Threads.cpp revision 223fd5c9738e9665e495904d37d4632414b68c1e
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 true /* FIXME force oneway contrary to .aidl */); 667 } else { 668 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 669 binder, 670 getWakeLockTag(), 671 String16("media"), 672 true /* FIXME force oneway contrary to .aidl */); 673 } 674 if (status == NO_ERROR) { 675 mWakeLockToken = binder; 676 } 677 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 678 } 679} 680 681void AudioFlinger::ThreadBase::releaseWakeLock() 682{ 683 Mutex::Autolock _l(mLock); 684 releaseWakeLock_l(); 685} 686 687void AudioFlinger::ThreadBase::releaseWakeLock_l() 688{ 689 if (mWakeLockToken != 0) { 690 ALOGV("releaseWakeLock_l() %s", mName); 691 if (mPowerManager != 0) { 692 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 693 true /* FIXME force oneway contrary to .aidl */); 694 } 695 mWakeLockToken.clear(); 696 } 697} 698 699void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 700 Mutex::Autolock _l(mLock); 701 updateWakeLockUids_l(uids); 702} 703 704void AudioFlinger::ThreadBase::getPowerManager_l() { 705 706 if (mPowerManager == 0) { 707 // use checkService() to avoid blocking if power service is not up yet 708 sp<IBinder> binder = 709 defaultServiceManager()->checkService(String16("power")); 710 if (binder == 0) { 711 ALOGW("Thread %s cannot connect to the power manager service", mName); 712 } else { 713 mPowerManager = interface_cast<IPowerManager>(binder); 714 binder->linkToDeath(mDeathRecipient); 715 } 716 } 717} 718 719void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 720 721 getPowerManager_l(); 722 if (mWakeLockToken == NULL) { 723 ALOGE("no wake lock to update!"); 724 return; 725 } 726 if (mPowerManager != 0) { 727 sp<IBinder> binder = new BBinder(); 728 status_t status; 729 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 730 true /* FIXME force oneway contrary to .aidl */); 731 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 732 } 733} 734 735void AudioFlinger::ThreadBase::clearPowerManager() 736{ 737 Mutex::Autolock _l(mLock); 738 releaseWakeLock_l(); 739 mPowerManager.clear(); 740} 741 742void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 743{ 744 sp<ThreadBase> thread = mThread.promote(); 745 if (thread != 0) { 746 thread->clearPowerManager(); 747 } 748 ALOGW("power manager service died !!!"); 749} 750 751void AudioFlinger::ThreadBase::setEffectSuspended( 752 const effect_uuid_t *type, bool suspend, int sessionId) 753{ 754 Mutex::Autolock _l(mLock); 755 setEffectSuspended_l(type, suspend, sessionId); 756} 757 758void AudioFlinger::ThreadBase::setEffectSuspended_l( 759 const effect_uuid_t *type, bool suspend, int sessionId) 760{ 761 sp<EffectChain> chain = getEffectChain_l(sessionId); 762 if (chain != 0) { 763 if (type != NULL) { 764 chain->setEffectSuspended_l(type, suspend); 765 } else { 766 chain->setEffectSuspendedAll_l(suspend); 767 } 768 } 769 770 updateSuspendedSessions_l(type, suspend, sessionId); 771} 772 773void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 774{ 775 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 776 if (index < 0) { 777 return; 778 } 779 780 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 781 mSuspendedSessions.valueAt(index); 782 783 for (size_t i = 0; i < sessionEffects.size(); i++) { 784 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 785 for (int j = 0; j < desc->mRefCount; j++) { 786 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 787 chain->setEffectSuspendedAll_l(true); 788 } else { 789 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 790 desc->mType.timeLow); 791 chain->setEffectSuspended_l(&desc->mType, true); 792 } 793 } 794 } 795} 796 797void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 798 bool suspend, 799 int sessionId) 800{ 801 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 802 803 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 804 805 if (suspend) { 806 if (index >= 0) { 807 sessionEffects = mSuspendedSessions.valueAt(index); 808 } else { 809 mSuspendedSessions.add(sessionId, sessionEffects); 810 } 811 } else { 812 if (index < 0) { 813 return; 814 } 815 sessionEffects = mSuspendedSessions.valueAt(index); 816 } 817 818 819 int key = EffectChain::kKeyForSuspendAll; 820 if (type != NULL) { 821 key = type->timeLow; 822 } 823 index = sessionEffects.indexOfKey(key); 824 825 sp<SuspendedSessionDesc> desc; 826 if (suspend) { 827 if (index >= 0) { 828 desc = sessionEffects.valueAt(index); 829 } else { 830 desc = new SuspendedSessionDesc(); 831 if (type != NULL) { 832 desc->mType = *type; 833 } 834 sessionEffects.add(key, desc); 835 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 836 } 837 desc->mRefCount++; 838 } else { 839 if (index < 0) { 840 return; 841 } 842 desc = sessionEffects.valueAt(index); 843 if (--desc->mRefCount == 0) { 844 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 845 sessionEffects.removeItemsAt(index); 846 if (sessionEffects.isEmpty()) { 847 ALOGV("updateSuspendedSessions_l() restore removing session %d", 848 sessionId); 849 mSuspendedSessions.removeItem(sessionId); 850 } 851 } 852 } 853 if (!sessionEffects.isEmpty()) { 854 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 855 } 856} 857 858void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 859 bool enabled, 860 int sessionId) 861{ 862 Mutex::Autolock _l(mLock); 863 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 864} 865 866void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 867 bool enabled, 868 int sessionId) 869{ 870 if (mType != RECORD) { 871 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 872 // another session. This gives the priority to well behaved effect control panels 873 // and applications not using global effects. 874 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 875 // global effects 876 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 877 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 878 } 879 } 880 881 sp<EffectChain> chain = getEffectChain_l(sessionId); 882 if (chain != 0) { 883 chain->checkSuspendOnEffectEnabled(effect, enabled); 884 } 885} 886 887// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 888sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 889 const sp<AudioFlinger::Client>& client, 890 const sp<IEffectClient>& effectClient, 891 int32_t priority, 892 int sessionId, 893 effect_descriptor_t *desc, 894 int *enabled, 895 status_t *status) 896{ 897 sp<EffectModule> effect; 898 sp<EffectHandle> handle; 899 status_t lStatus; 900 sp<EffectChain> chain; 901 bool chainCreated = false; 902 bool effectCreated = false; 903 bool effectRegistered = false; 904 905 lStatus = initCheck(); 906 if (lStatus != NO_ERROR) { 907 ALOGW("createEffect_l() Audio driver not initialized."); 908 goto Exit; 909 } 910 911 // Reject any effect on Direct output threads for now, since the format of 912 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 913 if (mType == DIRECT) { 914 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 915 desc->name, mName); 916 lStatus = BAD_VALUE; 917 goto Exit; 918 } 919 920 // Reject any effect on mixer or duplicating multichannel sinks. 921 // TODO: fix both format and multichannel issues with effects. 922 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 923 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 924 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 925 lStatus = BAD_VALUE; 926 goto Exit; 927 } 928 929 // Allow global effects only on offloaded and mixer threads 930 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 931 switch (mType) { 932 case MIXER: 933 case OFFLOAD: 934 break; 935 case DIRECT: 936 case DUPLICATING: 937 case RECORD: 938 default: 939 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 940 lStatus = BAD_VALUE; 941 goto Exit; 942 } 943 } 944 945 // Only Pre processor effects are allowed on input threads and only on input threads 946 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 947 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 948 desc->name, desc->flags, mType); 949 lStatus = BAD_VALUE; 950 goto Exit; 951 } 952 953 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 954 955 { // scope for mLock 956 Mutex::Autolock _l(mLock); 957 958 // check for existing effect chain with the requested audio session 959 chain = getEffectChain_l(sessionId); 960 if (chain == 0) { 961 // create a new chain for this session 962 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 963 chain = new EffectChain(this, sessionId); 964 addEffectChain_l(chain); 965 chain->setStrategy(getStrategyForSession_l(sessionId)); 966 chainCreated = true; 967 } else { 968 effect = chain->getEffectFromDesc_l(desc); 969 } 970 971 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 972 973 if (effect == 0) { 974 int id = mAudioFlinger->nextUniqueId(); 975 // Check CPU and memory usage 976 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 977 if (lStatus != NO_ERROR) { 978 goto Exit; 979 } 980 effectRegistered = true; 981 // create a new effect module if none present in the chain 982 effect = new EffectModule(this, chain, desc, id, sessionId); 983 lStatus = effect->status(); 984 if (lStatus != NO_ERROR) { 985 goto Exit; 986 } 987 effect->setOffloaded(mType == OFFLOAD, mId); 988 989 lStatus = chain->addEffect_l(effect); 990 if (lStatus != NO_ERROR) { 991 goto Exit; 992 } 993 effectCreated = true; 994 995 effect->setDevice(mOutDevice); 996 effect->setDevice(mInDevice); 997 effect->setMode(mAudioFlinger->getMode()); 998 effect->setAudioSource(mAudioSource); 999 } 1000 // create effect handle and connect it to effect module 1001 handle = new EffectHandle(effect, client, effectClient, priority); 1002 lStatus = handle->initCheck(); 1003 if (lStatus == OK) { 1004 lStatus = effect->addHandle(handle.get()); 1005 } 1006 if (enabled != NULL) { 1007 *enabled = (int)effect->isEnabled(); 1008 } 1009 } 1010 1011Exit: 1012 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1013 Mutex::Autolock _l(mLock); 1014 if (effectCreated) { 1015 chain->removeEffect_l(effect); 1016 } 1017 if (effectRegistered) { 1018 AudioSystem::unregisterEffect(effect->id()); 1019 } 1020 if (chainCreated) { 1021 removeEffectChain_l(chain); 1022 } 1023 handle.clear(); 1024 } 1025 1026 *status = lStatus; 1027 return handle; 1028} 1029 1030sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1031{ 1032 Mutex::Autolock _l(mLock); 1033 return getEffect_l(sessionId, effectId); 1034} 1035 1036sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1037{ 1038 sp<EffectChain> chain = getEffectChain_l(sessionId); 1039 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1040} 1041 1042// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1043// PlaybackThread::mLock held 1044status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1045{ 1046 // check for existing effect chain with the requested audio session 1047 int sessionId = effect->sessionId(); 1048 sp<EffectChain> chain = getEffectChain_l(sessionId); 1049 bool chainCreated = false; 1050 1051 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1052 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1053 this, effect->desc().name, effect->desc().flags); 1054 1055 if (chain == 0) { 1056 // create a new chain for this session 1057 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1058 chain = new EffectChain(this, sessionId); 1059 addEffectChain_l(chain); 1060 chain->setStrategy(getStrategyForSession_l(sessionId)); 1061 chainCreated = true; 1062 } 1063 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1064 1065 if (chain->getEffectFromId_l(effect->id()) != 0) { 1066 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1067 this, effect->desc().name, chain.get()); 1068 return BAD_VALUE; 1069 } 1070 1071 effect->setOffloaded(mType == OFFLOAD, mId); 1072 1073 status_t status = chain->addEffect_l(effect); 1074 if (status != NO_ERROR) { 1075 if (chainCreated) { 1076 removeEffectChain_l(chain); 1077 } 1078 return status; 1079 } 1080 1081 effect->setDevice(mOutDevice); 1082 effect->setDevice(mInDevice); 1083 effect->setMode(mAudioFlinger->getMode()); 1084 effect->setAudioSource(mAudioSource); 1085 return NO_ERROR; 1086} 1087 1088void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1089 1090 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1091 effect_descriptor_t desc = effect->desc(); 1092 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1093 detachAuxEffect_l(effect->id()); 1094 } 1095 1096 sp<EffectChain> chain = effect->chain().promote(); 1097 if (chain != 0) { 1098 // remove effect chain if removing last effect 1099 if (chain->removeEffect_l(effect) == 0) { 1100 removeEffectChain_l(chain); 1101 } 1102 } else { 1103 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1104 } 1105} 1106 1107void AudioFlinger::ThreadBase::lockEffectChains_l( 1108 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1109{ 1110 effectChains = mEffectChains; 1111 for (size_t i = 0; i < mEffectChains.size(); i++) { 1112 mEffectChains[i]->lock(); 1113 } 1114} 1115 1116void AudioFlinger::ThreadBase::unlockEffectChains( 1117 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1118{ 1119 for (size_t i = 0; i < effectChains.size(); i++) { 1120 effectChains[i]->unlock(); 1121 } 1122} 1123 1124sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1125{ 1126 Mutex::Autolock _l(mLock); 1127 return getEffectChain_l(sessionId); 1128} 1129 1130sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1131{ 1132 size_t size = mEffectChains.size(); 1133 for (size_t i = 0; i < size; i++) { 1134 if (mEffectChains[i]->sessionId() == sessionId) { 1135 return mEffectChains[i]; 1136 } 1137 } 1138 return 0; 1139} 1140 1141void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1142{ 1143 Mutex::Autolock _l(mLock); 1144 size_t size = mEffectChains.size(); 1145 for (size_t i = 0; i < size; i++) { 1146 mEffectChains[i]->setMode_l(mode); 1147 } 1148} 1149 1150void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1151{ 1152 config->type = AUDIO_PORT_TYPE_MIX; 1153 config->ext.mix.handle = mId; 1154 config->sample_rate = mSampleRate; 1155 config->format = mFormat; 1156 config->channel_mask = mChannelMask; 1157 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1158 AUDIO_PORT_CONFIG_FORMAT; 1159} 1160 1161 1162// ---------------------------------------------------------------------------- 1163// Playback 1164// ---------------------------------------------------------------------------- 1165 1166AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1167 AudioStreamOut* output, 1168 audio_io_handle_t id, 1169 audio_devices_t device, 1170 type_t type) 1171 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1172 mNormalFrameCount(0), mSinkBuffer(NULL), 1173 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1174 mMixerBuffer(NULL), 1175 mMixerBufferSize(0), 1176 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1177 mMixerBufferValid(false), 1178 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1179 mEffectBuffer(NULL), 1180 mEffectBufferSize(0), 1181 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1182 mEffectBufferValid(false), 1183 mSuspended(0), mBytesWritten(0), 1184 mActiveTracksGeneration(0), 1185 // mStreamTypes[] initialized in constructor body 1186 mOutput(output), 1187 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1188 mMixerStatus(MIXER_IDLE), 1189 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1190 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1191 mBytesRemaining(0), 1192 mCurrentWriteLength(0), 1193 mUseAsyncWrite(false), 1194 mWriteAckSequence(0), 1195 mDrainSequence(0), 1196 mSignalPending(false), 1197 mScreenState(AudioFlinger::mScreenState), 1198 // index 0 is reserved for normal mixer's submix 1199 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1200 // mLatchD, mLatchQ, 1201 mLatchDValid(false), mLatchQValid(false) 1202{ 1203 snprintf(mName, kNameLength, "AudioOut_%X", id); 1204 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1205 1206 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1207 // it would be safer to explicitly pass initial masterVolume/masterMute as 1208 // parameter. 1209 // 1210 // If the HAL we are using has support for master volume or master mute, 1211 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1212 // and the mute set to false). 1213 mMasterVolume = audioFlinger->masterVolume_l(); 1214 mMasterMute = audioFlinger->masterMute_l(); 1215 if (mOutput && mOutput->audioHwDev) { 1216 if (mOutput->audioHwDev->canSetMasterVolume()) { 1217 mMasterVolume = 1.0; 1218 } 1219 1220 if (mOutput->audioHwDev->canSetMasterMute()) { 1221 mMasterMute = false; 1222 } 1223 } 1224 1225 readOutputParameters_l(); 1226 1227 // ++ operator does not compile 1228 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1229 stream = (audio_stream_type_t) (stream + 1)) { 1230 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1231 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1232 } 1233} 1234 1235AudioFlinger::PlaybackThread::~PlaybackThread() 1236{ 1237 mAudioFlinger->unregisterWriter(mNBLogWriter); 1238 free(mSinkBuffer); 1239 free(mMixerBuffer); 1240 free(mEffectBuffer); 1241} 1242 1243void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1244{ 1245 dumpInternals(fd, args); 1246 dumpTracks(fd, args); 1247 dumpEffectChains(fd, args); 1248} 1249 1250void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1251{ 1252 const size_t SIZE = 256; 1253 char buffer[SIZE]; 1254 String8 result; 1255 1256 result.appendFormat(" Stream volumes in dB: "); 1257 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1258 const stream_type_t *st = &mStreamTypes[i]; 1259 if (i > 0) { 1260 result.appendFormat(", "); 1261 } 1262 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1263 if (st->mute) { 1264 result.append("M"); 1265 } 1266 } 1267 result.append("\n"); 1268 write(fd, result.string(), result.length()); 1269 result.clear(); 1270 1271 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1272 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1273 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1274 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1275 1276 size_t numtracks = mTracks.size(); 1277 size_t numactive = mActiveTracks.size(); 1278 dprintf(fd, " %d Tracks", numtracks); 1279 size_t numactiveseen = 0; 1280 if (numtracks) { 1281 dprintf(fd, " of which %d are active\n", numactive); 1282 Track::appendDumpHeader(result); 1283 for (size_t i = 0; i < numtracks; ++i) { 1284 sp<Track> track = mTracks[i]; 1285 if (track != 0) { 1286 bool active = mActiveTracks.indexOf(track) >= 0; 1287 if (active) { 1288 numactiveseen++; 1289 } 1290 track->dump(buffer, SIZE, active); 1291 result.append(buffer); 1292 } 1293 } 1294 } else { 1295 result.append("\n"); 1296 } 1297 if (numactiveseen != numactive) { 1298 // some tracks in the active list were not in the tracks list 1299 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1300 " not in the track list\n"); 1301 result.append(buffer); 1302 Track::appendDumpHeader(result); 1303 for (size_t i = 0; i < numactive; ++i) { 1304 sp<Track> track = mActiveTracks[i].promote(); 1305 if (track != 0 && mTracks.indexOf(track) < 0) { 1306 track->dump(buffer, SIZE, true); 1307 result.append(buffer); 1308 } 1309 } 1310 } 1311 1312 write(fd, result.string(), result.size()); 1313} 1314 1315void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1316{ 1317 dprintf(fd, "\nOutput thread %p:\n", this); 1318 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1319 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1320 dprintf(fd, " Total writes: %d\n", mNumWrites); 1321 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1322 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1323 dprintf(fd, " Suspend count: %d\n", mSuspended); 1324 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1325 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1326 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1327 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1328 1329 dumpBase(fd, args); 1330} 1331 1332// Thread virtuals 1333 1334void AudioFlinger::PlaybackThread::onFirstRef() 1335{ 1336 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1337} 1338 1339// ThreadBase virtuals 1340void AudioFlinger::PlaybackThread::preExit() 1341{ 1342 ALOGV(" preExit()"); 1343 // FIXME this is using hard-coded strings but in the future, this functionality will be 1344 // converted to use audio HAL extensions required to support tunneling 1345 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1346} 1347 1348// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1349sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1350 const sp<AudioFlinger::Client>& client, 1351 audio_stream_type_t streamType, 1352 uint32_t sampleRate, 1353 audio_format_t format, 1354 audio_channel_mask_t channelMask, 1355 size_t *pFrameCount, 1356 const sp<IMemory>& sharedBuffer, 1357 int sessionId, 1358 IAudioFlinger::track_flags_t *flags, 1359 pid_t tid, 1360 int uid, 1361 status_t *status) 1362{ 1363 size_t frameCount = *pFrameCount; 1364 sp<Track> track; 1365 status_t lStatus; 1366 1367 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1368 1369 // client expresses a preference for FAST, but we get the final say 1370 if (*flags & IAudioFlinger::TRACK_FAST) { 1371 if ( 1372 // not timed 1373 (!isTimed) && 1374 // either of these use cases: 1375 ( 1376 // use case 1: shared buffer with any frame count 1377 ( 1378 (sharedBuffer != 0) 1379 ) || 1380 // use case 2: callback handler and frame count is default or at least as large as HAL 1381 ( 1382 (tid != -1) && 1383 ((frameCount == 0) || 1384 (frameCount >= mFrameCount)) 1385 ) 1386 ) && 1387 // PCM data 1388 audio_is_linear_pcm(format) && 1389 // identical channel mask to sink, or mono in and stereo sink 1390 (channelMask == mChannelMask || 1391 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1392 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1393 // hardware sample rate 1394 (sampleRate == mSampleRate) && 1395 // normal mixer has an associated fast mixer 1396 hasFastMixer() && 1397 // there are sufficient fast track slots available 1398 (mFastTrackAvailMask != 0) 1399 // FIXME test that MixerThread for this fast track has a capable output HAL 1400 // FIXME add a permission test also? 1401 ) { 1402 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1403 if (frameCount == 0) { 1404 // read the fast track multiplier property the first time it is needed 1405 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1406 if (ok != 0) { 1407 ALOGE("%s pthread_once failed: %d", __func__, ok); 1408 } 1409 frameCount = mFrameCount * sFastTrackMultiplier; 1410 } 1411 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1412 frameCount, mFrameCount); 1413 } else { 1414 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1415 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1416 "sampleRate=%u mSampleRate=%u " 1417 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1418 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1419 audio_is_linear_pcm(format), 1420 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1421 *flags &= ~IAudioFlinger::TRACK_FAST; 1422 // For compatibility with AudioTrack calculation, buffer depth is forced 1423 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1424 // This is probably too conservative, but legacy application code may depend on it. 1425 // If you change this calculation, also review the start threshold which is related. 