Threads.cpp revision 2b217bb3aee87ce8486014f261c0f498f6209e80
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 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1228 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1229 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1230 stream = (audio_stream_type_t) (stream + 1)) { 1231 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1232 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1233 } 1234 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1235 // because mAudioFlinger doesn't have one to copy from 1236} 1237 1238AudioFlinger::PlaybackThread::~PlaybackThread() 1239{ 1240 mAudioFlinger->unregisterWriter(mNBLogWriter); 1241 free(mSinkBuffer); 1242 free(mMixerBuffer); 1243 free(mEffectBuffer); 1244} 1245 1246void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1247{ 1248 dumpInternals(fd, args); 1249 dumpTracks(fd, args); 1250 dumpEffectChains(fd, args); 1251} 1252 1253void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1254{ 1255 const size_t SIZE = 256; 1256 char buffer[SIZE]; 1257 String8 result; 1258 1259 result.appendFormat(" Stream volumes in dB: "); 1260 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1261 const stream_type_t *st = &mStreamTypes[i]; 1262 if (i > 0) { 1263 result.appendFormat(", "); 1264 } 1265 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1266 if (st->mute) { 1267 result.append("M"); 1268 } 1269 } 1270 result.append("\n"); 1271 write(fd, result.string(), result.length()); 1272 result.clear(); 1273 1274 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1275 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1276 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1277 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1278 1279 size_t numtracks = mTracks.size(); 1280 size_t numactive = mActiveTracks.size(); 1281 dprintf(fd, " %d Tracks", numtracks); 1282 size_t numactiveseen = 0; 1283 if (numtracks) { 1284 dprintf(fd, " of which %d are active\n", numactive); 1285 Track::appendDumpHeader(result); 1286 for (size_t i = 0; i < numtracks; ++i) { 1287 sp<Track> track = mTracks[i]; 1288 if (track != 0) { 1289 bool active = mActiveTracks.indexOf(track) >= 0; 1290 if (active) { 1291 numactiveseen++; 1292 } 1293 track->dump(buffer, SIZE, active); 1294 result.append(buffer); 1295 } 1296 } 1297 } else { 1298 result.append("\n"); 1299 } 1300 if (numactiveseen != numactive) { 1301 // some tracks in the active list were not in the tracks list 1302 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1303 " not in the track list\n"); 1304 result.append(buffer); 1305 Track::appendDumpHeader(result); 1306 for (size_t i = 0; i < numactive; ++i) { 1307 sp<Track> track = mActiveTracks[i].promote(); 1308 if (track != 0 && mTracks.indexOf(track) < 0) { 1309 track->dump(buffer, SIZE, true); 1310 result.append(buffer); 1311 } 1312 } 1313 } 1314 1315 write(fd, result.string(), result.size()); 1316} 1317 1318void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1319{ 1320 dprintf(fd, "\nOutput thread %p:\n", this); 1321 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1322 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1323 dprintf(fd, " Total writes: %d\n", mNumWrites); 1324 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1325 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1326 dprintf(fd, " Suspend count: %d\n", mSuspended); 1327 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1328 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1329 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1330 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1331 1332 dumpBase(fd, args); 1333} 1334 1335// Thread virtuals 1336 1337void AudioFlinger::PlaybackThread::onFirstRef() 1338{ 1339 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1340} 1341 1342// ThreadBase virtuals 1343void AudioFlinger::PlaybackThread::preExit() 1344{ 1345 ALOGV(" preExit()"); 1346 // FIXME this is using hard-coded strings but in the future, this functionality will be 1347 // converted to use audio HAL extensions required to support tunneling 1348 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1349} 1350 1351// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1352sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1353 const sp<AudioFlinger::Client>& client, 1354 audio_stream_type_t streamType, 1355 uint32_t sampleRate, 1356 audio_format_t format, 1357 audio_channel_mask_t channelMask, 1358 size_t *pFrameCount, 1359 const sp<IMemory>& sharedBuffer, 1360 int sessionId, 1361 IAudioFlinger::track_flags_t *flags, 1362 pid_t tid, 1363 int uid, 1364 status_t *status) 1365{ 1366 size_t frameCount = *pFrameCount; 1367 sp<Track> track; 1368 status_t lStatus; 1369 1370 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1371 1372 // client expresses a preference for FAST, but we get the final say 1373 if (*flags & IAudioFlinger::TRACK_FAST) { 1374 if ( 1375 // not timed 1376 (!isTimed) && 1377 // either of these use cases: 1378 ( 1379 // use case 1: shared buffer with any frame count 1380 ( 1381 (sharedBuffer != 0) 1382 ) || 1383 // use case 2: callback handler and frame count is default or at least as large as HAL 1384 ( 1385 (tid != -1) && 1386 ((frameCount == 0) || 1387 (frameCount >= mFrameCount)) 1388 ) 1389 ) && 1390 // PCM data 1391 audio_is_linear_pcm(format) && 1392 // identical channel mask to sink, or mono in and stereo sink 1393 (channelMask == mChannelMask || 1394 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1395 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1396 // hardware sample rate 1397 (sampleRate == mSampleRate) && 1398 // normal mixer has an associated fast mixer 1399 hasFastMixer() && 1400 // there are sufficient fast track slots available 1401 (mFastTrackAvailMask != 0) 1402 // FIXME test that MixerThread for this fast track has a capable output HAL 1403 // FIXME add a permission test also? 1404 ) { 1405 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1406 if (frameCount == 0) { 1407 // read the fast track multiplier property the first time it is needed 1408 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1409 if (ok != 0) { 1410 ALOGE("%s pthread_once failed: %d", __func__, ok); 1411 } 1412 frameCount = mFrameCount * sFastTrackMultiplier; 1413 } 1414 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1415 frameCount, mFrameCount); 1416 } else { 1417 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1418 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1419 "sampleRate=%u mSampleRate=%u " 1420 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1421 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1422 audio_is_linear_pcm(format), 1423 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1424 *flags &= ~IAudioFlinger::TRACK_FAST; 1425 // For compatibility with AudioTrack calculation, buffer depth is forced 1426 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1427 // This is probably too conservative, but legacy application code may depend on it. 1428 // If you change this calculation, also review the start threshold which is related. 1429 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1430 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1431 if (minBufCount < 2) { 1432 minBufCount = 2; 1433 } 1434 size_t minFrameCount = mNormalFrameCount * minBufCount; 1435 if (frameCount < minFrameCount) { 1436 frameCount = minFrameCount; 1437 } 1438 } 1439 } 1440 *pFrameCount = frameCount; 1441 1442 switch (mType) { 1443 1444 case DIRECT: 1445 if (audio_is_linear_pcm(format)) { 1446 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1447 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1448 "for output %p with format %#x", 1449 sampleRate, format, channelMask, mOutput, mFormat); 1450 lStatus = BAD_VALUE; 1451 goto Exit; 1452 } 1453 } 1454 break; 1455 1456 case OFFLOAD: 1457 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1458 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1459 "for output %p with format %#x", 1460 sampleRate, format, channelMask, mOutput, mFormat); 1461 lStatus = BAD_VALUE; 1462 goto Exit; 1463 } 1464 break; 1465 1466 default: 1467 if (!audio_is_linear_pcm(format)) { 1468 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1469 "for output %p with format %#x", 1470 format, mOutput, mFormat); 1471 lStatus = BAD_VALUE; 1472 goto Exit; 1473 } 1474 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1475 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1476 lStatus = BAD_VALUE; 1477 goto Exit; 1478 } 1479 break; 1480 1481 } 1482 1483 lStatus = initCheck(); 1484 if (lStatus != NO_ERROR) { 1485 ALOGE("createTrack_l() audio driver not initialized"); 1486 goto Exit; 1487 } 1488 1489 { // scope for mLock 1490 Mutex::Autolock _l(mLock); 1491 1492 // all tracks in same audio session must share the same routing strategy otherwise 1493 // conflicts will happen when tracks are moved from one output to another by audio policy 1494 // manager 1495 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1496 for (size_t i = 0; i < mTracks.size(); ++i) { 1497 sp<Track> t = mTracks[i]; 1498 if (t != 0 && t->isExternalTrack()) { 1499 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1500 if (sessionId == t->sessionId() && strategy != actual) { 1501 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1502 strategy, actual); 1503 lStatus = BAD_VALUE; 1504 goto Exit; 1505 } 1506 } 1507 } 1508 1509 if (!isTimed) { 1510 track = new Track(this, client, streamType, sampleRate, format, 1511 channelMask, frameCount, NULL, sharedBuffer, 1512 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1513 } else { 1514 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1515 channelMask, frameCount, sharedBuffer, sessionId, uid); 1516 } 1517 1518 // new Track always returns non-NULL, 1519 // but TimedTrack::create() is a factory that could fail by returning NULL 1520 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1521 if (lStatus != NO_ERROR) { 1522 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1523 // track must be cleared from the caller as the caller has the AF lock 1524 goto Exit; 1525 } 1526 mTracks.add(track); 1527 1528 sp<EffectChain> chain = getEffectChain_l(sessionId); 1529 if (chain != 0) { 1530 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1531 track->setMainBuffer(chain->inBuffer()); 1532 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1533 chain->incTrackCnt(); 1534 } 1535 1536 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1537 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1538 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1539 // so ask activity manager to do this on our behalf 1540 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1541 } 1542 } 1543 1544 lStatus = NO_ERROR; 1545 1546Exit: 1547 *status = lStatus; 1548 return track; 1549} 1550 1551uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1552{ 1553 return latency; 1554} 1555 1556uint32_t AudioFlinger::PlaybackThread::latency() const 1557{ 1558 Mutex::Autolock _l(mLock); 1559 return latency_l(); 1560} 1561uint32_t AudioFlinger::PlaybackThread::latency_l() const 1562{ 1563 if (initCheck() == NO_ERROR) { 1564 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1565 } else { 1566 return 0; 1567 } 1568} 1569 1570void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1571{ 1572 Mutex::Autolock _l(mLock); 1573 // Don't apply master volume in SW if our HAL can do it for us. 1574 if (mOutput && mOutput->audioHwDev && 1575 mOutput->audioHwDev->canSetMasterVolume()) { 1576 mMasterVolume = 1.0; 1577 } else { 1578 mMasterVolume = value; 1579 } 1580} 1581 1582void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1583{ 1584 Mutex::Autolock _l(mLock); 1585 // Don't apply master mute in SW if our HAL can do it for us. 1586 if (mOutput && mOutput->audioHwDev && 1587 mOutput->audioHwDev->canSetMasterMute()) { 1588 mMasterMute = false; 1589 } else { 1590 mMasterMute = muted; 1591 } 1592} 1593 1594void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1595{ 1596 Mutex::Autolock _l(mLock); 1597 mStreamTypes[stream].volume = value; 1598 broadcast_l(); 1599} 1600 1601void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1602{ 1603 Mutex::Autolock _l(mLock); 1604 mStreamTypes[stream].mute = muted; 1605 broadcast_l(); 1606} 1607 1608float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1609{ 1610 Mutex::Autolock _l(mLock); 1611 return mStreamTypes[stream].volume; 1612} 1613 1614// addTrack_l() must be called with ThreadBase::mLock held 1615status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1616{ 1617 status_t status = ALREADY_EXISTS; 1618 1619 // set retry count for buffer fill 1620 track->mRetryCount = kMaxTrackStartupRetries; 1621 if (mActiveTracks.indexOf(track) < 0) { 1622 // the track is newly added, make sure it fills up all its 1623 // buffers before playing. This is to ensure the client will 1624 // effectively get the latency it requested. 1625 if (track->isExternalTrack()) { 1626 TrackBase::track_state state = track->mState; 1627 mLock.unlock(); 1628 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1629 mLock.lock(); 1630 // abort track was stopped/paused while we released the lock 1631 if (state != track->mState) { 1632 if (status == NO_ERROR) { 1633 mLock.unlock(); 1634 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1635 mLock.lock(); 1636 } 1637 return INVALID_OPERATION; 1638 } 1639 // abort if start is rejected by audio policy manager 1640 if (status != NO_ERROR) { 1641 return PERMISSION_DENIED; 1642 } 1643#ifdef ADD_BATTERY_DATA 1644 // to track the speaker usage 1645 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1646#endif 1647 } 1648 1649 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1650 track->mResetDone = false; 1651 track->mPresentationCompleteFrames = 0; 1652 mActiveTracks.add(track); 1653 mWakeLockUids.add(track->uid()); 1654 mActiveTracksGeneration++; 1655 mLatestActiveTrack = track; 1656 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1657 if (chain != 0) { 1658 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1659 track->sessionId()); 1660 chain->incActiveTrackCnt(); 1661 } 1662 1663 status = NO_ERROR; 1664 } 1665 1666 onAddNewTrack_l(); 1667 return status; 1668} 1669 1670bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1671{ 1672 track->terminate(); 1673 // active tracks are removed by threadLoop() 1674 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1675 track->mState = TrackBase::STOPPED; 1676 if (!trackActive) { 1677 removeTrack_l(track); 1678 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1679 track->mState = TrackBase::STOPPING_1; 1680 } 1681 1682 return trackActive; 1683} 1684 1685void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1686{ 1687 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1688 mTracks.remove(track); 1689 deleteTrackName_l(track->name()); 1690 // redundant as track is about to be destroyed, for dumpsys only 1691 track->mName = -1; 1692 if (track->isFastTrack()) { 1693 int index = track->mFastIndex; 1694 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1695 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1696 mFastTrackAvailMask |= 1 << index; 1697 // redundant as track is about to be destroyed, for dumpsys only 1698 track->mFastIndex = -1; 1699 } 1700 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1701 if (chain != 0) { 1702 chain->decTrackCnt(); 1703 } 1704} 1705 1706void AudioFlinger::PlaybackThread::broadcast_l() 1707{ 1708 // Thread could be blocked waiting for async 1709 // so signal it to handle state changes immediately 1710 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1711 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1712 mSignalPending = true; 1713 mWaitWorkCV.broadcast(); 1714} 1715 1716String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1717{ 1718 Mutex::Autolock _l(mLock); 1719 if (initCheck() != NO_ERROR) { 1720 return String8(); 1721 } 1722 1723 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1724 const String8 out_s8(s); 1725 free(s); 1726 return out_s8; 1727} 1728 1729void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1730 AudioSystem::OutputDescriptor desc; 1731 void *param2 = NULL; 1732 1733 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1734 param); 1735 1736 switch (event) { 1737 case AudioSystem::OUTPUT_OPENED: 1738 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1739 desc.channelMask = mChannelMask; 1740 desc.samplingRate = mSampleRate; 1741 desc.format = mFormat; 1742 desc.frameCount = mNormalFrameCount; // FIXME see 1743 // AudioFlinger::frameCount(audio_io_handle_t) 1744 desc.latency = latency_l(); 1745 param2 = &desc; 1746 break; 1747 1748 case AudioSystem::STREAM_CONFIG_CHANGED: 1749 param2 = ¶m; 1750 case AudioSystem::OUTPUT_CLOSED: 1751 default: 1752 break; 1753 } 1754 mAudioFlinger->audioConfigChanged(event, mId, param2); 1755} 1756 1757void AudioFlinger::PlaybackThread::writeCallback() 1758{ 1759 ALOG_ASSERT(mCallbackThread != 0); 1760 mCallbackThread->resetWriteBlocked(); 1761} 1762 1763void AudioFlinger::PlaybackThread::drainCallback() 1764{ 1765 ALOG_ASSERT(mCallbackThread != 0); 1766 mCallbackThread->resetDraining(); 1767} 1768 1769void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1770{ 1771 Mutex::Autolock _l(mLock); 1772 // reject out of sequence requests 1773 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1774 mWriteAckSequence &= ~1; 1775 mWaitWorkCV.signal(); 1776 } 1777} 1778 1779void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1780{ 1781 Mutex::Autolock _l(mLock); 1782 // reject out of sequence requests 1783 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1784 mDrainSequence &= ~1; 1785 mWaitWorkCV.signal(); 1786 } 1787} 1788 1789// static 1790int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1791 void *param __unused, 1792 void *cookie) 1793{ 1794 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1795 ALOGV("asyncCallback() event %d", event); 1796 switch (event) { 1797 case STREAM_CBK_EVENT_WRITE_READY: 1798 me->writeCallback(); 1799 break; 1800 case STREAM_CBK_EVENT_DRAIN_READY: 1801 me->drainCallback(); 1802 break; 1803 default: 1804 ALOGW("asyncCallback() unknown event %d", event); 1805 break; 1806 } 1807 return 0; 1808} 1809 1810void AudioFlinger::PlaybackThread::readOutputParameters_l() 1811{ 1812 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1813 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1814 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1815 if (!audio_is_output_channel(mChannelMask)) { 1816 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1817 } 1818 if ((mType == MIXER || mType == DUPLICATING) 1819 && !isValidPcmSinkChannelMask(mChannelMask)) { 1820 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1821 mChannelMask); 1822 } 1823 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1824 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1825 mFormat = mHALFormat; 1826 if (!audio_is_valid_format(mFormat)) { 1827 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1828 } 1829 if ((mType == MIXER || mType == DUPLICATING) 1830 && !isValidPcmSinkFormat(mFormat)) { 1831 LOG_FATAL("HAL format %#x not supported for mixed output", 1832 mFormat); 1833 } 1834 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1835 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1836 mFrameCount = mBufferSize / mFrameSize; 1837 if (mFrameCount & 15) { 1838 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1839 mFrameCount); 1840 } 1841 1842 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1843 (mOutput->stream->set_callback != NULL)) { 1844 if (mOutput->stream->set_callback(mOutput->stream, 1845 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1846 mUseAsyncWrite = true; 1847 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1848 } 1849 } 1850 1851 // Calculate size of normal sink buffer relative to the HAL output buffer size 1852 double multiplier = 1.0; 1853 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1854 kUseFastMixer == FastMixer_Dynamic)) { 1855 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1856 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1857 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1858 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1859 maxNormalFrameCount = maxNormalFrameCount & ~15; 1860 if (maxNormalFrameCount < minNormalFrameCount) { 1861 maxNormalFrameCount = minNormalFrameCount; 1862 } 1863 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1864 if (multiplier <= 1.0) { 1865 multiplier = 1.0; 1866 } else if (multiplier <= 2.0) { 1867 if (2 * mFrameCount <= maxNormalFrameCount) { 1868 multiplier = 2.0; 1869 } else { 1870 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1871 } 1872 } else { 1873 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1874 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1875 // track, but we sometimes have to do this to satisfy the maximum frame count 1876 // constraint) 1877 // FIXME this rounding up should not be done if no HAL SRC 1878 uint32_t truncMult = (uint32_t) multiplier; 1879 if ((truncMult & 1)) { 1880 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1881 ++truncMult; 1882 } 1883 } 1884 multiplier = (double) truncMult; 1885 } 1886 } 1887 mNormalFrameCount = multiplier * mFrameCount; 1888 // round up to nearest 16 frames to satisfy AudioMixer 1889 if (mType == MIXER || mType == DUPLICATING) { 1890 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1891 } 1892 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1893 mNormalFrameCount); 1894 1895 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1896 // Originally this was int16_t[] array, need to remove legacy implications. 