Threads.cpp revision 9d1cad2ba6a35168fa27a322518150193f19e53b
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <media/AudioResamplerPublic.h> 30#include <utils/Log.h> 31#include <utils/Trace.h> 32 33#include <private/media/AudioTrackShared.h> 34#include <hardware/audio.h> 35#include <audio_effects/effect_ns.h> 36#include <audio_effects/effect_aec.h> 37#include <audio_utils/primitives.h> 38#include <audio_utils/format.h> 39#include <audio_utils/minifloat.h> 40 41// NBAIO implementations 42#include <media/nbaio/AudioStreamInSource.h> 43#include <media/nbaio/AudioStreamOutSink.h> 44#include <media/nbaio/MonoPipe.h> 45#include <media/nbaio/MonoPipeReader.h> 46#include <media/nbaio/Pipe.h> 47#include <media/nbaio/PipeReader.h> 48#include <media/nbaio/SourceAudioBufferProvider.h> 49 50#include <powermanager/PowerManager.h> 51 52#include <common_time/cc_helper.h> 53#include <common_time/local_clock.h> 54 55#include "AudioFlinger.h" 56#include "AudioMixer.h" 57#include "FastMixer.h" 58#include "FastCapture.h" 59#include "ServiceUtilities.h" 60#include "SchedulingPolicyService.h" 61 62#ifdef ADD_BATTERY_DATA 63#include <media/IMediaPlayerService.h> 64#include <media/IMediaDeathNotifier.h> 65#endif 66 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72// ---------------------------------------------------------------------------- 73 74// Note: the following macro is used for extremely verbose logging message. In 75// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 76// 0; but one side effect of this is to turn all LOGV's as well. Some messages 77// are so verbose that we want to suppress them even when we have ALOG_ASSERT 78// turned on. Do not uncomment the #def below unless you really know what you 79// are doing and want to see all of the extremely verbose messages. 80//#define VERY_VERY_VERBOSE_LOGGING 81#ifdef VERY_VERY_VERBOSE_LOGGING 82#define ALOGVV ALOGV 83#else 84#define ALOGVV(a...) do { } while(0) 85#endif 86 87#define max(a, b) ((a) > (b) ? (a) : (b)) 88 89namespace android { 90 91// retry counts for buffer fill timeout 92// 50 * ~20msecs = 1 second 93static const int8_t kMaxTrackRetries = 50; 94static const int8_t kMaxTrackStartupRetries = 50; 95// allow less retry attempts on direct output thread. 96// direct outputs can be a scarce resource in audio hardware and should 97// be released as quickly as possible. 98static const int8_t kMaxTrackRetriesDirect = 2; 99 100// don't warn about blocked writes or record buffer overflows more often than this 101static const nsecs_t kWarningThrottleNs = seconds(5); 102 103// RecordThread loop sleep time upon application overrun or audio HAL read error 104static const int kRecordThreadSleepUs = 5000; 105 106// maximum time to wait in sendConfigEvent_l() for a status to be received 107static const nsecs_t kConfigEventTimeoutNs = seconds(2); 108 109// minimum sleep time for the mixer thread loop when tracks are active but in underrun 110static const uint32_t kMinThreadSleepTimeUs = 5000; 111// maximum divider applied to the active sleep time in the mixer thread loop 112static const uint32_t kMaxThreadSleepTimeShift = 2; 113 114// minimum normal sink buffer size, expressed in milliseconds rather than frames 115static const uint32_t kMinNormalSinkBufferSizeMs = 20; 116// maximum normal sink buffer size 117static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 118 119// Offloaded output thread standby delay: allows track transition without going to standby 120static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 121 122// Whether to use fast mixer 123static const enum { 124 FastMixer_Never, // never initialize or use: for debugging only 125 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 126 // normal mixer multiplier is 1 127 FastMixer_Static, // initialize if needed, then use all the time if initialized, 128 // multiplier is calculated based on min & max normal mixer buffer size 129 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 130 // multiplier is calculated based on min & max normal mixer buffer size 131 // FIXME for FastMixer_Dynamic: 132 // Supporting this option will require fixing HALs that can't handle large writes. 133 // For example, one HAL implementation returns an error from a large write, 134 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 135 // We could either fix the HAL implementations, or provide a wrapper that breaks 136 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 137} kUseFastMixer = FastMixer_Static; 138 139// Whether to use fast capture 140static const enum { 141 FastCapture_Never, // never initialize or use: for debugging only 142 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 143 FastCapture_Static, // initialize if needed, then use all the time if initialized 144} kUseFastCapture = FastCapture_Static; 145 146// Priorities for requestPriority 147static const int kPriorityAudioApp = 2; 148static const int kPriorityFastMixer = 3; 149static const int kPriorityFastCapture = 3; 150 151// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 152// for the track. The client then sub-divides this into smaller buffers for its use. 153// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 154// So for now we just assume that client is double-buffered for fast tracks. 155// FIXME It would be better for client to tell AudioFlinger the value of N, 156// so AudioFlinger could allocate the right amount of memory. 157// See the client's minBufCount and mNotificationFramesAct calculations for details. 158 159// This is the default value, if not specified by property. 160static const int kFastTrackMultiplier = 2; 161 162// The minimum and maximum allowed values 163static const int kFastTrackMultiplierMin = 1; 164static const int kFastTrackMultiplierMax = 2; 165 166// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 167static int sFastTrackMultiplier = kFastTrackMultiplier; 168 169// See Thread::readOnlyHeap(). 170// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 171// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 172// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 173static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 174 175// ---------------------------------------------------------------------------- 176 177static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 178 179static void sFastTrackMultiplierInit() 180{ 181 char value[PROPERTY_VALUE_MAX]; 182 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 183 char *endptr; 184 unsigned long ul = strtoul(value, &endptr, 0); 185 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 186 sFastTrackMultiplier = (int) ul; 187 } 188 } 189} 190 191// ---------------------------------------------------------------------------- 192 193#ifdef ADD_BATTERY_DATA 194// To collect the amplifier usage 195static void addBatteryData(uint32_t params) { 196 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 197 if (service == NULL) { 198 // it already logged 199 return; 200 } 201 202 service->addBatteryData(params); 203} 204#endif 205 206 207// ---------------------------------------------------------------------------- 208// CPU Stats 209// ---------------------------------------------------------------------------- 210 211class CpuStats { 212public: 213 CpuStats(); 214 void sample(const String8 &title); 215#ifdef DEBUG_CPU_USAGE 216private: 217 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 218 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 219 220 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 221 222 int mCpuNum; // thread's current CPU number 223 int mCpukHz; // frequency of thread's current CPU in kHz 224#endif 225}; 226 227CpuStats::CpuStats() 228#ifdef DEBUG_CPU_USAGE 229 : mCpuNum(-1), mCpukHz(-1) 230#endif 231{ 232} 233 234void CpuStats::sample(const String8 &title 235#ifndef DEBUG_CPU_USAGE 236 __unused 237#endif 238 ) { 239#ifdef DEBUG_CPU_USAGE 240 // get current thread's delta CPU time in wall clock ns 241 double wcNs; 242 bool valid = mCpuUsage.sampleAndEnable(wcNs); 243 244 // record sample for wall clock statistics 245 if (valid) { 246 mWcStats.sample(wcNs); 247 } 248 249 // get the current CPU number 250 int cpuNum = sched_getcpu(); 251 252 // get the current CPU frequency in kHz 253 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 254 255 // check if either CPU number or frequency changed 256 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 257 mCpuNum = cpuNum; 258 mCpukHz = cpukHz; 259 // ignore sample for purposes of cycles 260 valid = false; 261 } 262 263 // if no change in CPU number or frequency, then record sample for cycle statistics 264 if (valid && mCpukHz > 0) { 265 double cycles = wcNs * cpukHz * 0.000001; 266 mHzStats.sample(cycles); 267 } 268 269 unsigned n = mWcStats.n(); 270 // mCpuUsage.elapsed() is expensive, so don't call it every loop 271 if ((n & 127) == 1) { 272 long long elapsed = mCpuUsage.elapsed(); 273 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 274 double perLoop = elapsed / (double) n; 275 double perLoop100 = perLoop * 0.01; 276 double perLoop1k = perLoop * 0.001; 277 double mean = mWcStats.mean(); 278 double stddev = mWcStats.stddev(); 279 double minimum = mWcStats.minimum(); 280 double maximum = mWcStats.maximum(); 281 double meanCycles = mHzStats.mean(); 282 double stddevCycles = mHzStats.stddev(); 283 double minCycles = mHzStats.minimum(); 284 double maxCycles = mHzStats.maximum(); 285 mCpuUsage.resetElapsed(); 286 mWcStats.reset(); 287 mHzStats.reset(); 288 ALOGD("CPU usage for %s over past %.1f secs\n" 289 " (%u mixer loops at %.1f mean ms per loop):\n" 290 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 291 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 292 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 293 title.string(), 294 elapsed * .000000001, n, perLoop * .000001, 295 mean * .001, 296 stddev * .001, 297 minimum * .001, 298 maximum * .001, 299 mean / perLoop100, 300 stddev / perLoop100, 301 minimum / perLoop100, 302 maximum / perLoop100, 303 meanCycles / perLoop1k, 304 stddevCycles / perLoop1k, 305 minCycles / perLoop1k, 306 maxCycles / perLoop1k); 307 308 } 309 } 310#endif 311}; 312 313// ---------------------------------------------------------------------------- 314// ThreadBase 315// ---------------------------------------------------------------------------- 316 317AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 318 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 319 : Thread(false /*canCallJava*/), 320 mType(type), 321 mAudioFlinger(audioFlinger), 322 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 323 // are set by PlaybackThread::readOutputParameters_l() or 324 // RecordThread::readInputParameters_l() 325 //FIXME: mStandby should be true here. Is this some kind of hack? 326 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 327 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 328 // mName will be set by concrete (non-virtual) subclass 329 mDeathRecipient(new PMDeathRecipient(this)) 330{ 331} 332 333AudioFlinger::ThreadBase::~ThreadBase() 334{ 335 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 336 mConfigEvents.clear(); 337 338 // do not lock the mutex in destructor 339 releaseWakeLock_l(); 340 if (mPowerManager != 0) { 341 sp<IBinder> binder = mPowerManager->asBinder(); 342 binder->unlinkToDeath(mDeathRecipient); 343 } 344} 345 346status_t AudioFlinger::ThreadBase::readyToRun() 347{ 348 status_t status = initCheck(); 349 if (status == NO_ERROR) { 350 ALOGI("AudioFlinger's thread %p ready to run", this); 351 } else { 352 ALOGE("No working audio driver found."); 353 } 354 return status; 355} 356 357void AudioFlinger::ThreadBase::exit() 358{ 359 ALOGV("ThreadBase::exit"); 360 // do any cleanup required for exit to succeed 361 preExit(); 362 { 363 // This lock prevents the following race in thread (uniprocessor for illustration): 364 // if (!exitPending()) { 365 // // context switch from here to exit() 366 // // exit() calls requestExit(), what exitPending() observes 367 // // exit() calls signal(), which is dropped since no waiters 368 // // context switch back from exit() to here 369 // mWaitWorkCV.wait(...); 370 // // now thread is hung 371 // } 372 AutoMutex lock(mLock); 373 requestExit(); 374 mWaitWorkCV.broadcast(); 375 } 376 // When Thread::requestExitAndWait is made virtual and this method is renamed to 377 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 378 requestExitAndWait(); 379} 380 381status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 382{ 383 status_t status; 384 385 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 386 Mutex::Autolock _l(mLock); 387 388 return sendSetParameterConfigEvent_l(keyValuePairs); 389} 390 391// sendConfigEvent_l() must be called with ThreadBase::mLock held 392// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 393status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 394{ 395 status_t status = NO_ERROR; 396 397 mConfigEvents.add(event); 398 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 399 mWaitWorkCV.signal(); 400 mLock.unlock(); 401 { 402 Mutex::Autolock _l(event->mLock); 403 while (event->mWaitStatus) { 404 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 405 event->mStatus = TIMED_OUT; 406 event->mWaitStatus = false; 407 } 408 } 409 status = event->mStatus; 410 } 411 mLock.lock(); 412 return status; 413} 414 415void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 416{ 417 Mutex::Autolock _l(mLock); 418 sendIoConfigEvent_l(event, param); 419} 420 421// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 422void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 423{ 424 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 425 sendConfigEvent_l(configEvent); 426} 427 428// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 429void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 430{ 431 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 432 sendConfigEvent_l(configEvent); 433} 434 435// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 436status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 437{ 438 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 439 return sendConfigEvent_l(configEvent); 440} 441 442status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 443 const struct audio_patch *patch, 444 audio_patch_handle_t *handle) 445{ 446 Mutex::Autolock _l(mLock); 447 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 448 status_t status = sendConfigEvent_l(configEvent); 449 if (status == NO_ERROR) { 450 CreateAudioPatchConfigEventData *data = 451 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 452 *handle = data->mHandle; 453 } 454 return status; 455} 456 457status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 458 const audio_patch_handle_t handle) 459{ 460 Mutex::Autolock _l(mLock); 461 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 462 return sendConfigEvent_l(configEvent); 463} 464 465 466// post condition: mConfigEvents.isEmpty() 467void AudioFlinger::ThreadBase::processConfigEvents_l() 468{ 469 bool configChanged = false; 470 471 while (!mConfigEvents.isEmpty()) { 472 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 473 sp<ConfigEvent> event = mConfigEvents[0]; 474 mConfigEvents.removeAt(0); 475 switch (event->mType) { 476 case CFG_EVENT_PRIO: { 477 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 478 // FIXME Need to understand why this has to be done asynchronously 479 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 480 true /*asynchronous*/); 481 if (err != 0) { 482 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 483 data->mPrio, data->mPid, data->mTid, err); 484 } 485 } break; 486 case CFG_EVENT_IO: { 487 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 488 audioConfigChanged(data->mEvent, data->mParam); 489 } break; 490 case CFG_EVENT_SET_PARAMETER: { 491 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 492 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 493 configChanged = true; 494 } 495 } break; 496 case CFG_EVENT_CREATE_AUDIO_PATCH: { 497 CreateAudioPatchConfigEventData *data = 498 (CreateAudioPatchConfigEventData *)event->mData.get(); 499 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 500 } break; 501 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 502 ReleaseAudioPatchConfigEventData *data = 503 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 504 event->mStatus = releaseAudioPatch_l(data->mHandle); 505 } break; 506 default: 507 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 508 break; 509 } 510 { 511 Mutex::Autolock _l(event->mLock); 512 if (event->mWaitStatus) { 513 event->mWaitStatus = false; 514 event->mCond.signal(); 515 } 516 } 517 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 518 } 519 520 if (configChanged) { 521 cacheParameters_l(); 522 } 523} 524 525String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 526 String8 s; 527 if (output) { 528 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 529 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 530 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 531 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 532 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 533 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 534 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 535 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 536 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 537 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 538 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 539 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 540 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 541 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 542 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 543 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 544 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 545 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 546 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 547 } else { 548 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 549 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 550 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 551 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 552 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 553 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 554 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 555 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 556 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 557 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 558 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 559 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 560 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 561 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 562 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 563 } 564 int len = s.length(); 565 if (s.length() > 2) { 566 char *str = s.lockBuffer(len); 567 s.unlockBuffer(len - 2); 568 } 569 return s; 570} 571 572void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 573{ 574 const size_t SIZE = 256; 575 char buffer[SIZE]; 576 String8 result; 577 578 bool locked = AudioFlinger::dumpTryLock(mLock); 579 if (!locked) { 580 dprintf(fd, "thread %p maybe dead locked\n", this); 581 } 582 583 dprintf(fd, " I/O handle: %d\n", mId); 584 dprintf(fd, " TID: %d\n", getTid()); 585 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 586 dprintf(fd, " Sample rate: %u\n", mSampleRate); 587 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 588 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 589 dprintf(fd, " Channel Count: %u\n", mChannelCount); 590 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 591 channelMaskToString(mChannelMask, mType != RECORD).string()); 592 dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 593 dprintf(fd, " Frame size: %zu\n", mFrameSize); 594 dprintf(fd, " Pending config events:"); 595 size_t numConfig = mConfigEvents.size(); 596 if (numConfig) { 597 for (size_t i = 0; i < numConfig; i++) { 598 mConfigEvents[i]->dump(buffer, SIZE); 599 dprintf(fd, "\n %s", buffer); 600 } 601 dprintf(fd, "\n"); 602 } else { 603 dprintf(fd, " none\n"); 604 } 605 606 if (locked) { 607 mLock.unlock(); 608 } 609} 610 611void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 612{ 613 const size_t SIZE = 256; 614 char buffer[SIZE]; 615 String8 result; 616 617 size_t numEffectChains = mEffectChains.size(); 618 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 619 write(fd, buffer, strlen(buffer)); 620 621 for (size_t i = 0; i < numEffectChains; ++i) { 622 sp<EffectChain> chain = mEffectChains[i]; 623 if (chain != 0) { 624 chain->dump(fd, args); 625 } 626 } 627} 628 629void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 630{ 631 Mutex::Autolock _l(mLock); 632 acquireWakeLock_l(uid); 633} 634 635String16 AudioFlinger::ThreadBase::getWakeLockTag() 636{ 637 switch (mType) { 638 case MIXER: 639 return String16("AudioMix"); 640 case DIRECT: 641 return String16("AudioDirectOut"); 642 case DUPLICATING: 643 return String16("AudioDup"); 644 case RECORD: 645 return String16("AudioIn"); 646 case OFFLOAD: 647 return String16("AudioOffload"); 648 default: 649 ALOG_ASSERT(false); 650 return String16("AudioUnknown"); 651 } 652} 653 654void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 655{ 656 getPowerManager_l(); 657 if (mPowerManager != 0) { 658 sp<IBinder> binder = new BBinder(); 659 status_t status; 660 if (uid >= 0) { 661 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 662 binder, 663 getWakeLockTag(), 664 String16("media"), 665 uid); 666 } else { 667 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 668 binder, 669 getWakeLockTag(), 670 String16("media")); 671 } 672 if (status == NO_ERROR) { 673 mWakeLockToken = binder; 674 } 675 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 676 } 677} 678 679void AudioFlinger::ThreadBase::releaseWakeLock() 680{ 681 Mutex::Autolock _l(mLock); 682 releaseWakeLock_l(); 683} 684 685void AudioFlinger::ThreadBase::releaseWakeLock_l() 686{ 687 if (mWakeLockToken != 0) { 688 ALOGV("releaseWakeLock_l() %s", mName); 689 if (mPowerManager != 0) { 690 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 691 } 692 mWakeLockToken.clear(); 693 } 694} 695 696void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 697 Mutex::Autolock _l(mLock); 698 updateWakeLockUids_l(uids); 699} 700 701void AudioFlinger::ThreadBase::getPowerManager_l() { 702 703 if (mPowerManager == 0) { 704 // use checkService() to avoid blocking if power service is not up yet 705 sp<IBinder> binder = 706 defaultServiceManager()->checkService(String16("power")); 707 if (binder == 0) { 708 ALOGW("Thread %s cannot connect to the power manager service", mName); 709 } else { 710 mPowerManager = interface_cast<IPowerManager>(binder); 711 binder->linkToDeath(mDeathRecipient); 712 } 713 } 714} 715 716void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 717 718 getPowerManager_l(); 719 if (mWakeLockToken == NULL) { 720 ALOGE("no wake lock to update!"); 721 return; 722 } 723 if (mPowerManager != 0) { 724 sp<IBinder> binder = new BBinder(); 725 status_t status; 726 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 727 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 728 } 729} 730 731void AudioFlinger::ThreadBase::clearPowerManager() 732{ 733 Mutex::Autolock _l(mLock); 734 releaseWakeLock_l(); 735 mPowerManager.clear(); 736} 737 738void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 739{ 740 sp<ThreadBase> thread = mThread.promote(); 741 if (thread != 0) { 742 thread->clearPowerManager(); 743 } 744 ALOGW("power manager service died !!!"); 745} 746 747void AudioFlinger::ThreadBase::setEffectSuspended( 748 const effect_uuid_t *type, bool suspend, int sessionId) 749{ 750 Mutex::Autolock _l(mLock); 751 setEffectSuspended_l(type, suspend, sessionId); 752} 753 754void AudioFlinger::ThreadBase::setEffectSuspended_l( 755 const effect_uuid_t *type, bool suspend, int sessionId) 756{ 757 sp<EffectChain> chain = getEffectChain_l(sessionId); 758 if (chain != 0) { 759 if (type != NULL) { 760 chain->setEffectSuspended_l(type, suspend); 761 } else { 762 chain->setEffectSuspendedAll_l(suspend); 763 } 764 } 765 766 updateSuspendedSessions_l(type, suspend, sessionId); 767} 768 769void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 770{ 771 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 772 if (index < 0) { 773 return; 774 } 775 776 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 777 mSuspendedSessions.valueAt(index); 778 779 for (size_t i = 0; i < sessionEffects.size(); i++) { 780 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 781 for (int j = 0; j < desc->mRefCount; j++) { 782 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 783 chain->setEffectSuspendedAll_l(true); 784 } else { 785 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 786 desc->mType.timeLow); 787 chain->setEffectSuspended_l(&desc->mType, true); 788 } 789 } 790 } 791} 792 793void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 794 bool suspend, 795 int sessionId) 796{ 797 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 798 799 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 800 801 if (suspend) { 802 if (index >= 0) { 803 sessionEffects = mSuspendedSessions.valueAt(index); 804 } else { 805 mSuspendedSessions.add(sessionId, sessionEffects); 806 } 807 } else { 808 if (index < 0) { 809 return; 810 } 811 sessionEffects = mSuspendedSessions.valueAt(index); 812 } 813 814 815 int key = EffectChain::kKeyForSuspendAll; 816 if (type != NULL) { 817 key = type->timeLow; 818 } 819 index = sessionEffects.indexOfKey(key); 820 821 sp<SuspendedSessionDesc> desc; 822 if (suspend) { 823 if (index >= 0) { 824 desc = sessionEffects.valueAt(index); 825 } else { 826 desc = new SuspendedSessionDesc(); 827 if (type != NULL) { 828 desc->mType = *type; 829 } 830 sessionEffects.add(key, desc); 831 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 832 } 833 desc->mRefCount++; 834 } else { 835 if (index < 0) { 836 return; 837 } 838 desc = sessionEffects.valueAt(index); 839 if (--desc->mRefCount == 0) { 840 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 841 sessionEffects.removeItemsAt(index); 842 if (sessionEffects.isEmpty()) { 843 ALOGV("updateSuspendedSessions_l() restore removing session %d", 844 sessionId); 845 mSuspendedSessions.removeItem(sessionId); 846 } 847 } 848 } 849 if (!sessionEffects.isEmpty()) { 850 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 851 } 852} 853 854void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 855 bool enabled, 856 int sessionId) 857{ 858 Mutex::Autolock _l(mLock); 859 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 860} 861 862void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 863 bool enabled, 864 int sessionId) 865{ 866 if (mType != RECORD) { 867 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 868 // another session. This gives the priority to well behaved effect control panels 869 // and applications not using global effects. 