1426 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1427 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1428 if (minBufCount < 2) { 1429 minBufCount = 2; 1430 } 1431 size_t minFrameCount = mNormalFrameCount * minBufCount; 1432 if (frameCount < minFrameCount) { 1433 frameCount = minFrameCount; 1434 } 1435 } 1436 } 1437 *pFrameCount = frameCount; 1438 1439 switch (mType) { 1440 1441 case DIRECT: 1442 if (audio_is_linear_pcm(format)) { 1443 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1444 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1445 "for output %p with format %#x", 1446 sampleRate, format, channelMask, mOutput, mFormat); 1447 lStatus = BAD_VALUE; 1448 goto Exit; 1449 } 1450 } 1451 break; 1452 1453 case OFFLOAD: 1454 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1455 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1456 "for output %p with format %#x", 1457 sampleRate, format, channelMask, mOutput, mFormat); 1458 lStatus = BAD_VALUE; 1459 goto Exit; 1460 } 1461 break; 1462 1463 default: 1464 if (!audio_is_linear_pcm(format)) { 1465 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1466 "for output %p with format %#x", 1467 format, mOutput, mFormat); 1468 lStatus = BAD_VALUE; 1469 goto Exit; 1470 } 1471 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1472 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1473 lStatus = BAD_VALUE; 1474 goto Exit; 1475 } 1476 break; 1477 1478 } 1479 1480 lStatus = initCheck(); 1481 if (lStatus != NO_ERROR) { 1482 ALOGE("createTrack_l() audio driver not initialized"); 1483 goto Exit; 1484 } 1485 1486 { // scope for mLock 1487 Mutex::Autolock _l(mLock); 1488 1489 // all tracks in same audio session must share the same routing strategy otherwise 1490 // conflicts will happen when tracks are moved from one output to another by audio policy 1491 // manager 1492 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1493 for (size_t i = 0; i < mTracks.size(); ++i) { 1494 sp<Track> t = mTracks[i]; 1495 if (t != 0 && t->isExternalTrack()) { 1496 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1497 if (sessionId == t->sessionId() && strategy != actual) { 1498 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1499 strategy, actual); 1500 lStatus = BAD_VALUE; 1501 goto Exit; 1502 } 1503 } 1504 } 1505 1506 if (!isTimed) { 1507 track = new Track(this, client, streamType, sampleRate, format, 1508 channelMask, frameCount, NULL, sharedBuffer, 1509 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1510 } else { 1511 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1512 channelMask, frameCount, sharedBuffer, sessionId, uid); 1513 } 1514 1515 // new Track always returns non-NULL, 1516 // but TimedTrack::create() is a factory that could fail by returning NULL 1517 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1518 if (lStatus != NO_ERROR) { 1519 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1520 // track must be cleared from the caller as the caller has the AF lock 1521 goto Exit; 1522 } 1523 mTracks.add(track); 1524 1525 sp<EffectChain> chain = getEffectChain_l(sessionId); 1526 if (chain != 0) { 1527 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1528 track->setMainBuffer(chain->inBuffer()); 1529 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1530 chain->incTrackCnt(); 1531 } 1532 1533 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1534 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1535 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1536 // so ask activity manager to do this on our behalf 1537 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1538 } 1539 } 1540 1541 lStatus = NO_ERROR; 1542 1543Exit: 1544 *status = lStatus; 1545 return track; 1546} 1547 1548uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1549{ 1550 return latency; 1551} 1552 1553uint32_t AudioFlinger::PlaybackThread::latency() const 1554{ 1555 Mutex::Autolock _l(mLock); 1556 return latency_l(); 1557} 1558uint32_t AudioFlinger::PlaybackThread::latency_l() const 1559{ 1560 if (initCheck() == NO_ERROR) { 1561 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1562 } else { 1563 return 0; 1564 } 1565} 1566 1567void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1568{ 1569 Mutex::Autolock _l(mLock); 1570 // Don't apply master volume in SW if our HAL can do it for us. 1571 if (mOutput && mOutput->audioHwDev && 1572 mOutput->audioHwDev->canSetMasterVolume()) { 1573 mMasterVolume = 1.0; 1574 } else { 1575 mMasterVolume = value; 1576 } 1577} 1578 1579void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1580{ 1581 Mutex::Autolock _l(mLock); 1582 // Don't apply master mute in SW if our HAL can do it for us. 1583 if (mOutput && mOutput->audioHwDev && 1584 mOutput->audioHwDev->canSetMasterMute()) { 1585 mMasterMute = false; 1586 } else { 1587 mMasterMute = muted; 1588 } 1589} 1590 1591void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1592{ 1593 Mutex::Autolock _l(mLock); 1594 mStreamTypes[stream].volume = value; 1595 broadcast_l(); 1596} 1597 1598void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1599{ 1600 Mutex::Autolock _l(mLock); 1601 mStreamTypes[stream].mute = muted; 1602 broadcast_l(); 1603} 1604 1605float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1606{ 1607 Mutex::Autolock _l(mLock); 1608 return mStreamTypes[stream].volume; 1609} 1610 1611// addTrack_l() must be called with ThreadBase::mLock held 1612status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1613{ 1614 status_t status = ALREADY_EXISTS; 1615 1616 // set retry count for buffer fill 1617 track->mRetryCount = kMaxTrackStartupRetries; 1618 if (mActiveTracks.indexOf(track) < 0) { 1619 // the track is newly added, make sure it fills up all its 1620 // buffers before playing. This is to ensure the client will 1621 // effectively get the latency it requested. 1622 if (track->isExternalTrack()) { 1623 TrackBase::track_state state = track->mState; 1624 mLock.unlock(); 1625 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1626 mLock.lock(); 1627 // abort track was stopped/paused while we released the lock 1628 if (state != track->mState) { 1629 if (status == NO_ERROR) { 1630 mLock.unlock(); 1631 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1632 mLock.lock(); 1633 } 1634 return INVALID_OPERATION; 1635 } 1636 // abort if start is rejected by audio policy manager 1637 if (status != NO_ERROR) { 1638 return PERMISSION_DENIED; 1639 } 1640#ifdef ADD_BATTERY_DATA 1641 // to track the speaker usage 1642 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1643#endif 1644 } 1645 1646 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1647 track->mResetDone = false; 1648 track->mPresentationCompleteFrames = 0; 1649 mActiveTracks.add(track); 1650 mWakeLockUids.add(track->uid()); 1651 mActiveTracksGeneration++; 1652 mLatestActiveTrack = track; 1653 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1654 if (chain != 0) { 1655 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1656 track->sessionId()); 1657 chain->incActiveTrackCnt(); 1658 } 1659 1660 status = NO_ERROR; 1661 } 1662 1663 onAddNewTrack_l(); 1664 return status; 1665} 1666 1667bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1668{ 1669 track->terminate(); 1670 // active tracks are removed by threadLoop() 1671 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1672 track->mState = TrackBase::STOPPED; 1673 if (!trackActive) { 1674 removeTrack_l(track); 1675 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1676 track->mState = TrackBase::STOPPING_1; 1677 } 1678 1679 return trackActive; 1680} 1681 1682void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1683{ 1684 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1685 mTracks.remove(track); 1686 deleteTrackName_l(track->name()); 1687 // redundant as track is about to be destroyed, for dumpsys only 1688 track->mName = -1; 1689 if (track->isFastTrack()) { 1690 int index = track->mFastIndex; 1691 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1692 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1693 mFastTrackAvailMask |= 1 << index; 1694 // redundant as track is about to be destroyed, for dumpsys only 1695 track->mFastIndex = -1; 1696 } 1697 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1698 if (chain != 0) { 1699 chain->decTrackCnt(); 1700 } 1701} 1702 1703void AudioFlinger::PlaybackThread::broadcast_l() 1704{ 1705 // Thread could be blocked waiting for async 1706 // so signal it to handle state changes immediately 1707 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1708 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1709 mSignalPending = true; 1710 mWaitWorkCV.broadcast(); 1711} 1712 1713String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1714{ 1715 Mutex::Autolock _l(mLock); 1716 if (initCheck() != NO_ERROR) { 1717 return String8(); 1718 } 1719 1720 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1721 const String8 out_s8(s); 1722 free(s); 1723 return out_s8; 1724} 1725 1726void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1727 AudioSystem::OutputDescriptor desc; 1728 void *param2 = NULL; 1729 1730 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1731 param); 1732 1733 switch (event) { 1734 case AudioSystem::OUTPUT_OPENED: 1735 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1736 desc.channelMask = mChannelMask; 1737 desc.samplingRate = mSampleRate; 1738 desc.format = mFormat; 1739 desc.frameCount = mNormalFrameCount; // FIXME see 1740 // AudioFlinger::frameCount(audio_io_handle_t) 1741 desc.latency = latency_l(); 1742 param2 = &desc; 1743 break; 1744 1745 case AudioSystem::STREAM_CONFIG_CHANGED: 1746 param2 = ¶m; 1747 case AudioSystem::OUTPUT_CLOSED: 1748 default: 1749 break; 1750 } 1751 mAudioFlinger->audioConfigChanged(event, mId, param2); 1752} 1753 1754void AudioFlinger::PlaybackThread::writeCallback() 1755{ 1756 ALOG_ASSERT(mCallbackThread != 0); 1757 mCallbackThread->resetWriteBlocked(); 1758} 1759 1760void AudioFlinger::PlaybackThread::drainCallback() 1761{ 1762 ALOG_ASSERT(mCallbackThread != 0); 1763 mCallbackThread->resetDraining(); 1764} 1765 1766void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1767{ 1768 Mutex::Autolock _l(mLock); 1769 // reject out of sequence requests 1770 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1771 mWriteAckSequence &= ~1; 1772 mWaitWorkCV.signal(); 1773 } 1774} 1775 1776void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1777{ 1778 Mutex::Autolock _l(mLock); 1779 // reject out of sequence requests 1780 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1781 mDrainSequence &= ~1; 1782 mWaitWorkCV.signal(); 1783 } 1784} 1785 1786// static 1787int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1788 void *param __unused, 1789 void *cookie) 1790{ 1791 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1792 ALOGV("asyncCallback() event %d", event); 1793 switch (event) { 1794 case STREAM_CBK_EVENT_WRITE_READY: 1795 me->writeCallback(); 1796 break; 1797 case STREAM_CBK_EVENT_DRAIN_READY: 1798 me->drainCallback(); 1799 break; 1800 default: 1801 ALOGW("asyncCallback() unknown event %d", event); 1802 break; 1803 } 1804 return 0; 1805} 1806 1807void AudioFlinger::PlaybackThread::readOutputParameters_l() 1808{ 1809 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1810 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1811 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1812 if (!audio_is_output_channel(mChannelMask)) { 1813 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1814 } 1815 if ((mType == MIXER || mType == DUPLICATING) 1816 && !isValidPcmSinkChannelMask(mChannelMask)) { 1817 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1818 mChannelMask); 1819 } 1820 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1821 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1822 mFormat = mHALFormat; 1823 if (!audio_is_valid_format(mFormat)) { 1824 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1825 } 1826 if ((mType == MIXER || mType == DUPLICATING) 1827 && !isValidPcmSinkFormat(mFormat)) { 1828 LOG_FATAL("HAL format %#x not supported for mixed output", 1829 mFormat); 1830 } 1831 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1832 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1833 mFrameCount = mBufferSize / mFrameSize; 1834 if (mFrameCount & 15) { 1835 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1836 mFrameCount); 1837 } 1838 1839 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1840 (mOutput->stream->set_callback != NULL)) { 1841 if (mOutput->stream->set_callback(mOutput->stream, 1842 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1843 mUseAsyncWrite = true; 1844 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1845 } 1846 } 1847 1848 // Calculate size of normal sink buffer relative to the HAL output buffer size 1849 double multiplier = 1.0; 1850 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1851 kUseFastMixer == FastMixer_Dynamic)) { 1852 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1853 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1854 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1855 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1856 maxNormalFrameCount = maxNormalFrameCount & ~15; 1857 if (maxNormalFrameCount < minNormalFrameCount) { 1858 maxNormalFrameCount = minNormalFrameCount; 1859 } 1860 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1861 if (multiplier <= 1.0) { 1862 multiplier = 1.0; 1863 } else if (multiplier <= 2.0) { 1864 if (2 * mFrameCount <= maxNormalFrameCount) { 1865 multiplier = 2.0; 1866 } else { 1867 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1868 } 1869 } else { 1870 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1871 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1872 // track, but we sometimes have to do this to satisfy the maximum frame count 1873 // constraint) 1874 // FIXME this rounding up should not be done if no HAL SRC 1875 uint32_t truncMult = (uint32_t) multiplier; 1876 if ((truncMult & 1)) { 1877 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1878 ++truncMult; 1879 } 1880 } 1881 multiplier = (double) truncMult; 1882 } 1883 } 1884 mNormalFrameCount = multiplier * mFrameCount; 1885 // round up to nearest 16 frames to satisfy AudioMixer 1886 if (mType == MIXER || mType == DUPLICATING) { 1887 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1888 } 1889 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1890 mNormalFrameCount); 1891 1892 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1893 // Originally this was int16_t[] array, need to remove legacy implications. 1894 free(mSinkBuffer); 1895 mSinkBuffer = NULL; 1896 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1897 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1898 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1899 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1900 1901 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1902 // drives the output. 1903 free(mMixerBuffer); 1904 mMixerBuffer = NULL; 1905 if (mMixerBufferEnabled) { 1906 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1907 mMixerBufferSize = mNormalFrameCount * mChannelCount 1908 * audio_bytes_per_sample(mMixerBufferFormat); 1909 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1910 } 1911 free(mEffectBuffer); 1912 mEffectBuffer = NULL; 1913 if (mEffectBufferEnabled) { 1914 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1915 mEffectBufferSize = mNormalFrameCount * mChannelCount 1916 * audio_bytes_per_sample(mEffectBufferFormat); 1917 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1918 } 1919 1920 // force reconfiguration of effect chains and engines to take new buffer size and audio 1921 // parameters into account 1922 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1923 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1924 // matter. 1925 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1926 Vector< sp<EffectChain> > effectChains = mEffectChains; 1927 for (size_t i = 0; i < effectChains.size(); i ++) { 1928 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1929 } 1930} 1931 1932 1933status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1934{ 1935 if (halFrames == NULL || dspFrames == NULL) { 1936 return BAD_VALUE; 1937 } 1938 Mutex::Autolock _l(mLock); 1939 if (initCheck() != NO_ERROR) { 1940 return INVALID_OPERATION; 1941 } 1942 size_t framesWritten = mBytesWritten / mFrameSize; 1943 *halFrames = framesWritten; 1944 1945 if (isSuspended()) { 1946 // return an estimation of rendered frames when the output is suspended 1947 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1948 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1949 return NO_ERROR; 1950 } else { 1951 status_t status; 1952 uint32_t frames; 1953 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1954 *dspFrames = (size_t)frames; 1955 return status; 1956 } 1957} 1958 1959uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1960{ 1961 Mutex::Autolock _l(mLock); 1962 uint32_t result = 0; 1963 if (getEffectChain_l(sessionId) != 0) { 1964 result = EFFECT_SESSION; 1965 } 1966 1967 for (size_t i = 0; i < mTracks.size(); ++i) { 1968 sp<Track> track = mTracks[i]; 1969 if (sessionId == track->sessionId() && !track->isInvalid()) { 1970 result |= TRACK_SESSION; 1971 break; 1972 } 1973 } 1974 1975 return result; 1976} 1977 1978uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1979{ 1980 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1981 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1982 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1983 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1984 } 1985 for (size_t i = 0; i < mTracks.size(); i++) { 1986 sp<Track> track = mTracks[i]; 1987 if (sessionId == track->sessionId() && !track->isInvalid()) { 1988 return AudioSystem::getStrategyForStream(track->streamType()); 1989 } 1990 } 1991 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1992} 1993 1994 1995AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1996{ 1997 Mutex::Autolock _l(mLock); 1998 return mOutput; 1999} 2000 2001AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2002{ 2003 Mutex::Autolock _l(mLock); 2004 AudioStreamOut *output = mOutput; 2005 mOutput = NULL; 2006 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2007 // must push a NULL and wait for ack 2008 mOutputSink.clear(); 2009 mPipeSink.clear(); 2010 mNormalSink.clear(); 2011 return output; 2012} 2013 2014// this method must always be called either with ThreadBase mLock held or inside the thread loop 2015audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2016{ 2017 if (mOutput == NULL) { 2018 return NULL; 2019 } 2020 return &mOutput->stream->common; 2021} 2022 2023uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2024{ 2025 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2026} 2027 2028status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2029{ 2030 if (!isValidSyncEvent(event)) { 2031 return BAD_VALUE; 2032 } 2033 2034 Mutex::Autolock _l(mLock); 2035 2036 for (size_t i = 0; i < mTracks.size(); ++i) { 2037 sp<Track> track = mTracks[i]; 2038 if (event->triggerSession() == track->sessionId()) { 2039 (void) track->setSyncEvent(event); 2040 return NO_ERROR; 2041 } 2042 } 2043 2044 return NAME_NOT_FOUND; 2045} 2046 2047bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2048{ 2049 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2050} 2051 2052void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2053 const Vector< sp<Track> >& tracksToRemove) 2054{ 2055 size_t count = tracksToRemove.size(); 2056 if (count > 0) { 2057 for (size_t i = 0 ; i < count ; i++) { 2058 const sp<Track>& track = tracksToRemove.itemAt(i); 2059 if (track->isExternalTrack()) { 2060 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2061#ifdef ADD_BATTERY_DATA 2062 // to track the speaker usage 2063 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2064#endif 2065 if (track->isTerminated()) { 2066 AudioSystem::releaseOutput(mId); 2067 } 2068 } 2069 } 2070 } 2071} 2072 2073void AudioFlinger::PlaybackThread::checkSilentMode_l() 2074{ 2075 if (!mMasterMute) { 2076 char value[PROPERTY_VALUE_MAX]; 2077 if (property_get("ro.audio.silent", value, "0") > 0) { 2078 char *endptr; 2079 unsigned long ul = strtoul(value, &endptr, 0); 2080 if (*endptr == '\0' && ul != 0) { 2081 ALOGD("Silence is golden"); 2082 // The setprop command will not allow a property to be changed after 2083 // the first time it is set, so we don't have to worry about un-muting. 2084 setMasterMute_l(true); 2085 } 2086 } 2087 } 2088} 2089 2090// shared by MIXER and DIRECT, overridden by DUPLICATING 2091ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2092{ 2093 // FIXME rewrite to reduce number of system calls 2094 mLastWriteTime = systemTime(); 2095 mInWrite = true; 2096 ssize_t bytesWritten; 2097 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2098 2099 // If an NBAIO sink is present, use it to write the normal mixer's submix 2100 if (mNormalSink != 0) { 2101 2102 const size_t count = mBytesRemaining / mFrameSize; 2103 2104 ATRACE_BEGIN("write"); 2105 // update the setpoint when AudioFlinger::mScreenState changes 2106 uint32_t screenState = AudioFlinger::mScreenState; 2107 if (screenState != mScreenState) { 2108 mScreenState = screenState; 2109 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2110 if (pipe != NULL) { 2111 pipe->setAvgFrames((mScreenState & 1) ? 