1897 free(mSinkBuffer); 1898 mSinkBuffer = NULL; 1899 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1900 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1901 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1902 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1903 1904 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1905 // drives the output. 1906 free(mMixerBuffer); 1907 mMixerBuffer = NULL; 1908 if (mMixerBufferEnabled) { 1909 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1910 mMixerBufferSize = mNormalFrameCount * mChannelCount 1911 * audio_bytes_per_sample(mMixerBufferFormat); 1912 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1913 } 1914 free(mEffectBuffer); 1915 mEffectBuffer = NULL; 1916 if (mEffectBufferEnabled) { 1917 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1918 mEffectBufferSize = mNormalFrameCount * mChannelCount 1919 * audio_bytes_per_sample(mEffectBufferFormat); 1920 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1921 } 1922 1923 // force reconfiguration of effect chains and engines to take new buffer size and audio 1924 // parameters into account 1925 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1926 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1927 // matter. 1928 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1929 Vector< sp<EffectChain> > effectChains = mEffectChains; 1930 for (size_t i = 0; i < effectChains.size(); i ++) { 1931 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1932 } 1933} 1934 1935 1936status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1937{ 1938 if (halFrames == NULL || dspFrames == NULL) { 1939 return BAD_VALUE; 1940 } 1941 Mutex::Autolock _l(mLock); 1942 if (initCheck() != NO_ERROR) { 1943 return INVALID_OPERATION; 1944 } 1945 size_t framesWritten = mBytesWritten / mFrameSize; 1946 *halFrames = framesWritten; 1947 1948 if (isSuspended()) { 1949 // return an estimation of rendered frames when the output is suspended 1950 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1951 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1952 return NO_ERROR; 1953 } else { 1954 status_t status; 1955 uint32_t frames; 1956 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1957 *dspFrames = (size_t)frames; 1958 return status; 1959 } 1960} 1961 1962uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1963{ 1964 Mutex::Autolock _l(mLock); 1965 uint32_t result = 0; 1966 if (getEffectChain_l(sessionId) != 0) { 1967 result = EFFECT_SESSION; 1968 } 1969 1970 for (size_t i = 0; i < mTracks.size(); ++i) { 1971 sp<Track> track = mTracks[i]; 1972 if (sessionId == track->sessionId() && !track->isInvalid()) { 1973 result |= TRACK_SESSION; 1974 break; 1975 } 1976 } 1977 1978 return result; 1979} 1980 1981uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1982{ 1983 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1984 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1985 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1986 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1987 } 1988 for (size_t i = 0; i < mTracks.size(); i++) { 1989 sp<Track> track = mTracks[i]; 1990 if (sessionId == track->sessionId() && !track->isInvalid()) { 1991 return AudioSystem::getStrategyForStream(track->streamType()); 1992 } 1993 } 1994 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1995} 1996 1997 1998AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1999{ 2000 Mutex::Autolock _l(mLock); 2001 return mOutput; 2002} 2003 2004AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2005{ 2006 Mutex::Autolock _l(mLock); 2007 AudioStreamOut *output = mOutput; 2008 mOutput = NULL; 2009 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2010 // must push a NULL and wait for ack 2011 mOutputSink.clear(); 2012 mPipeSink.clear(); 2013 mNormalSink.clear(); 2014 return output; 2015} 2016 2017// this method must always be called either with ThreadBase mLock held or inside the thread loop 2018audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2019{ 2020 if (mOutput == NULL) { 2021 return NULL; 2022 } 2023 return &mOutput->stream->common; 2024} 2025 2026uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2027{ 2028 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2029} 2030 2031status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2032{ 2033 if (!isValidSyncEvent(event)) { 2034 return BAD_VALUE; 2035 } 2036 2037 Mutex::Autolock _l(mLock); 2038 2039 for (size_t i = 0; i < mTracks.size(); ++i) { 2040 sp<Track> track = mTracks[i]; 2041 if (event->triggerSession() == track->sessionId()) { 2042 (void) track->setSyncEvent(event); 2043 return NO_ERROR; 2044 } 2045 } 2046 2047 return NAME_NOT_FOUND; 2048} 2049 2050bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2051{ 2052 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2053} 2054 2055void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2056 const Vector< sp<Track> >& tracksToRemove) 2057{ 2058 size_t count = tracksToRemove.size(); 2059 if (count > 0) { 2060 for (size_t i = 0 ; i < count ; i++) { 2061 const sp<Track>& track = tracksToRemove.itemAt(i); 2062 if (track->isExternalTrack()) { 2063 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2064#ifdef ADD_BATTERY_DATA 2065 // to track the speaker usage 2066 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2067#endif 2068 if (track->isTerminated()) { 2069 AudioSystem::releaseOutput(mId); 2070 } 2071 } 2072 } 2073 } 2074} 2075 2076void AudioFlinger::PlaybackThread::checkSilentMode_l() 2077{ 2078 if (!mMasterMute) { 2079 char value[PROPERTY_VALUE_MAX]; 2080 if (property_get("ro.audio.silent", value, "0") > 0) { 2081 char *endptr; 2082 unsigned long ul = strtoul(value, &endptr, 0); 2083 if (*endptr == '\0' && ul != 0) { 2084 ALOGD("Silence is golden"); 2085 // The setprop command will not allow a property to be changed after 2086 // the first time it is set, so we don't have to worry about un-muting. 2087 setMasterMute_l(true); 2088 } 2089 } 2090 } 2091} 2092 2093// shared by MIXER and DIRECT, overridden by DUPLICATING 2094ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2095{ 2096 // FIXME rewrite to reduce number of system calls 2097 mLastWriteTime = systemTime(); 2098 mInWrite = true; 2099 ssize_t bytesWritten; 2100 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2101 2102 // If an NBAIO sink is present, use it to write the normal mixer's submix 2103 if (mNormalSink != 0) { 2104 2105 const size_t count = mBytesRemaining / mFrameSize; 2106 2107 ATRACE_BEGIN("write"); 2108 // update the setpoint when AudioFlinger::mScreenState changes 2109 uint32_t screenState = AudioFlinger::mScreenState; 2110 if (screenState != mScreenState) { 2111 mScreenState = screenState; 2112 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2113 if (pipe != NULL) { 2114 pipe->setAvgFrames((mScreenState & 1) ? 2115 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2116 } 2117 } 2118 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2119 ATRACE_END(); 2120 if (framesWritten > 0) { 2121 bytesWritten = framesWritten * mFrameSize; 2122 } else { 2123 bytesWritten = framesWritten; 2124 } 2125 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2126 if (status == NO_ERROR) { 2127 size_t totalFramesWritten = mNormalSink->framesWritten(); 2128 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2129 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2130 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2131 mLatchDValid = true; 2132 } 2133 } 2134 // otherwise use the HAL / AudioStreamOut directly 2135 } else { 2136 // Direct output and offload threads 2137 2138 if (mUseAsyncWrite) { 2139 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2140 mWriteAckSequence += 2; 2141 mWriteAckSequence |= 1; 2142 ALOG_ASSERT(mCallbackThread != 0); 2143 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2144 } 2145 // FIXME We should have an implementation of timestamps for direct output threads. 2146 // They are used e.g for multichannel PCM playback over HDMI. 2147 bytesWritten = mOutput->stream->write(mOutput->stream, 2148 (char *)mSinkBuffer + offset, mBytesRemaining); 2149 if (mUseAsyncWrite && 2150 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2151 // do not wait for async callback in case of error of full write 2152 mWriteAckSequence &= ~1; 2153 ALOG_ASSERT(mCallbackThread != 0); 2154 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2155 } 2156 } 2157 2158 mNumWrites++; 2159 mInWrite = false; 2160 mStandby = false; 2161 return bytesWritten; 2162} 2163 2164void AudioFlinger::PlaybackThread::threadLoop_drain() 2165{ 2166 if (mOutput->stream->drain) { 2167 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2168 if (mUseAsyncWrite) { 2169 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2170 mDrainSequence |= 1; 2171 ALOG_ASSERT(mCallbackThread != 0); 2172 mCallbackThread->setDraining(mDrainSequence); 2173 } 2174 mOutput->stream->drain(mOutput->stream, 2175 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2176 : AUDIO_DRAIN_ALL); 2177 } 2178} 2179 2180void AudioFlinger::PlaybackThread::threadLoop_exit() 2181{ 2182 // Default implementation has nothing to do 2183} 2184 2185/* 2186The derived values that are cached: 2187 - mSinkBufferSize from frame count * frame size 2188 - activeSleepTime from activeSleepTimeUs() 2189 - idleSleepTime from idleSleepTimeUs() 2190 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2191 - maxPeriod from frame count and sample rate (MIXER only) 2192 2193The parameters that affect these derived values are: 2194 - frame count 2195 - frame size 2196 - sample rate 2197 - device type: A2DP or not 2198 - device latency 2199 - format: PCM or not 2200 - active sleep time 2201 - idle sleep time 2202*/ 2203 2204void AudioFlinger::PlaybackThread::cacheParameters_l() 2205{ 2206 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2207 activeSleepTime = activeSleepTimeUs(); 2208 idleSleepTime = idleSleepTimeUs(); 2209} 2210 2211void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2212{ 2213 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2214 this, streamType, mTracks.size()); 2215 Mutex::Autolock _l(mLock); 2216 2217 size_t size = mTracks.size(); 2218 for (size_t i = 0; i < size; i++) { 2219 sp<Track> t = mTracks[i]; 2220 if (t->streamType() == streamType) { 2221 t->invalidate(); 2222 } 2223 } 2224} 2225 2226status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2227{ 2228 int session = chain->sessionId(); 2229 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2230 ? mEffectBuffer : mSinkBuffer); 2231 bool ownsBuffer = false; 2232 2233 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2234 if (session > 0) { 2235 // Only one effect chain can be present in direct output thread and it uses 2236 // the sink buffer as input 2237 if (mType != DIRECT) { 2238 size_t numSamples = mNormalFrameCount * mChannelCount; 2239 buffer = new int16_t[numSamples]; 2240 memset(buffer, 0, numSamples * sizeof(int16_t)); 2241 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2242 ownsBuffer = true; 2243 } 2244 2245 // Attach all tracks with same session ID to this chain. 2246 for (size_t i = 0; i < mTracks.size(); ++i) { 2247 sp<Track> track = mTracks[i]; 2248 if (session == track->sessionId()) { 2249 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2250 buffer); 2251 track->setMainBuffer(buffer); 2252 chain->incTrackCnt(); 2253 } 2254 } 2255 2256 // indicate all active tracks in the chain 2257 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2258 sp<Track> track = mActiveTracks[i].promote(); 2259 if (track == 0) { 2260 continue; 2261 } 2262 if (session == track->sessionId()) { 2263 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2264 chain->incActiveTrackCnt(); 2265 } 2266 } 2267 } 2268 chain->setThread(this); 2269 chain->setInBuffer(buffer, ownsBuffer); 2270 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2271 ? mEffectBuffer : mSinkBuffer)); 2272 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2273 // chains list in order to be processed last as it contains output stage effects 2274 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2275 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2276 // after track specific effects and before output stage 2277 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2278 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2279 // Effect chain for other sessions are inserted at beginning of effect 2280 // chains list to be processed before output mix effects. Relative order between other 2281 // sessions is not important 2282 size_t size = mEffectChains.size(); 2283 size_t i = 0; 2284 for (i = 0; i < size; i++) { 2285 if (mEffectChains[i]->sessionId() < session) { 2286 break; 2287 } 2288 } 2289 mEffectChains.insertAt(chain, i); 2290 checkSuspendOnAddEffectChain_l(chain); 2291 2292 return NO_ERROR; 2293} 2294 2295size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2296{ 2297 int session = chain->sessionId(); 2298 2299 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2300 2301 for (size_t i = 0; i < mEffectChains.size(); i++) { 2302 if (chain == mEffectChains[i]) { 2303 mEffectChains.removeAt(i); 2304 // detach all active tracks from the chain 2305 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2306 sp<Track> track = mActiveTracks[i].promote(); 2307 if (track == 0) { 2308 continue; 2309 } 2310 if (session == track->sessionId()) { 2311 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2312 chain.get(), session); 2313 chain->decActiveTrackCnt(); 2314 } 2315 } 2316 2317 // detach all tracks with same session ID from this chain 2318 for (size_t i = 0; i < mTracks.size(); ++i) { 2319 sp<Track> track = mTracks[i]; 2320 if (session == track->sessionId()) { 2321 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2322 chain->decTrackCnt(); 2323 } 2324 } 2325 break; 2326 } 2327 } 2328 return mEffectChains.size(); 2329} 2330 2331status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2332 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2333{ 2334 Mutex::Autolock _l(mLock); 2335 return attachAuxEffect_l(track, EffectId); 2336} 2337 2338status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2339 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2340{ 2341 status_t status = NO_ERROR; 2342 2343 if (EffectId == 0) { 2344 track->setAuxBuffer(0, NULL); 2345 } else { 2346 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2347 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2348 if (effect != 0) { 2349 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2350 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2351 } else { 2352 status = INVALID_OPERATION; 2353 } 2354 } else { 2355 status = BAD_VALUE; 2356 } 2357 } 2358 return status; 2359} 2360 2361void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2362{ 2363 for (size_t i = 0; i < mTracks.size(); ++i) { 2364 sp<Track> track = mTracks[i]; 2365 if (track->auxEffectId() == effectId) { 2366 attachAuxEffect_l(track, 0); 2367 } 2368 } 2369} 2370 2371bool AudioFlinger::PlaybackThread::threadLoop() 2372{ 2373 Vector< sp<Track> > tracksToRemove; 2374 2375 standbyTime = systemTime(); 2376 2377 // MIXER 2378 nsecs_t lastWarning = 0; 2379 2380 // DUPLICATING 2381 // FIXME could this be made local to while loop? 2382 writeFrames = 0; 2383 2384 int lastGeneration = 0; 2385 2386 cacheParameters_l(); 2387 sleepTime = idleSleepTime; 2388 2389 if (mType == MIXER) { 2390 sleepTimeShift = 0; 2391 } 2392 2393 CpuStats cpuStats; 2394 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2395 2396 acquireWakeLock(); 2397 2398 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2399 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2400 // and then that string will be logged at the next convenient opportunity. 2401 const char *logString = NULL; 2402 2403 checkSilentMode_l(); 2404 2405 while (!exitPending()) 2406 { 2407 cpuStats.sample(myName); 2408 2409 Vector< sp<EffectChain> > effectChains; 2410 2411 { // scope for mLock 2412 2413 Mutex::Autolock _l(mLock); 2414 2415 processConfigEvents_l(); 2416 2417 if (logString != NULL) { 2418 mNBLogWriter->logTimestamp(); 2419 mNBLogWriter->log(logString); 2420 logString = NULL; 2421 } 2422 2423 // Gather the framesReleased counters for all active tracks, 2424 // and latch them atomically with the timestamp. 2425 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2426 mLatchD.mFramesReleased.clear(); 2427 size_t size = mActiveTracks.size(); 2428 for (size_t i = 0; i < size; i++) { 2429 sp<Track> t = mActiveTracks[i].promote(); 2430 if (t != 0) { 2431 mLatchD.mFramesReleased.add(t.get(), 2432 t->mAudioTrackServerProxy->framesReleased()); 2433 } 2434 } 2435 if (mLatchDValid) { 2436 mLatchQ = mLatchD; 2437 mLatchDValid = false; 2438 mLatchQValid = true; 2439 } 2440 2441 saveOutputTracks(); 2442 if (mSignalPending) { 2443 // A signal was raised while we were unlocked 2444 mSignalPending = false; 2445 } else if (waitingAsyncCallback_l()) { 2446 if (exitPending()) { 2447 break; 2448 } 2449 releaseWakeLock_l(); 2450 mWakeLockUids.clear(); 2451 mActiveTracksGeneration++; 2452 ALOGV("wait async completion"); 2453 mWaitWorkCV.wait(mLock); 2454 ALOGV("async completion/wake"); 2455 acquireWakeLock_l(); 2456 standbyTime = systemTime() + standbyDelay; 2457 sleepTime = 0; 2458 2459 continue; 2460 } 2461 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2462 isSuspended()) { 2463 // put audio hardware into standby after short delay 2464 if (shouldStandby_l()) { 2465 2466 threadLoop_standby(); 2467 2468 mStandby = true; 2469 } 2470 2471 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2472 // we're about to wait, flush the binder command buffer 2473 IPCThreadState::self()->flushCommands(); 2474 2475 clearOutputTracks(); 2476 2477 if (exitPending()) { 2478 break; 2479 } 2480 2481 releaseWakeLock_l(); 2482 mWakeLockUids.clear(); 2483 mActiveTracksGeneration++; 2484 // wait until we have something to do... 2485 ALOGV("%s going to sleep", myName.string()); 2486 mWaitWorkCV.wait(mLock); 2487 ALOGV("%s waking up", myName.string()); 2488 acquireWakeLock_l(); 2489 2490 mMixerStatus = MIXER_IDLE; 2491 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2492 mBytesWritten = 0; 2493 mBytesRemaining = 0; 2494 checkSilentMode_l(); 2495 2496 standbyTime = systemTime() + standbyDelay; 2497 sleepTime = idleSleepTime; 2498 if (mType == MIXER) { 2499 sleepTimeShift = 0; 2500 } 2501 2502 continue; 2503 } 2504 } 2505 // mMixerStatusIgnoringFastTracks is also updated internally 2506 mMixerStatus = prepareTracks_l(&tracksToRemove); 2507 2508 // compare with previously applied list 2509 if (lastGeneration != mActiveTracksGeneration) { 2510 // update wakelock 2511 updateWakeLockUids_l(mWakeLockUids); 2512 lastGeneration = mActiveTracksGeneration; 2513 } 2514 2515 // prevent any changes in effect chain list and in each effect chain 2516 // during mixing and effect process as the audio buffers could be deleted 2517 // or modified if an effect is created or deleted 2518 lockEffectChains_l(effectChains); 2519 } // mLock scope ends 2520 2521 if (mBytesRemaining == 0) { 2522 mCurrentWriteLength = 0; 2523 if (mMixerStatus == MIXER_TRACKS_READY) { 2524 // threadLoop_mix() sets mCurrentWriteLength 2525 threadLoop_mix(); 2526 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2527 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2528 // threadLoop_sleepTime sets sleepTime to 0 if data 2529 // must be written to HAL 2530 threadLoop_sleepTime(); 2531 if (sleepTime == 0) { 2532 mCurrentWriteLength = mSinkBufferSize; 2533 } 2534 } 2535 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2536 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2537 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2538 // or mSinkBuffer (if there are no effects). 