Threads.cpp revision 3458bb2356e711419487056fe1dd474e100466a3
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// ---------------------------------------------------------------------------- 316 317// static 318const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 319{ 320 switch (type) { 321 case MIXER: 322 return "MIXER"; 323 case DIRECT: 324 return "DIRECT"; 325 case DUPLICATING: 326 return "DUPLICATING"; 327 case RECORD: 328 return "RECORD"; 329 case OFFLOAD: 330 return "OFFLOAD"; 331 default: 332 return "unknown"; 333 } 334} 335 336static String8 outputFlagsToString(audio_output_flags_t flags) 337{ 338 static const struct mapping { 339 audio_output_flags_t mFlag; 340 const char * mString; 341 } mappings[] = { 342 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 343 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 344 AUDIO_OUTPUT_FLAG_FAST, "FAST", 345 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 346 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAAD", 347 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 348 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 349 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 350 }; 351 String8 result; 352 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 353 const mapping *entry; 354 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 355 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 356 if (flags & entry->mFlag) { 357 if (!result.isEmpty()) { 358 result.append("|"); 359 } 360 result.append(entry->mString); 361 } 362 } 363 if (flags & ~allFlags) { 364 if (!result.isEmpty()) { 365 result.append("|"); 366 } 367 result.appendFormat("0x%X", flags & ~allFlags); 368 } 369 if (result.isEmpty()) { 370 result.append(entry->mString); 371 } 372 return result; 373} 374 375AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 376 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 377 : Thread(false /*canCallJava*/), 378 mType(type), 379 mAudioFlinger(audioFlinger), 380 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 381 // are set by PlaybackThread::readOutputParameters_l() or 382 // RecordThread::readInputParameters_l() 383 //FIXME: mStandby should be true here. Is this some kind of hack? 384 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 385 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 386 // mName will be set by concrete (non-virtual) subclass 387 mDeathRecipient(new PMDeathRecipient(this)) 388{ 389} 390 391AudioFlinger::ThreadBase::~ThreadBase() 392{ 393 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 394 mConfigEvents.clear(); 395 396 // do not lock the mutex in destructor 397 releaseWakeLock_l(); 398 if (mPowerManager != 0) { 399 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 400 binder->unlinkToDeath(mDeathRecipient); 401 } 402} 403 404status_t AudioFlinger::ThreadBase::readyToRun() 405{ 406 status_t status = initCheck(); 407 if (status == NO_ERROR) { 408 ALOGI("AudioFlinger's thread %p ready to run", this); 409 } else { 410 ALOGE("No working audio driver found."); 411 } 412 return status; 413} 414 415void AudioFlinger::ThreadBase::exit() 416{ 417 ALOGV("ThreadBase::exit"); 418 // do any cleanup required for exit to succeed 419 preExit(); 420 { 421 // This lock prevents the following race in thread (uniprocessor for illustration): 422 // if (!exitPending()) { 423 // // context switch from here to exit() 424 // // exit() calls requestExit(), what exitPending() observes 425 // // exit() calls signal(), which is dropped since no waiters 426 // // context switch back from exit() to here 427 // mWaitWorkCV.wait(...); 428 // // now thread is hung 429 // } 430 AutoMutex lock(mLock); 431 requestExit(); 432 mWaitWorkCV.broadcast(); 433 } 434 // When Thread::requestExitAndWait is made virtual and this method is renamed to 435 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 436 requestExitAndWait(); 437} 438 439status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 440{ 441 status_t status; 442 443 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 444 Mutex::Autolock _l(mLock); 445 446 return sendSetParameterConfigEvent_l(keyValuePairs); 447} 448 449// sendConfigEvent_l() must be called with ThreadBase::mLock held 450// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 451status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 452{ 453 status_t status = NO_ERROR; 454 455 mConfigEvents.add(event); 456 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 457 mWaitWorkCV.signal(); 458 mLock.unlock(); 459 { 460 Mutex::Autolock _l(event->mLock); 461 while (event->mWaitStatus) { 462 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 463 event->mStatus = TIMED_OUT; 464 event->mWaitStatus = false; 465 } 466 } 467 status = event->mStatus; 468 } 469 mLock.lock(); 470 return status; 471} 472 473void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 474{ 475 Mutex::Autolock _l(mLock); 476 sendIoConfigEvent_l(event, param); 477} 478 479// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 480void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 481{ 482 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 483 sendConfigEvent_l(configEvent); 484} 485 486// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 487void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 488{ 489 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 490 sendConfigEvent_l(configEvent); 491} 492 493// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 494status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 495{ 496 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 497 return sendConfigEvent_l(configEvent); 498} 499 500status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 501 const struct audio_patch *patch, 502 audio_patch_handle_t *handle) 503{ 504 Mutex::Autolock _l(mLock); 505 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 506 status_t status = sendConfigEvent_l(configEvent); 507 if (status == NO_ERROR) { 508 CreateAudioPatchConfigEventData *data = 509 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 510 *handle = data->mHandle; 511 } 512 return status; 513} 514 515status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 516 const audio_patch_handle_t handle) 517{ 518 Mutex::Autolock _l(mLock); 519 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 520 return sendConfigEvent_l(configEvent); 521} 522 523 524// post condition: mConfigEvents.isEmpty() 525void AudioFlinger::ThreadBase::processConfigEvents_l() 526{ 527 bool configChanged = false; 528 529 while (!mConfigEvents.isEmpty()) { 530 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 531 sp<ConfigEvent> event = mConfigEvents[0]; 532 mConfigEvents.removeAt(0); 533 switch (event->mType) { 534 case CFG_EVENT_PRIO: { 535 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 536 // FIXME Need to understand why this has to be done asynchronously 537 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 538 true /*asynchronous*/); 539 if (err != 0) { 540 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 541 data->mPrio, data->mPid, data->mTid, err); 542 } 543 } break; 544 case CFG_EVENT_IO: { 545 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 546 audioConfigChanged(data->mEvent, data->mParam); 547 } break; 548 case CFG_EVENT_SET_PARAMETER: { 549 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 550 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 551 configChanged = true; 552 } 553 } break; 554 case CFG_EVENT_CREATE_AUDIO_PATCH: { 555 CreateAudioPatchConfigEventData *data = 556 (CreateAudioPatchConfigEventData *)event->mData.get(); 557 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 558 } break; 559 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 560 ReleaseAudioPatchConfigEventData *data = 561 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 562 event->mStatus = releaseAudioPatch_l(data->mHandle); 563 } break; 564 default: 565 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 566 break; 567 } 568 { 569 Mutex::Autolock _l(event->mLock); 570 if (event->mWaitStatus) { 571 event->mWaitStatus = false; 572 event->mCond.signal(); 573 } 574 } 575 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 576 } 577 578 if (configChanged) { 579 cacheParameters_l(); 580 } 581} 582 583String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 584 String8 s; 585 if (output) { 586 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 587 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 588 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 589 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 590 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 591 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 592 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 593 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 594 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 595 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 596 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 597 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 598 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 599 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 600 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 601 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 602 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 603 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 604 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 605 } else { 606 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 607 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 608 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 609 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 610 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 611 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 612 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 613 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 614 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 615 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 616 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 617 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 618 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 619 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 620 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 621 } 622 int len = s.length(); 623 if (s.length() > 2) { 624 char *str = s.lockBuffer(len); 625 s.unlockBuffer(len - 2); 626 } 627 return s; 628} 629 630void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 631{ 632 const size_t SIZE = 256; 633 char buffer[SIZE]; 634 String8 result; 635 636 bool locked = AudioFlinger::dumpTryLock(mLock); 637 if (!locked) { 638 dprintf(fd, "thread %p may be deadlocked\n", this); 639 } 640 641 dprintf(fd, " I/O handle: %d\n", mId); 642 dprintf(fd, " TID: %d\n", getTid()); 643 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 644 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 645 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 646 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 647 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 648 dprintf(fd, " Channel count: %u\n", mChannelCount); 649 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 650 channelMaskToString(mChannelMask, mType != RECORD).string()); 651 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 652 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 653 dprintf(fd, " Pending config events:"); 654 size_t numConfig = mConfigEvents.size(); 655 if (numConfig) { 656 for (size_t i = 0; i < numConfig; i++) { 657 mConfigEvents[i]->dump(buffer, SIZE); 658 dprintf(fd, "\n %s", buffer); 659 } 660 dprintf(fd, "\n"); 661 } else { 662 dprintf(fd, " none\n"); 663 } 664 665 if (locked) { 666 mLock.unlock(); 667 } 668} 669 670void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 671{ 672 const size_t SIZE = 256; 673 char buffer[SIZE]; 674 String8 result; 675 676 size_t numEffectChains = mEffectChains.size(); 677 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 678 write(fd, buffer, strlen(buffer)); 679 680 for (size_t i = 0; i < numEffectChains; ++i) { 681 sp<EffectChain> chain = mEffectChains[i]; 682 if (chain != 0) { 683 chain->dump(fd, args); 684 } 685 } 686} 687 688void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 689{ 690 Mutex::Autolock _l(mLock); 691 acquireWakeLock_l(uid); 692} 693 694String16 AudioFlinger::ThreadBase::getWakeLockTag() 695{ 696 switch (mType) { 697 case MIXER: 698 return String16("AudioMix"); 699 case DIRECT: 700 return String16("AudioDirectOut"); 701 case DUPLICATING: 702 return String16("AudioDup"); 703 case RECORD: 704 return String16("AudioIn"); 705 case OFFLOAD: 706 return String16("AudioOffload"); 707 default: 708 ALOG_ASSERT(false); 709 return String16("AudioUnknown"); 710 } 711} 712 713void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 714{ 715 getPowerManager_l(); 716 if (mPowerManager != 0) { 717 sp<IBinder> binder = new BBinder(); 718 status_t status; 719 if (uid >= 0) { 720 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 721 binder, 722 getWakeLockTag(), 723 String16("media"), 724 uid, 725 true /* FIXME force oneway contrary to .aidl */); 726 } else { 727 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 728 binder, 729 getWakeLockTag(), 730 String16("media"), 731 true /* FIXME force oneway contrary to .aidl */); 732 } 733 if (status == NO_ERROR) { 734 mWakeLockToken = binder; 735 } 736 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 737 } 738} 739 740void AudioFlinger::ThreadBase::releaseWakeLock() 741{ 742 Mutex::Autolock _l(mLock); 743 releaseWakeLock_l(); 744} 745 746void AudioFlinger::ThreadBase::releaseWakeLock_l() 747{ 748 if (mWakeLockToken != 0) { 749 ALOGV("releaseWakeLock_l() %s", mName); 750 if (mPowerManager != 0) { 751 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 752 true /* FIXME force oneway contrary to .aidl */); 753 } 754 mWakeLockToken.clear(); 755 } 756} 757 758void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 759 Mutex::Autolock _l(mLock); 760 updateWakeLockUids_l(uids); 761} 762 763void AudioFlinger::ThreadBase::getPowerManager_l() { 764 765 if (mPowerManager == 0) { 766 // use checkService() to avoid blocking if power service is not up yet 767 sp<IBinder> binder = 768 defaultServiceManager()->checkService(String16("power")); 769 if (binder == 0) { 770 ALOGW("Thread %s cannot connect to the power manager service", mName); 771 } else { 772 mPowerManager = interface_cast<IPowerManager>(binder); 773 binder->linkToDeath(mDeathRecipient); 774 } 775 } 776} 777 778void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 779 780 getPowerManager_l(); 781 if (mWakeLockToken == NULL) { 782 ALOGE("no wake lock to update!"); 783 return; 784 } 785 if (mPowerManager != 0) { 786 sp<IBinder> binder = new BBinder(); 787 status_t status; 788 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 789 true /* FIXME force oneway contrary to .aidl */); 790 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 791 } 792} 793 794void AudioFlinger::ThreadBase::clearPowerManager() 795{ 796 Mutex::Autolock _l(mLock); 797 releaseWakeLock_l(); 798 mPowerManager.clear(); 799} 800 801void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 802{ 803 sp<ThreadBase> thread = mThread.promote(); 804 if (thread != 0) { 805 thread->clearPowerManager(); 806 } 807 ALOGW("power manager service died !!!"); 808} 809 810void AudioFlinger::ThreadBase::setEffectSuspended( 811 const effect_uuid_t *type, bool suspend, int sessionId) 812{ 813 Mutex::Autolock _l(mLock); 814 setEffectSuspended_l(type, suspend, sessionId); 815} 816 817void AudioFlinger::ThreadBase::setEffectSuspended_l( 818 const effect_uuid_t *type, bool suspend, int sessionId) 819{ 820 sp<EffectChain> chain = getEffectChain_l(sessionId); 821 if (chain != 0) { 822 if (type != NULL) { 823 chain->setEffectSuspended_l(type, suspend); 824 } else { 825 chain->setEffectSuspendedAll_l(suspend); 826 } 827 } 828 829 updateSuspendedSessions_l(type, suspend, sessionId); 830} 831 832void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 833{ 834 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 835 if (index < 0) { 836 return; 837 } 838 839 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 840 mSuspendedSessions.valueAt(index); 841 842 for (size_t i = 0; i < sessionEffects.size(); i++) { 843 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 844 for (int j = 0; j < desc->mRefCount; j++) { 845 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 846 chain->setEffectSuspendedAll_l(true); 847 } else { 848 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 849 desc->mType.timeLow); 850 chain->setEffectSuspended_l(&desc->mType, true); 851 } 852 } 853 } 854} 855 856void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 857 bool suspend, 858 int sessionId) 859{ 860 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 861 862 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 863 864 if (suspend) { 865 if (index >= 0) { 866 sessionEffects = mSuspendedSessions.valueAt(index); 867 } else { 868 mSuspendedSessions.add(sessionId, sessionEffects); 869 } 870 } else { 871 if (index < 0) { 872 return; 873 } 874 sessionEffects = mSuspendedSessions.valueAt(index); 875 } 876 877 878 int key = EffectChain::kKeyForSuspendAll; 879 if (type != NULL) { 880 key = type->timeLow; 881 } 882 index = sessionEffects.indexOfKey(key); 883 884 sp<SuspendedSessionDesc> desc; 885 if (suspend) { 886 if (index >= 0) { 887 desc = sessionEffects.valueAt(index); 888 } else { 889 desc = new SuspendedSessionDesc(); 890 if (type != NULL) { 891 desc->mType = *type; 892 } 893 sessionEffects.add(key, desc); 894 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 895 } 896 desc->mRefCount++; 897 } else { 898 if (index < 0) { 899 return; 900 } 901 desc = sessionEffects.valueAt(index); 902 if (--desc->mRefCount == 0) { 903 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 904 sessionEffects.removeItemsAt(index); 905 if (sessionEffects.isEmpty()) { 906 ALOGV("updateSuspendedSessions_l() restore removing session %d", 907 sessionId); 908 mSuspendedSessions.removeItem(sessionId); 909 } 910 } 911 } 912 if (!sessionEffects.isEmpty()) { 913 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 914 } 915} 916 917void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 918 bool enabled, 919 int sessionId) 920{ 921 Mutex::Autolock _l(mLock); 922 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 923} 924 925void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 926 bool enabled, 927 int sessionId) 928{ 929 if (mType != RECORD) { 930 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 931 // another session. This gives the priority to well behaved effect control panels 932 // and applications not using global effects. 933 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 934 // global effects 935 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 936 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 937 } 938 } 939 940 sp<EffectChain> chain = getEffectChain_l(sessionId); 941 if (chain != 0) { 942 chain->checkSuspendOnEffectEnabled(effect, enabled); 943 } 944} 945 946// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 947sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 948 const sp<AudioFlinger::Client>& client, 949 const sp<IEffectClient>& effectClient, 950 int32_t priority, 951 int sessionId, 952 effect_descriptor_t *desc, 953 int *enabled, 954 status_t *status) 955{ 956 sp<EffectModule> effect; 957 sp<EffectHandle> handle; 958 status_t lStatus; 959 sp<EffectChain> chain; 960 bool chainCreated = false; 961 bool effectCreated = false; 962 bool effectRegistered = false; 963 964 lStatus = initCheck(); 965 if (lStatus != NO_ERROR) { 966 ALOGW("createEffect_l() Audio driver not initialized."); 967 goto Exit; 968 } 969 970 // Reject any effect on Direct output threads for now, since the format of 971 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 972 if (mType == DIRECT) { 973 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 974 desc->name, mName); 975 lStatus = BAD_VALUE; 976 goto Exit; 977 } 978 979 // Reject any effect on mixer or duplicating multichannel sinks. 980 // TODO: fix both format and multichannel issues with effects. 981 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 982 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 983 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 984 lStatus = BAD_VALUE; 985 goto Exit; 986 } 987 988 // Allow global effects only on offloaded and mixer threads 989 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 990 switch (mType) { 991 case MIXER: 992 case OFFLOAD: 993 break; 994 case DIRECT: 995 case DUPLICATING: 996 case RECORD: 997 default: 998 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 999 lStatus = BAD_VALUE; 1000 goto Exit; 1001 } 1002 } 1003 1004 // Only Pre processor effects are allowed on input threads and only on input threads 1005 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1006 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1007 desc->name, desc->flags, mType); 1008 lStatus = BAD_VALUE; 1009 goto Exit; 1010 } 1011 1012 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1013 1014 { // scope for mLock 1015 Mutex::Autolock _l(mLock); 1016 1017 // check for existing effect chain with the requested audio session 1018 chain = getEffectChain_l(sessionId); 1019 if (chain == 0) { 1020 // create a new chain for this session 1021 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1022 chain = new EffectChain(this, sessionId); 1023 addEffectChain_l(chain); 1024 chain->setStrategy(getStrategyForSession_l(sessionId)); 1025 chainCreated = true; 1026 } else { 1027 effect = chain->getEffectFromDesc_l(desc); 1028 } 1029 1030 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1031 1032 if (effect == 0) { 1033 int id = mAudioFlinger->nextUniqueId(); 1034 // Check CPU and memory usage 1035 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1036 if (lStatus != NO_ERROR) { 1037 goto Exit; 1038 } 1039 effectRegistered = true; 1040 // create a new effect module if none present in the chain 1041 effect = new EffectModule(this, chain, desc, id, sessionId); 1042 lStatus = effect->status(); 1043 if (lStatus != NO_ERROR) { 1044 goto Exit; 1045 } 1046 effect->setOffloaded(mType == OFFLOAD, mId); 1047 1048 lStatus = chain->addEffect_l(effect); 1049 if (lStatus != NO_ERROR) { 1050 goto Exit; 1051 } 1052 effectCreated = true; 1053 1054 effect->setDevice(mOutDevice); 1055 effect->setDevice(mInDevice); 1056 effect->setMode(mAudioFlinger->getMode()); 1057 effect->setAudioSource(mAudioSource); 1058 } 1059 // create effect handle and connect it to effect module 1060 handle = new EffectHandle(effect, client, effectClient, priority); 1061 lStatus = handle->initCheck(); 1062 if (lStatus == OK) { 1063 lStatus = effect->addHandle(handle.get()); 1064 } 1065 if (enabled != NULL) { 1066 *enabled = (int)effect->isEnabled(); 1067 } 1068 } 1069 1070Exit: 1071 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1072 Mutex::Autolock _l(mLock); 1073 if (effectCreated) { 1074 chain->removeEffect_l(effect); 1075 } 1076 if (effectRegistered) { 1077 AudioSystem::unregisterEffect(effect->id()); 1078 } 1079 if (chainCreated) { 1080 removeEffectChain_l(chain); 1081 } 1082 handle.clear(); 1083 } 1084 1085 *status = lStatus; 1086 return handle; 1087} 1088 1089sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1090{ 1091 Mutex::Autolock _l(mLock); 1092 return getEffect_l(sessionId, effectId); 1093} 1094 1095sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1096{ 1097 sp<EffectChain> chain = getEffectChain_l(sessionId); 1098 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1099} 1100 1101// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1102// PlaybackThread::mLock held 1103status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1104{ 1105 // check for existing effect chain with the requested audio session 1106 int sessionId = effect->sessionId(); 1107 sp<EffectChain> chain = getEffectChain_l(sessionId); 1108 bool chainCreated = false; 1109 1110 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1111 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1112 this, effect->desc().name, effect->desc().flags); 1113 1114 if (chain == 0) { 1115 // create a new chain for this session 1116 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1117 chain = new EffectChain(this, sessionId); 1118 addEffectChain_l(chain); 1119 