2112 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2113 } 2114 } 2115 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2116 ATRACE_END(); 2117 if (framesWritten > 0) { 2118 bytesWritten = framesWritten * mFrameSize; 2119 } else { 2120 bytesWritten = framesWritten; 2121 } 2122 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2123 if (status == NO_ERROR) { 2124 size_t totalFramesWritten = mNormalSink->framesWritten(); 2125 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2126 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2127 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2128 mLatchDValid = true; 2129 } 2130 } 2131 // otherwise use the HAL / AudioStreamOut directly 2132 } else { 2133 // Direct output and offload threads 2134 2135 if (mUseAsyncWrite) { 2136 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2137 mWriteAckSequence += 2; 2138 mWriteAckSequence |= 1; 2139 ALOG_ASSERT(mCallbackThread != 0); 2140 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2141 } 2142 // FIXME We should have an implementation of timestamps for direct output threads. 2143 // They are used e.g for multichannel PCM playback over HDMI. 2144 bytesWritten = mOutput->stream->write(mOutput->stream, 2145 (char *)mSinkBuffer + offset, mBytesRemaining); 2146 if (mUseAsyncWrite && 2147 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2148 // do not wait for async callback in case of error of full write 2149 mWriteAckSequence &= ~1; 2150 ALOG_ASSERT(mCallbackThread != 0); 2151 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2152 } 2153 } 2154 2155 mNumWrites++; 2156 mInWrite = false; 2157 mStandby = false; 2158 return bytesWritten; 2159} 2160 2161void AudioFlinger::PlaybackThread::threadLoop_drain() 2162{ 2163 if (mOutput->stream->drain) { 2164 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2165 if (mUseAsyncWrite) { 2166 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2167 mDrainSequence |= 1; 2168 ALOG_ASSERT(mCallbackThread != 0); 2169 mCallbackThread->setDraining(mDrainSequence); 2170 } 2171 mOutput->stream->drain(mOutput->stream, 2172 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2173 : AUDIO_DRAIN_ALL); 2174 } 2175} 2176 2177void AudioFlinger::PlaybackThread::threadLoop_exit() 2178{ 2179 // Default implementation has nothing to do 2180} 2181 2182/* 2183The derived values that are cached: 2184 - mSinkBufferSize from frame count * frame size 2185 - activeSleepTime from activeSleepTimeUs() 2186 - idleSleepTime from idleSleepTimeUs() 2187 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2188 - maxPeriod from frame count and sample rate (MIXER only) 2189 2190The parameters that affect these derived values are: 2191 - frame count 2192 - frame size 2193 - sample rate 2194 - device type: A2DP or not 2195 - device latency 2196 - format: PCM or not 2197 - active sleep time 2198 - idle sleep time 2199*/ 2200 2201void AudioFlinger::PlaybackThread::cacheParameters_l() 2202{ 2203 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2204 activeSleepTime = activeSleepTimeUs(); 2205 idleSleepTime = idleSleepTimeUs(); 2206} 2207 2208void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2209{ 2210 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2211 this, streamType, mTracks.size()); 2212 Mutex::Autolock _l(mLock); 2213 2214 size_t size = mTracks.size(); 2215 for (size_t i = 0; i < size; i++) { 2216 sp<Track> t = mTracks[i]; 2217 if (t->streamType() == streamType) { 2218 t->invalidate(); 2219 } 2220 } 2221} 2222 2223status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2224{ 2225 int session = chain->sessionId(); 2226 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2227 ? mEffectBuffer : mSinkBuffer); 2228 bool ownsBuffer = false; 2229 2230 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2231 if (session > 0) { 2232 // Only one effect chain can be present in direct output thread and it uses 2233 // the sink buffer as input 2234 if (mType != DIRECT) { 2235 size_t numSamples = mNormalFrameCount * mChannelCount; 2236 buffer = new int16_t[numSamples]; 2237 memset(buffer, 0, numSamples * sizeof(int16_t)); 2238 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2239 ownsBuffer = true; 2240 } 2241 2242 // Attach all tracks with same session ID to this chain. 2243 for (size_t i = 0; i < mTracks.size(); ++i) { 2244 sp<Track> track = mTracks[i]; 2245 if (session == track->sessionId()) { 2246 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2247 buffer); 2248 track->setMainBuffer(buffer); 2249 chain->incTrackCnt(); 2250 } 2251 } 2252 2253 // indicate all active tracks in the chain 2254 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2255 sp<Track> track = mActiveTracks[i].promote(); 2256 if (track == 0) { 2257 continue; 2258 } 2259 if (session == track->sessionId()) { 2260 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2261 chain->incActiveTrackCnt(); 2262 } 2263 } 2264 } 2265 chain->setThread(this); 2266 chain->setInBuffer(buffer, ownsBuffer); 2267 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2268 ? mEffectBuffer : mSinkBuffer)); 2269 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2270 // chains list in order to be processed last as it contains output stage effects 2271 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2272 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2273 // after track specific effects and before output stage 2274 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2275 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2276 // Effect chain for other sessions are inserted at beginning of effect 2277 // chains list to be processed before output mix effects. Relative order between other 2278 // sessions is not important 2279 size_t size = mEffectChains.size(); 2280 size_t i = 0; 2281 for (i = 0; i < size; i++) { 2282 if (mEffectChains[i]->sessionId() < session) { 2283 break; 2284 } 2285 } 2286 mEffectChains.insertAt(chain, i); 2287 checkSuspendOnAddEffectChain_l(chain); 2288 2289 return NO_ERROR; 2290} 2291 2292size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2293{ 2294 int session = chain->sessionId(); 2295 2296 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2297 2298 for (size_t i = 0; i < mEffectChains.size(); i++) { 2299 if (chain == mEffectChains[i]) { 2300 mEffectChains.removeAt(i); 2301 // detach all active tracks from the chain 2302 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2303 sp<Track> track = mActiveTracks[i].promote(); 2304 if (track == 0) { 2305 continue; 2306 } 2307 if (session == track->sessionId()) { 2308 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2309 chain.get(), session); 2310 chain->decActiveTrackCnt(); 2311 } 2312 } 2313 2314 // detach all tracks with same session ID from this chain 2315 for (size_t i = 0; i < mTracks.size(); ++i) { 2316 sp<Track> track = mTracks[i]; 2317 if (session == track->sessionId()) { 2318 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2319 chain->decTrackCnt(); 2320 } 2321 } 2322 break; 2323 } 2324 } 2325 return mEffectChains.size(); 2326} 2327 2328status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2329 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2330{ 2331 Mutex::Autolock _l(mLock); 2332 return attachAuxEffect_l(track, EffectId); 2333} 2334 2335status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2336 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2337{ 2338 status_t status = NO_ERROR; 2339 2340 if (EffectId == 0) { 2341 track->setAuxBuffer(0, NULL); 2342 } else { 2343 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2344 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2345 if (effect != 0) { 2346 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2347 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2348 } else { 2349 status = INVALID_OPERATION; 2350 } 2351 } else { 2352 status = BAD_VALUE; 2353 } 2354 } 2355 return status; 2356} 2357 2358void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2359{ 2360 for (size_t i = 0; i < mTracks.size(); ++i) { 2361 sp<Track> track = mTracks[i]; 2362 if (track->auxEffectId() == effectId) { 2363 attachAuxEffect_l(track, 0); 2364 } 2365 } 2366} 2367 2368bool AudioFlinger::PlaybackThread::threadLoop() 2369{ 2370 Vector< sp<Track> > tracksToRemove; 2371 2372 standbyTime = systemTime(); 2373 2374 // MIXER 2375 nsecs_t lastWarning = 0; 2376 2377 // DUPLICATING 2378 // FIXME could this be made local to while loop? 2379 writeFrames = 0; 2380 2381 int lastGeneration = 0; 2382 2383 cacheParameters_l(); 2384 sleepTime = idleSleepTime; 2385 2386 if (mType == MIXER) { 2387 sleepTimeShift = 0; 2388 } 2389 2390 CpuStats cpuStats; 2391 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2392 2393 acquireWakeLock(); 2394 2395 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2396 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2397 // and then that string will be logged at the next convenient opportunity. 2398 const char *logString = NULL; 2399 2400 checkSilentMode_l(); 2401 2402 while (!exitPending()) 2403 { 2404 cpuStats.sample(myName); 2405 2406 Vector< sp<EffectChain> > effectChains; 2407 2408 { // scope for mLock 2409 2410 Mutex::Autolock _l(mLock); 2411 2412 processConfigEvents_l(); 2413 2414 if (logString != NULL) { 2415 mNBLogWriter->logTimestamp(); 2416 mNBLogWriter->log(logString); 2417 logString = NULL; 2418 } 2419 2420 // Gather the framesReleased counters for all active tracks, 2421 // and latch them atomically with the timestamp. 2422 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2423 mLatchD.mFramesReleased.clear(); 2424 size_t size = mActiveTracks.size(); 2425 for (size_t i = 0; i < size; i++) { 2426 sp<Track> t = mActiveTracks[i].promote(); 2427 if (t != 0) { 2428 mLatchD.mFramesReleased.add(t.get(), 2429 t->mAudioTrackServerProxy->framesReleased()); 2430 } 2431 } 2432 if (mLatchDValid) { 2433 mLatchQ = mLatchD; 2434 mLatchDValid = false; 2435 mLatchQValid = true; 2436 } 2437 2438 saveOutputTracks(); 2439 if (mSignalPending) { 2440 // A signal was raised while we were unlocked 2441 mSignalPending = false; 2442 } else if (waitingAsyncCallback_l()) { 2443 if (exitPending()) { 2444 break; 2445 } 2446 releaseWakeLock_l(); 2447 mWakeLockUids.clear(); 2448 mActiveTracksGeneration++; 2449 ALOGV("wait async completion"); 2450 mWaitWorkCV.wait(mLock); 2451 ALOGV("async completion/wake"); 2452 acquireWakeLock_l(); 2453 standbyTime = systemTime() + standbyDelay; 2454 sleepTime = 0; 2455 2456 continue; 2457 } 2458 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2459 isSuspended()) { 2460 // put audio hardware into standby after short delay 2461 if (shouldStandby_l()) { 2462 2463 threadLoop_standby(); 2464 2465 mStandby = true; 2466 } 2467 2468 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2469 // we're about to wait, flush the binder command buffer 2470 IPCThreadState::self()->flushCommands(); 2471 2472 clearOutputTracks(); 2473 2474 if (exitPending()) { 2475 break; 2476 } 2477 2478 releaseWakeLock_l(); 2479 mWakeLockUids.clear(); 2480 mActiveTracksGeneration++; 2481 // wait until we have something to do... 2482 ALOGV("%s going to sleep", myName.string()); 2483 mWaitWorkCV.wait(mLock); 2484 ALOGV("%s waking up", myName.string()); 2485 acquireWakeLock_l(); 2486 2487 mMixerStatus = MIXER_IDLE; 2488 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2489 mBytesWritten = 0; 2490 mBytesRemaining = 0; 2491 checkSilentMode_l(); 2492 2493 standbyTime = systemTime() + standbyDelay; 2494 sleepTime = idleSleepTime; 2495 if (mType == MIXER) { 2496 sleepTimeShift = 0; 2497 } 2498 2499 continue; 2500 } 2501 } 2502 // mMixerStatusIgnoringFastTracks is also updated internally 2503 mMixerStatus = prepareTracks_l(&tracksToRemove); 2504 2505 // compare with previously applied list 2506 if (lastGeneration != mActiveTracksGeneration) { 2507 // update wakelock 2508 updateWakeLockUids_l(mWakeLockUids); 2509 lastGeneration = mActiveTracksGeneration; 2510 } 2511 2512 // prevent any changes in effect chain list and in each effect chain 2513 // during mixing and effect process as the audio buffers could be deleted 2514 // or modified if an effect is created or deleted 2515 lockEffectChains_l(effectChains); 2516 } // mLock scope ends 2517 2518 if (mBytesRemaining == 0) { 2519 mCurrentWriteLength = 0; 2520 if (mMixerStatus == MIXER_TRACKS_READY) { 2521 // threadLoop_mix() sets mCurrentWriteLength 2522 threadLoop_mix(); 2523 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2524 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2525 // threadLoop_sleepTime sets sleepTime to 0 if data 2526 // must be written to HAL 2527 threadLoop_sleepTime(); 2528 if (sleepTime == 0) { 2529 mCurrentWriteLength = mSinkBufferSize; 2530 } 2531 } 2532 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2533 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2534 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2535 // or mSinkBuffer (if there are no effects). 2536 // 2537 // This is done pre-effects computation; if effects change to 2538 // support higher precision, this needs to move. 2539 // 2540 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2541 // TODO use sleepTime == 0 as an additional condition. 2542 if (mMixerBufferValid) { 2543 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2544 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2545 2546 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2547 mNormalFrameCount * mChannelCount); 2548 } 2549 2550 mBytesRemaining = mCurrentWriteLength; 2551 if (isSuspended()) { 2552 sleepTime = suspendSleepTimeUs(); 2553 // simulate write to HAL when suspended 2554 mBytesWritten += mSinkBufferSize; 2555 mBytesRemaining = 0; 2556 } 2557 2558 // only process effects if we're going to write 2559 if (sleepTime == 0 && mType != OFFLOAD) { 2560 for (size_t i = 0; i < effectChains.size(); i ++) { 2561 effectChains[i]->process_l(); 2562 } 2563 } 2564 } 2565 // Process effect chains for offloaded thread even if no audio 2566 // was read from audio track: process only updates effect state 2567 // and thus does have to be synchronized with audio writes but may have 2568 // to be called while waiting for async write callback 2569 if (mType == OFFLOAD) { 2570 for (size_t i = 0; i < effectChains.size(); i ++) { 2571 effectChains[i]->process_l(); 2572 } 2573 } 2574 2575 // Only if the Effects buffer is enabled and there is data in the 2576 // Effects buffer (buffer valid), we need to 2577 // copy into the sink buffer. 2578 // TODO use sleepTime == 0 as an additional condition. 2579 if (mEffectBufferValid) { 2580 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2581 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2582 mNormalFrameCount * mChannelCount); 2583 } 2584 2585 // enable changes in effect chain 2586 unlockEffectChains(effectChains); 2587 2588 if (!waitingAsyncCallback()) { 2589 // sleepTime == 0 means we must write to audio hardware 2590 if (sleepTime == 0) { 2591 if (mBytesRemaining) { 2592 ssize_t ret = threadLoop_write(); 2593 if (ret < 0) { 2594 mBytesRemaining = 0; 2595 } else { 2596 mBytesWritten += ret; 2597 mBytesRemaining -= ret; 2598 } 2599 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2600 (mMixerStatus == MIXER_DRAIN_ALL)) { 2601 threadLoop_drain(); 2602 } 2603 if (mType == MIXER) { 2604 // write blocked detection 2605 nsecs_t now = systemTime(); 2606 nsecs_t delta = now - mLastWriteTime; 2607 if (!mStandby && delta > maxPeriod) { 2608 mNumDelayedWrites++; 2609 if ((now - lastWarning) > kWarningThrottleNs) { 2610 ATRACE_NAME("underrun"); 2611 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2612 ns2ms(delta), mNumDelayedWrites, this); 2613 lastWarning = now; 2614 } 2615 } 2616 } 2617 2618 } else { 2619 usleep(sleepTime); 2620 } 2621 } 2622 2623 // Finally let go of removed track(s), without the lock held 2624 // since we can't guarantee the destructors won't acquire that 2625 // same lock. This will also mutate and push a new fast mixer state. 2626 threadLoop_removeTracks(tracksToRemove); 2627 tracksToRemove.clear(); 2628 2629 // FIXME I don't understand the need for this here; 2630 // it was in the original code but maybe the 2631 // assignment in saveOutputTracks() makes this unnecessary? 2632 clearOutputTracks(); 2633 2634 // Effect chains will be actually deleted here if they were removed from 2635 // mEffectChains list during mixing or effects processing 2636 effectChains.clear(); 2637 2638 // FIXME Note that the above .clear() is no longer necessary since effectChains 2639 // is now local to this block, but will keep it for now (at least until merge done). 2640 } 2641 2642 threadLoop_exit(); 2643 2644 if (!mStandby) { 2645 threadLoop_standby(); 2646 mStandby = true; 2647 } 2648 2649 releaseWakeLock(); 2650 mWakeLockUids.clear(); 2651 mActiveTracksGeneration++; 2652 2653 ALOGV("Thread %p type %d exiting", this, mType); 2654 return false; 2655} 2656 2657// removeTracks_l() must be called with ThreadBase::mLock held 2658void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2659{ 2660 size_t count = tracksToRemove.size(); 2661 if (count > 0) { 2662 for (size_t i=0 ; i<count ; i++) { 2663 const sp<Track>& track = tracksToRemove.itemAt(i); 2664 mActiveTracks.remove(track); 2665 mWakeLockUids.remove(track->uid()); 2666 mActiveTracksGeneration++; 2667 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2668 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2669 if (chain != 0) { 2670 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2671 track->sessionId()); 2672 chain->decActiveTrackCnt(); 2673 } 2674 if (track->isTerminated()) { 2675 removeTrack_l(track); 2676 } 2677 } 2678 } 2679 2680} 2681 2682status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2683{ 2684 if (mNormalSink != 0) { 2685 return mNormalSink->getTimestamp(timestamp); 2686 } 2687 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2688 uint64_t position64; 2689 int ret = mOutput->stream->get_presentation_position( 2690 mOutput->stream, &position64, ×tamp.mTime); 2691 if (ret == 0) { 2692 timestamp.mPosition = (uint32_t)position64; 2693 return NO_ERROR; 2694 } 2695 } 2696 return INVALID_OPERATION; 2697} 2698 2699status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2700 audio_patch_handle_t *handle) 2701{ 2702 status_t status = NO_ERROR; 2703 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2704 // store new device and send to effects 2705 audio_devices_t type = AUDIO_DEVICE_NONE; 2706 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2707 type |= patch->sinks[i].ext.device.type; 2708 } 2709 mOutDevice = type; 2710 for (size_t i = 0; i < mEffectChains.size(); i++) { 2711 mEffectChains[i]->setDevice_l(mOutDevice); 2712 } 2713 2714 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2715 status = hwDevice->create_audio_patch(hwDevice, 2716 patch->num_sources, 2717 patch->sources, 2718 patch->num_sinks, 2719 patch->sinks, 2720 handle); 2721 } else { 2722 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2723 } 2724 return status; 2725} 2726 2727status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2728{ 2729 status_t status = NO_ERROR; 2730 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2731 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2732 status = hwDevice->release_audio_patch(hwDevice, handle); 2733 } else { 2734 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2735 } 2736 return status; 2737} 2738 2739void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2740{ 2741 Mutex::Autolock _l(mLock); 2742 mTracks.add(track); 2743} 2744 2745void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2746{ 2747 Mutex::Autolock _l(mLock); 2748 destroyTrack_l(track); 2749} 2750 2751void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2752{ 2753 ThreadBase::getAudioPortConfig(config); 2754 config->role = AUDIO_PORT_ROLE_SOURCE; 2755 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2756 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2757} 2758 2759// ---------------------------------------------------------------------------- 2760 2761AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2762 audio_io_handle_t id, audio_devices_t device, type_t type) 2763 : PlaybackThread(audioFlinger, output, id, device, type), 2764 // mAudioMixer below 2765 // mFastMixer below 2766 mFastMixerFutex(0) 2767 // mOutputSink below 2768 // mPipeSink below 2769 // mNormalSink below 2770{ 2771 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2772 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2773 "mFrameCount=%d, mNormalFrameCount=%d", 2774 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2775 mNormalFrameCount); 2776 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2777 2778 // create an NBAIO sink for the HAL output stream, and negotiate 2779 mOutputSink = new AudioStreamOutSink(output->stream); 2780 size_t numCounterOffers = 0; 2781 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2782 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2783 ALOG_ASSERT(index == 0); 2784 2785 // initialize fast mixer depending on configuration 2786 bool initFastMixer; 2787 switch (kUseFastMixer) { 2788 case FastMixer_Never: 2789 initFastMixer = false; 2790 break; 2791 case FastMixer_Always: 2792 initFastMixer = true; 2793 break; 2794 case FastMixer_Static: 2795 case FastMixer_Dynamic: 2796 initFastMixer = mFrameCount < mNormalFrameCount; 2797 break; 2798 } 2799 if (initFastMixer) { 2800 audio_format_t fastMixerFormat; 2801 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2802 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2803 } else { 2804 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2805 } 2806 if (mFormat != fastMixerFormat) { 2807 // change our Sink format to accept our intermediate precision 2808 mFormat = fastMixerFormat; 2809 free(mSinkBuffer); 2810 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2811 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2812 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2813 } 2814 2815 // create a MonoPipe to connect our submix to FastMixer 2816 NBAIO_Format format = mOutputSink->format(); 2817 NBAIO_Format origformat = 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, origformat); 2838 const NBAIO_Format offers2[1] = {origformat}; 2839 numCounterOffers = 0; 2840 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 2841 ALOG_ASSERT(index == 0); 2842 mTeeSink = teeSink; 2843 PipeReader *teeSource = new PipeReader(*teeSink); 2844 numCounterOffers = 0; 2845 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 2846 ALOG_ASSERT(index == 0); 2847 mTeeSource = teeSource; 2848 } 2849#endif 2850 2851 // create fast mixer and configure it initially with just one fast track for our submix 2852 mFastMixer = new FastMixer(); 2853 FastMixerStateQueue *sq = mFastMixer->sq(); 2854#ifdef STATE_QUEUE_DUMP 2855 sq->setObserverDump(&mStateQueueObserverDump); 2856 sq->setMutatorDump(&mStateQueueMutatorDump); 2857#endif 2858 FastMixerState *state = sq->begin(); 2859 FastTrack *fastTrack = &state->mFastTracks[0]; 2860 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2861 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2862 fastTrack->mVolumeProvider = NULL; 2863 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2864 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2865 fastTrack->mGeneration++; 2866 state->mFastTracksGen++; 2867 state->mTrackMask = 1; 2868 // fast mixer will use the HAL output sink 2869 state->mOutputSink = mOutputSink.get(); 2870 state->mOutputSinkGen++; 2871 state->mFrameCount = mFrameCount; 2872 state->mCommand = FastMixerState::COLD_IDLE; 2873 // already done in constructor initialization list 2874 //mFastMixerFutex = 0; 2875 state->mColdFutexAddr = &mFastMixerFutex; 2876 state->mColdGen++; 2877 state->mDumpState = &mFastMixerDumpState; 2878#ifdef TEE_SINK 2879 state->mTeeSink = mTeeSink.get(); 2880#endif 2881 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2882 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2883 sq->end(); 2884 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2885 2886 // start the fast mixer 2887 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2888 pid_t tid = mFastMixer->getTid(); 2889 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2890 if (err != 0) { 2891 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2892 kPriorityFastMixer, getpid_cached, tid, err); 2893 } 2894 2895#ifdef AUDIO_WATCHDOG 2896 // create and start the watchdog 2897 mAudioWatchdog = new AudioWatchdog(); 2898 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2899 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2900 tid = mAudioWatchdog->getTid(); 2901 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2902 if (err != 0) { 2903 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2904 kPriorityFastMixer, getpid_cached, tid, err); 2905 } 2906#endif 2907 2908 } 2909 2910 switch (kUseFastMixer) { 2911 case FastMixer_Never: 2912 case FastMixer_Dynamic: 2913 mNormalSink = mOutputSink; 2914 break; 2915 case FastMixer_Always: 2916 mNormalSink = mPipeSink; 2917 break; 2918 case FastMixer_Static: 2919 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2920 break; 2921 } 2922} 2923 2924AudioFlinger::MixerThread::~MixerThread() 2925{ 2926 if (mFastMixer != 0) { 2927 FastMixerStateQueue *sq = mFastMixer->sq(); 2928 FastMixerState *state = sq->begin(); 2929 if (state->mCommand == FastMixerState::COLD_IDLE) { 2930 int32_t old = android_atomic_inc(&mFastMixerFutex); 2931 if (old == -1) { 2932 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2933 } 2934 } 2935 state->mCommand = FastMixerState::EXIT; 2936 sq->end(); 2937 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2938 mFastMixer->join(); 2939 // Though the fast mixer thread has exited, it's state queue is still valid. 