2539 // 2540 // This is done pre-effects computation; if effects change to 2541 // support higher precision, this needs to move. 2542 // 2543 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2544 // TODO use sleepTime == 0 as an additional condition. 2545 if (mMixerBufferValid) { 2546 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2547 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2548 2549 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2550 mNormalFrameCount * mChannelCount); 2551 } 2552 2553 mBytesRemaining = mCurrentWriteLength; 2554 if (isSuspended()) { 2555 sleepTime = suspendSleepTimeUs(); 2556 // simulate write to HAL when suspended 2557 mBytesWritten += mSinkBufferSize; 2558 mBytesRemaining = 0; 2559 } 2560 2561 // only process effects if we're going to write 2562 if (sleepTime == 0 && mType != OFFLOAD) { 2563 for (size_t i = 0; i < effectChains.size(); i ++) { 2564 effectChains[i]->process_l(); 2565 } 2566 } 2567 } 2568 // Process effect chains for offloaded thread even if no audio 2569 // was read from audio track: process only updates effect state 2570 // and thus does have to be synchronized with audio writes but may have 2571 // to be called while waiting for async write callback 2572 if (mType == OFFLOAD) { 2573 for (size_t i = 0; i < effectChains.size(); i ++) { 2574 effectChains[i]->process_l(); 2575 } 2576 } 2577 2578 // Only if the Effects buffer is enabled and there is data in the 2579 // Effects buffer (buffer valid), we need to 2580 // copy into the sink buffer. 2581 // TODO use sleepTime == 0 as an additional condition. 2582 if (mEffectBufferValid) { 2583 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2584 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2585 mNormalFrameCount * mChannelCount); 2586 } 2587 2588 // enable changes in effect chain 2589 unlockEffectChains(effectChains); 2590 2591 if (!waitingAsyncCallback()) { 2592 // sleepTime == 0 means we must write to audio hardware 2593 if (sleepTime == 0) { 2594 if (mBytesRemaining) { 2595 ssize_t ret = threadLoop_write(); 2596 if (ret < 0) { 2597 mBytesRemaining = 0; 2598 } else { 2599 mBytesWritten += ret; 2600 mBytesRemaining -= ret; 2601 } 2602 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2603 (mMixerStatus == MIXER_DRAIN_ALL)) { 2604 threadLoop_drain(); 2605 } 2606 if (mType == MIXER) { 2607 // write blocked detection 2608 nsecs_t now = systemTime(); 2609 nsecs_t delta = now - mLastWriteTime; 2610 if (!mStandby && delta > maxPeriod) { 2611 mNumDelayedWrites++; 2612 if ((now - lastWarning) > kWarningThrottleNs) { 2613 ATRACE_NAME("underrun"); 2614 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2615 ns2ms(delta), mNumDelayedWrites, this); 2616 lastWarning = now; 2617 } 2618 } 2619 } 2620 2621 } else { 2622 usleep(sleepTime); 2623 } 2624 } 2625 2626 // Finally let go of removed track(s), without the lock held 2627 // since we can't guarantee the destructors won't acquire that 2628 // same lock. This will also mutate and push a new fast mixer state. 2629 threadLoop_removeTracks(tracksToRemove); 2630 tracksToRemove.clear(); 2631 2632 // FIXME I don't understand the need for this here; 2633 // it was in the original code but maybe the 2634 // assignment in saveOutputTracks() makes this unnecessary? 2635 clearOutputTracks(); 2636 2637 // Effect chains will be actually deleted here if they were removed from 2638 // mEffectChains list during mixing or effects processing 2639 effectChains.clear(); 2640 2641 // FIXME Note that the above .clear() is no longer necessary since effectChains 2642 // is now local to this block, but will keep it for now (at least until merge done). 2643 } 2644 2645 threadLoop_exit(); 2646 2647 if (!mStandby) { 2648 threadLoop_standby(); 2649 mStandby = true; 2650 } 2651 2652 releaseWakeLock(); 2653 mWakeLockUids.clear(); 2654 mActiveTracksGeneration++; 2655 2656 ALOGV("Thread %p type %d exiting", this, mType); 2657 return false; 2658} 2659 2660// removeTracks_l() must be called with ThreadBase::mLock held 2661void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2662{ 2663 size_t count = tracksToRemove.size(); 2664 if (count > 0) { 2665 for (size_t i=0 ; i<count ; i++) { 2666 const sp<Track>& track = tracksToRemove.itemAt(i); 2667 mActiveTracks.remove(track); 2668 mWakeLockUids.remove(track->uid()); 2669 mActiveTracksGeneration++; 2670 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2671 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2672 if (chain != 0) { 2673 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2674 track->sessionId()); 2675 chain->decActiveTrackCnt(); 2676 } 2677 if (track->isTerminated()) { 2678 removeTrack_l(track); 2679 } 2680 } 2681 } 2682 2683} 2684 2685status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2686{ 2687 if (mNormalSink != 0) { 2688 return mNormalSink->getTimestamp(timestamp); 2689 } 2690 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2691 uint64_t position64; 2692 int ret = mOutput->stream->get_presentation_position( 2693 mOutput->stream, &position64, ×tamp.mTime); 2694 if (ret == 0) { 2695 timestamp.mPosition = (uint32_t)position64; 2696 return NO_ERROR; 2697 } 2698 } 2699 return INVALID_OPERATION; 2700} 2701 2702status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2703 audio_patch_handle_t *handle) 2704{ 2705 status_t status = NO_ERROR; 2706 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2707 // store new device and send to effects 2708 audio_devices_t type = AUDIO_DEVICE_NONE; 2709 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2710 type |= patch->sinks[i].ext.device.type; 2711 } 2712 mOutDevice = type; 2713 for (size_t i = 0; i < mEffectChains.size(); i++) { 2714 mEffectChains[i]->setDevice_l(mOutDevice); 2715 } 2716 2717 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2718 status = hwDevice->create_audio_patch(hwDevice, 2719 patch->num_sources, 2720 patch->sources, 2721 patch->num_sinks, 2722 patch->sinks, 2723 handle); 2724 } else { 2725 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2726 } 2727 return status; 2728} 2729 2730status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2731{ 2732 status_t status = NO_ERROR; 2733 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2734 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2735 status = hwDevice->release_audio_patch(hwDevice, handle); 2736 } else { 2737 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2738 } 2739 return status; 2740} 2741 2742void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2743{ 2744 Mutex::Autolock _l(mLock); 2745 mTracks.add(track); 2746} 2747 2748void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2749{ 2750 Mutex::Autolock _l(mLock); 2751 destroyTrack_l(track); 2752} 2753 2754void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2755{ 2756 ThreadBase::getAudioPortConfig(config); 2757 config->role = AUDIO_PORT_ROLE_SOURCE; 2758 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2759 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2760} 2761 2762// ---------------------------------------------------------------------------- 2763 2764AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2765 audio_io_handle_t id, audio_devices_t device, type_t type) 2766 : PlaybackThread(audioFlinger, output, id, device, type), 2767 // mAudioMixer below 2768 // mFastMixer below 2769 mFastMixerFutex(0) 2770 // mOutputSink below 2771 // mPipeSink below 2772 // mNormalSink below 2773{ 2774 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2775 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2776 "mFrameCount=%d, mNormalFrameCount=%d", 2777 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2778 mNormalFrameCount); 2779 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2780 2781 // create an NBAIO sink for the HAL output stream, and negotiate 2782 mOutputSink = new AudioStreamOutSink(output->stream); 2783 size_t numCounterOffers = 0; 2784 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2785 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2786 ALOG_ASSERT(index == 0); 2787 2788 // initialize fast mixer depending on configuration 2789 bool initFastMixer; 2790 switch (kUseFastMixer) { 2791 case FastMixer_Never: 2792 initFastMixer = false; 2793 break; 2794 case FastMixer_Always: 2795 initFastMixer = true; 2796 break; 2797 case FastMixer_Static: 2798 case FastMixer_Dynamic: 2799 initFastMixer = mFrameCount < mNormalFrameCount; 2800 break; 2801 } 2802 if (initFastMixer) { 2803 audio_format_t fastMixerFormat; 2804 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2805 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2806 } else { 2807 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2808 } 2809 if (mFormat != fastMixerFormat) { 2810 // change our Sink format to accept our intermediate precision 2811 mFormat = fastMixerFormat; 2812 free(mSinkBuffer); 2813 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2814 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2815 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2816 } 2817 2818 // create a MonoPipe to connect our submix to FastMixer 2819 NBAIO_Format format = mOutputSink->format(); 2820 NBAIO_Format origformat = format; 2821 // adjust format to match that of the Fast Mixer 2822 format.mFormat = fastMixerFormat; 2823 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2824 2825 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2826 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2827 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2828 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2829 const NBAIO_Format offers[1] = {format}; 2830 size_t numCounterOffers = 0; 2831 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2832 ALOG_ASSERT(index == 0); 2833 monoPipe->setAvgFrames((mScreenState & 1) ? 2834 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2835 mPipeSink = monoPipe; 2836 2837#ifdef TEE_SINK 2838 if (mTeeSinkOutputEnabled) { 2839 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2840 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 2841 const NBAIO_Format offers2[1] = {origformat}; 2842 numCounterOffers = 0; 2843 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 2844 ALOG_ASSERT(index == 0); 2845 mTeeSink = teeSink; 2846 PipeReader *teeSource = new PipeReader(*teeSink); 2847 numCounterOffers = 0; 2848 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 2849 ALOG_ASSERT(index == 0); 2850 mTeeSource = teeSource; 2851 } 2852#endif 2853 2854 // create fast mixer and configure it initially with just one fast track for our submix 2855 mFastMixer = new FastMixer(); 2856 FastMixerStateQueue *sq = mFastMixer->sq(); 2857#ifdef STATE_QUEUE_DUMP 2858 sq->setObserverDump(&mStateQueueObserverDump); 2859 sq->setMutatorDump(&mStateQueueMutatorDump); 2860#endif 2861 FastMixerState *state = sq->begin(); 2862 FastTrack *fastTrack = &state->mFastTracks[0]; 2863 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2864 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2865 fastTrack->mVolumeProvider = NULL; 2866 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2867 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2868 fastTrack->mGeneration++; 2869 state->mFastTracksGen++; 2870 state->mTrackMask = 1; 2871 // fast mixer will use the HAL output sink 2872 state->mOutputSink = mOutputSink.get(); 2873 state->mOutputSinkGen++; 2874 state->mFrameCount = mFrameCount; 2875 state->mCommand = FastMixerState::COLD_IDLE; 2876 // already done in constructor initialization list 2877 //mFastMixerFutex = 0; 2878 state->mColdFutexAddr = &mFastMixerFutex; 2879 state->mColdGen++; 2880 state->mDumpState = &mFastMixerDumpState; 2881#ifdef TEE_SINK 2882 state->mTeeSink = mTeeSink.get(); 2883#endif 2884 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2885 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2886 sq->end(); 2887 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2888 2889 // start the fast mixer 2890 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2891 pid_t tid = mFastMixer->getTid(); 2892 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2893 if (err != 0) { 2894 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2895 kPriorityFastMixer, getpid_cached, tid, err); 2896 } 2897 2898#ifdef AUDIO_WATCHDOG 2899 // create and start the watchdog 2900 mAudioWatchdog = new AudioWatchdog(); 2901 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2902 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2903 tid = mAudioWatchdog->getTid(); 2904 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2905 if (err != 0) { 2906 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2907 kPriorityFastMixer, getpid_cached, tid, err); 2908 } 2909#endif 2910 2911 } 2912 2913 switch (kUseFastMixer) { 2914 case FastMixer_Never: 2915 case FastMixer_Dynamic: 2916 mNormalSink = mOutputSink; 2917 break; 2918 case FastMixer_Always: 2919 mNormalSink = mPipeSink; 2920 break; 2921 case FastMixer_Static: 2922 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2923 break; 2924 } 2925} 2926 2927AudioFlinger::MixerThread::~MixerThread() 2928{ 2929 if (mFastMixer != 0) { 2930 FastMixerStateQueue *sq = mFastMixer->sq(); 2931 FastMixerState *state = sq->begin(); 2932 if (state->mCommand == FastMixerState::COLD_IDLE) { 2933 int32_t old = android_atomic_inc(&mFastMixerFutex); 2934 if (old == -1) { 2935 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2936 } 2937 } 2938 state->mCommand = FastMixerState::EXIT; 2939 sq->end(); 2940 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2941 mFastMixer->join(); 2942 // Though the fast mixer thread has exited, it's state queue is still valid. 2943 // We'll use that extract the final state which contains one remaining fast track 2944 // corresponding to our sub-mix. 2945 state = sq->begin(); 2946 ALOG_ASSERT(state->mTrackMask == 1); 2947 FastTrack *fastTrack = &state->mFastTracks[0]; 2948 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2949 delete fastTrack->mBufferProvider; 2950 sq->end(false /*didModify*/); 2951 mFastMixer.clear(); 2952#ifdef AUDIO_WATCHDOG 2953 if (mAudioWatchdog != 0) { 2954 mAudioWatchdog->requestExit(); 2955 mAudioWatchdog->requestExitAndWait(); 2956 mAudioWatchdog.clear(); 2957 } 2958#endif 2959 } 2960 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2961 delete mAudioMixer; 2962} 2963 2964 2965uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2966{ 2967 if (mFastMixer != 0) { 2968 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2969 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2970 } 2971 return latency; 2972} 2973 2974 2975void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2976{ 2977 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2978} 2979 2980ssize_t AudioFlinger::MixerThread::threadLoop_write() 2981{ 2982 // FIXME we should only do one push per cycle; confirm this is true 2983 // Start the fast mixer if it's not already running 2984 if (mFastMixer != 0) { 2985 FastMixerStateQueue *sq = mFastMixer->sq(); 2986 FastMixerState *state = sq->begin(); 2987 if (state->mCommand != FastMixerState::MIX_WRITE && 2988 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2989 if (state->mCommand == FastMixerState::COLD_IDLE) { 2990 int32_t old = android_atomic_inc(&mFastMixerFutex); 2991 if (old == -1) { 2992 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2993 } 2994#ifdef AUDIO_WATCHDOG 2995 if (mAudioWatchdog != 0) { 2996 mAudioWatchdog->resume(); 2997 } 2998#endif 2999 } 3000 state->mCommand = FastMixerState::MIX_WRITE; 3001 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3002 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 3003 sq->end(); 3004 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3005 if (kUseFastMixer == FastMixer_Dynamic) { 3006 mNormalSink = mPipeSink; 3007 } 3008 } else { 3009 sq->end(false /*didModify*/); 3010 } 3011 } 3012 return PlaybackThread::threadLoop_write(); 3013} 3014 3015void AudioFlinger::MixerThread::threadLoop_standby() 3016{ 3017 // Idle the fast mixer if it's currently running 3018 if (mFastMixer != 0) { 3019 FastMixerStateQueue *sq = mFastMixer->sq(); 3020 FastMixerState *state = sq->begin(); 3021 if (!(state->mCommand & FastMixerState::IDLE)) { 3022 state->mCommand = FastMixerState::COLD_IDLE; 3023 state->mColdFutexAddr = &mFastMixerFutex; 3024 state->mColdGen++; 3025 mFastMixerFutex = 0; 3026 sq->end(); 3027 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3028 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3029 if (kUseFastMixer == FastMixer_Dynamic) { 3030 mNormalSink = mOutputSink; 3031 } 3032#ifdef AUDIO_WATCHDOG 3033 if (mAudioWatchdog != 0) { 3034 mAudioWatchdog->pause(); 3035 } 3036#endif 3037 } else { 3038 sq->end(false /*didModify*/); 3039 } 3040 } 3041 PlaybackThread::threadLoop_standby(); 3042} 3043 3044bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3045{ 3046 return false; 3047} 3048 3049bool AudioFlinger::PlaybackThread::shouldStandby_l() 3050{ 3051 return !mStandby; 3052} 3053 3054bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3055{ 3056 Mutex::Autolock _l(mLock); 3057 return waitingAsyncCallback_l(); 3058} 3059 3060// shared by MIXER and DIRECT, overridden by DUPLICATING 3061void AudioFlinger::PlaybackThread::threadLoop_standby() 3062{ 3063 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3064 mOutput->stream->common.standby(&mOutput->stream->common); 3065 if (mUseAsyncWrite != 0) { 3066 // discard any pending drain or write ack by incrementing sequence 3067 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3068 mDrainSequence = (mDrainSequence + 2) & ~1; 3069 ALOG_ASSERT(mCallbackThread != 0); 3070 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3071 mCallbackThread->setDraining(mDrainSequence); 3072 } 3073} 3074 3075void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3076{ 3077 ALOGV("signal playback thread"); 3078 broadcast_l(); 3079} 3080 3081void AudioFlinger::MixerThread::threadLoop_mix() 3082{ 3083 // obtain the presentation timestamp of the next output buffer 3084 int64_t pts; 3085 status_t status = INVALID_OPERATION; 3086 3087 if (mNormalSink != 0) { 3088 status = mNormalSink->getNextWriteTimestamp(&pts); 3089 } else { 3090 status = mOutputSink->getNextWriteTimestamp(&pts); 3091 } 3092 3093 if (status != NO_ERROR) { 3094 pts = AudioBufferProvider::kInvalidPTS; 3095 } 3096 3097 // mix buffers... 3098 mAudioMixer->process(pts); 3099 mCurrentWriteLength = mSinkBufferSize; 3100 // increase sleep time progressively when application underrun condition clears. 3101 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3102 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3103 // such that we would underrun the audio HAL. 3104 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3105 sleepTimeShift--; 3106 } 3107 sleepTime = 0; 3108 standbyTime = systemTime() + standbyDelay; 3109 //TODO: delay standby when effects have a tail 3110 3111} 3112 3113void AudioFlinger::MixerThread::threadLoop_sleepTime() 3114{ 3115 // If no tracks are ready, sleep once for the duration of an output 3116 // buffer size, then write 0s to the output 3117 if (sleepTime == 0) { 3118 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3119 sleepTime = activeSleepTime >> sleepTimeShift; 3120 if (sleepTime < kMinThreadSleepTimeUs) { 3121 sleepTime = kMinThreadSleepTimeUs; 3122 } 3123 // reduce sleep time in case of consecutive application underruns to avoid 3124 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3125 // duration we would end up writing less data than needed by the audio HAL if 3126 // the condition persists. 3127 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3128 sleepTimeShift++; 3129 } 3130 } else { 3131 sleepTime = idleSleepTime; 3132 } 3133 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3134 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3135 // before effects processing or output. 