chain->setStrategy(getStrategyForSession_l(sessionId)); 1120 chainCreated = true; 1121 } 1122 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1123 1124 if (chain->getEffectFromId_l(effect->id()) != 0) { 1125 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1126 this, effect->desc().name, chain.get()); 1127 return BAD_VALUE; 1128 } 1129 1130 effect->setOffloaded(mType == OFFLOAD, mId); 1131 1132 status_t status = chain->addEffect_l(effect); 1133 if (status != NO_ERROR) { 1134 if (chainCreated) { 1135 removeEffectChain_l(chain); 1136 } 1137 return status; 1138 } 1139 1140 effect->setDevice(mOutDevice); 1141 effect->setDevice(mInDevice); 1142 effect->setMode(mAudioFlinger->getMode()); 1143 effect->setAudioSource(mAudioSource); 1144 return NO_ERROR; 1145} 1146 1147void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1148 1149 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1150 effect_descriptor_t desc = effect->desc(); 1151 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1152 detachAuxEffect_l(effect->id()); 1153 } 1154 1155 sp<EffectChain> chain = effect->chain().promote(); 1156 if (chain != 0) { 1157 // remove effect chain if removing last effect 1158 if (chain->removeEffect_l(effect) == 0) { 1159 removeEffectChain_l(chain); 1160 } 1161 } else { 1162 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1163 } 1164} 1165 1166void AudioFlinger::ThreadBase::lockEffectChains_l( 1167 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1168{ 1169 effectChains = mEffectChains; 1170 for (size_t i = 0; i < mEffectChains.size(); i++) { 1171 mEffectChains[i]->lock(); 1172 } 1173} 1174 1175void AudioFlinger::ThreadBase::unlockEffectChains( 1176 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1177{ 1178 for (size_t i = 0; i < effectChains.size(); i++) { 1179 effectChains[i]->unlock(); 1180 } 1181} 1182 1183sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1184{ 1185 Mutex::Autolock _l(mLock); 1186 return getEffectChain_l(sessionId); 1187} 1188 1189sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1190{ 1191 size_t size = mEffectChains.size(); 1192 for (size_t i = 0; i < size; i++) { 1193 if (mEffectChains[i]->sessionId() == sessionId) { 1194 return mEffectChains[i]; 1195 } 1196 } 1197 return 0; 1198} 1199 1200void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1201{ 1202 Mutex::Autolock _l(mLock); 1203 size_t size = mEffectChains.size(); 1204 for (size_t i = 0; i < size; i++) { 1205 mEffectChains[i]->setMode_l(mode); 1206 } 1207} 1208 1209void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1210{ 1211 config->type = AUDIO_PORT_TYPE_MIX; 1212 config->ext.mix.handle = mId; 1213 config->sample_rate = mSampleRate; 1214 config->format = mFormat; 1215 config->channel_mask = mChannelMask; 1216 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1217 AUDIO_PORT_CONFIG_FORMAT; 1218} 1219 1220 1221// ---------------------------------------------------------------------------- 1222// Playback 1223// ---------------------------------------------------------------------------- 1224 1225AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1226 AudioStreamOut* output, 1227 audio_io_handle_t id, 1228 audio_devices_t device, 1229 type_t type) 1230 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1231 mNormalFrameCount(0), mSinkBuffer(NULL), 1232 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1233 mMixerBuffer(NULL), 1234 mMixerBufferSize(0), 1235 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1236 mMixerBufferValid(false), 1237 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1238 mEffectBuffer(NULL), 1239 mEffectBufferSize(0), 1240 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1241 mEffectBufferValid(false), 1242 mSuspended(0), mBytesWritten(0), 1243 mActiveTracksGeneration(0), 1244 // mStreamTypes[] initialized in constructor body 1245 mOutput(output), 1246 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1247 mMixerStatus(MIXER_IDLE), 1248 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1249 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1250 mBytesRemaining(0), 1251 mCurrentWriteLength(0), 1252 mUseAsyncWrite(false), 1253 mWriteAckSequence(0), 1254 mDrainSequence(0), 1255 mSignalPending(false), 1256 mScreenState(AudioFlinger::mScreenState), 1257 // index 0 is reserved for normal mixer's submix 1258 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1259 // mLatchD, mLatchQ, 1260 mLatchDValid(false), mLatchQValid(false) 1261{ 1262 snprintf(mName, kNameLength, "AudioOut_%X", id); 1263 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1264 1265 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1266 // it would be safer to explicitly pass initial masterVolume/masterMute as 1267 // parameter. 1268 // 1269 // If the HAL we are using has support for master volume or master mute, 1270 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1271 // and the mute set to false). 1272 mMasterVolume = audioFlinger->masterVolume_l(); 1273 mMasterMute = audioFlinger->masterMute_l(); 1274 if (mOutput && mOutput->audioHwDev) { 1275 if (mOutput->audioHwDev->canSetMasterVolume()) { 1276 mMasterVolume = 1.0; 1277 } 1278 1279 if (mOutput->audioHwDev->canSetMasterMute()) { 1280 mMasterMute = false; 1281 } 1282 } 1283 1284 readOutputParameters_l(); 1285 1286 // ++ operator does not compile 1287 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1288 stream = (audio_stream_type_t) (stream + 1)) { 1289 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1290 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1291 } 1292} 1293 1294AudioFlinger::PlaybackThread::~PlaybackThread() 1295{ 1296 mAudioFlinger->unregisterWriter(mNBLogWriter); 1297 free(mSinkBuffer); 1298 free(mMixerBuffer); 1299 free(mEffectBuffer); 1300} 1301 1302void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1303{ 1304 dumpInternals(fd, args); 1305 dumpTracks(fd, args); 1306 dumpEffectChains(fd, args); 1307} 1308 1309void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1310{ 1311 const size_t SIZE = 256; 1312 char buffer[SIZE]; 1313 String8 result; 1314 1315 result.appendFormat(" Stream volumes in dB: "); 1316 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1317 const stream_type_t *st = &mStreamTypes[i]; 1318 if (i > 0) { 1319 result.appendFormat(", "); 1320 } 1321 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1322 if (st->mute) { 1323 result.append("M"); 1324 } 1325 } 1326 result.append("\n"); 1327 write(fd, result.string(), result.length()); 1328 result.clear(); 1329 1330 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1331 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1332 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1333 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1334 1335 size_t numtracks = mTracks.size(); 1336 size_t numactive = mActiveTracks.size(); 1337 dprintf(fd, " %d Tracks", numtracks); 1338 size_t numactiveseen = 0; 1339 if (numtracks) { 1340 dprintf(fd, " of which %d are active\n", numactive); 1341 Track::appendDumpHeader(result); 1342 for (size_t i = 0; i < numtracks; ++i) { 1343 sp<Track> track = mTracks[i]; 1344 if (track != 0) { 1345 bool active = mActiveTracks.indexOf(track) >= 0; 1346 if (active) { 1347 numactiveseen++; 1348 } 1349 track->dump(buffer, SIZE, active); 1350 result.append(buffer); 1351 } 1352 } 1353 } else { 1354 result.append("\n"); 1355 } 1356 if (numactiveseen != numactive) { 1357 // some tracks in the active list were not in the tracks list 1358 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1359 " not in the track list\n"); 1360 result.append(buffer); 1361 Track::appendDumpHeader(result); 1362 for (size_t i = 0; i < numactive; ++i) { 1363 sp<Track> track = mActiveTracks[i].promote(); 1364 if (track != 0 && mTracks.indexOf(track) < 0) { 1365 track->dump(buffer, SIZE, true); 1366 result.append(buffer); 1367 } 1368 } 1369 } 1370 1371 write(fd, result.string(), result.size()); 1372} 1373 1374void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1375{ 1376 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1377 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1378 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1379 dprintf(fd, " Total writes: %d\n", mNumWrites); 1380 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1381 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1382 dprintf(fd, " Suspend count: %d\n", mSuspended); 1383 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1384 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1385 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1386 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1387 AudioStreamOut *output = mOutput; 1388 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1389 String8 flagsAsString = outputFlagsToString(flags); 1390 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1391 1392 dumpBase(fd, args); 1393} 1394 1395// Thread virtuals 1396 1397void AudioFlinger::PlaybackThread::onFirstRef() 1398{ 1399 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1400} 1401 1402// ThreadBase virtuals 1403void AudioFlinger::PlaybackThread::preExit() 1404{ 1405 ALOGV(" preExit()"); 1406 // FIXME this is using hard-coded strings but in the future, this functionality will be 1407 // converted to use audio HAL extensions required to support tunneling 1408 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1409} 1410 1411// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1412sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1413 const sp<AudioFlinger::Client>& client, 1414 audio_stream_type_t streamType, 1415 uint32_t sampleRate, 1416 audio_format_t format, 1417 audio_channel_mask_t channelMask, 1418 size_t *pFrameCount, 1419 const sp<IMemory>& sharedBuffer, 1420 int sessionId, 1421 IAudioFlinger::track_flags_t *flags, 1422 pid_t tid, 1423 int uid, 1424 status_t *status) 1425{ 1426 size_t frameCount = *pFrameCount; 1427 sp<Track> track; 1428 status_t lStatus; 1429 1430 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1431 1432 // client expresses a preference for FAST, but we get the final say 1433 if (*flags & IAudioFlinger::TRACK_FAST) { 1434 if ( 1435 // not timed 1436 (!isTimed) && 1437 // either of these use cases: 1438 ( 1439 // use case 1: shared buffer with any frame count 1440 ( 1441 (sharedBuffer != 0) 1442 ) || 1443 // use case 2: callback handler and frame count is default or at least as large as HAL 1444 ( 1445 (tid != -1) && 1446 ((frameCount == 0) || 1447 (frameCount >= mFrameCount)) 1448 ) 1449 ) && 1450 // PCM data 1451 audio_is_linear_pcm(format) && 1452 // identical channel mask to sink, or mono in and stereo sink 1453 (channelMask == mChannelMask || 1454 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1455 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1456 // hardware sample rate 1457 (sampleRate == mSampleRate) && 1458 // normal mixer has an associated fast mixer 1459 hasFastMixer() && 1460 // there are sufficient fast track slots available 1461 (mFastTrackAvailMask != 0) 1462 // FIXME test that MixerThread for this fast track has a capable output HAL 1463 // FIXME add a permission test also? 1464 ) { 1465 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1466 if (frameCount == 0) { 1467 // read the fast track multiplier property the first time it is needed 1468 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1469 if (ok != 0) { 1470 ALOGE("%s pthread_once failed: %d", __func__, ok); 1471 } 1472 frameCount = mFrameCount * sFastTrackMultiplier; 1473 } 1474 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1475 frameCount, mFrameCount); 1476 } else { 1477 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1478 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1479 "sampleRate=%u mSampleRate=%u " 1480 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1481 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1482 audio_is_linear_pcm(format), 1483 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1484 *flags &= ~IAudioFlinger::TRACK_FAST; 1485 // For compatibility with AudioTrack calculation, buffer depth is forced 1486 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1487 // This is probably too conservative, but legacy application code may depend on it. 1488 // If you change this calculation, also review the start threshold which is related. 1489 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1490 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1491 if (minBufCount < 2) { 1492 minBufCount = 2; 1493 } 1494 size_t minFrameCount = mNormalFrameCount * minBufCount; 1495 if (frameCount < minFrameCount) { 1496 frameCount = minFrameCount; 1497 } 1498 } 1499 } 1500 *pFrameCount = frameCount; 1501 1502 switch (mType) { 1503 1504 case DIRECT: 1505 if (audio_is_linear_pcm(format)) { 1506 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1507 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1508 "for output %p with format %#x", 1509 sampleRate, format, channelMask, mOutput, mFormat); 1510 lStatus = BAD_VALUE; 1511 goto Exit; 1512 } 1513 } 1514 break; 1515 1516 case OFFLOAD: 1517 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1518 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1519 "for output %p with format %#x", 1520 sampleRate, format, channelMask, mOutput, mFormat); 1521 lStatus = BAD_VALUE; 1522 goto Exit; 1523 } 1524 break; 1525 1526 default: 1527 if (!audio_is_linear_pcm(format)) { 1528 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1529 "for output %p with format %#x", 1530 format, mOutput, mFormat); 1531 lStatus = BAD_VALUE; 1532 goto Exit; 1533 } 1534 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1535 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1536 lStatus = BAD_VALUE; 1537 goto Exit; 1538 } 1539 break; 1540 1541 } 1542 1543 lStatus = initCheck(); 1544 if (lStatus != NO_ERROR) { 1545 ALOGE("createTrack_l() audio driver not initialized"); 1546 goto Exit; 1547 } 1548 1549 { // scope for mLock 1550 Mutex::Autolock _l(mLock); 1551 1552 // all tracks in same audio session must share the same routing strategy otherwise 1553 // conflicts will happen when tracks are moved from one output to another by audio policy 1554 // manager 1555 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1556 for (size_t i = 0; i < mTracks.size(); ++i) { 1557 sp<Track> t = mTracks[i]; 1558 if (t != 0 && t->isExternalTrack()) { 1559 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1560 if (sessionId == t->sessionId() && strategy != actual) { 1561 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1562 strategy, actual); 1563 lStatus = BAD_VALUE; 1564 goto Exit; 1565 } 1566 } 1567 } 1568 1569 if (!isTimed) { 1570 track = new Track(this, client, streamType, sampleRate, format, 1571 channelMask, frameCount, NULL, sharedBuffer, 1572 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1573 } else { 1574 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1575 channelMask, frameCount, sharedBuffer, sessionId, uid); 1576 } 1577 1578 // new Track always returns non-NULL, 1579 // but TimedTrack::create() is a factory that could fail by returning NULL 1580 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1581 if (lStatus != NO_ERROR) { 1582 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1583 // track must be cleared from the caller as the caller has the AF lock 1584 goto Exit; 1585 } 1586 mTracks.add(track); 1587 1588 sp<EffectChain> chain = getEffectChain_l(sessionId); 1589 if (chain != 0) { 1590 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1591 track->setMainBuffer(chain->inBuffer()); 1592 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1593 chain->incTrackCnt(); 1594 } 1595 1596 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1597 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1598 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1599 // so ask activity manager to do this on our behalf 1600 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1601 } 1602 } 1603 1604 lStatus = NO_ERROR; 1605 1606Exit: 1607 *status = lStatus; 1608 return track; 1609} 1610 1611uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1612{ 1613 return latency; 1614} 1615 1616uint32_t AudioFlinger::PlaybackThread::latency() const 1617{ 1618 Mutex::Autolock _l(mLock); 1619 return latency_l(); 1620} 1621uint32_t AudioFlinger::PlaybackThread::latency_l() const 1622{ 1623 if (initCheck() == NO_ERROR) { 1624 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1625 } else { 1626 return 0; 1627 } 1628} 1629 1630void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1631{ 1632 Mutex::Autolock _l(mLock); 1633 // Don't apply master volume in SW if our HAL can do it for us. 1634 if (mOutput && mOutput->audioHwDev && 1635 mOutput->audioHwDev->canSetMasterVolume()) { 1636 mMasterVolume = 1.0; 1637 } else { 1638 mMasterVolume = value; 1639 } 1640} 1641 1642void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1643{ 1644 Mutex::Autolock _l(mLock); 1645 // Don't apply master mute in SW if our HAL can do it for us. 1646 if (mOutput && mOutput->audioHwDev && 1647 mOutput->audioHwDev->canSetMasterMute()) { 1648 mMasterMute = false; 1649 } else { 1650 mMasterMute = muted; 1651 } 1652} 1653 1654void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1655{ 1656 Mutex::Autolock _l(mLock); 1657 mStreamTypes[stream].volume = value; 1658 broadcast_l(); 1659} 1660 1661void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1662{ 1663 Mutex::Autolock _l(mLock); 1664 mStreamTypes[stream].mute = muted; 1665 broadcast_l(); 1666} 1667 1668float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1669{ 1670 Mutex::Autolock _l(mLock); 1671 return mStreamTypes[stream].volume; 1672} 1673 1674// addTrack_l() must be called with ThreadBase::mLock held 1675status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1676{ 1677 status_t status = ALREADY_EXISTS; 1678 1679 // set retry count for buffer fill 1680 track->mRetryCount = kMaxTrackStartupRetries; 1681 if (mActiveTracks.indexOf(track) < 0) { 1682 // the track is newly added, make sure it fills up all its 1683 // buffers before playing. This is to ensure the client will 1684 // effectively get the latency it requested. 1685 if (track->isExternalTrack()) { 1686 TrackBase::track_state state = track->mState; 1687 mLock.unlock(); 1688 status = AudioSystem::startOutput(mId, track->streamType(), 1689 (audio_session_t)track->sessionId()); 1690 mLock.lock(); 1691 // abort track was stopped/paused while we released the lock 1692 if (state != track->mState) { 1693 if (status == NO_ERROR) { 1694 mLock.unlock(); 1695 AudioSystem::stopOutput(mId, track->streamType(), 1696 (audio_session_t)track->sessionId()); 1697 mLock.lock(); 1698 } 1699 return INVALID_OPERATION; 1700 } 1701 // abort if start is rejected by audio policy manager 1702 if (status != NO_ERROR) { 1703 return PERMISSION_DENIED; 1704 } 1705#ifdef ADD_BATTERY_DATA 1706 // to track the speaker usage 1707 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1708#endif 1709 } 1710 1711 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1712 track->mResetDone = false; 1713 track->mPresentationCompleteFrames = 0; 1714 mActiveTracks.add(track); 1715 mWakeLockUids.add(track->uid()); 1716 mActiveTracksGeneration++; 1717 mLatestActiveTrack = track; 1718 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1719 if (chain != 0) { 1720 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1721 track->sessionId()); 1722 chain->incActiveTrackCnt(); 1723 } 1724 1725 status = NO_ERROR; 1726 } 1727 1728 onAddNewTrack_l(); 1729 return status; 1730} 1731 1732bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1733{ 1734 track->terminate(); 1735 // active tracks are removed by threadLoop() 1736 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1737 track->mState = TrackBase::STOPPED; 1738 if (!trackActive) { 1739 removeTrack_l(track); 1740 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1741 track->mState = TrackBase::STOPPING_1; 1742 } 1743 1744 return trackActive; 1745} 1746 1747void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1748{ 1749 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1750 mTracks.remove(track); 1751 deleteTrackName_l(track->name()); 1752 // redundant as track is about to be destroyed, for dumpsys only 1753 track->mName = -1; 1754 if (track->isFastTrack()) { 1755 int index = track->mFastIndex; 1756 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1757 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1758 mFastTrackAvailMask |= 1 << index; 1759 // redundant as track is about to be destroyed, for dumpsys only 1760 track->mFastIndex = -1; 1761 } 1762 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1763 if (chain != 0) { 1764 chain->decTrackCnt(); 1765 } 1766} 1767 1768void AudioFlinger::PlaybackThread::broadcast_l() 1769{ 1770 // Thread could be blocked waiting for async 1771 // so signal it to handle state changes immediately 1772 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1773 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1774 mSignalPending = true; 1775 mWaitWorkCV.broadcast(); 1776} 1777 1778String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1779{ 1780 Mutex::Autolock _l(mLock); 1781 if (initCheck() != NO_ERROR) { 1782 return String8(); 1783 } 1784 1785 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1786 const String8 out_s8(s); 1787 free(s); 1788 return out_s8; 1789} 1790 1791void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1792 AudioSystem::OutputDescriptor desc; 1793 void *param2 = NULL; 1794 1795 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1796 param); 1797 1798 switch (event) { 1799 case AudioSystem::OUTPUT_OPENED: 1800 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1801 desc.channelMask = mChannelMask; 1802 desc.samplingRate = mSampleRate; 1803 desc.format = mFormat; 1804 desc.frameCount = mNormalFrameCount; // FIXME see 1805 // AudioFlinger::frameCount(audio_io_handle_t) 1806 desc.latency = latency_l(); 1807 param2 = &desc; 1808 break; 1809 1810 case AudioSystem::STREAM_CONFIG_CHANGED: 1811 param2 = ¶m; 1812 case AudioSystem::OUTPUT_CLOSED: 1813 default: 1814 break; 1815 } 1816 mAudioFlinger->audioConfigChanged(event, mId, param2); 1817} 1818 1819void AudioFlinger::PlaybackThread::writeCallback() 1820{ 1821 ALOG_ASSERT(mCallbackThread != 0); 1822 mCallbackThread->resetWriteBlocked(); 1823} 1824 1825void AudioFlinger::PlaybackThread::drainCallback() 1826{ 1827 ALOG_ASSERT(mCallbackThread != 0); 1828 mCallbackThread->resetDraining(); 1829} 1830 1831void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1832{ 1833 Mutex::Autolock _l(mLock); 1834 // reject out of sequence requests 1835 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1836 mWriteAckSequence &= ~1; 1837 mWaitWorkCV.signal(); 1838 } 1839} 1840 1841void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1842{ 1843 Mutex::Autolock _l(mLock); 1844 // reject out of sequence requests 1845 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1846 mDrainSequence &= ~1; 1847 mWaitWorkCV.signal(); 1848 } 1849} 1850 1851// static 1852int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1853 void *param __unused, 1854 void *cookie) 1855{ 1856 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1857 ALOGV("asyncCallback() event %d", event); 1858 switch (event) { 1859 case STREAM_CBK_EVENT_WRITE_READY: 1860 me->writeCallback(); 1861 break; 1862 case STREAM_CBK_EVENT_DRAIN_READY: 1863 me->drainCallback(); 1864 break; 1865 default: 1866 ALOGW("asyncCallback() unknown event %d", event); 1867 break; 1868 } 1869 return 0; 1870} 1871 1872void AudioFlinger::PlaybackThread::readOutputParameters_l() 1873{ 1874 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1875 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1876 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1877 if (!audio_is_output_channel(mChannelMask)) { 1878 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1879 } 1880 if ((mType == MIXER || mType == DUPLICATING) 1881 && !isValidPcmSinkChannelMask(mChannelMask)) { 1882 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1883 mChannelMask); 1884 } 1885 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1886 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1887 mFormat = mHALFormat; 1888 if (!audio_is_valid_format(mFormat)) { 1889 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1890 } 1891 if ((mType == MIXER || mType == DUPLICATING) 1892 && !isValidPcmSinkFormat(mFormat)) { 1893 LOG_FATAL("HAL format %#x not supported for mixed output", 1894 mFormat); 1895 } 1896 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1897 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1898 mFrameCount = mBufferSize / mFrameSize; 1899 if (mFrameCount & 15) { 1900 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1901 mFrameCount); 1902 } 1903 1904 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1905 (mOutput->stream->set_callback != NULL)) { 1906 if (mOutput->stream->set_callback(mOutput->stream, 1907 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1908 mUseAsyncWrite = true; 1909 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1910 } 1911 } 1912 1913 // Calculate size of normal sink buffer relative to the HAL output buffer size 1914 double multiplier = 1.0; 1915 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1916 kUseFastMixer == FastMixer_Dynamic)) { 1917 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1918 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1919 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1920 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1921 maxNormalFrameCount = maxNormalFrameCount & ~15; 1922 if (maxNormalFrameCount < minNormalFrameCount) { 1923 maxNormalFrameCount = minNormalFrameCount; 1924 } 1925 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1926 if (multiplier <= 1.0) { 1927 multiplier = 1.0; 1928 } else if (multiplier <= 2.0) { 1929 if (2 * mFrameCount <= maxNormalFrameCount) { 1930 multiplier = 2.0; 1931 } else { 1932 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1933 } 1934 } else { 1935 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1936 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1937 // track, but we sometimes have to do this to satisfy the maximum frame count 1938 // constraint) 1939 // FIXME this rounding up should not be done if no HAL SRC 1940 uint32_t truncMult = (uint32_t) multiplier; 1941 if ((truncMult & 1)) { 1942 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1943 ++truncMult; 1944 } 1945 } 1946 multiplier = (double) truncMult; 1947 } 1948 } 1949 mNormalFrameCount = multiplier * mFrameCount; 1950 // round up to nearest 16 frames to satisfy AudioMixer 1951 if (mType == MIXER || mType == DUPLICATING) { 1952 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1953 } 1954 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1955 mNormalFrameCount); 1956 1957 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1958 // Originally this was int16_t[] array, need to remove legacy implications. 