dump(fd); 3861#endif 3862 3863#ifdef TEE_SINK 3864 // Write the tee output to a .wav file 3865 dumpTee(fd, mTeeSource, mId); 3866#endif 3867 3868#ifdef AUDIO_WATCHDOG 3869 if (mAudioWatchdog != 0) { 3870 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3871 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3872 wdCopy.dump(fd); 3873 } 3874#endif 3875} 3876 3877uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3878{ 3879 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3880} 3881 3882uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3883{ 3884 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3885} 3886 3887void AudioFlinger::MixerThread::cacheParameters_l() 3888{ 3889 PlaybackThread::cacheParameters_l(); 3890 3891 // FIXME: Relaxed timing because of a certain device that can't meet latency 3892 // Should be reduced to 2x after the vendor fixes the driver issue 3893 // increase threshold again due to low power audio mode. The way this warning 3894 // threshold is calculated and its usefulness should be reconsidered anyway. 3895 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3896} 3897 3898// ---------------------------------------------------------------------------- 3899 3900AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3901 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3902 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3903 // mLeftVolFloat, mRightVolFloat 3904{ 3905} 3906 3907AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3908 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3909 ThreadBase::type_t type) 3910 : PlaybackThread(audioFlinger, output, id, device, type) 3911 // mLeftVolFloat, mRightVolFloat 3912{ 3913} 3914 3915AudioFlinger::DirectOutputThread::~DirectOutputThread() 3916{ 3917} 3918 3919void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3920{ 3921 audio_track_cblk_t* cblk = track->cblk(); 3922 float left, right; 3923 3924 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3925 left = right = 0; 3926 } else { 3927 float typeVolume = mStreamTypes[track->streamType()].volume; 3928 float v = mMasterVolume * typeVolume; 3929 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3930 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3931 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3932 if (left > GAIN_FLOAT_UNITY) { 3933 left = GAIN_FLOAT_UNITY; 3934 } 3935 left *= v; 3936 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3937 if (right > GAIN_FLOAT_UNITY) { 3938 right = GAIN_FLOAT_UNITY; 3939 } 3940 right *= v; 3941 } 3942 3943 if (lastTrack) { 3944 if (left != mLeftVolFloat || right != mRightVolFloat) { 3945 mLeftVolFloat = left; 3946 mRightVolFloat = right; 3947 3948 // Convert volumes from float to 8.24 3949 uint32_t vl = (uint32_t)(left * (1 << 24)); 3950 uint32_t vr = (uint32_t)(right * (1 << 24)); 3951 3952 // Delegate volume control to effect in track effect chain if needed 3953 // only one effect chain can be present on DirectOutputThread, so if 3954 // there is one, the track is connected to it 3955 if (!mEffectChains.isEmpty()) { 3956 mEffectChains[0]->setVolume_l(&vl, &vr); 3957 left = (float)vl / (1 << 24); 3958 right = (float)vr / (1 << 24); 3959 } 3960 if (mOutput->stream->set_volume) { 3961 mOutput->stream->set_volume(mOutput->stream, left, right); 3962 } 3963 } 3964 } 3965} 3966 3967 3968AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3969 Vector< sp<Track> > *tracksToRemove 3970) 3971{ 3972 size_t count = mActiveTracks.size(); 3973 mixer_state mixerStatus = MIXER_IDLE; 3974 3975 // find out which tracks need to be processed 3976 for (size_t i = 0; i < count; i++) { 3977 sp<Track> t = mActiveTracks[i].promote(); 3978 // The track died recently 3979 if (t == 0) { 3980 continue; 3981 } 3982 3983 Track* const track = t.get(); 3984 audio_track_cblk_t* cblk = track->cblk(); 3985 // Only consider last track started for volume and mixer state control. 3986 // In theory an older track could underrun and restart after the new one starts 3987 // but as we only care about the transition phase between two tracks on a 3988 // direct output, it is not a problem to ignore the underrun case. 3989 sp<Track> l = mLatestActiveTrack.promote(); 3990 bool last = l.get() == track; 3991 3992 // The first time a track is added we wait 3993 // for all its buffers to be filled before processing it 3994 uint32_t minFrames; 3995 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 3996 minFrames = mNormalFrameCount; 3997 } else { 3998 minFrames = 1; 3999 } 4000 4001 ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ", 4002 minFrames, track->mState, track->framesReady()); 4003 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4004 !track->isStopping_2() && !track->isStopped()) 4005 { 4006 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4007 4008 if (track->mFillingUpStatus == Track::FS_FILLED) { 4009 track->mFillingUpStatus = Track::FS_ACTIVE; 4010 // make sure processVolume_l() will apply new volume even if 0 4011 mLeftVolFloat = mRightVolFloat = -1.0; 4012 if (track->mState == TrackBase::RESUMING) { 4013 track->mState = TrackBase::ACTIVE; 4014 } 4015 } 4016 4017 // compute volume for this track 4018 processVolume_l(track, last); 4019 if (last) { 4020 // reset retry count 4021 track->mRetryCount = kMaxTrackRetriesDirect; 4022 mActiveTrack = t; 4023 mixerStatus = MIXER_TRACKS_READY; 4024 } 4025 } else { 4026 // clear effect chain input buffer if the last active track started underruns 4027 // to avoid sending previous audio buffer again to effects 4028 if (!mEffectChains.isEmpty() && last) { 4029 mEffectChains[0]->clearInputBuffer(); 4030 } 4031 if (track->isStopping_1()) { 4032 track->mState = TrackBase::STOPPING_2; 4033 } 4034 if ((track->sharedBuffer() != 0) || track->isStopped() || 4035 track->isStopping_2() || track->isPaused()) { 4036 // We have consumed all the buffers of this track. 4037 // Remove it from the list of active tracks. 4038 size_t audioHALFrames; 4039 if (audio_is_linear_pcm(mFormat)) { 4040 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4041 } else { 4042 audioHALFrames = 0; 4043 } 4044 4045 size_t framesWritten = mBytesWritten / mFrameSize; 4046 if (mStandby || !last || 4047 track->presentationComplete(framesWritten, audioHALFrames)) { 4048 if (track->isStopping_2()) { 4049 track->mState = TrackBase::STOPPED; 4050 } 4051 if (track->isStopped()) { 4052 track->reset(); 4053 } 4054 tracksToRemove->add(track); 4055 } 4056 } else { 4057 // No buffers for this track. Give it a few chances to 4058 // fill a buffer, then remove it from active list. 4059 // Only consider last track started for mixer state control 4060 if (--(track->mRetryCount) <= 0) { 4061 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4062 tracksToRemove->add(track); 4063 // indicate to client process that the track was disabled because of underrun; 4064 // it will then automatically call start() when data is available 4065 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4066 } else if (last) { 4067 mixerStatus = MIXER_TRACKS_ENABLED; 4068 } 4069 } 4070 } 4071 } 4072 4073 // remove all the tracks that need to be... 4074 removeTracks_l(*tracksToRemove); 4075 4076 return mixerStatus; 4077} 4078 4079void AudioFlinger::DirectOutputThread::threadLoop_mix() 4080{ 4081 size_t frameCount = mFrameCount; 4082 int8_t *curBuf = (int8_t *)mSinkBuffer; 4083 // output audio to hardware 4084 while (frameCount) { 4085 AudioBufferProvider::Buffer buffer; 4086 buffer.frameCount = frameCount; 4087 mActiveTrack->getNextBuffer(&buffer); 4088 if (buffer.raw == NULL) { 4089 memset(curBuf, 0, frameCount * mFrameSize); 4090 break; 4091 } 4092 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4093 frameCount -= buffer.frameCount; 4094 curBuf += buffer.frameCount * mFrameSize; 4095 mActiveTrack->releaseBuffer(&buffer); 4096 } 4097 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4098 sleepTime = 0; 4099 standbyTime = systemTime() + standbyDelay; 4100 mActiveTrack.clear(); 4101} 4102 4103void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4104{ 4105 if (sleepTime == 0) { 4106 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4107 sleepTime = activeSleepTime; 4108 } else { 4109 sleepTime = idleSleepTime; 4110 } 4111 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4112 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4113 sleepTime = 0; 4114 } 4115} 4116 4117// getTrackName_l() must be called with ThreadBase::mLock held 4118int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4119 audio_format_t format __unused, int sessionId __unused) 4120{ 4121 return 0; 4122} 4123 4124// deleteTrackName_l() must be called with ThreadBase::mLock held 4125void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4126{ 4127} 4128 4129// checkForNewParameter_l() must be called with ThreadBase::mLock held 4130bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4131 status_t& status) 4132{ 4133 bool reconfig = false; 4134 4135 status = NO_ERROR; 4136 4137 AudioParameter param = AudioParameter(keyValuePair); 4138 int value; 4139 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4140 // forward device change to effects that have requested to be 4141 // aware of attached audio device. 