870 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 871 // global effects 872 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 873 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 874 } 875 } 876 877 sp<EffectChain> chain = getEffectChain_l(sessionId); 878 if (chain != 0) { 879 chain->checkSuspendOnEffectEnabled(effect, enabled); 880 } 881} 882 883// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 884sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 885 const sp<AudioFlinger::Client>& client, 886 const sp<IEffectClient>& effectClient, 887 int32_t priority, 888 int sessionId, 889 effect_descriptor_t *desc, 890 int *enabled, 891 status_t *status) 892{ 893 sp<EffectModule> effect; 894 sp<EffectHandle> handle; 895 status_t lStatus; 896 sp<EffectChain> chain; 897 bool chainCreated = false; 898 bool effectCreated = false; 899 bool effectRegistered = false; 900 901 lStatus = initCheck(); 902 if (lStatus != NO_ERROR) { 903 ALOGW("createEffect_l() Audio driver not initialized."); 904 goto Exit; 905 } 906 907 // Reject any effect on Direct output threads for now, since the format of 908 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 909 if (mType == DIRECT) { 910 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 911 desc->name, mName); 912 lStatus = BAD_VALUE; 913 goto Exit; 914 } 915 916 // Reject any effect on mixer or duplicating multichannel sinks. 917 // TODO: fix both format and multichannel issues with effects. 918 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 919 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 920 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 921 lStatus = BAD_VALUE; 922 goto Exit; 923 } 924 925 // Allow global effects only on offloaded and mixer threads 926 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 927 switch (mType) { 928 case MIXER: 929 case OFFLOAD: 930 break; 931 case DIRECT: 932 case DUPLICATING: 933 case RECORD: 934 default: 935 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 936 lStatus = BAD_VALUE; 937 goto Exit; 938 } 939 } 940 941 // Only Pre processor effects are allowed on input threads and only on input threads 942 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 943 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 944 desc->name, desc->flags, mType); 945 lStatus = BAD_VALUE; 946 goto Exit; 947 } 948 949 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 950 951 { // scope for mLock 952 Mutex::Autolock _l(mLock); 953 954 // check for existing effect chain with the requested audio session 955 chain = getEffectChain_l(sessionId); 956 if (chain == 0) { 957 // create a new chain for this session 958 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 959 chain = new EffectChain(this, sessionId); 960 addEffectChain_l(chain); 961 chain->setStrategy(getStrategyForSession_l(sessionId)); 962 chainCreated = true; 963 } else { 964 effect = chain->getEffectFromDesc_l(desc); 965 } 966 967 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 968 969 if (effect == 0) { 970 int id = mAudioFlinger->nextUniqueId(); 971 // Check CPU and memory usage 972 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 973 if (lStatus != NO_ERROR) { 974 goto Exit; 975 } 976 effectRegistered = true; 977 // create a new effect module if none present in the chain 978 effect = new EffectModule(this, chain, desc, id, sessionId); 979 lStatus = effect->status(); 980 if (lStatus != NO_ERROR) { 981 goto Exit; 982 } 983 effect->setOffloaded(mType == OFFLOAD, mId); 984 985 lStatus = chain->addEffect_l(effect); 986 if (lStatus != NO_ERROR) { 987 goto Exit; 988 } 989 effectCreated = true; 990 991 effect->setDevice(mOutDevice); 992 effect->setDevice(mInDevice); 993 effect->setMode(mAudioFlinger->getMode()); 994 effect->setAudioSource(mAudioSource); 995 } 996 // create effect handle and connect it to effect module 997 handle = new EffectHandle(effect, client, effectClient, priority); 998 lStatus = handle->initCheck(); 999 if (lStatus == OK) { 1000 lStatus = effect->addHandle(handle.get()); 1001 } 1002 if (enabled != NULL) { 1003 *enabled = (int)effect->isEnabled(); 1004 } 1005 } 1006 1007Exit: 1008 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1009 Mutex::Autolock _l(mLock); 1010 if (effectCreated) { 1011 chain->removeEffect_l(effect); 1012 } 1013 if (effectRegistered) { 1014 AudioSystem::unregisterEffect(effect->id()); 1015 } 1016 if (chainCreated) { 1017 removeEffectChain_l(chain); 1018 } 1019 handle.clear(); 1020 } 1021 1022 *status = lStatus; 1023 return handle; 1024} 1025 1026sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1027{ 1028 Mutex::Autolock _l(mLock); 1029 return getEffect_l(sessionId, effectId); 1030} 1031 1032sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1033{ 1034 sp<EffectChain> chain = getEffectChain_l(sessionId); 1035 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1036} 1037 1038// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1039// PlaybackThread::mLock held 1040status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1041{ 1042 // check for existing effect chain with the requested audio session 1043 int sessionId = effect->sessionId(); 1044 sp<EffectChain> chain = getEffectChain_l(sessionId); 1045 bool chainCreated = false; 1046 1047 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1048 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1049 this, effect->desc().name, effect->desc().flags); 1050 1051 if (chain == 0) { 1052 // create a new chain for this session 1053 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1054 chain = new EffectChain(this, sessionId); 1055 addEffectChain_l(chain); 1056 chain->setStrategy(getStrategyForSession_l(sessionId)); 1057 chainCreated = true; 1058 } 1059 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1060 1061 if (chain->getEffectFromId_l(effect->id()) != 0) { 1062 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1063 this, effect->desc().name, chain.get()); 1064 return BAD_VALUE; 1065 } 1066 1067 effect->setOffloaded(mType == OFFLOAD, mId); 1068 1069 status_t status = chain->addEffect_l(effect); 1070 if (status != NO_ERROR) { 1071 if (chainCreated) { 1072 removeEffectChain_l(chain); 1073 } 1074 return status; 1075 } 1076 1077 effect->setDevice(mOutDevice); 1078 effect->setDevice(mInDevice); 1079 effect->setMode(mAudioFlinger->getMode()); 1080 effect->setAudioSource(mAudioSource); 1081 return NO_ERROR; 1082} 1083 1084void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1085 1086 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1087 effect_descriptor_t desc = effect->desc(); 1088 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1089 detachAuxEffect_l(effect->id()); 1090 } 1091 1092 sp<EffectChain> chain = effect->chain().promote(); 1093 if (chain != 0) { 1094 // remove effect chain if removing last effect 1095 if (chain->removeEffect_l(effect) == 0) { 1096 removeEffectChain_l(chain); 1097 } 1098 } else { 1099 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1100 } 1101} 1102 1103void AudioFlinger::ThreadBase::lockEffectChains_l( 1104 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1105{ 1106 effectChains = mEffectChains; 1107 for (size_t i = 0; i < mEffectChains.size(); i++) { 1108 mEffectChains[i]->lock(); 1109 } 1110} 1111 1112void AudioFlinger::ThreadBase::unlockEffectChains( 1113 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1114{ 1115 for (size_t i = 0; i < effectChains.size(); i++) { 1116 effectChains[i]->unlock(); 1117 } 1118} 1119 1120sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1121{ 1122 Mutex::Autolock _l(mLock); 1123 return getEffectChain_l(sessionId); 1124} 1125 1126sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1127{ 1128 size_t size = mEffectChains.size(); 1129 for (size_t i = 0; i < size; i++) { 1130 if (mEffectChains[i]->sessionId() == sessionId) { 1131 return mEffectChains[i]; 1132 } 1133 } 1134 return 0; 1135} 1136 1137void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1138{ 1139 Mutex::Autolock _l(mLock); 1140 size_t size = mEffectChains.size(); 1141 for (size_t i = 0; i < size; i++) { 1142 mEffectChains[i]->setMode_l(mode); 1143 } 1144} 1145 1146void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1147 EffectHandle *handle, 1148 bool unpinIfLast) { 1149 1150 Mutex::Autolock _l(mLock); 1151 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1152 // delete the effect module if removing last handle on it 1153 if (effect->removeHandle(handle) == 0) { 1154 if (!effect->isPinned() || unpinIfLast) { 1155 removeEffect_l(effect); 1156 AudioSystem::unregisterEffect(effect->id()); 1157 } 1158 } 1159} 1160 1161void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1162{ 1163 config->type = AUDIO_PORT_TYPE_MIX; 1164 config->ext.mix.handle = mId; 1165 config->sample_rate = mSampleRate; 1166 config->format = mFormat; 1167 config->channel_mask = mChannelMask; 1168 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1169 AUDIO_PORT_CONFIG_FORMAT; 1170} 1171 1172 1173// ---------------------------------------------------------------------------- 1174// Playback 1175// ---------------------------------------------------------------------------- 1176 1177AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1178 AudioStreamOut* output, 1179 audio_io_handle_t id, 1180 audio_devices_t device, 1181 type_t type) 1182 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1183 mNormalFrameCount(0), mSinkBuffer(NULL), 1184 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1185 mMixerBuffer(NULL), 1186 mMixerBufferSize(0), 1187 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1188 mMixerBufferValid(false), 1189 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1190 mEffectBuffer(NULL), 1191 mEffectBufferSize(0), 1192 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1193 mEffectBufferValid(false), 1194 mSuspended(0), mBytesWritten(0), 1195 mActiveTracksGeneration(0), 1196 // mStreamTypes[] initialized in constructor body 1197 mOutput(output), 1198 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1199 mMixerStatus(MIXER_IDLE), 1200 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1201 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1202 mBytesRemaining(0), 1203 mCurrentWriteLength(0), 1204 mUseAsyncWrite(false), 1205 mWriteAckSequence(0), 1206 mDrainSequence(0), 1207 mSignalPending(false), 1208 mScreenState(AudioFlinger::mScreenState), 1209 // index 0 is reserved for normal mixer's submix 1210 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1211 // mLatchD, mLatchQ, 1212 mLatchDValid(false), mLatchQValid(false) 1213{ 1214 snprintf(mName, kNameLength, "AudioOut_%X", id); 1215 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1216 1217 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1218 // it would be safer to explicitly pass initial masterVolume/masterMute as 1219 // parameter. 1220 // 1221 // If the HAL we are using has support for master volume or master mute, 1222 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1223 // and the mute set to false). 1224 mMasterVolume = audioFlinger->masterVolume_l(); 1225 mMasterMute = audioFlinger->masterMute_l(); 1226 if (mOutput && mOutput->audioHwDev) { 1227 if (mOutput->audioHwDev->canSetMasterVolume()) { 1228 mMasterVolume = 1.0; 1229 } 1230 1231 if (mOutput->audioHwDev->canSetMasterMute()) { 1232 mMasterMute = false; 1233 } 1234 } 1235 1236 readOutputParameters_l(); 1237 1238 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1239 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1240 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1241 stream = (audio_stream_type_t) (stream + 1)) { 1242 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1243 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1244 } 1245 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1246 // because mAudioFlinger doesn't have one to copy from 1247} 1248 1249AudioFlinger::PlaybackThread::~PlaybackThread() 1250{ 1251 mAudioFlinger->unregisterWriter(mNBLogWriter); 1252 free(mSinkBuffer); 1253 free(mMixerBuffer); 1254 free(mEffectBuffer); 1255} 1256 1257void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1258{ 1259 dumpInternals(fd, args); 1260 dumpTracks(fd, args); 1261 dumpEffectChains(fd, args); 1262} 1263 1264void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1265{ 1266 const size_t SIZE = 256; 1267 char buffer[SIZE]; 1268 String8 result; 1269 1270 result.appendFormat(" Stream volumes in dB: "); 1271 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1272 const stream_type_t *st = &mStreamTypes[i]; 1273 if (i > 0) { 1274 result.appendFormat(", "); 1275 } 1276 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1277 if (st->mute) { 1278 result.append("M"); 1279 } 1280 } 1281 result.append("\n"); 1282 write(fd, result.string(), result.length()); 1283 result.clear(); 1284 1285 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1286 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1287 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1288 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1289 1290 size_t numtracks = mTracks.size(); 1291 size_t numactive = mActiveTracks.size(); 1292 dprintf(fd, " %d Tracks", numtracks); 1293 size_t numactiveseen = 0; 1294 if (numtracks) { 1295 dprintf(fd, " of which %d are active\n", numactive); 1296 Track::appendDumpHeader(result); 1297 for (size_t i = 0; i < numtracks; ++i) { 1298 sp<Track> track = mTracks[i]; 1299 if (track != 0) { 1300 bool active = mActiveTracks.indexOf(track) >= 0; 1301 if (active) { 1302 numactiveseen++; 1303 } 1304 track->dump(buffer, SIZE, active); 1305 result.append(buffer); 1306 } 1307 } 1308 } else { 1309 result.append("\n"); 1310 } 1311 if (numactiveseen != numactive) { 1312 // some tracks in the active list were not in the tracks list 1313 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1314 " not in the track list\n"); 1315 result.append(buffer); 1316 Track::appendDumpHeader(result); 1317 for (size_t i = 0; i < numactive; ++i) { 1318 sp<Track> track = mActiveTracks[i].promote(); 1319 if (track != 0 && mTracks.indexOf(track) < 0) { 1320 track->dump(buffer, SIZE, true); 1321 result.append(buffer); 1322 } 1323 } 1324 } 1325 1326 write(fd, result.string(), result.size()); 1327} 1328 1329void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1330{ 1331 dprintf(fd, "\nOutput thread %p:\n", this); 1332 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1333 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1334 dprintf(fd, " Total writes: %d\n", mNumWrites); 1335 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1336 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1337 dprintf(fd, " Suspend count: %d\n", mSuspended); 1338 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1339 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1340 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1341 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1342 1343 dumpBase(fd, args); 1344} 1345 1346// Thread virtuals 1347 1348void AudioFlinger::PlaybackThread::onFirstRef() 1349{ 1350 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1351} 1352 1353// ThreadBase virtuals 1354void AudioFlinger::PlaybackThread::preExit() 1355{ 1356 ALOGV(" preExit()"); 1357 // FIXME this is using hard-coded strings but in the future, this functionality will be 1358 // converted to use audio HAL extensions required to support tunneling 1359 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1360} 1361 1362// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1363sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1364 const sp<AudioFlinger::Client>& client, 1365 audio_stream_type_t streamType, 1366 uint32_t sampleRate, 1367 audio_format_t format, 1368 audio_channel_mask_t channelMask, 1369 size_t *pFrameCount, 1370 const sp<IMemory>& sharedBuffer, 1371 int sessionId, 1372 IAudioFlinger::track_flags_t *flags, 1373 pid_t tid, 1374 int uid, 1375 status_t *status) 1376{ 1377 size_t frameCount = *pFrameCount; 1378 sp<Track> track; 1379 status_t lStatus; 1380 1381 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1382 1383 // client expresses a preference for FAST, but we get the final say 1384 if (*flags & IAudioFlinger::TRACK_FAST) { 1385 if ( 1386 // not timed 1387 (!isTimed) && 1388 // either of these use cases: 1389 ( 1390 // use case 1: shared buffer with any frame count 1391 ( 1392 (sharedBuffer != 0) 1393 ) || 1394 // use case 2: callback handler and frame count is default or at least as large as HAL 1395 ( 1396 (tid != -1) && 1397 ((frameCount == 0) || 1398 (frameCount >= mFrameCount)) 1399 ) 1400 ) && 1401 // PCM data 1402 audio_is_linear_pcm(format) && 1403 // identical channel mask to sink, or mono in and stereo sink 1404 (channelMask == mChannelMask || 1405 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1406 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1407 // hardware sample rate 1408 (sampleRate == mSampleRate) && 1409 // normal mixer has an associated fast mixer 1410 hasFastMixer() && 1411 // there are sufficient fast track slots available 1412 (mFastTrackAvailMask != 0) 1413 // FIXME test that MixerThread for this fast track has a capable output HAL 1414 // FIXME add a permission test also? 1415 ) { 1416 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1417 if (frameCount == 0) { 1418 // read the fast track multiplier property the first time it is needed 1419 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1420 if (ok != 0) { 1421 ALOGE("%s pthread_once failed: %d", __func__, ok); 1422 } 1423 frameCount = mFrameCount * sFastTrackMultiplier; 1424 } 1425 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1426 frameCount, mFrameCount); 1427 } else { 1428 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1429 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1430 "sampleRate=%u mSampleRate=%u " 1431 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1432 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1433 audio_is_linear_pcm(format), 1434 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1435 *flags &= ~IAudioFlinger::TRACK_FAST; 1436 // For compatibility with AudioTrack calculation, buffer depth is forced 1437 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1438 // This is probably too conservative, but legacy application code may depend on it. 1439 // If you change this calculation, also review the start threshold which is related. 1440 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1441 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1442 if (minBufCount < 2) { 1443 minBufCount = 2; 1444 } 1445 size_t minFrameCount = mNormalFrameCount * minBufCount; 1446 if (frameCount < minFrameCount) { 1447 frameCount = minFrameCount; 1448 } 1449 } 1450 } 1451 *pFrameCount = frameCount; 1452 1453 switch (mType) { 1454 1455 case DIRECT: 1456 if (audio_is_linear_pcm(format)) { 1457 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1458 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1459 "for output %p with format %#x", 1460 sampleRate, format, channelMask, mOutput, mFormat); 1461 lStatus = BAD_VALUE; 1462 goto Exit; 1463 } 1464 } 1465 break; 1466 1467 case OFFLOAD: 1468 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1469 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1470 "for output %p with format %#x", 1471 sampleRate, format, channelMask, mOutput, mFormat); 1472 lStatus = BAD_VALUE; 1473 goto Exit; 1474 } 1475 break; 1476 1477 default: 1478 if (!audio_is_linear_pcm(format)) { 1479 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1480 "for output %p with format %#x", 1481 format, mOutput, mFormat); 1482 lStatus = BAD_VALUE; 1483 goto Exit; 1484 } 1485 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1486 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1487 lStatus = BAD_VALUE; 1488 goto Exit; 1489 } 1490 break; 1491 1492 } 1493 1494 lStatus = initCheck(); 1495 if (lStatus != NO_ERROR) { 1496 ALOGE("createTrack_l() audio driver not initialized"); 1497 goto Exit; 1498 } 1499 1500 { // scope for mLock 1501 Mutex::Autolock _l(mLock); 1502 1503 // all tracks in same audio session must share the same routing strategy otherwise 1504 // conflicts will happen when tracks are moved from one output to another by audio policy 1505 // manager 1506 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1507 for (size_t i = 0; i < mTracks.size(); ++i) { 1508 sp<Track> t = mTracks[i]; 1509 if (t != 0 && t->isExternalTrack()) { 1510 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1511 if (sessionId == t->sessionId() && strategy != actual) { 1512 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1513 strategy, actual); 1514 lStatus = BAD_VALUE; 1515 goto Exit; 1516 } 1517 } 1518 } 1519 1520 if (!isTimed) { 1521 track = new Track(this, client, streamType, sampleRate, format, 1522 channelMask, frameCount, NULL, sharedBuffer, 1523 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1524 } else { 1525 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1526 channelMask, frameCount, sharedBuffer, sessionId, uid); 1527 } 1528 1529 // new Track always returns non-NULL, 1530 // but TimedTrack::create() is a factory that could fail by returning NULL 1531 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1532 if (lStatus != NO_ERROR) { 1533 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1534 // track must be cleared from the caller as the caller has the AF lock 1535 goto Exit; 1536 } 1537 mTracks.add(track); 1538 1539 sp<EffectChain> chain = getEffectChain_l(sessionId); 1540 if (chain != 0) { 1541 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1542 track->setMainBuffer(chain->inBuffer()); 1543 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1544 chain->incTrackCnt(); 1545 } 1546 1547 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1548 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1549 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1550 // so ask activity manager to do this on our behalf 1551 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1552 } 1553 } 1554 1555 lStatus = NO_ERROR; 1556 1557Exit: 1558 *status = lStatus; 1559 return track; 1560} 1561 1562uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1563{ 1564 return latency; 1565} 1566 1567uint32_t AudioFlinger::PlaybackThread::latency() const 1568{ 1569 Mutex::Autolock _l(mLock); 1570 return latency_l(); 1571} 1572uint32_t AudioFlinger::PlaybackThread::latency_l() const 1573{ 1574 if (initCheck() == NO_ERROR) { 1575 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1576 } else { 1577 return 0; 1578 } 1579} 1580 1581void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1582{ 1583 Mutex::Autolock _l(mLock); 1584 // Don't apply master volume in SW if our HAL can do it for us. 1585 if (mOutput && mOutput->audioHwDev && 1586 mOutput->audioHwDev->canSetMasterVolume()) { 1587 mMasterVolume = 1.0; 1588 } else { 1589 mMasterVolume = value; 1590 } 1591} 1592 1593void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1594{ 1595 Mutex::Autolock _l(mLock); 1596 // Don't apply master mute in SW if our HAL can do it for us. 1597 if (mOutput && mOutput->audioHwDev && 1598 mOutput->audioHwDev->canSetMasterMute()) { 1599 mMasterMute = false; 1600 } else { 1601 mMasterMute = muted; 1602 } 1603} 1604 1605void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1606{ 1607 Mutex::Autolock _l(mLock); 1608 mStreamTypes[stream].volume = value; 1609 broadcast_l(); 1610} 1611 1612void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1613{ 1614 Mutex::Autolock _l(mLock); 1615 mStreamTypes[stream].mute = muted; 1616 broadcast_l(); 1617} 1618 1619float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1620{ 1621 Mutex::Autolock _l(mLock); 1622 return mStreamTypes[stream].volume; 1623} 1624 1625// addTrack_l() must be called with ThreadBase::mLock held 1626status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1627{ 1628 status_t status = ALREADY_EXISTS; 1629 1630 // set retry count for buffer fill 1631 track->mRetryCount = kMaxTrackStartupRetries; 1632 if (mActiveTracks.indexOf(track) < 0) { 1633 // the track is newly added, make sure it fills up all its 1634 // buffers before playing. This is to ensure the client will 1635 // effectively get the latency it requested. 