1959 free(mSinkBuffer); 1960 mSinkBuffer = NULL; 1961 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1962 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1963 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1964 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1965 1966 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1967 // drives the output. 1968 free(mMixerBuffer); 1969 mMixerBuffer = NULL; 1970 if (mMixerBufferEnabled) { 1971 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1972 mMixerBufferSize = mNormalFrameCount * mChannelCount 1973 * audio_bytes_per_sample(mMixerBufferFormat); 1974 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1975 } 1976 free(mEffectBuffer); 1977 mEffectBuffer = NULL; 1978 if (mEffectBufferEnabled) { 1979 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1980 mEffectBufferSize = mNormalFrameCount * mChannelCount 1981 * audio_bytes_per_sample(mEffectBufferFormat); 1982 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1983 } 1984 1985 // force reconfiguration of effect chains and engines to take new buffer size and audio 1986 // parameters into account 1987 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1988 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1989 // matter. 1990 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1991 Vector< sp<EffectChain> > effectChains = mEffectChains; 1992 for (size_t i = 0; i < effectChains.size(); i ++) { 1993 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1994 } 1995} 1996 1997 1998status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1999{ 2000 if (halFrames == NULL || dspFrames == NULL) { 2001 return BAD_VALUE; 2002 } 2003 Mutex::Autolock _l(mLock); 2004 if (initCheck() != NO_ERROR) { 2005 return INVALID_OPERATION; 2006 } 2007 size_t framesWritten = mBytesWritten / mFrameSize; 2008 *halFrames = framesWritten; 2009 2010 if (isSuspended()) { 2011 // return an estimation of rendered frames when the output is suspended 2012 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2013 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2014 return NO_ERROR; 2015 } else { 2016 status_t status; 2017 uint32_t frames; 2018 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 2019 *dspFrames = (size_t)frames; 2020 return status; 2021 } 2022} 2023 2024uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2025{ 2026 Mutex::Autolock _l(mLock); 2027 uint32_t result = 0; 2028 if (getEffectChain_l(sessionId) != 0) { 2029 result = EFFECT_SESSION; 2030 } 2031 2032 for (size_t i = 0; i < mTracks.size(); ++i) { 2033 sp<Track> track = mTracks[i]; 2034 if (sessionId == track->sessionId() && !track->isInvalid()) { 2035 result |= TRACK_SESSION; 2036 break; 2037 } 2038 } 2039 2040 return result; 2041} 2042 2043uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2044{ 2045 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2046 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2047 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2048 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2049 } 2050 for (size_t i = 0; i < mTracks.size(); i++) { 2051 sp<Track> track = mTracks[i]; 2052 if (sessionId == track->sessionId() && !track->isInvalid()) { 2053 return AudioSystem::getStrategyForStream(track->streamType()); 2054 } 2055 } 2056 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2057} 2058 2059 2060AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2061{ 2062 Mutex::Autolock _l(mLock); 2063 return mOutput; 2064} 2065 2066AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2067{ 2068 Mutex::Autolock _l(mLock); 2069 AudioStreamOut *output = mOutput; 2070 mOutput = NULL; 2071 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2072 // must push a NULL and wait for ack 2073 mOutputSink.clear(); 2074 mPipeSink.clear(); 2075 mNormalSink.clear(); 2076 return output; 2077} 2078 2079// this method must always be called either with ThreadBase mLock held or inside the thread loop 2080audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2081{ 2082 if (mOutput == NULL) { 2083 return NULL; 2084 } 2085 return &mOutput->stream->common; 2086} 2087 2088uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2089{ 2090 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2091} 2092 2093status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2094{ 2095 if (!isValidSyncEvent(event)) { 2096 return BAD_VALUE; 2097 } 2098 2099 Mutex::Autolock _l(mLock); 2100 2101 for (size_t i = 0; i < mTracks.size(); ++i) { 2102 sp<Track> track = mTracks[i]; 2103 if (event->triggerSession() == track->sessionId()) { 2104 (void) track->setSyncEvent(event); 2105 return NO_ERROR; 2106 } 2107 } 2108 2109 return NAME_NOT_FOUND; 2110} 2111 2112bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2113{ 2114 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2115} 2116 2117void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2118 const Vector< sp<Track> >& tracksToRemove) 2119{ 2120 size_t count = tracksToRemove.size(); 2121 if (count > 0) { 2122 for (size_t i = 0 ; i < count ; i++) { 2123 const sp<Track>& track = tracksToRemove.itemAt(i); 2124 if (track->isExternalTrack()) { 2125 AudioSystem::stopOutput(mId, track->streamType(), 2126 (audio_session_t)track->sessionId()); 2127#ifdef ADD_BATTERY_DATA 2128 // to track the speaker usage 2129 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2130#endif 2131 if (track->isTerminated()) { 2132 AudioSystem::releaseOutput(mId, track->streamType(), 2133 (audio_session_t)track->sessionId()); 2134 } 2135 } 2136 } 2137 } 2138} 2139 2140void AudioFlinger::PlaybackThread::checkSilentMode_l() 2141{ 2142 if (!mMasterMute) { 2143 char value[PROPERTY_VALUE_MAX]; 2144 if (property_get("ro.audio.silent", value, "0") > 0) { 2145 char *endptr; 2146 unsigned long ul = strtoul(value, &endptr, 0); 2147 if (*endptr == '\0' && ul != 0) { 2148 ALOGD("Silence is golden"); 2149 // The setprop command will not allow a property to be changed after 2150 // the first time it is set, so we don't have to worry about un-muting. 2151 setMasterMute_l(true); 2152 } 2153 } 2154 } 2155} 2156 2157// shared by MIXER and DIRECT, overridden by DUPLICATING 2158ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2159{ 2160 // FIXME rewrite to reduce number of system calls 2161 mLastWriteTime = systemTime(); 2162 mInWrite = true; 2163 ssize_t bytesWritten; 2164 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2165 2166 // If an NBAIO sink is present, use it to write the normal mixer's submix 2167 if (mNormalSink != 0) { 2168 2169 const size_t count = mBytesRemaining / mFrameSize; 2170 2171 ATRACE_BEGIN("write"); 2172 // update the setpoint when AudioFlinger::mScreenState changes 2173 uint32_t screenState = AudioFlinger::mScreenState; 2174 if (screenState != mScreenState) { 2175 mScreenState = screenState; 2176 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2177 if (pipe != NULL) { 2178 pipe->setAvgFrames((mScreenState & 1) ? 2179 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2180 } 2181 } 2182 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2183 ATRACE_END(); 2184 if (framesWritten > 0) { 2185 bytesWritten = framesWritten * mFrameSize; 2186 } else { 2187 bytesWritten = framesWritten; 2188 } 2189 mLatchDValid = false; 2190 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2191 if (status == NO_ERROR) { 2192 size_t totalFramesWritten = mNormalSink->framesWritten(); 2193 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2194 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2195 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2196 mLatchDValid = true; 2197 } 2198 } 2199 // otherwise use the HAL / AudioStreamOut directly 2200 } else { 2201 // Direct output and offload threads 2202 2203 if (mUseAsyncWrite) { 2204 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2205 mWriteAckSequence += 2; 2206 mWriteAckSequence |= 1; 2207 ALOG_ASSERT(mCallbackThread != 0); 2208 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2209 } 2210 // FIXME We should have an implementation of timestamps for direct output threads. 2211 // They are used e.g for multichannel PCM playback over HDMI. 2212 bytesWritten = mOutput->stream->write(mOutput->stream, 2213 (char *)mSinkBuffer + offset, mBytesRemaining); 2214 if (mUseAsyncWrite && 2215 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2216 // do not wait for async callback in case of error of full write 2217 mWriteAckSequence &= ~1; 2218 ALOG_ASSERT(mCallbackThread != 0); 2219 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2220 } 2221 } 2222 2223 mNumWrites++; 2224 mInWrite = false; 2225 mStandby = false; 2226 return bytesWritten; 2227} 2228 2229void AudioFlinger::PlaybackThread::threadLoop_drain() 2230{ 2231 if (mOutput->stream->drain) { 2232 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2233 if (mUseAsyncWrite) { 2234 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2235 mDrainSequence |= 1; 2236 ALOG_ASSERT(mCallbackThread != 0); 2237 mCallbackThread->setDraining(mDrainSequence); 2238 } 2239 mOutput->stream->drain(mOutput->stream, 2240 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2241 : AUDIO_DRAIN_ALL); 2242 } 2243} 2244 2245void AudioFlinger::PlaybackThread::threadLoop_exit() 2246{ 2247 // Default implementation has nothing to do 2248} 2249 2250/* 2251The derived values that are cached: 2252 - mSinkBufferSize from frame count * frame size 2253 - activeSleepTime from activeSleepTimeUs() 2254 - idleSleepTime from idleSleepTimeUs() 2255 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2256 - maxPeriod from frame count and sample rate (MIXER only) 2257 2258The parameters that affect these derived values are: 2259 - frame count 2260 - frame size 2261 - sample rate 2262 - device type: A2DP or not 2263 - device latency 2264 - format: PCM or not 2265 - active sleep time 2266 - idle sleep time 2267*/ 2268 2269void AudioFlinger::PlaybackThread::cacheParameters_l() 2270{ 2271 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2272 activeSleepTime = activeSleepTimeUs(); 2273 idleSleepTime = idleSleepTimeUs(); 2274} 2275 2276void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2277{ 2278 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2279 this, streamType, mTracks.size()); 2280 Mutex::Autolock _l(mLock); 2281 2282 size_t size = mTracks.size(); 2283 for (size_t i = 0; i < size; i++) { 2284 sp<Track> t = mTracks[i]; 2285 if (t->streamType() == streamType) { 2286 t->invalidate(); 2287 } 2288 } 2289} 2290 2291status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2292{ 2293 int session = chain->sessionId(); 2294 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2295 ? mEffectBuffer : mSinkBuffer); 2296 bool ownsBuffer = false; 2297 2298 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2299 if (session > 0) { 2300 // Only one effect chain can be present in direct output thread and it uses 2301 // the sink buffer as input 2302 if (mType != DIRECT) { 2303 size_t numSamples = mNormalFrameCount * mChannelCount; 2304 buffer = new int16_t[numSamples]; 2305 memset(buffer, 0, numSamples * sizeof(int16_t)); 2306 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2307 ownsBuffer = true; 2308 } 2309 2310 // Attach all tracks with same session ID to this chain. 2311 for (size_t i = 0; i < mTracks.size(); ++i) { 2312 sp<Track> track = mTracks[i]; 2313 if (session == track->sessionId()) { 2314 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2315 buffer); 2316 track->setMainBuffer(buffer); 2317 chain->incTrackCnt(); 2318 } 2319 } 2320 2321 // indicate all active tracks in the chain 2322 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2323 sp<Track> track = mActiveTracks[i].promote(); 2324 if (track == 0) { 2325 continue; 2326 } 2327 if (session == track->sessionId()) { 2328 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2329 chain->incActiveTrackCnt(); 2330 } 2331 } 2332 } 2333 chain->setThread(this); 2334 chain->setInBuffer(buffer, ownsBuffer); 2335 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2336 ? mEffectBuffer : mSinkBuffer)); 2337 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2338 // chains list in order to be processed last as it contains output stage effects 2339 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2340 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2341 // after track specific effects and before output stage 2342 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2343 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2344 // Effect chain for other sessions are inserted at beginning of effect 2345 // chains list to be processed before output mix effects. Relative order between other 2346 // sessions is not important 2347 size_t size = mEffectChains.size(); 2348 size_t i = 0; 2349 for (i = 0; i < size; i++) { 2350 if (mEffectChains[i]->sessionId() < session) { 2351 break; 2352 } 2353 } 2354 mEffectChains.insertAt(chain, i); 2355 checkSuspendOnAddEffectChain_l(chain); 2356 2357 return NO_ERROR; 2358} 2359 2360size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2361{ 2362 int session = chain->sessionId(); 2363 2364 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2365 2366 for (size_t i = 0; i < mEffectChains.size(); i++) { 2367 if (chain == mEffectChains[i]) { 2368 mEffectChains.removeAt(i); 2369 // detach all active tracks from the chain 2370 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2371 sp<Track> track = mActiveTracks[i].promote(); 2372 if (track == 0) { 2373 continue; 2374 } 2375 if (session == track->sessionId()) { 2376 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2377 chain.get(), session); 2378 chain->decActiveTrackCnt(); 2379 } 2380 } 2381 2382 // detach all tracks with same session ID from this chain 2383 for (size_t i = 0; i < mTracks.size(); ++i) { 2384 sp<Track> track = mTracks[i]; 2385 if (session == track->sessionId()) { 2386 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2387 chain->decTrackCnt(); 2388 } 2389 } 2390 break; 2391 } 2392 } 2393 return mEffectChains.size(); 2394} 2395 2396status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2397 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2398{ 2399 Mutex::Autolock _l(mLock); 2400 return attachAuxEffect_l(track, EffectId); 2401} 2402 2403status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2404 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2405{ 2406 status_t status = NO_ERROR; 2407 2408 if (EffectId == 0) { 2409 track->setAuxBuffer(0, NULL); 2410 } else { 2411 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2412 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2413 if (effect != 0) { 2414 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2415 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2416 } else { 2417 status = INVALID_OPERATION; 2418 } 2419 } else { 2420 status = BAD_VALUE; 2421 } 2422 } 2423 return status; 2424} 2425 2426void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2427{ 2428 for (size_t i = 0; i < mTracks.size(); ++i) { 2429 sp<Track> track = mTracks[i]; 2430 if (track->auxEffectId() == effectId) { 2431 attachAuxEffect_l(track, 0); 2432 } 2433 } 2434} 2435 2436bool AudioFlinger::PlaybackThread::threadLoop() 2437{ 2438 Vector< sp<Track> > tracksToRemove; 2439 2440 standbyTime = systemTime(); 2441 2442 // MIXER 2443 nsecs_t lastWarning = 0; 2444 2445 // DUPLICATING 2446 // FIXME could this be made local to while loop? 2447 writeFrames = 0; 2448 2449 int lastGeneration = 0; 2450 2451 cacheParameters_l(); 2452 sleepTime = idleSleepTime; 2453 2454 if (mType == MIXER) { 2455 sleepTimeShift = 0; 2456 } 2457 2458 CpuStats cpuStats; 2459 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2460 2461 acquireWakeLock(); 2462 2463 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2464 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2465 // and then that string will be logged at the next convenient opportunity. 2466 const char *logString = NULL; 2467 2468 checkSilentMode_l(); 2469 2470 while (!exitPending()) 2471 { 2472 cpuStats.sample(myName); 2473 2474 Vector< sp<EffectChain> > effectChains; 2475 2476 { // scope for mLock 2477 2478 Mutex::Autolock _l(mLock); 2479 2480 processConfigEvents_l(); 2481 2482 if (logString != NULL) { 2483 mNBLogWriter->logTimestamp(); 2484 mNBLogWriter->log(logString); 2485 logString = NULL; 2486 } 2487 2488 // Gather the framesReleased counters for all active tracks, 2489 // and latch them atomically with the timestamp. 2490 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2491 mLatchD.mFramesReleased.clear(); 2492 size_t size = mActiveTracks.size(); 2493 for (size_t i = 0; i < size; i++) { 2494 sp<Track> t = mActiveTracks[i].promote(); 2495 if (t != 0) { 2496 mLatchD.mFramesReleased.add(t.get(), 2497 t->mAudioTrackServerProxy->framesReleased()); 2498 } 2499 } 2500 if (mLatchDValid) { 2501 mLatchQ = mLatchD; 2502 mLatchDValid = false; 2503 mLatchQValid = true; 2504 } 2505 2506 saveOutputTracks(); 2507 if (mSignalPending) { 2508 // A signal was raised while we were unlocked 2509 mSignalPending = false; 2510 } else if (waitingAsyncCallback_l()) { 2511 if (exitPending()) { 2512 break; 2513 } 2514 releaseWakeLock_l(); 2515 mWakeLockUids.clear(); 2516 mActiveTracksGeneration++; 2517 ALOGV("wait async completion"); 2518 mWaitWorkCV.wait(mLock); 2519 ALOGV("async completion/wake"); 2520 acquireWakeLock_l(); 2521 standbyTime = systemTime() + standbyDelay; 2522 sleepTime = 0; 2523 2524 continue; 2525 } 2526 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2527 isSuspended()) { 2528 // put audio hardware into standby after short delay 2529 if (shouldStandby_l()) { 2530 2531 threadLoop_standby(); 2532 2533 mStandby = true; 2534 } 2535 2536 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2537 // we're about to wait, flush the binder command buffer 2538 IPCThreadState::self()->flushCommands(); 2539 2540 clearOutputTracks(); 2541 2542 if (exitPending()) { 2543 break; 2544 } 2545 2546 releaseWakeLock_l(); 2547 mWakeLockUids.clear(); 2548 mActiveTracksGeneration++; 2549 // wait until we have something to do... 2550 ALOGV("%s going to sleep", myName.string()); 2551 mWaitWorkCV.wait(mLock); 2552 ALOGV("%s waking up", myName.string()); 2553 acquireWakeLock_l(); 2554 2555 mMixerStatus = MIXER_IDLE; 2556 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2557 mBytesWritten = 0; 2558 mBytesRemaining = 0; 2559 checkSilentMode_l(); 2560 2561 standbyTime = systemTime() + standbyDelay; 2562 sleepTime = idleSleepTime; 2563 if (mType == MIXER) { 2564 sleepTimeShift = 0; 2565 } 2566 2567 continue; 2568 } 2569 } 2570 // mMixerStatusIgnoringFastTracks is also updated internally 2571 mMixerStatus = prepareTracks_l(&tracksToRemove); 2572 2573 // compare with previously applied list 2574 if (lastGeneration != mActiveTracksGeneration) { 2575 // update wakelock 2576 updateWakeLockUids_l(mWakeLockUids); 2577 lastGeneration = mActiveTracksGeneration; 2578 } 2579 2580 // prevent any changes in effect chain list and in each effect chain 2581 // during mixing and effect process as the audio buffers could be deleted 2582 // or modified if an effect is created or deleted 2583 lockEffectChains_l(effectChains); 2584 } // mLock scope ends 2585 2586 if (mBytesRemaining == 0) { 2587 mCurrentWriteLength = 0; 2588 if (mMixerStatus == MIXER_TRACKS_READY) { 2589 // threadLoop_mix() sets mCurrentWriteLength 2590 threadLoop_mix(); 2591 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2592 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2593 // threadLoop_sleepTime sets sleepTime to 0 if data 2594 // must be written to HAL 2595 threadLoop_sleepTime(); 2596 if (sleepTime == 0) { 2597 mCurrentWriteLength = mSinkBufferSize; 2598 } 2599 } 2600 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2601 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2602 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2603 // or mSinkBuffer (if there are no effects). 2604 // 2605 // This is done pre-effects computation; if effects change to 2606 // support higher precision, this needs to move. 2607 // 2608 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2609 // TODO use sleepTime == 0 as an additional condition. 2610 if (mMixerBufferValid) { 2611 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2612 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2613 2614 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2615 mNormalFrameCount * mChannelCount); 2616 } 2617 2618 mBytesRemaining = mCurrentWriteLength; 2619 if (isSuspended()) { 2620 sleepTime = suspendSleepTimeUs(); 2621 // simulate write to HAL when suspended 2622 mBytesWritten += mSinkBufferSize; 2623 mBytesRemaining = 0; 2624 } 2625 2626 // only process effects if we're going to write 2627 if (sleepTime == 0 && mType != OFFLOAD) { 2628 for (size_t i = 0; i < effectChains.size(); i ++) { 2629 effectChains[i]->process_l(); 2630 } 2631 } 2632 } 2633 // Process effect chains for offloaded thread even if no audio 2634 // was read from audio track: process only updates effect state 2635 // and thus does have to be synchronized with audio writes but may have 2636 // to be called while waiting for async write callback 2637 if (mType == OFFLOAD) { 2638 for (size_t i = 0; i < effectChains.size(); i ++) { 2639 effectChains[i]->process_l(); 2640 } 2641 } 2642 2643 // Only if the Effects buffer is enabled and there is data in the 2644 // Effects buffer (buffer valid), we need to 2645 // copy into the sink buffer. 2646 // TODO use sleepTime == 0 as an additional condition. 