4142 if (value != AUDIO_DEVICE_NONE) { 4143 mOutDevice = value; 4144 for (size_t i = 0; i < mEffectChains.size(); i++) { 4145 mEffectChains[i]->setDevice_l(mOutDevice); 4146 } 4147 } 4148 } 4149 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4150 // do not accept frame count changes if tracks are open as the track buffer 4151 // size depends on frame count and correct behavior would not be garantied 4152 // if frame count is changed after track creation 4153 if (!mTracks.isEmpty()) { 4154 status = INVALID_OPERATION; 4155 } else { 4156 reconfig = true; 4157 } 4158 } 4159 if (status == NO_ERROR) { 4160 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4161 keyValuePair.string()); 4162 if (!mStandby && status == INVALID_OPERATION) { 4163 mOutput->stream->common.standby(&mOutput->stream->common); 4164 mStandby = true; 4165 mBytesWritten = 0; 4166 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4167 keyValuePair.string()); 4168 } 4169 if (status == NO_ERROR && reconfig) { 4170 readOutputParameters_l(); 4171 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4172 } 4173 } 4174 4175 return reconfig; 4176} 4177 4178uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4179{ 4180 uint32_t time; 4181 if (audio_is_linear_pcm(mFormat)) { 4182 time = PlaybackThread::activeSleepTimeUs(); 4183 } else { 4184 time = 10000; 4185 } 4186 return time; 4187} 4188 4189uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4190{ 4191 uint32_t time; 4192 if (audio_is_linear_pcm(mFormat)) { 4193 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4194 } else { 4195 time = 10000; 4196 } 4197 return time; 4198} 4199 4200uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4201{ 4202 uint32_t time; 4203 if (audio_is_linear_pcm(mFormat)) { 4204 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4205 } else { 4206 time = 10000; 4207 } 4208 return time; 4209} 4210 4211void AudioFlinger::DirectOutputThread::cacheParameters_l() 4212{ 4213 PlaybackThread::cacheParameters_l(); 4214 4215 // use shorter standby delay as on normal output to release 4216 // hardware resources as soon as possible 4217 if (audio_is_linear_pcm(mFormat)) { 4218 standbyDelay = microseconds(activeSleepTime*2); 4219 } else { 4220 standbyDelay = kOffloadStandbyDelayNs; 4221 } 4222} 4223 4224// ---------------------------------------------------------------------------- 4225 4226AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4227 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4228 : Thread(false /*canCallJava*/), 4229 mPlaybackThread(playbackThread), 4230 mWriteAckSequence(0), 4231 mDrainSequence(0) 4232{ 4233} 4234 4235AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4236{ 4237} 4238 4239void AudioFlinger::AsyncCallbackThread::onFirstRef() 4240{ 4241 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4242} 4243 4244bool AudioFlinger::AsyncCallbackThread::threadLoop() 4245{ 4246 while (!exitPending()) { 4247 uint32_t writeAckSequence; 4248 uint32_t drainSequence; 4249 4250 { 4251 Mutex::Autolock _l(mLock); 4252 while (!((mWriteAckSequence & 1) || 4253 (mDrainSequence & 1) || 4254 exitPending())) { 4255 mWaitWorkCV.wait(mLock); 4256 } 4257 4258 if (exitPending()) { 4259 break; 4260 } 4261 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4262 mWriteAckSequence, mDrainSequence); 4263 writeAckSequence = mWriteAckSequence; 4264 mWriteAckSequence &= ~1; 4265 drainSequence = mDrainSequence; 4266 mDrainSequence &= ~1; 4267 } 4268 { 4269 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4270 if (playbackThread != 0) { 4271 if (writeAckSequence & 1) { 4272 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4273 } 4274 if (drainSequence & 1) { 4275 playbackThread->resetDraining(drainSequence >> 1); 4276 } 4277 } 4278 } 4279 } 4280 return false; 4281} 4282 4283void AudioFlinger::AsyncCallbackThread::exit() 4284{ 4285 ALOGV("AsyncCallbackThread::exit"); 4286 Mutex::Autolock _l(mLock); 4287 requestExit(); 4288 mWaitWorkCV.broadcast(); 4289} 4290 4291void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4292{ 4293 Mutex::Autolock _l(mLock); 4294 // bit 0 is cleared 4295 mWriteAckSequence = sequence << 1; 4296} 4297 4298void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4299{ 4300 Mutex::Autolock _l(mLock); 4301 // ignore unexpected callbacks 4302 if (mWriteAckSequence & 2) { 4303 mWriteAckSequence |= 1; 4304 mWaitWorkCV.signal(); 4305 } 4306} 4307 4308void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4309{ 4310 Mutex::Autolock _l(mLock); 4311 // bit 0 is cleared 4312 mDrainSequence = sequence << 1; 4313} 4314 4315void AudioFlinger::AsyncCallbackThread::resetDraining() 4316{ 4317 Mutex::Autolock _l(mLock); 4318 // ignore unexpected callbacks 4319 if (mDrainSequence & 2) { 4320 mDrainSequence |= 1; 4321 mWaitWorkCV.signal(); 4322 } 4323} 4324 4325 4326// ---------------------------------------------------------------------------- 4327AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4328 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4329 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4330 mHwPaused(false), 4331 mFlushPending(false), 4332 mPausedBytesRemaining(0) 4333{ 4334 //FIXME: mStandby should be set to true by ThreadBase constructor 4335 mStandby = true; 4336} 4337 4338void AudioFlinger::OffloadThread::threadLoop_exit() 4339{ 4340 if (mFlushPending || mHwPaused) { 4341 // If a flush is pending or track was paused, just discard buffered data 4342 flushHw_l(); 4343 } else { 4344 mMixerStatus = MIXER_DRAIN_ALL; 4345 threadLoop_drain(); 4346 } 4347 if (mUseAsyncWrite) { 4348 ALOG_ASSERT(mCallbackThread != 0); 4349 mCallbackThread->exit(); 4350 } 4351 PlaybackThread::threadLoop_exit(); 4352} 4353 4354AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4355 Vector< sp<Track> > *tracksToRemove 4356) 4357{ 4358 size_t count = mActiveTracks.size(); 4359 4360 mixer_state mixerStatus = MIXER_IDLE; 4361 bool doHwPause = false; 4362 bool doHwResume = false; 4363 4364 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4365 4366 // find out which tracks need to be processed 4367 for (size_t i = 0; i < count; i++) { 4368 sp<Track> t = mActiveTracks[i].promote(); 4369 // The track died recently 4370 if (t == 0) { 4371 continue; 4372 } 4373 Track* const track = t.get(); 4374 audio_track_cblk_t* cblk = track->cblk(); 4375 // Only consider last track started for volume and mixer state control. 4376 // In theory an older track could underrun and restart after the new one starts 4377 // but as we only care about the transition phase between two tracks on a 4378 // direct output, it is not a problem to ignore the underrun case. 4379 sp<Track> l = mLatestActiveTrack.promote(); 4380 bool last = l.get() == track; 4381 4382 if (track->isInvalid()) { 4383 ALOGW("An invalidated track shouldn't be in active list"); 4384 tracksToRemove->add(track); 4385 continue; 4386 } 4387 4388 if (track->mState == TrackBase::IDLE) { 4389 ALOGW("An idle track shouldn't be in active list"); 4390 continue; 4391 } 4392 4393 if (track->isPausing()) { 4394 track->setPaused(); 4395 if (last) { 4396 if (!mHwPaused) { 4397 doHwPause = true; 4398 mHwPaused = true; 4399 } 4400 // If we were part way through writing the mixbuffer to 4401 // the HAL we must save this until we resume 4402 // BUG - this will be wrong if a different track is made active, 4403 // in that case we want to discard the pending data in the 4404 // mixbuffer and tell the client to present it again when the 4405 // track is resumed 4406 mPausedWriteLength = mCurrentWriteLength; 4407 mPausedBytesRemaining = mBytesRemaining; 4408 mBytesRemaining = 0; // stop writing 4409 } 4410 tracksToRemove->add(track); 4411 } else if (track->isFlushPending()) { 4412 track->flushAck(); 4413 if (last) { 4414 mFlushPending = true; 4415 } 4416 } else if (track->isResumePending()){ 4417 track->resumeAck(); 4418 if (last) { 4419 if (mPausedBytesRemaining) { 4420 // Need to continue write that was interrupted 4421 mCurrentWriteLength = mPausedWriteLength; 4422 mBytesRemaining = mPausedBytesRemaining; 4423 mPausedBytesRemaining = 0; 4424 } 4425 if (mHwPaused) { 4426 doHwResume = true; 4427 mHwPaused = false; 4428 // threadLoop_mix() will handle the case that we need to 4429 // resume an interrupted write 4430 } 4431 // enable write to audio HAL 4432 sleepTime = 0; 4433 4434 // Do not handle new data in this iteration even if track->framesReady() 4435 mixerStatus = MIXER_TRACKS_ENABLED; 4436 } 4437 } else if (track->framesReady() && track->isReady() && 4438 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4439 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4440 if (track->mFillingUpStatus == Track::FS_FILLED) { 4441 track->mFillingUpStatus = Track::FS_ACTIVE; 4442 // make sure processVolume_l() will apply new volume even if 0 4443 mLeftVolFloat = mRightVolFloat = -1.0; 4444 } 4445 4446 if (last) { 4447 sp<Track> previousTrack = mPreviousTrack.promote(); 4448 if (previousTrack != 0) { 4449 if (track != previousTrack.get()) { 4450 // Flush any data still being written from last track 4451 mBytesRemaining = 0; 4452 if (mPausedBytesRemaining) { 4453 // Last track was paused so we also need to flush saved 4454 // mixbuffer state and invalidate track so that it will 4455 // re-submit that unwritten data when it is next resumed 4456 mPausedBytesRemaining = 0; 4457 // Invalidate is a bit drastic - would be more efficient 4458 // to have a flag to tell client that some of the 4459 // previously written data was lost 4460 previousTrack->invalidate(); 4461 } 4462 // flush data already sent to the DSP if changing audio session as audio 4463 // comes from a different source. Also invalidate previous track to force a 4464 // seek when resuming. 