1636 if (track->isExternalTrack()) { 1637 TrackBase::track_state state = track->mState; 1638 mLock.unlock(); 1639 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1640 mLock.lock(); 1641 // abort track was stopped/paused while we released the lock 1642 if (state != track->mState) { 1643 if (status == NO_ERROR) { 1644 mLock.unlock(); 1645 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1646 mLock.lock(); 1647 } 1648 return INVALID_OPERATION; 1649 } 1650 // abort if start is rejected by audio policy manager 1651 if (status != NO_ERROR) { 1652 return PERMISSION_DENIED; 1653 } 1654#ifdef ADD_BATTERY_DATA 1655 // to track the speaker usage 1656 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1657#endif 1658 } 1659 1660 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1661 track->mResetDone = false; 1662 track->mPresentationCompleteFrames = 0; 1663 mActiveTracks.add(track); 1664 mWakeLockUids.add(track->uid()); 1665 mActiveTracksGeneration++; 1666 mLatestActiveTrack = track; 1667 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1668 if (chain != 0) { 1669 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1670 track->sessionId()); 1671 chain->incActiveTrackCnt(); 1672 } 1673 1674 status = NO_ERROR; 1675 } 1676 1677 onAddNewTrack_l(); 1678 return status; 1679} 1680 1681bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1682{ 1683 track->terminate(); 1684 // active tracks are removed by threadLoop() 1685 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1686 track->mState = TrackBase::STOPPED; 1687 if (!trackActive) { 1688 removeTrack_l(track); 1689 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1690 track->mState = TrackBase::STOPPING_1; 1691 } 1692 1693 return trackActive; 1694} 1695 1696void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1697{ 1698 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1699 mTracks.remove(track); 1700 deleteTrackName_l(track->name()); 1701 // redundant as track is about to be destroyed, for dumpsys only 1702 track->mName = -1; 1703 if (track->isFastTrack()) { 1704 int index = track->mFastIndex; 1705 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1706 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1707 mFastTrackAvailMask |= 1 << index; 1708 // redundant as track is about to be destroyed, for dumpsys only 1709 track->mFastIndex = -1; 1710 } 1711 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1712 if (chain != 0) { 1713 chain->decTrackCnt(); 1714 } 1715} 1716 1717void AudioFlinger::PlaybackThread::broadcast_l() 1718{ 1719 // Thread could be blocked waiting for async 1720 // so signal it to handle state changes immediately 1721 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1722 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1723 mSignalPending = true; 1724 mWaitWorkCV.broadcast(); 1725} 1726 1727String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1728{ 1729 Mutex::Autolock _l(mLock); 1730 if (initCheck() != NO_ERROR) { 1731 return String8(); 1732 } 1733 1734 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1735 const String8 out_s8(s); 1736 free(s); 1737 return out_s8; 1738} 1739 1740void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1741 AudioSystem::OutputDescriptor desc; 1742 void *param2 = NULL; 1743 1744 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1745 param); 1746 1747 switch (event) { 1748 case AudioSystem::OUTPUT_OPENED: 1749 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1750 desc.channelMask = mChannelMask; 1751 desc.samplingRate = mSampleRate; 1752 desc.format = mFormat; 1753 desc.frameCount = mNormalFrameCount; // FIXME see 1754 // AudioFlinger::frameCount(audio_io_handle_t) 1755 desc.latency = latency_l(); 1756 param2 = &desc; 1757 break; 1758 1759 case AudioSystem::STREAM_CONFIG_CHANGED: 1760 param2 = ¶m; 1761 case AudioSystem::OUTPUT_CLOSED: 1762 default: 1763 break; 1764 } 1765 mAudioFlinger->audioConfigChanged(event, mId, param2); 1766} 1767 1768void AudioFlinger::PlaybackThread::writeCallback() 1769{ 1770 ALOG_ASSERT(mCallbackThread != 0); 1771 mCallbackThread->resetWriteBlocked(); 1772} 1773 1774void AudioFlinger::PlaybackThread::drainCallback() 1775{ 1776 ALOG_ASSERT(mCallbackThread != 0); 1777 mCallbackThread->resetDraining(); 1778} 1779 1780void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1781{ 1782 Mutex::Autolock _l(mLock); 1783 // reject out of sequence requests 1784 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1785 mWriteAckSequence &= ~1; 1786 mWaitWorkCV.signal(); 1787 } 1788} 1789 1790void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1791{ 1792 Mutex::Autolock _l(mLock); 1793 // reject out of sequence requests 1794 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1795 mDrainSequence &= ~1; 1796 mWaitWorkCV.signal(); 1797 } 1798} 1799 1800// static 1801int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1802 void *param __unused, 1803 void *cookie) 1804{ 1805 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1806 ALOGV("asyncCallback() event %d", event); 1807 switch (event) { 1808 case STREAM_CBK_EVENT_WRITE_READY: 1809 me->writeCallback(); 1810 break; 1811 case STREAM_CBK_EVENT_DRAIN_READY: 1812 me->drainCallback(); 1813 break; 1814 default: 1815 ALOGW("asyncCallback() unknown event %d", event); 1816 break; 1817 } 1818 return 0; 1819} 1820 1821void AudioFlinger::PlaybackThread::readOutputParameters_l() 1822{ 1823 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1824 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1825 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1826 if (!audio_is_output_channel(mChannelMask)) { 1827 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1828 } 1829 if ((mType == MIXER || mType == DUPLICATING) 1830 && !isValidPcmSinkChannelMask(mChannelMask)) { 1831 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1832 mChannelMask); 1833 } 1834 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1835 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1836 mFormat = mHALFormat; 1837 if (!audio_is_valid_format(mFormat)) { 1838 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1839 } 1840 if ((mType == MIXER || mType == DUPLICATING) 1841 && !isValidPcmSinkFormat(mFormat)) { 1842 LOG_FATAL("HAL format %#x not supported for mixed output", 1843 mFormat); 1844 } 1845 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1846 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1847 mFrameCount = mBufferSize / mFrameSize; 1848 if (mFrameCount & 15) { 1849 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1850 mFrameCount); 1851 } 1852 1853 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1854 (mOutput->stream->set_callback != NULL)) { 1855 if (mOutput->stream->set_callback(mOutput->stream, 1856 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1857 mUseAsyncWrite = true; 1858 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1859 } 1860 } 1861 1862 // Calculate size of normal sink buffer relative to the HAL output buffer size 1863 double multiplier = 1.0; 1864 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1865 kUseFastMixer == FastMixer_Dynamic)) { 1866 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1867 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1868 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1869 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1870 maxNormalFrameCount = maxNormalFrameCount & ~15; 1871 if (maxNormalFrameCount < minNormalFrameCount) { 1872 maxNormalFrameCount = minNormalFrameCount; 1873 } 1874 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1875 if (multiplier <= 1.0) { 1876 multiplier = 1.0; 1877 } else if (multiplier <= 2.0) { 1878 if (2 * mFrameCount <= maxNormalFrameCount) { 1879 multiplier = 2.0; 1880 } else { 1881 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1882 } 1883 } else { 1884 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1885 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1886 // track, but we sometimes have to do this to satisfy the maximum frame count 1887 // constraint) 1888 // FIXME this rounding up should not be done if no HAL SRC 1889 uint32_t truncMult = (uint32_t) multiplier; 1890 if ((truncMult & 1)) { 1891 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1892 ++truncMult; 1893 } 1894 } 1895 multiplier = (double) truncMult; 1896 } 1897 } 1898 mNormalFrameCount = multiplier * mFrameCount; 1899 // round up to nearest 16 frames to satisfy AudioMixer 1900 if (mType == MIXER || mType == DUPLICATING) { 1901 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1902 } 1903 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1904 mNormalFrameCount); 1905 1906 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1907 // Originally this was int16_t[] array, need to remove legacy implications. 1908 free(mSinkBuffer); 1909 mSinkBuffer = NULL; 1910 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1911 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1912 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1913 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1914 1915 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1916 // drives the output. 1917 free(mMixerBuffer); 1918 mMixerBuffer = NULL; 1919 if (mMixerBufferEnabled) { 1920 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1921 mMixerBufferSize = mNormalFrameCount * mChannelCount 1922 * audio_bytes_per_sample(mMixerBufferFormat); 1923 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1924 } 1925 free(mEffectBuffer); 1926 mEffectBuffer = NULL; 1927 if (mEffectBufferEnabled) { 1928 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1929 mEffectBufferSize = mNormalFrameCount * mChannelCount 1930 * audio_bytes_per_sample(mEffectBufferFormat); 1931 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1932 } 1933 1934 // force reconfiguration of effect chains and engines to take new buffer size and audio 1935 // parameters into account 1936 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1937 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1938 // matter. 1939 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1940 Vector< sp<EffectChain> > effectChains = mEffectChains; 1941 for (size_t i = 0; i < effectChains.size(); i ++) { 1942 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1943 } 1944} 1945 1946 1947status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1948{ 1949 if (halFrames == NULL || dspFrames == NULL) { 1950 return BAD_VALUE; 1951 } 1952 Mutex::Autolock _l(mLock); 1953 if (initCheck() != NO_ERROR) { 1954 return INVALID_OPERATION; 1955 } 1956 size_t framesWritten = mBytesWritten / mFrameSize; 1957 *halFrames = framesWritten; 1958 1959 if (isSuspended()) { 1960 // return an estimation of rendered frames when the output is suspended 1961 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1962 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1963 return NO_ERROR; 1964 } else { 1965 status_t status; 1966 uint32_t frames; 1967 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1968 *dspFrames = (size_t)frames; 1969 return status; 1970 } 1971} 1972 1973uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1974{ 1975 Mutex::Autolock _l(mLock); 1976 uint32_t result = 0; 1977 if (getEffectChain_l(sessionId) != 0) { 1978 result = EFFECT_SESSION; 1979 } 1980 1981 for (size_t i = 0; i < mTracks.size(); ++i) { 1982 sp<Track> track = mTracks[i]; 1983 if (sessionId == track->sessionId() && !track->isInvalid()) { 1984 result |= TRACK_SESSION; 1985 break; 1986 } 1987 } 1988 1989 return result; 1990} 1991 1992uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1993{ 1994 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1995 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1996 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1997 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1998 } 1999 for (size_t i = 0; i < mTracks.size(); i++) { 2000 sp<Track> track = mTracks[i]; 2001 if (sessionId == track->sessionId() && !track->isInvalid()) { 2002 return AudioSystem::getStrategyForStream(track->streamType()); 2003 } 2004 } 2005 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2006} 2007 2008 2009AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2010{ 2011 Mutex::Autolock _l(mLock); 2012 return mOutput; 2013} 2014 2015AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2016{ 2017 Mutex::Autolock _l(mLock); 2018 AudioStreamOut *output = mOutput; 2019 mOutput = NULL; 2020 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2021 // must push a NULL and wait for ack 2022 mOutputSink.clear(); 2023 mPipeSink.clear(); 2024 mNormalSink.clear(); 2025 return output; 2026} 2027 2028// this method must always be called either with ThreadBase mLock held or inside the thread loop 2029audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2030{ 2031 if (mOutput == NULL) { 2032 return NULL; 2033 } 2034 return &mOutput->stream->common; 2035} 2036 2037uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2038{ 2039 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2040} 2041 2042status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2043{ 2044 if (!isValidSyncEvent(event)) { 2045 return BAD_VALUE; 2046 } 2047 2048 Mutex::Autolock _l(mLock); 2049 2050 for (size_t i = 0; i < mTracks.size(); ++i) { 2051 sp<Track> track = mTracks[i]; 2052 if (event->triggerSession() == track->sessionId()) { 2053 (void) track->setSyncEvent(event); 2054 return NO_ERROR; 2055 } 2056 } 2057 2058 return NAME_NOT_FOUND; 2059} 2060 2061bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2062{ 2063 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2064} 2065 2066void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2067 const Vector< sp<Track> >& tracksToRemove) 2068{ 2069 size_t count = tracksToRemove.size(); 2070 if (count > 0) { 2071 for (size_t i = 0 ; i < count ; i++) { 2072 const sp<Track>& track = tracksToRemove.itemAt(i); 2073 if (track->isExternalTrack()) { 2074 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2075#ifdef ADD_BATTERY_DATA 2076 // to track the speaker usage 2077 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2078#endif 2079 if (track->isTerminated()) { 2080 AudioSystem::releaseOutput(mId); 2081 } 2082 } 2083 } 2084 } 2085} 2086 2087void AudioFlinger::PlaybackThread::checkSilentMode_l() 2088{ 2089 if (!mMasterMute) { 2090 char value[PROPERTY_VALUE_MAX]; 2091 if (property_get("ro.audio.silent", value, "0") > 0) { 2092 char *endptr; 2093 unsigned long ul = strtoul(value, &endptr, 0); 2094 if (*endptr == '\0' && ul != 0) { 2095 ALOGD("Silence is golden"); 2096 // The setprop command will not allow a property to be changed after 2097 // the first time it is set, so we don't have to worry about un-muting. 2098 setMasterMute_l(true); 2099 } 2100 } 2101 } 2102} 2103 2104// shared by MIXER and DIRECT, overridden by DUPLICATING 2105ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2106{ 2107 // FIXME rewrite to reduce number of system calls 2108 mLastWriteTime = systemTime(); 2109 mInWrite = true; 2110 ssize_t bytesWritten; 2111 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2112 2113 // If an NBAIO sink is present, use it to write the normal mixer's submix 2114 if (mNormalSink != 0) { 2115 const size_t count = mBytesRemaining / mFrameSize; 2116 2117 ATRACE_BEGIN("write"); 2118 // update the setpoint when AudioFlinger::mScreenState changes 2119 uint32_t screenState = AudioFlinger::mScreenState; 2120 if (screenState != mScreenState) { 2121 mScreenState = screenState; 2122 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2123 if (pipe != NULL) { 2124 pipe->setAvgFrames((mScreenState & 1) ? 2125 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2126 } 2127 } 2128 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2129 ATRACE_END(); 2130 if (framesWritten > 0) { 2131 bytesWritten = framesWritten * mFrameSize; 2132 } else { 2133 bytesWritten = framesWritten; 2134 } 2135 mLatchDValid = false; 2136 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2137 if (status == NO_ERROR) { 2138 size_t totalFramesWritten = mNormalSink->framesWritten(); 2139 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2140 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2141 mLatchDValid = true; 2142 } 2143 } 2144 // otherwise use the HAL / AudioStreamOut directly 2145 } else { 2146 // Direct output and offload threads 2147 2148 if (mUseAsyncWrite) { 2149 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2150 mWriteAckSequence += 2; 2151 mWriteAckSequence |= 1; 2152 ALOG_ASSERT(mCallbackThread != 0); 2153 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2154 } 2155 // FIXME We should have an implementation of timestamps for direct output threads. 2156 // They are used e.g for multichannel PCM playback over HDMI. 2157 bytesWritten = mOutput->stream->write(mOutput->stream, 2158 (char *)mSinkBuffer + offset, mBytesRemaining); 2159 if (mUseAsyncWrite && 2160 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2161 // do not wait for async callback in case of error of full write 2162 mWriteAckSequence &= ~1; 2163 ALOG_ASSERT(mCallbackThread != 0); 2164 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2165 } 2166 } 2167 2168 mNumWrites++; 2169 mInWrite = false; 2170 mStandby = false; 2171 return bytesWritten; 2172} 2173 2174void AudioFlinger::PlaybackThread::threadLoop_drain() 2175{ 2176 if (mOutput->stream->drain) { 2177 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2178 if (mUseAsyncWrite) { 2179 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2180 mDrainSequence |= 1; 2181 ALOG_ASSERT(mCallbackThread != 0); 2182 mCallbackThread->setDraining(mDrainSequence); 2183 } 2184 mOutput->stream->drain(mOutput->stream, 2185 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2186 : AUDIO_DRAIN_ALL); 2187 } 2188} 2189 2190void AudioFlinger::PlaybackThread::threadLoop_exit() 2191{ 2192 // Default implementation has nothing to do 2193} 2194 2195/* 2196The derived values that are cached: 2197 - mSinkBufferSize from frame count * frame size 2198 - activeSleepTime from activeSleepTimeUs() 2199 - idleSleepTime from idleSleepTimeUs() 2200 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2201 - maxPeriod from frame count and sample rate (MIXER only) 2202 2203The parameters that affect these derived values are: 2204 - frame count 2205 - frame size 2206 - sample rate 2207 - device type: A2DP or not 2208 - device latency 2209 - format: PCM or not 2210 - active sleep time 2211 - idle sleep time 2212*/ 2213 2214void AudioFlinger::PlaybackThread::cacheParameters_l() 2215{ 2216 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2217 activeSleepTime = activeSleepTimeUs(); 2218 idleSleepTime = idleSleepTimeUs(); 2219} 2220 2221void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2222{ 2223 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2224 this, streamType, mTracks.size()); 2225 Mutex::Autolock _l(mLock); 2226 2227 size_t size = mTracks.size(); 2228 for (size_t i = 0; i < size; i++) { 2229 sp<Track> t = mTracks[i]; 2230 if (t->streamType() == streamType) { 2231 t->invalidate(); 2232 } 2233 } 2234} 2235 2236status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2237{ 2238 int session = chain->sessionId(); 2239 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2240 ? mEffectBuffer : mSinkBuffer); 2241 bool ownsBuffer = false; 2242 2243 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2244 if (session > 0) { 2245 // Only one effect chain can be present in direct output thread and it uses 2246 // the sink buffer as input 2247 if (mType != DIRECT) { 2248 size_t numSamples = mNormalFrameCount * mChannelCount; 2249 buffer = new int16_t[numSamples]; 2250 memset(buffer, 0, numSamples * sizeof(int16_t)); 2251 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2252 ownsBuffer = true; 2253 } 2254 2255 // Attach all tracks with same session ID to this chain. 2256 for (size_t i = 0; i < mTracks.size(); ++i) { 2257 sp<Track> track = mTracks[i]; 2258 if (session == track->sessionId()) { 2259 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2260 buffer); 2261 track->setMainBuffer(buffer); 2262 chain->incTrackCnt(); 2263 } 2264 } 2265 2266 // indicate all active tracks in the chain 2267 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2268 sp<Track> track = mActiveTracks[i].promote(); 2269 if (track == 0) { 2270 continue; 2271 } 2272 if (session == track->sessionId()) { 2273 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2274 chain->incActiveTrackCnt(); 2275 } 2276 } 2277 } 2278 2279 chain->setInBuffer(buffer, ownsBuffer); 2280 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2281 ? mEffectBuffer : mSinkBuffer)); 2282 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2283 // chains list in order to be processed last as it contains output stage effects 2284 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2285 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2286 // after track specific effects and before output stage 2287 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2288 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2289 // Effect chain for other sessions are inserted at beginning of effect 2290 // chains list to be processed before output mix effects. Relative order between other 2291 // sessions is not important 2292 size_t size = mEffectChains.size(); 2293 size_t i = 0; 2294 for (i = 0; i < size; i++) { 2295 if (mEffectChains[i]->sessionId() < session) { 2296 break; 2297 } 2298 } 2299 mEffectChains.insertAt(chain, i); 2300 checkSuspendOnAddEffectChain_l(chain); 2301 2302 return NO_ERROR; 2303} 2304 2305size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2306{ 2307 int session = chain->sessionId(); 2308 2309 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2310 2311 for (size_t i = 0; i < mEffectChains.size(); i++) { 2312 if (chain == mEffectChains[i]) { 2313 mEffectChains.removeAt(i); 2314 // detach all active tracks from the chain 2315 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2316 sp<Track> track = mActiveTracks[i].promote(); 2317 if (track == 0) { 2318 continue; 2319 } 2320 if (session == track->sessionId()) { 2321 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2322 chain.get(), session); 2323 chain->decActiveTrackCnt(); 2324 } 2325 } 2326 2327 // detach all tracks with same session ID from this chain 2328 for (size_t i = 0; i < mTracks.size(); ++i) { 2329 sp<Track> track = mTracks[i]; 2330 if (session == track->sessionId()) { 2331 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2332 chain->decTrackCnt(); 2333 } 2334 } 2335 break; 2336 } 2337 } 2338 return mEffectChains.size(); 2339} 2340 2341status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2342 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2343{ 2344 Mutex::Autolock _l(mLock); 2345 return attachAuxEffect_l(track, EffectId); 2346} 2347 2348status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2349 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2350{ 2351 status_t status = NO_ERROR; 2352 2353 if (EffectId == 0) { 2354 track->setAuxBuffer(0, NULL); 2355 } else { 2356 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2357 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2358 if (effect != 0) { 2359 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2360 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2361 } else { 2362 status = INVALID_OPERATION; 2363 } 2364 } else { 2365 status = BAD_VALUE; 2366 } 2367 } 2368 return status; 2369} 2370 2371void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2372{ 2373 for (size_t i = 0; i < mTracks.size(); ++i) { 2374 sp<Track> track = mTracks[i]; 2375 if (track->auxEffectId() == effectId) { 2376 attachAuxEffect_l(track, 0); 2377 } 2378 } 2379} 2380 2381bool AudioFlinger::PlaybackThread::threadLoop() 2382{ 2383 Vector< sp<Track> > tracksToRemove; 2384 2385 standbyTime = systemTime(); 2386 2387 // MIXER 2388 nsecs_t lastWarning = 0; 2389 2390 // DUPLICATING 2391 // FIXME could this be made local to while loop? 2392 writeFrames = 0; 2393 2394 int lastGeneration = 0; 2395 2396 cacheParameters_l(); 2397 sleepTime = idleSleepTime; 2398 2399 if (mType == MIXER) { 2400 sleepTimeShift = 0; 2401 } 2402 2403 CpuStats cpuStats; 2404 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2405 2406 acquireWakeLock(); 2407 2408 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2409 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2410 // and then that string will be logged at the next convenient opportunity. 