Threads.cpp revision aaa44478a373232d8416657035a9020f9c7aa7c3
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 const size_t count = mBytesRemaining / mFrameSize; 2105 2106 ATRACE_BEGIN("write"); 2107 // update the setpoint when AudioFlinger::mScreenState changes 2108 uint32_t screenState = AudioFlinger::mScreenState; 2109 if (screenState != mScreenState) { 2110 mScreenState = screenState; 2111 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2112 if (pipe != NULL) { 2113 pipe->setAvgFrames((mScreenState & 1) ? 2114 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2115 } 2116 } 2117 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2118 ATRACE_END(); 2119 if (framesWritten > 0) { 2120 bytesWritten = framesWritten * mFrameSize; 2121 } else { 2122 bytesWritten = framesWritten; 2123 } 2124 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2125 if (status == NO_ERROR) { 2126 size_t totalFramesWritten = mNormalSink->framesWritten(); 2127 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2128 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2129 mLatchDValid = true; 2130 } 2131 } 2132 // otherwise use the HAL / AudioStreamOut directly 2133 } else { 2134 // Direct output and offload threads 2135 2136 if (mUseAsyncWrite) { 2137 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2138 mWriteAckSequence += 2; 2139 mWriteAckSequence |= 1; 2140 ALOG_ASSERT(mCallbackThread != 0); 2141 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2142 } 2143 // FIXME We should have an implementation of timestamps for direct output threads. 2144 // They are used e.g for multichannel PCM playback over HDMI. 2145 bytesWritten = mOutput->stream->write(mOutput->stream, 2146 (char *)mSinkBuffer + offset, mBytesRemaining); 2147 if (mUseAsyncWrite && 2148 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2149 // do not wait for async callback in case of error of full write 2150 mWriteAckSequence &= ~1; 2151 ALOG_ASSERT(mCallbackThread != 0); 2152 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2153 } 2154 } 2155 2156 mNumWrites++; 2157 mInWrite = false; 2158 mStandby = false; 2159 return bytesWritten; 2160} 2161 2162void AudioFlinger::PlaybackThread::threadLoop_drain() 2163{ 2164 if (mOutput->stream->drain) { 2165 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2166 if (mUseAsyncWrite) { 2167 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2168 mDrainSequence |= 1; 2169 ALOG_ASSERT(mCallbackThread != 0); 2170 mCallbackThread->setDraining(mDrainSequence); 2171 } 2172 mOutput->stream->drain(mOutput->stream, 2173 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2174 : AUDIO_DRAIN_ALL); 2175 } 2176} 2177 2178void AudioFlinger::PlaybackThread::threadLoop_exit() 2179{ 2180 // Default implementation has nothing to do 2181} 2182 2183/* 2184The derived values that are cached: 2185 - mSinkBufferSize from frame count * frame size 2186 - activeSleepTime from activeSleepTimeUs() 2187 - idleSleepTime from idleSleepTimeUs() 2188 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2189 - maxPeriod from frame count and sample rate (MIXER only) 2190 2191The parameters that affect these derived values are: 2192 - frame count 2193 - frame size 2194 - sample rate 2195 - device type: A2DP or not 2196 - device latency 2197 - format: PCM or not 2198 - active sleep time 2199 - idle sleep time 2200*/ 2201 2202void AudioFlinger::PlaybackThread::cacheParameters_l() 2203{ 2204 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2205 activeSleepTime = activeSleepTimeUs(); 2206 idleSleepTime = idleSleepTimeUs(); 2207} 2208 2209void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2210{ 2211 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2212 this, streamType, mTracks.size()); 2213 Mutex::Autolock _l(mLock); 2214 2215 size_t size = mTracks.size(); 2216 for (size_t i = 0; i < size; i++) { 2217 sp<Track> t = mTracks[i]; 2218 if (t->streamType() == streamType) { 2219 t->invalidate(); 2220 } 2221 } 2222} 2223 2224status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2225{ 2226 int session = chain->sessionId(); 2227 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2228 ? mEffectBuffer : mSinkBuffer); 2229 bool ownsBuffer = false; 2230 2231 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2232 if (session > 0) { 2233 // Only one effect chain can be present in direct output thread and it uses 2234 // the sink buffer as input 2235 if (mType != DIRECT) { 2236 size_t numSamples = mNormalFrameCount * mChannelCount; 2237 buffer = new int16_t[numSamples]; 2238 memset(buffer, 0, numSamples * sizeof(int16_t)); 2239 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2240 ownsBuffer = true; 2241 } 2242 2243 // Attach all tracks with same session ID to this chain. 