2647 if (mEffectBufferValid) { 2648 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2649 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2650 mNormalFrameCount * mChannelCount); 2651 } 2652 2653 // enable changes in effect chain 2654 unlockEffectChains(effectChains); 2655 2656 if (!waitingAsyncCallback()) { 2657 // sleepTime == 0 means we must write to audio hardware 2658 if (sleepTime == 0) { 2659 if (mBytesRemaining) { 2660 ssize_t ret = threadLoop_write(); 2661 if (ret < 0) { 2662 mBytesRemaining = 0; 2663 } else { 2664 mBytesWritten += ret; 2665 mBytesRemaining -= ret; 2666 } 2667 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2668 (mMixerStatus == MIXER_DRAIN_ALL)) { 2669 threadLoop_drain(); 2670 } 2671 if (mType == MIXER) { 2672 // write blocked detection 2673 nsecs_t now = systemTime(); 2674 nsecs_t delta = now - mLastWriteTime; 2675 if (!mStandby && delta > maxPeriod) { 2676 mNumDelayedWrites++; 2677 if ((now - lastWarning) > kWarningThrottleNs) { 2678 ATRACE_NAME("underrun"); 2679 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2680 ns2ms(delta), mNumDelayedWrites, this); 2681 lastWarning = now; 2682 } 2683 } 2684 } 2685 2686 } else { 2687 ATRACE_BEGIN("sleep"); 2688 usleep(sleepTime); 2689 ATRACE_END(); 2690 } 2691 } 2692 2693 // Finally let go of removed track(s), without the lock held 2694 // since we can't guarantee the destructors won't acquire that 2695 // same lock. This will also mutate and push a new fast mixer state. 2696 threadLoop_removeTracks(tracksToRemove); 2697 tracksToRemove.clear(); 2698 2699 // FIXME I don't understand the need for this here; 2700 // it was in the original code but maybe the 2701 // assignment in saveOutputTracks() makes this unnecessary? 2702 clearOutputTracks(); 2703 2704 // Effect chains will be actually deleted here if they were removed from 2705 // mEffectChains list during mixing or effects processing 2706 effectChains.clear(); 2707 2708 // FIXME Note that the above .clear() is no longer necessary since effectChains 2709 // is now local to this block, but will keep it for now (at least until merge done). 2710 } 2711 2712 threadLoop_exit(); 2713 2714 if (!mStandby) { 2715 threadLoop_standby(); 2716 mStandby = true; 2717 } 2718 2719 releaseWakeLock(); 2720 mWakeLockUids.clear(); 2721 mActiveTracksGeneration++; 2722 2723 ALOGV("Thread %p type %d exiting", this, mType); 2724 return false; 2725} 2726 2727// removeTracks_l() must be called with ThreadBase::mLock held 2728void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2729{ 2730 size_t count = tracksToRemove.size(); 2731 if (count > 0) { 2732 for (size_t i=0 ; i<count ; i++) { 2733 const sp<Track>& track = tracksToRemove.itemAt(i); 2734 mActiveTracks.remove(track); 2735 mWakeLockUids.remove(track->uid()); 2736 mActiveTracksGeneration++; 2737 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2738 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2739 if (chain != 0) { 2740 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2741 track->sessionId()); 2742 chain->decActiveTrackCnt(); 2743 } 2744 if (track->isTerminated()) { 2745 removeTrack_l(track); 2746 } 2747 } 2748 } 2749 2750} 2751 2752status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2753{ 2754 if (mNormalSink != 0) { 2755 return mNormalSink->getTimestamp(timestamp); 2756 } 2757 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2758 uint64_t position64; 2759 int ret = mOutput->stream->get_presentation_position( 2760 mOutput->stream, &position64, ×tamp.mTime); 2761 if (ret == 0) { 2762 timestamp.mPosition = (uint32_t)position64; 2763 return NO_ERROR; 2764 } 2765 } 2766 return INVALID_OPERATION; 2767} 2768 2769status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2770 audio_patch_handle_t *handle) 2771{ 2772 status_t status = NO_ERROR; 2773 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2774 // store new device and send to effects 2775 audio_devices_t type = AUDIO_DEVICE_NONE; 2776 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2777 type |= patch->sinks[i].ext.device.type; 2778 } 2779 mOutDevice = type; 2780 for (size_t i = 0; i < mEffectChains.size(); i++) { 2781 mEffectChains[i]->setDevice_l(mOutDevice); 2782 } 2783 2784 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2785 status = hwDevice->create_audio_patch(hwDevice, 2786 patch->num_sources, 2787 patch->sources, 2788 patch->num_sinks, 2789 patch->sinks, 2790 handle); 2791 } else { 2792 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2793 } 2794 return status; 2795} 2796 2797status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2798{ 2799 status_t status = NO_ERROR; 2800 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2801 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2802 status = hwDevice->release_audio_patch(hwDevice, handle); 2803 } else { 2804 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2805 } 2806 return status; 2807} 2808 2809void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2810{ 2811 Mutex::Autolock _l(mLock); 2812 mTracks.add(track); 2813} 2814 2815void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2816{ 2817 Mutex::Autolock _l(mLock); 2818 destroyTrack_l(track); 2819} 2820 2821void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2822{ 2823 ThreadBase::getAudioPortConfig(config); 2824 config->role = AUDIO_PORT_ROLE_SOURCE; 2825 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2826 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2827} 2828 2829// ---------------------------------------------------------------------------- 2830 2831AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2832 audio_io_handle_t id, audio_devices_t device, type_t type) 2833 : PlaybackThread(audioFlinger, output, id, device, type), 2834 // mAudioMixer below 2835 // mFastMixer below 2836 mFastMixerFutex(0) 2837 // mOutputSink below 2838 // mPipeSink below 2839 // mNormalSink below 2840{ 2841 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2842 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2843 "mFrameCount=%d, mNormalFrameCount=%d", 2844 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2845 mNormalFrameCount); 2846 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2847 2848 // create an NBAIO sink for the HAL output stream, and negotiate 2849 mOutputSink = new AudioStreamOutSink(output->stream); 2850 size_t numCounterOffers = 0; 2851 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2852 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2853 ALOG_ASSERT(index == 0); 2854 2855 // initialize fast mixer depending on configuration 2856 bool initFastMixer; 2857 switch (kUseFastMixer) { 2858 case FastMixer_Never: 2859 initFastMixer = false; 2860 break; 2861 case FastMixer_Always: 2862 initFastMixer = true; 2863 break; 2864 case FastMixer_Static: 2865 case FastMixer_Dynamic: 2866 initFastMixer = mFrameCount < mNormalFrameCount; 2867 break; 2868 } 2869 if (initFastMixer) { 2870 audio_format_t fastMixerFormat; 2871 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2872 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2873 } else { 2874 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2875 } 2876 if (mFormat != fastMixerFormat) { 2877 // change our Sink format to accept our intermediate precision 2878 mFormat = fastMixerFormat; 2879 free(mSinkBuffer); 2880 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2881 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2882 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2883 } 2884 2885 // create a MonoPipe to connect our submix to FastMixer 2886 NBAIO_Format format = mOutputSink->format(); 2887 NBAIO_Format origformat = format; 2888 // adjust format to match that of the Fast Mixer 2889 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 2890 format.mFormat = fastMixerFormat; 2891 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2892 2893 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2894 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2895 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2896 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2897 const NBAIO_Format offers[1] = {format}; 2898 size_t numCounterOffers = 0; 2899 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2900 ALOG_ASSERT(index == 0); 2901 monoPipe->setAvgFrames((mScreenState & 1) ? 2902 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2903 mPipeSink = monoPipe; 2904 2905#ifdef TEE_SINK 2906 if (mTeeSinkOutputEnabled) { 2907 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2908 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 2909 const NBAIO_Format offers2[1] = {origformat}; 2910 numCounterOffers = 0; 2911 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 2912 ALOG_ASSERT(index == 0); 2913 mTeeSink = teeSink; 2914 PipeReader *teeSource = new PipeReader(*teeSink); 2915 numCounterOffers = 0; 2916 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 2917 ALOG_ASSERT(index == 0); 2918 mTeeSource = teeSource; 2919 } 2920#endif 2921 2922 // create fast mixer and configure it initially with just one fast track for our submix 2923 mFastMixer = new FastMixer(); 2924 FastMixerStateQueue *sq = mFastMixer->sq(); 2925#ifdef STATE_QUEUE_DUMP 2926 sq->setObserverDump(&mStateQueueObserverDump); 2927 sq->setMutatorDump(&mStateQueueMutatorDump); 2928#endif 2929 FastMixerState *state = sq->begin(); 2930 FastTrack *fastTrack = &state->mFastTracks[0]; 2931 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2932 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2933 fastTrack->mVolumeProvider = NULL; 2934 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2935 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2936 fastTrack->mGeneration++; 2937 state->mFastTracksGen++; 2938 state->mTrackMask = 1; 2939 // fast mixer will use the HAL output sink 2940 state->mOutputSink = mOutputSink.get(); 2941 state->mOutputSinkGen++; 2942 state->mFrameCount = mFrameCount; 2943 state->mCommand = FastMixerState::COLD_IDLE; 2944 // already done in constructor initialization list 2945 //mFastMixerFutex = 0; 2946 state->mColdFutexAddr = &mFastMixerFutex; 2947 state->mColdGen++; 2948 state->mDumpState = &mFastMixerDumpState; 2949#ifdef TEE_SINK 2950 state->mTeeSink = mTeeSink.get(); 2951#endif 2952 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2953 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2954 sq->end(); 2955 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2956 2957 // start the fast mixer 2958 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2959 pid_t tid = mFastMixer->getTid(); 2960 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2961 if (err != 0) { 2962 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2963 kPriorityFastMixer, getpid_cached, tid, err); 2964 } 2965 2966#ifdef AUDIO_WATCHDOG 2967 // create and start the watchdog 2968 mAudioWatchdog = new AudioWatchdog(); 2969 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2970 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2971 tid = mAudioWatchdog->getTid(); 2972 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2973 if (err != 0) { 2974 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2975 kPriorityFastMixer, getpid_cached, tid, err); 2976 } 2977#endif 2978 2979 } 2980 2981 switch (kUseFastMixer) { 2982 case FastMixer_Never: 2983 case FastMixer_Dynamic: 2984 mNormalSink = mOutputSink; 2985 break; 2986 case FastMixer_Always: 2987 mNormalSink = mPipeSink; 2988 break; 2989 case FastMixer_Static: 2990 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2991 break; 2992 } 2993} 2994 2995AudioFlinger::MixerThread::~MixerThread() 2996{ 2997 if (mFastMixer != 0) { 2998 FastMixerStateQueue *sq = mFastMixer->sq(); 2999 FastMixerState *state = sq->begin(); 3000 if (state->mCommand == FastMixerState::COLD_IDLE) { 3001 int32_t old = android_atomic_inc(&mFastMixerFutex); 3002 if (old == -1) { 3003 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3004 } 3005 } 3006 state->mCommand = FastMixerState::EXIT; 3007 sq->end(); 3008 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3009 mFastMixer->join(); 3010 // Though the fast mixer thread has exited, it's state queue is still valid. 3011 // We'll use that extract the final state which contains one remaining fast track 3012 // corresponding to our sub-mix. 3013 state = sq->begin(); 3014 ALOG_ASSERT(state->mTrackMask == 1); 3015 FastTrack *fastTrack = &state->mFastTracks[0]; 3016 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3017 delete fastTrack->mBufferProvider; 3018 sq->end(false /*didModify*/); 3019 mFastMixer.clear(); 3020#ifdef AUDIO_WATCHDOG 3021 if (mAudioWatchdog != 0) { 3022 mAudioWatchdog->requestExit(); 3023 mAudioWatchdog->requestExitAndWait(); 3024 mAudioWatchdog.clear(); 3025 } 3026#endif 3027 } 3028 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3029 delete mAudioMixer; 3030} 3031 3032 3033uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3034{ 3035 if (mFastMixer != 0) { 3036 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3037 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3038 } 3039 return latency; 3040} 3041 3042 3043void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3044{ 3045 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3046} 3047 3048ssize_t AudioFlinger::MixerThread::threadLoop_write() 3049{ 3050 // FIXME we should only do one push per cycle; confirm this is true 3051 // Start the fast mixer if it's not already running 3052 if (mFastMixer != 0) { 3053 FastMixerStateQueue *sq = mFastMixer->sq(); 3054 FastMixerState *state = sq->begin(); 3055 if (state->mCommand != FastMixerState::MIX_WRITE && 3056 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3057 if (state->mCommand == FastMixerState::COLD_IDLE) { 3058 int32_t old = android_atomic_inc(&mFastMixerFutex); 3059 if (old == -1) { 3060 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3061 } 3062#ifdef AUDIO_WATCHDOG 3063 if (mAudioWatchdog != 0) { 3064 mAudioWatchdog->resume(); 3065 } 3066#endif 3067 } 3068 state->mCommand = FastMixerState::MIX_WRITE; 3069 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3070 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 3071 sq->end(); 3072 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3073 if (kUseFastMixer == FastMixer_Dynamic) { 3074 mNormalSink = mPipeSink; 3075 } 3076 } else { 3077 sq->end(false /*didModify*/); 3078 } 3079 } 3080 return PlaybackThread::threadLoop_write(); 3081} 3082 3083void AudioFlinger::MixerThread::threadLoop_standby() 3084{ 3085 // Idle the fast mixer if it's currently running 3086 if (mFastMixer != 0) { 3087 FastMixerStateQueue *sq = mFastMixer->sq(); 3088 FastMixerState *state = sq->begin(); 3089 if (!(state->mCommand & FastMixerState::IDLE)) { 3090 state->mCommand = FastMixerState::COLD_IDLE; 3091 state->mColdFutexAddr = &mFastMixerFutex; 3092 state->mColdGen++; 3093 mFastMixerFutex = 0; 3094 sq->end(); 3095 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3096 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3097 if (kUseFastMixer == FastMixer_Dynamic) { 3098 mNormalSink = mOutputSink; 3099 } 3100#ifdef AUDIO_WATCHDOG 3101 if (mAudioWatchdog != 0) { 3102 mAudioWatchdog->pause(); 3103 } 3104#endif 3105 } else { 3106 sq->end(false /*didModify*/); 3107 } 3108 } 3109 PlaybackThread::threadLoop_standby(); 3110} 3111 3112bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3113{ 3114 return false; 3115} 3116 3117bool AudioFlinger::PlaybackThread::shouldStandby_l() 3118{ 3119 return !mStandby; 3120} 3121 3122bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3123{ 3124 Mutex::Autolock _l(mLock); 3125 return waitingAsyncCallback_l(); 3126} 3127 3128// shared by MIXER and DIRECT, overridden by DUPLICATING 3129void AudioFlinger::PlaybackThread::threadLoop_standby() 3130{ 3131 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3132 mOutput->stream->common.standby(&mOutput->stream->common); 3133 if (mUseAsyncWrite != 0) { 3134 // discard any pending drain or write ack by incrementing sequence 3135 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3136 mDrainSequence = (mDrainSequence + 2) & ~1; 3137 ALOG_ASSERT(mCallbackThread != 0); 3138 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3139 mCallbackThread->setDraining(mDrainSequence); 3140 } 3141} 3142 3143void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3144{ 3145 ALOGV("signal playback thread"); 3146 broadcast_l(); 3147} 3148 3149void AudioFlinger::MixerThread::threadLoop_mix() 3150{ 3151 // obtain the presentation timestamp of the next output buffer 3152 int64_t pts; 3153 status_t status = INVALID_OPERATION; 3154 3155 if (mNormalSink != 0) { 3156 status = mNormalSink->getNextWriteTimestamp(&pts); 3157 } else { 3158 status = mOutputSink->getNextWriteTimestamp(&pts); 3159 } 3160 3161 if (status != NO_ERROR) { 3162 pts = AudioBufferProvider::kInvalidPTS; 3163 } 3164 3165 // mix buffers... 3166 mAudioMixer->process(pts); 3167 mCurrentWriteLength = mSinkBufferSize; 3168 // increase sleep time progressively when application underrun condition clears. 3169 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3170 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3171 // such that we would underrun the audio HAL. 3172 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3173 sleepTimeShift--; 3174 } 3175 sleepTime = 0; 3176 standbyTime = systemTime() + standbyDelay; 3177 //TODO: delay standby when effects have a tail 3178 3179} 3180 3181void AudioFlinger::MixerThread::threadLoop_sleepTime() 3182{ 3183 // If no tracks are ready, sleep once for the duration of an output 3184 // buffer size, then write 0s to the output 3185 if (sleepTime == 0) { 3186 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3187 sleepTime = activeSleepTime >> sleepTimeShift; 3188 if (sleepTime < kMinThreadSleepTimeUs) { 3189 sleepTime = kMinThreadSleepTimeUs; 3190 } 3191 // reduce sleep time in case of consecutive application underruns to avoid 3192 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3193 // duration we would end up writing less data than needed by the audio HAL if 3194 // the condition persists. 3195 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3196 sleepTimeShift++; 3197 } 3198 } else { 3199 sleepTime = idleSleepTime; 3200 } 3201 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3202 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3203 // before effects processing or output. 3204 if (mMixerBufferValid) { 3205 memset(mMixerBuffer, 0, mMixerBufferSize); 3206 } else { 3207 memset(mSinkBuffer, 0, mSinkBufferSize); 3208 } 3209 sleepTime = 0; 3210 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3211 "anticipated start"); 3212 } 3213 // TODO add standby time extension fct of effect tail 3214} 3215 3216// prepareTracks_l() must be called with ThreadBase::mLock held 3217AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3218 Vector< sp<Track> > *tracksToRemove) 3219{ 3220 3221 mixer_state mixerStatus = MIXER_IDLE; 3222 // find out which tracks need to be processed 3223 size_t count = mActiveTracks.size(); 3224 size_t mixedTracks = 0; 3225 size_t tracksWithEffect = 0; 3226 // counts only _active_ fast tracks 3227 size_t fastTracks = 0; 3228 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3229 3230 float masterVolume = mMasterVolume; 3231 bool masterMute = mMasterMute; 3232 3233 if (masterMute) { 3234 masterVolume = 0; 3235 } 3236 // Delegate master volume control to effect in output mix effect chain if needed 3237 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3238 if (chain != 0) { 3239 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3240 chain->setVolume_l(&v, &v); 3241 masterVolume = (float)((v + (1 << 23)) >> 24); 3242 chain.clear(); 3243 } 3244 3245 // prepare a new state to push 3246 FastMixerStateQueue *sq = NULL; 3247 FastMixerState *state = NULL; 3248 bool didModify = false; 3249 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3250 if (mFastMixer != 0) { 3251 sq = mFastMixer->sq(); 3252 state = sq->begin(); 3253 } 3254 3255 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3256 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3257 3258 for (size_t i=0 ; i<count ; i++) { 3259 const sp<Track> t = mActiveTracks[i].promote(); 3260 if (t == 0) { 3261 continue; 3262 } 3263 3264 // this const just means the local variable doesn't change 3265 Track* const track = t.get(); 3266 3267 // process fast tracks 3268 if (track->isFastTrack()) { 3269 3270 // It's theoretically possible (though unlikely) for a fast track to be created 3271 // and then removed within the same normal mix cycle. This is not a problem, as 3272 // the track never becomes active so it's fast mixer slot is never touched. 3273 // The converse, of removing an (active) track and then creating a new track 3274 // at the identical fast mixer slot within the same normal mix cycle, 3275 // is impossible because the slot isn't marked available until the end of each cycle. 3276 int j = track->mFastIndex; 3277 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3278 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3279 FastTrack *fastTrack = &state->mFastTracks[j]; 3280 3281 // Determine whether the track is currently in underrun condition, 3282 // and whether it had a recent underrun. 3283 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3284 FastTrackUnderruns underruns = ftDump->mUnderruns; 3285 uint32_t recentFull = (underruns.mBitFields.mFull - 3286 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3287 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3288 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3289 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3290 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3291 uint32_t recentUnderruns = recentPartial + recentEmpty; 3292 track->mObservedUnderruns = underruns; 3293 // don't count underruns that occur while stopping or pausing 3294 // or stopped which can occur when flush() is called while active 3295 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3296 recentUnderruns > 0) { 3297 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3298 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3299 } 3300 3301 // This is similar to the state machine for normal tracks, 3302 // with a few modifications for fast tracks. 3303 bool isActive = true; 3304 switch (track->mState) { 3305 case TrackBase::STOPPING_1: 3306 // track stays active in STOPPING_1 state until first underrun 3307 if (recentUnderruns > 0 || track->isTerminated()) { 3308 track->mState = TrackBase::STOPPING_2; 3309 } 3310 break; 3311 case TrackBase::PAUSING: 3312 // ramp down is not yet implemented 3313 track->setPaused(); 3314 break; 3315 case TrackBase::RESUMING: 3316 // ramp up is not yet implemented 3317 track->mState = TrackBase::ACTIVE; 3318 break; 3319 case TrackBase::ACTIVE: 3320 if (recentFull > 0 || recentPartial > 0) { 3321 // track has provided at least some frames recently: reset retry count 3322 track->mRetryCount = kMaxTrackRetries; 3323 } 3324 if (recentUnderruns == 0) { 3325 // no recent underruns: stay active 3326 break; 3327 } 3328 // there has recently been an underrun of some kind 3329 if (track->sharedBuffer() == 0) { 3330 // were any of the recent underruns "empty" (no frames available)? 3331 if (recentEmpty == 0) { 3332 // no, then ignore the partial underruns as they are allowed indefinitely 3333 break; 3334 } 3335 // there has recently been an "empty" underrun: decrement the retry counter 3336 if (--(track->mRetryCount) > 0) { 3337 break; 3338 } 3339 // indicate to client process that the track was disabled because of underrun; 3340 // it will then automatically call start() when data is available 3341 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3342 // remove from active list, but state remains ACTIVE [confusing but true] 3343 isActive = false; 3344 break; 3345 } 3346 // fall through 3347 case TrackBase::STOPPING_2: 3348 case TrackBase::PAUSED: 3349 case TrackBase::STOPPED: 3350 case TrackBase::FLUSHED: // flush() while active 3351 // Check for presentation complete if track is inactive 3352 // We have consumed all the buffers of this track. 3353 // This would be incomplete if we auto-paused on underrun 3354 { 3355 size_t audioHALFrames = 3356 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3357 size_t framesWritten = mBytesWritten / mFrameSize; 3358 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3359 // track stays in active list until presentation is complete 3360 break; 3361 } 3362 } 3363 if (track->isStopping_2()) { 3364 track->mState = TrackBase::STOPPED; 3365 } 3366 if (track->isStopped()) { 3367 // Can't reset directly, as fast mixer is still polling this track 3368 // track->reset(); 3369 // So instead mark this track as needing to be reset after push with ack 3370 resetMask |= 1 << i; 3371 } 3372 isActive = false; 3373 break; 3374 case TrackBase::IDLE: 3375 default: 3376 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3377 } 3378 3379 if (isActive) { 3380 // was it previously inactive? 3381 if (!