2411 const char *logString = NULL; 2412 2413 checkSilentMode_l(); 2414 2415 while (!exitPending()) 2416 { 2417 cpuStats.sample(myName); 2418 2419 Vector< sp<EffectChain> > effectChains; 2420 2421 { // scope for mLock 2422 2423 Mutex::Autolock _l(mLock); 2424 2425 processConfigEvents_l(); 2426 2427 if (logString != NULL) { 2428 mNBLogWriter->logTimestamp(); 2429 mNBLogWriter->log(logString); 2430 logString = NULL; 2431 } 2432 2433 if (mLatchDValid) { 2434 mLatchQ = mLatchD; 2435 mLatchDValid = false; 2436 mLatchQValid = true; 2437 } 2438 2439 saveOutputTracks(); 2440 if (mSignalPending) { 2441 // A signal was raised while we were unlocked 2442 mSignalPending = false; 2443 } else if (waitingAsyncCallback_l()) { 2444 if (exitPending()) { 2445 break; 2446 } 2447 releaseWakeLock_l(); 2448 mWakeLockUids.clear(); 2449 mActiveTracksGeneration++; 2450 ALOGV("wait async completion"); 2451 mWaitWorkCV.wait(mLock); 2452 ALOGV("async completion/wake"); 2453 acquireWakeLock_l(); 2454 standbyTime = systemTime() + standbyDelay; 2455 sleepTime = 0; 2456 2457 continue; 2458 } 2459 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2460 isSuspended()) { 2461 // put audio hardware into standby after short delay 2462 if (shouldStandby_l()) { 2463 2464 threadLoop_standby(); 2465 2466 mStandby = true; 2467 } 2468 2469 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2470 // we're about to wait, flush the binder command buffer 2471 IPCThreadState::self()->flushCommands(); 2472 2473 clearOutputTracks(); 2474 2475 if (exitPending()) { 2476 break; 2477 } 2478 2479 releaseWakeLock_l(); 2480 mWakeLockUids.clear(); 2481 mActiveTracksGeneration++; 2482 // wait until we have something to do... 2483 ALOGV("%s going to sleep", myName.string()); 2484 mWaitWorkCV.wait(mLock); 2485 ALOGV("%s waking up", myName.string()); 2486 acquireWakeLock_l(); 2487 2488 mMixerStatus = MIXER_IDLE; 2489 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2490 mBytesWritten = 0; 2491 mBytesRemaining = 0; 2492 checkSilentMode_l(); 2493 2494 standbyTime = systemTime() + standbyDelay; 2495 sleepTime = idleSleepTime; 2496 if (mType == MIXER) { 2497 sleepTimeShift = 0; 2498 } 2499 2500 continue; 2501 } 2502 } 2503 // mMixerStatusIgnoringFastTracks is also updated internally 2504 mMixerStatus = prepareTracks_l(&tracksToRemove); 2505 2506 // compare with previously applied list 2507 if (lastGeneration != mActiveTracksGeneration) { 2508 // update wakelock 2509 updateWakeLockUids_l(mWakeLockUids); 2510 lastGeneration = mActiveTracksGeneration; 2511 } 2512 2513 // prevent any changes in effect chain list and in each effect chain 2514 // during mixing and effect process as the audio buffers could be deleted 2515 // or modified if an effect is created or deleted 2516 lockEffectChains_l(effectChains); 2517 } // mLock scope ends 2518 2519 if (mBytesRemaining == 0) { 2520 mCurrentWriteLength = 0; 2521 if (mMixerStatus == MIXER_TRACKS_READY) { 2522 // threadLoop_mix() sets mCurrentWriteLength 2523 threadLoop_mix(); 2524 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2525 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2526 // threadLoop_sleepTime sets sleepTime to 0 if data 2527 // must be written to HAL 2528 threadLoop_sleepTime(); 2529 if (sleepTime == 0) { 2530 mCurrentWriteLength = mSinkBufferSize; 2531 } 2532 } 2533 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2534 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2535 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2536 // or mSinkBuffer (if there are no effects). 2537 // 2538 // This is done pre-effects computation; if effects change to 2539 // support higher precision, this needs to move. 2540 // 2541 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2542 // TODO use sleepTime == 0 as an additional condition. 2543 if (mMixerBufferValid) { 2544 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2545 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2546 2547 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2548 mNormalFrameCount * mChannelCount); 2549 } 2550 2551 mBytesRemaining = mCurrentWriteLength; 2552 if (isSuspended()) { 2553 sleepTime = suspendSleepTimeUs(); 2554 // simulate write to HAL when suspended 2555 mBytesWritten += mSinkBufferSize; 2556 mBytesRemaining = 0; 2557 } 2558 2559 // only process effects if we're going to write 2560 if (sleepTime == 0 && mType != OFFLOAD) { 2561 for (size_t i = 0; i < effectChains.size(); i ++) { 2562 effectChains[i]->process_l(); 2563 } 2564 } 2565 } 2566 // Process effect chains for offloaded thread even if no audio 2567 // was read from audio track: process only updates effect state 2568 // and thus does have to be synchronized with audio writes but may have 2569 // to be called while waiting for async write callback 2570 if (mType == OFFLOAD) { 2571 for (size_t i = 0; i < effectChains.size(); i ++) { 2572 effectChains[i]->process_l(); 2573 } 2574 } 2575 2576 // Only if the Effects buffer is enabled and there is data in the 2577 // Effects buffer (buffer valid), we need to 2578 // copy into the sink buffer. 2579 // TODO use sleepTime == 0 as an additional condition. 2580 if (mEffectBufferValid) { 2581 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2582 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2583 mNormalFrameCount * mChannelCount); 2584 } 2585 2586 // enable changes in effect chain 2587 unlockEffectChains(effectChains); 2588 2589 if (!waitingAsyncCallback()) { 2590 // sleepTime == 0 means we must write to audio hardware 2591 if (sleepTime == 0) { 2592 if (mBytesRemaining) { 2593 ssize_t ret = threadLoop_write(); 2594 if (ret < 0) { 2595 mBytesRemaining = 0; 2596 } else { 2597 mBytesWritten += ret; 2598 mBytesRemaining -= ret; 2599 } 2600 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2601 (mMixerStatus == MIXER_DRAIN_ALL)) { 2602 threadLoop_drain(); 2603 } 2604 if (mType == MIXER) { 2605 // write blocked detection 2606 nsecs_t now = systemTime(); 2607 nsecs_t delta = now - mLastWriteTime; 2608 if (!mStandby && delta > maxPeriod) { 2609 mNumDelayedWrites++; 2610 if ((now - lastWarning) > kWarningThrottleNs) { 2611 ATRACE_NAME("underrun"); 2612 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2613 ns2ms(delta), mNumDelayedWrites, this); 2614 lastWarning = now; 2615 } 2616 } 2617 } 2618 2619 } else { 2620 usleep(sleepTime); 2621 } 2622 } 2623 2624 // Finally let go of removed track(s), without the lock held 2625 // since we can't guarantee the destructors won't acquire that 2626 // same lock. This will also mutate and push a new fast mixer state. 2627 threadLoop_removeTracks(tracksToRemove); 2628 tracksToRemove.clear(); 2629 2630 // FIXME I don't understand the need for this here; 2631 // it was in the original code but maybe the 2632 // assignment in saveOutputTracks() makes this unnecessary? 2633 clearOutputTracks(); 2634 2635 // Effect chains will be actually deleted here if they were removed from 2636 // mEffectChains list during mixing or effects processing 2637 effectChains.clear(); 2638 2639 // FIXME Note that the above .clear() is no longer necessary since effectChains 2640 // is now local to this block, but will keep it for now (at least until merge done). 2641 } 2642 2643 threadLoop_exit(); 2644 2645 if (!mStandby) { 2646 threadLoop_standby(); 2647 mStandby = true; 2648 } 2649 2650 releaseWakeLock(); 2651 mWakeLockUids.clear(); 2652 mActiveTracksGeneration++; 2653 2654 ALOGV("Thread %p type %d exiting", this, mType); 2655 return false; 2656} 2657 2658// removeTracks_l() must be called with ThreadBase::mLock held 2659void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2660{ 2661 size_t count = tracksToRemove.size(); 2662 if (count > 0) { 2663 for (size_t i=0 ; i<count ; i++) { 2664 const sp<Track>& track = tracksToRemove.itemAt(i); 2665 mActiveTracks.remove(track); 2666 mWakeLockUids.remove(track->uid()); 2667 mActiveTracksGeneration++; 2668 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2669 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2670 if (chain != 0) { 2671 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2672 track->sessionId()); 2673 chain->decActiveTrackCnt(); 2674 } 2675 if (track->isTerminated()) { 2676 removeTrack_l(track); 2677 } 2678 } 2679 } 2680 2681} 2682 2683status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2684{ 2685 if (mNormalSink != 0) { 2686 return mNormalSink->getTimestamp(timestamp); 2687 } 2688 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2689 uint64_t position64; 2690 int ret = mOutput->stream->get_presentation_position( 2691 mOutput->stream, &position64, ×tamp.mTime); 2692 if (ret == 0) { 2693 timestamp.mPosition = (uint32_t)position64; 2694 return NO_ERROR; 2695 } 2696 } 2697 return INVALID_OPERATION; 2698} 2699 2700status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2701 audio_patch_handle_t *handle) 2702{ 2703 status_t status = NO_ERROR; 2704 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2705 // store new device and send to effects 2706 audio_devices_t type = AUDIO_DEVICE_NONE; 2707 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2708 type |= patch->sinks[i].ext.device.type; 2709 } 2710 mOutDevice = type; 2711 for (size_t i = 0; i < mEffectChains.size(); i++) { 2712 mEffectChains[i]->setDevice_l(mOutDevice); 2713 } 2714 2715 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2716 status = hwDevice->create_audio_patch(hwDevice, 2717 patch->num_sources, 2718 patch->sources, 2719 patch->num_sinks, 2720 patch->sinks, 2721 handle); 2722 } else { 2723 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2724 } 2725 return status; 2726} 2727 2728status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2729{ 2730 status_t status = NO_ERROR; 2731 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2732 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2733 status = hwDevice->release_audio_patch(hwDevice, handle); 2734 } else { 2735 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2736 } 2737 return status; 2738} 2739 2740void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2741{ 2742 Mutex::Autolock _l(mLock); 2743 mTracks.add(track); 2744} 2745 2746void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2747{ 2748 Mutex::Autolock _l(mLock); 2749 destroyTrack_l(track); 2750} 2751 2752void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2753{ 2754 ThreadBase::getAudioPortConfig(config); 2755 config->role = AUDIO_PORT_ROLE_SOURCE; 2756 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2757 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2758} 2759 2760// ---------------------------------------------------------------------------- 2761 2762AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2763 audio_io_handle_t id, audio_devices_t device, type_t type) 2764 : PlaybackThread(audioFlinger, output, id, device, type), 2765 // mAudioMixer below 2766 // mFastMixer below 2767 mFastMixerFutex(0) 2768 // mOutputSink below 2769 // mPipeSink below 2770 // mNormalSink below 2771{ 2772 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2773 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2774 "mFrameCount=%d, mNormalFrameCount=%d", 2775 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2776 mNormalFrameCount); 2777 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2778 2779 // create an NBAIO sink for the HAL output stream, and negotiate 2780 mOutputSink = new AudioStreamOutSink(output->stream); 2781 size_t numCounterOffers = 0; 2782 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2783 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2784 ALOG_ASSERT(index == 0); 2785 2786 // initialize fast mixer depending on configuration 2787 bool initFastMixer; 2788 switch (kUseFastMixer) { 2789 case FastMixer_Never: 2790 initFastMixer = false; 2791 break; 2792 case FastMixer_Always: 2793 initFastMixer = true; 2794 break; 2795 case FastMixer_Static: 2796 case FastMixer_Dynamic: 2797 initFastMixer = mFrameCount < mNormalFrameCount; 2798 break; 2799 } 2800 if (initFastMixer) { 2801 audio_format_t fastMixerFormat; 2802 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2803 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2804 } else { 2805 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2806 } 2807 if (mFormat != fastMixerFormat) { 2808 // change our Sink format to accept our intermediate precision 2809 mFormat = fastMixerFormat; 2810 free(mSinkBuffer); 2811 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2812 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2813 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2814 } 2815 2816 // create a MonoPipe to connect our submix to FastMixer 2817 NBAIO_Format format = mOutputSink->format(); 2818 // adjust format to match that of the Fast Mixer 2819 format.mFormat = fastMixerFormat; 2820 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2821 2822 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2823 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2824 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2825 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2826 const NBAIO_Format offers[1] = {format}; 2827 size_t numCounterOffers = 0; 2828 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2829 ALOG_ASSERT(index == 0); 2830 monoPipe->setAvgFrames((mScreenState & 1) ? 2831 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2832 mPipeSink = monoPipe; 2833 2834#ifdef TEE_SINK 2835 if (mTeeSinkOutputEnabled) { 2836 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2837 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2838 numCounterOffers = 0; 2839 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2840 ALOG_ASSERT(index == 0); 2841 mTeeSink = teeSink; 2842 PipeReader *teeSource = new PipeReader(*teeSink); 2843 numCounterOffers = 0; 2844 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2845 ALOG_ASSERT(index == 0); 2846 mTeeSource = teeSource; 2847 } 2848#endif 2849 2850 // create fast mixer and configure it initially with just one fast track for our submix 2851 mFastMixer = new FastMixer(); 2852 FastMixerStateQueue *sq = mFastMixer->sq(); 2853#ifdef STATE_QUEUE_DUMP 2854 sq->setObserverDump(&mStateQueueObserverDump); 2855 sq->setMutatorDump(&mStateQueueMutatorDump); 2856#endif 2857 FastMixerState *state = sq->begin(); 2858 FastTrack *fastTrack = &state->mFastTracks[0]; 2859 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2860 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2861 fastTrack->mVolumeProvider = NULL; 2862 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2863 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2864 fastTrack->mGeneration++; 2865 state->mFastTracksGen++; 2866 state->mTrackMask = 1; 2867 // fast mixer will use the HAL output sink 2868 state->mOutputSink = mOutputSink.get(); 2869 state->mOutputSinkGen++; 2870 state->mFrameCount = mFrameCount; 2871 state->mCommand = FastMixerState::COLD_IDLE; 2872 // already done in constructor initialization list 2873 //mFastMixerFutex = 0; 2874 state->mColdFutexAddr = &mFastMixerFutex; 2875 state->mColdGen++; 2876 state->mDumpState = &mFastMixerDumpState; 2877#ifdef TEE_SINK 2878 state->mTeeSink = mTeeSink.get(); 2879#endif 2880 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2881 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2882 sq->end(); 2883 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2884 2885 // start the fast mixer 2886 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2887 pid_t tid = mFastMixer->getTid(); 2888 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2889 if (err != 0) { 2890 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2891 kPriorityFastMixer, getpid_cached, tid, err); 2892 } 2893 2894#ifdef AUDIO_WATCHDOG 2895 // create and start the watchdog 2896 mAudioWatchdog = new AudioWatchdog(); 2897 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2898 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2899 tid = mAudioWatchdog->getTid(); 2900 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2901 if (err != 0) { 2902 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2903 kPriorityFastMixer, getpid_cached, tid, err); 2904 } 2905#endif 2906 2907 } 2908 2909 switch (kUseFastMixer) { 2910 case FastMixer_Never: 2911 case FastMixer_Dynamic: 2912 mNormalSink = mOutputSink; 2913 break; 2914 case FastMixer_Always: 2915 mNormalSink = mPipeSink; 2916 break; 2917 case FastMixer_Static: 2918 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2919 break; 2920 } 2921} 2922 2923AudioFlinger::MixerThread::~MixerThread() 2924{ 2925 if (mFastMixer != 0) { 2926 FastMixerStateQueue *sq = mFastMixer->sq(); 2927 FastMixerState *state = sq->begin(); 2928 if (state->mCommand == FastMixerState::COLD_IDLE) { 2929 int32_t old = android_atomic_inc(&mFastMixerFutex); 2930 if (old == -1) { 2931 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2932 } 2933 } 2934 state->mCommand = FastMixerState::EXIT; 2935 sq->end(); 2936 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2937 mFastMixer->join(); 2938 // Though the fast mixer thread has exited, it's state queue is still valid. 2939 // We'll use that extract the final state which contains one remaining fast track 2940 // corresponding to our sub-mix. 2941 state = sq->begin(); 2942 ALOG_ASSERT(state->mTrackMask == 1); 2943 FastTrack *fastTrack = &state->mFastTracks[0]; 2944 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2945 delete fastTrack->mBufferProvider; 2946 sq->end(false /*didModify*/); 2947 mFastMixer.clear(); 2948#ifdef AUDIO_WATCHDOG 2949 if (mAudioWatchdog != 0) { 2950 mAudioWatchdog->requestExit(); 2951 mAudioWatchdog->requestExitAndWait(); 2952 mAudioWatchdog.clear(); 2953 } 2954#endif 2955 } 2956 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2957 delete mAudioMixer; 2958} 2959 2960 2961uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2962{ 2963 if (mFastMixer != 0) { 2964 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2965 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2966 } 2967 return latency; 2968} 2969 2970 2971void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2972{ 2973 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2974} 2975 2976ssize_t AudioFlinger::MixerThread::threadLoop_write() 2977{ 2978 // FIXME we should only do one push per cycle; confirm this is true 2979 // Start the fast mixer if it's not already running 2980 if (mFastMixer != 0) { 2981 FastMixerStateQueue *sq = mFastMixer->sq(); 2982 FastMixerState *state = sq->begin(); 2983 if (state->mCommand != FastMixerState::MIX_WRITE && 2984 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2985 if (state->mCommand == FastMixerState::COLD_IDLE) { 2986 int32_t old = android_atomic_inc(&mFastMixerFutex); 2987 if (old == -1) { 2988 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2989 } 2990#ifdef AUDIO_WATCHDOG 2991 if (mAudioWatchdog != 0) { 2992 mAudioWatchdog->resume(); 2993 } 2994#endif 2995 } 2996 state->mCommand = FastMixerState::MIX_WRITE; 2997 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2998 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2999 sq->end(); 3000 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3001 if (kUseFastMixer == FastMixer_Dynamic) { 3002 mNormalSink = mPipeSink; 3003 } 3004 } else { 3005 sq->end(false /*didModify*/); 3006 } 3007 } 3008 return PlaybackThread::threadLoop_write(); 3009} 3010 3011void AudioFlinger::MixerThread::threadLoop_standby() 3012{ 3013 // Idle the fast mixer if it's currently running 3014 if (mFastMixer != 0) { 3015 FastMixerStateQueue *sq = mFastMixer->sq(); 3016 FastMixerState *state = sq->begin(); 3017 if (!(state->mCommand & FastMixerState::IDLE)) { 3018 state->mCommand = FastMixerState::COLD_IDLE; 3019 state->mColdFutexAddr = &mFastMixerFutex; 3020 state->mColdGen++; 3021 mFastMixerFutex = 0; 3022 sq->end(); 3023 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3024 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3025 if (kUseFastMixer == FastMixer_Dynamic) { 3026 mNormalSink = mOutputSink; 3027 } 3028#ifdef AUDIO_WATCHDOG 3029 if (mAudioWatchdog != 0) { 3030 mAudioWatchdog->pause(); 3031 } 3032#endif 3033 } else { 3034 sq->end(false /*didModify*/); 3035 } 3036 } 3037 PlaybackThread::threadLoop_standby(); 3038} 3039 3040bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3041{ 3042 return false; 3043} 3044 3045bool AudioFlinger::PlaybackThread::shouldStandby_l() 3046{ 3047 return !mStandby; 3048} 3049 3050bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3051{ 3052 Mutex::Autolock _l(mLock); 3053 return waitingAsyncCallback_l(); 3054} 3055 3056// shared by MIXER and DIRECT, overridden by DUPLICATING 3057void AudioFlinger::PlaybackThread::threadLoop_standby() 3058{ 3059 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3060 mOutput->stream->common.standby(&mOutput->stream->common); 3061 if (mUseAsyncWrite != 0) { 3062 // discard any pending drain or write ack by incrementing sequence 3063 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3064 mDrainSequence = (mDrainSequence + 2) & ~1; 3065 ALOG_ASSERT(mCallbackThread != 0); 3066 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3067 mCallbackThread->setDraining(mDrainSequence); 3068 } 3069} 3070 3071void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3072{ 3073 ALOGV("signal playback thread"); 3074 broadcast_l(); 3075} 3076 3077void AudioFlinger::MixerThread::threadLoop_mix() 3078{ 3079 // obtain the presentation timestamp of the next output buffer 3080 int64_t pts; 3081 status_t status = INVALID_OPERATION; 3082 3083 if (mNormalSink != 0) { 3084 status = mNormalSink->getNextWriteTimestamp(&pts); 3085 } else { 3086 status = mOutputSink->getNextWriteTimestamp(&pts); 3087 } 3088 3089 if (status != NO_ERROR) { 3090 pts = AudioBufferProvider::kInvalidPTS; 3091 } 3092 3093 // mix buffers... 3094 mAudioMixer->process(pts); 3095 mCurrentWriteLength = mSinkBufferSize; 3096 // increase sleep time progressively when application underrun condition clears. 3097 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3098 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3099 // such that we would underrun the audio HAL. 3100 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3101 sleepTimeShift--; 3102 } 3103 sleepTime = 0; 3104 standbyTime = systemTime() + standbyDelay; 3105 //TODO: delay standby when effects have a tail 3106} 3107 3108void AudioFlinger::MixerThread::threadLoop_sleepTime() 3109{ 3110 // If no tracks are ready, sleep once for the duration of an output 3111 // buffer size, then write 0s to the output 3112 if (sleepTime == 0) { 3113 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3114 sleepTime = activeSleepTime >> sleepTimeShift; 3115 if (sleepTime < kMinThreadSleepTimeUs) { 3116 sleepTime = kMinThreadSleepTimeUs; 3117 } 3118 // reduce sleep time in case of consecutive application underruns to avoid 3119 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3120 // duration we would end up writing less data than needed by the audio HAL if 3121 // the condition persists. 3122 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3123 sleepTimeShift++; 3124 } 3125 } else { 3126 sleepTime = idleSleepTime; 3127 } 3128 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3129 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3130 // before effects processing or output. 3131 if (mMixerBufferValid) { 3132 memset(mMixerBuffer, 0, mMixerBufferSize); 3133 } else { 3134 memset(mSinkBuffer, 0, mSinkBufferSize); 3135 } 3136 sleepTime = 0; 3137 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3138 "anticipated start"); 3139 } 3140 // TODO add standby time extension fct of effect tail 3141} 3142 3143// prepareTracks_l() must be called with ThreadBase::mLock held 3144AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3145 Vector< sp<Track> > *tracksToRemove) 3146{ 3147 3148 mixer_state mixerStatus = MIXER_IDLE; 3149 // find out which tracks need to be processed 3150 size_t count = mActiveTracks.size(); 3151 size_t mixedTracks = 0; 3152 size_t tracksWithEffect = 0; 3153 // counts only _active_ fast tracks 3154 size_t fastTracks = 0; 3155 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3156 3157 float masterVolume = mMasterVolume; 3158 bool masterMute = mMasterMute; 3159 3160 if (masterMute) { 3161 masterVolume = 0; 3162 } 3163 // Delegate master volume control to effect in output mix effect chain if needed 3164 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3165 if (chain != 0) { 3166 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3167 chain->setVolume_l(&v, &v); 3168 masterVolume = (float)((v + (1 << 23)) >> 24); 3169 chain.clear(); 3170 } 3171 3172 // prepare a new state to push 3173 FastMixerStateQueue *sq = NULL; 3174 FastMixerState *state = NULL; 3175 bool didModify = false; 3176 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3177 if (mFastMixer != 0) { 3178 sq = mFastMixer->sq(); 3179 state = sq->begin(); 3180 } 3181 3182 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3183 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3184 3185 for (size_t i=0 ; i<count ; i++) { 3186 const sp<Track> t = mActiveTracks[i].promote(); 3187 if (t == 0) { 3188 continue; 3189 } 3190 3191 // this const just means the local variable doesn't change 3192 Track* const track = t.get(); 3193 3194 // process fast tracks 3195 if (track->isFastTrack()) { 3196 3197 // It's theoretically possible (though unlikely) for a fast track to be created 3198 // and then removed within the same normal mix cycle. This is not a problem, as 3199 // the track never becomes active so it's fast mixer slot is never touched. 