3136 if (mMixerBufferValid) { 3137 memset(mMixerBuffer, 0, mMixerBufferSize); 3138 } else { 3139 memset(mSinkBuffer, 0, mSinkBufferSize); 3140 } 3141 sleepTime = 0; 3142 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3143 "anticipated start"); 3144 } 3145 // TODO add standby time extension fct of effect tail 3146} 3147 3148// prepareTracks_l() must be called with ThreadBase::mLock held 3149AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3150 Vector< sp<Track> > *tracksToRemove) 3151{ 3152 3153 mixer_state mixerStatus = MIXER_IDLE; 3154 // find out which tracks need to be processed 3155 size_t count = mActiveTracks.size(); 3156 size_t mixedTracks = 0; 3157 size_t tracksWithEffect = 0; 3158 // counts only _active_ fast tracks 3159 size_t fastTracks = 0; 3160 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3161 3162 float masterVolume = mMasterVolume; 3163 bool masterMute = mMasterMute; 3164 3165 if (masterMute) { 3166 masterVolume = 0; 3167 } 3168 // Delegate master volume control to effect in output mix effect chain if needed 3169 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3170 if (chain != 0) { 3171 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3172 chain->setVolume_l(&v, &v); 3173 masterVolume = (float)((v + (1 << 23)) >> 24); 3174 chain.clear(); 3175 } 3176 3177 // prepare a new state to push 3178 FastMixerStateQueue *sq = NULL; 3179 FastMixerState *state = NULL; 3180 bool didModify = false; 3181 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3182 if (mFastMixer != 0) { 3183 sq = mFastMixer->sq(); 3184 state = sq->begin(); 3185 } 3186 3187 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3188 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3189 3190 for (size_t i=0 ; i<count ; i++) { 3191 const sp<Track> t = mActiveTracks[i].promote(); 3192 if (t == 0) { 3193 continue; 3194 } 3195 3196 // this const just means the local variable doesn't change 3197 Track* const track = t.get(); 3198 3199 // process fast tracks 3200 if (track->isFastTrack()) { 3201 3202 // It's theoretically possible (though unlikely) for a fast track to be created 3203 // and then removed within the same normal mix cycle. This is not a problem, as 3204 // the track never becomes active so it's fast mixer slot is never touched. 3205 // The converse, of removing an (active) track and then creating a new track 3206 // at the identical fast mixer slot within the same normal mix cycle, 3207 // is impossible because the slot isn't marked available until the end of each cycle. 3208 int j = track->mFastIndex; 3209 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3210 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3211 FastTrack *fastTrack = &state->mFastTracks[j]; 3212 3213 // Determine whether the track is currently in underrun condition, 3214 // and whether it had a recent underrun. 3215 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3216 FastTrackUnderruns underruns = ftDump->mUnderruns; 3217 uint32_t recentFull = (underruns.mBitFields.mFull - 3218 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3219 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3220 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3221 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3222 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3223 uint32_t recentUnderruns = recentPartial + recentEmpty; 3224 track->mObservedUnderruns = underruns; 3225 // don't count underruns that occur while stopping or pausing 3226 // or stopped which can occur when flush() is called while active 3227 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3228 recentUnderruns > 0) { 3229 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3230 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3231 } 3232 3233 // This is similar to the state machine for normal tracks, 3234 // with a few modifications for fast tracks. 3235 bool isActive = true; 3236 switch (track->mState) { 3237 case TrackBase::STOPPING_1: 3238 // track stays active in STOPPING_1 state until first underrun 3239 if (recentUnderruns > 0 || track->isTerminated()) { 3240 track->mState = TrackBase::STOPPING_2; 3241 } 3242 break; 3243 case TrackBase::PAUSING: 3244 // ramp down is not yet implemented 3245 track->setPaused(); 3246 break; 3247 case TrackBase::RESUMING: 3248 // ramp up is not yet implemented 3249 track->mState = TrackBase::ACTIVE; 3250 break; 3251 case TrackBase::ACTIVE: 3252 if (recentFull > 0 || recentPartial > 0) { 3253 // track has provided at least some frames recently: reset retry count 3254 track->mRetryCount = kMaxTrackRetries; 3255 } 3256 if (recentUnderruns == 0) { 3257 // no recent underruns: stay active 3258 break; 3259 } 3260 // there has recently been an underrun of some kind 3261 if (track->sharedBuffer() == 0) { 3262 // were any of the recent underruns "empty" (no frames available)? 3263 if (recentEmpty == 0) { 3264 // no, then ignore the partial underruns as they are allowed indefinitely 3265 break; 3266 } 3267 // there has recently been an "empty" underrun: decrement the retry counter 3268 if (--(track->mRetryCount) > 0) { 3269 break; 3270 } 3271 // indicate to client process that the track was disabled because of underrun; 3272 // it will then automatically call start() when data is available 3273 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3274 // remove from active list, but state remains ACTIVE [confusing but true] 3275 isActive = false; 3276 break; 3277 } 3278 // fall through 3279 case TrackBase::STOPPING_2: 3280 case TrackBase::PAUSED: 3281 case TrackBase::STOPPED: 3282 case TrackBase::FLUSHED: // flush() while active 3283 // Check for presentation complete if track is inactive 3284 // We have consumed all the buffers of this track. 3285 // This would be incomplete if we auto-paused on underrun 3286 { 3287 size_t audioHALFrames = 3288 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3289 size_t framesWritten = mBytesWritten / mFrameSize; 3290 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3291 // track stays in active list until presentation is complete 3292 break; 3293 } 3294 } 3295 if (track->isStopping_2()) { 3296 track->mState = TrackBase::STOPPED; 3297 } 3298 if (track->isStopped()) { 3299 // Can't reset directly, as fast mixer is still polling this track 3300 // track->reset(); 3301 // So instead mark this track as needing to be reset after push with ack 3302 resetMask |= 1 << i; 3303 } 3304 isActive = false; 3305 break; 3306 case TrackBase::IDLE: 3307 default: 3308 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3309 } 3310 3311 if (isActive) { 3312 // was it previously inactive? 3313 if (!(state->mTrackMask & (1 << j))) { 3314 ExtendedAudioBufferProvider *eabp = track; 3315 VolumeProvider *vp = track; 3316 fastTrack->mBufferProvider = eabp; 3317 fastTrack->mVolumeProvider = vp; 3318 fastTrack->mChannelMask = track->mChannelMask; 3319 fastTrack->mFormat = track->mFormat; 3320 fastTrack->mGeneration++; 3321 state->mTrackMask |= 1 << j; 3322 didModify = true; 3323 // no acknowledgement required for newly active tracks 3324 } 3325 // cache the combined master volume and stream type volume for fast mixer; this 3326 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3327 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3328 ++fastTracks; 3329 } else { 3330 // was it previously active? 3331 if (state->mTrackMask & (1 << j)) { 3332 fastTrack->mBufferProvider = NULL; 3333 fastTrack->mGeneration++; 3334 state->mTrackMask &= ~(1 << j); 3335 didModify = true; 3336 // If any fast tracks were removed, we must wait for acknowledgement 3337 // because we're about to decrement the last sp<> on those tracks. 3338 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3339 } else { 3340 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3341 } 3342 tracksToRemove->add(track); 3343 // Avoids a misleading display in dumpsys 3344 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3345 } 3346 continue; 3347 } 3348 3349 { // local variable scope to avoid goto warning 3350 3351 audio_track_cblk_t* cblk = track->cblk(); 3352 3353 // The first time a track is added we wait 3354 // for all its buffers to be filled before processing it 3355 int name = track->name(); 3356 // make sure that we have enough frames to mix one full buffer. 3357 // enforce this condition only once to enable draining the buffer in case the client 3358 // app does not call stop() and relies on underrun to stop: 3359 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3360 // during last round 3361 size_t desiredFrames; 3362 uint32_t sr = track->sampleRate(); 3363 if (sr == mSampleRate) { 3364 desiredFrames = mNormalFrameCount; 3365 } else { 3366 // +1 for rounding and +1 for additional sample needed for interpolation 3367 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3368 // add frames already consumed but not yet released by the resampler 3369 // because mAudioTrackServerProxy->framesReady() will include these frames 3370 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3371#if 0 3372 // the minimum track buffer size is normally twice the number of frames necessary 3373 // to fill one buffer and the resampler should not leave more than one buffer worth 3374 // of unreleased frames after each pass, but just in case... 3375 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3376#endif 3377 } 3378 uint32_t minFrames = 1; 3379 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3380 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3381 minFrames = desiredFrames; 3382 } 3383 3384 size_t framesReady = track->framesReady(); 3385 if ((framesReady >= minFrames) && track->isReady() && 3386 !track->isPaused() && !track->isTerminated()) 3387 { 3388 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3389 3390 mixedTracks++; 3391 3392 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3393 // there is an effect chain connected to the track 3394 chain.clear(); 3395 if (track->mainBuffer() != mSinkBuffer && 3396 track->mainBuffer() != mMixerBuffer) { 3397 if (mEffectBufferEnabled) { 3398 mEffectBufferValid = true; // Later can set directly. 3399 } 3400 chain = getEffectChain_l(track->sessionId()); 3401 // Delegate volume control to effect in track effect chain if needed 3402 if (chain != 0) { 3403 tracksWithEffect++; 3404 } else { 3405 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3406 "session %d", 3407 name, track->sessionId()); 3408 } 3409 } 3410 3411 3412 int param = AudioMixer::VOLUME; 3413 if (track->mFillingUpStatus == Track::FS_FILLED) { 3414 // no ramp for the first volume setting 3415 track->mFillingUpStatus = Track::FS_ACTIVE; 3416 if (track->mState == TrackBase::RESUMING) { 3417 track->mState = TrackBase::ACTIVE; 3418 param = AudioMixer::RAMP_VOLUME; 3419 } 3420 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3421 // FIXME should not make a decision based on mServer 3422 } else if (cblk->mServer != 0) { 3423 // If the track is stopped before the first frame was mixed, 3424 // do not apply ramp 3425 param = AudioMixer::RAMP_VOLUME; 3426 } 3427 3428 // compute volume for this track 3429 uint32_t vl, vr; // in U8.24 integer format 3430 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3431 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3432 vl = vr = 0; 3433 vlf = vrf = vaf = 0.; 3434 if (track->isPausing()) { 3435 track->setPaused(); 3436 } 3437 } else { 3438 3439 // read original volumes with volume control 3440 float typeVolume = mStreamTypes[track->streamType()].volume; 3441 float v = masterVolume * typeVolume; 3442 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3443 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3444 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3445 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3446 // track volumes come from shared memory, so can't be trusted and must be clamped 3447 if (vlf > GAIN_FLOAT_UNITY) { 3448 ALOGV("Track left volume out of range: %.3g", vlf); 3449 vlf = GAIN_FLOAT_UNITY; 3450 } 3451 if (vrf > GAIN_FLOAT_UNITY) { 3452 ALOGV("Track right volume out of range: %.3g", vrf); 3453 vrf = GAIN_FLOAT_UNITY; 3454 } 3455 // now apply the master volume and stream type volume 3456 vlf *= v; 3457 vrf *= v; 3458 // assuming master volume and stream type volume each go up to 1.0, 3459 // then derive vl and vr as U8.24 versions for the effect chain 3460 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3461 vl = (uint32_t) (scaleto8_24 * vlf); 3462 vr = (uint32_t) (scaleto8_24 * vrf); 3463 // vl and vr are now in U8.24 format 3464 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3465 // send level comes from shared memory and so may be corrupt 3466 if (sendLevel > MAX_GAIN_INT) { 3467 ALOGV("Track send level out of range: %04X", sendLevel); 3468 sendLevel = MAX_GAIN_INT; 3469 } 3470 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3471 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3472 } 3473 3474 // Delegate volume control to effect in track effect chain if needed 3475 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3476 // Do not ramp volume if volume is controlled by effect 3477 param = AudioMixer::VOLUME; 3478 // Update remaining floating point volume levels 3479 vlf = (float)vl / (1 << 24); 3480 vrf = (float)vr / (1 << 24); 3481 track->mHasVolumeController = true; 3482 } else { 3483 // force no volume ramp when volume controller was just disabled or removed 3484 // from effect chain to avoid volume spike 3485 if (track->mHasVolumeController) { 3486 param = AudioMixer::VOLUME; 3487 } 3488 track->mHasVolumeController = false; 3489 } 3490 3491 // XXX: these things DON'T need to be done each time 3492 mAudioMixer->setBufferProvider(name, track); 3493 mAudioMixer->enable(name); 3494 3495 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3496 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3497 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3498 mAudioMixer->setParameter( 3499 name, 3500 AudioMixer::TRACK, 3501 AudioMixer::FORMAT, (void *)track->format()); 3502 mAudioMixer->setParameter( 3503 name, 3504 AudioMixer::TRACK, 3505 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3506 mAudioMixer->setParameter( 3507 name, 3508 AudioMixer::TRACK, 3509 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3510 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3511 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3512 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3513 if (reqSampleRate == 0) { 3514 reqSampleRate = mSampleRate; 3515 } else if (reqSampleRate > maxSampleRate) { 3516 reqSampleRate = maxSampleRate; 3517 } 3518 mAudioMixer->setParameter( 3519 name, 3520 AudioMixer::RESAMPLE, 3521 AudioMixer::SAMPLE_RATE, 3522 (void *)(uintptr_t)reqSampleRate); 3523 /* 3524 * Select the appropriate output buffer for the track. 3525 * 3526 * Tracks with effects go into their own effects chain buffer 3527 * and from there into either mEffectBuffer or mSinkBuffer. 3528 * 3529 * Other tracks can use mMixerBuffer for higher precision 3530 * channel accumulation. If this buffer is enabled 3531 * (mMixerBufferEnabled true), then selected tracks will accumulate 3532 * into it. 3533 * 3534 */ 3535 if (mMixerBufferEnabled 3536 && (track->mainBuffer() == mSinkBuffer 3537 || track->mainBuffer() == mMixerBuffer)) { 3538 mAudioMixer->setParameter( 3539 name, 3540 AudioMixer::TRACK, 3541 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3542 mAudioMixer->setParameter( 3543 name, 3544 AudioMixer::TRACK, 3545 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3546 // TODO: override track->mainBuffer()? 3547 mMixerBufferValid = true; 3548 } else { 3549 mAudioMixer->setParameter( 3550 name, 3551 AudioMixer::TRACK, 3552 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3553 mAudioMixer->setParameter( 3554 name, 3555 AudioMixer::TRACK, 3556 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3557 } 3558 mAudioMixer->setParameter( 3559 name, 3560 AudioMixer::TRACK, 3561 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3562 3563 // reset retry count 3564 track->mRetryCount = kMaxTrackRetries; 3565 3566 // If one track is ready, set the mixer ready if: 3567 // - the mixer was not ready during previous round OR 3568 // - no other track is not ready 3569 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3570 mixerStatus != MIXER_TRACKS_ENABLED) { 3571 mixerStatus = MIXER_TRACKS_READY; 3572 } 3573 } else { 3574 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3575 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3576 } 3577 // clear effect chain input buffer if an active track underruns to avoid sending 3578 // previous audio buffer again to effects 3579 chain = getEffectChain_l(track->sessionId()); 3580 if (chain != 0) { 3581 chain->clearInputBuffer(); 3582 } 3583 3584 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3585 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3586 track->isStopped() || track->isPaused()) { 3587 // We have consumed all the buffers of this track. 3588 // Remove it from the list of active tracks. 3589 // TODO: use actual buffer filling status instead of latency when available from 3590 // audio HAL 3591 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3592 size_t framesWritten = mBytesWritten / mFrameSize; 3593 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3594 if (track->isStopped()) { 3595 track->reset(); 3596 } 3597 tracksToRemove->add(track); 3598 } 3599 } else { 3600 // No buffers for this track. Give it a few chances to 3601 // fill a buffer, then remove it from active list. 3602 if (--(track->mRetryCount) <= 0) { 3603 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3604 tracksToRemove->add(track); 3605 // indicate to client process that the track was disabled because of underrun; 3606 // it will then automatically call start() when data is available 3607 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3608 // If one track is not ready, mark the mixer also not ready if: 3609 // - the mixer was ready during previous round OR 3610 // - no other track is ready 3611 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3612 mixerStatus != MIXER_TRACKS_READY) { 3613 mixerStatus = MIXER_TRACKS_ENABLED; 3614 } 3615 } 3616 mAudioMixer->disable(name); 3617 } 3618 3619 } // local variable scope to avoid goto warning 3620track_is_ready: ; 3621 3622 } 3623 3624 // Push the new FastMixer state if necessary 3625 bool pauseAudioWatchdog = false; 3626 if (didModify) { 3627 state->mFastTracksGen++; 3628 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3629 if (kUseFastMixer == FastMixer_Dynamic && 3630 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3631 state->mCommand = FastMixerState::COLD_IDLE; 3632 state->mColdFutexAddr = &mFastMixerFutex; 3633 state->mColdGen++; 3634 mFastMixerFutex = 0; 3635 if (kUseFastMixer == FastMixer_Dynamic) { 3636 mNormalSink = mOutputSink; 3637 } 3638 // If we go into cold idle, need to wait for acknowledgement 3639 // so that fast mixer stops doing I/O. 3640 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3641 pauseAudioWatchdog = true; 3642 } 3643 } 3644 if (sq != NULL) { 3645 sq->end(didModify); 3646 sq->push(block); 3647 } 3648#ifdef AUDIO_WATCHDOG 3649 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3650 mAudioWatchdog->pause(); 3651 } 3652#endif 3653 3654 // Now perform the deferred reset on fast tracks that have stopped 3655 while (resetMask != 0) { 3656 size_t i = __builtin_ctz(resetMask); 3657 ALOG_ASSERT(i < count); 3658 resetMask &= ~(1 << i); 3659 sp<Track> t = mActiveTracks[i].promote(); 3660 if (t == 0) { 3661 continue; 3662 } 3663 Track* track = t.get(); 3664 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3665 track->reset(); 3666 } 3667 3668 // remove all the tracks that need to be... 3669 removeTracks_l(*tracksToRemove); 3670 3671 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3672 mEffectBufferValid = true; 3673 } 3674 3675 // sink or mix buffer must be cleared if all tracks are connected to an 3676 // effect chain as in this case the mixer will not write to the sink or mix buffer 3677 // and track effects will accumulate into it 3678 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3679 (mixedTracks == 0 && fastTracks > 0))) { 3680 // FIXME as a performance optimization, should remember previous zero status 3681 if (mMixerBufferValid) { 3682 memset(mMixerBuffer, 0, mMixerBufferSize); 3683 // TODO: In testing, mSinkBuffer below need not be cleared because 3684 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3685 // after mixing. 3686 // 3687 // To enforce this guarantee: 3688 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3689 // (mixedTracks == 0 && fastTracks > 0)) 3690 // must imply MIXER_TRACKS_READY. 3691 // Later, we may clear buffers regardless, and skip much of this logic. 