(state->mTrackMask & (1 << j))) { 3382 ExtendedAudioBufferProvider *eabp = track; 3383 VolumeProvider *vp = track; 3384 fastTrack->mBufferProvider = eabp; 3385 fastTrack->mVolumeProvider = vp; 3386 fastTrack->mChannelMask = track->mChannelMask; 3387 fastTrack->mFormat = track->mFormat; 3388 fastTrack->mGeneration++; 3389 state->mTrackMask |= 1 << j; 3390 didModify = true; 3391 // no acknowledgement required for newly active tracks 3392 } 3393 // cache the combined master volume and stream type volume for fast mixer; this 3394 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3395 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3396 ++fastTracks; 3397 } else { 3398 // was it previously active? 3399 if (state->mTrackMask & (1 << j)) { 3400 fastTrack->mBufferProvider = NULL; 3401 fastTrack->mGeneration++; 3402 state->mTrackMask &= ~(1 << j); 3403 didModify = true; 3404 // If any fast tracks were removed, we must wait for acknowledgement 3405 // because we're about to decrement the last sp<> on those tracks. 3406 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3407 } else { 3408 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3409 } 3410 tracksToRemove->add(track); 3411 // Avoids a misleading display in dumpsys 3412 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3413 } 3414 continue; 3415 } 3416 3417 { // local variable scope to avoid goto warning 3418 3419 audio_track_cblk_t* cblk = track->cblk(); 3420 3421 // The first time a track is added we wait 3422 // for all its buffers to be filled before processing it 3423 int name = track->name(); 3424 // make sure that we have enough frames to mix one full buffer. 3425 // enforce this condition only once to enable draining the buffer in case the client 3426 // app does not call stop() and relies on underrun to stop: 3427 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3428 // during last round 3429 size_t desiredFrames; 3430 uint32_t sr = track->sampleRate(); 3431 if (sr == mSampleRate) { 3432 desiredFrames = mNormalFrameCount; 3433 } else { 3434 // +1 for rounding and +1 for additional sample needed for interpolation 3435 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3436 // add frames already consumed but not yet released by the resampler 3437 // because mAudioTrackServerProxy->framesReady() will include these frames 3438 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3439#if 0 3440 // the minimum track buffer size is normally twice the number of frames necessary 3441 // to fill one buffer and the resampler should not leave more than one buffer worth 3442 // of unreleased frames after each pass, but just in case... 3443 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3444#endif 3445 } 3446 uint32_t minFrames = 1; 3447 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3448 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3449 minFrames = desiredFrames; 3450 } 3451 3452 size_t framesReady = track->framesReady(); 3453 if (ATRACE_ENABLED()) { 3454 // I wish we had formatted trace names 3455 char traceName[16]; 3456 strcpy(traceName, "nRdy"); 3457 int name = track->name(); 3458 if (AudioMixer::TRACK0 <= name && 3459 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3460 name -= AudioMixer::TRACK0; 3461 traceName[4] = (name / 10) + '0'; 3462 traceName[5] = (name % 10) + '0'; 3463 } else { 3464 traceName[4] = '?'; 3465 traceName[5] = '?'; 3466 } 3467 traceName[6] = '\0'; 3468 ATRACE_INT(traceName, framesReady); 3469 } 3470 if ((framesReady >= minFrames) && track->isReady() && 3471 !track->isPaused() && !track->isTerminated()) 3472 { 3473 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3474 3475 mixedTracks++; 3476 3477 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3478 // there is an effect chain connected to the track 3479 chain.clear(); 3480 if (track->mainBuffer() != mSinkBuffer && 3481 track->mainBuffer() != mMixerBuffer) { 3482 if (mEffectBufferEnabled) { 3483 mEffectBufferValid = true; // Later can set directly. 3484 } 3485 chain = getEffectChain_l(track->sessionId()); 3486 // Delegate volume control to effect in track effect chain if needed 3487 if (chain != 0) { 3488 tracksWithEffect++; 3489 } else { 3490 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3491 "session %d", 3492 name, track->sessionId()); 3493 } 3494 } 3495 3496 3497 int param = AudioMixer::VOLUME; 3498 if (track->mFillingUpStatus == Track::FS_FILLED) { 3499 // no ramp for the first volume setting 3500 track->mFillingUpStatus = Track::FS_ACTIVE; 3501 if (track->mState == TrackBase::RESUMING) { 3502 track->mState = TrackBase::ACTIVE; 3503 param = AudioMixer::RAMP_VOLUME; 3504 } 3505 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3506 // FIXME should not make a decision based on mServer 3507 } else if (cblk->mServer != 0) { 3508 // If the track is stopped before the first frame was mixed, 3509 // do not apply ramp 3510 param = AudioMixer::RAMP_VOLUME; 3511 } 3512 3513 // compute volume for this track 3514 uint32_t vl, vr; // in U8.24 integer format 3515 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3516 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3517 vl = vr = 0; 3518 vlf = vrf = vaf = 0.; 3519 if (track->isPausing()) { 3520 track->setPaused(); 3521 } 3522 } else { 3523 3524 // read original volumes with volume control 3525 float typeVolume = mStreamTypes[track->streamType()].volume; 3526 float v = masterVolume * typeVolume; 3527 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3528 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3529 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3530 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3531 // track volumes come from shared memory, so can't be trusted and must be clamped 3532 if (vlf > GAIN_FLOAT_UNITY) { 3533 ALOGV("Track left volume out of range: %.3g", vlf); 3534 vlf = GAIN_FLOAT_UNITY; 3535 } 3536 if (vrf > GAIN_FLOAT_UNITY) { 3537 ALOGV("Track right volume out of range: %.3g", vrf); 3538 vrf = GAIN_FLOAT_UNITY; 3539 } 3540 // now apply the master volume and stream type volume 3541 vlf *= v; 3542 vrf *= v; 3543 // assuming master volume and stream type volume each go up to 1.0, 3544 // then derive vl and vr as U8.24 versions for the effect chain 3545 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3546 vl = (uint32_t) (scaleto8_24 * vlf); 3547 vr = (uint32_t) (scaleto8_24 * vrf); 3548 // vl and vr are now in U8.24 format 3549 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3550 // send level comes from shared memory and so may be corrupt 3551 if (sendLevel > MAX_GAIN_INT) { 3552 ALOGV("Track send level out of range: %04X", sendLevel); 3553 sendLevel = MAX_GAIN_INT; 3554 } 3555 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3556 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3557 } 3558 3559 // Delegate volume control to effect in track effect chain if needed 3560 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3561 // Do not ramp volume if volume is controlled by effect 3562 param = AudioMixer::VOLUME; 3563 // Update remaining floating point volume levels 3564 vlf = (float)vl / (1 << 24); 3565 vrf = (float)vr / (1 << 24); 3566 track->mHasVolumeController = true; 3567 } else { 3568 // force no volume ramp when volume controller was just disabled or removed 3569 // from effect chain to avoid volume spike 3570 if (track->mHasVolumeController) { 3571 param = AudioMixer::VOLUME; 3572 } 3573 track->mHasVolumeController = false; 3574 } 3575 3576 // XXX: these things DON'T need to be done each time 3577 mAudioMixer->setBufferProvider(name, track); 3578 mAudioMixer->enable(name); 3579 3580 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3581 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3582 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3583 mAudioMixer->setParameter( 3584 name, 3585 AudioMixer::TRACK, 3586 AudioMixer::FORMAT, (void *)track->format()); 3587 mAudioMixer->setParameter( 3588 name, 3589 AudioMixer::TRACK, 3590 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3591 mAudioMixer->setParameter( 3592 name, 3593 AudioMixer::TRACK, 3594 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3595 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3596 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3597 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3598 if (reqSampleRate == 0) { 3599 reqSampleRate = mSampleRate; 3600 } else if (reqSampleRate > maxSampleRate) { 3601 reqSampleRate = maxSampleRate; 3602 } 3603 mAudioMixer->setParameter( 3604 name, 3605 AudioMixer::RESAMPLE, 3606 AudioMixer::SAMPLE_RATE, 3607 (void *)(uintptr_t)reqSampleRate); 3608 /* 3609 * Select the appropriate output buffer for the track. 3610 * 3611 * Tracks with effects go into their own effects chain buffer 3612 * and from there into either mEffectBuffer or mSinkBuffer. 3613 * 3614 * Other tracks can use mMixerBuffer for higher precision 3615 * channel accumulation. If this buffer is enabled 3616 * (mMixerBufferEnabled true), then selected tracks will accumulate 3617 * into it. 3618 * 3619 */ 3620 if (mMixerBufferEnabled 3621 && (track->mainBuffer() == mSinkBuffer 3622 || track->mainBuffer() == mMixerBuffer)) { 3623 mAudioMixer->setParameter( 3624 name, 3625 AudioMixer::TRACK, 3626 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3627 mAudioMixer->setParameter( 3628 name, 3629 AudioMixer::TRACK, 3630 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3631 // TODO: override track->mainBuffer()? 3632 mMixerBufferValid = true; 3633 } else { 3634 mAudioMixer->setParameter( 3635 name, 3636 AudioMixer::TRACK, 3637 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3638 mAudioMixer->setParameter( 3639 name, 3640 AudioMixer::TRACK, 3641 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3642 } 3643 mAudioMixer->setParameter( 3644 name, 3645 AudioMixer::TRACK, 3646 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3647 3648 // reset retry count 3649 track->mRetryCount = kMaxTrackRetries; 3650 3651 // If one track is ready, set the mixer ready if: 3652 // - the mixer was not ready during previous round OR 3653 // - no other track is not ready 3654 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3655 mixerStatus != MIXER_TRACKS_ENABLED) { 3656 mixerStatus = MIXER_TRACKS_READY; 3657 } 3658 } else { 3659 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3660 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3661 } 3662 // clear effect chain input buffer if an active track underruns to avoid sending 3663 // previous audio buffer again to effects 3664 chain = getEffectChain_l(track->sessionId()); 3665 if (chain != 0) { 3666 chain->clearInputBuffer(); 3667 } 3668 3669 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3670 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3671 track->isStopped() || track->isPaused()) { 3672 // We have consumed all the buffers of this track. 3673 // Remove it from the list of active tracks. 3674 // TODO: use actual buffer filling status instead of latency when available from 3675 // audio HAL 3676 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3677 size_t framesWritten = mBytesWritten / mFrameSize; 3678 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3679 if (track->isStopped()) { 3680 track->reset(); 3681 } 3682 tracksToRemove->add(track); 3683 } 3684 } else { 3685 // No buffers for this track. Give it a few chances to 3686 // fill a buffer, then remove it from active list. 3687 if (--(track->mRetryCount) <= 0) { 3688 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3689 tracksToRemove->add(track); 3690 // indicate to client process that the track was disabled because of underrun; 3691 // it will then automatically call start() when data is available 3692 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3693 // If one track is not ready, mark the mixer also not ready if: 3694 // - the mixer was ready during previous round OR 3695 // - no other track is ready 3696 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3697 mixerStatus != MIXER_TRACKS_READY) { 3698 mixerStatus = MIXER_TRACKS_ENABLED; 3699 } 3700 } 3701 mAudioMixer->disable(name); 3702 } 3703 3704 } // local variable scope to avoid goto warning 3705track_is_ready: ; 3706 3707 } 3708 3709 // Push the new FastMixer state if necessary 3710 bool pauseAudioWatchdog = false; 3711 if (didModify) { 3712 state->mFastTracksGen++; 3713 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3714 if (kUseFastMixer == FastMixer_Dynamic && 3715 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3716 state->mCommand = FastMixerState::COLD_IDLE; 3717 state->mColdFutexAddr = &mFastMixerFutex; 3718 state->mColdGen++; 3719 mFastMixerFutex = 0; 3720 if (kUseFastMixer == FastMixer_Dynamic) { 3721 mNormalSink = mOutputSink; 3722 } 3723 // If we go into cold idle, need to wait for acknowledgement 3724 // so that fast mixer stops doing I/O. 3725 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3726 pauseAudioWatchdog = true; 3727 } 3728 } 3729 if (sq != NULL) { 3730 sq->end(didModify); 3731 sq->push(block); 3732 } 3733#ifdef AUDIO_WATCHDOG 3734 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3735 mAudioWatchdog->pause(); 3736 } 3737#endif 3738 3739 // Now perform the deferred reset on fast tracks that have stopped 3740 while (resetMask != 0) { 3741 size_t i = __builtin_ctz(resetMask); 3742 ALOG_ASSERT(i < count); 3743 resetMask &= ~(1 << i); 3744 sp<Track> t = mActiveTracks[i].promote(); 3745 if (t == 0) { 3746 continue; 3747 } 3748 Track* track = t.get(); 3749 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3750 track->reset(); 3751 } 3752 3753 // remove all the tracks that need to be... 3754 removeTracks_l(*tracksToRemove); 3755 3756 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3757 mEffectBufferValid = true; 3758 } 3759 3760 if (mEffectBufferValid) { 3761 // as long as there are effects we should clear the effects buffer, to avoid 3762 // passing a non-clean buffer to the effect chain 3763 memset(mEffectBuffer, 0, mEffectBufferSize); 3764 } 3765 // sink or mix buffer must be cleared if all tracks are connected to an 3766 // effect chain as in this case the mixer will not write to the sink or mix buffer 3767 // and track effects will accumulate into it 3768 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3769 (mixedTracks == 0 && fastTracks > 0))) { 3770 // FIXME as a performance optimization, should remember previous zero status 3771 if (mMixerBufferValid) { 3772 memset(mMixerBuffer, 0, mMixerBufferSize); 3773 // TODO: In testing, mSinkBuffer below need not be cleared because 3774 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3775 // after mixing. 3776 // 3777 // To enforce this guarantee: 3778 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3779 // (mixedTracks == 0 && fastTracks > 0)) 3780 // must imply MIXER_TRACKS_READY. 3781 // Later, we may clear buffers regardless, and skip much of this logic. 3782 } 3783 // FIXME as a performance optimization, should remember previous zero status 3784 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3785 } 3786 3787 // if any fast tracks, then status is ready 3788 mMixerStatusIgnoringFastTracks = mixerStatus; 3789 if (fastTracks > 0) { 3790 mixerStatus = MIXER_TRACKS_READY; 3791 } 3792 return mixerStatus; 3793} 3794 3795// getTrackName_l() must be called with ThreadBase::mLock held 3796int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3797 audio_format_t format, int sessionId) 3798{ 3799 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3800} 3801 3802// deleteTrackName_l() must be called with ThreadBase::mLock held 3803void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3804{ 3805 ALOGV("remove track (%d) and delete from mixer", name); 3806 mAudioMixer->deleteTrackName(name); 3807} 3808 3809// checkForNewParameter_l() must be called with ThreadBase::mLock held 3810bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3811 status_t& status) 3812{ 3813 bool reconfig = false; 3814 3815 status = NO_ERROR; 3816 3817 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3818 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3819 if (mFastMixer != 0) { 3820 FastMixerStateQueue *sq = mFastMixer->sq(); 3821 FastMixerState *state = sq->begin(); 3822 if (!(state->mCommand & FastMixerState::IDLE)) { 3823 previousCommand = state->mCommand; 3824 state->mCommand = FastMixerState::HOT_IDLE; 3825 sq->end(); 3826 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3827 } else { 3828 sq->end(false /*didModify*/); 3829 } 3830 } 3831 3832 AudioParameter param = AudioParameter(keyValuePair); 3833 int value; 3834 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3835 reconfig = true; 3836 } 3837 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3838 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3839 status = BAD_VALUE; 3840 } else { 3841 // no need to save value, since it's constant 3842 reconfig = true; 3843 } 3844 } 3845 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3846 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3847 status = BAD_VALUE; 3848 } else { 3849 // no need to save value, since it's constant 3850 reconfig = true; 3851 } 3852 } 3853 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3854 // do not accept frame count changes if tracks are open as the track buffer 3855 // size depends on frame count and correct behavior would not be guaranteed 3856 // if frame count is changed after track creation 3857 if (!mTracks.isEmpty()) { 3858 status = INVALID_OPERATION; 3859 } else { 3860 reconfig = true; 3861 } 3862 } 3863 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3864#ifdef ADD_BATTERY_DATA 3865 // when changing the audio output device, call addBatteryData to notify 3866 // the change 3867 if (mOutDevice != value) { 3868 uint32_t params = 0; 3869 // check whether speaker is on 3870 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3871 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3872 } 3873 3874 audio_devices_t deviceWithoutSpeaker 3875 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3876 // check if any other device (except speaker) is on 3877 if (value & deviceWithoutSpeaker ) { 3878 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3879 } 3880 3881 if (params != 0) { 3882 addBatteryData(params); 3883 } 3884 } 3885#endif 3886 3887 // forward device change to effects that have requested to be 3888 // aware of attached audio device. 3889 if (value != AUDIO_DEVICE_NONE) { 3890 mOutDevice = value; 3891 for (size_t i = 0; i < mEffectChains.size(); i++) { 3892 mEffectChains[i]->setDevice_l(mOutDevice); 3893 } 3894 } 3895 } 3896 3897 if (status == NO_ERROR) { 3898 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3899 keyValuePair.string()); 3900 if (!mStandby && status == INVALID_OPERATION) { 3901 mOutput->stream->common.standby(&mOutput->stream->common); 3902 mStandby = true; 3903 mBytesWritten = 0; 3904 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3905 keyValuePair.string()); 3906 } 3907 if (status == NO_ERROR && reconfig) { 3908 readOutputParameters_l(); 3909 delete mAudioMixer; 3910 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3911 for (size_t i = 0; i < mTracks.size() ; i++) { 3912 int name = getTrackName_l(mTracks[i]->mChannelMask, 3913 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3914 if (name < 0) { 3915 break; 3916 } 3917 mTracks[i]->mName = name; 3918 } 3919 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3920 } 3921 } 3922 3923 if (!(previousCommand & FastMixerState::IDLE)) { 3924 ALOG_ASSERT(mFastMixer != 0); 3925 FastMixerStateQueue *sq = mFastMixer->sq(); 3926 FastMixerState *state = sq->begin(); 3927 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3928 state->mCommand = previousCommand; 3929 sq->end(); 3930 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3931 } 3932 3933 return reconfig; 3934} 3935 3936 3937void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3938{ 3939 const size_t SIZE = 256; 3940 char buffer[SIZE]; 3941 String8 result; 3942 3943 PlaybackThread::dumpInternals(fd, args); 3944 3945 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3946 3947 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3948 const FastMixerDumpState copy(mFastMixerDumpState); 3949 copy.dump(fd); 3950 3951#ifdef STATE_QUEUE_DUMP 3952 // Similar for state queue 3953 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3954 observerCopy.dump(fd); 3955 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3956 mutatorCopy.dump(fd); 3957#endif 3958 3959#ifdef TEE_SINK 3960 // Write the tee output to a .wav file 3961 dumpTee(fd, mTeeSource, mId); 3962#endif 3963 3964#ifdef AUDIO_WATCHDOG 3965 if (mAudioWatchdog != 0) { 3966 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3967 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3968 wdCopy.dump(fd); 3969 } 3970#endif 3971} 3972 3973uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3974{ 3975 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3976} 3977 3978uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3979{ 3980 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3981} 3982 3983void AudioFlinger::MixerThread::cacheParameters_l() 3984{ 3985 PlaybackThread::cacheParameters_l(); 3986 3987 // FIXME: Relaxed timing because of a certain device that can't meet latency 3988 // Should be reduced to 2x after the vendor fixes the driver issue 3989 // increase threshold again due to low power audio mode. The way this warning 3990 // threshold is calculated and its usefulness should be reconsidered anyway. 3991 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3992} 3993 3994// ---------------------------------------------------------------------------- 3995 3996AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3997 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3998 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3999 // mLeftVolFloat, mRightVolFloat 4000{ 4001} 4002 4003AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4004 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4005 ThreadBase::type_t type) 4006 : PlaybackThread(audioFlinger, output, id, device, type) 4007 // mLeftVolFloat, mRightVolFloat 4008{ 4009} 4010 4011AudioFlinger::DirectOutputThread::~DirectOutputThread() 4012{ 4013} 4014 4015void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4016{ 4017 audio_track_cblk_t* cblk = track->cblk(); 4018 float left, right; 4019 4020 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4021 left = right = 0; 4022 } else { 4023 float typeVolume = mStreamTypes[track->streamType()].volume; 4024 float v = mMasterVolume * typeVolume; 4025 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4026 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4027 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4028 if (left > GAIN_FLOAT_UNITY) { 4029 left = GAIN_FLOAT_UNITY; 4030 } 4031 left *= v; 4032 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4033 if (right > GAIN_FLOAT_UNITY) { 4034 right = GAIN_FLOAT_UNITY; 4035 } 4036 right *= v; 4037 } 4038 4039 if (lastTrack) { 4040 if (left != mLeftVolFloat || right != mRightVolFloat) { 4041 mLeftVolFloat = left; 4042 mRightVolFloat = right; 4043 4044 // Convert volumes from float to 8.24 4045 uint32_t vl = (uint32_t)(left * (1 << 24)); 4046 uint32_t vr = (uint32_t)(right * (1 << 24)); 4047 4048 // Delegate volume control to effect in track effect chain if needed 4049 // only one effect chain can be present on DirectOutputThread, so if 4050 // there is one, the track is connected to it 4051 if (!mEffectChains.isEmpty()) { 4052 mEffectChains[0]->setVolume_l(&vl, &vr); 4053 left = (float)vl / (1 << 24); 4054 right = (float)vr / (1 << 24); 4055 } 4056 if (mOutput->stream->set_volume) { 4057 mOutput->stream->set_volume(mOutput->stream, left, right); 4058 } 4059 } 4060 } 4061} 4062 4063 4064AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4065 Vector< sp<Track> > *tracksToRemove 4066) 4067{ 4068 size_t count = mActiveTracks.size(); 4069 mixer_state mixerStatus = MIXER_IDLE; 4070 4071 // find out which tracks need to be processed 4072 for (size_t i = 0; i < count; i++) { 4073 sp<Track> t = mActiveTracks[i].promote(); 4074 // The track died recently 4075 if (t == 0) { 4076 continue; 4077 } 4078 4079 Track* const track = t.get(); 4080 audio_track_cblk_t* cblk = track->cblk(); 4081 // Only consider last track started for volume and mixer state control. 