4465 if (previousTrack->sessionId() != track->sessionId()) { 4466 previousTrack->invalidate(); 4467 } 4468 } 4469 } 4470 mPreviousTrack = track; 4471 // reset retry count 4472 track->mRetryCount = kMaxTrackRetriesOffload; 4473 mActiveTrack = t; 4474 mixerStatus = MIXER_TRACKS_READY; 4475 } 4476 } else { 4477 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4478 if (track->isStopping_1()) { 4479 // Hardware buffer can hold a large amount of audio so we must 4480 // wait for all current track's data to drain before we say 4481 // that the track is stopped. 4482 if (mBytesRemaining == 0) { 4483 // Only start draining when all data in mixbuffer 4484 // has been written 4485 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4486 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4487 // do not drain if no data was ever sent to HAL (mStandby == true) 4488 if (last && !mStandby) { 4489 // do not modify drain sequence if we are already draining. This happens 4490 // when resuming from pause after drain. 4491 if ((mDrainSequence & 1) == 0) { 4492 sleepTime = 0; 4493 standbyTime = systemTime() + standbyDelay; 4494 mixerStatus = MIXER_DRAIN_TRACK; 4495 mDrainSequence += 2; 4496 } 4497 if (mHwPaused) { 4498 // It is possible to move from PAUSED to STOPPING_1 without 4499 // a resume so we must ensure hardware is running 4500 doHwResume = true; 4501 mHwPaused = false; 4502 } 4503 } 4504 } 4505 } else if (track->isStopping_2()) { 4506 // Drain has completed or we are in standby, signal presentation complete 4507 if (!(mDrainSequence & 1) || !last || mStandby) { 4508 track->mState = TrackBase::STOPPED; 4509 size_t audioHALFrames = 4510 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4511 size_t framesWritten = 4512 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4513 track->presentationComplete(framesWritten, audioHALFrames); 4514 track->reset(); 4515 tracksToRemove->add(track); 4516 } 4517 } else { 4518 // No buffers for this track. Give it a few chances to 4519 // fill a buffer, then remove it from active list. 4520 if (--(track->mRetryCount) <= 0) { 4521 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4522 track->name()); 4523 tracksToRemove->add(track); 4524 // indicate to client process that the track was disabled because of underrun; 4525 // it will then automatically call start() when data is available 4526 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4527 } else if (last){ 4528 mixerStatus = MIXER_TRACKS_ENABLED; 4529 } 4530 } 4531 } 4532 // compute volume for this track 4533 processVolume_l(track, last); 4534 } 4535 4536 // make sure the pause/flush/resume sequence is executed in the right order. 4537 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4538 // before flush and then resume HW. This can happen in case of pause/flush/resume 4539 // if resume is received before pause is executed. 4540 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4541 mOutput->stream->pause(mOutput->stream); 4542 } 4543 if (mFlushPending) { 4544 flushHw_l(); 4545 mFlushPending = false; 4546 } 4547 if (!mStandby && doHwResume) { 4548 mOutput->stream->resume(mOutput->stream); 4549 } 4550 4551 // remove all the tracks that need to be... 4552 removeTracks_l(*tracksToRemove); 4553 4554 return mixerStatus; 4555} 4556 4557// must be called with thread mutex locked 4558bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4559{ 4560 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4561 mWriteAckSequence, mDrainSequence); 4562 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4563 return true; 4564 } 4565 return false; 4566} 4567 4568// must be called with thread mutex locked 4569bool AudioFlinger::OffloadThread::shouldStandby_l() 4570{ 4571 bool trackPaused = false; 4572 4573 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4574 // after a timeout and we will enter standby then. 4575 if (mTracks.size() > 0) { 4576 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4577 } 4578 4579 return !mStandby && !trackPaused; 4580} 4581 4582 4583bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4584{ 4585 Mutex::Autolock _l(mLock); 4586 return waitingAsyncCallback_l(); 4587} 4588 4589void AudioFlinger::OffloadThread::flushHw_l() 4590{ 4591 mOutput->stream->flush(mOutput->stream); 4592 // Flush anything still waiting in the mixbuffer 4593 mCurrentWriteLength = 0; 4594 mBytesRemaining = 0; 4595 mPausedWriteLength = 0; 4596 mPausedBytesRemaining = 0; 4597 mHwPaused = false; 4598 4599 if (mUseAsyncWrite) { 4600 // discard any pending drain or write ack by incrementing sequence 4601 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4602 mDrainSequence = (mDrainSequence + 2) & ~1; 4603 ALOG_ASSERT(mCallbackThread != 0); 4604 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4605 mCallbackThread->setDraining(mDrainSequence); 4606 } 4607} 4608 4609void AudioFlinger::OffloadThread::onAddNewTrack_l() 4610{ 4611 sp<Track> previousTrack = mPreviousTrack.promote(); 4612 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4613 4614 if (previousTrack != 0 && latestTrack != 0 && 4615 (previousTrack->sessionId() != latestTrack->sessionId())) { 4616 mFlushPending = true; 4617 } 4618 PlaybackThread::onAddNewTrack_l(); 4619} 4620 4621// ---------------------------------------------------------------------------- 4622 4623AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4624 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4625 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4626 DUPLICATING), 4627 mWaitTimeMs(UINT_MAX) 4628{ 4629 addOutputTrack(mainThread); 4630} 4631 4632AudioFlinger::DuplicatingThread::~DuplicatingThread() 4633{ 4634 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4635 mOutputTracks[i]->destroy(); 4636 } 4637} 4638 4639void AudioFlinger::DuplicatingThread::threadLoop_mix() 4640{ 4641 // mix buffers... 4642 if (outputsReady(outputTracks)) { 4643 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4644 } else { 4645 memset(mSinkBuffer, 0, mSinkBufferSize); 4646 } 4647 sleepTime = 0; 4648 writeFrames = mNormalFrameCount; 4649 mCurrentWriteLength = mSinkBufferSize; 4650 standbyTime = systemTime() + standbyDelay; 4651} 4652 4653void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4654{ 4655 if (sleepTime == 0) { 4656 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4657 sleepTime = activeSleepTime; 4658 } else { 4659 sleepTime = idleSleepTime; 4660 } 4661 } else if (mBytesWritten != 0) { 4662 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4663 writeFrames = mNormalFrameCount; 4664 memset(mSinkBuffer, 0, mSinkBufferSize); 4665 } else { 4666 // flush remaining overflow buffers in output tracks 4667 writeFrames = 0; 4668 } 4669 sleepTime = 0; 4670 } 4671} 4672 4673ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4674{ 4675 for (size_t i = 0; i < outputTracks.size(); i++) { 4676 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4677 // for delivery downstream as needed. This in-place conversion is safe as 4678 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4679 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4680 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4681 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4682 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4683 } 4684 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4685 } 4686 mStandby = false; 4687 return (ssize_t)mSinkBufferSize; 4688} 4689 4690void AudioFlinger::DuplicatingThread::threadLoop_standby() 4691{ 4692 // DuplicatingThread implements standby by stopping all tracks 4693 for (size_t i = 0; i < outputTracks.size(); i++) { 4694 outputTracks[i]->stop(); 4695 } 4696} 4697 4698void AudioFlinger::DuplicatingThread::saveOutputTracks() 4699{ 4700 outputTracks = mOutputTracks; 4701} 4702 4703void AudioFlinger::DuplicatingThread::clearOutputTracks() 4704{ 4705 outputTracks.clear(); 4706} 4707 4708void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4709{ 4710 Mutex::Autolock _l(mLock); 4711 // FIXME explain this formula 4712 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4713 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4714 // due to current usage case and restrictions on the AudioBufferProvider. 4715 // Actual buffer conversion is done in threadLoop_write(). 4716 // 4717 // TODO: This may change in the future, depending on multichannel 4718 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4719 OutputTrack *outputTrack = new OutputTrack(thread, 4720 this, 4721 mSampleRate, 4722 AUDIO_FORMAT_PCM_16_BIT, 4723 mChannelMask, 4724 frameCount, 4725 IPCThreadState::self()->getCallingUid()); 4726 if (outputTrack->cblk() != NULL) { 4727 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4728 mOutputTracks.add(outputTrack); 4729 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4730 updateWaitTime_l(); 4731 } 4732} 4733 4734void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4735{ 4736 Mutex::Autolock _l(mLock); 4737 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4738 if (mOutputTracks[i]->thread() == thread) { 4739 mOutputTracks[i]->destroy(); 4740 mOutputTracks.removeAt(i); 4741 updateWaitTime_l(); 4742 return; 4743 } 4744 } 4745 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4746} 4747 4748// caller must hold mLock 4749void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4750{ 4751 mWaitTimeMs = UINT_MAX; 4752 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4753 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4754 if (strong != 0) { 4755 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4756 if (waitTimeMs < mWaitTimeMs) { 4757 mWaitTimeMs = waitTimeMs; 4758 } 4759 } 4760 } 4761} 4762 4763 4764bool AudioFlinger::DuplicatingThread::outputsReady( 4765 const SortedVector< sp<OutputTrack> > &outputTracks) 