2940 // We'll use that extract the final state which contains one remaining fast track 2941 // corresponding to our sub-mix. 2942 state = sq->begin(); 2943 ALOG_ASSERT(state->mTrackMask == 1); 2944 FastTrack *fastTrack = &state->mFastTracks[0]; 2945 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2946 delete fastTrack->mBufferProvider; 2947 sq->end(false /*didModify*/); 2948 mFastMixer.clear(); 2949#ifdef AUDIO_WATCHDOG 2950 if (mAudioWatchdog != 0) { 2951 mAudioWatchdog->requestExit(); 2952 mAudioWatchdog->requestExitAndWait(); 2953 mAudioWatchdog.clear(); 2954 } 2955#endif 2956 } 2957 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2958 delete mAudioMixer; 2959} 2960 2961 2962uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2963{ 2964 if (mFastMixer != 0) { 2965 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2966 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2967 } 2968 return latency; 2969} 2970 2971 2972void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2973{ 2974 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2975} 2976 2977ssize_t AudioFlinger::MixerThread::threadLoop_write() 2978{ 2979 // FIXME we should only do one push per cycle; confirm this is true 2980 // Start the fast mixer if it's not already running 2981 if (mFastMixer != 0) { 2982 FastMixerStateQueue *sq = mFastMixer->sq(); 2983 FastMixerState *state = sq->begin(); 2984 if (state->mCommand != FastMixerState::MIX_WRITE && 2985 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2986 if (state->mCommand == FastMixerState::COLD_IDLE) { 2987 int32_t old = android_atomic_inc(&mFastMixerFutex); 2988 if (old == -1) { 2989 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2990 } 2991#ifdef AUDIO_WATCHDOG 2992 if (mAudioWatchdog != 0) { 2993 mAudioWatchdog->resume(); 2994 } 2995#endif 2996 } 2997 state->mCommand = FastMixerState::MIX_WRITE; 2998 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2999 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 3000 sq->end(); 3001 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3002 if (kUseFastMixer == FastMixer_Dynamic) { 3003 mNormalSink = mPipeSink; 3004 } 3005 } else { 3006 sq->end(false /*didModify*/); 3007 } 3008 } 3009 return PlaybackThread::threadLoop_write(); 3010} 3011 3012void AudioFlinger::MixerThread::threadLoop_standby() 3013{ 3014 // Idle the fast mixer if it's currently running 3015 if (mFastMixer != 0) { 3016 FastMixerStateQueue *sq = mFastMixer->sq(); 3017 FastMixerState *state = sq->begin(); 3018 if (!(state->mCommand & FastMixerState::IDLE)) { 3019 state->mCommand = FastMixerState::COLD_IDLE; 3020 state->mColdFutexAddr = &mFastMixerFutex; 3021 state->mColdGen++; 3022 mFastMixerFutex = 0; 3023 sq->end(); 3024 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3025 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3026 if (kUseFastMixer == FastMixer_Dynamic) { 3027 mNormalSink = mOutputSink; 3028 } 3029#ifdef AUDIO_WATCHDOG 3030 if (mAudioWatchdog != 0) { 3031 mAudioWatchdog->pause(); 3032 } 3033#endif 3034 } else { 3035 sq->end(false /*didModify*/); 3036 } 3037 } 3038 PlaybackThread::threadLoop_standby(); 3039} 3040 3041bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3042{ 3043 return false; 3044} 3045 3046bool AudioFlinger::PlaybackThread::shouldStandby_l() 3047{ 3048 return !mStandby; 3049} 3050 3051bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3052{ 3053 Mutex::Autolock _l(mLock); 3054 return waitingAsyncCallback_l(); 3055} 3056 3057// shared by MIXER and DIRECT, overridden by DUPLICATING 3058void AudioFlinger::PlaybackThread::threadLoop_standby() 3059{ 3060 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3061 mOutput->stream->common.standby(&mOutput->stream->common); 3062 if (mUseAsyncWrite != 0) { 3063 // discard any pending drain or write ack by incrementing sequence 3064 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3065 mDrainSequence = (mDrainSequence + 2) & ~1; 3066 ALOG_ASSERT(mCallbackThread != 0); 3067 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3068 mCallbackThread->setDraining(mDrainSequence); 3069 } 3070} 3071 3072void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3073{ 3074 ALOGV("signal playback thread"); 3075 broadcast_l(); 3076} 3077 3078void AudioFlinger::MixerThread::threadLoop_mix() 3079{ 3080 // obtain the presentation timestamp of the next output buffer 3081 int64_t pts; 3082 status_t status = INVALID_OPERATION; 3083 3084 if (mNormalSink != 0) { 3085 status = mNormalSink->getNextWriteTimestamp(&pts); 3086 } else { 3087 status = mOutputSink->getNextWriteTimestamp(&pts); 3088 } 3089 3090 if (status != NO_ERROR) { 3091 pts = AudioBufferProvider::kInvalidPTS; 3092 } 3093 3094 // mix buffers... 3095 mAudioMixer->process(pts); 3096 mCurrentWriteLength = mSinkBufferSize; 3097 // increase sleep time progressively when application underrun condition clears. 3098 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3099 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3100 // such that we would underrun the audio HAL. 3101 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3102 sleepTimeShift--; 3103 } 3104 sleepTime = 0; 3105 standbyTime = systemTime() + standbyDelay; 3106 //TODO: delay standby when effects have a tail 3107 3108} 3109 3110void AudioFlinger::MixerThread::threadLoop_sleepTime() 3111{ 3112 // If no tracks are ready, sleep once for the duration of an output 3113 // buffer size, then write 0s to the output 3114 if (sleepTime == 0) { 3115 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3116 sleepTime = activeSleepTime >> sleepTimeShift; 3117 if (sleepTime < kMinThreadSleepTimeUs) { 3118 sleepTime = kMinThreadSleepTimeUs; 3119 } 3120 // reduce sleep time in case of consecutive application underruns to avoid 3121 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3122 // duration we would end up writing less data than needed by the audio HAL if 3123 // the condition persists. 3124 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3125 sleepTimeShift++; 3126 } 3127 } else { 3128 sleepTime = idleSleepTime; 3129 } 3130 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3131 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3132 // before effects processing or output. 3133 if (mMixerBufferValid) { 3134 memset(mMixerBuffer, 0, mMixerBufferSize); 3135 } else { 3136 memset(mSinkBuffer, 0, mSinkBufferSize); 3137 } 3138 sleepTime = 0; 3139 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3140 "anticipated start"); 3141 } 3142 // TODO add standby time extension fct of effect tail 3143} 3144 3145// prepareTracks_l() must be called with ThreadBase::mLock held 3146AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3147 Vector< sp<Track> > *tracksToRemove) 3148{ 3149 3150 mixer_state mixerStatus = MIXER_IDLE; 3151 // find out which tracks need to be processed 3152 size_t count = mActiveTracks.size(); 3153 size_t mixedTracks = 0; 3154 size_t tracksWithEffect = 0; 3155 // counts only _active_ fast tracks 3156 size_t fastTracks = 0; 3157 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3158 3159 float masterVolume = mMasterVolume; 3160 bool masterMute = mMasterMute; 3161 3162 if (masterMute) { 3163 masterVolume = 0; 3164 } 3165 // Delegate master volume control to effect in output mix effect chain if needed 3166 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3167 if (chain != 0) { 3168 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3169 chain->setVolume_l(&v, &v); 3170 masterVolume = (float)((v + (1 << 23)) >> 24); 3171 chain.clear(); 3172 } 3173 3174 // prepare a new state to push 3175 FastMixerStateQueue *sq = NULL; 3176 FastMixerState *state = NULL; 3177 bool didModify = false; 3178 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3179 if (mFastMixer != 0) { 3180 sq = mFastMixer->sq(); 3181 state = sq->begin(); 3182 } 3183 3184 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3185 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3186 3187 for (size_t i=0 ; i<count ; i++) { 3188 const sp<Track> t = mActiveTracks[i].promote(); 3189 if (t == 0) { 3190 continue; 3191 } 3192 3193 // this const just means the local variable doesn't change 3194 Track* const track = t.get(); 3195 3196 // process fast tracks 3197 if (track->isFastTrack()) { 3198 3199 // It's theoretically possible (though unlikely) for a fast track to be created 3200 // and then removed within the same normal mix cycle. This is not a problem, as 3201 // the track never becomes active so it's fast mixer slot is never touched. 3202 // The converse, of removing an (active) track and then creating a new track 3203 // at the identical fast mixer slot within the same normal mix cycle, 3204 // is impossible because the slot isn't marked available until the end of each cycle. 3205 int j = track->mFastIndex; 3206 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3207 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3208 FastTrack *fastTrack = &state->mFastTracks[j]; 3209 3210 // Determine whether the track is currently in underrun condition, 3211 // and whether it had a recent underrun. 3212 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3213 FastTrackUnderruns underruns = ftDump->mUnderruns; 3214 uint32_t recentFull = (underruns.mBitFields.mFull - 3215 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3216 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3217 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3218 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3219 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3220 uint32_t recentUnderruns = recentPartial + recentEmpty; 3221 track->mObservedUnderruns = underruns; 3222 // don't count underruns that occur while stopping or pausing 3223 // or stopped which can occur when flush() is called while active 3224 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3225 recentUnderruns > 0) { 3226 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3227 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3228 } 3229 3230 // This is similar to the state machine for normal tracks, 3231 // with a few modifications for fast tracks. 3232 bool isActive = true; 3233 switch (track->mState) { 3234 case TrackBase::STOPPING_1: 3235 // track stays active in STOPPING_1 state until first underrun 3236 if (recentUnderruns > 0 || track->isTerminated()) { 3237 track->mState = TrackBase::STOPPING_2; 3238 } 3239 break; 3240 case TrackBase::PAUSING: 3241 // ramp down is not yet implemented 3242 track->setPaused(); 3243 break; 3244 case TrackBase::RESUMING: 3245 // ramp up is not yet implemented 3246 track->mState = TrackBase::ACTIVE; 3247 break; 3248 case TrackBase::ACTIVE: 3249 if (recentFull > 0 || recentPartial > 0) { 3250 // track has provided at least some frames recently: reset retry count 3251 track->mRetryCount = kMaxTrackRetries; 3252 } 3253 if (recentUnderruns == 0) { 3254 // no recent underruns: stay active 3255 break; 3256 } 3257 // there has recently been an underrun of some kind 3258 if (track->sharedBuffer() == 0) { 3259 // were any of the recent underruns "empty" (no frames available)? 3260 if (recentEmpty == 0) { 3261 // no, then ignore the partial underruns as they are allowed indefinitely 3262 break; 3263 } 3264 // there has recently been an "empty" underrun: decrement the retry counter 3265 if (--(track->mRetryCount) > 0) { 3266 break; 3267 } 3268 // indicate to client process that the track was disabled because of underrun; 3269 // it will then automatically call start() when data is available 3270 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3271 // remove from active list, but state remains ACTIVE [confusing but true] 3272 isActive = false; 3273 break; 3274 } 3275 // fall through 3276 case TrackBase::STOPPING_2: 3277 case TrackBase::PAUSED: 3278 case TrackBase::STOPPED: 3279 case TrackBase::FLUSHED: // flush() while active 3280 // Check for presentation complete if track is inactive 3281 // We have consumed all the buffers of this track. 3282 // This would be incomplete if we auto-paused on underrun 3283 { 3284 size_t audioHALFrames = 3285 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3286 size_t framesWritten = mBytesWritten / mFrameSize; 3287 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3288 // track stays in active list until presentation is complete 3289 break; 3290 } 3291 } 3292 if (track->isStopping_2()) { 3293 track->mState = TrackBase::STOPPED; 3294 } 3295 if (track->isStopped()) { 3296 // Can't reset directly, as fast mixer is still polling this track 3297 // track->reset(); 3298 // So instead mark this track as needing to be reset after push with ack 3299 resetMask |= 1 << i; 3300 } 3301 isActive = false; 3302 break; 3303 case TrackBase::IDLE: 3304 default: 3305 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3306 } 3307 3308 if (isActive) { 3309 // was it previously inactive? 3310 if (!(state->mTrackMask & (1 << j))) { 3311 ExtendedAudioBufferProvider *eabp = track; 3312 VolumeProvider *vp = track; 3313 fastTrack->mBufferProvider = eabp; 3314 fastTrack->mVolumeProvider = vp; 3315 fastTrack->mChannelMask = track->mChannelMask; 3316 fastTrack->mFormat = track->mFormat; 3317 fastTrack->mGeneration++; 3318 state->mTrackMask |= 1 << j; 3319 didModify = true; 3320 // no acknowledgement required for newly active tracks 3321 } 3322 // cache the combined master volume and stream type volume for fast mixer; this 3323 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3324 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3325 ++fastTracks; 3326 } else { 3327 // was it previously active? 3328 if (state->mTrackMask & (1 << j)) { 3329 fastTrack->mBufferProvider = NULL; 3330 fastTrack->mGeneration++; 3331 state->mTrackMask &= ~(1 << j); 3332 didModify = true; 3333 // If any fast tracks were removed, we must wait for acknowledgement 3334 // because we're about to decrement the last sp<> on those tracks. 3335 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3336 } else { 3337 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3338 } 3339 tracksToRemove->add(track); 3340 // Avoids a misleading display in dumpsys 3341 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3342 } 3343 continue; 3344 } 3345 3346 { // local variable scope to avoid goto warning 3347 3348 audio_track_cblk_t* cblk = track->cblk(); 3349 3350 // The first time a track is added we wait 3351 // for all its buffers to be filled before processing it 3352 int name = track->name(); 3353 // make sure that we have enough frames to mix one full buffer. 3354 // enforce this condition only once to enable draining the buffer in case the client 3355 // app does not call stop() and relies on underrun to stop: 3356 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3357 // during last round 3358 size_t desiredFrames; 3359 uint32_t sr = track->sampleRate(); 3360 if (sr == mSampleRate) { 3361 desiredFrames = mNormalFrameCount; 3362 } else { 3363 // +1 for rounding and +1 for additional sample needed for interpolation 3364 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3365 // add frames already consumed but not yet released by the resampler 3366 // because mAudioTrackServerProxy->framesReady() will include these frames 3367 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3368#if 0 3369 // the minimum track buffer size is normally twice the number of frames necessary 3370 // to fill one buffer and the resampler should not leave more than one buffer worth 3371 // of unreleased frames after each pass, but just in case... 3372 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3373#endif 3374 } 3375 uint32_t minFrames = 1; 3376 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3377 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3378 minFrames = desiredFrames; 3379 } 3380 3381 size_t framesReady = track->framesReady(); 3382 if ((framesReady >= minFrames) && track->isReady() && 3383 !track->isPaused() && !track->isTerminated()) 3384 { 3385 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3386 3387 mixedTracks++; 3388 3389 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3390 // there is an effect chain connected to the track 3391 chain.clear(); 3392 if (track->mainBuffer() != mSinkBuffer && 3393 track->mainBuffer() != mMixerBuffer) { 3394 if (mEffectBufferEnabled) { 3395 mEffectBufferValid = true; // Later can set directly. 3396 } 3397 chain = getEffectChain_l(track->sessionId()); 3398 // Delegate volume control to effect in track effect chain if needed 3399 if (chain != 0) { 3400 tracksWithEffect++; 3401 } else { 3402 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3403 "session %d", 3404 name, track->sessionId()); 3405 } 3406 } 3407 3408 3409 int param = AudioMixer::VOLUME; 3410 if (track->mFillingUpStatus == Track::FS_FILLED) { 3411 // no ramp for the first volume setting 3412 track->mFillingUpStatus = Track::FS_ACTIVE; 3413 if (track->mState == TrackBase::RESUMING) { 3414 track->mState = TrackBase::ACTIVE; 3415 param = AudioMixer::RAMP_VOLUME; 3416 } 3417 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3418 // FIXME should not make a decision based on mServer 3419 } else if (cblk->mServer != 0) { 3420 // If the track is stopped before the first frame was mixed, 3421 // do not apply ramp 3422 param = AudioMixer::RAMP_VOLUME; 3423 } 3424 3425 // compute volume for this track 3426 uint32_t vl, vr; // in U8.24 integer format 3427 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3428 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3429 vl = vr = 0; 3430 vlf = vrf = vaf = 0.; 3431 if (track->isPausing()) { 3432 track->setPaused(); 3433 } 3434 } else { 3435 3436 // read original volumes with volume control 3437 float typeVolume = mStreamTypes[track->streamType()].volume; 3438 float v = masterVolume * typeVolume; 3439 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3440 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3441 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3442 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3443 // track volumes come from shared memory, so can't be trusted and must be clamped 3444 if (vlf > GAIN_FLOAT_UNITY) { 3445 ALOGV("Track left volume out of range: %.3g", vlf); 3446 vlf = GAIN_FLOAT_UNITY; 3447 } 3448 if (vrf > GAIN_FLOAT_UNITY) { 3449 ALOGV("Track right volume out of range: %.3g", vrf); 3450 vrf = GAIN_FLOAT_UNITY; 3451 } 3452 // now apply the master volume and stream type volume 3453 vlf *= v; 3454 vrf *= v; 3455 // assuming master volume and stream type volume each go up to 1.0, 3456 // then derive vl and vr as U8.24 versions for the effect chain 3457 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3458 vl = (uint32_t) (scaleto8_24 * vlf); 3459 vr = (uint32_t) (scaleto8_24 * vrf); 3460 // vl and vr are now in U8.24 format 3461 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3462 // send level comes from shared memory and so may be corrupt 3463 if (sendLevel > MAX_GAIN_INT) { 3464 ALOGV("Track send level out of range: %04X", sendLevel); 3465 sendLevel = MAX_GAIN_INT; 3466 } 3467 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3468 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3469 } 3470 3471 // Delegate volume control to effect in track effect chain if needed 3472 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3473 // Do not ramp volume if volume is controlled by effect 3474 param = AudioMixer::VOLUME; 3475 // Update remaining floating point volume levels 3476 vlf = (float)vl / (1 << 24); 3477 vrf = (float)vr / (1 << 24); 3478 track->mHasVolumeController = true; 3479 } else { 3480 // force no volume ramp when volume controller was just disabled or removed 3481 // from effect chain to avoid volume spike 3482 if (track->mHasVolumeController) { 3483 param = AudioMixer::VOLUME; 3484 } 3485 track->mHasVolumeController = false; 3486 } 3487 3488 // XXX: these things DON'T need to be done each time 3489 mAudioMixer->setBufferProvider(name, track); 3490 mAudioMixer->enable(name); 3491 3492 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3493 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3494 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3495 mAudioMixer->setParameter( 3496 name, 3497 AudioMixer::TRACK, 3498 AudioMixer::FORMAT, (void *)track->format()); 3499 mAudioMixer->setParameter( 3500 name, 3501 AudioMixer::TRACK, 3502 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3503 mAudioMixer->setParameter( 3504 name, 3505 AudioMixer::TRACK, 3506 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3507 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3508 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3509 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3510 if (reqSampleRate == 0) { 3511 reqSampleRate = mSampleRate; 3512 } else if (reqSampleRate > maxSampleRate) { 3513 reqSampleRate = maxSampleRate; 3514 } 3515 mAudioMixer->setParameter( 3516 name, 3517 AudioMixer::RESAMPLE, 3518 AudioMixer::SAMPLE_RATE, 3519 (void *)(uintptr_t)reqSampleRate); 3520 /* 3521 * Select the appropriate output buffer for the track. 