4082 // In theory an older track could underrun and restart after the new one starts 4083 // but as we only care about the transition phase between two tracks on a 4084 // direct output, it is not a problem to ignore the underrun case. 4085 sp<Track> l = mLatestActiveTrack.promote(); 4086 bool last = l.get() == track; 4087 4088 // The first time a track is added we wait 4089 // for all its buffers to be filled before processing it 4090 uint32_t minFrames; 4091 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 4092 minFrames = mNormalFrameCount; 4093 } else { 4094 minFrames = 1; 4095 } 4096 4097 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4098 !track->isStopping_2() && !track->isStopped()) 4099 { 4100 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4101 4102 if (track->mFillingUpStatus == Track::FS_FILLED) { 4103 track->mFillingUpStatus = Track::FS_ACTIVE; 4104 // make sure processVolume_l() will apply new volume even if 0 4105 mLeftVolFloat = mRightVolFloat = -1.0; 4106 if (track->mState == TrackBase::RESUMING) { 4107 track->mState = TrackBase::ACTIVE; 4108 } 4109 } 4110 4111 // compute volume for this track 4112 processVolume_l(track, last); 4113 if (last) { 4114 // reset retry count 4115 track->mRetryCount = kMaxTrackRetriesDirect; 4116 mActiveTrack = t; 4117 mixerStatus = MIXER_TRACKS_READY; 4118 } 4119 } else { 4120 // clear effect chain input buffer if the last active track started underruns 4121 // to avoid sending previous audio buffer again to effects 4122 if (!mEffectChains.isEmpty() && last) { 4123 mEffectChains[0]->clearInputBuffer(); 4124 } 4125 if (track->isStopping_1()) { 4126 track->mState = TrackBase::STOPPING_2; 4127 } 4128 if ((track->sharedBuffer() != 0) || track->isStopped() || 4129 track->isStopping_2() || track->isPaused()) { 4130 // We have consumed all the buffers of this track. 4131 // Remove it from the list of active tracks. 4132 size_t audioHALFrames; 4133 if (audio_is_linear_pcm(mFormat)) { 4134 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4135 } else { 4136 audioHALFrames = 0; 4137 } 4138 4139 size_t framesWritten = mBytesWritten / mFrameSize; 4140 if (mStandby || !last || 4141 track->presentationComplete(framesWritten, audioHALFrames)) { 4142 if (track->isStopping_2()) { 4143 track->mState = TrackBase::STOPPED; 4144 } 4145 if (track->isStopped()) { 4146 if (track->mState == TrackBase::FLUSHED) { 4147 flushHw_l(); 4148 } 4149 track->reset(); 4150 } 4151 tracksToRemove->add(track); 4152 } 4153 } else { 4154 // No buffers for this track. Give it a few chances to 4155 // fill a buffer, then remove it from active list. 4156 // Only consider last track started for mixer state control 4157 if (--(track->mRetryCount) <= 0) { 4158 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4159 tracksToRemove->add(track); 4160 // indicate to client process that the track was disabled because of underrun; 4161 // it will then automatically call start() when data is available 4162 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4163 } else if (last) { 4164 mixerStatus = MIXER_TRACKS_ENABLED; 4165 } 4166 } 4167 } 4168 } 4169 4170 // remove all the tracks that need to be... 4171 removeTracks_l(*tracksToRemove); 4172 4173 return mixerStatus; 4174} 4175 4176void AudioFlinger::DirectOutputThread::threadLoop_mix() 4177{ 4178 size_t frameCount = mFrameCount; 4179 int8_t *curBuf = (int8_t *)mSinkBuffer; 4180 // output audio to hardware 4181 while (frameCount) { 4182 AudioBufferProvider::Buffer buffer; 4183 buffer.frameCount = frameCount; 4184 mActiveTrack->getNextBuffer(&buffer); 4185 if (buffer.raw == NULL) { 4186 memset(curBuf, 0, frameCount * mFrameSize); 4187 break; 4188 } 4189 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4190 frameCount -= buffer.frameCount; 4191 curBuf += buffer.frameCount * mFrameSize; 4192 mActiveTrack->releaseBuffer(&buffer); 4193 } 4194 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4195 sleepTime = 0; 4196 standbyTime = systemTime() + standbyDelay; 4197 mActiveTrack.clear(); 4198} 4199 4200void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4201{ 4202 if (sleepTime == 0) { 4203 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4204 sleepTime = activeSleepTime; 4205 } else { 4206 sleepTime = idleSleepTime; 4207 } 4208 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4209 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4210 sleepTime = 0; 4211 } 4212} 4213 4214// getTrackName_l() must be called with ThreadBase::mLock held 4215int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4216 audio_format_t format __unused, int sessionId __unused) 4217{ 4218 return 0; 4219} 4220 4221// deleteTrackName_l() must be called with ThreadBase::mLock held 4222void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4223{ 4224} 4225 4226// checkForNewParameter_l() must be called with ThreadBase::mLock held 4227bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4228 status_t& status) 4229{ 4230 bool reconfig = false; 4231 4232 status = NO_ERROR; 4233 4234 AudioParameter param = AudioParameter(keyValuePair); 4235 int value; 4236 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4237 // forward device change to effects that have requested to be 4238 // aware of attached audio device. 4239 if (value != AUDIO_DEVICE_NONE) { 4240 mOutDevice = value; 4241 for (size_t i = 0; i < mEffectChains.size(); i++) { 4242 mEffectChains[i]->setDevice_l(mOutDevice); 4243 } 4244 } 4245 } 4246 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4247 // do not accept frame count changes if tracks are open as the track buffer 4248 // size depends on frame count and correct behavior would not be garantied 4249 // if frame count is changed after track creation 4250 if (!mTracks.isEmpty()) { 4251 status = INVALID_OPERATION; 4252 } else { 4253 reconfig = true; 4254 } 4255 } 4256 if (status == NO_ERROR) { 4257 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4258 keyValuePair.string()); 4259 if (!mStandby && status == INVALID_OPERATION) { 4260 mOutput->stream->common.standby(&mOutput->stream->common); 4261 mStandby = true; 4262 mBytesWritten = 0; 4263 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4264 keyValuePair.string()); 4265 } 4266 if (status == NO_ERROR && reconfig) { 4267 readOutputParameters_l(); 4268 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4269 } 4270 } 4271 4272 return reconfig; 4273} 4274 4275uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4276{ 4277 uint32_t time; 4278 if (audio_is_linear_pcm(mFormat)) { 4279 time = PlaybackThread::activeSleepTimeUs(); 4280 } else { 4281 time = 10000; 4282 } 4283 return time; 4284} 4285 4286uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4287{ 4288 uint32_t time; 4289 if (audio_is_linear_pcm(mFormat)) { 4290 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4291 } else { 4292 time = 10000; 4293 } 4294 return time; 4295} 4296 4297uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4298{ 4299 uint32_t time; 4300 if (audio_is_linear_pcm(mFormat)) { 4301 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4302 } else { 4303 time = 10000; 4304 } 4305 return time; 4306} 4307 4308void AudioFlinger::DirectOutputThread::cacheParameters_l() 4309{ 4310 PlaybackThread::cacheParameters_l(); 4311 4312 // use shorter standby delay as on normal output to release 4313 // hardware resources as soon as possible 4314 if (audio_is_linear_pcm(mFormat)) { 4315 standbyDelay = microseconds(activeSleepTime*2); 4316 } else { 4317 standbyDelay = kOffloadStandbyDelayNs; 4318 } 4319} 4320 4321void AudioFlinger::DirectOutputThread::flushHw_l() 4322{ 4323 if (mOutput->stream->flush != NULL) 4324 mOutput->stream->flush(mOutput->stream); 4325} 4326 4327// ---------------------------------------------------------------------------- 4328 4329AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4330 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4331 : Thread(false /*canCallJava*/), 4332 mPlaybackThread(playbackThread), 4333 mWriteAckSequence(0), 4334 mDrainSequence(0) 4335{ 4336} 4337 4338AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4339{ 4340} 4341 4342void AudioFlinger::AsyncCallbackThread::onFirstRef() 4343{ 4344 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4345} 4346 4347bool AudioFlinger::AsyncCallbackThread::threadLoop() 4348{ 4349 while (!exitPending()) { 4350 uint32_t writeAckSequence; 4351 uint32_t drainSequence; 4352 4353 { 4354 Mutex::Autolock _l(mLock); 4355 while (!((mWriteAckSequence & 1) || 4356 (mDrainSequence & 1) || 4357 exitPending())) { 4358 mWaitWorkCV.wait(mLock); 4359 } 4360 4361 if (exitPending()) { 4362 break; 4363 } 4364 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4365 mWriteAckSequence, mDrainSequence); 4366 writeAckSequence = mWriteAckSequence; 4367 mWriteAckSequence &= ~1; 4368 drainSequence = mDrainSequence; 4369 mDrainSequence &= ~1; 4370 } 4371 { 4372 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4373 if (playbackThread != 0) { 4374 if (writeAckSequence & 1) { 4375 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4376 } 4377 if (drainSequence & 1) { 4378 playbackThread->resetDraining(drainSequence >> 1); 4379 } 4380 } 4381 } 4382 } 4383 return false; 4384} 4385 4386void AudioFlinger::AsyncCallbackThread::exit() 4387{ 4388 ALOGV("AsyncCallbackThread::exit"); 4389 Mutex::Autolock _l(mLock); 4390 requestExit(); 4391 mWaitWorkCV.broadcast(); 4392} 4393 4394void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4395{ 4396 Mutex::Autolock _l(mLock); 4397 // bit 0 is cleared 4398 mWriteAckSequence = sequence << 1; 4399} 4400 4401void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4402{ 4403 Mutex::Autolock _l(mLock); 4404 // ignore unexpected callbacks 4405 if (mWriteAckSequence & 2) { 4406 mWriteAckSequence |= 1; 4407 mWaitWorkCV.signal(); 4408 } 4409} 4410 4411void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4412{ 4413 Mutex::Autolock _l(mLock); 4414 // bit 0 is cleared 4415 mDrainSequence = sequence << 1; 4416} 4417 4418void AudioFlinger::AsyncCallbackThread::resetDraining() 4419{ 4420 Mutex::Autolock _l(mLock); 4421 // ignore unexpected callbacks 4422 if (mDrainSequence & 2) { 4423 mDrainSequence |= 1; 4424 mWaitWorkCV.signal(); 4425 } 4426} 4427 4428 4429// ---------------------------------------------------------------------------- 4430AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4431 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4432 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4433 mHwPaused(false), 4434 mFlushPending(false), 4435 mPausedBytesRemaining(0) 4436{ 4437 //FIXME: mStandby should be set to true by ThreadBase constructor 4438 mStandby = true; 4439} 4440 4441void AudioFlinger::OffloadThread::threadLoop_exit() 4442{ 4443 if (mFlushPending || mHwPaused) { 4444 // If a flush is pending or track was paused, just discard buffered data 4445 flushHw_l(); 4446 } else { 4447 mMixerStatus = MIXER_DRAIN_ALL; 4448 threadLoop_drain(); 4449 } 4450 if (mUseAsyncWrite) { 4451 ALOG_ASSERT(mCallbackThread != 0); 4452 mCallbackThread->exit(); 4453 } 4454 PlaybackThread::threadLoop_exit(); 4455} 4456 4457AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4458 Vector< sp<Track> > *tracksToRemove 4459) 4460{ 4461 size_t count = mActiveTracks.size(); 4462 4463 mixer_state mixerStatus = MIXER_IDLE; 4464 bool doHwPause = false; 4465 bool doHwResume = false; 4466 4467 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4468 4469 // find out which tracks need to be processed 4470 for (size_t i = 0; i < count; i++) { 4471 sp<Track> t = mActiveTracks[i].promote(); 4472 // The track died recently 4473 if (t == 0) { 4474 continue; 4475 } 4476 Track* const track = t.get(); 4477 audio_track_cblk_t* cblk = track->cblk(); 4478 // Only consider last track started for volume and mixer state control. 4479 // In theory an older track could underrun and restart after the new one starts 4480 // but as we only care about the transition phase between two tracks on a 4481 // direct output, it is not a problem to ignore the underrun case. 4482 sp<Track> l = mLatestActiveTrack.promote(); 4483 bool last = l.get() == track; 4484 4485 if (track->isInvalid()) { 4486 ALOGW("An invalidated track shouldn't be in active list"); 4487 tracksToRemove->add(track); 4488 continue; 4489 } 4490 4491 if (track->mState == TrackBase::IDLE) { 4492 ALOGW("An idle track shouldn't be in active list"); 4493 continue; 4494 } 4495 4496 if (track->isPausing()) { 4497 track->setPaused(); 4498 if (last) { 4499 if (!mHwPaused) { 4500 doHwPause = true; 4501 mHwPaused = true; 4502 } 4503 // If we were part way through writing the mixbuffer to 4504 // the HAL we must save this until we resume 4505 // BUG - this will be wrong if a different track is made active, 4506 // in that case we want to discard the pending data in the 4507 // mixbuffer and tell the client to present it again when the 4508 // track is resumed 4509 mPausedWriteLength = mCurrentWriteLength; 4510 mPausedBytesRemaining = mBytesRemaining; 4511 mBytesRemaining = 0; // stop writing 4512 } 4513 tracksToRemove->add(track); 4514 } else if (track->isFlushPending()) { 4515 track->flushAck(); 4516 if (last) { 4517 mFlushPending = true; 4518 } 4519 } else if (track->isResumePending()){ 4520 track->resumeAck(); 4521 if (last) { 4522 if (mPausedBytesRemaining) { 4523 // Need to continue write that was interrupted 4524 mCurrentWriteLength = mPausedWriteLength; 4525 mBytesRemaining = mPausedBytesRemaining; 4526 mPausedBytesRemaining = 0; 4527 } 4528 if (mHwPaused) { 4529 doHwResume = true; 4530 mHwPaused = false; 4531 // threadLoop_mix() will handle the case that we need to 4532 // resume an interrupted write 4533 } 4534 // enable write to audio HAL 4535 sleepTime = 0; 4536 4537 // Do not handle new data in this iteration even if track->framesReady() 4538 mixerStatus = MIXER_TRACKS_ENABLED; 4539 } 4540 } else if (track->framesReady() && track->isReady() && 4541 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4542 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4543 if (track->mFillingUpStatus == Track::FS_FILLED) { 4544 track->mFillingUpStatus = Track::FS_ACTIVE; 4545 // make sure processVolume_l() will apply new volume even if 0 4546 mLeftVolFloat = mRightVolFloat = -1.0; 4547 } 4548 4549 if (last) { 4550 sp<Track> previousTrack = mPreviousTrack.promote(); 4551 if (previousTrack != 0) { 4552 if (track != previousTrack.get()) { 4553 // Flush any data still being written from last track 4554 mBytesRemaining = 0; 4555 if (mPausedBytesRemaining) { 4556 // Last track was paused so we also need to flush saved 4557 // mixbuffer state and invalidate track so that it will 4558 // re-submit that unwritten data when it is next resumed 4559 mPausedBytesRemaining = 0; 4560 // Invalidate is a bit drastic - would be more efficient 4561 // to have a flag to tell client that some of the 4562 // previously written data was lost 4563 previousTrack->invalidate(); 4564 } 4565 // flush data already sent to the DSP if changing audio session as audio 4566 // comes from a different source. Also invalidate previous track to force a 4567 // seek when resuming. 4568 if (previousTrack->sessionId() != track->sessionId()) { 4569 previousTrack->invalidate(); 4570 } 4571 } 4572 } 4573 mPreviousTrack = track; 4574 // reset retry count 4575 track->mRetryCount = kMaxTrackRetriesOffload; 4576 mActiveTrack = t; 4577 mixerStatus = MIXER_TRACKS_READY; 4578 } 4579 } else { 4580 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4581 if (track->isStopping_1()) { 4582 // Hardware buffer can hold a large amount of audio so we must 4583 // wait for all current track's data to drain before we say 4584 // that the track is stopped. 4585 if (mBytesRemaining == 0) { 4586 // Only start draining when all data in mixbuffer 4587 // has been written 4588 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4589 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4590 // do not drain if no data was ever sent to HAL (mStandby == true) 4591 if (last && !mStandby) { 4592 // do not modify drain sequence if we are already draining. This happens 4593 // when resuming from pause after drain. 4594 if ((mDrainSequence & 1) == 0) { 4595 sleepTime = 0; 4596 standbyTime = systemTime() + standbyDelay; 4597 mixerStatus = MIXER_DRAIN_TRACK; 4598 mDrainSequence += 2; 4599 } 4600 if (mHwPaused) { 4601 // It is possible to move from PAUSED to STOPPING_1 without 4602 // a resume so we must ensure hardware is running 4603 doHwResume = true; 4604 mHwPaused = false; 4605 } 4606 } 4607 } 4608 } else if (track->isStopping_2()) { 4609 // Drain has completed or we are in standby, signal presentation complete 4610 if (!(mDrainSequence & 1) || !last || mStandby) { 4611 track->mState = TrackBase::STOPPED; 4612 size_t audioHALFrames = 4613 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4614 size_t framesWritten = 4615 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4616 track->presentationComplete(framesWritten, audioHALFrames); 4617 track->reset(); 4618 tracksToRemove->add(track); 4619 } 4620 } else { 4621 // No buffers for this track. Give it a few chances to 4622 // fill a buffer, then remove it from active list. 4623 if (--(track->mRetryCount) <= 0) { 4624 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4625 track->name()); 4626 tracksToRemove->add(track); 4627 // indicate to client process that the track was disabled because of underrun; 4628 // it will then automatically call start() when data is available 4629 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4630 } else if (last){ 4631 mixerStatus = MIXER_TRACKS_ENABLED; 4632 } 4633 } 4634 } 4635 // compute volume for this track 4636 processVolume_l(track, last); 4637 } 4638 4639 // make sure the pause/flush/resume sequence is executed in the right order. 4640 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4641 // before flush and then resume HW. This can happen in case of pause/flush/resume 4642 // if resume is received before pause is executed. 4643 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4644 mOutput->stream->pause(mOutput->stream); 4645 } 4646 if (mFlushPending) { 4647 flushHw_l(); 4648 mFlushPending = false; 4649 } 4650 if (!mStandby && doHwResume) { 4651 mOutput->stream->resume(mOutput->stream); 4652 } 4653 4654 // remove all the tracks that need to be... 4655 removeTracks_l(*tracksToRemove); 4656 4657 return mixerStatus; 4658} 4659 4660// must be called with thread mutex locked 4661bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4662{ 4663 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4664 mWriteAckSequence, mDrainSequence); 4665 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4666 return true; 4667 } 4668 return false; 4669} 4670 4671// must be called with thread mutex locked 4672bool AudioFlinger::OffloadThread::shouldStandby_l() 4673{ 4674 bool trackPaused = false; 4675 4676 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4677 // after a timeout and we will enter standby then. 4678 if (mTracks.size() > 0) { 4679 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4680 } 4681 4682 return !mStandby && !trackPaused; 4683} 4684 4685 4686bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4687{ 4688 Mutex::Autolock _l(mLock); 4689 return waitingAsyncCallback_l(); 4690} 4691 4692void AudioFlinger::OffloadThread::flushHw_l() 4693{ 4694 DirectOutputThread::flushHw_l(); 4695 // Flush anything still waiting in the mixbuffer 4696 mCurrentWriteLength = 0; 4697 mBytesRemaining = 0; 4698 mPausedWriteLength = 0; 4699 mPausedBytesRemaining = 0; 4700 mHwPaused = false; 4701 4702 if (mUseAsyncWrite) { 4703 // discard any pending drain or write ack by incrementing sequence 4704 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4705 mDrainSequence = (mDrainSequence + 2) & ~1; 4706 ALOG_ASSERT(mCallbackThread != 0); 4707 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4708 mCallbackThread->setDraining(mDrainSequence); 4709 } 4710} 4711 4712void AudioFlinger::OffloadThread::onAddNewTrack_l() 4713{ 4714 sp<Track> previousTrack = mPreviousTrack.promote(); 4715 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4716 4717 if (previousTrack != 0 && latestTrack != 0 && 4718 (previousTrack->sessionId() != latestTrack->sessionId())) { 4719 mFlushPending = true; 4720 } 4721 PlaybackThread::onAddNewTrack_l(); 4722} 4723 4724// ---------------------------------------------------------------------------- 4725 4726AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4727 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4728 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4729 DUPLICATING), 4730 mWaitTimeMs(UINT_MAX) 4731{ 4732 addOutputTrack(mainThread); 4733} 4734 4735AudioFlinger::DuplicatingThread::~DuplicatingThread() 4736{ 4737 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4738 mOutputTracks[i]->destroy(); 4739 } 4740} 4741 4742void AudioFlinger::DuplicatingThread::threadLoop_mix() 4743{ 4744 // mix buffers... 4745 if (outputsReady(outputTracks)) { 4746 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4747 } else { 4748 if (mMixerBufferValid) { 4749 memset(mMixerBuffer, 0, mMixerBufferSize); 4750 } else { 4751 memset(mSinkBuffer, 0, mSinkBufferSize); 4752 } 4753 } 4754 sleepTime = 0; 4755 writeFrames = mNormalFrameCount; 4756 mCurrentWriteLength = mSinkBufferSize; 4757 standbyTime = systemTime() + standbyDelay; 4758} 4759 4760void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4761{ 4762 if (sleepTime == 0) { 4763 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4764 sleepTime = activeSleepTime; 4765 } else { 4766 sleepTime = idleSleepTime; 4767 } 4768 } else if (mBytesWritten != 0) { 4769 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4770 writeFrames = mNormalFrameCount; 4771 memset(mSinkBuffer, 0, mSinkBufferSize); 4772 } else { 4773 // flush remaining overflow buffers in output tracks 4774 writeFrames = 0; 4775 } 4776 sleepTime = 0; 4777 } 4778} 4779 4780ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4781{ 4782 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4783 // for delivery downstream as needed. This in-place conversion is safe as 4784 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4785 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4786 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4787 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4788 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4789 } 4790 for (size_t i = 0; i < outputTracks.size(); i++) { 4791 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4792 } 4793 mStandby = false; 4794 return (ssize_t)mSinkBufferSize; 4795} 4796 4797void AudioFlinger::DuplicatingThread::threadLoop_standby() 4798{ 4799 // DuplicatingThread implements standby by stopping all tracks 4800 for (size_t i = 0; i < outputTracks.size(); i++) { 4801 outputTracks[i]->stop(); 4802 } 4803} 4804 4805void AudioFlinger::DuplicatingThread::saveOutputTracks() 4806{ 4807 outputTracks = mOutputTracks; 4808} 4809 4810void AudioFlinger::DuplicatingThread::clearOutputTracks() 4811{ 4812 outputTracks.clear(); 4813} 4814 4815void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4816{ 4817 Mutex::Autolock _l(mLock); 4818 // FIXME explain this formula 4819 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4820 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4821 // due to current usage case and restrictions on the AudioBufferProvider. 4822 // Actual buffer conversion is done in threadLoop_write(). 