2244 for (size_t i = 0; i < mTracks.size(); ++i) { 2245 sp<Track> track = mTracks[i]; 2246 if (session == track->sessionId()) { 2247 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2248 buffer); 2249 track->setMainBuffer(buffer); 2250 chain->incTrackCnt(); 2251 } 2252 } 2253 2254 // indicate all active tracks in the chain 2255 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2256 sp<Track> track = mActiveTracks[i].promote(); 2257 if (track == 0) { 2258 continue; 2259 } 2260 if (session == track->sessionId()) { 2261 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2262 chain->incActiveTrackCnt(); 2263 } 2264 } 2265 } 2266 chain->setThread(this); 2267 chain->setInBuffer(buffer, ownsBuffer); 2268 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2269 ? mEffectBuffer : mSinkBuffer)); 2270 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2271 // chains list in order to be processed last as it contains output stage effects 2272 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2273 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2274 // after track specific effects and before output stage 2275 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2276 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2277 // Effect chain for other sessions are inserted at beginning of effect 2278 // chains list to be processed before output mix effects. Relative order between other 2279 // sessions is not important 2280 size_t size = mEffectChains.size(); 2281 size_t i = 0; 2282 for (i = 0; i < size; i++) { 2283 if (mEffectChains[i]->sessionId() < session) { 2284 break; 2285 } 2286 } 2287 mEffectChains.insertAt(chain, i); 2288 checkSuspendOnAddEffectChain_l(chain); 2289 2290 return NO_ERROR; 2291} 2292 2293size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2294{ 2295 int session = chain->sessionId(); 2296 2297 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2298 2299 for (size_t i = 0; i < mEffectChains.size(); i++) { 2300 if (chain == mEffectChains[i]) { 2301 mEffectChains.removeAt(i); 2302 // detach all active tracks from the chain 2303 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2304 sp<Track> track = mActiveTracks[i].promote(); 2305 if (track == 0) { 2306 continue; 2307 } 2308 if (session == track->sessionId()) { 2309 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2310 chain.get(), session); 2311 chain->decActiveTrackCnt(); 2312 } 2313 } 2314 2315 // detach all tracks with same session ID from this chain 2316 for (size_t i = 0; i < mTracks.size(); ++i) { 2317 sp<Track> track = mTracks[i]; 2318 if (session == track->sessionId()) { 2319 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2320 chain->decTrackCnt(); 2321 } 2322 } 2323 break; 2324 } 2325 } 2326 return mEffectChains.size(); 2327} 2328 2329status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2330 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2331{ 2332 Mutex::Autolock _l(mLock); 2333 return attachAuxEffect_l(track, EffectId); 2334} 2335 2336status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2337 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2338{ 2339 status_t status = NO_ERROR; 2340 2341 if (EffectId == 0) { 2342 track->setAuxBuffer(0, NULL); 2343 } else { 2344 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2345 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2346 if (effect != 0) { 2347 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2348 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2349 } else { 2350 status = INVALID_OPERATION; 2351 } 2352 } else { 2353 status = BAD_VALUE; 2354 } 2355 } 2356 return status; 2357} 2358 2359void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2360{ 2361 for (size_t i = 0; i < mTracks.size(); ++i) { 2362 sp<Track> track = mTracks[i]; 2363 if (track->auxEffectId() == effectId) { 2364 attachAuxEffect_l(track, 0); 2365 } 2366 } 2367} 2368 2369bool AudioFlinger::PlaybackThread::threadLoop() 2370{ 2371 Vector< sp<Track> > tracksToRemove; 2372 2373 standbyTime = systemTime(); 2374 2375 // MIXER 2376 nsecs_t lastWarning = 0; 2377 2378 // DUPLICATING 2379 // FIXME could this be made local to while loop? 2380 writeFrames = 0; 2381 2382 int lastGeneration = 0; 2383 2384 cacheParameters_l(); 2385 sleepTime = idleSleepTime; 2386 2387 if (mType == MIXER) { 2388 sleepTimeShift = 0; 2389 } 2390 2391 CpuStats cpuStats; 2392 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2393 2394 acquireWakeLock(); 2395 2396 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2397 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2398 // and then that string will be logged at the next convenient opportunity. 