3692 } 3693 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3694 if (mEffectBufferValid) { 3695 memset(mEffectBuffer, 0, mEffectBufferSize); 3696 } 3697 // FIXME as a performance optimization, should remember previous zero status 3698 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3699 } 3700 3701 // if any fast tracks, then status is ready 3702 mMixerStatusIgnoringFastTracks = mixerStatus; 3703 if (fastTracks > 0) { 3704 mixerStatus = MIXER_TRACKS_READY; 3705 } 3706 return mixerStatus; 3707} 3708 3709// getTrackName_l() must be called with ThreadBase::mLock held 3710int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3711 audio_format_t format, int sessionId) 3712{ 3713 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3714} 3715 3716// deleteTrackName_l() must be called with ThreadBase::mLock held 3717void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3718{ 3719 ALOGV("remove track (%d) and delete from mixer", name); 3720 mAudioMixer->deleteTrackName(name); 3721} 3722 3723// checkForNewParameter_l() must be called with ThreadBase::mLock held 3724bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3725 status_t& status) 3726{ 3727 bool reconfig = false; 3728 3729 status = NO_ERROR; 3730 3731 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3732 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3733 if (mFastMixer != 0) { 3734 FastMixerStateQueue *sq = mFastMixer->sq(); 3735 FastMixerState *state = sq->begin(); 3736 if (!(state->mCommand & FastMixerState::IDLE)) { 3737 previousCommand = state->mCommand; 3738 state->mCommand = FastMixerState::HOT_IDLE; 3739 sq->end(); 3740 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3741 } else { 3742 sq->end(false /*didModify*/); 3743 } 3744 } 3745 3746 AudioParameter param = AudioParameter(keyValuePair); 3747 int value; 3748 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3749 reconfig = true; 3750 } 3751 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3752 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3753 status = BAD_VALUE; 3754 } else { 3755 // no need to save value, since it's constant 3756 reconfig = true; 3757 } 3758 } 3759 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3760 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3761 status = BAD_VALUE; 3762 } else { 3763 // no need to save value, since it's constant 3764 reconfig = true; 3765 } 3766 } 3767 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3768 // do not accept frame count changes if tracks are open as the track buffer 3769 // size depends on frame count and correct behavior would not be guaranteed 3770 // if frame count is changed after track creation 3771 if (!mTracks.isEmpty()) { 3772 status = INVALID_OPERATION; 3773 } else { 3774 reconfig = true; 3775 } 3776 } 3777 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3778#ifdef ADD_BATTERY_DATA 3779 // when changing the audio output device, call addBatteryData to notify 3780 // the change 3781 if (mOutDevice != value) { 3782 uint32_t params = 0; 3783 // check whether speaker is on 3784 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3785 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3786 } 3787 3788 audio_devices_t deviceWithoutSpeaker 3789 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3790 // check if any other device (except speaker) is on 3791 if (value & deviceWithoutSpeaker ) { 3792 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3793 } 3794 3795 if (params != 0) { 3796 addBatteryData(params); 3797 } 3798 } 3799#endif 3800 3801 // forward device change to effects that have requested to be 3802 // aware of attached audio device. 3803 if (value != AUDIO_DEVICE_NONE) { 3804 mOutDevice = value; 3805 for (size_t i = 0; i < mEffectChains.size(); i++) { 3806 mEffectChains[i]->setDevice_l(mOutDevice); 3807 } 3808 } 3809 } 3810 3811 if (status == NO_ERROR) { 3812 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3813 keyValuePair.string()); 3814 if (!mStandby && status == INVALID_OPERATION) { 3815 mOutput->stream->common.standby(&mOutput->stream->common); 3816 mStandby = true; 3817 mBytesWritten = 0; 3818 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3819 keyValuePair.string()); 3820 } 3821 if (status == NO_ERROR && reconfig) { 3822 readOutputParameters_l(); 3823 delete mAudioMixer; 3824 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3825 for (size_t i = 0; i < mTracks.size() ; i++) { 3826 int name = getTrackName_l(mTracks[i]->mChannelMask, 3827 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3828 if (name < 0) { 3829 break; 3830 } 3831 mTracks[i]->mName = name; 3832 } 3833 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3834 } 3835 } 3836 3837 if (!(previousCommand & FastMixerState::IDLE)) { 3838 ALOG_ASSERT(mFastMixer != 0); 3839 FastMixerStateQueue *sq = mFastMixer->sq(); 3840 FastMixerState *state = sq->begin(); 3841 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3842 state->mCommand = previousCommand; 3843 sq->end(); 3844 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3845 } 3846 3847 return reconfig; 3848} 3849 3850 3851void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3852{ 3853 const size_t SIZE = 256; 3854 char buffer[SIZE]; 3855 String8 result; 3856 3857 PlaybackThread::dumpInternals(fd, args); 3858 3859 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3860 3861 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3862 const FastMixerDumpState copy(mFastMixerDumpState); 3863 copy.dump(fd); 3864 3865#ifdef STATE_QUEUE_DUMP 3866 // Similar for state queue 3867 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3868 observerCopy.dump(fd); 3869 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3870 mutatorCopy.dump(fd); 3871#endif 3872 3873#ifdef TEE_SINK 3874 // Write the tee output to a .wav file 3875 dumpTee(fd, mTeeSource, mId); 3876#endif 3877 3878#ifdef AUDIO_WATCHDOG 3879 if (mAudioWatchdog != 0) { 3880 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3881 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3882 wdCopy.dump(fd); 3883 } 3884#endif 3885} 3886 3887uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3888{ 3889 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3890} 3891 3892uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3893{ 3894 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3895} 3896 3897void AudioFlinger::MixerThread::cacheParameters_l() 3898{ 3899 PlaybackThread::cacheParameters_l(); 3900 3901 // FIXME: Relaxed timing because of a certain device that can't meet latency 3902 // Should be reduced to 2x after the vendor fixes the driver issue 3903 // increase threshold again due to low power audio mode. The way this warning 3904 // threshold is calculated and its usefulness should be reconsidered anyway. 3905 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3906} 3907 3908// ---------------------------------------------------------------------------- 3909 3910AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3911 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3912 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3913 // mLeftVolFloat, mRightVolFloat 3914{ 3915} 3916 3917AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3918 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3919 ThreadBase::type_t type) 3920 : PlaybackThread(audioFlinger, output, id, device, type) 3921 // mLeftVolFloat, mRightVolFloat 3922{ 3923} 3924 3925AudioFlinger::DirectOutputThread::~DirectOutputThread() 3926{ 3927} 3928 3929void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3930{ 3931 audio_track_cblk_t* cblk = track->cblk(); 3932 float left, right; 3933 3934 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3935 left = right = 0; 3936 } else { 3937 float typeVolume = mStreamTypes[track->streamType()].volume; 3938 float v = mMasterVolume * typeVolume; 3939 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3940 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3941 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3942 if (left > GAIN_FLOAT_UNITY) { 3943 left = GAIN_FLOAT_UNITY; 3944 } 3945 left *= v; 3946 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3947 if (right > GAIN_FLOAT_UNITY) { 3948 right = GAIN_FLOAT_UNITY; 3949 } 3950 right *= v; 3951 } 3952 3953 if (lastTrack) { 3954 if (left != mLeftVolFloat || right != mRightVolFloat) { 3955 mLeftVolFloat = left; 3956 mRightVolFloat = right; 3957 3958 // Convert volumes from float to 8.24 3959 uint32_t vl = (uint32_t)(left * (1 << 24)); 3960 uint32_t vr = (uint32_t)(right * (1 << 24)); 3961 3962 // Delegate volume control to effect in track effect chain if needed 3963 // only one effect chain can be present on DirectOutputThread, so if 3964 // there is one, the track is connected to it 3965 if (!mEffectChains.isEmpty()) { 3966 mEffectChains[0]->setVolume_l(&vl, &vr); 3967 left = (float)vl / (1 << 24); 3968 right = (float)vr / (1 << 24); 3969 } 3970 if (mOutput->stream->set_volume) { 3971 mOutput->stream->set_volume(mOutput->stream, left, right); 3972 } 3973 } 3974 } 3975} 3976 3977 3978AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3979 Vector< sp<Track> > *tracksToRemove 3980) 3981{ 3982 size_t count = mActiveTracks.size(); 3983 mixer_state mixerStatus = MIXER_IDLE; 3984 3985 // find out which tracks need to be processed 3986 for (size_t i = 0; i < count; i++) { 3987 sp<Track> t = mActiveTracks[i].promote(); 3988 // The track died recently 3989 if (t == 0) { 3990 continue; 3991 } 3992 3993 Track* const track = t.get(); 3994 audio_track_cblk_t* cblk = track->cblk(); 3995 // Only consider last track started for volume and mixer state control. 3996 // In theory an older track could underrun and restart after the new one starts 3997 // but as we only care about the transition phase between two tracks on a 3998 // direct output, it is not a problem to ignore the underrun case. 3999 sp<Track> l = mLatestActiveTrack.promote(); 4000 bool last = l.get() == track; 4001 4002 // The first time a track is added we wait 4003 // for all its buffers to be filled before processing it 4004 uint32_t minFrames; 4005 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 4006 minFrames = mNormalFrameCount; 4007 } else { 4008 minFrames = 1; 4009 } 4010 4011 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4012 !track->isStopping_2() && !track->isStopped()) 4013 { 4014 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4015 4016 if (track->mFillingUpStatus == Track::FS_FILLED) { 4017 track->mFillingUpStatus = Track::FS_ACTIVE; 4018 // make sure processVolume_l() will apply new volume even if 0 4019 mLeftVolFloat = mRightVolFloat = -1.0; 4020 if (track->mState == TrackBase::RESUMING) { 4021 track->mState = TrackBase::ACTIVE; 4022 } 4023 } 4024 4025 // compute volume for this track 4026 processVolume_l(track, last); 4027 if (last) { 4028 // reset retry count 4029 track->mRetryCount = kMaxTrackRetriesDirect; 4030 mActiveTrack = t; 4031 mixerStatus = MIXER_TRACKS_READY; 4032 } 4033 } else { 4034 // clear effect chain input buffer if the last active track started underruns 4035 // to avoid sending previous audio buffer again to effects 4036 if (!mEffectChains.isEmpty() && last) { 4037 mEffectChains[0]->clearInputBuffer(); 4038 } 4039 if (track->isStopping_1()) { 4040 track->mState = TrackBase::STOPPING_2; 4041 } 4042 if ((track->sharedBuffer() != 0) || track->isStopped() || 4043 track->isStopping_2() || track->isPaused()) { 4044 // We have consumed all the buffers of this track. 4045 // Remove it from the list of active tracks. 4046 size_t audioHALFrames; 4047 if (audio_is_linear_pcm(mFormat)) { 4048 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4049 } else { 4050 audioHALFrames = 0; 4051 } 4052 4053 size_t framesWritten = mBytesWritten / mFrameSize; 4054 if (mStandby || !last || 4055 track->presentationComplete(framesWritten, audioHALFrames)) { 4056 if (track->isStopping_2()) { 4057 track->mState = TrackBase::STOPPED; 4058 } 4059 if (track->isStopped()) { 4060 if (track->mState == TrackBase::FLUSHED) { 4061 flushHw_l(); 4062 } 4063 track->reset(); 4064 } 4065 tracksToRemove->add(track); 4066 } 4067 } else { 4068 // No buffers for this track. Give it a few chances to 4069 // fill a buffer, then remove it from active list. 4070 // Only consider last track started for mixer state control 4071 if (--(track->mRetryCount) <= 0) { 4072 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4073 tracksToRemove->add(track); 4074 // indicate to client process that the track was disabled because of underrun; 4075 // it will then automatically call start() when data is available 4076 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4077 } else if (last) { 4078 mixerStatus = MIXER_TRACKS_ENABLED; 4079 } 4080 } 4081 } 4082 } 4083 4084 // remove all the tracks that need to be... 4085 removeTracks_l(*tracksToRemove); 4086 4087 return mixerStatus; 4088} 4089 4090void AudioFlinger::DirectOutputThread::threadLoop_mix() 4091{ 4092 size_t frameCount = mFrameCount; 4093 int8_t *curBuf = (int8_t *)mSinkBuffer; 4094 // output audio to hardware 4095 while (frameCount) { 4096 AudioBufferProvider::Buffer buffer; 4097 buffer.frameCount = frameCount; 4098 mActiveTrack->getNextBuffer(&buffer); 4099 if (buffer.raw == NULL) { 4100 memset(curBuf, 0, frameCount * mFrameSize); 4101 break; 4102 } 4103 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4104 frameCount -= buffer.frameCount; 4105 curBuf += buffer.frameCount * mFrameSize; 4106 mActiveTrack->releaseBuffer(&buffer); 4107 } 4108 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4109 sleepTime = 0; 4110 standbyTime = systemTime() + standbyDelay; 4111 mActiveTrack.clear(); 4112} 4113 4114void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4115{ 4116 if (sleepTime == 0) { 4117 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4118 sleepTime = activeSleepTime; 4119 } else { 4120 sleepTime = idleSleepTime; 4121 } 4122 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4123 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4124 sleepTime = 0; 4125 } 4126} 4127 4128// getTrackName_l() must be called with ThreadBase::mLock held 4129int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4130 audio_format_t format __unused, int sessionId __unused) 4131{ 4132 return 0; 4133} 4134 4135// deleteTrackName_l() must be called with ThreadBase::mLock held 4136void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4137{ 4138} 4139 4140// checkForNewParameter_l() must be called with ThreadBase::mLock held 4141bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4142 status_t& status) 4143{ 4144 bool reconfig = false; 4145 4146 status = NO_ERROR; 4147 4148 AudioParameter param = AudioParameter(keyValuePair); 4149 int value; 4150 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4151 // forward device change to effects that have requested to be 4152 // aware of attached audio device. 4153 if (value != AUDIO_DEVICE_NONE) { 4154 mOutDevice = value; 4155 for (size_t i = 0; i < mEffectChains.size(); i++) { 4156 mEffectChains[i]->setDevice_l(mOutDevice); 4157 } 4158 } 4159 } 4160 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4161 // do not accept frame count changes if tracks are open as the track buffer 4162 // size depends on frame count and correct behavior would not be garantied 4163 // if frame count is changed after track creation 4164 if (!mTracks.isEmpty()) { 4165 status = INVALID_OPERATION; 4166 } else { 4167 reconfig = true; 4168 } 4169 } 4170 if (status == NO_ERROR) { 4171 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4172 keyValuePair.string()); 4173 if (!mStandby && status == INVALID_OPERATION) { 4174 mOutput->stream->common.standby(&mOutput->stream->common); 4175 mStandby = true; 4176 mBytesWritten = 0; 4177 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4178 keyValuePair.string()); 4179 } 4180 if (status == NO_ERROR && reconfig) { 4181 readOutputParameters_l(); 4182 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4183 } 4184 } 4185 4186 return reconfig; 4187} 4188 4189uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4190{ 4191 uint32_t time; 4192 if (audio_is_linear_pcm(mFormat)) { 4193 time = PlaybackThread::activeSleepTimeUs(); 4194 } else { 4195 time = 10000; 4196 } 4197 return time; 4198} 4199 4200uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4201{ 4202 uint32_t time; 4203 if (audio_is_linear_pcm(mFormat)) { 4204 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4205 } else { 4206 time = 10000; 4207 } 4208 return time; 4209} 4210 4211uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4212{ 4213 uint32_t time; 4214 if (audio_is_linear_pcm(mFormat)) { 4215 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4216 } else { 4217 time = 10000; 4218 } 4219 return time; 4220} 4221 4222void AudioFlinger::DirectOutputThread::cacheParameters_l() 4223{ 4224 PlaybackThread::cacheParameters_l(); 4225 4226 // use shorter standby delay as on normal output to release 4227 // hardware resources as soon as possible 4228 if (audio_is_linear_pcm(mFormat)) { 4229 standbyDelay = microseconds(activeSleepTime*2); 4230 } else { 4231 standbyDelay = kOffloadStandbyDelayNs; 4232 } 4233} 4234 4235void AudioFlinger::DirectOutputThread::flushHw_l() 4236{ 4237 if (mOutput->stream->flush != NULL) 4238 mOutput->stream->flush(mOutput->stream); 4239} 4240 4241// ---------------------------------------------------------------------------- 4242 4243AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4244 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4245 : Thread(false /*canCallJava*/), 4246 mPlaybackThread(playbackThread), 4247 mWriteAckSequence(0), 4248 mDrainSequence(0) 4249{ 4250} 4251 4252AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4253{ 4254} 4255 4256void AudioFlinger::AsyncCallbackThread::onFirstRef() 4257{ 4258 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4259} 4260 4261bool AudioFlinger::AsyncCallbackThread::threadLoop() 4262{ 4263 while (!exitPending()) { 4264 uint32_t writeAckSequence; 4265 uint32_t drainSequence; 4266 4267 { 4268 Mutex::Autolock _l(mLock); 4269 while (!((mWriteAckSequence & 1) || 4270 (mDrainSequence & 1) || 4271 exitPending())) { 4272 mWaitWorkCV.wait(mLock); 4273 } 4274 4275 if (exitPending()) { 4276 break; 4277 } 4278 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4279 mWriteAckSequence, mDrainSequence); 4280 writeAckSequence = mWriteAckSequence; 4281 mWriteAckSequence &= ~1; 4282 drainSequence = mDrainSequence; 4283 mDrainSequence &= ~1; 4284 } 4285 { 4286 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4287 if (playbackThread != 0) { 4288 if (writeAckSequence & 1) { 4289 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4290 } 4291 if (drainSequence & 1) { 4292 playbackThread->resetDraining(drainSequence >> 1); 4293 } 4294 } 4295 } 4296 } 4297 return false; 4298} 4299 4300void AudioFlinger::AsyncCallbackThread::exit() 4301{ 4302 ALOGV("AsyncCallbackThread::exit"); 4303 Mutex::Autolock _l(mLock); 4304 requestExit(); 4305 mWaitWorkCV.broadcast(); 4306} 4307 4308void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4309{ 4310 Mutex::Autolock _l(mLock); 4311 // bit 0 is cleared 4312 mWriteAckSequence = sequence << 1; 4313} 4314 4315void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4316{ 4317 Mutex::Autolock _l(mLock); 4318 // ignore unexpected callbacks 4319 if (mWriteAckSequence & 2) { 4320 mWriteAckSequence |= 1; 4321 mWaitWorkCV.signal(); 4322 } 4323} 4324 4325void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4326{ 4327 Mutex::Autolock _l(mLock); 4328 // bit 0 is cleared 4329 mDrainSequence = sequence << 1; 4330} 4331 4332void AudioFlinger::AsyncCallbackThread::resetDraining() 4333{ 4334 Mutex::Autolock _l(mLock); 4335 // ignore unexpected callbacks 4336 if (mDrainSequence & 2) { 4337 mDrainSequence |= 1; 4338 mWaitWorkCV.signal(); 4339 } 4340} 4341 4342 4343// ---------------------------------------------------------------------------- 4344AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4345 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4346 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4347 mHwPaused(false), 4348 mFlushPending(false), 4349 mPausedBytesRemaining(0) 4350{ 4351 //FIXME: mStandby should be set to true by ThreadBase constructor 4352 mStandby = true; 4353} 4354 4355void AudioFlinger::OffloadThread::threadLoop_exit() 4356{ 4357 if (mFlushPending || mHwPaused) { 4358 // If a flush is pending or track was paused, just discard buffered data 4359 flushHw_l(); 4360 } else { 4361 mMixerStatus = MIXER_DRAIN_ALL; 4362 threadLoop_drain(); 4363 } 4364 if (mUseAsyncWrite) { 4365 ALOG_ASSERT(mCallbackThread != 0); 4366 mCallbackThread->exit(); 4367 } 4368 PlaybackThread::threadLoop_exit(); 4369} 4370 4371AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4372 Vector< sp<Track> > *tracksToRemove 4373) 4374{ 4375 size_t count = mActiveTracks.size(); 4376 4377 mixer_state mixerStatus = MIXER_IDLE; 4378 bool doHwPause = false; 4379 bool doHwResume = false; 4380 4381 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4382 4383 // find out which tracks need to be processed 4384 for (size_t i = 0; i < count; i++) { 4385 sp<Track> t = mActiveTracks[i].promote(); 4386 // The track died recently 4387 if (t == 0) { 4388 continue; 4389 } 4390 Track* const track = t.get(); 4391 audio_track_cblk_t* cblk = track->cblk(); 4392 // Only consider last track started for volume and mixer state control. 