4766{ 4767 for (size_t i = 0; i < outputTracks.size(); i++) { 4768 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4769 if (thread == 0) { 4770 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4771 outputTracks[i].get()); 4772 return false; 4773 } 4774 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4775 // see note at standby() declaration 4776 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4777 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4778 thread.get()); 4779 return false; 4780 } 4781 } 4782 return true; 4783} 4784 4785uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4786{ 4787 return (mWaitTimeMs * 1000) / 2; 4788} 4789 4790void AudioFlinger::DuplicatingThread::cacheParameters_l() 4791{ 4792 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4793 updateWaitTime_l(); 4794 4795 MixerThread::cacheParameters_l(); 4796} 4797 4798// ---------------------------------------------------------------------------- 4799// Record 4800// ---------------------------------------------------------------------------- 4801 4802AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4803 AudioStreamIn *input, 4804 audio_io_handle_t id, 4805 audio_devices_t outDevice, 4806 audio_devices_t inDevice 4807#ifdef TEE_SINK 4808 , const sp<NBAIO_Sink>& teeSink 4809#endif 4810 ) : 4811 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4812 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4813 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4814 mRsmpInRear(0) 4815#ifdef TEE_SINK 4816 , mTeeSink(teeSink) 4817#endif 4818 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4819 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4820 // mFastCapture below 4821 , mFastCaptureFutex(0) 4822 // mInputSource 4823 // mPipeSink 4824 // mPipeSource 4825 , mPipeFramesP2(0) 4826 // mPipeMemory 4827 // mFastCaptureNBLogWriter 4828 , mFastTrackAvail(false) 4829{ 4830 snprintf(mName, kNameLength, "AudioIn_%X", id); 4831 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4832 4833 readInputParameters_l(); 4834 4835 // create an NBAIO source for the HAL input stream, and negotiate 4836 mInputSource = new AudioStreamInSource(input->stream); 4837 size_t numCounterOffers = 0; 4838 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4839 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4840 ALOG_ASSERT(index == 0); 4841 4842 // initialize fast capture depending on configuration 4843 bool initFastCapture; 4844 switch (kUseFastCapture) { 4845 case FastCapture_Never: 4846 initFastCapture = false; 4847 break; 4848 case FastCapture_Always: 4849 initFastCapture = true; 4850 break; 4851 case FastCapture_Static: 4852 uint32_t primaryOutputSampleRate; 4853 { 4854 AutoMutex _l(audioFlinger->mHardwareLock); 4855 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4856 } 4857 initFastCapture = 4858 // either capture sample rate is same as (a reasonable) primary output sample rate 4859 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4860 (mSampleRate == primaryOutputSampleRate)) || 4861 // or primary output sample rate is unknown, and capture sample rate is reasonable 4862 ((primaryOutputSampleRate == 0) && 4863 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4864 // and the buffer size is < 12 ms 4865 (mFrameCount * 1000) / mSampleRate < 12; 4866 break; 4867 // case FastCapture_Dynamic: 4868 } 4869 4870 if (initFastCapture) { 4871 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4872 NBAIO_Format format = mInputSource->format(); 4873 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4874 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4875 void *pipeBuffer; 4876 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4877 sp<IMemory> pipeMemory; 4878 if ((roHeap == 0) || 4879 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4880 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4881 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4882 goto failed; 4883 } 4884 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4885 memset(pipeBuffer, 0, pipeSize); 4886 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4887 const NBAIO_Format offers[1] = {format}; 4888 size_t numCounterOffers = 0; 4889 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4890 ALOG_ASSERT(index == 0); 4891 mPipeSink = pipe; 4892 PipeReader *pipeReader = new PipeReader(*pipe); 4893 numCounterOffers = 0; 4894 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4895 ALOG_ASSERT(index == 0); 4896 mPipeSource = pipeReader; 4897 mPipeFramesP2 = pipeFramesP2; 4898 mPipeMemory = pipeMemory; 4899 4900 // create fast capture 4901 mFastCapture = new FastCapture(); 4902 FastCaptureStateQueue *sq = mFastCapture->sq(); 4903#ifdef STATE_QUEUE_DUMP 4904 // FIXME 4905#endif 4906 FastCaptureState *state = sq->begin(); 4907 state->mCblk = NULL; 4908 state->mInputSource = mInputSource.get(); 4909 state->mInputSourceGen++; 4910 state->mPipeSink = pipe; 4911 state->mPipeSinkGen++; 4912 state->mFrameCount = mFrameCount; 4913 state->mCommand = FastCaptureState::COLD_IDLE; 4914 // already done in constructor initialization list 4915 //mFastCaptureFutex = 0; 4916 state->mColdFutexAddr = &mFastCaptureFutex; 4917 state->mColdGen++; 4918 state->mDumpState = &mFastCaptureDumpState; 4919#ifdef TEE_SINK 4920 // FIXME 4921#endif 4922 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4923 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4924 sq->end(); 4925 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4926 4927 // start the fast capture 4928 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4929 pid_t tid = mFastCapture->getTid(); 4930 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4931 if (err != 0) { 4932 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4933 kPriorityFastCapture, getpid_cached, tid, err); 4934 } 4935 4936#ifdef AUDIO_WATCHDOG 4937 // FIXME 4938#endif 4939 4940 mFastTrackAvail = true; 4941 } 4942failed: ; 4943 4944 // FIXME mNormalSource 4945} 4946 4947 4948AudioFlinger::RecordThread::~RecordThread() 4949{ 4950 if (mFastCapture != 0) { 4951 FastCaptureStateQueue *sq = mFastCapture->sq(); 4952 FastCaptureState *state = sq->begin(); 4953 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4954 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4955 if (old == -1) { 4956 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4957 } 4958 } 4959 state->mCommand = FastCaptureState::EXIT; 4960 sq->end(); 4961 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4962 mFastCapture->join(); 4963 mFastCapture.clear(); 4964 } 4965 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4966 mAudioFlinger->unregisterWriter(mNBLogWriter); 4967 delete[] mRsmpInBuffer; 4968} 4969 4970void AudioFlinger::RecordThread::onFirstRef() 4971{ 4972 run(mName, PRIORITY_URGENT_AUDIO); 4973} 4974 4975bool AudioFlinger::RecordThread::threadLoop() 4976{ 4977 nsecs_t lastWarning = 0; 4978 4979 inputStandBy(); 4980 4981reacquire_wakelock: 4982 sp<RecordTrack> activeTrack; 4983 int activeTracksGen; 4984 { 4985 Mutex::Autolock _l(mLock); 4986 size_t size = mActiveTracks.size(); 4987 activeTracksGen = mActiveTracksGen; 4988 if (size > 0) { 4989 // FIXME an arbitrary choice 4990 activeTrack = mActiveTracks[0]; 4991 acquireWakeLock_l(activeTrack->uid()); 4992 if (size > 1) { 4993 SortedVector<int> tmp; 4994 for (size_t i = 0; i < size; i++) { 4995 tmp.add(mActiveTracks[i]->uid()); 4996 } 4997 updateWakeLockUids_l(tmp); 4998 } 4999 } else { 5000 acquireWakeLock_l(-1); 5001 } 5002 } 5003 5004 // used to request a deferred sleep, to be executed later while mutex is unlocked 5005 uint32_t sleepUs = 0; 5006 5007 // loop while there is work to do 5008 for (;;) { 5009 Vector< sp<EffectChain> > effectChains; 5010 5011 // sleep with mutex unlocked 5012 if (sleepUs > 0) { 5013 usleep(sleepUs); 5014 sleepUs = 0; 5015 } 5016 5017 // activeTracks accumulates a copy of a subset of mActiveTracks 5018 Vector< sp<RecordTrack> > activeTracks; 5019 5020 // reference to the (first and only) fast track 5021 sp<RecordTrack> fastTrack; 5022 5023 { // scope for mLock 5024 Mutex::Autolock _l(mLock); 5025 5026 processConfigEvents_l(); 5027 5028 // check exitPending here because checkForNewParameters_l() and 5029 // checkForNewParameters_l() can temporarily release mLock 5030 if (exitPending()) { 5031 break; 5032 } 5033 5034 // if no active track(s), then standby and release wakelock 5035 size_t size = mActiveTracks.size(); 5036 if (size == 0) { 5037 standbyIfNotAlreadyInStandby(); 5038 // exitPending() can't become true here 5039 releaseWakeLock_l(); 5040 ALOGV("RecordThread: loop stopping"); 5041 // go to sleep 5042 mWaitWorkCV.wait(mLock); 5043 ALOGV("RecordThread: loop starting"); 5044 goto reacquire_wakelock; 5045 } 5046 5047 if (mActiveTracksGen != activeTracksGen) { 5048 activeTracksGen = mActiveTracksGen; 5049 SortedVector<int> tmp; 5050 for (size_t i = 0; i < size; i++) { 5051 tmp.add(mActiveTracks[i]->uid()); 5052 } 5053 updateWakeLockUids_l(tmp); 5054 } 5055 5056 bool doBroadcast = false; 5057 for (size_t i = 0; i < size; ) { 5058 5059 activeTrack = mActiveTracks[i]; 5060 if (activeTrack->isTerminated()) { 5061 removeTrack_l(activeTrack); 5062 mActiveTracks.remove(activeTrack); 5063 mActiveTracksGen++; 5064 size--; 5065 continue; 5066 } 5067 5068 TrackBase::track_state activeTrackState = activeTrack->mState; 5069 switch (activeTrackState) { 5070 5071 case TrackBase::PAUSING: 5072 mActiveTracks.remove(activeTrack); 5073 mActiveTracksGen++; 5074 doBroadcast = true; 5075 size--; 5076 continue; 5077 5078 case TrackBase::STARTING_1: 