2399 const char *logString = NULL; 2400 2401 checkSilentMode_l(); 2402 2403 while (!exitPending()) 2404 { 2405 cpuStats.sample(myName); 2406 2407 Vector< sp<EffectChain> > effectChains; 2408 2409 { // scope for mLock 2410 2411 Mutex::Autolock _l(mLock); 2412 2413 processConfigEvents_l(); 2414 2415 if (logString != NULL) { 2416 mNBLogWriter->logTimestamp(); 2417 mNBLogWriter->log(logString); 2418 logString = NULL; 2419 } 2420 2421 if (mLatchDValid) { 2422 mLatchQ = mLatchD; 2423 mLatchDValid = false; 2424 mLatchQValid = true; 2425 } 2426 2427 saveOutputTracks(); 2428 if (mSignalPending) { 2429 // A signal was raised while we were unlocked 2430 mSignalPending = false; 2431 } else if (waitingAsyncCallback_l()) { 2432 if (exitPending()) { 2433 break; 2434 } 2435 releaseWakeLock_l(); 2436 mWakeLockUids.clear(); 2437 mActiveTracksGeneration++; 2438 ALOGV("wait async completion"); 2439 mWaitWorkCV.wait(mLock); 2440 ALOGV("async completion/wake"); 2441 acquireWakeLock_l(); 2442 standbyTime = systemTime() + standbyDelay; 2443 sleepTime = 0; 2444 2445 continue; 2446 } 2447 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2448 isSuspended()) { 2449 // put audio hardware into standby after short delay 2450 if (shouldStandby_l()) { 2451 2452 threadLoop_standby(); 2453 2454 mStandby = true; 2455 } 2456 2457 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2458 // we're about to wait, flush the binder command buffer 2459 IPCThreadState::self()->flushCommands(); 2460 2461 clearOutputTracks(); 2462 2463 if (exitPending()) { 2464 break; 2465 } 2466 2467 releaseWakeLock_l(); 2468 mWakeLockUids.clear(); 2469 mActiveTracksGeneration++; 2470 // wait until we have something to do... 2471 ALOGV("%s going to sleep", myName.string()); 2472 mWaitWorkCV.wait(mLock); 2473 ALOGV("%s waking up", myName.string()); 2474 acquireWakeLock_l(); 2475 2476 mMixerStatus = MIXER_IDLE; 2477 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2478 mBytesWritten = 0; 2479 mBytesRemaining = 0; 2480 checkSilentMode_l(); 2481 2482 standbyTime = systemTime() + standbyDelay; 2483 sleepTime = idleSleepTime; 2484 if (mType == MIXER) { 2485 sleepTimeShift = 0; 2486 } 2487 2488 continue; 2489 } 2490 } 2491 // mMixerStatusIgnoringFastTracks is also updated internally 2492 mMixerStatus = prepareTracks_l(&tracksToRemove); 2493 2494 // compare with previously applied list 2495 if (lastGeneration != mActiveTracksGeneration) { 2496 // update wakelock 2497 updateWakeLockUids_l(mWakeLockUids); 2498 lastGeneration = mActiveTracksGeneration; 2499 } 2500 2501 // prevent any changes in effect chain list and in each effect chain 2502 // during mixing and effect process as the audio buffers could be deleted 2503 // or modified if an effect is created or deleted 2504 lockEffectChains_l(effectChains); 2505 } // mLock scope ends 2506 2507 if (mBytesRemaining == 0) { 2508 mCurrentWriteLength = 0; 2509 if (mMixerStatus == MIXER_TRACKS_READY) { 2510 // threadLoop_mix() sets mCurrentWriteLength 2511 threadLoop_mix(); 2512 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2513 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2514 // threadLoop_sleepTime sets sleepTime to 0 if data 2515 // must be written to HAL 2516 threadLoop_sleepTime(); 2517 if (sleepTime == 0) { 2518 mCurrentWriteLength = mSinkBufferSize; 2519 } 2520 } 2521 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2522 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2523 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2524 // or mSinkBuffer (if there are no effects). 2525 // 2526 // This is done pre-effects computation; if effects change to 2527 // support higher precision, this needs to move. 2528 // 2529 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2530 // TODO use sleepTime == 0 as an additional condition. 