4393 // In theory an older track could underrun and restart after the new one starts 4394 // but as we only care about the transition phase between two tracks on a 4395 // direct output, it is not a problem to ignore the underrun case. 4396 sp<Track> l = mLatestActiveTrack.promote(); 4397 bool last = l.get() == track; 4398 4399 if (track->isInvalid()) { 4400 ALOGW("An invalidated track shouldn't be in active list"); 4401 tracksToRemove->add(track); 4402 continue; 4403 } 4404 4405 if (track->mState == TrackBase::IDLE) { 4406 ALOGW("An idle track shouldn't be in active list"); 4407 continue; 4408 } 4409 4410 if (track->isPausing()) { 4411 track->setPaused(); 4412 if (last) { 4413 if (!mHwPaused) { 4414 doHwPause = true; 4415 mHwPaused = true; 4416 } 4417 // If we were part way through writing the mixbuffer to 4418 // the HAL we must save this until we resume 4419 // BUG - this will be wrong if a different track is made active, 4420 // in that case we want to discard the pending data in the 4421 // mixbuffer and tell the client to present it again when the 4422 // track is resumed 4423 mPausedWriteLength = mCurrentWriteLength; 4424 mPausedBytesRemaining = mBytesRemaining; 4425 mBytesRemaining = 0; // stop writing 4426 } 4427 tracksToRemove->add(track); 4428 } else if (track->isFlushPending()) { 4429 track->flushAck(); 4430 if (last) { 4431 mFlushPending = true; 4432 } 4433 } else if (track->isResumePending()){ 4434 track->resumeAck(); 4435 if (last) { 4436 if (mPausedBytesRemaining) { 4437 // Need to continue write that was interrupted 4438 mCurrentWriteLength = mPausedWriteLength; 4439 mBytesRemaining = mPausedBytesRemaining; 4440 mPausedBytesRemaining = 0; 4441 } 4442 if (mHwPaused) { 4443 doHwResume = true; 4444 mHwPaused = false; 4445 // threadLoop_mix() will handle the case that we need to 4446 // resume an interrupted write 4447 } 4448 // enable write to audio HAL 4449 sleepTime = 0; 4450 4451 // Do not handle new data in this iteration even if track->framesReady() 4452 mixerStatus = MIXER_TRACKS_ENABLED; 4453 } 4454 } else if (track->framesReady() && track->isReady() && 4455 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4456 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4457 if (track->mFillingUpStatus == Track::FS_FILLED) { 4458 track->mFillingUpStatus = Track::FS_ACTIVE; 4459 // make sure processVolume_l() will apply new volume even if 0 4460 mLeftVolFloat = mRightVolFloat = -1.0; 4461 } 4462 4463 if (last) { 4464 sp<Track> previousTrack = mPreviousTrack.promote(); 4465 if (previousTrack != 0) { 4466 if (track != previousTrack.get()) { 4467 // Flush any data still being written from last track 4468 mBytesRemaining = 0; 4469 if (mPausedBytesRemaining) { 4470 // Last track was paused so we also need to flush saved 4471 // mixbuffer state and invalidate track so that it will 4472 // re-submit that unwritten data when it is next resumed 4473 mPausedBytesRemaining = 0; 4474 // Invalidate is a bit drastic - would be more efficient 4475 // to have a flag to tell client that some of the 4476 // previously written data was lost 4477 previousTrack->invalidate(); 4478 } 4479 // flush data already sent to the DSP if changing audio session as audio 4480 // comes from a different source. Also invalidate previous track to force a 4481 // seek when resuming. 4482 if (previousTrack->sessionId() != track->sessionId()) { 4483 previousTrack->invalidate(); 4484 } 4485 } 4486 } 4487 mPreviousTrack = track; 4488 // reset retry count 4489 track->mRetryCount = kMaxTrackRetriesOffload; 4490 mActiveTrack = t; 4491 mixerStatus = MIXER_TRACKS_READY; 4492 } 4493 } else { 4494 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4495 if (track->isStopping_1()) { 4496 // Hardware buffer can hold a large amount of audio so we must 4497 // wait for all current track's data to drain before we say 4498 // that the track is stopped. 4499 if (mBytesRemaining == 0) { 4500 // Only start draining when all data in mixbuffer 4501 // has been written 4502 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4503 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4504 // do not drain if no data was ever sent to HAL (mStandby == true) 4505 if (last && !mStandby) { 4506 // do not modify drain sequence if we are already draining. This happens 4507 // when resuming from pause after drain. 4508 if ((mDrainSequence & 1) == 0) { 4509 sleepTime = 0; 4510 standbyTime = systemTime() + standbyDelay; 4511 mixerStatus = MIXER_DRAIN_TRACK; 4512 mDrainSequence += 2; 4513 } 4514 if (mHwPaused) { 4515 // It is possible to move from PAUSED to STOPPING_1 without 4516 // a resume so we must ensure hardware is running 4517 doHwResume = true; 4518 mHwPaused = false; 4519 } 4520 } 4521 } 4522 } else if (track->isStopping_2()) { 4523 // Drain has completed or we are in standby, signal presentation complete 4524 if (!(mDrainSequence & 1) || !last || mStandby) { 4525 track->mState = TrackBase::STOPPED; 4526 size_t audioHALFrames = 4527 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4528 size_t framesWritten = 4529 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4530 track->presentationComplete(framesWritten, audioHALFrames); 4531 track->reset(); 4532 tracksToRemove->add(track); 4533 } 4534 } else { 4535 // No buffers for this track. Give it a few chances to 4536 // fill a buffer, then remove it from active list. 4537 if (--(track->mRetryCount) <= 0) { 4538 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4539 track->name()); 4540 tracksToRemove->add(track); 4541 // indicate to client process that the track was disabled because of underrun; 4542 // it will then automatically call start() when data is available 4543 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4544 } else if (last){ 4545 mixerStatus = MIXER_TRACKS_ENABLED; 4546 } 4547 } 4548 } 4549 // compute volume for this track 4550 processVolume_l(track, last); 4551 } 4552 4553 // make sure the pause/flush/resume sequence is executed in the right order. 4554 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4555 // before flush and then resume HW. This can happen in case of pause/flush/resume 4556 // if resume is received before pause is executed. 4557 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4558 mOutput->stream->pause(mOutput->stream); 4559 } 4560 if (mFlushPending) { 4561 flushHw_l(); 4562 mFlushPending = false; 4563 } 4564 if (!mStandby && doHwResume) { 4565 mOutput->stream->resume(mOutput->stream); 4566 } 4567 4568 // remove all the tracks that need to be... 4569 removeTracks_l(*tracksToRemove); 4570 4571 return mixerStatus; 4572} 4573 4574// must be called with thread mutex locked 4575bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4576{ 4577 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4578 mWriteAckSequence, mDrainSequence); 4579 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4580 return true; 4581 } 4582 return false; 4583} 4584 4585// must be called with thread mutex locked 4586bool AudioFlinger::OffloadThread::shouldStandby_l() 4587{ 4588 bool trackPaused = false; 4589 4590 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4591 // after a timeout and we will enter standby then. 4592 if (mTracks.size() > 0) { 4593 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4594 } 4595 4596 return !mStandby && !trackPaused; 4597} 4598 4599 4600bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4601{ 4602 Mutex::Autolock _l(mLock); 4603 return waitingAsyncCallback_l(); 4604} 4605 4606void AudioFlinger::OffloadThread::flushHw_l() 4607{ 4608 DirectOutputThread::flushHw_l(); 4609 // Flush anything still waiting in the mixbuffer 4610 mCurrentWriteLength = 0; 4611 mBytesRemaining = 0; 4612 mPausedWriteLength = 0; 4613 mPausedBytesRemaining = 0; 4614 mHwPaused = false; 4615 4616 if (mUseAsyncWrite) { 4617 // discard any pending drain or write ack by incrementing sequence 4618 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4619 mDrainSequence = (mDrainSequence + 2) & ~1; 4620 ALOG_ASSERT(mCallbackThread != 0); 4621 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4622 mCallbackThread->setDraining(mDrainSequence); 4623 } 4624} 4625 4626void AudioFlinger::OffloadThread::onAddNewTrack_l() 4627{ 4628 sp<Track> previousTrack = mPreviousTrack.promote(); 4629 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4630 4631 if (previousTrack != 0 && latestTrack != 0 && 4632 (previousTrack->sessionId() != latestTrack->sessionId())) { 4633 mFlushPending = true; 4634 } 4635 PlaybackThread::onAddNewTrack_l(); 4636} 4637 4638// ---------------------------------------------------------------------------- 4639 4640AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4641 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4642 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4643 DUPLICATING), 4644 mWaitTimeMs(UINT_MAX) 4645{ 4646 addOutputTrack(mainThread); 4647} 4648 4649AudioFlinger::DuplicatingThread::~DuplicatingThread() 4650{ 4651 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4652 mOutputTracks[i]->destroy(); 4653 } 4654} 4655 4656void AudioFlinger::DuplicatingThread::threadLoop_mix() 4657{ 4658 // mix buffers... 4659 if (outputsReady(outputTracks)) { 4660 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4661 } else { 4662 memset(mSinkBuffer, 0, mSinkBufferSize); 4663 } 4664 sleepTime = 0; 4665 writeFrames = mNormalFrameCount; 4666 mCurrentWriteLength = mSinkBufferSize; 4667 standbyTime = systemTime() + standbyDelay; 4668} 4669 4670void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4671{ 4672 if (sleepTime == 0) { 4673 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4674 sleepTime = activeSleepTime; 4675 } else { 4676 sleepTime = idleSleepTime; 4677 } 4678 } else if (mBytesWritten != 0) { 4679 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4680 writeFrames = mNormalFrameCount; 4681 memset(mSinkBuffer, 0, mSinkBufferSize); 4682 } else { 4683 // flush remaining overflow buffers in output tracks 4684 writeFrames = 0; 4685 } 4686 sleepTime = 0; 4687 } 4688} 4689 4690ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4691{ 4692 for (size_t i = 0; i < outputTracks.size(); i++) { 4693 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4694 // for delivery downstream as needed. This in-place conversion is safe as 4695 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4696 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4697 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4698 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4699 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4700 } 4701 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4702 } 4703 mStandby = false; 4704 return (ssize_t)mSinkBufferSize; 4705} 4706 4707void AudioFlinger::DuplicatingThread::threadLoop_standby() 4708{ 4709 // DuplicatingThread implements standby by stopping all tracks 4710 for (size_t i = 0; i < outputTracks.size(); i++) { 4711 outputTracks[i]->stop(); 4712 } 4713} 4714 4715void AudioFlinger::DuplicatingThread::saveOutputTracks() 4716{ 4717 outputTracks = mOutputTracks; 4718} 4719 4720void AudioFlinger::DuplicatingThread::clearOutputTracks() 4721{ 4722 outputTracks.clear(); 4723} 4724 4725void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4726{ 4727 Mutex::Autolock _l(mLock); 4728 // FIXME explain this formula 4729 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4730 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4731 // due to current usage case and restrictions on the AudioBufferProvider. 4732 // Actual buffer conversion is done in threadLoop_write(). 4733 // 4734 // TODO: This may change in the future, depending on multichannel 4735 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4736 OutputTrack *outputTrack = new OutputTrack(thread, 4737 this, 4738 mSampleRate, 4739 AUDIO_FORMAT_PCM_16_BIT, 4740 mChannelMask, 4741 frameCount, 4742 IPCThreadState::self()->getCallingUid()); 4743 if (outputTrack->cblk() != NULL) { 4744 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4745 mOutputTracks.add(outputTrack); 4746 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4747 updateWaitTime_l(); 4748 } 4749} 4750 4751void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4752{ 4753 Mutex::Autolock _l(mLock); 4754 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4755 if (mOutputTracks[i]->thread() == thread) { 4756 mOutputTracks[i]->destroy(); 4757 mOutputTracks.removeAt(i); 4758 updateWaitTime_l(); 4759 return; 4760 } 4761 } 4762 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4763} 4764 4765// caller must hold mLock 4766void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4767{ 4768 mWaitTimeMs = UINT_MAX; 4769 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4770 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4771 if (strong != 0) { 4772 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4773 if (waitTimeMs < mWaitTimeMs) { 4774 mWaitTimeMs = waitTimeMs; 4775 } 4776 } 4777 } 4778} 4779 4780 4781bool AudioFlinger::DuplicatingThread::outputsReady( 4782 const SortedVector< sp<OutputTrack> > &outputTracks) 4783{ 4784 for (size_t i = 0; i < outputTracks.size(); i++) { 4785 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4786 if (thread == 0) { 4787 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4788 outputTracks[i].get()); 4789 return false; 4790 } 4791 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4792 // see note at standby() declaration 4793 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4794 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4795 thread.get()); 4796 return false; 4797 } 4798 } 4799 return true; 4800} 4801 4802uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4803{ 4804 return (mWaitTimeMs * 1000) / 2; 4805} 4806 4807void AudioFlinger::DuplicatingThread::cacheParameters_l() 4808{ 4809 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4810 updateWaitTime_l(); 4811 4812 MixerThread::cacheParameters_l(); 4813} 4814 4815// ---------------------------------------------------------------------------- 4816// Record 4817// ---------------------------------------------------------------------------- 4818 4819AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4820 AudioStreamIn *input, 4821 audio_io_handle_t id, 4822 audio_devices_t outDevice, 4823 audio_devices_t inDevice 4824#ifdef TEE_SINK 4825 , const sp<NBAIO_Sink>& teeSink 4826#endif 4827 ) : 4828 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4829 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4830 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4831 mRsmpInRear(0) 4832#ifdef TEE_SINK 4833 , mTeeSink(teeSink) 4834#endif 4835 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4836 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4837 // mFastCapture below 4838 , mFastCaptureFutex(0) 4839 // mInputSource 4840 // mPipeSink 4841 // mPipeSource 4842 , mPipeFramesP2(0) 4843 // mPipeMemory 4844 // mFastCaptureNBLogWriter 4845 , mFastTrackAvail(false) 4846{ 4847 snprintf(mName, kNameLength, "AudioIn_%X", id); 4848 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4849 4850 readInputParameters_l(); 4851 4852 // create an NBAIO source for the HAL input stream, and negotiate 4853 mInputSource = new AudioStreamInSource(input->stream); 4854 size_t numCounterOffers = 0; 4855 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4856 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4857 ALOG_ASSERT(index == 0); 4858 4859 // initialize fast capture depending on configuration 4860 bool initFastCapture; 4861 switch (kUseFastCapture) { 4862 case FastCapture_Never: 4863 initFastCapture = false; 4864 break; 4865 case FastCapture_Always: 4866 initFastCapture = true; 4867 break; 4868 case FastCapture_Static: 4869 uint32_t primaryOutputSampleRate; 4870 { 4871 AutoMutex _l(audioFlinger->mHardwareLock); 4872 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4873 } 4874 initFastCapture = 4875 // either capture sample rate is same as (a reasonable) primary output sample rate 4876 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4877 (mSampleRate == primaryOutputSampleRate)) || 4878 // or primary output sample rate is unknown, and capture sample rate is reasonable 4879 ((primaryOutputSampleRate == 0) && 4880 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4881 // and the buffer size is < 12 ms 4882 (mFrameCount * 1000) / mSampleRate < 12; 4883 break; 4884 // case FastCapture_Dynamic: 4885 } 4886 4887 if (initFastCapture) { 4888 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4889 NBAIO_Format format = mInputSource->format(); 4890 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4891 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4892 void *pipeBuffer; 4893 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4894 sp<IMemory> pipeMemory; 4895 if ((roHeap == 0) || 4896 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4897 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4898 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4899 goto failed; 4900 } 4901 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4902 memset(pipeBuffer, 0, pipeSize); 4903 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4904 const NBAIO_Format offers[1] = {format}; 4905 size_t numCounterOffers = 0; 4906 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4907 ALOG_ASSERT(index == 0); 4908 mPipeSink = pipe; 4909 PipeReader *pipeReader = new PipeReader(*pipe); 4910 numCounterOffers = 0; 4911 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4912 ALOG_ASSERT(index == 0); 4913 mPipeSource = pipeReader; 4914 mPipeFramesP2 = pipeFramesP2; 4915 mPipeMemory = pipeMemory; 4916 4917 // create fast capture 4918 mFastCapture = new FastCapture(); 4919 FastCaptureStateQueue *sq = mFastCapture->sq(); 4920#ifdef STATE_QUEUE_DUMP 4921 // FIXME 4922#endif 4923 FastCaptureState *state = sq->begin(); 4924 state->mCblk = NULL; 4925 state->mInputSource = mInputSource.get(); 4926 state->mInputSourceGen++; 4927 state->mPipeSink = pipe; 4928 state->mPipeSinkGen++; 4929 state->mFrameCount = mFrameCount; 4930 state->mCommand = FastCaptureState::COLD_IDLE; 4931 // already done in constructor initialization list 4932 //mFastCaptureFutex = 0; 4933 state->mColdFutexAddr = &mFastCaptureFutex; 4934 state->mColdGen++; 4935 state->mDumpState = &mFastCaptureDumpState; 4936#ifdef TEE_SINK 4937 // FIXME 4938#endif 4939 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4940 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4941 sq->end(); 4942 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4943 4944 // start the fast capture 4945 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4946 pid_t tid = mFastCapture->getTid(); 4947 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4948 if (err != 0) { 4949 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4950 kPriorityFastCapture, getpid_cached, tid, err); 4951 } 4952 4953#ifdef AUDIO_WATCHDOG 4954 // FIXME 4955#endif 4956 4957 mFastTrackAvail = true; 4958 } 4959failed: ; 4960 4961 // FIXME mNormalSource 4962} 4963 4964 4965AudioFlinger::RecordThread::~RecordThread() 4966{ 4967 if (mFastCapture != 0) { 4968 FastCaptureStateQueue *sq = mFastCapture->sq(); 4969 FastCaptureState *state = sq->begin(); 4970 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4971 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4972 if (old == -1) { 4973 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4974 } 4975 } 4976 state->mCommand = FastCaptureState::EXIT; 4977 sq->end(); 4978 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4979 mFastCapture->join(); 4980 mFastCapture.clear(); 4981 } 4982 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4983 mAudioFlinger->unregisterWriter(mNBLogWriter); 4984 delete[] mRsmpInBuffer; 4985} 4986 4987void AudioFlinger::RecordThread::onFirstRef() 4988{ 4989 run(mName, PRIORITY_URGENT_AUDIO); 4990} 4991 4992bool AudioFlinger::RecordThread::threadLoop() 4993{ 4994 nsecs_t lastWarning = 0; 4995 4996 inputStandBy(); 4997 4998reacquire_wakelock: 4999 sp<RecordTrack> activeTrack; 5000 int activeTracksGen; 5001 { 5002 Mutex::Autolock _l(mLock); 5003 size_t size = mActiveTracks.size(); 5004 activeTracksGen = mActiveTracksGen; 5005 if (size > 0) { 5006 // FIXME an arbitrary choice 5007 activeTrack = mActiveTracks[0]; 5008 acquireWakeLock_l(activeTrack->uid()); 5009 if (size > 1) { 5010 SortedVector<int> tmp; 5011 for (size_t i = 0; i < size; i++) { 5012 tmp.add(mActiveTracks[i]->uid()); 5013 } 5014 