3522 * 3523 * Tracks with effects go into their own effects chain buffer 3524 * and from there into either mEffectBuffer or mSinkBuffer. 3525 * 3526 * Other tracks can use mMixerBuffer for higher precision 3527 * channel accumulation. If this buffer is enabled 3528 * (mMixerBufferEnabled true), then selected tracks will accumulate 3529 * into it. 3530 * 3531 */ 3532 if (mMixerBufferEnabled 3533 && (track->mainBuffer() == mSinkBuffer 3534 || track->mainBuffer() == mMixerBuffer)) { 3535 mAudioMixer->setParameter( 3536 name, 3537 AudioMixer::TRACK, 3538 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3539 mAudioMixer->setParameter( 3540 name, 3541 AudioMixer::TRACK, 3542 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3543 // TODO: override track->mainBuffer()? 3544 mMixerBufferValid = true; 3545 } else { 3546 mAudioMixer->setParameter( 3547 name, 3548 AudioMixer::TRACK, 3549 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3550 mAudioMixer->setParameter( 3551 name, 3552 AudioMixer::TRACK, 3553 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3554 } 3555 mAudioMixer->setParameter( 3556 name, 3557 AudioMixer::TRACK, 3558 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3559 3560 // reset retry count 3561 track->mRetryCount = kMaxTrackRetries; 3562 3563 // If one track is ready, set the mixer ready if: 3564 // - the mixer was not ready during previous round OR 3565 // - no other track is not ready 3566 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3567 mixerStatus != MIXER_TRACKS_ENABLED) { 3568 mixerStatus = MIXER_TRACKS_READY; 3569 } 3570 } else { 3571 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3572 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3573 } 3574 // clear effect chain input buffer if an active track underruns to avoid sending 3575 // previous audio buffer again to effects 3576 chain = getEffectChain_l(track->sessionId()); 3577 if (chain != 0) { 3578 chain->clearInputBuffer(); 3579 } 3580 3581 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3582 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3583 track->isStopped() || track->isPaused()) { 3584 // We have consumed all the buffers of this track. 3585 // Remove it from the list of active tracks. 3586 // TODO: use actual buffer filling status instead of latency when available from 3587 // audio HAL 3588 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3589 size_t framesWritten = mBytesWritten / mFrameSize; 3590 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3591 if (track->isStopped()) { 3592 track->reset(); 3593 } 3594 tracksToRemove->add(track); 3595 } 3596 } else { 3597 // No buffers for this track. Give it a few chances to 3598 // fill a buffer, then remove it from active list. 3599 if (--(track->mRetryCount) <= 0) { 3600 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3601 tracksToRemove->add(track); 3602 // indicate to client process that the track was disabled because of underrun; 3603 // it will then automatically call start() when data is available 3604 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3605 // If one track is not ready, mark the mixer also not ready if: 3606 // - the mixer was ready during previous round OR 3607 // - no other track is ready 3608 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3609 mixerStatus != MIXER_TRACKS_READY) { 3610 mixerStatus = MIXER_TRACKS_ENABLED; 3611 } 3612 } 3613 mAudioMixer->disable(name); 3614 } 3615 3616 } // local variable scope to avoid goto warning 3617track_is_ready: ; 3618 3619 } 3620 3621 // Push the new FastMixer state if necessary 3622 bool pauseAudioWatchdog = false; 3623 if (didModify) { 3624 state->mFastTracksGen++; 3625 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3626 if (kUseFastMixer == FastMixer_Dynamic && 3627 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3628 state->mCommand = FastMixerState::COLD_IDLE; 3629 state->mColdFutexAddr = &mFastMixerFutex; 3630 state->mColdGen++; 3631 mFastMixerFutex = 0; 3632 if (kUseFastMixer == FastMixer_Dynamic) { 3633 mNormalSink = mOutputSink; 3634 } 3635 // If we go into cold idle, need to wait for acknowledgement 3636 // so that fast mixer stops doing I/O. 3637 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3638 pauseAudioWatchdog = true; 3639 } 3640 } 3641 if (sq != NULL) { 3642 sq->end(didModify); 3643 sq->push(block); 3644 } 3645#ifdef AUDIO_WATCHDOG 3646 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3647 mAudioWatchdog->pause(); 3648 } 3649#endif 3650 3651 // Now perform the deferred reset on fast tracks that have stopped 3652 while (resetMask != 0) { 3653 size_t i = __builtin_ctz(resetMask); 3654 ALOG_ASSERT(i < count); 3655 resetMask &= ~(1 << i); 3656 sp<Track> t = mActiveTracks[i].promote(); 3657 if (t == 0) { 3658 continue; 3659 } 3660 Track* track = t.get(); 3661 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3662 track->reset(); 3663 } 3664 3665 // remove all the tracks that need to be... 3666 removeTracks_l(*tracksToRemove); 3667 3668 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3669 mEffectBufferValid = true; 3670 } 3671 3672 if (mEffectBufferValid) { 3673 // as long as there are effects we should clear the effects buffer, to avoid 3674 // passing a non-clean buffer to the effect chain 3675 memset(mEffectBuffer, 0, mEffectBufferSize); 3676 } 3677 // sink or mix buffer must be cleared if all tracks are connected to an 3678 // effect chain as in this case the mixer will not write to the sink or mix buffer 3679 // and track effects will accumulate into it 3680 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3681 (mixedTracks == 0 && fastTracks > 0))) { 3682 // FIXME as a performance optimization, should remember previous zero status 3683 if (mMixerBufferValid) { 3684 memset(mMixerBuffer, 0, mMixerBufferSize); 3685 // TODO: In testing, mSinkBuffer below need not be cleared because 3686 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3687 // after mixing. 3688 // 3689 // To enforce this guarantee: 3690 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3691 // (mixedTracks == 0 && fastTracks > 0)) 3692 // must imply MIXER_TRACKS_READY. 3693 // Later, we may clear buffers regardless, and skip much of this logic. 3694 } 3695 // FIXME as a performance optimization, should remember previous zero status 3696 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3697 } 3698 3699 // if any fast tracks, then status is ready 3700 mMixerStatusIgnoringFastTracks = mixerStatus; 3701 if (fastTracks > 0) { 3702 mixerStatus = MIXER_TRACKS_READY; 3703 } 3704 return mixerStatus; 3705} 3706 3707// getTrackName_l() must be called with ThreadBase::mLock held 3708int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3709 audio_format_t format, int sessionId) 3710{ 3711 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3712} 3713 3714// deleteTrackName_l() must be called with ThreadBase::mLock held 3715void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3716{ 3717 ALOGV("remove track (%d) and delete from mixer", name); 3718 mAudioMixer->deleteTrackName(name); 3719} 3720 3721// checkForNewParameter_l() must be called with ThreadBase::mLock held 3722bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3723 status_t& status) 3724{ 3725 bool reconfig = false; 3726 3727 status = NO_ERROR; 3728 3729 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3730 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3731 if (mFastMixer != 0) { 3732 FastMixerStateQueue *sq = mFastMixer->sq(); 3733 FastMixerState *state = sq->begin(); 3734 if (!(state->mCommand & FastMixerState::IDLE)) { 3735 previousCommand = state->mCommand; 3736 state->mCommand = FastMixerState::HOT_IDLE; 3737 sq->end(); 3738 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3739 } else { 3740 sq->end(false /*didModify*/); 3741 } 3742 } 3743 3744 AudioParameter param = AudioParameter(keyValuePair); 3745 int value; 3746 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3747 reconfig = true; 3748 } 3749 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3750 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3751 status = BAD_VALUE; 3752 } else { 3753 // no need to save value, since it's constant 3754 reconfig = true; 3755 } 3756 } 3757 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3758 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3759 status = BAD_VALUE; 3760 } else { 3761 // no need to save value, since it's constant 3762 reconfig = true; 3763 } 3764 } 3765 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3766 // do not accept frame count changes if tracks are open as the track buffer 3767 // size depends on frame count and correct behavior would not be guaranteed 3768 // if frame count is changed after track creation 3769 if (!mTracks.isEmpty()) { 3770 status = INVALID_OPERATION; 3771 } else { 3772 reconfig = true; 3773 } 3774 } 3775 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3776#ifdef ADD_BATTERY_DATA 3777 // when changing the audio output device, call addBatteryData to notify 3778 // the change 3779 if (mOutDevice != value) { 3780 uint32_t params = 0; 3781 // check whether speaker is on 3782 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3783 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3784 } 3785 3786 audio_devices_t deviceWithoutSpeaker 3787 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3788 // check if any other device (except speaker) is on 3789 if (value & deviceWithoutSpeaker ) { 3790 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3791 } 3792 3793 if (params != 0) { 3794 addBatteryData(params); 3795 } 3796 } 3797#endif 3798 3799 // forward device change to effects that have requested to be 3800 // aware of attached audio device. 3801 if (value != AUDIO_DEVICE_NONE) { 3802 mOutDevice = value; 3803 for (size_t i = 0; i < mEffectChains.size(); i++) { 3804 mEffectChains[i]->setDevice_l(mOutDevice); 3805 } 3806 } 3807 } 3808 3809 if (status == NO_ERROR) { 3810 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3811 keyValuePair.string()); 3812 if (!mStandby && status == INVALID_OPERATION) { 3813 mOutput->stream->common.standby(&mOutput->stream->common); 3814 mStandby = true; 3815 mBytesWritten = 0; 3816 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3817 keyValuePair.string()); 3818 } 3819 if (status == NO_ERROR && reconfig) { 3820 readOutputParameters_l(); 3821 delete mAudioMixer; 3822 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3823 for (size_t i = 0; i < mTracks.size() ; i++) { 3824 int name = getTrackName_l(mTracks[i]->mChannelMask, 3825 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3826 if (name < 0) { 3827 break; 3828 } 3829 mTracks[i]->mName = name; 3830 } 3831 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3832 } 3833 } 3834 3835 if (!(previousCommand & FastMixerState::IDLE)) { 3836 ALOG_ASSERT(mFastMixer != 0); 3837 FastMixerStateQueue *sq = mFastMixer->sq(); 3838 FastMixerState *state = sq->begin(); 3839 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3840 state->mCommand = previousCommand; 3841 sq->end(); 3842 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3843 } 3844 3845 return reconfig; 3846} 3847 3848 3849void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3850{ 3851 const size_t SIZE = 256; 3852 char buffer[SIZE]; 3853 String8 result; 3854 3855 PlaybackThread::dumpInternals(fd, args); 3856 3857 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3858 3859 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3860 const FastMixerDumpState copy(mFastMixerDumpState); 3861 copy.dump(fd); 3862 3863#ifdef STATE_QUEUE_DUMP 3864 // Similar for state queue 3865 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3866 observerCopy.dump(fd); 3867 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3868 mutatorCopy.dump(fd); 3869#endif 3870 3871#ifdef TEE_SINK 3872 // Write the tee output to a .wav file 3873 dumpTee(fd, mTeeSource, mId); 3874#endif 3875 3876#ifdef AUDIO_WATCHDOG 3877 if (mAudioWatchdog != 0) { 3878 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3879 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3880 wdCopy.dump(fd); 3881 } 3882#endif 3883} 3884 3885uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3886{ 3887 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3888} 3889 3890uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3891{ 3892 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3893} 3894 3895void AudioFlinger::MixerThread::cacheParameters_l() 3896{ 3897 PlaybackThread::cacheParameters_l(); 3898 3899 // FIXME: Relaxed timing because of a certain device that can't meet latency 3900 // Should be reduced to 2x after the vendor fixes the driver issue 3901 // increase threshold again due to low power audio mode. The way this warning 3902 // threshold is calculated and its usefulness should be reconsidered anyway. 3903 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3904} 3905 3906// ---------------------------------------------------------------------------- 3907 3908AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3909 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3910 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3911 // mLeftVolFloat, mRightVolFloat 3912{ 3913} 3914 3915AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3916 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3917 ThreadBase::type_t type) 3918 : PlaybackThread(audioFlinger, output, id, device, type) 3919 // mLeftVolFloat, mRightVolFloat 3920{ 3921} 3922 3923AudioFlinger::DirectOutputThread::~DirectOutputThread() 3924{ 3925} 3926 3927void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3928{ 3929 audio_track_cblk_t* cblk = track->cblk(); 3930 float left, right; 3931 3932 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3933 left = right = 0; 3934 } else { 3935 float typeVolume = mStreamTypes[track->streamType()].volume; 3936 float v = mMasterVolume * typeVolume; 3937 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3938 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3939 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3940 if (left > GAIN_FLOAT_UNITY) { 3941 left = GAIN_FLOAT_UNITY; 3942 } 3943 left *= v; 3944 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3945 if (right > GAIN_FLOAT_UNITY) { 3946 right = GAIN_FLOAT_UNITY; 3947 } 3948 right *= v; 3949 } 3950 3951 if (lastTrack) { 3952 if (left != mLeftVolFloat || right != mRightVolFloat) { 3953 mLeftVolFloat = left; 3954 mRightVolFloat = right; 3955 3956 // Convert volumes from float to 8.24 3957 uint32_t vl = (uint32_t)(left * (1 << 24)); 3958 uint32_t vr = (uint32_t)(right * (1 << 24)); 3959 3960 // Delegate volume control to effect in track effect chain if needed 3961 // only one effect chain can be present on DirectOutputThread, so if 3962 // there is one, the track is connected to it 3963 if (!mEffectChains.isEmpty()) { 3964 mEffectChains[0]->setVolume_l(&vl, &vr); 3965 left = (float)vl / (1 << 24); 3966 right = (float)vr / (1 << 24); 3967 } 3968 if (mOutput->stream->set_volume) { 3969 mOutput->stream->set_volume(mOutput->stream, left, right); 3970 } 3971 } 3972 } 3973} 3974 3975 3976AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3977 Vector< sp<Track> > *tracksToRemove 3978) 3979{ 3980 size_t count = mActiveTracks.size(); 3981 mixer_state mixerStatus = MIXER_IDLE; 3982 3983 // find out which tracks need to be processed 3984 for (size_t i = 0; i < count; i++) { 3985 sp<Track> t = mActiveTracks[i].promote(); 3986 // The track died recently 3987 if (t == 0) { 3988 continue; 3989 } 3990 3991 Track* const track = t.get(); 3992 audio_track_cblk_t* cblk = track->cblk(); 3993 // Only consider last track started for volume and mixer state control. 3994 // In theory an older track could underrun and restart after the new one starts 3995 // but as we only care about the transition phase between two tracks on a 3996 // direct output, it is not a problem to ignore the underrun case. 3997 sp<Track> l = mLatestActiveTrack.promote(); 3998 bool last = l.get() == track; 3999 4000 // The first time a track is added we wait 4001 // for all its buffers to be filled before processing it 4002 uint32_t minFrames; 4003 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 4004 minFrames = mNormalFrameCount; 4005 } else { 4006 minFrames = 1; 4007 } 4008 4009 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4010 !track->isStopping_2() && !track->isStopped()) 4011 { 4012 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4013 4014 if (track->mFillingUpStatus == Track::FS_FILLED) { 4015 track->mFillingUpStatus = Track::FS_ACTIVE; 4016 // make sure processVolume_l() will apply new volume even if 0 4017 mLeftVolFloat = mRightVolFloat = -1.0; 4018 if (track->mState == TrackBase::RESUMING) { 4019 track->mState = TrackBase::ACTIVE; 4020 } 4021 } 4022 4023 // compute volume for this track 4024 processVolume_l(track, last); 4025 if (last) { 4026 // reset retry count 4027 track->mRetryCount = kMaxTrackRetriesDirect; 4028 mActiveTrack = t; 4029 mixerStatus = MIXER_TRACKS_READY; 4030 } 4031 } else { 4032 // clear effect chain input buffer if the last active track started underruns 4033 // to avoid sending previous audio buffer again to effects 4034 if (!mEffectChains.isEmpty() && last) { 4035 mEffectChains[0]->clearInputBuffer(); 4036 } 4037 if (track->isStopping_1()) { 4038 track->mState = TrackBase::STOPPING_2; 4039 } 4040 if ((track->sharedBuffer() != 0) || track->isStopped() || 4041 track->isStopping_2() || track->isPaused()) { 4042 // We have consumed all the buffers of this track. 4043 // Remove it from the list of active tracks. 4044 size_t audioHALFrames; 4045 if (audio_is_linear_pcm(mFormat)) { 4046 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4047 } else { 4048 audioHALFrames = 0; 4049 } 4050 4051 size_t framesWritten = mBytesWritten / mFrameSize; 4052 if (mStandby || !last || 4053 track->presentationComplete(framesWritten, audioHALFrames)) { 4054 if (track->isStopping_2()) { 4055 track->mState = TrackBase::STOPPED; 4056 } 4057 if (track->isStopped()) { 4058 if (track->mState == TrackBase::FLUSHED) { 4059 flushHw_l(); 4060 } 4061 track->reset(); 4062 } 4063 tracksToRemove->add(track); 4064 } 4065 } else { 4066 // No buffers for this track. Give it a few chances to 4067 // fill a buffer, then remove it from active list. 4068 // Only consider last track started for mixer state control 4069 if (--(track->mRetryCount) <= 0) { 4070 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4071 tracksToRemove->add(track); 4072 // indicate to client process that the track was disabled because of underrun; 4073 // it will then automatically call start() when data is available 4074 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4075 } else if (last) { 4076 mixerStatus = MIXER_TRACKS_ENABLED; 4077 } 4078 } 4079 } 4080 } 4081 4082 // remove all the tracks that need to be... 4083 removeTracks_l(*tracksToRemove); 4084 4085 return mixerStatus; 4086} 4087 4088void AudioFlinger::DirectOutputThread::threadLoop_mix() 4089{ 4090 size_t frameCount = mFrameCount; 4091 int8_t *curBuf = (int8_t *)mSinkBuffer; 4092 // output audio to hardware 4093 while (frameCount) { 4094 AudioBufferProvider::Buffer buffer; 4095 buffer.frameCount = frameCount; 4096 mActiveTrack->getNextBuffer(&buffer); 4097 if (buffer.raw == NULL) { 4098 memset(curBuf, 0, frameCount * mFrameSize); 4099 break; 4100 } 4101 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4102 frameCount -= buffer.frameCount; 4103 curBuf += buffer.frameCount * mFrameSize; 4104 mActiveTrack->releaseBuffer(&buffer); 4105 } 4106 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4107 sleepTime = 0; 4108 standbyTime = systemTime() + standbyDelay; 4109 mActiveTrack.clear(); 4110} 4111 4112void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4113{ 4114 if (sleepTime == 0) { 4115 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4116 sleepTime = activeSleepTime; 4117 } else { 4118 sleepTime = idleSleepTime; 4119 } 4120 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4121 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4122 sleepTime = 0; 4123 } 4124} 4125 4126// getTrackName_l() must be called with ThreadBase::mLock held 4127int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4128 audio_format_t format __unused, int sessionId __unused) 4129{ 4130 return 0; 4131} 4132 4133// deleteTrackName_l() must be called with ThreadBase::mLock held 4134void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4135{ 4136} 4137 4138// checkForNewParameter_l() must be called with ThreadBase::mLock held 4139bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4140 status_t& status) 4141{ 4142 bool reconfig = false; 4143 4144 status = NO_ERROR; 4145 4146 AudioParameter param = AudioParameter(keyValuePair); 4147 int value; 4148 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4149 // forward device change to effects that have requested to be 4150 // aware of attached audio device. 