5079 sleepUs = 10000; 5080 i++; 5081 continue; 5082 5083 case TrackBase::STARTING_2: 5084 doBroadcast = true; 5085 mStandby = false; 5086 activeTrack->mState = TrackBase::ACTIVE; 5087 break; 5088 5089 case TrackBase::ACTIVE: 5090 break; 5091 5092 case TrackBase::IDLE: 5093 i++; 5094 continue; 5095 5096 default: 5097 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5098 } 5099 5100 activeTracks.add(activeTrack); 5101 i++; 5102 5103 if (activeTrack->isFastTrack()) { 5104 ALOG_ASSERT(!mFastTrackAvail); 5105 ALOG_ASSERT(fastTrack == 0); 5106 fastTrack = activeTrack; 5107 } 5108 } 5109 if (doBroadcast) { 5110 mStartStopCond.broadcast(); 5111 } 5112 5113 // sleep if there are no active tracks to process 5114 if (activeTracks.size() == 0) { 5115 if (sleepUs == 0) { 5116 sleepUs = kRecordThreadSleepUs; 5117 } 5118 continue; 5119 } 5120 sleepUs = 0; 5121 5122 lockEffectChains_l(effectChains); 5123 } 5124 5125 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5126 5127 size_t size = effectChains.size(); 5128 for (size_t i = 0; i < size; i++) { 5129 // thread mutex is not locked, but effect chain is locked 5130 effectChains[i]->process_l(); 5131 } 5132 5133 // Start the fast capture if it's not already running 5134 if (mFastCapture != 0) { 5135 FastCaptureStateQueue *sq = mFastCapture->sq(); 5136 FastCaptureState *state = sq->begin(); 5137 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5138 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5139 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5140 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5141 if (old == -1) { 5142 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5143 } 5144 } 5145 state->mCommand = FastCaptureState::READ_WRITE; 5146#if 0 // FIXME 5147 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5148 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5149#endif 5150 state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL; 5151 sq->end(); 5152 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5153#if 0 5154 if (kUseFastCapture == FastCapture_Dynamic) { 5155 mNormalSource = mPipeSource; 5156 } 5157#endif 5158 } else { 5159 sq->end(false /*didModify*/); 5160 } 5161 } 5162 5163 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5164 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5165 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5166 // If destination is non-contiguous, first read past the nominal end of buffer, then 5167 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5168 5169 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5170 ssize_t framesRead; 5171 5172 // If an NBAIO source is present, use it to read the normal capture's data 5173 if (mPipeSource != 0) { 5174 size_t framesToRead = mBufferSize / mFrameSize; 5175 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5176 framesToRead, AudioBufferProvider::kInvalidPTS); 5177 if (framesRead == 0) { 5178 // since pipe is non-blocking, simulate blocking input 5179 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5180 } 5181 // otherwise use the HAL / AudioStreamIn directly 5182 } else { 5183 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5184 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5185 if (bytesRead < 0) { 5186 framesRead = bytesRead; 5187 } else { 5188 framesRead = bytesRead / mFrameSize; 5189 } 5190 } 5191 5192 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5193 ALOGE("read failed: framesRead=%d", framesRead); 5194 // Force input into standby so that it tries to recover at next read attempt 5195 inputStandBy(); 5196 sleepUs = kRecordThreadSleepUs; 5197 } 5198 if (framesRead <= 0) { 5199 goto unlock; 5200 } 5201 ALOG_ASSERT(framesRead > 0); 5202 5203 if (mTeeSink != 0) { 5204 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5205 } 5206 // If destination is non-contiguous, we now correct for reading past end of buffer. 5207 { 5208 size_t part1 = mRsmpInFramesP2 - rear; 5209 if ((size_t) framesRead > part1) { 5210 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5211 (framesRead - part1) * mFrameSize); 5212 } 5213 } 5214 rear = mRsmpInRear += framesRead; 5215 5216 size = activeTracks.size(); 5217 // loop over each active track 5218 for (size_t i = 0; i < size; i++) { 5219 activeTrack = activeTracks[i]; 5220 5221 // skip fast tracks, as those are handled directly by FastCapture 5222 if (activeTrack->isFastTrack()) { 5223 continue; 5224 } 5225 5226 enum { 5227 OVERRUN_UNKNOWN, 5228 OVERRUN_TRUE, 5229 OVERRUN_FALSE 5230 } overrun = OVERRUN_UNKNOWN; 5231 5232 // loop over getNextBuffer to handle circular sink 5233 for (;;) { 5234 5235 activeTrack->mSink.frameCount = ~0; 5236 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5237 size_t framesOut = activeTrack->mSink.frameCount; 5238 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5239 5240 int32_t front = activeTrack->mRsmpInFront; 5241 ssize_t filled = rear - front; 5242 size_t framesIn; 5243 5244 if (filled < 0) { 5245 // should not happen, but treat like a massive overrun and re-sync 5246 framesIn = 0; 5247 activeTrack->mRsmpInFront = rear; 5248 overrun = OVERRUN_TRUE; 5249 } else if ((size_t) filled <= mRsmpInFrames) { 5250 framesIn = (size_t) filled; 5251 } else { 5252 // client is not keeping up with server, but give it latest data 5253 framesIn = mRsmpInFrames; 5254 activeTrack->mRsmpInFront = front = rear - framesIn; 5255 overrun = OVERRUN_TRUE; 5256 } 5257 5258 if (framesOut == 0 || framesIn == 0) { 5259 break; 5260 } 5261 5262 if (activeTrack->mResampler == NULL) { 5263 // no resampling 5264 if (framesIn > framesOut) { 5265 framesIn = framesOut; 5266 } else { 5267 framesOut = framesIn; 5268 } 5269 int8_t *dst = activeTrack->mSink.i8; 5270 while (framesIn > 0) { 5271 front &= mRsmpInFramesP2 - 1; 5272 size_t part1 = mRsmpInFramesP2 - front; 5273 if (part1 > framesIn) { 5274 part1 = framesIn; 5275 } 5276 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5277 if (mChannelCount == activeTrack->mChannelCount) { 5278 memcpy(dst, src, part1 * mFrameSize); 5279 } else if (mChannelCount == 1) { 5280 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5281 part1); 5282 } else { 5283 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5284 part1); 5285 } 5286 dst += part1 * activeTrack->mFrameSize; 5287 front += part1; 5288 framesIn -= part1; 5289 } 5290 activeTrack->mRsmpInFront += framesOut; 5291 5292 } else { 5293 // resampling 5294 // FIXME framesInNeeded should really be part of resampler API, and should 5295 // depend on the SRC ratio 5296 // to keep mRsmpInBuffer full so resampler always has sufficient input 5297 size_t framesInNeeded; 5298 // FIXME only re-calculate when it changes, and optimize for common ratios 5299 // Do not precompute in/out because floating point is not associative 5300 // e.g. a*b/c != a*(b/c). 5301 const double in(mSampleRate); 5302 const double out(activeTrack->mSampleRate); 5303 framesInNeeded = ceil(framesOut * in / out) + 1; 5304 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5305 framesInNeeded, framesOut, in / out); 5306 // Although we theoretically have framesIn in circular buffer, some of those are 5307 // unreleased frames, and thus must be discounted for purpose of budgeting. 5308 size_t unreleased = activeTrack->mRsmpInUnrel; 5309 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5310 if (framesIn < framesInNeeded) { 5311 ALOGV("not enough to resample: have %u frames in but need %u in to " 5312 "produce %u out given in/out ratio of %.4g", 5313 framesIn, framesInNeeded, framesOut, in / out); 5314 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5315 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5316 if (newFramesOut == 0) { 5317 break; 5318 } 5319 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5320 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5321 framesInNeeded, newFramesOut, out / in); 5322 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5323 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5324 "given in/out ratio of %.4g", 5325 framesIn, framesInNeeded, newFramesOut, in / out); 5326 framesOut = newFramesOut; 5327 } else { 5328 ALOGV("success 1: have %u in and need %u in to produce %u out " 5329 "given in/out ratio of %.4g", 5330 framesIn, framesInNeeded, framesOut, in / out); 5331 } 5332 5333 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5334 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5335 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5336 delete[] activeTrack->mRsmpOutBuffer; 5337 // resampler always outputs stereo 5338 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5339 activeTrack->mRsmpOutFrameCount = framesOut; 5340 } 5341 5342 // resampler accumulates, but we only have one source track 5343 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5344 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5345 // FIXME how about having activeTrack implement this interface itself? 