updateWakeLockUids_l(tmp); 5015 } 5016 } else { 5017 acquireWakeLock_l(-1); 5018 } 5019 } 5020 5021 // used to request a deferred sleep, to be executed later while mutex is unlocked 5022 uint32_t sleepUs = 0; 5023 5024 // loop while there is work to do 5025 for (;;) { 5026 Vector< sp<EffectChain> > effectChains; 5027 5028 // sleep with mutex unlocked 5029 if (sleepUs > 0) { 5030 usleep(sleepUs); 5031 sleepUs = 0; 5032 } 5033 5034 // activeTracks accumulates a copy of a subset of mActiveTracks 5035 Vector< sp<RecordTrack> > activeTracks; 5036 5037 // reference to the (first and only) active fast track 5038 sp<RecordTrack> fastTrack; 5039 5040 // reference to a fast track which is about to be removed 5041 sp<RecordTrack> fastTrackToRemove; 5042 5043 { // scope for mLock 5044 Mutex::Autolock _l(mLock); 5045 5046 processConfigEvents_l(); 5047 5048 // check exitPending here because checkForNewParameters_l() and 5049 // checkForNewParameters_l() can temporarily release mLock 5050 if (exitPending()) { 5051 break; 5052 } 5053 5054 // if no active track(s), then standby and release wakelock 5055 size_t size = mActiveTracks.size(); 5056 if (size == 0) { 5057 standbyIfNotAlreadyInStandby(); 5058 // exitPending() can't become true here 5059 releaseWakeLock_l(); 5060 ALOGV("RecordThread: loop stopping"); 5061 // go to sleep 5062 mWaitWorkCV.wait(mLock); 5063 ALOGV("RecordThread: loop starting"); 5064 goto reacquire_wakelock; 5065 } 5066 5067 if (mActiveTracksGen != activeTracksGen) { 5068 activeTracksGen = mActiveTracksGen; 5069 SortedVector<int> tmp; 5070 for (size_t i = 0; i < size; i++) { 5071 tmp.add(mActiveTracks[i]->uid()); 5072 } 5073 updateWakeLockUids_l(tmp); 5074 } 5075 5076 bool doBroadcast = false; 5077 for (size_t i = 0; i < size; ) { 5078 5079 activeTrack = mActiveTracks[i]; 5080 if (activeTrack->isTerminated()) { 5081 if (activeTrack->isFastTrack()) { 5082 ALOG_ASSERT(fastTrackToRemove == 0); 5083 fastTrackToRemove = activeTrack; 5084 } 5085 removeTrack_l(activeTrack); 5086 mActiveTracks.remove(activeTrack); 5087 mActiveTracksGen++; 5088 size--; 5089 continue; 5090 } 5091 5092 TrackBase::track_state activeTrackState = activeTrack->mState; 5093 switch (activeTrackState) { 5094 5095 case TrackBase::PAUSING: 5096 mActiveTracks.remove(activeTrack); 5097 mActiveTracksGen++; 5098 doBroadcast = true; 5099 size--; 5100 continue; 5101 5102 case TrackBase::STARTING_1: 5103 sleepUs = 10000; 5104 i++; 5105 continue; 5106 5107 case TrackBase::STARTING_2: 5108 doBroadcast = true; 5109 mStandby = false; 5110 activeTrack->mState = TrackBase::ACTIVE; 5111 break; 5112 5113 case TrackBase::ACTIVE: 5114 break; 5115 5116 case TrackBase::IDLE: 5117 i++; 5118 continue; 5119 5120 default: 5121 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5122 } 5123 5124 activeTracks.add(activeTrack); 5125 i++; 5126 5127 if (activeTrack->isFastTrack()) { 5128 ALOG_ASSERT(!mFastTrackAvail); 5129 ALOG_ASSERT(fastTrack == 0); 5130 fastTrack = activeTrack; 5131 } 5132 } 5133 if (doBroadcast) { 5134 mStartStopCond.broadcast(); 5135 } 5136 5137 // sleep if there are no active tracks to process 5138 if (activeTracks.size() == 0) { 5139 if (sleepUs == 0) { 5140 sleepUs = kRecordThreadSleepUs; 5141 } 5142 continue; 5143 } 5144 sleepUs = 0; 5145 5146 lockEffectChains_l(effectChains); 5147 } 5148 5149 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5150 5151 size_t size = effectChains.size(); 5152 for (size_t i = 0; i < size; i++) { 5153 // thread mutex is not locked, but effect chain is locked 5154 effectChains[i]->process_l(); 5155 } 5156 5157 // Push a new fast capture state if fast capture is not already running, or cblk change 5158 if (mFastCapture != 0) { 5159 FastCaptureStateQueue *sq = mFastCapture->sq(); 5160 FastCaptureState *state = sq->begin(); 5161 bool didModify = false; 5162 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5163 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5164 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5165 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5166 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5167 if (old == -1) { 5168 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5169 } 5170 } 5171 state->mCommand = FastCaptureState::READ_WRITE; 5172#if 0 // FIXME 5173 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5174 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5175#endif 5176 didModify = true; 5177 } 5178 audio_track_cblk_t *cblkOld = state->mCblk; 5179 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5180 if (cblkNew != cblkOld) { 5181 state->mCblk = cblkNew; 5182 // block until acked if removing a fast track 5183 if (cblkOld != NULL) { 5184 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5185 } 5186 didModify = true; 5187 } 5188 sq->end(didModify); 5189 if (didModify) { 5190 sq->push(block); 5191#if 0 5192 if (kUseFastCapture == FastCapture_Dynamic) { 5193 mNormalSource = mPipeSource; 5194 } 5195#endif 5196 } 5197 } 5198 5199 // now run the fast track destructor with thread mutex unlocked 5200 fastTrackToRemove.clear(); 5201 5202 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5203 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5204 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5205 // If destination is non-contiguous, first read past the nominal end of buffer, then 5206 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5207 5208 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5209 ssize_t framesRead; 5210 5211 // If an NBAIO source is present, use it to read the normal capture's data 5212 if (mPipeSource != 0) { 5213 size_t framesToRead = mBufferSize / mFrameSize; 5214 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5215 framesToRead, AudioBufferProvider::kInvalidPTS); 5216 if (framesRead == 0) { 5217 // since pipe is non-blocking, simulate blocking input 5218 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5219 } 5220 // otherwise use the HAL / AudioStreamIn directly 5221 } else { 5222 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5223 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5224 if (bytesRead < 0) { 5225 framesRead = bytesRead; 5226 } else { 5227 framesRead = bytesRead / mFrameSize; 5228 } 5229 } 5230 5231 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5232 ALOGE("read failed: framesRead=%d", framesRead); 5233 // Force input into standby so that it tries to recover at next read attempt 5234 inputStandBy(); 5235 sleepUs = kRecordThreadSleepUs; 5236 } 5237 if (framesRead <= 0) { 5238 goto unlock; 5239 } 5240 ALOG_ASSERT(framesRead > 0); 5241 5242 if (mTeeSink != 0) { 5243 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5244 } 5245 // If destination is non-contiguous, we now correct for reading past end of buffer. 5246 { 5247 size_t part1 = mRsmpInFramesP2 - rear; 5248 if ((size_t) framesRead > part1) { 5249 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5250 (framesRead - part1) * mFrameSize); 5251 } 5252 } 5253 rear = mRsmpInRear += framesRead; 5254 5255 size = activeTracks.size(); 5256 // loop over each active track 5257 for (size_t i = 0; i < size; i++) { 5258 activeTrack = activeTracks[i]; 5259 5260 // skip fast tracks, as those are handled directly by FastCapture 5261 if (activeTrack->isFastTrack()) { 5262 continue; 5263 } 5264 5265 enum { 5266 OVERRUN_UNKNOWN, 5267 OVERRUN_TRUE, 5268 OVERRUN_FALSE 5269 } overrun = OVERRUN_UNKNOWN; 5270 5271 // loop over getNextBuffer to handle circular sink 5272 for (;;) { 5273 5274 activeTrack->mSink.frameCount = ~0; 5275 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5276 size_t framesOut = activeTrack->mSink.frameCount; 5277 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5278 5279 int32_t front = activeTrack->mRsmpInFront; 5280 ssize_t filled = rear - front; 5281 size_t framesIn; 5282 5283 if (filled < 0) { 5284 // should not happen, but treat like a massive overrun and re-sync 5285 framesIn = 0; 5286 activeTrack->mRsmpInFront = rear; 5287 overrun = OVERRUN_TRUE; 5288 } else if ((size_t) filled <= mRsmpInFrames) { 5289 framesIn = (size_t) filled; 5290 } else { 5291 // client is not keeping up with server, but give it latest data 5292 framesIn = mRsmpInFrames; 5293 activeTrack->mRsmpInFront = front = rear - framesIn; 5294 overrun = OVERRUN_TRUE; 5295 } 5296 5297 if (framesOut == 0 || framesIn == 0) { 5298 break; 5299 } 5300 5301 if (activeTrack->mResampler == NULL) { 5302 // no resampling 5303 if (framesIn > framesOut) { 5304 framesIn = framesOut; 5305 } else { 5306 framesOut = framesIn; 5307 } 5308 int8_t *dst = activeTrack->mSink.i8; 5309 while (framesIn > 0) { 5310 front &= mRsmpInFramesP2 - 1; 5311 size_t part1 = mRsmpInFramesP2 - front; 5312 if (part1 > framesIn) { 5313 part1 = framesIn; 5314 } 5315 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5316 if (mChannelCount == activeTrack->mChannelCount) { 5317 memcpy(dst, src, part1 * mFrameSize); 5318 } else if (mChannelCount == 1) { 5319 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5320 part1); 5321 } else { 5322 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5323 part1); 5324 } 5325 dst += part1 * activeTrack->mFrameSize; 5326 front += part1; 5327 framesIn -= part1; 5328 } 5329 activeTrack->mRsmpInFront += framesOut; 5330 5331 } else { 5332 // resampling 5333 // FIXME framesInNeeded should really be part of resampler API, and should 5334 // depend on the SRC ratio 5335 // to keep mRsmpInBuffer full so resampler always has sufficient input 5336 size_t framesInNeeded; 5337 // FIXME only re-calculate when it changes, and optimize for common ratios 5338 // Do not precompute in/out because floating point is not associative 5339 // e.g. a*b/c != a*(b/c). 5340 const double in(mSampleRate); 5341 const double out(activeTrack->mSampleRate); 5342 framesInNeeded = ceil(framesOut * in / out) + 1; 5343 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5344 framesInNeeded, framesOut, in / out); 5345 // Although we theoretically have framesIn in circular buffer, some of those are 5346 // unreleased frames, and thus must be discounted for purpose of budgeting. 5347 size_t unreleased = activeTrack->mRsmpInUnrel; 5348 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5349 if (framesIn < framesInNeeded) { 5350 ALOGV("not enough to resample: have %u frames in but need %u in to " 5351 "produce %u out given in/out ratio of %.4g", 5352 framesIn, framesInNeeded, framesOut, in / out); 5353 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5354 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5355 if (newFramesOut == 0) { 5356 break; 5357 } 5358 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5359 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5360 framesInNeeded, newFramesOut, out / in); 5361 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5362 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5363 "given in/out ratio of %.4g", 5364 framesIn, framesInNeeded, newFramesOut, in / out); 5365 framesOut = newFramesOut; 5366 } else { 5367 ALOGV("success 1: have %u in and need %u in to produce %u out " 5368 "given in/out ratio of %.4g", 5369 framesIn, framesInNeeded, framesOut, in / out); 5370 } 5371 5372 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5373 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5374 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5375 delete[] activeTrack->mRsmpOutBuffer; 5376 // resampler always outputs stereo 5377 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5378 activeTrack->mRsmpOutFrameCount = framesOut; 5379 } 5380 5381 // resampler accumulates, but we only have one source track 5382 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5383 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5384 // FIXME how about having activeTrack implement this interface itself? 5385 activeTrack->mResamplerBufferProvider 5386 /*this*/ /* AudioBufferProvider* */); 5387 // ditherAndClamp() works as long as all buffers returned by 5388 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5389 if (activeTrack->mChannelCount == 1) { 5390 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5391 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5392 framesOut); 5393 // the resampler always outputs stereo samples: 5394 // do post stereo to mono conversion 5395 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5396 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5397 } else { 5398 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5399 activeTrack->mRsmpOutBuffer, framesOut); 5400 } 5401 // now done with mRsmpOutBuffer 5402 5403 } 5404 5405 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5406 overrun = OVERRUN_FALSE; 5407 } 5408 5409 if (activeTrack->mFramesToDrop == 0) { 5410 if (framesOut > 0) { 5411 activeTrack->mSink.frameCount = framesOut; 5412 activeTrack->releaseBuffer(&activeTrack->mSink); 5413 } 5414 } else { 5415 // FIXME could do a partial drop of framesOut 5416 if (activeTrack->mFramesToDrop > 0) { 5417 activeTrack->mFramesToDrop -= framesOut; 5418 if (activeTrack->mFramesToDrop <= 0) { 5419 activeTrack->clearSyncStartEvent(); 5420 } 5421 } else { 5422 activeTrack->mFramesToDrop += framesOut; 5423 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5424 activeTrack->mSyncStartEvent->isCancelled()) { 5425 ALOGW("Synced record %s, session %d, trigger session %d", 5426 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5427 activeTrack->sessionId(), 5428 (activeTrack->mSyncStartEvent != 0) ? 5429 activeTrack->mSyncStartEvent->triggerSession() : 0); 5430 activeTrack->clearSyncStartEvent(); 5431 } 5432 } 5433 } 5434 5435 if (framesOut == 0) { 5436 break; 5437 } 5438 } 5439 5440 switch (overrun) { 5441 case OVERRUN_TRUE: 5442 // client isn't retrieving buffers fast enough 5443 if (!activeTrack->setOverflow()) { 5444 nsecs_t now = systemTime(); 5445 // FIXME should lastWarning per track? 5446 if ((now - lastWarning) > kWarningThrottleNs) { 5447 ALOGW("RecordThread: buffer overflow"); 5448 lastWarning = now; 5449 } 5450 } 5451 break; 5452 case OVERRUN_FALSE: 5453 activeTrack->clearOverflow(); 5454 break; 5455 case OVERRUN_UNKNOWN: 5456 break; 5457 } 5458 5459 } 5460 5461unlock: 5462 // enable changes in effect chain 5463 unlockEffectChains(effectChains); 5464 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5465 } 5466 5467 standbyIfNotAlreadyInStandby(); 5468 5469 { 5470 Mutex::Autolock _l(mLock); 5471 for (size_t i = 0; i < mTracks.size(); i++) { 5472 sp<RecordTrack> track = mTracks[i]; 5473 track->invalidate(); 5474 } 5475 mActiveTracks.clear(); 5476 mActiveTracksGen++; 5477 mStartStopCond.broadcast(); 5478 } 5479 5480 releaseWakeLock(); 5481 5482 ALOGV("RecordThread %p exiting", this); 5483 return false; 5484} 5485 5486void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5487{ 5488 if (!mStandby) { 5489 inputStandBy(); 5490 mStandby = true; 5491 } 5492} 5493 5494void AudioFlinger::RecordThread::inputStandBy() 5495{ 5496 // Idle the fast capture if it's currently running 5497 if (mFastCapture != 0) { 5498 FastCaptureStateQueue *sq = mFastCapture->sq(); 5499 FastCaptureState *state = sq->begin(); 5500 if (!(state->mCommand & FastCaptureState::IDLE)) { 5501 state->mCommand = FastCaptureState::COLD_IDLE; 5502 state->mColdFutexAddr = &mFastCaptureFutex; 5503 state->mColdGen++; 5504 mFastCaptureFutex = 0; 5505 sq->end(); 5506 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5507 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5508#if 0 5509 if (kUseFastCapture == FastCapture_Dynamic) { 5510 // FIXME 5511 } 5512#endif 5513#ifdef AUDIO_WATCHDOG 5514 // FIXME 5515#endif 5516 } else { 5517 sq->end(false /*didModify*/); 5518 } 5519 } 5520 mInput->stream->common.standby(&mInput->stream->common); 5521} 5522 5523// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5524sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5525 const sp<AudioFlinger::Client>& client, 5526 uint32_t sampleRate, 5527 audio_format_t format, 5528 audio_channel_mask_t channelMask, 5529 size_t *pFrameCount, 5530 int sessionId, 5531 size_t *notificationFrames, 5532 int uid, 5533 IAudioFlinger::track_flags_t *flags, 5534 pid_t tid, 5535 status_t *status) 5536{ 5537 size_t frameCount = *pFrameCount; 5538 sp<RecordTrack> track; 5539 status_t lStatus; 5540 5541 // client expresses a preference for FAST, but we get the final say 5542 if (*flags & IAudioFlinger::TRACK_FAST) { 5543 if ( 5544 // use case: callback handler 5545 (tid != -1) && 5546 // frame count is not specified, or is exactly the pipe depth 5547 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5548 // PCM data 5549 audio_is_linear_pcm(format) && 5550 // native format 5551 (format == mFormat) && 5552 // native channel mask 5553 (channelMask == mChannelMask) && 5554 // native hardware sample rate 5555 (sampleRate == mSampleRate) && 5556 // record thread has an associated fast capture 5557 hasFastCapture() && 5558 // there are sufficient fast track slots available 5559 mFastTrackAvail 5560 ) { 5561 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5562 frameCount, mFrameCount); 5563 } else { 5564 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5565 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5566 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5567 frameCount, mFrameCount, mPipeFramesP2, 5568 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5569 hasFastCapture(), tid, mFastTrackAvail); 5570 *flags &= ~IAudioFlinger::TRACK_FAST; 5571 } 5572 } 5573 5574 // compute track buffer size in frames, and suggest the notification frame count 5575 if (*flags & IAudioFlinger::TRACK_FAST) { 5576 // fast track: frame count is exactly the pipe depth 5577 frameCount = mPipeFramesP2; 5578 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5579 *notificationFrames = mFrameCount; 5580 } else { 5581 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5582 // or 20 ms if there is a fast capture 5583 // TODO This could be a roundupRatio inline, and const 5584 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5585 * sampleRate + mSampleRate - 1) / mSampleRate; 5586 // minimum number of notification periods is at least kMinNotifications, 5587 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5588 static const size_t kMinNotifications = 3; 5589 static const uint32_t kMinMs = 30; 5590 // TODO This could be a roundupRatio inline 5591 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5592 // TODO This could be a roundupRatio inline 5593 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5594 maxNotificationFrames; 5595 const size_t minFrameCount = maxNotificationFrames * 5596 max(kMinNotifications, minNotificationsByMs); 5597 frameCount = max(frameCount, minFrameCount); 5598 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5599 *notificationFrames = maxNotificationFrames; 5600 } 5601 } 5602 *pFrameCount = frameCount; 5603 5604 lStatus = initCheck(); 5605 if (lStatus != NO_ERROR) { 5606 ALOGE("createRecordTrack_l() audio driver not initialized"); 5607 goto Exit; 5608 } 5609 5610 { // scope for mLock 5611 Mutex::Autolock _l(mLock); 5612 5613 track = new RecordTrack(this, client, sampleRate, 5614 format, channelMask, frameCount, NULL, sessionId, uid, 5615 *flags, TrackBase::TYPE_DEFAULT); 5616 5617 lStatus = track->initCheck(); 5618 if (lStatus != NO_ERROR) { 5619 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5620 // track must be cleared from the caller as the caller has the AF lock 5621 goto Exit; 5622 } 5623 mTracks.add(track); 5624 5625 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5626 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5627 mAudioFlinger->btNrecIsOff(); 5628 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5629 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5630 5631 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5632 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5633 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5634 // so ask activity manager to do this on our behalf 5635 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5636 } 5637 } 5638 5639 lStatus = NO_ERROR; 5640 5641Exit: 5642 *status = lStatus; 5643 return track; 5644} 5645 5646status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5647 AudioSystem::sync_event_t event, 5648 int triggerSession) 5649{ 5650 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5651 sp<ThreadBase> strongMe = this; 5652 status_t status = NO_ERROR; 5653 5654 if (event == AudioSystem::SYNC_EVENT_NONE) { 5655 recordTrack->clearSyncStartEvent(); 5656 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5657 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5658 triggerSession, 5659 recordTrack->sessionId(), 5660 syncStartEventCallback, 5661 recordTrack); 5662 // Sync event can be cancelled by the trigger session if the track is not in a 5663 // compatible state in which case we start record immediately 5664 if (recordTrack->mSyncStartEvent->isCancelled()) { 5665 recordTrack->clearSyncStartEvent(); 5666 } else { 5667 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5668 recordTrack->mFramesToDrop = - 5669 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5670 } 5671 } 5672 5673 { 5674 // This section is a rendezvous between binder thread executing start() and RecordThread 5675 AutoMutex lock(mLock); 5676 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5677 if (recordTrack->mState == TrackBase::PAUSING) { 5678 ALOGV("active record track PAUSING -> ACTIVE"); 5679 recordTrack->mState = TrackBase::ACTIVE; 5680 } else { 5681 ALOGV("active record track state %d", recordTrack->mState); 5682 } 5683 return status; 5684 } 5685 5686 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5687 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5688 // or using a separate command thread 5689 recordTrack->mState = TrackBase::STARTING_1; 5690 mActiveTracks.add(recordTrack); 5691 mActiveTracksGen++; 5692 status_t status = NO_ERROR; 5693 if (recordTrack->isExternalTrack()) { 5694 mLock.unlock(); 5695 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5696 mLock.lock(); 5697 // FIXME should verify that recordTrack is still in mActiveTracks 5698 if (status != NO_ERROR) { 5699 mActiveTracks.remove(recordTrack); 5700 mActiveTracksGen++; 5701 recordTrack->clearSyncStartEvent(); 5702 ALOGV("RecordThread::start error %d", status); 5703 return status; 5704 } 5705 } 5706 // Catch up with current buffer indices if thread is already running. 