4823 // 4824 // TODO: This may change in the future, depending on multichannel 4825 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4826 OutputTrack *outputTrack = new OutputTrack(thread, 4827 this, 4828 mSampleRate, 4829 AUDIO_FORMAT_PCM_16_BIT, 4830 mChannelMask, 4831 frameCount, 4832 IPCThreadState::self()->getCallingUid()); 4833 if (outputTrack->cblk() != NULL) { 4834 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 4835 mOutputTracks.add(outputTrack); 4836 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4837 updateWaitTime_l(); 4838 } 4839} 4840 4841void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4842{ 4843 Mutex::Autolock _l(mLock); 4844 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4845 if (mOutputTracks[i]->thread() == thread) { 4846 mOutputTracks[i]->destroy(); 4847 mOutputTracks.removeAt(i); 4848 updateWaitTime_l(); 4849 return; 4850 } 4851 } 4852 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4853} 4854 4855// caller must hold mLock 4856void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4857{ 4858 mWaitTimeMs = UINT_MAX; 4859 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4860 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4861 if (strong != 0) { 4862 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4863 if (waitTimeMs < mWaitTimeMs) { 4864 mWaitTimeMs = waitTimeMs; 4865 } 4866 } 4867 } 4868} 4869 4870 4871bool AudioFlinger::DuplicatingThread::outputsReady( 4872 const SortedVector< sp<OutputTrack> > &outputTracks) 4873{ 4874 for (size_t i = 0; i < outputTracks.size(); i++) { 4875 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4876 if (thread == 0) { 4877 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4878 outputTracks[i].get()); 4879 return false; 4880 } 4881 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4882 // see note at standby() declaration 4883 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4884 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4885 thread.get()); 4886 return false; 4887 } 4888 } 4889 return true; 4890} 4891 4892uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4893{ 4894 return (mWaitTimeMs * 1000) / 2; 4895} 4896 4897void AudioFlinger::DuplicatingThread::cacheParameters_l() 4898{ 4899 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4900 updateWaitTime_l(); 4901 4902 MixerThread::cacheParameters_l(); 4903} 4904 4905// ---------------------------------------------------------------------------- 4906// Record 4907// ---------------------------------------------------------------------------- 4908 4909AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4910 AudioStreamIn *input, 4911 audio_io_handle_t id, 4912 audio_devices_t outDevice, 4913 audio_devices_t inDevice 4914#ifdef TEE_SINK 4915 , const sp<NBAIO_Sink>& teeSink 4916#endif 4917 ) : 4918 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4919 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4920 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4921 mRsmpInRear(0) 4922#ifdef TEE_SINK 4923 , mTeeSink(teeSink) 4924#endif 4925 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4926 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4927 // mFastCapture below 4928 , mFastCaptureFutex(0) 4929 // mInputSource 4930 // mPipeSink 4931 // mPipeSource 4932 , mPipeFramesP2(0) 4933 // mPipeMemory 4934 // mFastCaptureNBLogWriter 4935 , mFastTrackAvail(false) 4936{ 4937 snprintf(mName, kNameLength, "AudioIn_%X", id); 4938 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4939 4940 readInputParameters_l(); 4941 4942 // create an NBAIO source for the HAL input stream, and negotiate 4943 mInputSource = new AudioStreamInSource(input->stream); 4944 size_t numCounterOffers = 0; 4945 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4946 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4947 ALOG_ASSERT(index == 0); 4948 4949 // initialize fast capture depending on configuration 4950 bool initFastCapture; 4951 switch (kUseFastCapture) { 4952 case FastCapture_Never: 4953 initFastCapture = false; 4954 break; 4955 case FastCapture_Always: 4956 initFastCapture = true; 4957 break; 4958 case FastCapture_Static: 4959 uint32_t primaryOutputSampleRate; 4960 { 4961 AutoMutex _l(audioFlinger->mHardwareLock); 4962 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4963 } 4964 initFastCapture = 4965 // either capture sample rate is same as (a reasonable) primary output sample rate 4966 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4967 (mSampleRate == primaryOutputSampleRate)) || 4968 // or primary output sample rate is unknown, and capture sample rate is reasonable 4969 ((primaryOutputSampleRate == 0) && 4970 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4971 // and the buffer size is < 12 ms 4972 (mFrameCount * 1000) / mSampleRate < 12; 4973 break; 4974 // case FastCapture_Dynamic: 4975 } 4976 4977 if (initFastCapture) { 4978 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4979 NBAIO_Format format = mInputSource->format(); 4980 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4981 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4982 void *pipeBuffer; 4983 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4984 sp<IMemory> pipeMemory; 4985 if ((roHeap == 0) || 4986 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4987 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4988 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4989 goto failed; 4990 } 4991 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4992 memset(pipeBuffer, 0, pipeSize); 4993 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4994 const NBAIO_Format offers[1] = {format}; 4995 size_t numCounterOffers = 0; 4996 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4997 ALOG_ASSERT(index == 0); 4998 mPipeSink = pipe; 4999 PipeReader *pipeReader = new PipeReader(*pipe); 5000 numCounterOffers = 0; 5001 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5002 ALOG_ASSERT(index == 0); 5003 mPipeSource = pipeReader; 5004 mPipeFramesP2 = pipeFramesP2; 5005 mPipeMemory = pipeMemory; 5006 5007 // create fast capture 5008 mFastCapture = new FastCapture(); 5009 FastCaptureStateQueue *sq = mFastCapture->sq(); 5010#ifdef STATE_QUEUE_DUMP 5011 // FIXME 5012#endif 5013 FastCaptureState *state = sq->begin(); 5014 state->mCblk = NULL; 5015 state->mInputSource = mInputSource.get(); 5016 state->mInputSourceGen++; 5017 state->mPipeSink = pipe; 5018 state->mPipeSinkGen++; 5019 state->mFrameCount = mFrameCount; 5020 state->mCommand = FastCaptureState::COLD_IDLE; 5021 // already done in constructor initialization list 5022 //mFastCaptureFutex = 0; 5023 state->mColdFutexAddr = &mFastCaptureFutex; 5024 state->mColdGen++; 5025 state->mDumpState = &mFastCaptureDumpState; 5026#ifdef TEE_SINK 5027 // FIXME 5028#endif 5029 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5030 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5031 sq->end(); 5032 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5033 5034 // start the fast capture 5035 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5036 pid_t tid = mFastCapture->getTid(); 5037 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5038 if (err != 0) { 5039 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5040 kPriorityFastCapture, getpid_cached, tid, err); 5041 } 5042 5043#ifdef AUDIO_WATCHDOG 5044 // FIXME 5045#endif 5046 5047 mFastTrackAvail = true; 5048 } 5049failed: ; 5050 5051 // FIXME mNormalSource 5052} 5053 5054 5055AudioFlinger::RecordThread::~RecordThread() 5056{ 5057 if (mFastCapture != 0) { 5058 FastCaptureStateQueue *sq = mFastCapture->sq(); 5059 FastCaptureState *state = sq->begin(); 5060 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5061 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5062 if (old == -1) { 5063 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5064 } 5065 } 5066 state->mCommand = FastCaptureState::EXIT; 5067 sq->end(); 5068 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5069 mFastCapture->join(); 5070 mFastCapture.clear(); 5071 } 5072 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5073 mAudioFlinger->unregisterWriter(mNBLogWriter); 5074 delete[] mRsmpInBuffer; 5075} 5076 5077void AudioFlinger::RecordThread::onFirstRef() 5078{ 5079 run(mName, PRIORITY_URGENT_AUDIO); 5080} 5081 5082bool AudioFlinger::RecordThread::threadLoop() 5083{ 5084 nsecs_t lastWarning = 0; 5085 5086 inputStandBy(); 5087 5088reacquire_wakelock: 5089 sp<RecordTrack> activeTrack; 5090 int activeTracksGen; 5091 { 5092 Mutex::Autolock _l(mLock); 5093 size_t size = mActiveTracks.size(); 5094 activeTracksGen = mActiveTracksGen; 5095 if (size > 0) { 5096 // FIXME an arbitrary choice 5097 activeTrack = mActiveTracks[0]; 5098 acquireWakeLock_l(activeTrack->uid()); 5099 if (size > 1) { 5100 SortedVector<int> tmp; 5101 for (size_t i = 0; i < size; i++) { 5102 tmp.add(mActiveTracks[i]->uid()); 5103 } 5104 updateWakeLockUids_l(tmp); 5105 } 5106 } else { 5107 acquireWakeLock_l(-1); 5108 } 5109 } 5110 5111 // used to request a deferred sleep, to be executed later while mutex is unlocked 5112 uint32_t sleepUs = 0; 5113 5114 // loop while there is work to do 5115 for (;;) { 5116 Vector< sp<EffectChain> > effectChains; 5117 5118 // sleep with mutex unlocked 5119 if (sleepUs > 0) { 5120 ATRACE_BEGIN("sleep"); 5121 usleep(sleepUs); 5122 ATRACE_END(); 5123 sleepUs = 0; 5124 } 5125 5126 // activeTracks accumulates a copy of a subset of mActiveTracks 5127 Vector< sp<RecordTrack> > activeTracks; 5128 5129 // reference to the (first and only) active fast track 5130 sp<RecordTrack> fastTrack; 5131 5132 // reference to a fast track which is about to be removed 5133 sp<RecordTrack> fastTrackToRemove; 5134 5135 { // scope for mLock 5136 Mutex::Autolock _l(mLock); 5137 5138 processConfigEvents_l(); 5139 5140 // check exitPending here because checkForNewParameters_l() and 5141 // checkForNewParameters_l() can temporarily release mLock 5142 if (exitPending()) { 5143 break; 5144 } 5145 5146 // if no active track(s), then standby and release wakelock 5147 size_t size = mActiveTracks.size(); 5148 if (size == 0) { 5149 standbyIfNotAlreadyInStandby(); 5150 // exitPending() can't become true here 5151 releaseWakeLock_l(); 5152 ALOGV("RecordThread: loop stopping"); 5153 // go to sleep 5154 mWaitWorkCV.wait(mLock); 5155 ALOGV("RecordThread: loop starting"); 5156 goto reacquire_wakelock; 5157 } 5158 5159 if (mActiveTracksGen != activeTracksGen) { 5160 activeTracksGen = mActiveTracksGen; 5161 SortedVector<int> tmp; 5162 for (size_t i = 0; i < size; i++) { 5163 tmp.add(mActiveTracks[i]->uid()); 5164 } 5165 updateWakeLockUids_l(tmp); 5166 } 5167 5168 bool doBroadcast = false; 5169 for (size_t i = 0; i < size; ) { 5170 5171 activeTrack = mActiveTracks[i]; 5172 if (activeTrack->isTerminated()) { 5173 if (activeTrack->isFastTrack()) { 5174 ALOG_ASSERT(fastTrackToRemove == 0); 5175 fastTrackToRemove = activeTrack; 5176 } 5177 removeTrack_l(activeTrack); 5178 mActiveTracks.remove(activeTrack); 5179 mActiveTracksGen++; 5180 size--; 5181 continue; 5182 } 5183 5184 TrackBase::track_state activeTrackState = activeTrack->mState; 5185 switch (activeTrackState) { 5186 5187 case TrackBase::PAUSING: 5188 mActiveTracks.remove(activeTrack); 5189 mActiveTracksGen++; 5190 doBroadcast = true; 5191 size--; 5192 continue; 5193 5194 case TrackBase::STARTING_1: 5195 sleepUs = 10000; 5196 i++; 5197 continue; 5198 5199 case TrackBase::STARTING_2: 5200 doBroadcast = true; 5201 mStandby = false; 5202 activeTrack->mState = TrackBase::ACTIVE; 5203 break; 5204 5205 case TrackBase::ACTIVE: 5206 break; 5207 5208 case TrackBase::IDLE: 5209 i++; 5210 continue; 5211 5212 default: 5213 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5214 } 5215 5216 activeTracks.add(activeTrack); 5217 i++; 5218 5219 if (activeTrack->isFastTrack()) { 5220 ALOG_ASSERT(!mFastTrackAvail); 5221 ALOG_ASSERT(fastTrack == 0); 5222 fastTrack = activeTrack; 5223 } 5224 } 5225 if (doBroadcast) { 5226 mStartStopCond.broadcast(); 5227 } 5228 5229 // sleep if there are no active tracks to process 5230 if (activeTracks.size() == 0) { 5231 if (sleepUs == 0) { 5232 sleepUs = kRecordThreadSleepUs; 5233 } 5234 continue; 5235 } 5236 sleepUs = 0; 5237 5238 lockEffectChains_l(effectChains); 5239 } 5240 5241 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5242 5243 size_t size = effectChains.size(); 5244 for (size_t i = 0; i < size; i++) { 5245 // thread mutex is not locked, but effect chain is locked 5246 effectChains[i]->process_l(); 5247 } 5248 5249 // Push a new fast capture state if fast capture is not already running, or cblk change 5250 if (mFastCapture != 0) { 5251 FastCaptureStateQueue *sq = mFastCapture->sq(); 5252 FastCaptureState *state = sq->begin(); 5253 bool didModify = false; 5254 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5255 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5256 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5257 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5258 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5259 if (old == -1) { 5260 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5261 } 5262 } 5263 state->mCommand = FastCaptureState::READ_WRITE; 5264#if 0 // FIXME 5265 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5266 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5267#endif 5268 didModify = true; 5269 } 5270 audio_track_cblk_t *cblkOld = state->mCblk; 5271 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5272 if (cblkNew != cblkOld) { 5273 state->mCblk = cblkNew; 5274 // block until acked if removing a fast track 5275 if (cblkOld != NULL) { 5276 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5277 } 5278 didModify = true; 5279 } 5280 sq->end(didModify); 5281 if (didModify) { 5282 sq->push(block); 5283#if 0 5284 if (kUseFastCapture == FastCapture_Dynamic) { 5285 mNormalSource = mPipeSource; 5286 } 5287#endif 5288 } 5289 } 5290 5291 // now run the fast track destructor with thread mutex unlocked 5292 fastTrackToRemove.clear(); 5293 5294 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5295 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5296 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5297 // If destination is non-contiguous, first read past the nominal end of buffer, then 5298 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5299 5300 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5301 ssize_t framesRead; 5302 5303 // If an NBAIO source is present, use it to read the normal capture's data 5304 if (mPipeSource != 0) { 5305 size_t framesToRead = mBufferSize / mFrameSize; 5306 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5307 framesToRead, AudioBufferProvider::kInvalidPTS); 5308 if (framesRead == 0) { 5309 // since pipe is non-blocking, simulate blocking input 5310 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5311 } 5312 // otherwise use the HAL / AudioStreamIn directly 5313 } else { 5314 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5315 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5316 if (bytesRead < 0) { 5317 framesRead = bytesRead; 5318 } else { 5319 framesRead = bytesRead / mFrameSize; 5320 } 5321 } 5322 5323 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5324 ALOGE("read failed: framesRead=%d", framesRead); 5325 // Force input into standby so that it tries to recover at next read attempt 5326 inputStandBy(); 5327 sleepUs = kRecordThreadSleepUs; 5328 } 5329 if (framesRead <= 0) { 5330 goto unlock; 5331 } 5332 ALOG_ASSERT(framesRead > 0); 5333 5334 if (mTeeSink != 0) { 5335 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5336 } 5337 // If destination is non-contiguous, we now correct for reading past end of buffer. 5338 { 5339 size_t part1 = mRsmpInFramesP2 - rear; 5340 if ((size_t) framesRead > part1) { 5341 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5342 (framesRead - part1) * mFrameSize); 5343 } 5344 } 5345 rear = mRsmpInRear += framesRead; 5346 5347 size = activeTracks.size(); 5348 // loop over each active track 5349 for (size_t i = 0; i < size; i++) { 5350 activeTrack = activeTracks[i]; 5351 5352 // skip fast tracks, as those are handled directly by FastCapture 5353 if (activeTrack->isFastTrack()) { 5354 continue; 5355 } 5356 5357 enum { 5358 OVERRUN_UNKNOWN, 5359 OVERRUN_TRUE, 5360 OVERRUN_FALSE 5361 } overrun = OVERRUN_UNKNOWN; 5362 5363 // loop over getNextBuffer to handle circular sink 5364 for (;;) { 5365 5366 activeTrack->mSink.frameCount = ~0; 5367 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5368 size_t framesOut = activeTrack->mSink.frameCount; 5369 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5370 5371 int32_t front = activeTrack->mRsmpInFront; 5372 ssize_t filled = rear - front; 5373 size_t framesIn; 5374 5375 if (filled < 0) { 5376 // should not happen, but treat like a massive overrun and re-sync 5377 framesIn = 0; 5378 activeTrack->mRsmpInFront = rear; 5379 overrun = OVERRUN_TRUE; 5380 } else if ((size_t) filled <= mRsmpInFrames) { 5381 framesIn = (size_t) filled; 5382 } else { 5383 // client is not keeping up with server, but give it latest data 5384 framesIn = mRsmpInFrames; 5385 activeTrack->mRsmpInFront = front = rear - framesIn; 5386 overrun = OVERRUN_TRUE; 5387 } 5388 5389 if (framesOut == 0 || framesIn == 0) { 5390 break; 5391 } 5392 5393 if (activeTrack->mResampler == NULL) { 5394 // no resampling 5395 if (framesIn > framesOut) { 5396 framesIn = framesOut; 5397 } else { 5398 framesOut = framesIn; 5399 } 5400 int8_t *dst = activeTrack->mSink.i8; 5401 while (framesIn > 0) { 5402 front &= mRsmpInFramesP2 - 1; 5403 size_t part1 = mRsmpInFramesP2 - front; 5404 if (part1 > framesIn) { 5405 part1 = framesIn; 5406 } 5407 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5408 if (mChannelCount == activeTrack->mChannelCount) { 5409 memcpy(dst, src, part1 * mFrameSize); 5410 } else if (mChannelCount == 1) { 5411 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5412 part1); 5413 } else { 5414 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5415 part1); 5416 } 5417 dst += part1 * activeTrack->mFrameSize; 5418 front += part1; 5419 framesIn -= part1; 5420 } 5421 activeTrack->mRsmpInFront += framesOut; 5422 5423 } else { 5424 // resampling 5425 // FIXME framesInNeeded should really be part of resampler API, and should 5426 // depend on the SRC ratio 5427 // to keep mRsmpInBuffer full so resampler always has sufficient input 5428 size_t framesInNeeded; 5429 // FIXME only re-calculate when it changes, and optimize for common ratios 5430 // Do not precompute in/out because floating point is not associative 5431 // e.g. a*b/c != a*(b/c). 5432 const double in(mSampleRate); 5433 const double out(activeTrack->mSampleRate); 5434 framesInNeeded = ceil(framesOut * in / out) + 1; 5435 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5436 framesInNeeded, framesOut, in / out); 5437 // Although we theoretically have framesIn in circular buffer, some of those are 5438 // unreleased frames, and thus must be discounted for purpose of budgeting. 5439 size_t unreleased = activeTrack->mRsmpInUnrel; 5440 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5441 if (framesIn < framesInNeeded) { 5442 ALOGV("not enough to resample: have %u frames in but need %u in to " 5443 "produce %u out given in/out ratio of %.4g", 5444 framesIn, framesInNeeded, framesOut, in / out); 5445 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5446 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5447 if (newFramesOut == 0) { 5448 break; 5449 } 5450 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5451 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5452 framesInNeeded, newFramesOut, out / in); 5453 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5454 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5455 "given in/out ratio of %.4g", 5456 framesIn, framesInNeeded, newFramesOut, in / out); 5457 framesOut = newFramesOut; 5458 } else { 5459 ALOGV("success 1: have %u in and need %u in to produce %u out " 5460 "given in/out ratio of %.4g", 5461 framesIn, framesInNeeded, framesOut, in / out); 5462 } 5463 5464 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5465 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5466 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5467 delete[] activeTrack->mRsmpOutBuffer; 5468 // resampler always outputs stereo 5469 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5470 activeTrack->mRsmpOutFrameCount = framesOut; 5471 } 5472 5473 // resampler accumulates, but we only have one source track 5474 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5475 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5476 // FIXME how about having activeTrack implement this interface itself? 5477 activeTrack->mResamplerBufferProvider 5478 /*this*/ /* AudioBufferProvider* */); 5479 // ditherAndClamp() works as long as all buffers returned by 5480 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5481 if (activeTrack->mChannelCount == 1) { 5482 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5483 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5484 framesOut); 5485 // the resampler always outputs stereo samples: 5486 // do post stereo to mono conversion 5487 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5488 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5489 } else { 5490 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5491 activeTrack->mRsmpOutBuffer, framesOut); 5492 } 5493 // now done with mRsmpOutBuffer 5494 5495 } 5496 5497 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5498 overrun = OVERRUN_FALSE; 5499 } 5500 5501 if (activeTrack->mFramesToDrop == 0) { 5502 if (framesOut > 0) { 5503 activeTrack->mSink.frameCount = framesOut; 5504 activeTrack->releaseBuffer(&activeTrack->mSink); 5505 } 5506 } else { 5507 // FIXME could do a partial drop of framesOut 5508 if (activeTrack->mFramesToDrop > 0) { 5509 activeTrack->mFramesToDrop -= framesOut; 5510 if (activeTrack->mFramesToDrop <= 0) { 5511 activeTrack->clearSyncStartEvent(); 5512 } 5513 } else { 5514 activeTrack->mFramesToDrop += framesOut; 5515 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5516 activeTrack->mSyncStartEvent->isCancelled()) { 5517 ALOGW("Synced record %s, session %d, trigger session %d", 5518 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5519 activeTrack->sessionId(), 5520 (activeTrack->mSyncStartEvent != 0) ? 5521 activeTrack->mSyncStartEvent->triggerSession() : 0); 5522 activeTrack->clearSyncStartEvent(); 5523 } 5524 } 5525 } 5526 5527 if (framesOut == 0) { 5528 break; 5529 } 5530 } 5531 5532 switch (overrun) { 5533 case OVERRUN_TRUE: 5534 // client isn't retrieving buffers fast enough 5535 if (!activeTrack->setOverflow()) { 5536 nsecs_t now = systemTime(); 5537 // FIXME should lastWarning per track? 5538 if ((now - lastWarning) > kWarningThrottleNs) { 5539 ALOGW("RecordThread: buffer overflow"); 5540 lastWarning = now; 5541 } 5542 } 5543 break; 5544 case OVERRUN_FALSE: 5545 activeTrack->clearOverflow(); 5546 break; 5547 case OVERRUN_UNKNOWN: 5548 break; 5549 } 5550 5551 } 5552 5553unlock: 5554 // enable changes in effect chain 5555 unlockEffectChains(effectChains); 5556 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5557 } 5558 5559 standbyIfNotAlreadyInStandby(); 5560 5561 { 5562 Mutex::Autolock _l(mLock); 5563 for (size_t i = 0; i < mTracks.size(); i++) { 5564 sp<RecordTrack> track = mTracks[i]; 5565 track->invalidate(); 5566 } 5567 mActiveTracks.clear(); 5568 mActiveTracksGen++; 5569 mStartStopCond.broadcast(); 5570 } 5571 5572 releaseWakeLock(); 5573 5574 ALOGV("RecordThread %p exiting", this); 5575 return false; 5576} 5577 5578void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5579{ 5580 if (!mStandby) { 5581 inputStandBy(); 5582 mStandby = true; 5583 } 5584} 5585 5586void AudioFlinger::RecordThread::inputStandBy() 5587{ 5588 // Idle the fast capture if it's currently running 5589 if (mFastCapture != 0) { 5590 FastCaptureStateQueue *sq = mFastCapture->sq(); 5591 FastCaptureState *state = sq->begin(); 5592 if (!