2531 if (mMixerBufferValid) { 2532 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2533 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2534 2535 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2536 mNormalFrameCount * mChannelCount); 2537 } 2538 2539 mBytesRemaining = mCurrentWriteLength; 2540 if (isSuspended()) { 2541 sleepTime = suspendSleepTimeUs(); 2542 // simulate write to HAL when suspended 2543 mBytesWritten += mSinkBufferSize; 2544 mBytesRemaining = 0; 2545 } 2546 2547 // only process effects if we're going to write 2548 if (sleepTime == 0 && mType != OFFLOAD) { 2549 for (size_t i = 0; i < effectChains.size(); i ++) { 2550 effectChains[i]->process_l(); 2551 } 2552 } 2553 } 2554 // Process effect chains for offloaded thread even if no audio 2555 // was read from audio track: process only updates effect state 2556 // and thus does have to be synchronized with audio writes but may have 2557 // to be called while waiting for async write callback 2558 if (mType == OFFLOAD) { 2559 for (size_t i = 0; i < effectChains.size(); i ++) { 2560 effectChains[i]->process_l(); 2561 } 2562 } 2563 2564 // Only if the Effects buffer is enabled and there is data in the 2565 // Effects buffer (buffer valid), we need to 2566 // copy into the sink buffer. 2567 // TODO use sleepTime == 0 as an additional condition. 2568 if (mEffectBufferValid) { 2569 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2570 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2571 mNormalFrameCount * mChannelCount); 2572 } 2573 2574 // enable changes in effect chain 2575 unlockEffectChains(effectChains); 2576 2577 if (!waitingAsyncCallback()) { 2578 // sleepTime == 0 means we must write to audio hardware 2579 if (sleepTime == 0) { 2580 if (mBytesRemaining) { 2581 ssize_t ret = threadLoop_write(); 2582 if (ret < 0) { 2583 mBytesRemaining = 0; 2584 } else { 2585 mBytesWritten += ret; 2586 mBytesRemaining -= ret; 2587 } 2588 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2589 (mMixerStatus == MIXER_DRAIN_ALL)) { 2590 threadLoop_drain(); 2591 } 2592 if (mType == MIXER) { 2593 // write blocked detection 2594 nsecs_t now = systemTime(); 2595 nsecs_t delta = now - mLastWriteTime; 2596 if (!mStandby && delta > maxPeriod) { 2597 mNumDelayedWrites++; 2598 if ((now - lastWarning) > kWarningThrottleNs) { 2599 ATRACE_NAME("underrun"); 2600 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2601 ns2ms(delta), mNumDelayedWrites, this); 2602 lastWarning = now; 2603 } 2604 } 2605 } 2606 2607 } else { 2608 usleep(sleepTime); 2609 } 2610 } 2611 2612 // Finally let go of removed track(s), without the lock held 2613 // since we can't guarantee the destructors won't acquire that 2614 // same lock. This will also mutate and push a new fast mixer state. 2615 threadLoop_removeTracks(tracksToRemove); 2616 tracksToRemove.clear(); 2617 2618 // FIXME I don't understand the need for this here; 2619 // it was in the original code but maybe the 2620 // assignment in saveOutputTracks() makes this unnecessary? 2621 clearOutputTracks(); 2622 2623 // Effect chains will be actually deleted here if they were removed from 2624 // mEffectChains list during mixing or effects processing 2625 effectChains.clear(); 2626 2627 // FIXME Note that the above .clear() is no longer necessary since effectChains 2628 // is now local to this block, but will keep it for now (at least until merge done). 2629 } 2630 2631 threadLoop_exit(); 2632 2633 if (!mStandby) { 2634 threadLoop_standby(); 2635 mStandby = true; 2636 } 2637 2638 releaseWakeLock(); 2639 mWakeLockUids.clear(); 2640 mActiveTracksGeneration++; 2641 2642 ALOGV("Thread %p type %d exiting", this, mType); 2643 return false; 2644} 2645 2646// removeTracks_l() must be called with ThreadBase::mLock held 2647void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2648{ 2649 size_t count = tracksToRemove.size(); 2650 if (count > 0) { 2651 for (size_t i=0 ; i<count ; i++) { 2652 const sp<Track>& track = tracksToRemove.itemAt(i); 2653 mActiveTracks.remove(track); 2654 mWakeLockUids.remove(track->uid()); 2655 mActiveTracksGeneration++; 2656 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2657 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2658 if (chain != 0) { 2659 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2660 track->sessionId()); 2661 chain->decActiveTrackCnt(); 2662 } 2663 if (track->isTerminated()) { 2664 removeTrack_l(track); 2665 } 2666 } 2667 } 2668 2669} 2670 2671status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2672{ 