5707 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5708 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5709 // see previously buffered data before it called start(), but with greater risk of overrun. 5710 5711 recordTrack->mRsmpInFront = mRsmpInRear; 5712 recordTrack->mRsmpInUnrel = 0; 5713 // FIXME why reset? 5714 if (recordTrack->mResampler != NULL) { 5715 recordTrack->mResampler->reset(); 5716 } 5717 recordTrack->mState = TrackBase::STARTING_2; 5718 // signal thread to start 5719 mWaitWorkCV.broadcast(); 5720 if (mActiveTracks.indexOf(recordTrack) < 0) { 5721 ALOGV("Record failed to start"); 5722 status = BAD_VALUE; 5723 goto startError; 5724 } 5725 return status; 5726 } 5727 5728startError: 5729 if (recordTrack->isExternalTrack()) { 5730 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5731 } 5732 recordTrack->clearSyncStartEvent(); 5733 // FIXME I wonder why we do not reset the state here? 5734 return status; 5735} 5736 5737void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5738{ 5739 sp<SyncEvent> strongEvent = event.promote(); 5740 5741 if (strongEvent != 0) { 5742 sp<RefBase> ptr = strongEvent->cookie().promote(); 5743 if (ptr != 0) { 5744 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5745 recordTrack->handleSyncStartEvent(strongEvent); 5746 } 5747 } 5748} 5749 5750bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5751 ALOGV("RecordThread::stop"); 5752 AutoMutex _l(mLock); 5753 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5754 return false; 5755 } 5756 // note that threadLoop may still be processing the track at this point [without lock] 5757 recordTrack->mState = TrackBase::PAUSING; 5758 // do not wait for mStartStopCond if exiting 5759 if (exitPending()) { 5760 return true; 5761 } 5762 // FIXME incorrect usage of wait: no explicit predicate or loop 5763 mStartStopCond.wait(mLock); 5764 // if we have been restarted, recordTrack is in mActiveTracks here 5765 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5766 ALOGV("Record stopped OK"); 5767 return true; 5768 } 5769 return false; 5770} 5771 5772bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5773{ 5774 return false; 5775} 5776 5777status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5778{ 5779#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5780 if (!isValidSyncEvent(event)) { 5781 return BAD_VALUE; 5782 } 5783 5784 int eventSession = event->triggerSession(); 5785 status_t ret = NAME_NOT_FOUND; 5786 5787 Mutex::Autolock _l(mLock); 5788 5789 for (size_t i = 0; i < mTracks.size(); i++) { 5790 sp<RecordTrack> track = mTracks[i]; 5791 if (eventSession == track->sessionId()) { 5792 (void) track->setSyncEvent(event); 5793 ret = NO_ERROR; 5794 } 5795 } 5796 return ret; 5797#else 5798 return BAD_VALUE; 5799#endif 5800} 5801 5802// destroyTrack_l() must be called with ThreadBase::mLock held 5803void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5804{ 5805 track->terminate(); 5806 track->mState = TrackBase::STOPPED; 5807 // active tracks are removed by threadLoop() 5808 if (mActiveTracks.indexOf(track) < 0) { 5809 removeTrack_l(track); 5810 } 5811} 5812 5813void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5814{ 5815 mTracks.remove(track); 5816 // need anything related to effects here? 5817 if (track->isFastTrack()) { 5818 ALOG_ASSERT(!mFastTrackAvail); 5819 mFastTrackAvail = true; 5820 } 5821} 5822 5823void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5824{ 5825 dumpInternals(fd, args); 5826 dumpTracks(fd, args); 5827 dumpEffectChains(fd, args); 5828} 5829 5830void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5831{ 5832 dprintf(fd, "\nInput thread %p:\n", this); 5833 5834 if (mActiveTracks.size() > 0) { 5835 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5836 } else { 5837 dprintf(fd, " No active record clients\n"); 5838 } 5839 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5840 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5841 5842 dumpBase(fd, args); 5843} 5844 5845void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5846{ 5847 const size_t SIZE = 256; 5848 char buffer[SIZE]; 5849 String8 result; 5850 5851 size_t numtracks = mTracks.size(); 5852 size_t numactive = mActiveTracks.size(); 5853 size_t numactiveseen = 0; 5854 dprintf(fd, " %d Tracks", numtracks); 5855 if (numtracks) { 5856 dprintf(fd, " of which %d are active\n", numactive); 5857 RecordTrack::appendDumpHeader(result); 5858 for (size_t i = 0; i < numtracks ; ++i) { 5859 sp<RecordTrack> track = mTracks[i]; 5860 if (track != 0) { 5861 bool active = mActiveTracks.indexOf(track) >= 0; 5862 if (active) { 5863 numactiveseen++; 5864 } 5865 track->dump(buffer, SIZE, active); 5866 result.append(buffer); 5867 } 5868 } 5869 } else { 5870 dprintf(fd, "\n"); 5871 } 5872 5873 if (numactiveseen != numactive) { 5874 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5875 " not in the track list\n"); 5876 result.append(buffer); 5877 RecordTrack::appendDumpHeader(result); 5878 for (size_t i = 0; i < numactive; ++i) { 5879 sp<RecordTrack> track = mActiveTracks[i]; 5880 if (mTracks.indexOf(track) < 0) { 5881 track->dump(buffer, SIZE, true); 5882 result.append(buffer); 5883 } 5884 } 5885 5886 } 5887 write(fd, result.string(), result.size()); 5888} 5889 5890// AudioBufferProvider interface 5891status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5892 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5893{ 5894 RecordTrack *activeTrack = mRecordTrack; 5895 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5896 if (threadBase == 0) { 5897 buffer->frameCount = 0; 5898 buffer->raw = NULL; 5899 return NOT_ENOUGH_DATA; 5900 } 5901 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5902 int32_t rear = recordThread->mRsmpInRear; 5903 int32_t front = activeTrack->mRsmpInFront; 5904 ssize_t filled = rear - front; 5905 // FIXME should not be P2 (don't want to increase latency) 5906 // FIXME if client not keeping up, discard 5907 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5908 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5909 front &= recordThread->mRsmpInFramesP2 - 1; 5910 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5911 if (part1 > (size_t) filled) { 5912 part1 = filled; 5913 } 5914 size_t ask = buffer->frameCount; 5915 ALOG_ASSERT(ask > 0); 5916 if (part1 > ask) { 5917 part1 = ask; 5918 } 5919 if (part1 == 0) { 5920 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5921 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5922 buffer->raw = NULL; 5923 buffer->frameCount = 0; 5924 activeTrack->mRsmpInUnrel = 0; 5925 return NOT_ENOUGH_DATA; 5926 } 5927 5928 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5929 buffer->frameCount = part1; 5930 activeTrack->mRsmpInUnrel = part1; 5931 return NO_ERROR; 5932} 5933 5934// AudioBufferProvider interface 5935void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5936 AudioBufferProvider::Buffer* buffer) 5937{ 5938 RecordTrack *activeTrack = mRecordTrack; 5939 size_t stepCount = buffer->frameCount; 5940 if (stepCount == 0) { 5941 return; 5942 } 5943 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5944 activeTrack->mRsmpInUnrel -= stepCount; 5945 activeTrack->mRsmpInFront += stepCount; 5946 buffer->raw = NULL; 5947 buffer->frameCount = 0; 5948} 5949 5950bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5951 status_t& status) 5952{ 5953 bool reconfig = false; 5954 5955 status = NO_ERROR; 5956 5957 audio_format_t reqFormat = mFormat; 5958 uint32_t samplingRate = mSampleRate; 5959 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5960 5961 AudioParameter param = AudioParameter(keyValuePair); 5962 int value; 5963 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5964 // channel count change can be requested. Do we mandate the first client defines the 5965 // HAL sampling rate and channel count or do we allow changes on the fly? 5966 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5967 samplingRate = value; 5968 reconfig = true; 5969 } 5970 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5971 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5972 status = BAD_VALUE; 5973 } else { 5974 reqFormat = (audio_format_t) value; 5975 reconfig = true; 5976 } 5977 } 5978 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5979 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5980 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5981 status = BAD_VALUE; 5982 } else { 5983 channelMask = mask; 5984 reconfig = true; 5985 } 5986 } 5987 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5988 // do not accept frame count changes if tracks are open as the track buffer 5989 // size depends on frame count and correct behavior would not be guaranteed 5990 // if frame count is changed after track creation 5991 if (mActiveTracks.size() > 0) { 5992 status = INVALID_OPERATION; 5993 } else { 5994 reconfig = true; 5995 } 5996 } 5997 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5998 // forward device change to effects that have requested to be 5999 // aware of attached audio device. 6000 for (size_t i = 0; i < mEffectChains.size(); i++) { 6001 mEffectChains[i]->setDevice_l(value); 6002 } 6003 6004 // store input device and output device but do not forward output device to audio HAL. 6005 // Note that status is ignored by the caller for output device 6006 // (see AudioFlinger::setParameters() 6007 if (audio_is_output_devices(value)) { 6008 mOutDevice = value; 6009 status = BAD_VALUE; 6010 } else { 6011 mInDevice = value; 6012 // disable AEC and NS if the device is a BT SCO headset supporting those 6013 // pre processings 6014 if (mTracks.size() > 0) { 6015 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6016 mAudioFlinger->btNrecIsOff(); 6017 for (size_t i = 0; i < mTracks.size(); i++) { 6018 sp<RecordTrack> track = mTracks[i]; 6019 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6020 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6021 } 6022 } 6023 } 6024 } 6025 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6026 mAudioSource != (audio_source_t)value) { 6027 // forward device change to effects that have requested to be 6028 // aware of attached audio device. 6029 for (size_t i = 0; i < mEffectChains.size(); i++) { 6030 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6031 } 6032 mAudioSource = (audio_source_t)value; 6033 } 6034 6035 if (status == NO_ERROR) { 6036 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6037 keyValuePair.string()); 6038 if (status == INVALID_OPERATION) { 6039 inputStandBy(); 6040 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6041 keyValuePair.string()); 6042 } 6043 if (reconfig) { 6044 if (status == BAD_VALUE && 6045 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6046 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6047 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6048 <= (2 * samplingRate)) && 6049 audio_channel_count_from_in_mask( 6050 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6051 (channelMask == AUDIO_CHANNEL_IN_MONO || 6052 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6053 status = NO_ERROR; 6054 } 6055 if (status == NO_ERROR) { 6056 readInputParameters_l(); 6057 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6058 } 6059 } 6060 } 6061 6062 return reconfig; 6063} 6064 6065String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6066{ 6067 Mutex::Autolock _l(mLock); 6068 if (initCheck() != NO_ERROR) { 6069 return String8(); 6070 } 6071 6072 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6073 const String8 out_s8(s); 6074 free(s); 6075 return out_s8; 6076} 6077 6078void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6079 AudioSystem::OutputDescriptor desc; 6080 const void *param2 = NULL; 6081 6082 switch (event) { 6083 case AudioSystem::INPUT_OPENED: 6084 case AudioSystem::INPUT_CONFIG_CHANGED: 6085 desc.channelMask = mChannelMask; 6086 desc.samplingRate = mSampleRate; 6087 desc.format = mFormat; 6088 desc.frameCount = mFrameCount; 6089 desc.latency = 0; 6090 param2 = &desc; 6091 break; 6092 6093 case AudioSystem::INPUT_CLOSED: 6094 default: 6095 break; 6096 } 6097 mAudioFlinger->audioConfigChanged(event, mId, param2); 6098} 6099 6100void AudioFlinger::RecordThread::readInputParameters_l() 6101{ 6102 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6103 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6104 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6105 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6106 mFormat = mHALFormat; 6107 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6108 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6109 } 6110 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6111 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6112 mFrameCount = mBufferSize / mFrameSize; 6113 // This is the formula for calculating the temporary buffer size. 6114 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6115 // 1 full output buffer, regardless of the alignment of the available input. 6116 // The value is somewhat arbitrary, and could probably be even larger. 6117 // A larger value should allow more old data to be read after a track calls start(), 6118 // without increasing latency. 6119 mRsmpInFrames = mFrameCount * 7; 6120 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6121 delete[] mRsmpInBuffer; 6122 6123 // TODO optimize audio capture buffer sizes ... 6124 // Here we calculate the size of the sliding buffer used as a source 6125 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6126 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6127 // be better to have it derived from the pipe depth in the long term. 6128 // The current value is higher than necessary. However it should not add to latency. 6129 6130 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6131 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6132 6133 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6134 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6135} 6136 6137uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6138{ 6139 Mutex::Autolock _l(mLock); 6140 if (initCheck() != NO_ERROR) { 6141 return 0; 6142 } 6143 6144 return mInput->stream->get_input_frames_lost(mInput->stream); 6145} 6146 6147uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6148{ 6149 Mutex::Autolock _l(mLock); 6150 uint32_t result = 0; 6151 if (getEffectChain_l(sessionId) != 0) { 6152 result = EFFECT_SESSION; 6153 } 6154 6155 for (size_t i = 0; i < mTracks.size(); ++i) { 6156 if (sessionId == mTracks[i]->sessionId()) { 6157 result |= TRACK_SESSION; 6158 break; 6159 } 6160 } 6161 6162 return result; 6163} 6164 6165KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6166{ 6167 KeyedVector<int, bool> ids; 6168 Mutex::Autolock _l(mLock); 6169 for (size_t j = 0; j < mTracks.size(); ++j) { 6170 sp<RecordThread::RecordTrack> track = mTracks[j]; 6171 int sessionId = track->sessionId(); 6172 if (ids.indexOfKey(sessionId) < 0) { 6173 ids.add(sessionId, true); 6174 } 6175 } 6176 return ids; 6177} 6178 6179AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6180{ 6181 Mutex::Autolock _l(mLock); 6182 AudioStreamIn *input = mInput; 6183 mInput = NULL; 6184 return input; 6185} 6186 6187// this method must always be called either with ThreadBase mLock held or inside the thread loop 6188audio_stream_t* AudioFlinger::RecordThread::stream() const 6189{ 6190 if (mInput == NULL) { 6191 return NULL; 6192 } 6193 return &mInput->stream->common; 6194} 6195 6196status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6197{ 6198 // only one chain per input thread 6199 if (mEffectChains.size() != 0) { 6200 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6201 return INVALID_OPERATION; 6202 } 6203 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6204 chain->setThread(this); 6205 chain->setInBuffer(NULL); 6206 chain->setOutBuffer(NULL); 6207 6208 checkSuspendOnAddEffectChain_l(chain); 6209 6210 // make sure enabled pre processing effects state is communicated to the HAL as we 6211 // just moved them to a new input stream. 6212 chain->syncHalEffectsState(); 6213 6214 mEffectChains.add(chain); 6215 6216 return NO_ERROR; 6217} 6218 6219size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6220{ 6221 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6222 ALOGW_IF(mEffectChains.size() != 1, 6223 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6224 chain.get(), mEffectChains.size(), this); 6225 if (mEffectChains.size() == 1) { 6226 mEffectChains.removeAt(0); 6227 } 6228 return 0; 6229} 6230 6231status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6232 audio_patch_handle_t *handle) 6233{ 6234 status_t status = NO_ERROR; 6235 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6236 // store new device and send to effects 6237 mInDevice = patch->sources[0].ext.device.type; 6238 for (size_t i = 0; i < mEffectChains.size(); i++) { 6239 mEffectChains[i]->setDevice_l(mInDevice); 6240 } 6241 6242 // disable AEC and NS if the device is a BT SCO headset supporting those 6243 // pre processings 6244 if (mTracks.size() > 0) { 6245 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6246 mAudioFlinger->btNrecIsOff(); 6247 for (size_t i = 0; i < mTracks.size(); i++) { 6248 sp<RecordTrack> track = mTracks[i]; 6249 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6250 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6251 } 6252 } 6253 6254 // store new source and send to effects 6255 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6256 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6257 for (size_t i = 0; i < mEffectChains.size(); i++) { 6258 mEffectChains[i]->setAudioSource_l(mAudioSource); 6259 } 6260 } 6261 6262 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6263 status = hwDevice->create_audio_patch(hwDevice, 6264 patch->num_sources, 6265 patch->sources, 6266 patch->num_sinks, 6267 patch->sinks, 6268 handle); 6269 } else { 6270 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6271 } 6272 return status; 6273} 6274 6275status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6276{ 6277 status_t status = NO_ERROR; 6278 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6279 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6280 status = hwDevice->release_audio_patch(hwDevice, handle); 6281 } else { 6282 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6283 } 6284 return status; 6285} 6286 6287void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6288{ 6289 Mutex::Autolock _l(mLock); 6290 mTracks.add(record); 6291} 6292 6293void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6294{ 6295 Mutex::Autolock _l(mLock); 6296 destroyTrack_l(record); 6297} 6298 6299void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6300{ 6301 ThreadBase::getAudioPortConfig(config); 6302 config->role = AUDIO_PORT_ROLE_SINK; 6303 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6304 config->ext.mix.usecase.source = mAudioSource; 6305} 6306 6307}; // namespace android 6308