(state->mCommand & FastCaptureState::IDLE)) { 5593 state->mCommand = FastCaptureState::COLD_IDLE; 5594 state->mColdFutexAddr = &mFastCaptureFutex; 5595 state->mColdGen++; 5596 mFastCaptureFutex = 0; 5597 sq->end(); 5598 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5599 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5600#if 0 5601 if (kUseFastCapture == FastCapture_Dynamic) { 5602 // FIXME 5603 } 5604#endif 5605#ifdef AUDIO_WATCHDOG 5606 // FIXME 5607#endif 5608 } else { 5609 sq->end(false /*didModify*/); 5610 } 5611 } 5612 mInput->stream->common.standby(&mInput->stream->common); 5613} 5614 5615// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5616sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5617 const sp<AudioFlinger::Client>& client, 5618 uint32_t sampleRate, 5619 audio_format_t format, 5620 audio_channel_mask_t channelMask, 5621 size_t *pFrameCount, 5622 int sessionId, 5623 size_t *notificationFrames, 5624 int uid, 5625 IAudioFlinger::track_flags_t *flags, 5626 pid_t tid, 5627 status_t *status) 5628{ 5629 size_t frameCount = *pFrameCount; 5630 sp<RecordTrack> track; 5631 status_t lStatus; 5632 5633 // client expresses a preference for FAST, but we get the final say 5634 if (*flags & IAudioFlinger::TRACK_FAST) { 5635 if ( 5636 // use case: callback handler 5637 (tid != -1) && 5638 // frame count is not specified, or is exactly the pipe depth 5639 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5640 // PCM data 5641 audio_is_linear_pcm(format) && 5642 // native format 5643 (format == mFormat) && 5644 // native channel mask 5645 (channelMask == mChannelMask) && 5646 // native hardware sample rate 5647 (sampleRate == mSampleRate) && 5648 // record thread has an associated fast capture 5649 hasFastCapture() && 5650 // there are sufficient fast track slots available 5651 mFastTrackAvail 5652 ) { 5653 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5654 frameCount, mFrameCount); 5655 } else { 5656 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5657 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5658 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5659 frameCount, mFrameCount, mPipeFramesP2, 5660 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5661 hasFastCapture(), tid, mFastTrackAvail); 5662 *flags &= ~IAudioFlinger::TRACK_FAST; 5663 } 5664 } 5665 5666 // compute track buffer size in frames, and suggest the notification frame count 5667 if (*flags & IAudioFlinger::TRACK_FAST) { 5668 // fast track: frame count is exactly the pipe depth 5669 frameCount = mPipeFramesP2; 5670 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5671 *notificationFrames = mFrameCount; 5672 } else { 5673 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5674 // or 20 ms if there is a fast capture 5675 // TODO This could be a roundupRatio inline, and const 5676 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5677 * sampleRate + mSampleRate - 1) / mSampleRate; 5678 // minimum number of notification periods is at least kMinNotifications, 5679 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5680 static const size_t kMinNotifications = 3; 5681 static const uint32_t kMinMs = 30; 5682 // TODO This could be a roundupRatio inline 5683 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5684 // TODO This could be a roundupRatio inline 5685 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5686 maxNotificationFrames; 5687 const size_t minFrameCount = maxNotificationFrames * 5688 max(kMinNotifications, minNotificationsByMs); 5689 frameCount = max(frameCount, minFrameCount); 5690 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5691 *notificationFrames = maxNotificationFrames; 5692 } 5693 } 5694 *pFrameCount = frameCount; 5695 5696 lStatus = initCheck(); 5697 if (lStatus != NO_ERROR) { 5698 ALOGE("createRecordTrack_l() audio driver not initialized"); 5699 goto Exit; 5700 } 5701 5702 { // scope for mLock 5703 Mutex::Autolock _l(mLock); 5704 5705 track = new RecordTrack(this, client, sampleRate, 5706 format, channelMask, frameCount, NULL, sessionId, uid, 5707 *flags, TrackBase::TYPE_DEFAULT); 5708 5709 lStatus = track->initCheck(); 5710 if (lStatus != NO_ERROR) { 5711 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5712 // track must be cleared from the caller as the caller has the AF lock 5713 goto Exit; 5714 } 5715 mTracks.add(track); 5716 5717 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5718 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5719 mAudioFlinger->btNrecIsOff(); 5720 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5721 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5722 5723 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5724 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5725 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5726 // so ask activity manager to do this on our behalf 5727 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5728 } 5729 } 5730 5731 lStatus = NO_ERROR; 5732 5733Exit: 5734 *status = lStatus; 5735 return track; 5736} 5737 5738status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5739 AudioSystem::sync_event_t event, 5740 int triggerSession) 5741{ 5742 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5743 sp<ThreadBase> strongMe = this; 5744 status_t status = NO_ERROR; 5745 5746 if (event == AudioSystem::SYNC_EVENT_NONE) { 5747 recordTrack->clearSyncStartEvent(); 5748 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5749 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5750 triggerSession, 5751 recordTrack->sessionId(), 5752 syncStartEventCallback, 5753 recordTrack); 5754 // Sync event can be cancelled by the trigger session if the track is not in a 5755 // compatible state in which case we start record immediately 5756 if (recordTrack->mSyncStartEvent->isCancelled()) { 5757 recordTrack->clearSyncStartEvent(); 5758 } else { 5759 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5760 recordTrack->mFramesToDrop = - 5761 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5762 } 5763 } 5764 5765 { 5766 // This section is a rendezvous between binder thread executing start() and RecordThread 5767 AutoMutex lock(mLock); 5768 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5769 if (recordTrack->mState == TrackBase::PAUSING) { 5770 ALOGV("active record track PAUSING -> ACTIVE"); 5771 recordTrack->mState = TrackBase::ACTIVE; 5772 } else { 5773 ALOGV("active record track state %d", recordTrack->mState); 5774 } 5775 return status; 5776 } 5777 5778 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5779 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5780 // or using a separate command thread 5781 recordTrack->mState = TrackBase::STARTING_1; 5782 mActiveTracks.add(recordTrack); 5783 mActiveTracksGen++; 5784 status_t status = NO_ERROR; 5785 if (recordTrack->isExternalTrack()) { 5786 mLock.unlock(); 5787 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5788 mLock.lock(); 5789 // FIXME should verify that recordTrack is still in mActiveTracks 5790 if (status != NO_ERROR) { 5791 mActiveTracks.remove(recordTrack); 5792 mActiveTracksGen++; 5793 recordTrack->clearSyncStartEvent(); 5794 ALOGV("RecordThread::start error %d", status); 5795 return status; 5796 } 5797 } 5798 // Catch up with current buffer indices if thread is already running. 5799 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5800 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5801 // see previously buffered data before it called start(), but with greater risk of overrun. 5802 5803 recordTrack->mRsmpInFront = mRsmpInRear; 5804 recordTrack->mRsmpInUnrel = 0; 5805 // FIXME why reset? 5806 if (recordTrack->mResampler != NULL) { 5807 recordTrack->mResampler->reset(); 5808 } 5809 recordTrack->mState = TrackBase::STARTING_2; 5810 // signal thread to start 5811 mWaitWorkCV.broadcast(); 5812 if (mActiveTracks.indexOf(recordTrack) < 0) { 5813 ALOGV("Record failed to start"); 5814 status = BAD_VALUE; 5815 goto startError; 5816 } 5817 return status; 5818 } 5819 5820startError: 5821 if (recordTrack->isExternalTrack()) { 5822 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5823 } 5824 recordTrack->clearSyncStartEvent(); 5825 // FIXME I wonder why we do not reset the state here? 5826 return status; 5827} 5828 5829void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5830{ 5831 sp<SyncEvent> strongEvent = event.promote(); 5832 5833 if (strongEvent != 0) { 5834 sp<RefBase> ptr = strongEvent->cookie().promote(); 5835 if (ptr != 0) { 5836 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5837 recordTrack->handleSyncStartEvent(strongEvent); 5838 } 5839 } 5840} 5841 5842bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5843 ALOGV("RecordThread::stop"); 5844 AutoMutex _l(mLock); 5845 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5846 return false; 5847 } 5848 // note that threadLoop may still be processing the track at this point [without lock] 5849 recordTrack->mState = TrackBase::PAUSING; 5850 // do not wait for mStartStopCond if exiting 5851 if (exitPending()) { 5852 return true; 5853 } 5854 // FIXME incorrect usage of wait: no explicit predicate or loop 5855 mStartStopCond.wait(mLock); 5856 // if we have been restarted, recordTrack is in mActiveTracks here 5857 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5858 ALOGV("Record stopped OK"); 5859 return true; 5860 } 5861 return false; 5862} 5863 5864bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5865{ 5866 return false; 5867} 5868 5869status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5870{ 5871#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5872 if (!isValidSyncEvent(event)) { 5873 return BAD_VALUE; 5874 } 5875 5876 int eventSession = event->triggerSession(); 5877 status_t ret = NAME_NOT_FOUND; 5878 5879 Mutex::Autolock _l(mLock); 5880 5881 for (size_t i = 0; i < mTracks.size(); i++) { 5882 sp<RecordTrack> track = mTracks[i]; 5883 if (eventSession == track->sessionId()) { 5884 (void) track->setSyncEvent(event); 5885 ret = NO_ERROR; 5886 } 5887 } 5888 return ret; 5889#else 5890 return BAD_VALUE; 5891#endif 5892} 5893 5894// destroyTrack_l() must be called with ThreadBase::mLock held 5895void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5896{ 5897 track->terminate(); 5898 track->mState = TrackBase::STOPPED; 5899 // active tracks are removed by threadLoop() 5900 if (mActiveTracks.indexOf(track) < 0) { 5901 removeTrack_l(track); 5902 } 5903} 5904 5905void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5906{ 5907 mTracks.remove(track); 5908 // need anything related to effects here? 5909 if (track->isFastTrack()) { 5910 ALOG_ASSERT(!mFastTrackAvail); 5911 mFastTrackAvail = true; 5912 } 5913} 5914 5915void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5916{ 5917 dumpInternals(fd, args); 5918 dumpTracks(fd, args); 5919 dumpEffectChains(fd, args); 5920} 5921 5922void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5923{ 5924 dprintf(fd, "\nInput thread %p:\n", this); 5925 5926 if (mActiveTracks.size() > 0) { 5927 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5928 } else { 5929 dprintf(fd, " No active record clients\n"); 5930 } 5931 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5932 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5933 5934 dumpBase(fd, args); 5935} 5936 5937void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5938{ 5939 const size_t SIZE = 256; 5940 char buffer[SIZE]; 5941 String8 result; 5942 5943 size_t numtracks = mTracks.size(); 5944 size_t numactive = mActiveTracks.size(); 5945 size_t numactiveseen = 0; 5946 dprintf(fd, " %d Tracks", numtracks); 5947 if (numtracks) { 5948 dprintf(fd, " of which %d are active\n", numactive); 5949 RecordTrack::appendDumpHeader(result); 5950 for (size_t i = 0; i < numtracks ; ++i) { 5951 sp<RecordTrack> track = mTracks[i]; 5952 if (track != 0) { 5953 bool active = mActiveTracks.indexOf(track) >= 0; 5954 if (active) { 5955 numactiveseen++; 5956 } 5957 track->dump(buffer, SIZE, active); 5958 result.append(buffer); 5959 } 5960 } 5961 } else { 5962 dprintf(fd, "\n"); 5963 } 5964 5965 if (numactiveseen != numactive) { 5966 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5967 " not in the track list\n"); 5968 result.append(buffer); 5969 RecordTrack::appendDumpHeader(result); 5970 for (size_t i = 0; i < numactive; ++i) { 5971 sp<RecordTrack> track = mActiveTracks[i]; 5972 if (mTracks.indexOf(track) < 0) { 5973 track->dump(buffer, SIZE, true); 5974 result.append(buffer); 5975 } 5976 } 5977 5978 } 5979 write(fd, result.string(), result.size()); 5980} 5981 5982// AudioBufferProvider interface 5983status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5984 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5985{ 5986 RecordTrack *activeTrack = mRecordTrack; 5987 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5988 if (threadBase == 0) { 5989 buffer->frameCount = 0; 5990 buffer->raw = NULL; 5991 return NOT_ENOUGH_DATA; 5992 } 5993 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5994 int32_t rear = recordThread->mRsmpInRear; 5995 int32_t front = activeTrack->mRsmpInFront; 5996 ssize_t filled = rear - front; 5997 // FIXME should not be P2 (don't want to increase latency) 5998 // FIXME if client not keeping up, discard 5999 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6000 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6001 front &= recordThread->mRsmpInFramesP2 - 1; 6002 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6003 if (part1 > (size_t) filled) { 6004 part1 = filled; 6005 } 6006 size_t ask = buffer->frameCount; 6007 ALOG_ASSERT(ask > 0); 6008 if (part1 > ask) { 6009 part1 = ask; 6010 } 6011 if (part1 == 0) { 6012 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 6013 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 6014 buffer->raw = NULL; 6015 buffer->frameCount = 0; 6016 activeTrack->mRsmpInUnrel = 0; 6017 return NOT_ENOUGH_DATA; 6018 } 6019 6020 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6021 buffer->frameCount = part1; 6022 activeTrack->mRsmpInUnrel = part1; 6023 return NO_ERROR; 6024} 6025 6026// AudioBufferProvider interface 6027void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6028 AudioBufferProvider::Buffer* buffer) 6029{ 6030 RecordTrack *activeTrack = mRecordTrack; 6031 size_t stepCount = buffer->frameCount; 6032 if (stepCount == 0) { 6033 return; 6034 } 6035 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 6036 activeTrack->mRsmpInUnrel -= stepCount; 6037 activeTrack->mRsmpInFront += stepCount; 6038 buffer->raw = NULL; 6039 buffer->frameCount = 0; 6040} 6041 6042bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6043 status_t& status) 6044{ 6045 bool reconfig = false; 6046 6047 status = NO_ERROR; 6048 6049 audio_format_t reqFormat = mFormat; 6050 uint32_t samplingRate = mSampleRate; 6051 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6052 6053 AudioParameter param = AudioParameter(keyValuePair); 6054 int value; 6055 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6056 // channel count change can be requested. Do we mandate the first client defines the 6057 // HAL sampling rate and channel count or do we allow changes on the fly? 6058 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6059 samplingRate = value; 6060 reconfig = true; 6061 } 6062 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6063 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 6064 status = BAD_VALUE; 6065 } else { 6066 reqFormat = (audio_format_t) value; 6067 reconfig = true; 6068 } 6069 } 6070 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6071 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6072 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6073 status = BAD_VALUE; 6074 } else { 6075 channelMask = mask; 6076 reconfig = true; 6077 } 6078 } 6079 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6080 // do not accept frame count changes if tracks are open as the track buffer 6081 // size depends on frame count and correct behavior would not be guaranteed 6082 // if frame count is changed after track creation 6083 if (mActiveTracks.size() > 0) { 6084 status = INVALID_OPERATION; 6085 } else { 6086 reconfig = true; 6087 } 6088 } 6089 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6090 // forward device change to effects that have requested to be 6091 // aware of attached audio device. 6092 for (size_t i = 0; i < mEffectChains.size(); i++) { 6093 mEffectChains[i]->setDevice_l(value); 6094 } 6095 6096 // store input device and output device but do not forward output device to audio HAL. 6097 // Note that status is ignored by the caller for output device 6098 // (see AudioFlinger::setParameters() 6099 if (audio_is_output_devices(value)) { 6100 mOutDevice = value; 6101 status = BAD_VALUE; 6102 } else { 6103 mInDevice = value; 6104 // disable AEC and NS if the device is a BT SCO headset supporting those 6105 // pre processings 6106 if (mTracks.size() > 0) { 6107 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6108 mAudioFlinger->btNrecIsOff(); 6109 for (size_t i = 0; i < mTracks.size(); i++) { 6110 sp<RecordTrack> track = mTracks[i]; 6111 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6112 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6113 } 6114 } 6115 } 6116 } 6117 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6118 mAudioSource != (audio_source_t)value) { 6119 // forward device change to effects that have requested to be 6120 // aware of attached audio device. 6121 for (size_t i = 0; i < mEffectChains.size(); i++) { 6122 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6123 } 6124 mAudioSource = (audio_source_t)value; 6125 } 6126 6127 if (status == NO_ERROR) { 6128 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6129 keyValuePair.string()); 6130 if (status == INVALID_OPERATION) { 6131 inputStandBy(); 6132 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6133 keyValuePair.string()); 6134 } 6135 if (reconfig) { 6136 if (status == BAD_VALUE && 6137 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6138 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6139 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6140 <= (2 * samplingRate)) && 6141 audio_channel_count_from_in_mask( 6142 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6143 (channelMask == AUDIO_CHANNEL_IN_MONO || 6144 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6145 status = NO_ERROR; 6146 } 6147 if (status == NO_ERROR) { 6148 readInputParameters_l(); 6149 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6150 } 6151 } 6152 } 6153 6154 return reconfig; 6155} 6156 6157String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6158{ 6159 Mutex::Autolock _l(mLock); 6160 if (initCheck() != NO_ERROR) { 6161 return String8(); 6162 } 6163 6164 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6165 const String8 out_s8(s); 6166 free(s); 6167 return out_s8; 6168} 6169 6170void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6171 AudioSystem::OutputDescriptor desc; 6172 const void *param2 = NULL; 6173 6174 switch (event) { 6175 case AudioSystem::INPUT_OPENED: 6176 case AudioSystem::INPUT_CONFIG_CHANGED: 6177 desc.channelMask = mChannelMask; 6178 desc.samplingRate = mSampleRate; 6179 desc.format = mFormat; 6180 desc.frameCount = mFrameCount; 6181 desc.latency = 0; 6182 param2 = &desc; 6183 break; 6184 6185 case AudioSystem::INPUT_CLOSED: 6186 default: 6187 break; 6188 } 6189 mAudioFlinger->audioConfigChanged(event, mId, param2); 6190} 6191 6192void AudioFlinger::RecordThread::readInputParameters_l() 6193{ 6194 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6195 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6196 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6197 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6198 mFormat = mHALFormat; 6199 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6200 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6201 } 6202 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6203 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6204 mFrameCount = mBufferSize / mFrameSize; 6205 // This is the formula for calculating the temporary buffer size. 6206 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6207 // 1 full output buffer, regardless of the alignment of the available input. 6208 // The value is somewhat arbitrary, and could probably be even larger. 6209 // A larger value should allow more old data to be read after a track calls start(), 6210 // without increasing latency. 6211 mRsmpInFrames = mFrameCount * 7; 6212 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6213 delete[] mRsmpInBuffer; 6214 6215 // TODO optimize audio capture buffer sizes ... 6216 // Here we calculate the size of the sliding buffer used as a source 6217 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6218 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6219 // be better to have it derived from the pipe depth in the long term. 6220 // The current value is higher than necessary. However it should not add to latency. 6221 6222 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6223 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6224 6225 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6226 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6227} 6228 6229uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6230{ 6231 Mutex::Autolock _l(mLock); 6232 if (initCheck() != NO_ERROR) { 6233 return 0; 6234 } 6235 6236 return mInput->stream->get_input_frames_lost(mInput->stream); 6237} 6238 6239uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6240{ 6241 Mutex::Autolock _l(mLock); 6242 uint32_t result = 0; 6243 if (getEffectChain_l(sessionId) != 0) { 6244 result = EFFECT_SESSION; 6245 } 6246 6247 for (size_t i = 0; i < mTracks.size(); ++i) { 6248 if (sessionId == mTracks[i]->sessionId()) { 6249 result |= TRACK_SESSION; 6250 break; 6251 } 6252 } 6253 6254 return result; 6255} 6256 6257KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6258{ 6259 KeyedVector<int, bool> ids; 6260 Mutex::Autolock _l(mLock); 6261 for (size_t j = 0; j < mTracks.size(); ++j) { 6262 sp<RecordThread::RecordTrack> track = mTracks[j]; 6263 int sessionId = track->sessionId(); 6264 if (ids.indexOfKey(sessionId) < 0) { 6265 ids.add(sessionId, true); 6266 } 6267 } 6268 return ids; 6269} 6270 6271AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6272{ 6273 Mutex::Autolock _l(mLock); 6274 AudioStreamIn *input = mInput; 6275 mInput = NULL; 6276 return input; 6277} 6278 6279// this method must always be called either with ThreadBase mLock held or inside the thread loop 6280audio_stream_t* AudioFlinger::RecordThread::stream() const 6281{ 6282 if (mInput == NULL) { 6283 return NULL; 6284 } 6285 return &mInput->stream->common; 6286} 6287 6288status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6289{ 6290 // only one chain per input thread 6291 if (mEffectChains.size() != 0) { 6292 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6293 return INVALID_OPERATION; 6294 } 6295 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6296 chain->setThread(this); 6297 chain->setInBuffer(NULL); 6298 chain->setOutBuffer(NULL); 6299 6300 checkSuspendOnAddEffectChain_l(chain); 6301 6302 // make sure enabled pre processing effects state is communicated to the HAL as we 6303 // just moved them to a new input stream. 6304 chain->syncHalEffectsState(); 6305 6306 mEffectChains.add(chain); 6307 6308 return NO_ERROR; 6309} 6310 6311size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6312{ 6313 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6314 ALOGW_IF(mEffectChains.size() != 1, 6315 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6316 chain.get(), mEffectChains.size(), this); 6317 if (mEffectChains.size() == 1) { 6318 mEffectChains.removeAt(0); 6319 } 6320 return 0; 6321} 6322 6323status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6324 audio_patch_handle_t *handle) 6325{ 6326 status_t status = NO_ERROR; 6327 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6328 // store new device and send to effects 6329 mInDevice = patch->sources[0].ext.device.type; 6330 for (size_t i = 0; i < mEffectChains.size(); i++) { 6331 mEffectChains[i]->setDevice_l(mInDevice); 6332 } 6333 6334 // disable AEC and NS if the device is a BT SCO headset supporting those 6335 // pre processings 6336 if (mTracks.size() > 0) { 6337 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6338 mAudioFlinger->btNrecIsOff(); 6339 for (size_t i = 0; i < mTracks.size(); i++) { 6340 sp<RecordTrack> track = mTracks[i]; 6341 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6342 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6343 } 6344 } 6345 6346 // store new source and send to effects 6347 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6348 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6349 for (size_t i = 0; i < mEffectChains.size(); i++) { 6350 mEffectChains[i]->setAudioSource_l(mAudioSource); 6351 } 6352 } 6353 6354 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6355 status = hwDevice->create_audio_patch(hwDevice, 6356 patch->num_sources, 6357 patch->sources, 6358 patch->num_sinks, 6359 patch->sinks, 6360 handle); 6361 } else { 6362 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6363 } 6364 return status; 6365} 6366 6367status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6368{ 6369 status_t status = NO_ERROR; 6370 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6371 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6372 status = hwDevice->release_audio_patch(hwDevice, handle); 6373 } else { 6374 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6375 } 6376 return status; 6377} 6378 6379void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6380{ 6381 Mutex::Autolock _l(mLock); 6382 mTracks.add(record); 6383} 6384 6385void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6386{ 6387 Mutex::Autolock _l(mLock); 6388 destroyTrack_l(record); 6389} 6390 6391void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6392{ 6393 ThreadBase::getAudioPortConfig(config); 6394 config->role = AUDIO_PORT_ROLE_SINK; 6395 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6396 config->ext.mix.usecase.source = mAudioSource; 6397} 6398 6399}; // namespace android 6400