5346 activeTrack->mResamplerBufferProvider 5347 /*this*/ /* AudioBufferProvider* */); 5348 // ditherAndClamp() works as long as all buffers returned by 5349 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5350 if (activeTrack->mChannelCount == 1) { 5351 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5352 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5353 framesOut); 5354 // the resampler always outputs stereo samples: 5355 // do post stereo to mono conversion 5356 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5357 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5358 } else { 5359 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5360 activeTrack->mRsmpOutBuffer, framesOut); 5361 } 5362 // now done with mRsmpOutBuffer 5363 5364 } 5365 5366 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5367 overrun = OVERRUN_FALSE; 5368 } 5369 5370 if (activeTrack->mFramesToDrop == 0) { 5371 if (framesOut > 0) { 5372 activeTrack->mSink.frameCount = framesOut; 5373 activeTrack->releaseBuffer(&activeTrack->mSink); 5374 } 5375 } else { 5376 // FIXME could do a partial drop of framesOut 5377 if (activeTrack->mFramesToDrop > 0) { 5378 activeTrack->mFramesToDrop -= framesOut; 5379 if (activeTrack->mFramesToDrop <= 0) { 5380 activeTrack->clearSyncStartEvent(); 5381 } 5382 } else { 5383 activeTrack->mFramesToDrop += framesOut; 5384 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5385 activeTrack->mSyncStartEvent->isCancelled()) { 5386 ALOGW("Synced record %s, session %d, trigger session %d", 5387 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5388 activeTrack->sessionId(), 5389 (activeTrack->mSyncStartEvent != 0) ? 5390 activeTrack->mSyncStartEvent->triggerSession() : 0); 5391 activeTrack->clearSyncStartEvent(); 5392 } 5393 } 5394 } 5395 5396 if (framesOut == 0) { 5397 break; 5398 } 5399 } 5400 5401 switch (overrun) { 5402 case OVERRUN_TRUE: 5403 // client isn't retrieving buffers fast enough 5404 if (!activeTrack->setOverflow()) { 5405 nsecs_t now = systemTime(); 5406 // FIXME should lastWarning per track? 5407 if ((now - lastWarning) > kWarningThrottleNs) { 5408 ALOGW("RecordThread: buffer overflow"); 5409 lastWarning = now; 5410 } 5411 } 5412 break; 5413 case OVERRUN_FALSE: 5414 activeTrack->clearOverflow(); 5415 break; 5416 case OVERRUN_UNKNOWN: 5417 break; 5418 } 5419 5420 } 5421 5422unlock: 5423 // enable changes in effect chain 5424 unlockEffectChains(effectChains); 5425 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5426 } 5427 5428 standbyIfNotAlreadyInStandby(); 5429 5430 { 5431 Mutex::Autolock _l(mLock); 5432 for (size_t i = 0; i < mTracks.size(); i++) { 5433 sp<RecordTrack> track = mTracks[i]; 5434 track->invalidate(); 5435 } 5436 mActiveTracks.clear(); 5437 mActiveTracksGen++; 5438 mStartStopCond.broadcast(); 5439 } 5440 5441 releaseWakeLock(); 5442 5443 ALOGV("RecordThread %p exiting", this); 5444 return false; 5445} 5446 5447void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5448{ 5449 if (!mStandby) { 5450 inputStandBy(); 5451 mStandby = true; 5452 } 5453} 5454 5455void AudioFlinger::RecordThread::inputStandBy() 5456{ 5457 // Idle the fast capture if it's currently running 5458 if (mFastCapture != 0) { 5459 FastCaptureStateQueue *sq = mFastCapture->sq(); 5460 FastCaptureState *state = sq->begin(); 5461 if (!(state->mCommand & FastCaptureState::IDLE)) { 5462 state->mCommand = FastCaptureState::COLD_IDLE; 5463 state->mColdFutexAddr = &mFastCaptureFutex; 5464 state->mColdGen++; 5465 mFastCaptureFutex = 0; 5466 sq->end(); 5467 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5468 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5469#if 0 5470 if (kUseFastCapture == FastCapture_Dynamic) { 5471 // FIXME 5472 } 5473#endif 5474#ifdef AUDIO_WATCHDOG 5475 // FIXME 5476#endif 5477 } else { 5478 sq->end(false /*didModify*/); 5479 } 5480 } 5481 mInput->stream->common.standby(&mInput->stream->common); 5482} 5483 5484// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5485sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5486 const sp<AudioFlinger::Client>& client, 5487 uint32_t sampleRate, 5488 audio_format_t format, 5489 audio_channel_mask_t channelMask, 5490 size_t *pFrameCount, 5491 int sessionId, 5492 size_t *notificationFrames, 5493 int uid, 5494 IAudioFlinger::track_flags_t *flags, 5495 pid_t tid, 5496 status_t *status) 5497{ 5498 size_t frameCount = *pFrameCount; 5499 sp<RecordTrack> track; 5500 status_t lStatus; 5501 5502 // client expresses a preference for FAST, but we get the final say 5503 if (*flags & IAudioFlinger::TRACK_FAST) { 5504 if ( 5505 // use case: callback handler 5506 (tid != -1) && 5507 // frame count is not specified, or is exactly the pipe depth 5508 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5509 // PCM data 5510 audio_is_linear_pcm(format) && 5511 // native format 5512 (format == mFormat) && 5513 // native channel mask 5514 (channelMask == mChannelMask) && 5515 // native hardware sample rate 5516 (sampleRate == mSampleRate) && 5517 // record thread has an associated fast capture 5518 hasFastCapture() && 5519 // there are sufficient fast track slots available 5520 mFastTrackAvail 5521 ) { 5522 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5523 frameCount, mFrameCount); 5524 } else { 5525 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5526 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5527 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5528 frameCount, mFrameCount, mPipeFramesP2, 5529 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5530 hasFastCapture(), tid, mFastTrackAvail); 5531 *flags &= ~IAudioFlinger::TRACK_FAST; 5532 } 5533 } 5534 5535 // compute track buffer size in frames, and suggest the notification frame count 5536 if (*flags & IAudioFlinger::TRACK_FAST) { 5537 // fast track: frame count is exactly the pipe depth 5538 frameCount = mPipeFramesP2; 5539 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5540 *notificationFrames = mFrameCount; 5541 } else { 5542 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5543 // or 20 ms if there is a fast capture 5544 // TODO This could be a roundupRatio inline, and const 5545 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5546 * sampleRate + mSampleRate - 1) / mSampleRate; 5547 // minimum number of notification periods is at least kMinNotifications, 5548 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5549 static const size_t kMinNotifications = 3; 5550 static const uint32_t kMinMs = 30; 5551 // TODO This could be a roundupRatio inline 5552 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5553 // TODO This could be a roundupRatio inline 5554 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5555 maxNotificationFrames; 5556 const size_t minFrameCount = maxNotificationFrames * 5557 max(kMinNotifications, minNotificationsByMs); 5558 frameCount = max(frameCount, minFrameCount); 5559 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5560 *notificationFrames = maxNotificationFrames; 5561 } 5562 } 5563 *pFrameCount = frameCount; 5564 5565 lStatus = initCheck(); 5566 if (lStatus != NO_ERROR) { 5567 ALOGE("createRecordTrack_l() audio driver not initialized"); 5568 goto Exit; 5569 } 5570 5571 { // scope for mLock 5572 Mutex::Autolock _l(mLock); 5573 5574 track = new RecordTrack(this, client, sampleRate, 5575 format, channelMask, frameCount, NULL, sessionId, uid, 5576 *flags, TrackBase::TYPE_DEFAULT); 5577 5578 lStatus = track->initCheck(); 5579 if (lStatus != NO_ERROR) { 5580 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5581 // track must be cleared from the caller as the caller has the AF lock 5582 goto Exit; 5583 } 5584 mTracks.add(track); 5585 5586 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5587 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5588 mAudioFlinger->btNrecIsOff(); 5589 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5590 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5591 5592 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5593 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5594 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5595 // so ask activity manager to do this on our behalf 5596 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5597 } 5598 } 5599 5600 lStatus = NO_ERROR; 5601 5602Exit: 5603 *status = lStatus; 5604 return track; 5605} 5606 5607status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5608 AudioSystem::sync_event_t event, 5609 int triggerSession) 5610{ 5611 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5612 sp<ThreadBase> strongMe = this; 5613 status_t status = NO_ERROR; 5614 5615 if (event == AudioSystem::SYNC_EVENT_NONE) { 5616 recordTrack->clearSyncStartEvent(); 5617 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5618 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5619 triggerSession, 5620 recordTrack->sessionId(), 5621 syncStartEventCallback, 5622 recordTrack); 5623 // Sync event can be cancelled by the trigger session if the track is not in a 5624 // compatible state in which case we start record immediately 5625 if (recordTrack->mSyncStartEvent->isCancelled()) { 5626 recordTrack->clearSyncStartEvent(); 5627 } else { 5628 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5629 recordTrack->mFramesToDrop = - 5630 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5631 } 5632 } 5633 5634 { 5635 // This section is a rendezvous between binder thread executing start() and RecordThread 5636 AutoMutex lock(mLock); 5637 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5638 if (recordTrack->mState == TrackBase::PAUSING) { 5639 ALOGV("active record track PAUSING -> ACTIVE"); 5640 recordTrack->mState = TrackBase::ACTIVE; 5641 } else { 5642 ALOGV("active record track state %d", recordTrack->mState); 5643 } 5644 return status; 5645 } 5646 5647 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5648 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5649 // or using a separate command thread 5650 recordTrack->mState = TrackBase::STARTING_1; 5651 mActiveTracks.add(recordTrack); 5652 mActiveTracksGen++; 5653 status_t status = NO_ERROR; 5654 if (recordTrack->isExternalTrack()) { 5655 mLock.unlock(); 5656 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5657 mLock.lock(); 5658 // FIXME should verify that recordTrack is still in mActiveTracks 5659 if (status != NO_ERROR) { 5660 mActiveTracks.remove(recordTrack); 5661 mActiveTracksGen++; 5662 recordTrack->clearSyncStartEvent(); 5663 ALOGV("RecordThread::start error %d", status); 5664 return status; 5665 } 5666 } 5667 // Catch up with current buffer indices if thread is already running. 