2673 if (mNormalSink != 0) { 2674 return mNormalSink->getTimestamp(timestamp); 2675 } 2676 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2677 uint64_t position64; 2678 int ret = mOutput->stream->get_presentation_position( 2679 mOutput->stream, &position64, ×tamp.mTime); 2680 if (ret == 0) { 2681 timestamp.mPosition = (uint32_t)position64; 2682 return NO_ERROR; 2683 } 2684 } 2685 return INVALID_OPERATION; 2686} 2687 2688status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2689 audio_patch_handle_t *handle) 2690{ 2691 status_t status = NO_ERROR; 2692 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2693 // store new device and send to effects 2694 audio_devices_t type = AUDIO_DEVICE_NONE; 2695 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2696 type |= patch->sinks[i].ext.device.type; 2697 } 2698 mOutDevice = type; 2699 for (size_t i = 0; i < mEffectChains.size(); i++) { 2700 mEffectChains[i]->setDevice_l(mOutDevice); 2701 } 2702 2703 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2704 status = hwDevice->create_audio_patch(hwDevice, 2705 patch->num_sources, 2706 patch->sources, 2707 patch->num_sinks, 2708 patch->sinks, 2709 handle); 2710 } else { 2711 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2712 } 2713 return status; 2714} 2715 2716status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2717{ 2718 status_t status = NO_ERROR; 2719 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2720 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2721 status = hwDevice->release_audio_patch(hwDevice, handle); 2722 } else { 2723 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2724 } 2725 return status; 2726} 2727 2728void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2729{ 2730 Mutex::Autolock _l(mLock); 2731 mTracks.add(track); 2732} 2733 2734void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2735{ 2736 Mutex::Autolock _l(mLock); 2737 destroyTrack_l(track); 2738} 2739 2740void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2741{ 2742 ThreadBase::getAudioPortConfig(config); 2743 config->role = AUDIO_PORT_ROLE_SOURCE; 2744 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2745 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2746} 2747 2748// ---------------------------------------------------------------------------- 2749 2750AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2751 audio_io_handle_t id, audio_devices_t device, type_t type) 2752 : PlaybackThread(audioFlinger, output, id, device, type), 2753 // mAudioMixer below 2754 // mFastMixer below 2755 mFastMixerFutex(0) 2756 // mOutputSink below 2757 // mPipeSink below 2758 // mNormalSink below 2759{ 2760 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2761 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2762 "mFrameCount=%d, mNormalFrameCount=%d", 2763 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2764 mNormalFrameCount); 2765 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2766 2767 // create an NBAIO sink for the HAL output stream, and negotiate 2768 mOutputSink = new AudioStreamOutSink(output->stream); 2769 size_t numCounterOffers = 0; 2770 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2771 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2772 ALOG_ASSERT(index == 0); 2773 2774 // initialize fast mixer depending on configuration 2775 bool initFastMixer; 2776 switch (kUseFastMixer) { 2777 case FastMixer_Never: 2778 initFastMixer = false; 2779 break; 2780 case FastMixer_Always: 2781 initFastMixer = true; 2782 break; 2783 case FastMixer_Static: 2784 case FastMixer_Dynamic: 2785 initFastMixer = mFrameCount < mNormalFrameCount; 2786 break; 2787 } 2788 if (initFastMixer) { 2789 audio_format_t fastMixerFormat; 2790 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2791 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2792 } else { 2793 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2794 } 2795 if (mFormat != fastMixerFormat) { 2796 // change our Sink format to accept our intermediate precision 2797 mFormat = fastMixerFormat; 2798 free(mSinkBuffer); 2799 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2800 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2801 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2802 } 2803 2804 // create a MonoPipe to connect our submix to FastMixer 2805 NBAIO_Format format = mOutputSink->format(); 2806 // adjust format to match that of the Fast Mixer 2807 format.mFormat = fastMixerFormat; 2808 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2809 2810 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2811 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2812 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2813 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2814 const NBAIO_Format offers[1] = {format}; 2815 size_t numCounterOffers = 0; 2816 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2817 ALOG_ASSERT(index == 0); 2818 monoPipe->setAvgFrames((mScreenState & 1) ? 