4151 if (value != AUDIO_DEVICE_NONE) { 4152 mOutDevice = value; 4153 for (size_t i = 0; i < mEffectChains.size(); i++) { 4154 mEffectChains[i]->setDevice_l(mOutDevice); 4155 } 4156 } 4157 } 4158 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4159 // do not accept frame count changes if tracks are open as the track buffer 4160 // size depends on frame count and correct behavior would not be garantied 4161 // if frame count is changed after track creation 4162 if (!mTracks.isEmpty()) { 4163 status = INVALID_OPERATION; 4164 } else { 4165 reconfig = true; 4166 } 4167 } 4168 if (status == NO_ERROR) { 4169 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4170 keyValuePair.string()); 4171 if (!mStandby && status == INVALID_OPERATION) { 4172 mOutput->stream->common.standby(&mOutput->stream->common); 4173 mStandby = true; 4174 mBytesWritten = 0; 4175 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4176 keyValuePair.string()); 4177 } 4178 if (status == NO_ERROR && reconfig) { 4179 readOutputParameters_l(); 4180 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4181 } 4182 } 4183 4184 return reconfig; 4185} 4186 4187uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4188{ 4189 uint32_t time; 4190 if (audio_is_linear_pcm(mFormat)) { 4191 time = PlaybackThread::activeSleepTimeUs(); 4192 } else { 4193 time = 10000; 4194 } 4195 return time; 4196} 4197 4198uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4199{ 4200 uint32_t time; 4201 if (audio_is_linear_pcm(mFormat)) { 4202 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4203 } else { 4204 time = 10000; 4205 } 4206 return time; 4207} 4208 4209uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4210{ 4211 uint32_t time; 4212 if (audio_is_linear_pcm(mFormat)) { 4213 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4214 } else { 4215 time = 10000; 4216 } 4217 return time; 4218} 4219 4220void AudioFlinger::DirectOutputThread::cacheParameters_l() 4221{ 4222 PlaybackThread::cacheParameters_l(); 4223 4224 // use shorter standby delay as on normal output to release 4225 // hardware resources as soon as possible 4226 if (audio_is_linear_pcm(mFormat)) { 4227 standbyDelay = microseconds(activeSleepTime*2); 4228 } else { 4229 standbyDelay = kOffloadStandbyDelayNs; 4230 } 4231} 4232 4233void AudioFlinger::DirectOutputThread::flushHw_l() 4234{ 4235 if (mOutput->stream->flush != NULL) 4236 mOutput->stream->flush(mOutput->stream); 4237} 4238 4239// ---------------------------------------------------------------------------- 4240 4241AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4242 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4243 : Thread(false /*canCallJava*/), 4244 mPlaybackThread(playbackThread), 4245 mWriteAckSequence(0), 4246 mDrainSequence(0) 4247{ 4248} 4249 4250AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4251{ 4252} 4253 4254void AudioFlinger::AsyncCallbackThread::onFirstRef() 4255{ 4256 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4257} 4258 4259bool AudioFlinger::AsyncCallbackThread::threadLoop() 4260{ 4261 while (!exitPending()) { 4262 uint32_t writeAckSequence; 4263 uint32_t drainSequence; 4264 4265 { 4266 Mutex::Autolock _l(mLock); 4267 while (!((mWriteAckSequence & 1) || 4268 (mDrainSequence & 1) || 4269 exitPending())) { 4270 mWaitWorkCV.wait(mLock); 4271 } 4272 4273 if (exitPending()) { 4274 break; 4275 } 4276 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4277 mWriteAckSequence, mDrainSequence); 4278 writeAckSequence = mWriteAckSequence; 4279 mWriteAckSequence &= ~1; 4280 drainSequence = mDrainSequence; 4281 mDrainSequence &= ~1; 4282 } 4283 { 4284 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4285 if (playbackThread != 0) { 4286 if (writeAckSequence & 1) { 4287 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4288 } 4289 if (drainSequence & 1) { 4290 playbackThread->resetDraining(drainSequence >> 1); 4291 } 4292 } 4293 } 4294 } 4295 return false; 4296} 4297 4298void AudioFlinger::AsyncCallbackThread::exit() 4299{ 4300 ALOGV("AsyncCallbackThread::exit"); 4301 Mutex::Autolock _l(mLock); 4302 requestExit(); 4303 mWaitWorkCV.broadcast(); 4304} 4305 4306void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4307{ 4308 Mutex::Autolock _l(mLock); 4309 // bit 0 is cleared 4310 mWriteAckSequence = sequence << 1; 4311} 4312 4313void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4314{ 4315 Mutex::Autolock _l(mLock); 4316 // ignore unexpected callbacks 4317 if (mWriteAckSequence & 2) { 4318 mWriteAckSequence |= 1; 4319 mWaitWorkCV.signal(); 4320 } 4321} 4322 4323void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4324{ 4325 Mutex::Autolock _l(mLock); 4326 // bit 0 is cleared 4327 mDrainSequence = sequence << 1; 4328} 4329 4330void AudioFlinger::AsyncCallbackThread::resetDraining() 4331{ 4332 Mutex::Autolock _l(mLock); 4333 // ignore unexpected callbacks 4334 if (mDrainSequence & 2) { 4335 mDrainSequence |= 1; 4336 mWaitWorkCV.signal(); 4337 } 4338} 4339 4340 4341// ---------------------------------------------------------------------------- 4342AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4343 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4344 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4345 mHwPaused(false), 4346 mFlushPending(false), 4347 mPausedBytesRemaining(0) 4348{ 4349 //FIXME: mStandby should be set to true by ThreadBase constructor 4350 mStandby = true; 4351} 4352 4353void AudioFlinger::OffloadThread::threadLoop_exit() 4354{ 4355 if (mFlushPending || mHwPaused) { 4356 // If a flush is pending or track was paused, just discard buffered data 4357 flushHw_l(); 4358 } else { 4359 mMixerStatus = MIXER_DRAIN_ALL; 4360 threadLoop_drain(); 4361 } 4362 if (mUseAsyncWrite) { 4363 ALOG_ASSERT(mCallbackThread != 0); 4364 mCallbackThread->exit(); 4365 } 4366 PlaybackThread::threadLoop_exit(); 4367} 4368 4369AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4370 Vector< sp<Track> > *tracksToRemove 4371) 4372{ 4373 size_t count = mActiveTracks.size(); 4374 4375 mixer_state mixerStatus = MIXER_IDLE; 4376 bool doHwPause = false; 4377 bool doHwResume = false; 4378 4379 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4380 4381 // find out which tracks need to be processed 4382 for (size_t i = 0; i < count; i++) { 4383 sp<Track> t = mActiveTracks[i].promote(); 4384 // The track died recently 4385 if (t == 0) { 4386 continue; 4387 } 4388 Track* const track = t.get(); 4389 audio_track_cblk_t* cblk = track->cblk(); 4390 // Only consider last track started for volume and mixer state control. 4391 // In theory an older track could underrun and restart after the new one starts 4392 // but as we only care about the transition phase between two tracks on a 4393 // direct output, it is not a problem to ignore the underrun case. 4394 sp<Track> l = mLatestActiveTrack.promote(); 4395 bool last = l.get() == track; 4396 4397 if (track->isInvalid()) { 4398 ALOGW("An invalidated track shouldn't be in active list"); 4399 tracksToRemove->add(track); 4400 continue; 4401 } 4402 4403 if (track->mState == TrackBase::IDLE) { 4404 ALOGW("An idle track shouldn't be in active list"); 4405 continue; 4406 } 4407 4408 if (track->isPausing()) { 4409 track->setPaused(); 4410 if (last) { 4411 if (!mHwPaused) { 4412 doHwPause = true; 4413 mHwPaused = true; 4414 } 4415 // If we were part way through writing the mixbuffer to 4416 // the HAL we must save this until we resume 4417 // BUG - this will be wrong if a different track is made active, 4418 // in that case we want to discard the pending data in the 4419 // mixbuffer and tell the client to present it again when the 4420 // track is resumed 4421 mPausedWriteLength = mCurrentWriteLength; 4422 mPausedBytesRemaining = mBytesRemaining; 4423 mBytesRemaining = 0; // stop writing 4424 } 4425 tracksToRemove->add(track); 4426 } else if (track->isFlushPending()) { 4427 track->flushAck(); 4428 if (last) { 4429 mFlushPending = true; 4430 } 4431 } else if (track->isResumePending()){ 4432 track->resumeAck(); 4433 if (last) { 4434 if (mPausedBytesRemaining) { 4435 // Need to continue write that was interrupted 4436 mCurrentWriteLength = mPausedWriteLength; 4437 mBytesRemaining = mPausedBytesRemaining; 4438 mPausedBytesRemaining = 0; 4439 } 4440 if (mHwPaused) { 4441 doHwResume = true; 4442 mHwPaused = false; 4443 // threadLoop_mix() will handle the case that we need to 4444 // resume an interrupted write 4445 } 4446 // enable write to audio HAL 4447 sleepTime = 0; 4448 4449 // Do not handle new data in this iteration even if track->framesReady() 4450 mixerStatus = MIXER_TRACKS_ENABLED; 4451 } 4452 } else if (track->framesReady() && track->isReady() && 4453 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4454 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4455 if (track->mFillingUpStatus == Track::FS_FILLED) { 4456 track->mFillingUpStatus = Track::FS_ACTIVE; 4457 // make sure processVolume_l() will apply new volume even if 0 4458 mLeftVolFloat = mRightVolFloat = -1.0; 4459 } 4460 4461 if (last) { 4462 sp<Track> previousTrack = mPreviousTrack.promote(); 4463 if (previousTrack != 0) { 4464 if (track != previousTrack.get()) { 4465 // Flush any data still being written from last track 4466 mBytesRemaining = 0; 4467 if (mPausedBytesRemaining) { 4468 // Last track was paused so we also need to flush saved 4469 // mixbuffer state and invalidate track so that it will 4470 // re-submit that unwritten data when it is next resumed 4471 mPausedBytesRemaining = 0; 4472 // Invalidate is a bit drastic - would be more efficient 4473 // to have a flag to tell client that some of the 4474 // previously written data was lost 4475 previousTrack->invalidate(); 4476 } 4477 // flush data already sent to the DSP if changing audio session as audio 4478 // comes from a different source. Also invalidate previous track to force a 4479 // seek when resuming. 4480 if (previousTrack->sessionId() != track->sessionId()) { 4481 previousTrack->invalidate(); 4482 } 4483 } 4484 } 4485 mPreviousTrack = track; 4486 // reset retry count 4487 track->mRetryCount = kMaxTrackRetriesOffload; 4488 mActiveTrack = t; 4489 mixerStatus = MIXER_TRACKS_READY; 4490 } 4491 } else { 4492 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4493 if (track->isStopping_1()) { 4494 // Hardware buffer can hold a large amount of audio so we must 4495 // wait for all current track's data to drain before we say 4496 // that the track is stopped. 4497 if (mBytesRemaining == 0) { 4498 // Only start draining when all data in mixbuffer 4499 // has been written 4500 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4501 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4502 // do not drain if no data was ever sent to HAL (mStandby == true) 4503 if (last && !mStandby) { 4504 // do not modify drain sequence if we are already draining. This happens 4505 // when resuming from pause after drain. 4506 if ((mDrainSequence & 1) == 0) { 4507 sleepTime = 0; 4508 standbyTime = systemTime() + standbyDelay; 4509 mixerStatus = MIXER_DRAIN_TRACK; 4510 mDrainSequence += 2; 4511 } 4512 if (mHwPaused) { 4513 // It is possible to move from PAUSED to STOPPING_1 without 4514 // a resume so we must ensure hardware is running 4515 doHwResume = true; 4516 mHwPaused = false; 4517 } 4518 } 4519 } 4520 } else if (track->isStopping_2()) { 4521 // Drain has completed or we are in standby, signal presentation complete 4522 if (!(mDrainSequence & 1) || !last || mStandby) { 4523 track->mState = TrackBase::STOPPED; 4524 size_t audioHALFrames = 4525 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4526 size_t framesWritten = 4527 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4528 track->presentationComplete(framesWritten, audioHALFrames); 4529 track->reset(); 4530 tracksToRemove->add(track); 4531 } 4532 } else { 4533 // No buffers for this track. Give it a few chances to 4534 // fill a buffer, then remove it from active list. 4535 if (--(track->mRetryCount) <= 0) { 4536 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4537 track->name()); 4538 tracksToRemove->add(track); 4539 // indicate to client process that the track was disabled because of underrun; 4540 // it will then automatically call start() when data is available 4541 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4542 } else if (last){ 4543 mixerStatus = MIXER_TRACKS_ENABLED; 4544 } 4545 } 4546 } 4547 // compute volume for this track 4548 processVolume_l(track, last); 4549 } 4550 4551 // make sure the pause/flush/resume sequence is executed in the right order. 4552 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4553 // before flush and then resume HW. This can happen in case of pause/flush/resume 4554 // if resume is received before pause is executed. 4555 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4556 mOutput->stream->pause(mOutput->stream); 4557 } 4558 if (mFlushPending) { 4559 flushHw_l(); 4560 mFlushPending = false; 4561 } 4562 if (!mStandby && doHwResume) { 4563 mOutput->stream->resume(mOutput->stream); 4564 } 4565 4566 // remove all the tracks that need to be... 4567 removeTracks_l(*tracksToRemove); 4568 4569 return mixerStatus; 4570} 4571 4572// must be called with thread mutex locked 4573bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4574{ 4575 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4576 mWriteAckSequence, mDrainSequence); 4577 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4578 return true; 4579 } 4580 return false; 4581} 4582 4583// must be called with thread mutex locked 4584bool AudioFlinger::OffloadThread::shouldStandby_l() 4585{ 4586 bool trackPaused = false; 4587 4588 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4589 // after a timeout and we will enter standby then. 4590 if (mTracks.size() > 0) { 4591 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4592 } 4593 4594 return !mStandby && !trackPaused; 4595} 4596 4597 4598bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4599{ 4600 Mutex::Autolock _l(mLock); 4601 return waitingAsyncCallback_l(); 4602} 4603 4604void AudioFlinger::OffloadThread::flushHw_l() 4605{ 4606 DirectOutputThread::flushHw_l(); 4607 // Flush anything still waiting in the mixbuffer 4608 mCurrentWriteLength = 0; 4609 mBytesRemaining = 0; 4610 mPausedWriteLength = 0; 4611 mPausedBytesRemaining = 0; 4612 mHwPaused = false; 4613 4614 if (mUseAsyncWrite) { 4615 // discard any pending drain or write ack by incrementing sequence 4616 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4617 mDrainSequence = (mDrainSequence + 2) & ~1; 4618 ALOG_ASSERT(mCallbackThread != 0); 4619 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4620 mCallbackThread->setDraining(mDrainSequence); 4621 } 4622} 4623 4624void AudioFlinger::OffloadThread::onAddNewTrack_l() 4625{ 4626 sp<Track> previousTrack = mPreviousTrack.promote(); 4627 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4628 4629 if (previousTrack != 0 && latestTrack != 0 && 4630 (previousTrack->sessionId() != latestTrack->sessionId())) { 4631 mFlushPending = true; 4632 } 4633 PlaybackThread::onAddNewTrack_l(); 4634} 4635 4636// ---------------------------------------------------------------------------- 4637 4638AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4639 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4640 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4641 DUPLICATING), 4642 mWaitTimeMs(UINT_MAX) 4643{ 4644 addOutputTrack(mainThread); 4645} 4646 4647AudioFlinger::DuplicatingThread::~DuplicatingThread() 4648{ 4649 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4650 mOutputTracks[i]->destroy(); 4651 } 4652} 4653 4654void AudioFlinger::DuplicatingThread::threadLoop_mix() 4655{ 4656 // mix buffers... 4657 if (outputsReady(outputTracks)) { 4658 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4659 } else { 4660 if (mMixerBufferValid) { 4661 memset(mMixerBuffer, 0, mMixerBufferSize); 4662 } else { 4663 memset(mSinkBuffer, 0, mSinkBufferSize); 4664 } 4665 } 4666 sleepTime = 0; 4667 writeFrames = mNormalFrameCount; 4668 mCurrentWriteLength = mSinkBufferSize; 4669 standbyTime = systemTime() + standbyDelay; 4670} 4671 4672void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4673{ 4674 if (sleepTime == 0) { 4675 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4676 sleepTime = activeSleepTime; 4677 } else { 4678 sleepTime = idleSleepTime; 4679 } 4680 } else if (mBytesWritten != 0) { 4681 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4682 writeFrames = mNormalFrameCount; 4683 memset(mSinkBuffer, 0, mSinkBufferSize); 4684 } else { 4685 // flush remaining overflow buffers in output tracks 4686 writeFrames = 0; 4687 } 4688 sleepTime = 0; 4689 } 4690} 4691 4692ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4693{ 4694 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4695 // for delivery downstream as needed. This in-place conversion is safe as 4696 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4697 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4698 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4699 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4700 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4701 } 4702 for (size_t i = 0; i < outputTracks.size(); i++) { 4703 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4704 } 4705 mStandby = false; 4706 return (ssize_t)mSinkBufferSize; 4707} 4708 4709void AudioFlinger::DuplicatingThread::threadLoop_standby() 4710{ 4711 // DuplicatingThread implements standby by stopping all tracks 4712 for (size_t i = 0; i < outputTracks.size(); i++) { 4713 outputTracks[i]->stop(); 4714 } 4715} 4716 4717void AudioFlinger::DuplicatingThread::saveOutputTracks() 4718{ 4719 outputTracks = mOutputTracks; 4720} 4721 4722void AudioFlinger::DuplicatingThread::clearOutputTracks() 4723{ 4724 outputTracks.clear(); 4725} 4726 4727void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4728{ 4729 Mutex::Autolock _l(mLock); 4730 // FIXME explain this formula 4731 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4732 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4733 // due to current usage case and restrictions on the AudioBufferProvider. 4734 // Actual buffer conversion is done in threadLoop_write(). 4735 // 4736 // TODO: This may change in the future, depending on multichannel 4737 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4738 OutputTrack *outputTrack = new OutputTrack(thread, 4739 this, 4740 mSampleRate, 4741 AUDIO_FORMAT_PCM_16_BIT, 4742 mChannelMask, 4743 frameCount, 4744 IPCThreadState::self()->getCallingUid()); 4745 if (outputTrack->cblk() != NULL) { 4746 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 4747 mOutputTracks.add(outputTrack); 4748 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4749 updateWaitTime_l(); 4750 } 4751} 4752 4753void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4754{ 4755 Mutex::Autolock _l(mLock); 4756 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4757 if (mOutputTracks[i]->thread() == thread) { 4758 mOutputTracks[i]->destroy(); 4759 mOutputTracks.removeAt(i); 4760 updateWaitTime_l(); 4761 return; 4762 } 4763 } 4764 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4765} 4766 4767// caller must hold mLock 4768void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4769{ 4770 mWaitTimeMs = UINT_MAX; 4771 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4772 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4773 if (strong != 0) { 4774 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4775 if (waitTimeMs < mWaitTimeMs) { 4776 mWaitTimeMs = waitTimeMs; 4777 } 4778 } 4779 } 4780} 4781 4782 4783bool AudioFlinger::DuplicatingThread::outputsReady( 4784 const SortedVector< sp<OutputTrack> > &outputTracks) 4785{ 4786 for (size_t i = 0; i < outputTracks.size(); i++) { 4787 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4788 if (thread == 0) { 4789 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4790 outputTracks[i].get()); 4791 return false; 4792 } 4793 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4794 // see note at standby() declaration 4795 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4796 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4797 thread.get()); 4798 return false; 4799 } 4800 } 4801 return true; 4802} 4803 4804uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4805{ 4806 return (mWaitTimeMs * 1000) / 2; 4807} 4808 4809void AudioFlinger::DuplicatingThread::cacheParameters_l() 4810{ 4811 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4812 updateWaitTime_l(); 4813 4814 MixerThread::cacheParameters_l(); 4815} 4816 4817// ---------------------------------------------------------------------------- 4818// Record 4819// ---------------------------------------------------------------------------- 4820 4821AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4822 AudioStreamIn *input, 4823 audio_io_handle_t id, 4824 audio_devices_t outDevice, 4825 audio_devices_t inDevice 4826#ifdef TEE_SINK 4827 , const sp<NBAIO_Sink>& teeSink 4828#endif 4829 ) : 4830 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4831 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4832 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4833 mRsmpInRear(0) 4834#ifdef TEE_SINK 4835 , mTeeSink(teeSink) 4836#endif 4837 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4838 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4839 // mFastCapture below 4840 , mFastCaptureFutex(0) 4841 // mInputSource 4842 // mPipeSink 4843 // mPipeSource 4844 , mPipeFramesP2(0) 4845 // mPipeMemory 4846 // mFastCaptureNBLogWriter 4847 , mFastTrackAvail(false) 4848{ 4849 snprintf(mName, kNameLength, "AudioIn_%X", id); 4850 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4851 4852 readInputParameters_l(); 4853 4854 // create an NBAIO source for the HAL input stream, and negotiate 4855 mInputSource = new AudioStreamInSource(input->stream); 4856 size_t numCounterOffers = 0; 4857 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4858 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4859 ALOG_ASSERT(index == 0); 4860 4861 // initialize fast capture depending on configuration 4862 bool initFastCapture; 4863 switch (kUseFastCapture) { 4864 case FastCapture_Never: 4865 initFastCapture = false; 4866 break; 4867 case FastCapture_Always: 4868 initFastCapture = true; 4869 break; 4870 case FastCapture_Static: 4871 uint32_t primaryOutputSampleRate; 4872 { 4873 AutoMutex _l(audioFlinger->mHardwareLock); 4874 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4875 } 4876 initFastCapture = 4877 // either capture sample rate is same as (a reasonable) primary output sample rate 4878 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4879 (mSampleRate == primaryOutputSampleRate)) || 4880 // or primary output sample rate is unknown, and capture sample rate is reasonable 4881 ((primaryOutputSampleRate == 0) && 4882 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4883 // and the buffer size is < 12 ms 4884 (mFrameCount * 1000) / mSampleRate < 12; 4885 break; 4886 // case FastCapture_Dynamic: 4887 } 4888 4889 if (initFastCapture) { 4890 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4891 