3200 // The converse, of removing an (active) track and then creating a new track 3201 // at the identical fast mixer slot within the same normal mix cycle, 3202 // is impossible because the slot isn't marked available until the end of each cycle. 3203 int j = track->mFastIndex; 3204 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3205 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3206 FastTrack *fastTrack = &state->mFastTracks[j]; 3207 3208 // Determine whether the track is currently in underrun condition, 3209 // and whether it had a recent underrun. 3210 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3211 FastTrackUnderruns underruns = ftDump->mUnderruns; 3212 uint32_t recentFull = (underruns.mBitFields.mFull - 3213 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3214 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3215 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3216 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3217 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3218 uint32_t recentUnderruns = recentPartial + recentEmpty; 3219 track->mObservedUnderruns = underruns; 3220 // don't count underruns that occur while stopping or pausing 3221 // or stopped which can occur when flush() is called while active 3222 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3223 recentUnderruns > 0) { 3224 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3225 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3226 } 3227 3228 // This is similar to the state machine for normal tracks, 3229 // with a few modifications for fast tracks. 3230 bool isActive = true; 3231 switch (track->mState) { 3232 case TrackBase::STOPPING_1: 3233 // track stays active in STOPPING_1 state until first underrun 3234 if (recentUnderruns > 0 || track->isTerminated()) { 3235 track->mState = TrackBase::STOPPING_2; 3236 } 3237 break; 3238 case TrackBase::PAUSING: 3239 // ramp down is not yet implemented 3240 track->setPaused(); 3241 break; 3242 case TrackBase::RESUMING: 3243 // ramp up is not yet implemented 3244 track->mState = TrackBase::ACTIVE; 3245 break; 3246 case TrackBase::ACTIVE: 3247 if (recentFull > 0 || recentPartial > 0) { 3248 // track has provided at least some frames recently: reset retry count 3249 track->mRetryCount = kMaxTrackRetries; 3250 } 3251 if (recentUnderruns == 0) { 3252 // no recent underruns: stay active 3253 break; 3254 } 3255 // there has recently been an underrun of some kind 3256 if (track->sharedBuffer() == 0) { 3257 // were any of the recent underruns "empty" (no frames available)? 3258 if (recentEmpty == 0) { 3259 // no, then ignore the partial underruns as they are allowed indefinitely 3260 break; 3261 } 3262 // there has recently been an "empty" underrun: decrement the retry counter 3263 if (--(track->mRetryCount) > 0) { 3264 break; 3265 } 3266 // indicate to client process that the track was disabled because of underrun; 3267 // it will then automatically call start() when data is available 3268 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3269 // remove from active list, but state remains ACTIVE [confusing but true] 3270 isActive = false; 3271 break; 3272 } 3273 // fall through 3274 case TrackBase::STOPPING_2: 3275 case TrackBase::PAUSED: 3276 case TrackBase::STOPPED: 3277 case TrackBase::FLUSHED: // flush() while active 3278 // Check for presentation complete if track is inactive 3279 // We have consumed all the buffers of this track. 3280 // This would be incomplete if we auto-paused on underrun 3281 { 3282 size_t audioHALFrames = 3283 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3284 size_t framesWritten = mBytesWritten / mFrameSize; 3285 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3286 // track stays in active list until presentation is complete 3287 break; 3288 } 3289 } 3290 if (track->isStopping_2()) { 3291 track->mState = TrackBase::STOPPED; 3292 } 3293 if (track->isStopped()) { 3294 // Can't reset directly, as fast mixer is still polling this track 3295 // track->reset(); 3296 // So instead mark this track as needing to be reset after push with ack 3297 resetMask |= 1 << i; 3298 } 3299 isActive = false; 3300 break; 3301 case TrackBase::IDLE: 3302 default: 3303 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3304 } 3305 3306 if (isActive) { 3307 // was it previously inactive? 3308 if (!(state->mTrackMask & (1 << j))) { 3309 ExtendedAudioBufferProvider *eabp = track; 3310 VolumeProvider *vp = track; 3311 fastTrack->mBufferProvider = eabp; 3312 fastTrack->mVolumeProvider = vp; 3313 fastTrack->mChannelMask = track->mChannelMask; 3314 fastTrack->mFormat = track->mFormat; 3315 fastTrack->mGeneration++; 3316 state->mTrackMask |= 1 << j; 3317 didModify = true; 3318 // no acknowledgement required for newly active tracks 3319 } 3320 // cache the combined master volume and stream type volume for fast mixer; this 3321 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3322 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3323 ++fastTracks; 3324 } else { 3325 // was it previously active? 3326 if (state->mTrackMask & (1 << j)) { 3327 fastTrack->mBufferProvider = NULL; 3328 fastTrack->mGeneration++; 3329 state->mTrackMask &= ~(1 << j); 3330 didModify = true; 3331 // If any fast tracks were removed, we must wait for acknowledgement 3332 // because we're about to decrement the last sp<> on those tracks. 3333 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3334 } else { 3335 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3336 } 3337 tracksToRemove->add(track); 3338 // Avoids a misleading display in dumpsys 3339 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3340 } 3341 continue; 3342 } 3343 3344 { // local variable scope to avoid goto warning 3345 3346 audio_track_cblk_t* cblk = track->cblk(); 3347 3348 // The first time a track is added we wait 3349 // for all its buffers to be filled before processing it 3350 int name = track->name(); 3351 // make sure that we have enough frames to mix one full buffer. 3352 // enforce this condition only once to enable draining the buffer in case the client 3353 // app does not call stop() and relies on underrun to stop: 3354 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3355 // during last round 3356 size_t desiredFrames; 3357 uint32_t sr = track->sampleRate(); 3358 if (sr == mSampleRate) { 3359 desiredFrames = mNormalFrameCount; 3360 } else { 3361 // +1 for rounding and +1 for additional sample needed for interpolation 3362 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3363 // add frames already consumed but not yet released by the resampler 3364 // because mAudioTrackServerProxy->framesReady() will include these frames 3365 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3366#if 0 3367 // the minimum track buffer size is normally twice the number of frames necessary 3368 // to fill one buffer and the resampler should not leave more than one buffer worth 3369 // of unreleased frames after each pass, but just in case... 3370 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3371#endif 3372 } 3373 uint32_t minFrames = 1; 3374 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3375 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3376 minFrames = desiredFrames; 3377 } 3378 3379 size_t framesReady = track->framesReady(); 3380 if ((framesReady >= minFrames) && track->isReady() && 3381 !track->isPaused() && !track->isTerminated()) 3382 { 3383 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3384 3385 mixedTracks++; 3386 3387 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3388 // there is an effect chain connected to the track 3389 chain.clear(); 3390 if (track->mainBuffer() != mSinkBuffer && 3391 track->mainBuffer() != mMixerBuffer) { 3392 if (mEffectBufferEnabled) { 3393 mEffectBufferValid = true; // Later can set directly. 3394 } 3395 chain = getEffectChain_l(track->sessionId()); 3396 // Delegate volume control to effect in track effect chain if needed 3397 if (chain != 0) { 3398 tracksWithEffect++; 3399 } else { 3400 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3401 "session %d", 3402 name, track->sessionId()); 3403 } 3404 } 3405 3406 3407 int param = AudioMixer::VOLUME; 3408 if (track->mFillingUpStatus == Track::FS_FILLED) { 3409 // no ramp for the first volume setting 3410 track->mFillingUpStatus = Track::FS_ACTIVE; 3411 if (track->mState == TrackBase::RESUMING) { 3412 track->mState = TrackBase::ACTIVE; 3413 param = AudioMixer::RAMP_VOLUME; 3414 } 3415 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3416 // FIXME should not make a decision based on mServer 3417 } else if (cblk->mServer != 0) { 3418 // If the track is stopped before the first frame was mixed, 3419 // do not apply ramp 3420 param = AudioMixer::RAMP_VOLUME; 3421 } 3422 3423 // compute volume for this track 3424 uint32_t vl, vr; // in U8.24 integer format 3425 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3426 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3427 vl = vr = 0; 3428 vlf = vrf = vaf = 0.; 3429 if (track->isPausing()) { 3430 track->setPaused(); 3431 } 3432 } else { 3433 3434 // read original volumes with volume control 3435 float typeVolume = mStreamTypes[track->streamType()].volume; 3436 float v = masterVolume * typeVolume; 3437 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3438 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3439 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3440 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3441 // track volumes come from shared memory, so can't be trusted and must be clamped 3442 if (vlf > GAIN_FLOAT_UNITY) { 3443 ALOGV("Track left volume out of range: %.3g", vlf); 3444 vlf = GAIN_FLOAT_UNITY; 3445 } 3446 if (vrf > GAIN_FLOAT_UNITY) { 3447 ALOGV("Track right volume out of range: %.3g", vrf); 3448 vrf = GAIN_FLOAT_UNITY; 3449 } 3450 // now apply the master volume and stream type volume 3451 vlf *= v; 3452 vrf *= v; 3453 // assuming master volume and stream type volume each go up to 1.0, 3454 // then derive vl and vr as U8.24 versions for the effect chain 3455 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3456 vl = (uint32_t) (scaleto8_24 * vlf); 3457 vr = (uint32_t) (scaleto8_24 * vrf); 3458 // vl and vr are now in U8.24 format 3459 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3460 // send level comes from shared memory and so may be corrupt 3461 if (sendLevel > MAX_GAIN_INT) { 3462 ALOGV("Track send level out of range: %04X", sendLevel); 3463 sendLevel = MAX_GAIN_INT; 3464 } 3465 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3466 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3467 } 3468 3469 // Delegate volume control to effect in track effect chain if needed 3470 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3471 // Do not ramp volume if volume is controlled by effect 3472 param = AudioMixer::VOLUME; 3473 // Update remaining floating point volume levels 3474 vlf = (float)vl / (1 << 24); 3475 vrf = (float)vr / (1 << 24); 3476 track->mHasVolumeController = true; 3477 } else { 3478 // force no volume ramp when volume controller was just disabled or removed 3479 // from effect chain to avoid volume spike 3480 if (track->mHasVolumeController) { 3481 param = AudioMixer::VOLUME; 3482 } 3483 track->mHasVolumeController = false; 3484 } 3485 3486 // XXX: these things DON'T need to be done each time 3487 mAudioMixer->setBufferProvider(name, track); 3488 mAudioMixer->enable(name); 3489 3490 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3491 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3492 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3493 mAudioMixer->setParameter( 3494 name, 3495 AudioMixer::TRACK, 3496 AudioMixer::FORMAT, (void *)track->format()); 3497 mAudioMixer->setParameter( 3498 name, 3499 AudioMixer::TRACK, 3500 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3501 mAudioMixer->setParameter( 3502 name, 3503 AudioMixer::TRACK, 3504 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3505 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3506 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3507 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3508 if (reqSampleRate == 0) { 3509 reqSampleRate = mSampleRate; 3510 } else if (reqSampleRate > maxSampleRate) { 3511 reqSampleRate = maxSampleRate; 3512 } 3513 mAudioMixer->setParameter( 3514 name, 3515 AudioMixer::RESAMPLE, 3516 AudioMixer::SAMPLE_RATE, 3517 (void *)(uintptr_t)reqSampleRate); 3518 /* 3519 * Select the appropriate output buffer for the track. 3520 * 3521 * Tracks with effects go into their own effects chain buffer 3522 * and from there into either mEffectBuffer or mSinkBuffer. 3523 * 3524 * Other tracks can use mMixerBuffer for higher precision 3525 * channel accumulation. If this buffer is enabled 3526 * (mMixerBufferEnabled true), then selected tracks will accumulate 3527 * into it. 3528 * 3529 */ 3530 if (mMixerBufferEnabled 3531 && (track->mainBuffer() == mSinkBuffer 3532 || track->mainBuffer() == mMixerBuffer)) { 3533 mAudioMixer->setParameter( 3534 name, 3535 AudioMixer::TRACK, 3536 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3537 mAudioMixer->setParameter( 3538 name, 3539 AudioMixer::TRACK, 3540 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3541 // TODO: override track->mainBuffer()? 3542 mMixerBufferValid = true; 3543 } else { 3544 mAudioMixer->setParameter( 3545 name, 3546 AudioMixer::TRACK, 3547 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3548 mAudioMixer->setParameter( 3549 name, 3550 AudioMixer::TRACK, 3551 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3552 } 3553 mAudioMixer->setParameter( 3554 name, 3555 AudioMixer::TRACK, 3556 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3557 3558 // reset retry count 3559 track->mRetryCount = kMaxTrackRetries; 3560 3561 // If one track is ready, set the mixer ready if: 3562 // - the mixer was not ready during previous round OR 3563 // - no other track is not ready 3564 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3565 mixerStatus != MIXER_TRACKS_ENABLED) { 3566 mixerStatus = MIXER_TRACKS_READY; 3567 } 3568 } else { 3569 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3570 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3571 } 3572 // clear effect chain input buffer if an active track underruns to avoid sending 3573 // previous audio buffer again to effects 3574 chain = getEffectChain_l(track->sessionId()); 3575 if (chain != 0) { 3576 chain->clearInputBuffer(); 3577 } 3578 3579 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3580 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3581 track->isStopped() || track->isPaused()) { 3582 // We have consumed all the buffers of this track. 3583 // Remove it from the list of active tracks. 3584 // TODO: use actual buffer filling status instead of latency when available from 3585 // audio HAL 3586 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3587 size_t framesWritten = mBytesWritten / mFrameSize; 3588 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3589 if (track->isStopped()) { 3590 track->reset(); 3591 } 3592 tracksToRemove->add(track); 3593 } 3594 } else { 3595 // No buffers for this track. Give it a few chances to 3596 // fill a buffer, then remove it from active list. 3597 if (--(track->mRetryCount) <= 0) { 3598 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3599 tracksToRemove->add(track); 3600 // indicate to client process that the track was disabled because of underrun; 3601 // it will then automatically call start() when data is available 3602 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3603 // If one track is not ready, mark the mixer also not ready if: 3604 // - the mixer was ready during previous round OR 3605 // - no other track is ready 3606 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3607 mixerStatus != MIXER_TRACKS_READY) { 3608 mixerStatus = MIXER_TRACKS_ENABLED; 3609 } 3610 } 3611 mAudioMixer->disable(name); 3612 } 3613 3614 } // local variable scope to avoid goto warning 3615track_is_ready: ; 3616 3617 } 3618 3619 // Push the new FastMixer state if necessary 3620 bool pauseAudioWatchdog = false; 3621 if (didModify) { 3622 state->mFastTracksGen++; 3623 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3624 if (kUseFastMixer == FastMixer_Dynamic && 3625 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3626 state->mCommand = FastMixerState::COLD_IDLE; 3627 state->mColdFutexAddr = &mFastMixerFutex; 3628 state->mColdGen++; 3629 mFastMixerFutex = 0; 3630 if (kUseFastMixer == FastMixer_Dynamic) { 3631 mNormalSink = mOutputSink; 3632 } 3633 // If we go into cold idle, need to wait for acknowledgement 3634 // so that fast mixer stops doing I/O. 3635 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3636 pauseAudioWatchdog = true; 3637 } 3638 } 3639 if (sq != NULL) { 3640 sq->end(didModify); 3641 sq->push(block); 3642 } 3643#ifdef AUDIO_WATCHDOG 3644 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3645 mAudioWatchdog->pause(); 3646 } 3647#endif 3648 3649 // Now perform the deferred reset on fast tracks that have stopped 3650 while (resetMask != 0) { 3651 size_t i = __builtin_ctz(resetMask); 3652 ALOG_ASSERT(i < count); 3653 resetMask &= ~(1 << i); 3654 sp<Track> t = mActiveTracks[i].promote(); 3655 if (t == 0) { 3656 continue; 3657 } 3658 Track* track = t.get(); 3659 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3660 track->reset(); 3661 } 3662 3663 // remove all the tracks that need to be... 3664 removeTracks_l(*tracksToRemove); 3665 3666 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3667 mEffectBufferValid = true; 3668 } 3669 3670 // sink or mix buffer must be cleared if all tracks are connected to an 3671 // effect chain as in this case the mixer will not write to the sink or mix buffer 3672 // and track effects will accumulate into it 3673 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3674 (mixedTracks == 0 && fastTracks > 0))) { 3675 // FIXME as a performance optimization, should remember previous zero status 3676 if (mMixerBufferValid) { 3677 memset(mMixerBuffer, 0, mMixerBufferSize); 3678 // TODO: In testing, mSinkBuffer below need not be cleared because 3679 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3680 // after mixing. 3681 // 3682 // To enforce this guarantee: 3683 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3684 // (mixedTracks == 0 && fastTracks > 0)) 3685 // must imply MIXER_TRACKS_READY. 3686 // Later, we may clear buffers regardless, and skip much of this logic. 3687 } 3688 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3689 if (mEffectBufferValid) { 3690 memset(mEffectBuffer, 0, mEffectBufferSize); 3691 } 3692 // FIXME as a performance optimization, should remember previous zero status 3693 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3694 } 3695 3696 // if any fast tracks, then status is ready 3697 mMixerStatusIgnoringFastTracks = mixerStatus; 3698 if (fastTracks > 0) { 3699 mixerStatus = MIXER_TRACKS_READY; 3700 } 3701 return mixerStatus; 3702} 3703 3704// getTrackName_l() must be called with ThreadBase::mLock held 3705int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3706 audio_format_t format, int sessionId) 3707{ 3708 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3709} 3710 3711// deleteTrackName_l() must be called with ThreadBase::mLock held 3712void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3713{ 3714 ALOGV("remove track (%d) and delete from mixer", name); 3715 mAudioMixer->deleteTrackName(name); 3716} 3717 3718// checkForNewParameter_l() must be called with ThreadBase::mLock held 3719bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3720 status_t& status) 3721{ 3722 bool reconfig = false; 3723 3724 status = NO_ERROR; 3725 3726 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3727 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3728 if (mFastMixer != 0) { 3729 FastMixerStateQueue *sq = mFastMixer->sq(); 3730 FastMixerState *state = sq->begin(); 3731 if (!(state->mCommand & FastMixerState::IDLE)) { 3732 previousCommand = state->mCommand; 3733 state->mCommand = FastMixerState::HOT_IDLE; 3734 sq->end(); 3735 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3736 } else { 3737 sq->end(false /*didModify*/); 3738 } 3739 } 3740 3741 AudioParameter param = AudioParameter(keyValuePair); 3742 int value; 3743 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3744 reconfig = true; 3745 } 3746 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3747 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3748 status = BAD_VALUE; 3749 } else { 3750 // no need to save value, since it's constant 3751 reconfig = true; 3752 } 3753 } 3754 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3755 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3756 status = BAD_VALUE; 3757 } else { 3758 // no need to save value, since it's constant 3759 reconfig = true; 3760 } 3761 } 3762 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3763 // do not accept frame count changes if tracks are open as the track buffer 3764 // size depends on frame count and correct behavior would not be guaranteed 3765 // if frame count is changed after track creation 3766 if (!mTracks.isEmpty()) { 3767 status = INVALID_OPERATION; 3768 } else { 3769 reconfig = true; 3770 } 3771 } 3772 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3773#ifdef ADD_BATTERY_DATA 3774 // when changing the audio output device, call addBatteryData to notify 3775 // the change 3776 if (mOutDevice != value) { 3777 uint32_t params = 0; 3778 // check whether speaker is on 3779 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3780 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3781 } 3782 3783 audio_devices_t deviceWithoutSpeaker 3784 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3785 // check if any other device (except speaker) is on 3786 if (value & deviceWithoutSpeaker ) { 3787 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3788 } 3789 3790 if (params != 0) { 3791 addBatteryData(params); 3792 } 3793 } 3794#endif 3795 3796 // forward device change to effects that have requested to be 3797 // aware of attached audio device. 3798 if (value != AUDIO_DEVICE_NONE) { 3799 mOutDevice = value; 3800 for (size_t i = 0; i < mEffectChains.size(); i++) { 3801 mEffectChains[i]->setDevice_l(mOutDevice); 3802 } 3803 } 3804 } 3805 3806 if (status == NO_ERROR) { 3807 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3808 keyValuePair.string()); 3809 if (!mStandby && status == INVALID_OPERATION) { 3810 mOutput->stream->common.standby(&mOutput->stream->common); 3811 mStandby = true; 3812 mBytesWritten = 0; 3813 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3814 keyValuePair.string()); 3815 } 3816 if (status == NO_ERROR && reconfig) { 3817 readOutputParameters_l(); 3818 delete mAudioMixer; 3819 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3820 for (size_t i = 0; i < mTracks.size() ; i++) { 3821 int name = getTrackName_l(mTracks[i]->mChannelMask, 3822 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3823 if (name < 0) { 3824 break; 3825 } 3826 mTracks[i]->mName = name; 3827 } 3828 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3829 } 3830 } 3831 3832 if (!(previousCommand & FastMixerState::IDLE)) { 3833 ALOG_ASSERT(mFastMixer != 0); 3834 FastMixerStateQueue *sq = mFastMixer->sq(); 3835 FastMixerState *state = sq->begin(); 3836 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3837 state->mCommand = previousCommand; 3838 sq->end(); 3839 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3840 } 3841 3842 return reconfig; 3843} 3844 3845 3846void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3847{ 3848 const size_t SIZE = 256; 3849 char buffer[SIZE]; 3850 String8 result; 3851 3852 PlaybackThread::dumpInternals(fd, args); 3853 3854 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3855 3856 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3857 const FastMixerDumpState copy(mFastMixerDumpState); 3858 copy.dump(fd); 3859 3860#ifdef STATE_QUEUE_DUMP 3861 // Similar for state queue 3862 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3863 observerCopy.dump(fd); 3864 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3865 mutatorCopy.dump(fd); 3866#endif 3867 3868#ifdef TEE_SINK 3869 // Write the tee output to a .wav file 3870 dumpTee(fd, mTeeSource, mId); 3871#endif 3872 3873#ifdef AUDIO_WATCHDOG 3874 if (mAudioWatchdog != 0) { 3875 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3876 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3877 wdCopy.dump(fd); 3878 } 3879#endif 3880} 3881 3882uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3883{ 3884 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3885} 3886 3887uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3888{ 3889 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3890} 3891 3892void AudioFlinger::MixerThread::cacheParameters_l() 3893{ 3894 PlaybackThread::cacheParameters_l(); 3895 3896 // FIXME: Relaxed timing because of a certain device that can't meet latency 3897 // Should be reduced to 2x after the vendor fixes the driver issue 3898 // increase threshold again due to low power audio mode. The way this warning 3899 // threshold is calculated and its usefulness should be reconsidered anyway. 