2819 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2820 mPipeSink = monoPipe; 2821 2822#ifdef TEE_SINK 2823 if (mTeeSinkOutputEnabled) { 2824 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2825 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2826 numCounterOffers = 0; 2827 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2828 ALOG_ASSERT(index == 0); 2829 mTeeSink = teeSink; 2830 PipeReader *teeSource = new PipeReader(*teeSink); 2831 numCounterOffers = 0; 2832 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2833 ALOG_ASSERT(index == 0); 2834 mTeeSource = teeSource; 2835 } 2836#endif 2837 2838 // create fast mixer and configure it initially with just one fast track for our submix 2839 mFastMixer = new FastMixer(); 2840 FastMixerStateQueue *sq = mFastMixer->sq(); 2841#ifdef STATE_QUEUE_DUMP 2842 sq->setObserverDump(&mStateQueueObserverDump); 2843 sq->setMutatorDump(&mStateQueueMutatorDump); 2844#endif 2845 FastMixerState *state = sq->begin(); 2846 FastTrack *fastTrack = &state->mFastTracks[0]; 2847 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2848 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2849 fastTrack->mVolumeProvider = NULL; 2850 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2851 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2852 fastTrack->mGeneration++; 2853 state->mFastTracksGen++; 2854 state->mTrackMask = 1; 2855 // fast mixer will use the HAL output sink 2856 state->mOutputSink = mOutputSink.get(); 2857 state->mOutputSinkGen++; 2858 state->mFrameCount = mFrameCount; 2859 state->mCommand = FastMixerState::COLD_IDLE; 2860 // already done in constructor initialization list 2861 //mFastMixerFutex = 0; 2862 state->mColdFutexAddr = &mFastMixerFutex; 2863 state->mColdGen++; 2864 state->mDumpState = &mFastMixerDumpState; 2865#ifdef TEE_SINK 2866 state->mTeeSink = mTeeSink.get(); 2867#endif 2868 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2869 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2870 sq->end(); 2871 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2872 2873 // start the fast mixer 2874 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2875 pid_t tid = mFastMixer->getTid(); 2876 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2877 if (err != 0) { 2878 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2879 kPriorityFastMixer, getpid_cached, tid, err); 2880 } 2881 2882#ifdef AUDIO_WATCHDOG 2883 // create and start the watchdog 2884 mAudioWatchdog = new AudioWatchdog(); 2885 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2886 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2887 tid = mAudioWatchdog->getTid(); 2888 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2889 if (err != 0) { 2890 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2891 kPriorityFastMixer, getpid_cached, tid, err); 2892 } 2893#endif 2894 2895 } 2896 2897 switch (kUseFastMixer) { 2898 case FastMixer_Never: 2899 case FastMixer_Dynamic: 2900 mNormalSink = mOutputSink; 2901 break; 2902 case FastMixer_Always: 2903 mNormalSink = mPipeSink; 2904 break; 2905 case FastMixer_Static: 2906 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2907 break; 2908 } 2909} 2910 2911AudioFlinger::MixerThread::~MixerThread() 2912{ 2913 if (mFastMixer != 0) { 2914 FastMixerStateQueue *sq = mFastMixer->sq(); 2915 FastMixerState *state = sq->begin(); 2916 if (state->mCommand == FastMixerState::COLD_IDLE) { 2917 int32_t old = android_atomic_inc(&mFastMixerFutex); 2918 if (old == -1) { 2919 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2920 } 2921 } 2922 state->mCommand = FastMixerState::EXIT; 2923 sq->end(); 2924 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2925 mFastMixer->join(); 2926 // Though the fast mixer thread has exited, it's state queue is still valid. 