NBAIO_Format format = mInputSource->format(); 4892 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4893 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4894 void *pipeBuffer; 4895 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4896 sp<IMemory> pipeMemory; 4897 if ((roHeap == 0) || 4898 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4899 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4900 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4901 goto failed; 4902 } 4903 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4904 memset(pipeBuffer, 0, pipeSize); 4905 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4906 const NBAIO_Format offers[1] = {format}; 4907 size_t numCounterOffers = 0; 4908 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4909 ALOG_ASSERT(index == 0); 4910 mPipeSink = pipe; 4911 PipeReader *pipeReader = new PipeReader(*pipe); 4912 numCounterOffers = 0; 4913 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4914 ALOG_ASSERT(index == 0); 4915 mPipeSource = pipeReader; 4916 mPipeFramesP2 = pipeFramesP2; 4917 mPipeMemory = pipeMemory; 4918 4919 // create fast capture 4920 mFastCapture = new FastCapture(); 4921 FastCaptureStateQueue *sq = mFastCapture->sq(); 4922#ifdef STATE_QUEUE_DUMP 4923 // FIXME 4924#endif 4925 FastCaptureState *state = sq->begin(); 4926 state->mCblk = NULL; 4927 state->mInputSource = mInputSource.get(); 4928 state->mInputSourceGen++; 4929 state->mPipeSink = pipe; 4930 state->mPipeSinkGen++; 4931 state->mFrameCount = mFrameCount; 4932 state->mCommand = FastCaptureState::COLD_IDLE; 4933 // already done in constructor initialization list 4934 //mFastCaptureFutex = 0; 4935 state->mColdFutexAddr = &mFastCaptureFutex; 4936 state->mColdGen++; 4937 state->mDumpState = &mFastCaptureDumpState; 4938#ifdef TEE_SINK 4939 // FIXME 4940#endif 4941 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4942 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4943 sq->end(); 4944 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4945 4946 // start the fast capture 4947 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4948 pid_t tid = mFastCapture->getTid(); 4949 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4950 if (err != 0) { 4951 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4952 kPriorityFastCapture, getpid_cached, tid, err); 4953 } 4954 4955#ifdef AUDIO_WATCHDOG 4956 // FIXME 4957#endif 4958 4959 mFastTrackAvail = true; 4960 } 4961failed: ; 4962 4963 // FIXME mNormalSource 4964} 4965 4966 4967AudioFlinger::RecordThread::~RecordThread() 4968{ 4969 if (mFastCapture != 0) { 4970 FastCaptureStateQueue *sq = mFastCapture->sq(); 4971 FastCaptureState *state = sq->begin(); 4972 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4973 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4974 if (old == -1) { 4975 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4976 } 4977 } 4978 state->mCommand = FastCaptureState::EXIT; 4979 sq->end(); 4980 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4981 mFastCapture->join(); 4982 mFastCapture.clear(); 4983 } 4984 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4985 mAudioFlinger->unregisterWriter(mNBLogWriter); 4986 delete[] mRsmpInBuffer; 4987} 4988 4989void AudioFlinger::RecordThread::onFirstRef() 4990{ 4991 run(mName, PRIORITY_URGENT_AUDIO); 4992} 4993 4994bool AudioFlinger::RecordThread::threadLoop() 4995{ 4996 nsecs_t lastWarning = 0; 4997 4998 inputStandBy(); 4999 5000reacquire_wakelock: 5001 sp<RecordTrack> activeTrack; 5002 int activeTracksGen; 5003 { 5004 Mutex::Autolock _l(mLock); 5005 size_t size = mActiveTracks.size(); 5006 activeTracksGen = mActiveTracksGen; 5007 if (size > 0) { 5008 // FIXME an arbitrary choice 5009 activeTrack = mActiveTracks[0]; 5010 acquireWakeLock_l(activeTrack->uid()); 5011 if (size > 1) { 5012 SortedVector<int> tmp; 5013 for (size_t i = 0; i < size; i++) { 5014 tmp.add(mActiveTracks[i]->uid()); 5015 } 5016 updateWakeLockUids_l(tmp); 5017 } 5018 } else { 5019 acquireWakeLock_l(-1); 5020 } 5021 } 5022 5023 // used to request a deferred sleep, to be executed later while mutex is unlocked 5024 uint32_t sleepUs = 0; 5025 5026 // loop while there is work to do 5027 for (;;) { 5028 Vector< sp<EffectChain> > effectChains; 5029 5030 // sleep with mutex unlocked 5031 if (sleepUs > 0) { 5032 usleep(sleepUs); 5033 sleepUs = 0; 5034 } 5035 5036 // activeTracks accumulates a copy of a subset of mActiveTracks 5037 Vector< sp<RecordTrack> > activeTracks; 5038 5039 // reference to the (first and only) active fast track 5040 sp<RecordTrack> fastTrack; 5041 5042 // reference to a fast track which is about to be removed 5043 sp<RecordTrack> fastTrackToRemove; 5044 5045 { // scope for mLock 5046 Mutex::Autolock _l(mLock); 5047 5048 processConfigEvents_l(); 5049 5050 // check exitPending here because checkForNewParameters_l() and 5051 // checkForNewParameters_l() can temporarily release mLock 5052 if (exitPending()) { 5053 break; 5054 } 5055 5056 // if no active track(s), then standby and release wakelock 5057 size_t size = mActiveTracks.size(); 5058 if (size == 0) { 5059 standbyIfNotAlreadyInStandby(); 5060 // exitPending() can't become true here 5061 releaseWakeLock_l(); 5062 ALOGV("RecordThread: loop stopping"); 5063 // go to sleep 5064 mWaitWorkCV.wait(mLock); 5065 ALOGV("RecordThread: loop starting"); 5066 goto reacquire_wakelock; 5067 } 5068 5069 if (mActiveTracksGen != activeTracksGen) { 5070 activeTracksGen = mActiveTracksGen; 5071 SortedVector<int> tmp; 5072 for (size_t i = 0; i < size; i++) { 5073 tmp.add(mActiveTracks[i]->uid()); 5074 } 5075 updateWakeLockUids_l(tmp); 5076 } 5077 5078 bool doBroadcast = false; 5079 for (size_t i = 0; i < size; ) { 5080 5081 activeTrack = mActiveTracks[i]; 5082 if (activeTrack->isTerminated()) { 5083 if (activeTrack->isFastTrack()) { 5084 ALOG_ASSERT(fastTrackToRemove == 0); 5085 fastTrackToRemove = activeTrack; 5086 } 5087 removeTrack_l(activeTrack); 5088 mActiveTracks.remove(activeTrack); 5089 mActiveTracksGen++; 5090 size--; 5091 continue; 5092 } 5093 5094 TrackBase::track_state activeTrackState = activeTrack->mState; 5095 switch (activeTrackState) { 5096 5097 case TrackBase::PAUSING: 5098 mActiveTracks.remove(activeTrack); 5099 mActiveTracksGen++; 5100 doBroadcast = true; 5101 size--; 5102 continue; 5103 5104 case TrackBase::STARTING_1: 5105 sleepUs = 10000; 5106 i++; 5107 continue; 5108 5109 case TrackBase::STARTING_2: 5110 doBroadcast = true; 5111 mStandby = false; 5112 activeTrack->mState = TrackBase::ACTIVE; 5113 break; 5114 5115 case TrackBase::ACTIVE: 5116 break; 5117 5118 case TrackBase::IDLE: 5119 i++; 5120 continue; 5121 5122 default: 5123 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5124 } 5125 5126 activeTracks.add(activeTrack); 5127 i++; 5128 5129 if (activeTrack->isFastTrack()) { 5130 ALOG_ASSERT(!mFastTrackAvail); 5131 ALOG_ASSERT(fastTrack == 0); 5132 fastTrack = activeTrack; 5133 } 5134 } 5135 if (doBroadcast) { 5136 mStartStopCond.broadcast(); 5137 } 5138 5139 // sleep if there are no active tracks to process 5140 if (activeTracks.size() == 0) { 5141 if (sleepUs == 0) { 5142 sleepUs = kRecordThreadSleepUs; 5143 } 5144 continue; 5145 } 5146 sleepUs = 0; 5147 5148 lockEffectChains_l(effectChains); 5149 } 5150 5151 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5152 5153 size_t size = effectChains.size(); 5154 for (size_t i = 0; i < size; i++) { 5155 // thread mutex is not locked, but effect chain is locked 5156 effectChains[i]->process_l(); 5157 } 5158 5159 // Push a new fast capture state if fast capture is not already running, or cblk change 5160 if (mFastCapture != 0) { 5161 FastCaptureStateQueue *sq = mFastCapture->sq(); 5162 FastCaptureState *state = sq->begin(); 5163 bool didModify = false; 5164 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5165 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5166 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5167 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5168 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5169 if (old == -1) { 5170 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5171 } 5172 } 5173 state->mCommand = FastCaptureState::READ_WRITE; 5174#if 0 // FIXME 5175 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5176 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5177#endif 5178 didModify = true; 5179 } 5180 audio_track_cblk_t *cblkOld = state->mCblk; 5181 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5182 if (cblkNew != cblkOld) { 5183 state->mCblk = cblkNew; 5184 // block until acked if removing a fast track 5185 if (cblkOld != NULL) { 5186 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5187 } 5188 didModify = true; 5189 } 5190 sq->end(didModify); 5191 if (didModify) { 5192 sq->push(block); 5193#if 0 5194 if (kUseFastCapture == FastCapture_Dynamic) { 5195 mNormalSource = mPipeSource; 5196 } 5197#endif 5198 } 5199 } 5200 5201 // now run the fast track destructor with thread mutex unlocked 5202 fastTrackToRemove.clear(); 5203 5204 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5205 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5206 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5207 // If destination is non-contiguous, first read past the nominal end of buffer, then 5208 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5209 5210 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5211 ssize_t framesRead; 5212 5213 // If an NBAIO source is present, use it to read the normal capture's data 5214 if (mPipeSource != 0) { 5215 size_t framesToRead = mBufferSize / mFrameSize; 5216 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5217 framesToRead, AudioBufferProvider::kInvalidPTS); 5218 if (framesRead == 0) { 5219 // since pipe is non-blocking, simulate blocking input 5220 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5221 } 5222 // otherwise use the HAL / AudioStreamIn directly 5223 } else { 5224 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5225 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5226 if (bytesRead < 0) { 5227 framesRead = bytesRead; 5228 } else { 5229 framesRead = bytesRead / mFrameSize; 5230 } 5231 } 5232 5233 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5234 ALOGE("read failed: framesRead=%d", framesRead); 5235 // Force input into standby so that it tries to recover at next read attempt 5236 inputStandBy(); 5237 sleepUs = kRecordThreadSleepUs; 5238 } 5239 if (framesRead <= 0) { 5240 goto unlock; 5241 } 5242 ALOG_ASSERT(framesRead > 0); 5243 5244 if (mTeeSink != 0) { 5245 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5246 } 5247 // If destination is non-contiguous, we now correct for reading past end of buffer. 5248 { 5249 size_t part1 = mRsmpInFramesP2 - rear; 5250 if ((size_t) framesRead > part1) { 5251 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5252 (framesRead - part1) * mFrameSize); 5253 } 5254 } 5255 rear = mRsmpInRear += framesRead; 5256 5257 size = activeTracks.size(); 5258 // loop over each active track 5259 for (size_t i = 0; i < size; i++) { 5260 activeTrack = activeTracks[i]; 5261 5262 // skip fast tracks, as those are handled directly by FastCapture 5263 if (activeTrack->isFastTrack()) { 5264 continue; 5265 } 5266 5267 enum { 5268 OVERRUN_UNKNOWN, 5269 OVERRUN_TRUE, 5270 OVERRUN_FALSE 5271 } overrun = OVERRUN_UNKNOWN; 5272 5273 // loop over getNextBuffer to handle circular sink 5274 for (;;) { 5275 5276 activeTrack->mSink.frameCount = ~0; 5277 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5278 size_t framesOut = activeTrack->mSink.frameCount; 5279 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5280 5281 int32_t front = activeTrack->mRsmpInFront; 5282 ssize_t filled = rear - front; 5283 size_t framesIn; 5284 5285 if (filled < 0) { 5286 // should not happen, but treat like a massive overrun and re-sync 5287 framesIn = 0; 5288 activeTrack->mRsmpInFront = rear; 5289 overrun = OVERRUN_TRUE; 5290 } else if ((size_t) filled <= mRsmpInFrames) { 5291 framesIn = (size_t) filled; 5292 } else { 5293 // client is not keeping up with server, but give it latest data 5294 framesIn = mRsmpInFrames; 5295 activeTrack->mRsmpInFront = front = rear - framesIn; 5296 overrun = OVERRUN_TRUE; 5297 } 5298 5299 if (framesOut == 0 || framesIn == 0) { 5300 break; 5301 } 5302 5303 if (activeTrack->mResampler == NULL) { 5304 // no resampling 5305 if (framesIn > framesOut) { 5306 framesIn = framesOut; 5307 } else { 5308 framesOut = framesIn; 5309 } 5310 int8_t *dst = activeTrack->mSink.i8; 5311 while (framesIn > 0) { 5312 front &= mRsmpInFramesP2 - 1; 5313 size_t part1 = mRsmpInFramesP2 - front; 5314 if (part1 > framesIn) { 5315 part1 = framesIn; 5316 } 5317 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5318 if (mChannelCount == activeTrack->mChannelCount) { 5319 memcpy(dst, src, part1 * mFrameSize); 5320 } else if (mChannelCount == 1) { 5321 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5322 part1); 5323 } else { 5324 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5325 part1); 5326 } 5327 dst += part1 * activeTrack->mFrameSize; 5328 front += part1; 5329 framesIn -= part1; 5330 } 5331 activeTrack->mRsmpInFront += framesOut; 5332 5333 } else { 5334 // resampling 5335 // FIXME framesInNeeded should really be part of resampler API, and should 5336 // depend on the SRC ratio 5337 // to keep mRsmpInBuffer full so resampler always has sufficient input 5338 size_t framesInNeeded; 5339 // FIXME only re-calculate when it changes, and optimize for common ratios 5340 // Do not precompute in/out because floating point is not associative 5341 // e.g. a*b/c != a*(b/c). 5342 const double in(mSampleRate); 5343 const double out(activeTrack->mSampleRate); 5344 framesInNeeded = ceil(framesOut * in / out) + 1; 5345 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5346 framesInNeeded, framesOut, in / out); 5347 // Although we theoretically have framesIn in circular buffer, some of those are 5348 // unreleased frames, and thus must be discounted for purpose of budgeting. 5349 size_t unreleased = activeTrack->mRsmpInUnrel; 5350 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5351 if (framesIn < framesInNeeded) { 5352 ALOGV("not enough to resample: have %u frames in but need %u in to " 5353 "produce %u out given in/out ratio of %.4g", 5354 framesIn, framesInNeeded, framesOut, in / out); 5355 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5356 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5357 if (newFramesOut == 0) { 5358 break; 5359 } 5360 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5361 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5362 framesInNeeded, newFramesOut, out / in); 5363 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5364 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5365 "given in/out ratio of %.4g", 5366 framesIn, framesInNeeded, newFramesOut, in / out); 5367 framesOut = newFramesOut; 5368 } else { 5369 ALOGV("success 1: have %u in and need %u in to produce %u out " 5370 "given in/out ratio of %.4g", 5371 framesIn, framesInNeeded, framesOut, in / out); 5372 } 5373 5374 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5375 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5376 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5377 delete[] activeTrack->mRsmpOutBuffer; 5378 // resampler always outputs stereo 5379 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5380 activeTrack->mRsmpOutFrameCount = framesOut; 5381 } 5382 5383 // resampler accumulates, but we only have one source track 5384 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5385 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5386 // FIXME how about having activeTrack implement this interface itself? 5387 activeTrack->mResamplerBufferProvider 5388 /*this*/ /* AudioBufferProvider* */); 5389 // ditherAndClamp() works as long as all buffers returned by 5390 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5391 if (activeTrack->mChannelCount == 1) { 5392 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5393 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5394 framesOut); 5395 // the resampler always outputs stereo samples: 5396 // do post stereo to mono conversion 5397 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5398 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5399 } else { 5400 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5401 activeTrack->mRsmpOutBuffer, framesOut); 5402 } 5403 // now done with mRsmpOutBuffer 5404 5405 } 5406 5407 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5408 overrun = OVERRUN_FALSE; 5409 } 5410 5411 if (activeTrack->mFramesToDrop == 0) { 5412 if (framesOut > 0) { 5413 activeTrack->mSink.frameCount = framesOut; 5414 activeTrack->releaseBuffer(&activeTrack->mSink); 5415 } 5416 } else { 5417 // FIXME could do a partial drop of framesOut 5418 if (activeTrack->mFramesToDrop > 0) { 5419 activeTrack->mFramesToDrop -= framesOut; 5420 if (activeTrack->mFramesToDrop <= 0) { 5421 activeTrack->clearSyncStartEvent(); 5422 } 5423 } else { 5424 activeTrack->mFramesToDrop += framesOut; 5425 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5426 activeTrack->mSyncStartEvent->isCancelled()) { 5427 ALOGW("Synced record %s, session %d, trigger session %d", 5428 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5429 activeTrack->sessionId(), 5430 (activeTrack->mSyncStartEvent != 0) ? 5431 activeTrack->mSyncStartEvent->triggerSession() : 0); 5432 activeTrack->clearSyncStartEvent(); 5433 } 5434 } 5435 } 5436 5437 if (framesOut == 0) { 5438 break; 5439 } 5440 } 5441 5442 switch (overrun) { 5443 case OVERRUN_TRUE: 5444 // client isn't retrieving buffers fast enough 5445 if (!activeTrack->setOverflow()) { 5446 nsecs_t now = systemTime(); 5447 // FIXME should lastWarning per track? 5448 if ((now - lastWarning) > kWarningThrottleNs) { 5449 ALOGW("RecordThread: buffer overflow"); 5450 lastWarning = now; 5451 } 5452 } 5453 break; 5454 case OVERRUN_FALSE: 5455 activeTrack->clearOverflow(); 5456 break; 5457 case OVERRUN_UNKNOWN: 5458 break; 5459 } 5460 5461 } 5462 5463unlock: 5464 // enable changes in effect chain 5465 unlockEffectChains(effectChains); 5466 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5467 } 5468 5469 standbyIfNotAlreadyInStandby(); 5470 5471 { 5472 Mutex::Autolock _l(mLock); 5473 for (size_t i = 0; i < mTracks.size(); i++) { 5474 sp<RecordTrack> track = mTracks[i]; 5475 track->invalidate(); 5476 } 5477 mActiveTracks.clear(); 5478 mActiveTracksGen++; 5479 mStartStopCond.broadcast(); 5480 } 5481 5482 releaseWakeLock(); 5483 5484 ALOGV("RecordThread %p exiting", this); 5485 return false; 5486} 5487 5488void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5489{ 5490 if (!mStandby) { 5491 inputStandBy(); 5492 mStandby = true; 5493 } 5494} 5495 5496void AudioFlinger::RecordThread::inputStandBy() 5497{ 5498 // Idle the fast capture if it's currently running 5499 if (mFastCapture != 0) { 5500 FastCaptureStateQueue *sq = mFastCapture->sq(); 5501 FastCaptureState *state = sq->begin(); 5502 if (!(state->mCommand & FastCaptureState::IDLE)) { 5503 state->mCommand = FastCaptureState::COLD_IDLE; 5504 state->mColdFutexAddr = &mFastCaptureFutex; 5505 state->mColdGen++; 5506 mFastCaptureFutex = 0; 5507 sq->end(); 5508 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5509 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5510#if 0 5511 if (kUseFastCapture == FastCapture_Dynamic) { 5512 // FIXME 5513 } 5514#endif 5515#ifdef AUDIO_WATCHDOG 5516 // FIXME 5517#endif 5518 } else { 5519 sq->end(false /*didModify*/); 5520 } 5521 } 5522 mInput->stream->common.standby(&mInput->stream->common); 5523} 5524 5525// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5526sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5527 const sp<AudioFlinger::Client>& client, 5528 uint32_t sampleRate, 5529 audio_format_t format, 5530 audio_channel_mask_t channelMask, 5531 size_t *pFrameCount, 5532 int sessionId, 5533 size_t *notificationFrames, 5534 int uid, 5535 IAudioFlinger::track_flags_t *flags, 5536 pid_t tid, 5537 status_t *status) 5538{ 5539 size_t frameCount = *pFrameCount; 5540 sp<RecordTrack> track; 5541 status_t lStatus; 5542 5543 // client expresses a preference for FAST, but we get the final say 5544 if (*flags & IAudioFlinger::TRACK_FAST) { 5545 if ( 5546 // use case: callback handler 5547 (tid != -1) && 5548 // frame count is not specified, or is exactly the pipe depth 5549 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5550 // PCM data 5551 audio_is_linear_pcm(format) && 5552 // native format 5553 (format == mFormat) && 5554 // native channel mask 5555 (channelMask == mChannelMask) && 5556 // native hardware sample rate 5557 (sampleRate == mSampleRate) && 5558 // record thread has an associated fast capture 5559 hasFastCapture() && 5560 // there are sufficient fast track slots available 5561 mFastTrackAvail 5562 ) { 5563 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5564 frameCount, mFrameCount); 5565 } else { 5566 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5567 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5568 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5569 frameCount, mFrameCount, mPipeFramesP2, 5570 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5571 hasFastCapture(), tid, mFastTrackAvail); 5572 *flags &= ~IAudioFlinger::TRACK_FAST; 5573 } 5574 } 5575 5576 // compute track buffer size in frames, and suggest the notification frame count 5577 if (*flags & IAudioFlinger::TRACK_FAST) { 5578 // fast track: frame count is exactly the pipe depth 5579 frameCount = mPipeFramesP2; 5580 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5581 *notificationFrames = mFrameCount; 5582 } else { 5583 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5584 // or 20 ms if there is a fast capture 5585 // TODO This could be a roundupRatio inline, and const 5586 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5587 * sampleRate + mSampleRate - 1) / mSampleRate; 5588 // minimum number of notification periods is at least kMinNotifications, 5589 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5590 static const size_t kMinNotifications = 3; 5591 static const uint32_t kMinMs = 30; 5592 // TODO This could be a roundupRatio inline 5593 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5594 // TODO This could be a roundupRatio inline 5595 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5596 maxNotificationFrames; 5597 const size_t minFrameCount = maxNotificationFrames * 5598 max(kMinNotifications, minNotificationsByMs); 5599 frameCount = max(frameCount, minFrameCount); 5600 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5601 *notificationFrames = maxNotificationFrames; 5602 } 5603 } 5604 *pFrameCount = frameCount; 5605 5606 lStatus = initCheck(); 5607 if (lStatus != NO_ERROR) { 5608 ALOGE("createRecordTrack_l() audio driver not initialized"); 5609 goto Exit; 5610 } 5611 5612 { // scope for mLock 5613 Mutex::Autolock _l(mLock); 5614 5615 track = new RecordTrack(this, client, sampleRate, 5616 format, channelMask, frameCount, NULL, sessionId, uid, 5617 *flags, TrackBase::TYPE_DEFAULT); 5618 5619 lStatus = track->initCheck(); 5620 if (lStatus != NO_ERROR) { 5621 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5622 // track must be cleared from the caller as the caller has the AF lock 5623 goto Exit; 5624 } 5625 mTracks.add(track); 5626 5627 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5628 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5629 mAudioFlinger->btNrecIsOff(); 5630 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5631 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5632 5633 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5634 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5635 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5636 // so ask activity manager to do this on our behalf 5637 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5638 } 5639 } 5640 5641 lStatus = NO_ERROR; 5642 5643Exit: 5644 *status = lStatus; 5645 return track; 5646} 5647 5648status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5649 AudioSystem::sync_event_t event, 5650 int triggerSession) 5651{ 5652 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5653 sp<ThreadBase> strongMe = this; 5654 status_t status = NO_ERROR; 5655 5656 if (event == AudioSystem::SYNC_EVENT_NONE) { 5657 recordTrack->clearSyncStartEvent(); 5658 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5659 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5660 triggerSession, 5661 recordTrack->sessionId(), 5662 syncStartEventCallback, 5663 recordTrack); 5664 // Sync event can be cancelled by the trigger session if the track is not in a 5665 // compatible state in which case we start record immediately 5666 if (recordTrack->mSyncStartEvent->isCancelled()) { 5667 recordTrack->clearSyncStartEvent(); 5668 } else { 5669 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5670 recordTrack->mFramesToDrop = - 5671 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5672 } 5673 } 5674 5675 { 5676 // This section is a rendezvous between binder thread executing start() and RecordThread 5677 AutoMutex lock(mLock); 5678 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5679 if (recordTrack->mState == TrackBase::PAUSING) { 5680 ALOGV("active record track PAUSING -> ACTIVE"); 5681 recordTrack->mState = TrackBase::ACTIVE; 5682 } else { 5683 ALOGV("active record track state %d", recordTrack->mState); 5684 } 5685 return status; 5686 } 5687 5688 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5689 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5690 // or using a separate command thread 5691 recordTrack->mState = TrackBase::STARTING_1; 5692 mActiveTracks.add(recordTrack); 5693 mActiveTracksGen++; 5694 status_t status = NO_ERROR; 5695 if (recordTrack->isExternalTrack()) { 5696 mLock.unlock(); 5697 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5698 mLock.lock(); 5699 // FIXME should verify that recordTrack is still in mActiveTracks 5700 if (status != NO_ERROR) { 5701 mActiveTracks.remove(recordTrack); 5702 mActiveTracksGen++; 5703 recordTrack->clearSyncStartEvent(); 5704 ALOGV("RecordThread::start error %d", status); 5705 return status; 5706 } 5707 } 5708 // Catch up with current buffer indices if thread is already running. 5709 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5710 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5711 // see previously buffered data before it called start(), but with greater risk of overrun. 5712 5713 recordTrack->mRsmpInFront = mRsmpInRear; 5714 recordTrack->mRsmpInUnrel = 0; 5715 // FIXME why reset? 5716 if (recordTrack->mResampler != NULL) { 5717 recordTrack->mResampler->reset(); 5718 } 5719 recordTrack->mState = TrackBase::STARTING_2; 5720 // signal thread to start 5721 mWaitWorkCV.broadcast(); 5722 if (mActiveTracks.indexOf(recordTrack) < 0) { 5723 ALOGV("Record failed to start"); 5724 status = BAD_VALUE; 5725 goto startError; 5726 } 5727 return status; 5728 } 5729 5730startError: 5731 if (recordTrack->isExternalTrack()) { 5732 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5733 } 5734 recordTrack->clearSyncStartEvent(); 5735 // FIXME I wonder why we do not reset the state here? 5736 return status; 5737} 5738 5739void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5740{ 5741 sp<SyncEvent> strongEvent = event.promote(); 5742 5743 if (strongEvent != 0) { 5744 sp<RefBase> ptr = strongEvent->cookie().promote(); 5745 if (ptr != 0) { 5746 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5747 recordTrack->handleSyncStartEvent(strongEvent); 5748 } 5749 } 5750} 5751 5752bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5753 ALOGV("RecordThread::stop"); 5754 AutoMutex _l(mLock); 5755 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5756 return false; 5757 } 5758 // note that threadLoop may still be processing the track at this point [without lock] 5759 recordTrack->mState = TrackBase::PAUSING; 5760 // do not wait for mStartStopCond if exiting 5761 if (exitPending()) { 5762 return true; 5763 } 5764 // FIXME incorrect usage of wait: no explicit predicate or loop 5765 mStartStopCond.wait(mLock); 5766 // if we have been restarted, recordTrack is in mActiveTracks here 5767 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5768 ALOGV("Record stopped OK"); 5769 return true; 5770 } 5771 return false; 5772} 5773 5774bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5775{ 5776 return false; 5777} 5778 5779status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5780{ 5781#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5782 if (!isValidSyncEvent(event)) { 5783 return BAD_VALUE; 5784 } 5785 5786 int eventSession = event->triggerSession(); 5787 status_t ret = NAME_NOT_FOUND; 5788 5789 Mutex::Autolock _l(mLock); 5790 5791 for (size_t i = 0; i < mTracks.size(); i++) { 5792 sp<RecordTrack> track = mTracks[i]; 5793 if (eventSession == track->sessionId()) { 5794 (void) track->setSyncEvent(event); 5795 ret = NO_ERROR; 5796 } 5797 } 5798 return ret; 5799#else 5800 return BAD_VALUE; 5801#endif 5802} 5803 5804// destroyTrack_l() must be called with ThreadBase::mLock held 5805void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5806{ 5807 track->terminate(); 5808 track->mState = TrackBase::STOPPED; 5809 // active tracks are removed by threadLoop() 5810 if (mActiveTracks.indexOf(track) < 0) { 5811 removeTrack_l(track); 5812 } 5813} 5814 5815void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5816{ 5817 mTracks.remove(track); 5818 // need anything related to effects here? 5819 if (track->isFastTrack()) { 5820 ALOG_ASSERT(!mFastTrackAvail); 5821 mFastTrackAvail = true; 5822 } 5823} 5824 5825void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5826{ 5827 dumpInternals(fd, args); 5828 dumpTracks(fd, args); 5829 dumpEffectChains(fd, args); 5830} 5831 5832void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5833{ 5834 dprintf(fd, "\nInput thread %p:\n", this); 5835 5836 if (mActiveTracks.size() > 0) { 5837 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5838 } else { 5839 dprintf(fd, " No active record clients\n"); 5840 } 5841 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5842 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5843 5844 dumpBase(fd, args); 5845} 5846 5847void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5848{ 5849 const size_t SIZE = 256; 5850 char buffer[SIZE]; 5851 String8 result; 5852 5853 size_t numtracks = mTracks.size(); 5854 size_t numactive = mActiveTracks.size(); 5855 size_t numactiveseen = 0; 5856 dprintf(fd, " %d Tracks", numtracks); 5857 if (numtracks) { 5858 dprintf(fd, " of which %d are active\n", numactive); 5859 RecordTrack::appendDumpHeader(result); 5860 for (size_t i = 0; i < numtracks ; ++i) { 5861 sp<RecordTrack> track = mTracks[i]; 5862 if (track != 0) { 5863 bool active = mActiveTracks.indexOf(track) >= 0; 5864 if (active) { 5865 numactiveseen++; 5866 } 5867 track->dump(buffer, SIZE, active); 5868 result.append(buffer); 5869 } 5870 } 5871 } else { 5872 dprintf(fd, "\n"); 5873 } 5874 5875 if (numactiveseen != numactive) { 5876 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5877 " not in the track list\n"); 5878 result.append(buffer); 5879 RecordTrack::appendDumpHeader(result); 5880 for (size_t i = 0; i < numactive; ++i) { 5881 sp<RecordTrack> track = mActiveTracks[i]; 5882 if (mTracks.indexOf(track) < 0) { 5883 track->dump(buffer, SIZE, true); 5884 result.append(buffer); 5885 } 5886 } 5887 5888 } 5889 write(fd, result.string(), result.size()); 5890} 5891 5892// AudioBufferProvider interface 5893status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5894 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5895{ 5896 RecordTrack *activeTrack = mRecordTrack; 5897 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5898 if (threadBase == 0) { 5899 buffer->frameCount = 0; 5900 buffer->raw = NULL; 5901 return NOT_ENOUGH_DATA; 5902 } 5903 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5904 int32_t rear = recordThread->mRsmpInRear; 5905 int32_t front = activeTrack->mRsmpInFront; 5906 ssize_t filled = rear - front; 5907 // FIXME should not be P2 (don't want to increase latency) 5908 // FIXME if client not keeping up, discard 5909 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5910 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5911 front &= recordThread->mRsmpInFramesP2 - 1; 5912 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5913 if (part1 > (size_t) filled) { 5914 part1 = filled; 5915 } 5916 size_t ask = buffer->frameCount; 5917 ALOG_ASSERT(ask > 0); 5918 if (part1 > ask) { 5919 part1 = ask; 5920 } 5921 if (part1 == 0) { 5922 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5923 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5924 buffer->raw = NULL; 5925 buffer->frameCount = 0; 5926 activeTrack->mRsmpInUnrel = 0; 5927 return NOT_ENOUGH_DATA; 5928 } 5929 5930 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5931 buffer->frameCount = part1; 5932 activeTrack->mRsmpInUnrel = part1; 5933 return NO_ERROR; 5934} 5935 5936// AudioBufferProvider interface 5937void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5938 AudioBufferProvider::Buffer* buffer) 5939{ 5940 RecordTrack *activeTrack = mRecordTrack; 5941 size_t stepCount = buffer->frameCount; 5942 if (stepCount == 0) { 5943 return; 5944 } 5945 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5946 activeTrack->mRsmpInUnrel -= stepCount; 5947 activeTrack->mRsmpInFront += stepCount; 5948 buffer->raw = NULL; 5949 buffer->frameCount = 0; 5950} 5951 5952bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5953 status_t& status) 5954{ 5955 bool reconfig = false; 5956 5957 status = NO_ERROR; 5958 5959 audio_format_t reqFormat = mFormat; 5960 uint32_t samplingRate = mSampleRate; 5961 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5962 5963 AudioParameter param = AudioParameter(keyValuePair); 5964 int value; 5965 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5966 // channel count change can be requested. Do we mandate the first client defines the 5967 // HAL sampling rate and channel count or do we allow changes on the fly? 5968 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5969 samplingRate = value; 5970 reconfig = true; 5971 } 5972 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5973 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5974 status = BAD_VALUE; 5975 } else { 5976 reqFormat = (audio_format_t) value; 5977 reconfig = true; 5978 } 5979 } 5980 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5981 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5982 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5983 status = BAD_VALUE; 5984 } else { 5985 channelMask = mask; 5986 reconfig = true; 5987 } 5988 } 5989 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5990 // do not accept frame count changes if tracks are open as the track buffer 5991 // size depends on frame count and correct behavior would not be guaranteed 5992 // if frame count is changed after track creation 5993 if (mActiveTracks.size() > 0) { 5994 status = INVALID_OPERATION; 5995 } else { 5996 reconfig = true; 5997 } 5998 } 5999 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6000 // forward device change to effects that have requested to be 6001 // aware of attached audio device. 6002 for (size_t i = 0; i < mEffectChains.size(); i++) { 6003 mEffectChains[i]->setDevice_l(value); 6004 } 6005 6006 // store input device and output device but do not forward output device to audio HAL. 6007 // Note that status is ignored by the caller for output device 6008 // (see AudioFlinger::setParameters() 6009 if (audio_is_output_devices(value)) { 6010 mOutDevice = value; 6011 status = BAD_VALUE; 6012 } else { 6013 mInDevice = value; 6014 // disable AEC and NS if the device is a BT SCO headset supporting those 6015 // pre processings 6016 if (mTracks.size() > 0) { 6017 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6018 mAudioFlinger->btNrecIsOff(); 6019 for (size_t i = 0; i < mTracks.size(); i++) { 6020 sp<RecordTrack> track = mTracks[i]; 6021 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6022 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6023 } 6024 } 6025 } 6026 } 6027 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6028 mAudioSource != (audio_source_t)value) { 6029 // forward device change to effects that have requested to be 6030 // aware of attached audio device. 6031 for (size_t i = 0; i < mEffectChains.size(); i++) { 6032 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6033 } 6034 mAudioSource = (audio_source_t)value; 6035 } 6036 6037 if (status == NO_ERROR) { 6038 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6039 keyValuePair.string()); 6040 if (status == INVALID_OPERATION) { 6041 inputStandBy(); 6042 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6043 keyValuePair.string()); 6044 } 6045 if (reconfig) { 6046 if (status == BAD_VALUE && 6047 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6048 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6049 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6050 <= (2 * samplingRate)) && 6051 audio_channel_count_from_in_mask( 6052 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6053 (channelMask == AUDIO_CHANNEL_IN_MONO || 6054 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6055 status = NO_ERROR; 6056 } 6057 if (status == NO_ERROR) { 6058 readInputParameters_l(); 6059 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6060 } 6061 } 6062 } 6063 6064 return reconfig; 6065} 6066 6067String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6068{ 6069 Mutex::Autolock _l(mLock); 6070 if (initCheck() != NO_ERROR) { 6071 return String8(); 6072 } 6073 6074 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6075 const String8 out_s8(s); 6076 free(s); 6077 return out_s8; 6078} 6079 6080void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6081 AudioSystem::OutputDescriptor desc; 6082 const void *param2 = NULL; 6083 6084 switch (event) { 6085 case AudioSystem::INPUT_OPENED: 6086 case AudioSystem::INPUT_CONFIG_CHANGED: 6087 desc.channelMask = mChannelMask; 6088 desc.samplingRate = mSampleRate; 6089 desc.format = mFormat; 6090 desc.frameCount = mFrameCount; 6091 desc.latency = 0; 6092 param2 = &desc; 6093 break; 6094 6095 case AudioSystem::INPUT_CLOSED: 6096 default: 6097 break; 6098 } 6099 mAudioFlinger->audioConfigChanged(event, mId, param2); 6100} 6101 6102void AudioFlinger::RecordThread::readInputParameters_l() 6103{ 6104 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6105 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6106 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6107 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6108 mFormat = mHALFormat; 6109 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6110 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6111 } 6112 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6113 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6114 mFrameCount = mBufferSize / mFrameSize; 6115 // This is the formula for calculating the temporary buffer size. 6116 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6117 // 1 full output buffer, regardless of the alignment of the available input. 6118 // The value is somewhat arbitrary, and could probably be even larger. 6119 // A larger value should allow more old data to be read after a track calls start(), 6120 // without increasing latency. 6121 mRsmpInFrames = mFrameCount * 7; 6122 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6123 delete[] mRsmpInBuffer; 6124 6125 // TODO optimize audio capture buffer sizes ... 6126 // Here we calculate the size of the sliding buffer used as a source 6127 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6128 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6129 // be better to have it derived from the pipe depth in the long term. 6130 // The current value is higher than necessary. However it should not add to latency. 6131 6132 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6133 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6134 6135 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6136 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6137} 6138 6139uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6140{ 6141 Mutex::Autolock _l(mLock); 6142 if (initCheck() != NO_ERROR) { 6143 return 0; 6144 } 6145 6146 return mInput->stream->get_input_frames_lost(mInput->stream); 6147} 6148 6149uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6150{ 6151 Mutex::Autolock _l(mLock); 6152 uint32_t result = 0; 6153 if (getEffectChain_l(sessionId) != 0) { 6154 result = EFFECT_SESSION; 6155 } 6156 6157 for (size_t i = 0; i < mTracks.size(); ++i) { 6158 if (sessionId == mTracks[i]->sessionId()) { 6159 result |= TRACK_SESSION; 6160 break; 6161 } 6162 } 6163 6164 return result; 6165} 6166 6167KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6168{ 6169 KeyedVector<int, bool> ids; 6170 Mutex::Autolock _l(mLock); 6171 for (size_t j = 0; j < mTracks.size(); ++j) { 6172 sp<RecordThread::RecordTrack> track = mTracks[j]; 6173 int sessionId = track->sessionId(); 6174 if (ids.indexOfKey(sessionId) < 0) { 6175 ids.add(sessionId, true); 6176 } 6177 } 6178 return ids; 6179} 6180 6181AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6182{ 6183 Mutex::Autolock _l(mLock); 6184 AudioStreamIn *input = mInput; 6185 mInput = NULL; 6186 return input; 6187} 6188 6189// this method must always be called either with ThreadBase mLock held or inside the thread loop 6190audio_stream_t* AudioFlinger::RecordThread::stream() const 6191{ 6192 if (mInput == NULL) { 6193 return NULL; 6194 } 6195 return &mInput->stream->common; 6196} 6197 6198status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6199{ 6200 // only one chain per input thread 6201 if (mEffectChains.size() != 0) { 6202 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6203 return INVALID_OPERATION; 6204 } 6205 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6206 chain->setThread(this); 6207 chain->setInBuffer(NULL); 6208 chain->setOutBuffer(NULL); 6209 6210 checkSuspendOnAddEffectChain_l(chain); 6211 6212 // make sure enabled pre processing effects state is communicated to the HAL as we 6213 // just moved them to a new input stream. 6214 chain->syncHalEffectsState(); 6215 6216 mEffectChains.add(chain); 6217 6218 return NO_ERROR; 6219} 6220 6221size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6222{ 6223 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6224 ALOGW_IF(mEffectChains.size() != 1, 6225 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6226 chain.get(), mEffectChains.size(), this); 6227 if (mEffectChains.size() == 1) { 6228 mEffectChains.removeAt(0); 6229 } 6230 return 0; 6231} 6232 6233status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6234 audio_patch_handle_t *handle) 6235{ 6236 status_t status = NO_ERROR; 6237 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6238 // store new device and send to effects 6239 mInDevice = patch->sources[0].ext.device.type; 6240 for (size_t i = 0; i < mEffectChains.size(); i++) { 6241 mEffectChains[i]->setDevice_l(mInDevice); 6242 } 6243 6244 // disable AEC and NS if the device is a BT SCO headset supporting those 6245 // pre processings 6246 if (mTracks.size() > 0) { 6247 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6248 mAudioFlinger->btNrecIsOff(); 6249 for (size_t i = 0; i < mTracks.size(); i++) { 6250 sp<RecordTrack> track = mTracks[i]; 6251 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6252 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6253 } 6254 } 6255 6256 // store new source and send to effects 6257 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6258 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6259 for (size_t i = 0; i < mEffectChains.size(); i++) { 6260 mEffectChains[i]->setAudioSource_l(mAudioSource); 6261 } 6262 } 6263 6264 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6265 status = hwDevice->create_audio_patch(hwDevice, 6266 patch->num_sources, 6267 patch->sources, 6268 patch->num_sinks, 6269 patch->sinks, 6270 handle); 6271 } else { 6272 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6273 } 6274 return status; 6275} 6276 6277status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6278{ 6279 status_t status = NO_ERROR; 6280 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6281 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6282 status = hwDevice->release_audio_patch(hwDevice, handle); 6283 } else { 6284 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6285 } 6286 return status; 6287} 6288 6289void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6290{ 6291 Mutex::Autolock _l(mLock); 6292 mTracks.add(record); 6293} 6294 6295void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6296{ 6297 Mutex::Autolock _l(mLock); 6298 destroyTrack_l(record); 6299} 6300 6301void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6302{ 6303 ThreadBase::getAudioPortConfig(config); 6304 config->role = AUDIO_PORT_ROLE_SINK; 6305 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6306 config->ext.mix.usecase.source = mAudioSource; 6307} 6308 6309}; // namespace android 6310