5668 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5669 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5670 // see previously buffered data before it called start(), but with greater risk of overrun. 5671 5672 recordTrack->mRsmpInFront = mRsmpInRear; 5673 recordTrack->mRsmpInUnrel = 0; 5674 // FIXME why reset? 5675 if (recordTrack->mResampler != NULL) { 5676 recordTrack->mResampler->reset(); 5677 } 5678 recordTrack->mState = TrackBase::STARTING_2; 5679 // signal thread to start 5680 mWaitWorkCV.broadcast(); 5681 if (mActiveTracks.indexOf(recordTrack) < 0) { 5682 ALOGV("Record failed to start"); 5683 status = BAD_VALUE; 5684 goto startError; 5685 } 5686 return status; 5687 } 5688 5689startError: 5690 if (recordTrack->isExternalTrack()) { 5691 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5692 } 5693 recordTrack->clearSyncStartEvent(); 5694 // FIXME I wonder why we do not reset the state here? 5695 return status; 5696} 5697 5698void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5699{ 5700 sp<SyncEvent> strongEvent = event.promote(); 5701 5702 if (strongEvent != 0) { 5703 sp<RefBase> ptr = strongEvent->cookie().promote(); 5704 if (ptr != 0) { 5705 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5706 recordTrack->handleSyncStartEvent(strongEvent); 5707 } 5708 } 5709} 5710 5711bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5712 ALOGV("RecordThread::stop"); 5713 AutoMutex _l(mLock); 5714 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5715 return false; 5716 } 5717 // note that threadLoop may still be processing the track at this point [without lock] 5718 recordTrack->mState = TrackBase::PAUSING; 5719 // do not wait for mStartStopCond if exiting 5720 if (exitPending()) { 5721 return true; 5722 } 5723 // FIXME incorrect usage of wait: no explicit predicate or loop 5724 mStartStopCond.wait(mLock); 5725 // if we have been restarted, recordTrack is in mActiveTracks here 5726 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5727 ALOGV("Record stopped OK"); 5728 return true; 5729 } 5730 return false; 5731} 5732 5733bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5734{ 5735 return false; 5736} 5737 5738status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5739{ 5740#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5741 if (!isValidSyncEvent(event)) { 5742 return BAD_VALUE; 5743 } 5744 5745 int eventSession = event->triggerSession(); 5746 status_t ret = NAME_NOT_FOUND; 5747 5748 Mutex::Autolock _l(mLock); 5749 5750 for (size_t i = 0; i < mTracks.size(); i++) { 5751 sp<RecordTrack> track = mTracks[i]; 5752 if (eventSession == track->sessionId()) { 5753 (void) track->setSyncEvent(event); 5754 ret = NO_ERROR; 5755 } 5756 } 5757 return ret; 5758#else 5759 return BAD_VALUE; 5760#endif 5761} 5762 5763// destroyTrack_l() must be called with ThreadBase::mLock held 5764void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5765{ 5766 track->terminate(); 5767 track->mState = TrackBase::STOPPED; 5768 // active tracks are removed by threadLoop() 5769 if (mActiveTracks.indexOf(track) < 0) { 5770 removeTrack_l(track); 5771 } 5772} 5773 5774void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5775{ 5776 mTracks.remove(track); 5777 // need anything related to effects here? 5778 if (track->isFastTrack()) { 5779 ALOG_ASSERT(!mFastTrackAvail); 5780 mFastTrackAvail = true; 5781 } 5782} 5783 5784void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5785{ 5786 dumpInternals(fd, args); 5787 dumpTracks(fd, args); 5788 dumpEffectChains(fd, args); 5789} 5790 5791void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5792{ 5793 dprintf(fd, "\nInput thread %p:\n", this); 5794 5795 if (mActiveTracks.size() > 0) { 5796 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5797 } else { 5798 dprintf(fd, " No active record clients\n"); 5799 } 5800 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5801 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5802 5803 dumpBase(fd, args); 5804} 5805 5806void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5807{ 5808 const size_t SIZE = 256; 5809 char buffer[SIZE]; 5810 String8 result; 5811 5812 size_t numtracks = mTracks.size(); 5813 size_t numactive = mActiveTracks.size(); 5814 size_t numactiveseen = 0; 5815 dprintf(fd, " %d Tracks", numtracks); 5816 if (numtracks) { 5817 dprintf(fd, " of which %d are active\n", numactive); 5818 RecordTrack::appendDumpHeader(result); 5819 for (size_t i = 0; i < numtracks ; ++i) { 5820 sp<RecordTrack> track = mTracks[i]; 5821 if (track != 0) { 5822 bool active = mActiveTracks.indexOf(track) >= 0; 5823 if (active) { 5824 numactiveseen++; 5825 } 5826 track->dump(buffer, SIZE, active); 5827 result.append(buffer); 5828 } 5829 } 5830 } else { 5831 dprintf(fd, "\n"); 5832 } 5833 5834 if (numactiveseen != numactive) { 5835 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5836 " not in the track list\n"); 5837 result.append(buffer); 5838 RecordTrack::appendDumpHeader(result); 5839 for (size_t i = 0; i < numactive; ++i) { 5840 sp<RecordTrack> track = mActiveTracks[i]; 5841 if (mTracks.indexOf(track) < 0) { 5842 track->dump(buffer, SIZE, true); 5843 result.append(buffer); 5844 } 5845 } 5846 5847 } 5848 write(fd, result.string(), result.size()); 5849} 5850 5851// AudioBufferProvider interface 5852status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5853 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5854{ 5855 RecordTrack *activeTrack = mRecordTrack; 5856 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5857 if (threadBase == 0) { 5858 buffer->frameCount = 0; 5859 buffer->raw = NULL; 5860 return NOT_ENOUGH_DATA; 5861 } 5862 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5863 int32_t rear = recordThread->mRsmpInRear; 5864 int32_t front = activeTrack->mRsmpInFront; 5865 ssize_t filled = rear - front; 5866 // FIXME should not be P2 (don't want to increase latency) 5867 // FIXME if client not keeping up, discard 5868 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5869 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5870 front &= recordThread->mRsmpInFramesP2 - 1; 5871 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5872 if (part1 > (size_t) filled) { 5873 part1 = filled; 5874 } 5875 size_t ask = buffer->frameCount; 5876 ALOG_ASSERT(ask > 0); 5877 if (part1 > ask) { 5878 part1 = ask; 5879 } 5880 if (part1 == 0) { 5881 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5882 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5883 buffer->raw = NULL; 5884 buffer->frameCount = 0; 5885 activeTrack->mRsmpInUnrel = 0; 5886 return NOT_ENOUGH_DATA; 5887 } 5888 5889 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5890 buffer->frameCount = part1; 5891 activeTrack->mRsmpInUnrel = part1; 5892 return NO_ERROR; 5893} 5894 5895// AudioBufferProvider interface 5896void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5897 AudioBufferProvider::Buffer* buffer) 5898{ 5899 RecordTrack *activeTrack = mRecordTrack; 5900 size_t stepCount = buffer->frameCount; 5901 if (stepCount == 0) { 5902 return; 5903 } 5904 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5905 activeTrack->mRsmpInUnrel -= stepCount; 5906 activeTrack->mRsmpInFront += stepCount; 5907 buffer->raw = NULL; 5908 buffer->frameCount = 0; 5909} 5910 5911bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5912 status_t& status) 5913{ 5914 bool reconfig = false; 5915 5916 status = NO_ERROR; 5917 5918 audio_format_t reqFormat = mFormat; 5919 uint32_t samplingRate = mSampleRate; 5920 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5921 5922 AudioParameter param = AudioParameter(keyValuePair); 5923 int value; 5924 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5925 // channel count change can be requested. Do we mandate the first client defines the 5926 // HAL sampling rate and channel count or do we allow changes on the fly? 5927 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5928 samplingRate = value; 5929 reconfig = true; 5930 } 5931 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5932 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5933 status = BAD_VALUE; 5934 } else { 5935 reqFormat = (audio_format_t) value; 5936 reconfig = true; 5937 } 5938 } 5939 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5940 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5941 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5942 status = BAD_VALUE; 5943 } else { 5944 channelMask = mask; 5945 reconfig = true; 5946 } 5947 } 5948 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5949 // do not accept frame count changes if tracks are open as the track buffer 5950 // size depends on frame count and correct behavior would not be guaranteed 5951 // if frame count is changed after track creation 5952 if (mActiveTracks.size() > 0) { 5953 status = INVALID_OPERATION; 5954 } else { 5955 reconfig = true; 5956 } 5957 } 5958 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5959 // forward device change to effects that have requested to be 5960 // aware of attached audio device. 