3900 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3901} 3902 3903// ---------------------------------------------------------------------------- 3904 3905AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3906 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3907 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3908 // mLeftVolFloat, mRightVolFloat 3909{ 3910} 3911 3912AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3913 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3914 ThreadBase::type_t type) 3915 : PlaybackThread(audioFlinger, output, id, device, type) 3916 // mLeftVolFloat, mRightVolFloat 3917{ 3918} 3919 3920AudioFlinger::DirectOutputThread::~DirectOutputThread() 3921{ 3922} 3923 3924void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3925{ 3926 audio_track_cblk_t* cblk = track->cblk(); 3927 float left, right; 3928 3929 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3930 left = right = 0; 3931 } else { 3932 float typeVolume = mStreamTypes[track->streamType()].volume; 3933 float v = mMasterVolume * typeVolume; 3934 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3935 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3936 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3937 if (left > GAIN_FLOAT_UNITY) { 3938 left = GAIN_FLOAT_UNITY; 3939 } 3940 left *= v; 3941 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3942 if (right > GAIN_FLOAT_UNITY) { 3943 right = GAIN_FLOAT_UNITY; 3944 } 3945 right *= v; 3946 } 3947 3948 if (lastTrack) { 3949 if (left != mLeftVolFloat || right != mRightVolFloat) { 3950 mLeftVolFloat = left; 3951 mRightVolFloat = right; 3952 3953 // Convert volumes from float to 8.24 3954 uint32_t vl = (uint32_t)(left * (1 << 24)); 3955 uint32_t vr = (uint32_t)(right * (1 << 24)); 3956 3957 // Delegate volume control to effect in track effect chain if needed 3958 // only one effect chain can be present on DirectOutputThread, so if 3959 // there is one, the track is connected to it 3960 if (!mEffectChains.isEmpty()) { 3961 mEffectChains[0]->setVolume_l(&vl, &vr); 3962 left = (float)vl / (1 << 24); 3963 right = (float)vr / (1 << 24); 3964 } 3965 if (mOutput->stream->set_volume) { 3966 mOutput->stream->set_volume(mOutput->stream, left, right); 3967 } 3968 } 3969 } 3970} 3971 3972 3973AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3974 Vector< sp<Track> > *tracksToRemove 3975) 3976{ 3977 size_t count = mActiveTracks.size(); 3978 mixer_state mixerStatus = MIXER_IDLE; 3979 3980 // find out which tracks need to be processed 3981 for (size_t i = 0; i < count; i++) { 3982 sp<Track> t = mActiveTracks[i].promote(); 3983 // The track died recently 3984 if (t == 0) { 3985 continue; 3986 } 3987 3988 Track* const track = t.get(); 3989 audio_track_cblk_t* cblk = track->cblk(); 3990 // Only consider last track started for volume and mixer state control. 3991 // In theory an older track could underrun and restart after the new one starts 3992 // but as we only care about the transition phase between two tracks on a 3993 // direct output, it is not a problem to ignore the underrun case. 3994 sp<Track> l = mLatestActiveTrack.promote(); 3995 bool last = l.get() == track; 3996 3997 // The first time a track is added we wait 3998 // for all its buffers to be filled before processing it 3999 uint32_t minFrames; 4000 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 4001 minFrames = mNormalFrameCount; 4002 } else { 4003 minFrames = 1; 4004 } 4005 4006 ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ", 4007 minFrames, track->mState, track->framesReady()); 4008 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4009 !track->isStopping_2() && !track->isStopped()) 4010 { 4011 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4012 4013 if (track->mFillingUpStatus == Track::FS_FILLED) { 4014 track->mFillingUpStatus = Track::FS_ACTIVE; 4015 // make sure processVolume_l() will apply new volume even if 0 4016 mLeftVolFloat = mRightVolFloat = -1.0; 4017 if (track->mState == TrackBase::RESUMING) { 4018 track->mState = TrackBase::ACTIVE; 4019 } 4020 } 4021 4022 // compute volume for this track 4023 processVolume_l(track, last); 4024 if (last) { 4025 // reset retry count 4026 track->mRetryCount = kMaxTrackRetriesDirect; 4027 mActiveTrack = t; 4028 mixerStatus = MIXER_TRACKS_READY; 4029 } 4030 } else { 4031 // clear effect chain input buffer if the last active track started underruns 4032 // to avoid sending previous audio buffer again to effects 4033 if (!mEffectChains.isEmpty() && last) { 4034 mEffectChains[0]->clearInputBuffer(); 4035 } 4036 if (track->isStopping_1()) { 4037 track->mState = TrackBase::STOPPING_2; 4038 } 4039 if ((track->sharedBuffer() != 0) || track->isStopped() || 4040 track->isStopping_2() || track->isPaused()) { 4041 // We have consumed all the buffers of this track. 4042 // Remove it from the list of active tracks. 4043 size_t audioHALFrames; 4044 if (audio_is_linear_pcm(mFormat)) { 4045 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4046 } else { 4047 audioHALFrames = 0; 4048 } 4049 4050 size_t framesWritten = mBytesWritten / mFrameSize; 4051 if (mStandby || !last || 4052 track->presentationComplete(framesWritten, audioHALFrames)) { 4053 if (track->isStopping_2()) { 4054 track->mState = TrackBase::STOPPED; 4055 } 4056 if (track->isStopped()) { 4057 track->reset(); 4058 } 4059 tracksToRemove->add(track); 4060 } 4061 } else { 4062 // No buffers for this track. Give it a few chances to 4063 // fill a buffer, then remove it from active list. 4064 // Only consider last track started for mixer state control 4065 if (--(track->mRetryCount) <= 0) { 4066 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4067 tracksToRemove->add(track); 4068 // indicate to client process that the track was disabled because of underrun; 4069 // it will then automatically call start() when data is available 4070 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4071 } else if (last) { 4072 mixerStatus = MIXER_TRACKS_ENABLED; 4073 } 4074 } 4075 } 4076 } 4077 4078 // remove all the tracks that need to be... 4079 removeTracks_l(*tracksToRemove); 4080 4081 return mixerStatus; 4082} 4083 4084void AudioFlinger::DirectOutputThread::threadLoop_mix() 4085{ 4086 size_t frameCount = mFrameCount; 4087 int8_t *curBuf = (int8_t *)mSinkBuffer; 4088 // output audio to hardware 4089 while (frameCount) { 4090 AudioBufferProvider::Buffer buffer; 4091 buffer.frameCount = frameCount; 4092 mActiveTrack->getNextBuffer(&buffer); 4093 if (buffer.raw == NULL) { 4094 memset(curBuf, 0, frameCount * mFrameSize); 4095 break; 4096 } 4097 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4098 frameCount -= buffer.frameCount; 4099 curBuf += buffer.frameCount * mFrameSize; 4100 mActiveTrack->releaseBuffer(&buffer); 4101 } 4102 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4103 sleepTime = 0; 4104 standbyTime = systemTime() + standbyDelay; 4105 mActiveTrack.clear(); 4106} 4107 4108void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4109{ 4110 if (sleepTime == 0) { 4111 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4112 sleepTime = activeSleepTime; 4113 } else { 4114 sleepTime = idleSleepTime; 4115 } 4116 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4117 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4118 sleepTime = 0; 4119 } 4120} 4121 4122// getTrackName_l() must be called with ThreadBase::mLock held 4123int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4124 audio_format_t format __unused, int sessionId __unused) 4125{ 4126 return 0; 4127} 4128 4129// deleteTrackName_l() must be called with ThreadBase::mLock held 4130void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4131{ 4132} 4133 4134// checkForNewParameter_l() must be called with ThreadBase::mLock held 4135bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4136 status_t& status) 4137{ 4138 bool reconfig = false; 4139 4140 status = NO_ERROR; 4141 4142 AudioParameter param = AudioParameter(keyValuePair); 4143 int value; 4144 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4145 // forward device change to effects that have requested to be 4146 // aware of attached audio device. 4147 if (value != AUDIO_DEVICE_NONE) { 4148 mOutDevice = value; 4149 for (size_t i = 0; i < mEffectChains.size(); i++) { 4150 mEffectChains[i]->setDevice_l(mOutDevice); 4151 } 4152 } 4153 } 4154 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4155 // do not accept frame count changes if tracks are open as the track buffer 4156 // size depends on frame count and correct behavior would not be garantied 4157 // if frame count is changed after track creation 4158 if (!mTracks.isEmpty()) { 4159 status = INVALID_OPERATION; 4160 } else { 4161 reconfig = true; 4162 } 4163 } 4164 if (status == NO_ERROR) { 4165 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4166 keyValuePair.string()); 4167 if (!mStandby && status == INVALID_OPERATION) { 4168 mOutput->stream->common.standby(&mOutput->stream->common); 4169 mStandby = true; 4170 mBytesWritten = 0; 4171 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4172 keyValuePair.string()); 4173 } 4174 if (status == NO_ERROR && reconfig) { 4175 readOutputParameters_l(); 4176 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4177 } 4178 } 4179 4180 return reconfig; 4181} 4182 4183uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4184{ 4185 uint32_t time; 4186 if (audio_is_linear_pcm(mFormat)) { 4187 time = PlaybackThread::activeSleepTimeUs(); 4188 } else { 4189 time = 10000; 4190 } 4191 return time; 4192} 4193 4194uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4195{ 4196 uint32_t time; 4197 if (audio_is_linear_pcm(mFormat)) { 4198 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4199 } else { 4200 time = 10000; 4201 } 4202 return time; 4203} 4204 4205uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4206{ 4207 uint32_t time; 4208 if (audio_is_linear_pcm(mFormat)) { 4209 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4210 } else { 4211 time = 10000; 4212 } 4213 return time; 4214} 4215 4216void AudioFlinger::DirectOutputThread::cacheParameters_l() 4217{ 4218 PlaybackThread::cacheParameters_l(); 4219 4220 // use shorter standby delay as on normal output to release 4221 // hardware resources as soon as possible 4222 if (audio_is_linear_pcm(mFormat)) { 4223 standbyDelay = microseconds(activeSleepTime*2); 4224 } else { 4225 standbyDelay = kOffloadStandbyDelayNs; 4226 } 4227} 4228 4229// ---------------------------------------------------------------------------- 4230 4231AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4232 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4233 : Thread(false /*canCallJava*/), 4234 mPlaybackThread(playbackThread), 4235 mWriteAckSequence(0), 4236 mDrainSequence(0) 4237{ 4238} 4239 4240AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4241{ 4242} 4243 4244void AudioFlinger::AsyncCallbackThread::onFirstRef() 4245{ 4246 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4247} 4248 4249bool AudioFlinger::AsyncCallbackThread::threadLoop() 4250{ 4251 while (!exitPending()) { 4252 uint32_t writeAckSequence; 4253 uint32_t drainSequence; 4254 4255 { 4256 Mutex::Autolock _l(mLock); 4257 while (!((mWriteAckSequence & 1) || 4258 (mDrainSequence & 1) || 4259 exitPending())) { 4260 mWaitWorkCV.wait(mLock); 4261 } 4262 4263 if (exitPending()) { 4264 break; 4265 } 4266 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4267 mWriteAckSequence, mDrainSequence); 4268 writeAckSequence = mWriteAckSequence; 4269 mWriteAckSequence &= ~1; 4270 drainSequence = mDrainSequence; 4271 mDrainSequence &= ~1; 4272 } 4273 { 4274 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4275 if (playbackThread != 0) { 4276 if (writeAckSequence & 1) { 4277 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4278 } 4279 if (drainSequence & 1) { 4280 playbackThread->resetDraining(drainSequence >> 1); 4281 } 4282 } 4283 } 4284 } 4285 return false; 4286} 4287 4288void AudioFlinger::AsyncCallbackThread::exit() 4289{ 4290 ALOGV("AsyncCallbackThread::exit"); 4291 Mutex::Autolock _l(mLock); 4292 requestExit(); 4293 mWaitWorkCV.broadcast(); 4294} 4295 4296void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4297{ 4298 Mutex::Autolock _l(mLock); 4299 // bit 0 is cleared 4300 mWriteAckSequence = sequence << 1; 4301} 4302 4303void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4304{ 4305 Mutex::Autolock _l(mLock); 4306 // ignore unexpected callbacks 4307 if (mWriteAckSequence & 2) { 4308 mWriteAckSequence |= 1; 4309 mWaitWorkCV.signal(); 4310 } 4311} 4312 4313void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4314{ 4315 Mutex::Autolock _l(mLock); 4316 // bit 0 is cleared 4317 mDrainSequence = sequence << 1; 4318} 4319 4320void AudioFlinger::AsyncCallbackThread::resetDraining() 4321{ 4322 Mutex::Autolock _l(mLock); 4323 // ignore unexpected callbacks 4324 if (mDrainSequence & 2) { 4325 mDrainSequence |= 1; 4326 mWaitWorkCV.signal(); 4327 } 4328} 4329 4330 4331// ---------------------------------------------------------------------------- 4332AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4333 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4334 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4335 mHwPaused(false), 4336 mFlushPending(false), 4337 mPausedBytesRemaining(0) 4338{ 4339 //FIXME: mStandby should be set to true by ThreadBase constructor 4340 mStandby = true; 4341} 4342 4343void AudioFlinger::OffloadThread::threadLoop_exit() 4344{ 4345 if (mFlushPending || mHwPaused) { 4346 // If a flush is pending or track was paused, just discard buffered data 4347 flushHw_l(); 4348 } else { 4349 mMixerStatus = MIXER_DRAIN_ALL; 4350 threadLoop_drain(); 4351 } 4352 if (mUseAsyncWrite) { 4353 ALOG_ASSERT(mCallbackThread != 0); 4354 mCallbackThread->exit(); 4355 } 4356 PlaybackThread::threadLoop_exit(); 4357} 4358 4359AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4360 Vector< sp<Track> > *tracksToRemove 4361) 4362{ 4363 size_t count = mActiveTracks.size(); 4364 4365 mixer_state mixerStatus = MIXER_IDLE; 4366 bool doHwPause = false; 4367 bool doHwResume = false; 4368 4369 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4370 4371 // find out which tracks need to be processed 4372 for (size_t i = 0; i < count; i++) { 4373 sp<Track> t = mActiveTracks[i].promote(); 4374 // The track died recently 4375 if (t == 0) { 4376 continue; 4377 } 4378 Track* const track = t.get(); 4379 audio_track_cblk_t* cblk = track->cblk(); 4380 // Only consider last track started for volume and mixer state control. 4381 // In theory an older track could underrun and restart after the new one starts 4382 // but as we only care about the transition phase between two tracks on a 4383 // direct output, it is not a problem to ignore the underrun case. 4384 sp<Track> l = mLatestActiveTrack.promote(); 4385 bool last = l.get() == track; 4386 4387 if (track->isInvalid()) { 4388 ALOGW("An invalidated track shouldn't be in active list"); 4389 tracksToRemove->add(track); 4390 continue; 4391 } 4392 4393 if (track->mState == TrackBase::IDLE) { 4394 ALOGW("An idle track shouldn't be in active list"); 4395 continue; 4396 } 4397 4398 if (track->isPausing()) { 4399 track->setPaused(); 4400 if (last) { 4401 if (!mHwPaused) { 4402 doHwPause = true; 4403 mHwPaused = true; 4404 } 4405 // If we were part way through writing the mixbuffer to 4406 // the HAL we must save this until we resume 4407 // BUG - this will be wrong if a different track is made active, 4408 // in that case we want to discard the pending data in the 4409 // mixbuffer and tell the client to present it again when the 4410 // track is resumed 4411 mPausedWriteLength = mCurrentWriteLength; 4412 mPausedBytesRemaining = mBytesRemaining; 4413 mBytesRemaining = 0; // stop writing 4414 } 4415 tracksToRemove->add(track); 4416 } else if (track->isFlushPending()) { 4417 track->flushAck(); 4418 if (last) { 4419 mFlushPending = true; 4420 } 4421 } else if (track->isResumePending()){ 4422 track->resumeAck(); 4423 if (last) { 4424 if (mPausedBytesRemaining) { 4425 // Need to continue write that was interrupted 4426 mCurrentWriteLength = mPausedWriteLength; 4427 mBytesRemaining = mPausedBytesRemaining; 4428 mPausedBytesRemaining = 0; 4429 } 4430 if (mHwPaused) { 4431 doHwResume = true; 4432 mHwPaused = false; 4433 // threadLoop_mix() will handle the case that we need to 4434 // resume an interrupted write 4435 } 4436 // enable write to audio HAL 4437 sleepTime = 0; 4438 4439 // Do not handle new data in this iteration even if track->framesReady() 4440 mixerStatus = MIXER_TRACKS_ENABLED; 4441 } 4442 } else if (track->framesReady() && track->isReady() && 4443 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4444 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4445 if (track->mFillingUpStatus == Track::FS_FILLED) { 4446 track->mFillingUpStatus = Track::FS_ACTIVE; 4447 // make sure processVolume_l() will apply new volume even if 0 4448 mLeftVolFloat = mRightVolFloat = -1.0; 4449 } 4450 4451 if (last) { 4452 sp<Track> previousTrack = mPreviousTrack.promote(); 4453 if (previousTrack != 0) { 4454 if (track != previousTrack.get()) { 4455 // Flush any data still being written from last track 4456 mBytesRemaining = 0; 4457 if (mPausedBytesRemaining) { 4458 // Last track was paused so we also need to flush saved 4459 // mixbuffer state and invalidate track so that it will 4460 // re-submit that unwritten data when it is next resumed 4461 mPausedBytesRemaining = 0; 4462 // Invalidate is a bit drastic - would be more efficient 4463 // to have a flag to tell client that some of the 4464 // previously written data was lost 4465 previousTrack->invalidate(); 4466 } 4467 // flush data already sent to the DSP if changing audio session as audio 4468 // comes from a different source. Also invalidate previous track to force a 4469 // seek when resuming. 4470 if (previousTrack->sessionId() != track->sessionId()) { 4471 previousTrack->invalidate(); 4472 } 4473 } 4474 } 4475 mPreviousTrack = track; 4476 // reset retry count 4477 track->mRetryCount = kMaxTrackRetriesOffload; 4478 mActiveTrack = t; 4479 mixerStatus = MIXER_TRACKS_READY; 4480 } 4481 } else { 4482 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4483 if (track->isStopping_1()) { 4484 // Hardware buffer can hold a large amount of audio so we must 4485 // wait for all current track's data to drain before we say 4486 // that the track is stopped. 4487 if (mBytesRemaining == 0) { 4488 // Only start draining when all data in mixbuffer 4489 // has been written 4490 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4491 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4492 // do not drain if no data was ever sent to HAL (mStandby == true) 4493 if (last && !mStandby) { 4494 // do not modify drain sequence if we are already draining. This happens 4495 // when resuming from pause after drain. 4496 if ((mDrainSequence & 1) == 0) { 4497 sleepTime = 0; 4498 standbyTime = systemTime() + standbyDelay; 4499 mixerStatus = MIXER_DRAIN_TRACK; 4500 mDrainSequence += 2; 4501 } 4502 if (mHwPaused) { 4503 // It is possible to move from PAUSED to STOPPING_1 without 4504 // a resume so we must ensure hardware is running 4505 doHwResume = true; 4506 mHwPaused = false; 4507 } 4508 } 4509 } 4510 } else if (track->isStopping_2()) { 4511 // Drain has completed or we are in standby, signal presentation complete 4512 if (!(mDrainSequence & 1) || !last || mStandby) { 4513 track->mState = TrackBase::STOPPED; 4514 size_t audioHALFrames = 4515 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4516 size_t framesWritten = 4517 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4518 track->presentationComplete(framesWritten, audioHALFrames); 4519 track->reset(); 4520 tracksToRemove->add(track); 4521 } 4522 } else { 4523 // No buffers for this track. Give it a few chances to 4524 // fill a buffer, then remove it from active list. 4525 if (--(track->mRetryCount) <= 0) { 4526 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4527 track->name()); 4528 tracksToRemove->add(track); 4529 // indicate to client process that the track was disabled because of underrun; 4530 // it will then automatically call start() when data is available 4531 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4532 } else if (last){ 4533 mixerStatus = MIXER_TRACKS_ENABLED; 4534 } 4535 } 4536 } 4537 // compute volume for this track 4538 processVolume_l(track, last); 4539 } 4540 4541 // make sure the pause/flush/resume sequence is executed in the right order. 4542 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4543 // before flush and then resume HW. This can happen in case of pause/flush/resume 4544 // if resume is received before pause is executed. 4545 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4546 mOutput->stream->pause(mOutput->stream); 4547 } 4548 if (mFlushPending) { 4549 flushHw_l(); 4550 mFlushPending = false; 4551 } 4552 if (!mStandby && doHwResume) { 4553 mOutput->stream->resume(mOutput->stream); 4554 } 4555 4556 // remove all the tracks that need to be... 4557 removeTracks_l(*tracksToRemove); 4558 4559 return mixerStatus; 4560} 4561 4562// must be called with thread mutex locked 4563bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4564{ 4565 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4566 mWriteAckSequence, mDrainSequence); 4567 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4568 return true; 4569 } 4570 return false; 4571} 4572 4573// must be called with thread mutex locked 4574bool AudioFlinger::OffloadThread::shouldStandby_l() 4575{ 4576 bool trackPaused = false; 4577 4578 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4579 // after a timeout and we will enter standby then. 4580 if (mTracks.size() > 0) { 4581 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4582 } 4583 4584 return !mStandby && !trackPaused; 4585} 4586 4587 4588bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4589{ 4590 Mutex::Autolock _l(mLock); 4591 return waitingAsyncCallback_l(); 4592} 4593 4594void AudioFlinger::OffloadThread::flushHw_l() 4595{ 4596 mOutput->stream->flush(mOutput->stream); 4597 // Flush anything still waiting in the mixbuffer 4598 mCurrentWriteLength = 0; 4599 mBytesRemaining = 0; 4600 mPausedWriteLength = 0; 4601 mPausedBytesRemaining = 0; 4602 mHwPaused = false; 4603 4604 if (mUseAsyncWrite) { 4605 // discard any pending drain or write ack by incrementing sequence 4606 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4607 mDrainSequence = (mDrainSequence + 2) & ~1; 4608 ALOG_ASSERT(mCallbackThread != 0); 4609 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4610 mCallbackThread->setDraining(mDrainSequence); 4611 } 4612} 4613 4614void AudioFlinger::OffloadThread::onAddNewTrack_l() 4615{ 4616 sp<Track> previousTrack = mPreviousTrack.promote(); 4617 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4618 4619 if (previousTrack != 0 && latestTrack != 0 && 4620 (previousTrack->sessionId() != latestTrack->sessionId())) { 4621 mFlushPending = true; 4622 } 4623 PlaybackThread::onAddNewTrack_l(); 4624} 4625 4626// ---------------------------------------------------------------------------- 4627 4628AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4629 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4630 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4631 DUPLICATING), 4632 mWaitTimeMs(UINT_MAX) 4633{ 4634 addOutputTrack(mainThread); 4635} 4636 4637AudioFlinger::DuplicatingThread::~DuplicatingThread() 4638{ 4639 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4640 mOutputTracks[i]->destroy(); 4641 } 4642} 4643 4644void AudioFlinger::DuplicatingThread::threadLoop_mix() 4645{ 4646 // mix buffers... 4647 if (outputsReady(outputTracks)) { 4648 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4649 } else { 4650 memset(mSinkBuffer, 0, mSinkBufferSize); 4651 } 4652 sleepTime = 0; 4653 writeFrames = mNormalFrameCount; 4654 mCurrentWriteLength = mSinkBufferSize; 4655 standbyTime = systemTime() + standbyDelay; 4656} 4657 4658void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4659{ 4660 if (sleepTime == 0) { 4661 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4662 sleepTime = activeSleepTime; 4663 } else { 4664 sleepTime = idleSleepTime; 4665 } 4666 } else if (mBytesWritten != 0) { 4667 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4668 writeFrames = mNormalFrameCount; 4669 memset(mSinkBuffer, 0, mSinkBufferSize); 4670 } else { 4671 // flush remaining overflow buffers in output tracks 4672 writeFrames = 0; 4673 } 4674 sleepTime = 0; 4675 } 4676} 4677 4678ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4679{ 4680 for (size_t i = 0; i < outputTracks.size(); i++) { 4681 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4682 // for delivery downstream as needed. This in-place conversion is safe as 4683 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4684 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4685 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4686 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4687 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4688 } 4689 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4690 } 4691 mStandby = false; 4692 return (ssize_t)mSinkBufferSize; 4693} 4694 4695void AudioFlinger::DuplicatingThread::threadLoop_standby() 4696{ 4697 // DuplicatingThread implements standby by stopping all tracks 4698 for (size_t i = 0; i < outputTracks.size(); i++) { 4699 outputTracks[i]->stop(); 4700 } 4701} 4702 4703void AudioFlinger::DuplicatingThread::saveOutputTracks() 4704{ 4705 outputTracks = mOutputTracks; 4706} 4707 4708void AudioFlinger::DuplicatingThread::clearOutputTracks() 4709{ 4710 outputTracks.clear(); 4711} 4712 4713void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4714{ 4715 Mutex::Autolock _l(mLock); 4716 // FIXME explain this formula 4717 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4718 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4719 // due to current usage case and restrictions on the AudioBufferProvider. 4720 // Actual buffer conversion is done in threadLoop_write(). 4721 // 4722 // TODO: This may change in the future, depending on multichannel 4723 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4724 OutputTrack *outputTrack = new OutputTrack(thread, 4725 this, 4726 mSampleRate, 4727 AUDIO_FORMAT_PCM_16_BIT, 4728 mChannelMask, 4729 frameCount, 4730 IPCThreadState::self()->getCallingUid()); 4731 if (outputTrack->cblk() != NULL) { 4732 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4733 mOutputTracks.add(outputTrack); 4734 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4735 updateWaitTime_l(); 4736 } 4737} 4738 4739void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4740{ 4741 Mutex::Autolock _l(mLock); 4742 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4743 if (mOutputTracks[i]->thread() == thread) { 4744 mOutputTracks[i]->destroy(); 4745 mOutputTracks.removeAt(i); 4746 updateWaitTime_l(); 4747 return; 4748 } 4749 } 4750 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4751} 4752 4753// caller must hold mLock 4754void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4755{ 4756 mWaitTimeMs = UINT_MAX; 4757 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4758 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4759 if (strong != 0) { 4760 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4761 if (waitTimeMs < mWaitTimeMs) { 4762 mWaitTimeMs = waitTimeMs; 4763 } 4764 } 4765 } 4766} 4767 4768 4769bool AudioFlinger::DuplicatingThread::outputsReady( 4770 const SortedVector< sp<OutputTrack> > &outputTracks) 4771{ 4772 for (size_t i = 0; i < outputTracks.size(); i++) { 4773 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4774 if (thread == 0) { 4775 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4776 outputTracks[i].get()); 4777 return false; 4778 } 4779 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4780 // see note at standby() declaration 4781 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4782 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4783 thread.get()); 4784 return false; 4785 } 4786 } 4787 return true; 4788} 4789 4790uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4791{ 4792 return (mWaitTimeMs * 1000) / 2; 4793} 4794 4795void AudioFlinger::DuplicatingThread::cacheParameters_l() 4796{ 4797 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4798 updateWaitTime_l(); 4799 4800 MixerThread::cacheParameters_l(); 4801} 4802 4803// ---------------------------------------------------------------------------- 4804// Record 4805// ---------------------------------------------------------------------------- 4806 4807AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4808 AudioStreamIn *input, 4809 audio_io_handle_t id, 4810 audio_devices_t outDevice, 4811 audio_devices_t inDevice 4812#ifdef TEE_SINK 4813 , const sp<NBAIO_Sink>& teeSink 4814#endif 4815 ) : 4816 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4817 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4818 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4819 mRsmpInRear(0) 4820#ifdef TEE_SINK 4821 , mTeeSink(teeSink) 4822#endif 4823 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4824 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4825 // mFastCapture below 4826 , mFastCaptureFutex(0) 4827 // mInputSource 4828 // mPipeSink 4829 // mPipeSource 4830 , mPipeFramesP2(0) 4831 // mPipeMemory 4832 // mFastCaptureNBLogWriter 4833 , mFastTrackAvail(false) 4834{ 4835 snprintf(mName, kNameLength, "AudioIn_%X", id); 4836 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4837 4838 readInputParameters_l(); 4839 4840 // create an NBAIO source for the HAL input stream, and negotiate 4841 mInputSource = new AudioStreamInSource(input->stream); 4842 size_t numCounterOffers = 0; 4843 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4844 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4845 ALOG_ASSERT(index == 0); 4846 4847 // initialize fast capture depending on configuration 4848 bool initFastCapture; 4849 switch (kUseFastCapture) { 4850 case FastCapture_Never: 4851 initFastCapture = false; 4852 break; 4853 case FastCapture_Always: 4854 initFastCapture = true; 4855 break; 4856 case FastCapture_Static: 4857 uint32_t primaryOutputSampleRate; 4858 { 4859 AutoMutex _l(audioFlinger->mHardwareLock); 4860 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4861 } 4862 initFastCapture = 4863 // either capture sample rate is same as (a reasonable) primary output sample rate 4864 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4865 (mSampleRate == primaryOutputSampleRate)) || 4866 // or primary output sample rate is unknown, and capture sample rate is reasonable 4867 ((primaryOutputSampleRate == 0) && 4868 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4869 // and the buffer size is < 12 ms 4870 (mFrameCount * 1000) / mSampleRate < 12; 4871 break; 4872 // case FastCapture_Dynamic: 4873 } 4874 4875 if (initFastCapture) { 4876 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4877 NBAIO_Format format = mInputSource->format(); 4878 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4879 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4880 void *pipeBuffer; 4881 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4882 sp<IMemory> pipeMemory; 4883 if ((roHeap == 0) || 4884 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4885 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4886 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4887 goto failed; 4888 } 4889 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4890 memset(pipeBuffer, 0, pipeSize); 4891 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4892 const NBAIO_Format offers[1] = {format}; 4893 size_t numCounterOffers = 0; 4894 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4895 ALOG_ASSERT(index == 0); 4896 mPipeSink = pipe; 4897 PipeReader *pipeReader = new PipeReader(*pipe); 4898 numCounterOffers = 0; 4899 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4900 ALOG_ASSERT(index == 0); 4901 mPipeSource = pipeReader; 4902 mPipeFramesP2 = pipeFramesP2; 4903 mPipeMemory = pipeMemory; 4904 4905 // create fast capture 4906 mFastCapture = new FastCapture(); 4907 FastCaptureStateQueue *sq = mFastCapture->sq(); 4908#ifdef STATE_QUEUE_DUMP 4909 // FIXME 4910#endif 4911 FastCaptureState *state = sq->begin(); 4912 state->mCblk = NULL; 4913 state->mInputSource = mInputSource.get(); 4914 state->mInputSourceGen++; 4915 state->mPipeSink = pipe; 4916 state->mPipeSinkGen++; 4917 state->mFrameCount = mFrameCount; 4918 state->mCommand = FastCaptureState::COLD_IDLE; 4919 // already done in constructor initialization list 4920 //mFastCaptureFutex = 0; 4921 state->mColdFutexAddr = &mFastCaptureFutex; 4922 state->mColdGen++; 4923 state->mDumpState = &mFastCaptureDumpState; 4924#ifdef TEE_SINK 4925 // FIXME 4926#endif 4927 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4928 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4929 sq->end(); 4930 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4931 4932 // start the fast capture 4933 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4934 pid_t tid = mFastCapture->getTid(); 4935 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4936 if (err != 0) { 4937 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4938 kPriorityFastCapture, getpid_cached, tid, err); 4939 } 4940 4941#ifdef AUDIO_WATCHDOG 4942 // FIXME 4943#endif 4944 4945 mFastTrackAvail = true; 4946 } 4947failed: ; 4948 4949 // FIXME mNormalSource 4950} 4951 4952 4953AudioFlinger::RecordThread::~RecordThread() 4954{ 4955 if (mFastCapture != 0) { 4956 FastCaptureStateQueue *sq = mFastCapture->sq(); 4957 FastCaptureState *state = sq->begin(); 4958 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4959 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4960 if (old == -1) { 4961 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4962 } 4963 } 4964 state->mCommand = FastCaptureState::EXIT; 4965 sq->end(); 4966 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4967 mFastCapture->join(); 4968 mFastCapture.clear(); 4969 } 4970 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4971 mAudioFlinger->unregisterWriter(mNBLogWriter); 4972 delete[] mRsmpInBuffer; 4973} 4974 4975void AudioFlinger::RecordThread::onFirstRef() 4976{ 4977 run(mName, PRIORITY_URGENT_AUDIO); 4978} 4979 4980bool AudioFlinger::RecordThread::threadLoop() 4981{ 4982 nsecs_t lastWarning = 0; 4983 4984 inputStandBy(); 4985 4986reacquire_wakelock: 4987 sp<RecordTrack> activeTrack; 4988 int activeTracksGen; 4989 { 4990 Mutex::Autolock _l(mLock); 4991 size_t size = mActiveTracks.size(); 4992 activeTracksGen = mActiveTracksGen; 4993 if (size > 0) { 4994 // FIXME an arbitrary choice 4995 activeTrack = mActiveTracks[0]; 4996 acquireWakeLock_l(activeTrack->uid()); 4997 if (size > 1) { 4998 SortedVector<int> tmp; 4999 for (size_t i = 0; i < size; i++) { 5000 tmp.add(mActiveTracks[i]->uid()); 5001 } 5002 updateWakeLockUids_l(tmp); 5003 } 5004 } else { 5005 acquireWakeLock_l(-1); 5006 } 5007 } 5008 5009 // used to request a deferred sleep, to be executed later while mutex is unlocked 5010 uint32_t sleepUs = 0; 5011 5012 // loop while there is work to do 5013 for (;;) { 5014 Vector< sp<EffectChain> > effectChains; 5015 5016 // sleep with mutex unlocked 5017 if (sleepUs > 0) { 5018 usleep(sleepUs); 5019 sleepUs = 0; 5020 } 5021 5022 // activeTracks accumulates a copy of a subset of mActiveTracks 5023 Vector< sp<RecordTrack> > activeTracks; 5024 5025 // reference to the (first and only) active fast track 5026 sp<RecordTrack> fastTrack; 5027 5028 // reference to a fast track which is about to be removed 5029 sp<RecordTrack> fastTrackToRemove; 5030 5031 { // scope for mLock 5032 Mutex::Autolock _l(mLock); 5033 5034 processConfigEvents_l(); 5035 5036 // check exitPending here because checkForNewParameters_l() and 5037 // checkForNewParameters_l() can temporarily release mLock 5038 if (exitPending()) { 5039 break; 5040 } 5041 5042 // if no active track(s), then standby and release wakelock 5043 size_t size = mActiveTracks.size(); 5044 if (size == 0) { 5045 standbyIfNotAlreadyInStandby(); 5046 // exitPending() can't become true here 5047 releaseWakeLock_l(); 5048 ALOGV("RecordThread: loop stopping"); 5049 // go to sleep 5050 mWaitWorkCV.wait(mLock); 5051 ALOGV("RecordThread: loop starting"); 5052 goto reacquire_wakelock; 5053 } 5054 5055 if (mActiveTracksGen != activeTracksGen) { 5056 activeTracksGen = mActiveTracksGen; 5057 SortedVector<int> tmp; 5058 for (size_t i = 0; i < size; i++) { 5059 tmp.add(mActiveTracks[i]->uid()); 5060 } 5061 updateWakeLockUids_l(tmp); 5062 } 5063 5064 bool doBroadcast = false; 5065 for (size_t i = 0; i < size; ) { 5066 5067 activeTrack = mActiveTracks[i]; 5068 if (activeTrack->isTerminated()) { 5069 if (activeTrack->isFastTrack()) { 5070 ALOG_ASSERT(fastTrackToRemove == 0); 5071 fastTrackToRemove = activeTrack; 5072 } 5073 removeTrack_l(activeTrack); 5074 mActiveTracks.remove(activeTrack); 5075 mActiveTracksGen++; 5076 size--; 5077 continue; 5078 } 5079 5080 TrackBase::track_state activeTrackState = activeTrack->mState; 5081 switch (activeTrackState) { 5082 5083 case TrackBase::PAUSING: 5084 mActiveTracks.remove(activeTrack); 5085 mActiveTracksGen++; 5086 doBroadcast = true; 5087 size--; 5088 continue; 5089 5090 case TrackBase::STARTING_1: 5091 sleepUs = 10000; 5092 i++; 5093 continue; 5094 5095 case TrackBase::STARTING_2: 5096 doBroadcast = true; 5097 mStandby = false; 5098 activeTrack->mState = TrackBase::ACTIVE; 5099 break; 5100 5101 case TrackBase::ACTIVE: 5102 break; 5103 5104 case TrackBase::IDLE: 5105 i++; 5106 continue; 5107 5108 default: 5109 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5110 } 5111 5112 activeTracks.add(activeTrack); 5113 i++; 5114 5115 if (activeTrack->isFastTrack()) { 5116 ALOG_ASSERT(!mFastTrackAvail); 5117 ALOG_ASSERT(fastTrack == 0); 5118 fastTrack = activeTrack; 5119 } 5120 } 5121 if (doBroadcast) { 5122 mStartStopCond.broadcast(); 5123 } 5124 5125 // sleep if there are no active tracks to process 5126 if (activeTracks.size() == 0) { 5127 if (sleepUs == 0) { 5128 sleepUs = kRecordThreadSleepUs; 5129 } 5130 continue; 5131 } 5132 sleepUs = 0; 5133 5134 lockEffectChains_l(effectChains); 5135 } 5136 5137 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5138 5139 size_t size = effectChains.size(); 5140 for (size_t i = 0; i < size; i++) { 5141 // thread mutex is not locked, but effect chain is locked 5142 effectChains[i]->process_l(); 5143 } 5144 5145 // Push a new fast capture state if fast capture is not already running, or cblk change 5146 if (mFastCapture != 0) { 5147 FastCaptureStateQueue *sq = mFastCapture->sq(); 5148 FastCaptureState *state = sq->begin(); 5149 bool didModify = false; 5150 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5151 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5152 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5153 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5154 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5155 if (old == -1) { 5156 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5157 } 5158 } 5159 state->mCommand = FastCaptureState::READ_WRITE; 5160#if 0 // FIXME 5161 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5162 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5163#endif 5164 didModify = true; 5165 } 5166 audio_track_cblk_t *cblkOld = state->mCblk; 5167 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5168 if (cblkNew != cblkOld) { 5169 state->mCblk = cblkNew; 5170 // block until acked if removing a fast track 5171 if (cblkOld != NULL) { 5172 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5173 } 5174 didModify = true; 5175 } 5176 sq->end(didModify); 5177 if (didModify) { 5178 sq->push(block); 5179#if 0 5180 if (kUseFastCapture == FastCapture_Dynamic) { 5181 mNormalSource = mPipeSource; 5182 } 5183#endif 5184 } 5185 } 5186 5187 // now run the fast track destructor with thread mutex unlocked 5188 fastTrackToRemove.clear(); 5189 5190 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5191 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5192 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5193 // If destination is non-contiguous, first read past the nominal end of buffer, then 5194 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5195 5196 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5197 ssize_t framesRead; 5198 5199 // If an NBAIO source is present, use it to read the normal capture's data 5200 if (mPipeSource != 0) { 5201 size_t framesToRead = mBufferSize / mFrameSize; 5202 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5203 framesToRead, AudioBufferProvider::kInvalidPTS); 5204 if (framesRead == 0) { 5205 // since pipe is non-blocking, simulate blocking input 5206 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5207 } 5208 // otherwise use the HAL / AudioStreamIn directly 5209 } else { 5210 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5211 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5212 if (bytesRead < 0) { 5213 framesRead = bytesRead; 5214 } else { 5215 framesRead = bytesRead / mFrameSize; 5216 } 5217 } 5218 5219 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5220 ALOGE("read failed: framesRead=%d", framesRead); 5221 // Force input into standby so that it tries to recover at next read attempt 5222 inputStandBy(); 5223 sleepUs = kRecordThreadSleepUs; 5224 } 5225 if (framesRead <= 0) { 5226 goto unlock; 5227 } 5228 ALOG_ASSERT(framesRead > 0); 5229 5230 if (mTeeSink != 0) { 5231 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5232 } 5233 // If destination is non-contiguous, we now correct for reading past end of buffer. 5234 { 5235 size_t part1 = mRsmpInFramesP2 - rear; 5236 if ((size_t) framesRead > part1) { 5237 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5238 (framesRead - part1) * mFrameSize); 5239 } 5240 } 5241 rear = mRsmpInRear += framesRead; 5242 5243 size = activeTracks.size(); 5244 // loop over each active track 5245 for (size_t i = 0; i < size; i++) { 5246 activeTrack = activeTracks[i]; 5247 5248 // skip fast tracks, as those are handled directly by FastCapture 5249 if (activeTrack->isFastTrack()) { 5250 continue; 5251 } 5252 5253 enum { 5254 OVERRUN_UNKNOWN, 5255 OVERRUN_TRUE, 5256 OVERRUN_FALSE 5257 } overrun = OVERRUN_UNKNOWN; 5258 5259 // loop over getNextBuffer to handle circular sink 5260 for (;;) { 5261 5262 activeTrack->mSink.frameCount = ~0; 5263 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5264 size_t framesOut = activeTrack->mSink.frameCount; 5265 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5266 5267 int32_t front = activeTrack->mRsmpInFront; 5268 ssize_t filled = rear - front; 5269 size_t framesIn; 5270 5271 if (filled < 0) { 5272 // should not happen, but treat like a massive overrun and re-sync 5273 framesIn = 0; 5274 activeTrack->mRsmpInFront = rear; 5275 overrun = OVERRUN_TRUE; 5276 } else if ((size_t) filled <= mRsmpInFrames) { 5277 framesIn = (size_t) filled; 5278 } else { 5279 // client is not keeping up with server, but give it latest data 5280 framesIn = mRsmpInFrames; 5281 activeTrack->mRsmpInFront = front = rear - framesIn; 5282 overrun = OVERRUN_TRUE; 5283 } 5284 5285 if (framesOut == 0 || framesIn == 0) { 5286 break; 5287 } 5288 5289 if (activeTrack->mResampler == NULL) { 5290 // no resampling 5291 if (framesIn > framesOut) { 5292 framesIn = framesOut; 5293 } else { 5294 framesOut = framesIn; 5295 } 5296 int8_t *dst = activeTrack->mSink.i8; 5297 while (framesIn > 0) { 5298 front &= mRsmpInFramesP2 - 1; 5299 size_t part1 = mRsmpInFramesP2 - front; 5300 if (part1 > framesIn) { 5301 part1 = framesIn; 5302 } 5303 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5304 if (mChannelCount == activeTrack->mChannelCount) { 5305 memcpy(dst, src, part1 * mFrameSize); 5306 } else if (mChannelCount == 1) { 5307 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5308 part1); 5309 } else { 5310 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5311 part1); 5312 } 5313 dst += part1 * activeTrack->mFrameSize; 5314 front += part1; 5315 framesIn -= part1; 5316 } 5317 activeTrack->mRsmpInFront += framesOut; 5318 5319 } else { 5320 // resampling 5321 // FIXME framesInNeeded should really be part of resampler API, and should 5322 // depend on the SRC ratio 5323 // to keep mRsmpInBuffer full so resampler always has sufficient input 5324 size_t framesInNeeded; 5325 // FIXME only re-calculate when it changes, and optimize for common ratios 5326 // Do not precompute in/out because floating point is not associative 5327 // e.g. a*b/c != a*(b/c). 5328 const double in(mSampleRate); 5329 const double out(activeTrack->mSampleRate); 5330 framesInNeeded = ceil(framesOut * in / out) + 1; 5331 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5332 framesInNeeded, framesOut, in / out); 5333 // Although we theoretically have framesIn in circular buffer, some of those are 5334 // unreleased frames, and thus must be discounted for purpose of budgeting. 5335 size_t unreleased = activeTrack->mRsmpInUnrel; 5336 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5337 if (framesIn < framesInNeeded) { 5338 ALOGV("not enough to resample: have %u frames in but need %u in to " 5339 "produce %u out given in/out ratio of %.4g", 5340 framesIn, framesInNeeded, framesOut, in / out); 5341 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5342 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5343 if (newFramesOut == 0) { 5344 break; 5345 } 5346 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5347 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5348 framesInNeeded, newFramesOut, out / in); 5349 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5350 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5351 "given in/out ratio of %.4g", 5352 framesIn, framesInNeeded, newFramesOut, in / out); 5353 framesOut = newFramesOut; 5354 } else { 5355 ALOGV("success 1: have %u in and need %u in to produce %u out " 5356 "given in/out ratio of %.4g", 5357 framesIn, framesInNeeded, framesOut, in / out); 5358 } 5359 5360 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5361 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5362 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5363 delete[] activeTrack->mRsmpOutBuffer; 5364 // resampler always outputs stereo 5365 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5366 activeTrack->mRsmpOutFrameCount = framesOut; 5367 } 5368 5369 // resampler accumulates, but we only have one source track 5370 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5371 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5372 // FIXME how about having activeTrack implement this interface itself? 5373 activeTrack->mResamplerBufferProvider 5374 /*this*/ /* AudioBufferProvider* */); 5375 // ditherAndClamp() works as long as all buffers returned by 5376 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5377 if (activeTrack->mChannelCount == 1) { 5378 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5379 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5380 framesOut); 5381 // the resampler always outputs stereo samples: 5382 // do post stereo to mono conversion 5383 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5384 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5385 } else { 5386 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5387 activeTrack->mRsmpOutBuffer, framesOut); 5388 } 5389 // now done with mRsmpOutBuffer 5390 5391 } 5392 5393 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5394 overrun = OVERRUN_FALSE; 5395 } 5396 5397 if (activeTrack->mFramesToDrop == 0) { 5398 if (framesOut > 0) { 5399 activeTrack->mSink.frameCount = framesOut; 5400 activeTrack->releaseBuffer(&activeTrack->mSink); 5401 } 5402 } else { 5403 // FIXME could do a partial drop of framesOut 5404 if (activeTrack->mFramesToDrop > 0) { 5405 activeTrack->mFramesToDrop -= framesOut; 5406 if (activeTrack->mFramesToDrop <= 0) { 5407 activeTrack->clearSyncStartEvent(); 5408 } 5409 } else { 5410 activeTrack->mFramesToDrop += framesOut; 5411 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5412 activeTrack->mSyncStartEvent->isCancelled()) { 5413 ALOGW("Synced record %s, session %d, trigger session %d", 5414 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5415 activeTrack->sessionId(), 5416 (activeTrack->mSyncStartEvent != 0) ? 5417 activeTrack->mSyncStartEvent->triggerSession() : 0); 5418 activeTrack->clearSyncStartEvent(); 5419 } 5420 } 5421 } 5422 5423 if (framesOut == 0) { 5424 break; 5425 } 5426 } 5427 5428 switch (overrun) { 5429 case OVERRUN_TRUE: 5430 // client isn't retrieving buffers fast enough 5431 if (!activeTrack->setOverflow()) { 5432 nsecs_t now = systemTime(); 5433 // FIXME should lastWarning per track? 5434 if ((now - lastWarning) > kWarningThrottleNs) { 5435 ALOGW("RecordThread: buffer overflow"); 5436 lastWarning = now; 5437 } 5438 } 5439 break; 5440 case OVERRUN_FALSE: 5441 activeTrack->clearOverflow(); 5442 break; 5443 case OVERRUN_UNKNOWN: 5444 break; 5445 } 5446 5447 } 5448 5449unlock: 5450 // enable changes in effect chain 5451 unlockEffectChains(effectChains); 5452 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5453 } 5454 5455 standbyIfNotAlreadyInStandby(); 5456 5457 { 5458 Mutex::Autolock _l(mLock); 5459 for (size_t i = 0; i < mTracks.size(); i++) { 5460 sp<RecordTrack> track = mTracks[i]; 5461 track->invalidate(); 5462 } 5463 mActiveTracks.clear(); 5464 mActiveTracksGen++; 5465 mStartStopCond.broadcast(); 5466 } 5467 5468 releaseWakeLock(); 5469 5470 ALOGV("RecordThread %p exiting", this); 5471 return false; 5472} 5473 5474void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5475{ 5476 if (!mStandby) { 5477 inputStandBy(); 5478 mStandby = true; 5479 } 5480} 5481 5482void AudioFlinger::RecordThread::inputStandBy() 5483{ 5484 // Idle the fast capture if it's currently running 5485 if (mFastCapture != 0) { 5486 FastCaptureStateQueue *sq = mFastCapture->sq(); 5487 FastCaptureState *state = sq->begin(); 5488 if (!