5961 for (size_t i = 0; i < mEffectChains.size(); i++) { 5962 mEffectChains[i]->setDevice_l(value); 5963 } 5964 5965 // store input device and output device but do not forward output device to audio HAL. 5966 // Note that status is ignored by the caller for output device 5967 // (see AudioFlinger::setParameters() 5968 if (audio_is_output_devices(value)) { 5969 mOutDevice = value; 5970 status = BAD_VALUE; 5971 } else { 5972 mInDevice = value; 5973 // disable AEC and NS if the device is a BT SCO headset supporting those 5974 // pre processings 5975 if (mTracks.size() > 0) { 5976 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5977 mAudioFlinger->btNrecIsOff(); 5978 for (size_t i = 0; i < mTracks.size(); i++) { 5979 sp<RecordTrack> track = mTracks[i]; 5980 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5981 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5982 } 5983 } 5984 } 5985 } 5986 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5987 mAudioSource != (audio_source_t)value) { 5988 // forward device change to effects that have requested to be 5989 // aware of attached audio device. 5990 for (size_t i = 0; i < mEffectChains.size(); i++) { 5991 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5992 } 5993 mAudioSource = (audio_source_t)value; 5994 } 5995 5996 if (status == NO_ERROR) { 5997 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5998 keyValuePair.string()); 5999 if (status == INVALID_OPERATION) { 6000 inputStandBy(); 6001 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6002 keyValuePair.string()); 6003 } 6004 if (reconfig) { 6005 if (status == BAD_VALUE && 6006 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6007 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6008 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6009 <= (2 * samplingRate)) && 6010 audio_channel_count_from_in_mask( 6011 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6012 (channelMask == AUDIO_CHANNEL_IN_MONO || 6013 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6014 status = NO_ERROR; 6015 } 6016 if (status == NO_ERROR) { 6017 readInputParameters_l(); 6018 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6019 } 6020 } 6021 } 6022 6023 return reconfig; 6024} 6025 6026String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6027{ 6028 Mutex::Autolock _l(mLock); 6029 if (initCheck() != NO_ERROR) { 6030 return String8(); 6031 } 6032 6033 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6034 const String8 out_s8(s); 6035 free(s); 6036 return out_s8; 6037} 6038 6039void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6040 AudioSystem::OutputDescriptor desc; 6041 const void *param2 = NULL; 6042 6043 switch (event) { 6044 case AudioSystem::INPUT_OPENED: 6045 case AudioSystem::INPUT_CONFIG_CHANGED: 6046 desc.channelMask = mChannelMask; 6047 desc.samplingRate = mSampleRate; 6048 desc.format = mFormat; 6049 desc.frameCount = mFrameCount; 6050 desc.latency = 0; 6051 param2 = &desc; 6052 break; 6053 6054 case AudioSystem::INPUT_CLOSED: 6055 default: 6056 break; 6057 } 6058 mAudioFlinger->audioConfigChanged(event, mId, param2); 6059} 6060 6061void AudioFlinger::RecordThread::readInputParameters_l() 6062{ 6063 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6064 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6065 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6066 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6067 mFormat = mHALFormat; 6068 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6069 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6070 } 6071 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6072 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6073 mFrameCount = mBufferSize / mFrameSize; 6074 // This is the formula for calculating the temporary buffer size. 6075 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6076 // 1 full output buffer, regardless of the alignment of the available input. 6077 // The value is somewhat arbitrary, and could probably be even larger. 6078 // A larger value should allow more old data to be read after a track calls start(), 6079 // without increasing latency. 6080 mRsmpInFrames = mFrameCount * 7; 6081 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6082 delete[] mRsmpInBuffer; 6083 6084 // TODO optimize audio capture buffer sizes ... 6085 // Here we calculate the size of the sliding buffer used as a source 6086 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6087 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6088 // be better to have it derived from the pipe depth in the long term. 6089 // The current value is higher than necessary. However it should not add to latency. 6090 6091 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6092 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6093 6094 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6095 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6096} 6097 6098uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6099{ 6100 Mutex::Autolock _l(mLock); 6101 if (initCheck() != NO_ERROR) { 6102 return 0; 6103 } 6104 6105 return mInput->stream->get_input_frames_lost(mInput->stream); 6106} 6107 6108uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6109{ 6110 Mutex::Autolock _l(mLock); 6111 uint32_t result = 0; 6112 if (getEffectChain_l(sessionId) != 0) { 6113 result = EFFECT_SESSION; 6114 } 6115 6116 for (size_t i = 0; i < mTracks.size(); ++i) { 6117 if (sessionId == mTracks[i]->sessionId()) { 6118 result |= TRACK_SESSION; 6119 break; 6120 } 6121 } 6122 6123 return result; 6124} 6125 6126KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6127{ 6128 KeyedVector<int, bool> ids; 6129 Mutex::Autolock _l(mLock); 6130 for (size_t j = 0; j < mTracks.size(); ++j) { 6131 sp<RecordThread::RecordTrack> track = mTracks[j]; 6132 int sessionId = track->sessionId(); 6133 if (ids.indexOfKey(sessionId) < 0) { 6134 ids.add(sessionId, true); 6135 } 6136 } 6137 return ids; 6138} 6139 6140AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6141{ 6142 Mutex::Autolock _l(mLock); 6143 AudioStreamIn *input = mInput; 6144 mInput = NULL; 6145 return input; 6146} 6147 6148// this method must always be called either with ThreadBase mLock held or inside the thread loop 6149audio_stream_t* AudioFlinger::RecordThread::stream() const 6150{ 6151 if (mInput == NULL) { 6152 return NULL; 6153 } 6154 return &mInput->stream->common; 6155} 6156 6157status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6158{ 6159 // only one chain per input thread 6160 if (mEffectChains.size() != 0) { 6161 return INVALID_OPERATION; 6162 } 6163 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6164 6165 chain->setInBuffer(NULL); 6166 chain->setOutBuffer(NULL); 6167 6168 checkSuspendOnAddEffectChain_l(chain); 6169 6170 mEffectChains.add(chain); 6171 6172 return NO_ERROR; 6173} 6174 6175size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6176{ 6177 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6178 ALOGW_IF(mEffectChains.size() != 1, 6179 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6180 chain.get(), mEffectChains.size(), this); 6181 if (mEffectChains.size() == 1) { 6182 mEffectChains.removeAt(0); 6183 } 6184 return 0; 6185} 6186 6187status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6188 audio_patch_handle_t *handle) 6189{ 6190 status_t status = NO_ERROR; 6191 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6192 // store new device and send to effects 6193 mInDevice = patch->sources[0].ext.device.type; 6194 for (size_t i = 0; i < mEffectChains.size(); i++) { 6195 mEffectChains[i]->setDevice_l(mInDevice); 6196 } 6197 6198 // disable AEC and NS if the device is a BT SCO headset supporting those 6199 // pre processings 6200 if (mTracks.size() > 0) { 6201 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6202 mAudioFlinger->btNrecIsOff(); 6203 for (size_t i = 0; i < mTracks.size(); i++) { 6204 sp<RecordTrack> track = mTracks[i]; 6205 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6206 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6207 } 6208 } 6209 6210 // store new source and send to effects 6211 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6212 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6213 for (size_t i = 0; i < mEffectChains.size(); i++) { 6214 mEffectChains[i]->setAudioSource_l(mAudioSource); 6215 } 6216 } 6217 6218 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6219 status = hwDevice->create_audio_patch(hwDevice, 6220 patch->num_sources, 6221 patch->sources, 6222 patch->num_sinks, 6223 patch->sinks, 6224 handle); 6225 } else { 6226 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6227 } 6228 return status; 6229} 6230 6231status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6232{ 6233 status_t status = NO_ERROR; 6234 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6235 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6236 status = hwDevice->release_audio_patch(hwDevice, handle); 6237 } else { 6238 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6239 } 6240 return status; 6241} 6242 6243void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6244{ 6245 Mutex::Autolock _l(mLock); 6246 mTracks.add(record); 6247} 6248 6249void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6250{ 6251 Mutex::Autolock _l(mLock); 6252 destroyTrack_l(record); 6253} 6254 6255void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6256{ 6257 ThreadBase::getAudioPortConfig(config); 6258 config->role = AUDIO_PORT_ROLE_SINK; 6259 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6260 config->ext.mix.usecase.source = mAudioSource; 6261} 6262 6263}; // namespace android 6264