(state->mCommand & FastCaptureState::IDLE)) { 5489 state->mCommand = FastCaptureState::COLD_IDLE; 5490 state->mColdFutexAddr = &mFastCaptureFutex; 5491 state->mColdGen++; 5492 mFastCaptureFutex = 0; 5493 sq->end(); 5494 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5495 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5496#if 0 5497 if (kUseFastCapture == FastCapture_Dynamic) { 5498 // FIXME 5499 } 5500#endif 5501#ifdef AUDIO_WATCHDOG 5502 // FIXME 5503#endif 5504 } else { 5505 sq->end(false /*didModify*/); 5506 } 5507 } 5508 mInput->stream->common.standby(&mInput->stream->common); 5509} 5510 5511// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5512sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5513 const sp<AudioFlinger::Client>& client, 5514 uint32_t sampleRate, 5515 audio_format_t format, 5516 audio_channel_mask_t channelMask, 5517 size_t *pFrameCount, 5518 int sessionId, 5519 size_t *notificationFrames, 5520 int uid, 5521 IAudioFlinger::track_flags_t *flags, 5522 pid_t tid, 5523 status_t *status) 5524{ 5525 size_t frameCount = *pFrameCount; 5526 sp<RecordTrack> track; 5527 status_t lStatus; 5528 5529 // client expresses a preference for FAST, but we get the final say 5530 if (*flags & IAudioFlinger::TRACK_FAST) { 5531 if ( 5532 // use case: callback handler 5533 (tid != -1) && 5534 // frame count is not specified, or is exactly the pipe depth 5535 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5536 // PCM data 5537 audio_is_linear_pcm(format) && 5538 // native format 5539 (format == mFormat) && 5540 // native channel mask 5541 (channelMask == mChannelMask) && 5542 // native hardware sample rate 5543 (sampleRate == mSampleRate) && 5544 // record thread has an associated fast capture 5545 hasFastCapture() && 5546 // there are sufficient fast track slots available 5547 mFastTrackAvail 5548 ) { 5549 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5550 frameCount, mFrameCount); 5551 } else { 5552 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5553 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5554 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5555 frameCount, mFrameCount, mPipeFramesP2, 5556 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5557 hasFastCapture(), tid, mFastTrackAvail); 5558 *flags &= ~IAudioFlinger::TRACK_FAST; 5559 } 5560 } 5561 5562 // compute track buffer size in frames, and suggest the notification frame count 5563 if (*flags & IAudioFlinger::TRACK_FAST) { 5564 // fast track: frame count is exactly the pipe depth 5565 frameCount = mPipeFramesP2; 5566 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5567 *notificationFrames = mFrameCount; 5568 } else { 5569 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5570 // or 20 ms if there is a fast capture 5571 // TODO This could be a roundupRatio inline, and const 5572 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5573 * sampleRate + mSampleRate - 1) / mSampleRate; 5574 // minimum number of notification periods is at least kMinNotifications, 5575 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5576 static const size_t kMinNotifications = 3; 5577 static const uint32_t kMinMs = 30; 5578 // TODO This could be a roundupRatio inline 5579 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5580 // TODO This could be a roundupRatio inline 5581 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5582 maxNotificationFrames; 5583 const size_t minFrameCount = maxNotificationFrames * 5584 max(kMinNotifications, minNotificationsByMs); 5585 frameCount = max(frameCount, minFrameCount); 5586 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5587 *notificationFrames = maxNotificationFrames; 5588 } 5589 } 5590 *pFrameCount = frameCount; 5591 5592 lStatus = initCheck(); 5593 if (lStatus != NO_ERROR) { 5594 ALOGE("createRecordTrack_l() audio driver not initialized"); 5595 goto Exit; 5596 } 5597 5598 { // scope for mLock 5599 Mutex::Autolock _l(mLock); 5600 5601 track = new RecordTrack(this, client, sampleRate, 5602 format, channelMask, frameCount, NULL, sessionId, uid, 5603 *flags, TrackBase::TYPE_DEFAULT); 5604 5605 lStatus = track->initCheck(); 5606 if (lStatus != NO_ERROR) { 5607 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5608 // track must be cleared from the caller as the caller has the AF lock 5609 goto Exit; 5610 } 5611 mTracks.add(track); 5612 5613 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5614 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5615 mAudioFlinger->btNrecIsOff(); 5616 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5617 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5618 5619 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5620 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5621 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5622 // so ask activity manager to do this on our behalf 5623 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5624 } 5625 } 5626 5627 lStatus = NO_ERROR; 5628 5629Exit: 5630 *status = lStatus; 5631 return track; 5632} 5633 5634status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5635 AudioSystem::sync_event_t event, 5636 int triggerSession) 5637{ 5638 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5639 sp<ThreadBase> strongMe = this; 5640 status_t status = NO_ERROR; 5641 5642 if (event == AudioSystem::SYNC_EVENT_NONE) { 5643 recordTrack->clearSyncStartEvent(); 5644 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5645 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5646 triggerSession, 5647 recordTrack->sessionId(), 5648 syncStartEventCallback, 5649 recordTrack); 5650 // Sync event can be cancelled by the trigger session if the track is not in a 5651 // compatible state in which case we start record immediately 5652 if (recordTrack->mSyncStartEvent->isCancelled()) { 5653 recordTrack->clearSyncStartEvent(); 5654 } else { 5655 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5656 recordTrack->mFramesToDrop = - 5657 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5658 } 5659 } 5660 5661 { 5662 // This section is a rendezvous between binder thread executing start() and RecordThread 5663 AutoMutex lock(mLock); 5664 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5665 if (recordTrack->mState == TrackBase::PAUSING) { 5666 ALOGV("active record track PAUSING -> ACTIVE"); 5667 recordTrack->mState = TrackBase::ACTIVE; 5668 } else { 5669 ALOGV("active record track state %d", recordTrack->mState); 5670 } 5671 return status; 5672 } 5673 5674 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5675 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5676 // or using a separate command thread 5677 recordTrack->mState = TrackBase::STARTING_1; 5678 mActiveTracks.add(recordTrack); 5679 mActiveTracksGen++; 5680 status_t status = NO_ERROR; 5681 if (recordTrack->isExternalTrack()) { 5682 mLock.unlock(); 5683 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5684 mLock.lock(); 5685 // FIXME should verify that recordTrack is still in mActiveTracks 5686 if (status != NO_ERROR) { 5687 mActiveTracks.remove(recordTrack); 5688 mActiveTracksGen++; 5689 recordTrack->clearSyncStartEvent(); 5690 ALOGV("RecordThread::start error %d", status); 5691 return status; 5692 } 5693 } 5694 // Catch up with current buffer indices if thread is already running. 5695 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5696 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5697 // see previously buffered data before it called start(), but with greater risk of overrun. 5698 5699 recordTrack->mRsmpInFront = mRsmpInRear; 5700 recordTrack->mRsmpInUnrel = 0; 5701 // FIXME why reset? 5702 if (recordTrack->mResampler != NULL) { 5703 recordTrack->mResampler->reset(); 5704 } 5705 recordTrack->mState = TrackBase::STARTING_2; 5706 // signal thread to start 5707 mWaitWorkCV.broadcast(); 5708 if (mActiveTracks.indexOf(recordTrack) < 0) { 5709 ALOGV("Record failed to start"); 5710 status = BAD_VALUE; 5711 goto startError; 5712 } 5713 return status; 5714 } 5715 5716startError: 5717 if (recordTrack->isExternalTrack()) { 5718 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5719 } 5720 recordTrack->clearSyncStartEvent(); 5721 // FIXME I wonder why we do not reset the state here? 5722 return status; 5723} 5724 5725void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5726{ 5727 sp<SyncEvent> strongEvent = event.promote(); 5728 5729 if (strongEvent != 0) { 5730 sp<RefBase> ptr = strongEvent->cookie().promote(); 5731 if (ptr != 0) { 5732 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5733 recordTrack->handleSyncStartEvent(strongEvent); 5734 } 5735 } 5736} 5737 5738bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5739 ALOGV("RecordThread::stop"); 5740 AutoMutex _l(mLock); 5741 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5742 return false; 5743 } 5744 // note that threadLoop may still be processing the track at this point [without lock] 5745 recordTrack->mState = TrackBase::PAUSING; 5746 // do not wait for mStartStopCond if exiting 5747 if (exitPending()) { 5748 return true; 5749 } 5750 // FIXME incorrect usage of wait: no explicit predicate or loop 5751 mStartStopCond.wait(mLock); 5752 // if we have been restarted, recordTrack is in mActiveTracks here 5753 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5754 ALOGV("Record stopped OK"); 5755 return true; 5756 } 5757 return false; 5758} 5759 5760bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5761{ 5762 return false; 5763} 5764 5765status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5766{ 5767#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5768 if (!isValidSyncEvent(event)) { 5769 return BAD_VALUE; 5770 } 5771 5772 int eventSession = event->triggerSession(); 5773 status_t ret = NAME_NOT_FOUND; 5774 5775 Mutex::Autolock _l(mLock); 5776 5777 for (size_t i = 0; i < mTracks.size(); i++) { 5778 sp<RecordTrack> track = mTracks[i]; 5779 if (eventSession == track->sessionId()) { 5780 (void) track->setSyncEvent(event); 5781 ret = NO_ERROR; 5782 } 5783 } 5784 return ret; 5785#else 5786 return BAD_VALUE; 5787#endif 5788} 5789 5790// destroyTrack_l() must be called with ThreadBase::mLock held 5791void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5792{ 5793 track->terminate(); 5794 track->mState = TrackBase::STOPPED; 5795 // active tracks are removed by threadLoop() 5796 if (mActiveTracks.indexOf(track) < 0) { 5797 removeTrack_l(track); 5798 } 5799} 5800 5801void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5802{ 5803 mTracks.remove(track); 5804 // need anything related to effects here? 5805 if (track->isFastTrack()) { 5806 ALOG_ASSERT(!mFastTrackAvail); 5807 mFastTrackAvail = true; 5808 } 5809} 5810 5811void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5812{ 5813 dumpInternals(fd, args); 5814 dumpTracks(fd, args); 5815 dumpEffectChains(fd, args); 5816} 5817 5818void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5819{ 5820 dprintf(fd, "\nInput thread %p:\n", this); 5821 5822 if (mActiveTracks.size() > 0) { 5823 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5824 } else { 5825 dprintf(fd, " No active record clients\n"); 5826 } 5827 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5828 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5829 5830 dumpBase(fd, args); 5831} 5832 5833void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5834{ 5835 const size_t SIZE = 256; 5836 char buffer[SIZE]; 5837 String8 result; 5838 5839 size_t numtracks = mTracks.size(); 5840 size_t numactive = mActiveTracks.size(); 5841 size_t numactiveseen = 0; 5842 dprintf(fd, " %d Tracks", numtracks); 5843 if (numtracks) { 5844 dprintf(fd, " of which %d are active\n", numactive); 5845 RecordTrack::appendDumpHeader(result); 5846 for (size_t i = 0; i < numtracks ; ++i) { 5847 sp<RecordTrack> track = mTracks[i]; 5848 if (track != 0) { 5849 bool active = mActiveTracks.indexOf(track) >= 0; 5850 if (active) { 5851 numactiveseen++; 5852 } 5853 track->dump(buffer, SIZE, active); 5854 result.append(buffer); 5855 } 5856 } 5857 } else { 5858 dprintf(fd, "\n"); 5859 } 5860 5861 if (numactiveseen != numactive) { 5862 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5863 " not in the track list\n"); 5864 result.append(buffer); 5865 RecordTrack::appendDumpHeader(result); 5866 for (size_t i = 0; i < numactive; ++i) { 5867 sp<RecordTrack> track = mActiveTracks[i]; 5868 if (mTracks.indexOf(track) < 0) { 5869 track->dump(buffer, SIZE, true); 5870 result.append(buffer); 5871 } 5872 } 5873 5874 } 5875 write(fd, result.string(), result.size()); 5876} 5877 5878// AudioBufferProvider interface 5879status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5880 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5881{ 5882 RecordTrack *activeTrack = mRecordTrack; 5883 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5884 if (threadBase == 0) { 5885 buffer->frameCount = 0; 5886 buffer->raw = NULL; 5887 return NOT_ENOUGH_DATA; 5888 } 5889 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5890 int32_t rear = recordThread->mRsmpInRear; 5891 int32_t front = activeTrack->mRsmpInFront; 5892 ssize_t filled = rear - front; 5893 // FIXME should not be P2 (don't want to increase latency) 5894 // FIXME if client not keeping up, discard 5895 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5896 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5897 front &= recordThread->mRsmpInFramesP2 - 1; 5898 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5899 if (part1 > (size_t) filled) { 5900 part1 = filled; 5901 } 5902 size_t ask = buffer->frameCount; 5903 ALOG_ASSERT(ask > 0); 5904 if (part1 > ask) { 5905 part1 = ask; 5906 } 5907 if (part1 == 0) { 5908 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5909 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5910 buffer->raw = NULL; 5911 buffer->frameCount = 0; 5912 activeTrack->mRsmpInUnrel = 0; 5913 return NOT_ENOUGH_DATA; 5914 } 5915 5916 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5917 buffer->frameCount = part1; 5918 activeTrack->mRsmpInUnrel = part1; 5919 return NO_ERROR; 5920} 5921 5922// AudioBufferProvider interface 5923void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5924 AudioBufferProvider::Buffer* buffer) 5925{ 5926 RecordTrack *activeTrack = mRecordTrack; 5927 size_t stepCount = buffer->frameCount; 5928 if (stepCount == 0) { 5929 return; 5930 } 5931 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5932 activeTrack->mRsmpInUnrel -= stepCount; 5933 activeTrack->mRsmpInFront += stepCount; 5934 buffer->raw = NULL; 5935 buffer->frameCount = 0; 5936} 5937 5938bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5939 status_t& status) 5940{ 5941 bool reconfig = false; 5942 5943 status = NO_ERROR; 5944 5945 audio_format_t reqFormat = mFormat; 5946 uint32_t samplingRate = mSampleRate; 5947 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5948 5949 AudioParameter param = AudioParameter(keyValuePair); 5950 int value; 5951 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5952 // channel count change can be requested. Do we mandate the first client defines the 5953 // HAL sampling rate and channel count or do we allow changes on the fly? 5954 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5955 samplingRate = value; 5956 reconfig = true; 5957 } 5958 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5959 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5960 status = BAD_VALUE; 5961 } else { 5962 reqFormat = (audio_format_t) value; 5963 reconfig = true; 5964 } 5965 } 5966 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5967 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5968 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5969 status = BAD_VALUE; 5970 } else { 5971 channelMask = mask; 5972 reconfig = true; 5973 } 5974 } 5975 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5976 // do not accept frame count changes if tracks are open as the track buffer 5977 // size depends on frame count and correct behavior would not be guaranteed 5978 // if frame count is changed after track creation 5979 if (mActiveTracks.size() > 0) { 5980 status = INVALID_OPERATION; 5981 } else { 5982 reconfig = true; 5983 } 5984 } 5985 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5986 // forward device change to effects that have requested to be 5987 // aware of attached audio device. 5988 for (size_t i = 0; i < mEffectChains.size(); i++) { 5989 mEffectChains[i]->setDevice_l(value); 5990 } 5991 5992 // store input device and output device but do not forward output device to audio HAL. 5993 // Note that status is ignored by the caller for output device 5994 // (see AudioFlinger::setParameters() 5995 if (audio_is_output_devices(value)) { 5996 mOutDevice = value; 5997 status = BAD_VALUE; 5998 } else { 5999 mInDevice = value; 6000 // disable AEC and NS if the device is a BT SCO headset supporting those 6001 // pre processings 6002 if (mTracks.size() > 0) { 6003 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6004 mAudioFlinger->btNrecIsOff(); 6005 for (size_t i = 0; i < mTracks.size(); i++) { 6006 sp<RecordTrack> track = mTracks[i]; 6007 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6008 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6009 } 6010 } 6011 } 6012 } 6013 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6014 mAudioSource != (audio_source_t)value) { 6015 // forward device change to effects that have requested to be 6016 // aware of attached audio device. 6017 for (size_t i = 0; i < mEffectChains.size(); i++) { 6018 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6019 } 6020 mAudioSource = (audio_source_t)value; 6021 } 6022 6023 if (status == NO_ERROR) { 6024 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6025 keyValuePair.string()); 6026 if (status == INVALID_OPERATION) { 6027 inputStandBy(); 6028 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6029 keyValuePair.string()); 6030 } 6031 if (reconfig) { 6032 if (status == BAD_VALUE && 6033 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6034 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6035 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6036 <= (2 * samplingRate)) && 6037 audio_channel_count_from_in_mask( 6038 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6039 (channelMask == AUDIO_CHANNEL_IN_MONO || 6040 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6041 status = NO_ERROR; 6042 } 6043 if (status == NO_ERROR) { 6044 readInputParameters_l(); 6045 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6046 } 6047 } 6048 } 6049 6050 return reconfig; 6051} 6052 6053String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6054{ 6055 Mutex::Autolock _l(mLock); 6056 if (initCheck() != NO_ERROR) { 6057 return String8(); 6058 } 6059 6060 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6061 const String8 out_s8(s); 6062 free(s); 6063 return out_s8; 6064} 6065 6066void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6067 AudioSystem::OutputDescriptor desc; 6068 const void *param2 = NULL; 6069 6070 switch (event) { 6071 case AudioSystem::INPUT_OPENED: 6072 case AudioSystem::INPUT_CONFIG_CHANGED: 6073 desc.channelMask = mChannelMask; 6074 desc.samplingRate = mSampleRate; 6075 desc.format = mFormat; 6076 desc.frameCount = mFrameCount; 6077 desc.latency = 0; 6078 param2 = &desc; 6079 break; 6080 6081 case AudioSystem::INPUT_CLOSED: 6082 default: 6083 break; 6084 } 6085 mAudioFlinger->audioConfigChanged(event, mId, param2); 6086} 6087 6088void AudioFlinger::RecordThread::readInputParameters_l() 6089{ 6090 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6091 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6092 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6093 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6094 mFormat = mHALFormat; 6095 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6096 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6097 } 6098 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6099 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6100 mFrameCount = mBufferSize / mFrameSize; 6101 // This is the formula for calculating the temporary buffer size. 6102 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6103 // 1 full output buffer, regardless of the alignment of the available input. 6104 // The value is somewhat arbitrary, and could probably be even larger. 6105 // A larger value should allow more old data to be read after a track calls start(), 6106 // without increasing latency. 6107 mRsmpInFrames = mFrameCount * 7; 6108 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6109 delete[] mRsmpInBuffer; 6110 6111 // TODO optimize audio capture buffer sizes ... 6112 // Here we calculate the size of the sliding buffer used as a source 6113 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6114 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6115 // be better to have it derived from the pipe depth in the long term. 6116 // The current value is higher than necessary. However it should not add to latency. 6117 6118 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6119 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6120 6121 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6122 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6123} 6124 6125uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6126{ 6127 Mutex::Autolock _l(mLock); 6128 if (initCheck() != NO_ERROR) { 6129 return 0; 6130 } 6131 6132 return mInput->stream->get_input_frames_lost(mInput->stream); 6133} 6134 6135uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6136{ 6137 Mutex::Autolock _l(mLock); 6138 uint32_t result = 0; 6139 if (getEffectChain_l(sessionId) != 0) { 6140 result = EFFECT_SESSION; 6141 } 6142 6143 for (size_t i = 0; i < mTracks.size(); ++i) { 6144 if (sessionId == mTracks[i]->sessionId()) { 6145 result |= TRACK_SESSION; 6146 break; 6147 } 6148 } 6149 6150 return result; 6151} 6152 6153KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6154{ 6155 KeyedVector<int, bool> ids; 6156 Mutex::Autolock _l(mLock); 6157 for (size_t j = 0; j < mTracks.size(); ++j) { 6158 sp<RecordThread::RecordTrack> track = mTracks[j]; 6159 int sessionId = track->sessionId(); 6160 if (ids.indexOfKey(sessionId) < 0) { 6161 ids.add(sessionId, true); 6162 } 6163 } 6164 return ids; 6165} 6166 6167AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6168{ 6169 Mutex::Autolock _l(mLock); 6170 AudioStreamIn *input = mInput; 6171 mInput = NULL; 6172 return input; 6173} 6174 6175// this method must always be called either with ThreadBase mLock held or inside the thread loop 6176audio_stream_t* AudioFlinger::RecordThread::stream() const 6177{ 6178 if (mInput == NULL) { 6179 return NULL; 6180 } 6181 return &mInput->stream->common; 6182} 6183 6184status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6185{ 6186 // only one chain per input thread 6187 if (mEffectChains.size() != 0) { 6188 return INVALID_OPERATION; 6189 } 6190 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6191 6192 chain->setInBuffer(NULL); 6193 chain->setOutBuffer(NULL); 6194 6195 checkSuspendOnAddEffectChain_l(chain); 6196 6197 mEffectChains.add(chain); 6198 6199 return NO_ERROR; 6200} 6201 6202size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6203{ 6204 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6205 ALOGW_IF(mEffectChains.size() != 1, 6206 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6207 chain.get(), mEffectChains.size(), this); 6208 if (mEffectChains.size() == 1) { 6209 mEffectChains.removeAt(0); 6210 } 6211 return 0; 6212} 6213 6214status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6215 audio_patch_handle_t *handle) 6216{ 6217 status_t status = NO_ERROR; 6218 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6219 // store new device and send to effects 6220 mInDevice = patch->sources[0].ext.device.type; 6221 for (size_t i = 0; i < mEffectChains.size(); i++) { 6222 mEffectChains[i]->setDevice_l(mInDevice); 6223 } 6224 6225 // disable AEC and NS if the device is a BT SCO headset supporting those 6226 // pre processings 6227 if (mTracks.size() > 0) { 6228 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6229 mAudioFlinger->btNrecIsOff(); 6230 for (size_t i = 0; i < mTracks.size(); i++) { 6231 sp<RecordTrack> track = mTracks[i]; 6232 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6233 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6234 } 6235 } 6236 6237 // store new source and send to effects 6238 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6239 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6240 for (size_t i = 0; i < mEffectChains.size(); i++) { 6241 mEffectChains[i]->setAudioSource_l(mAudioSource); 6242 } 6243 } 6244 6245 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6246 status = hwDevice->create_audio_patch(hwDevice, 6247 patch->num_sources, 6248 patch->sources, 6249 patch->num_sinks, 6250 patch->sinks, 6251 handle); 6252 } else { 6253 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6254 } 6255 return status; 6256} 6257 6258status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6259{ 6260 status_t status = NO_ERROR; 6261 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6262 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6263 status = hwDevice->release_audio_patch(hwDevice, handle); 6264 } else { 6265 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6266 } 6267 return status; 6268} 6269 6270void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6271{ 6272 Mutex::Autolock _l(mLock); 6273 mTracks.add(record); 6274} 6275 6276void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6277{ 6278 Mutex::Autolock _l(mLock); 6279 destroyTrack_l(record); 6280} 6281 6282void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6283{ 6284 ThreadBase::getAudioPortConfig(config); 6285 config->role = AUDIO_PORT_ROLE_SINK; 6286 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6287 config->ext.mix.usecase.source = mAudioSource; 6288} 6289 6290}; // namespace android 6291