2927 // We'll use that extract the final state which contains one remaining fast track 2928 // corresponding to our sub-mix. 2929 state = sq->begin(); 2930 ALOG_ASSERT(state->mTrackMask == 1); 2931 FastTrack *fastTrack = &state->mFastTracks[0]; 2932 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2933 delete fastTrack->mBufferProvider; 2934 sq->end(false /*didModify*/); 2935 mFastMixer.clear(); 2936#ifdef AUDIO_WATCHDOG 2937 if (mAudioWatchdog != 0) { 2938 mAudioWatchdog->requestExit(); 2939 mAudioWatchdog->requestExitAndWait(); 2940 mAudioWatchdog.clear(); 2941 } 2942#endif 2943 } 2944 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2945 delete mAudioMixer; 2946} 2947 2948 2949uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2950{ 2951 if (mFastMixer != 0) { 2952 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2953 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2954 } 2955 return latency; 2956} 2957 2958 2959void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2960{ 2961 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2962} 2963 2964ssize_t AudioFlinger::MixerThread::threadLoop_write() 2965{ 2966 // FIXME we should only do one push per cycle; confirm this is true 2967 // Start the fast mixer if it's not already running 2968 if (mFastMixer != 0) { 2969 FastMixerStateQueue *sq = mFastMixer->sq(); 2970 FastMixerState *state = sq->begin(); 2971 if (state->mCommand != FastMixerState::MIX_WRITE && 2972 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2973 if (state->mCommand == FastMixerState::COLD_IDLE) { 2974 int32_t old = android_atomic_inc(&mFastMixerFutex); 2975 if (old == -1) { 2976 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2977 } 2978#ifdef AUDIO_WATCHDOG 2979 if (mAudioWatchdog != 0) { 2980 mAudioWatchdog->resume(); 2981 } 2982#endif 2983 } 2984 state->mCommand = FastMixerState::MIX_WRITE; 2985 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2986 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2987 sq->end(); 2988 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2989 if (kUseFastMixer == FastMixer_Dynamic) { 2990 mNormalSink = mPipeSink; 2991 } 2992 } else { 2993 sq->end(false /*didModify*/); 2994 } 2995 } 2996 return PlaybackThread::threadLoop_write(); 2997} 2998 2999void AudioFlinger::MixerThread::threadLoop_standby() 3000{ 3001 // Idle the fast mixer if it's currently running 3002 if (mFastMixer != 0) { 3003 FastMixerStateQueue *sq = mFastMixer->sq(); 3004 FastMixerState *state = sq->begin(); 3005 if (!(state->mCommand & FastMixerState::IDLE)) { 3006 state->mCommand = FastMixerState::COLD_IDLE; 3007 state->mColdFutexAddr = &mFastMixerFutex; 3008 state->mColdGen++; 3009 mFastMixerFutex = 0; 3010 sq->end(); 3011 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3012 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3013 if (kUseFastMixer == FastMixer_Dynamic) { 3014 mNormalSink = mOutputSink; 3015 } 3016#ifdef AUDIO_WATCHDOG 3017 if (mAudioWatchdog != 0) { 3018 mAudioWatchdog->pause(); 3019 } 3020#endif 3021 } else { 3022 sq->end(false /*didModify*/); 3023 } 3024 } 3025 PlaybackThread::threadLoop_standby(); 3026} 3027 3028bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3029{ 3030 return false; 3031} 3032 3033bool AudioFlinger::PlaybackThread::shouldStandby_l() 3034{ 3035 return !mStandby; 3036} 3037 3038bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3039{ 3040 Mutex::Autolock _l(mLock); 3041 return waitingAsyncCallback_l(); 3042} 3043 3044// shared by MIXER and DIRECT, overridden by DUPLICATING 3045void AudioFlinger::PlaybackThread::threadLoop_standby() 3046{ 3047 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3048 mOutput->stream->common.standby(&mOutput->stream->common); 3049 if (mUseAsyncWrite != 0) { 3050 // discard any pending drain or write ack by incrementing sequence 3051 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3052 mDrainSequence = (mDrainSequence + 2) & ~1; 3053 ALOG_ASSERT(mCallbackThread != 0); 3054 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3055 mCallbackThread->setDraining(mDrainSequence); 3056 } 3057} 3058 3059void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3060{ 3061 ALOGV("signal playback thread"); 3062 broadcast_l(); 3063} 3064 3065void AudioFlinger::MixerThread::threadLoop_mix() 3066{ 3067 // obtain the presentation timestamp of the next output buffer 3068 int64_t pts; 3069 status_t status = INVALID_OPERATION; 3070 3071 if (mNormalSink != 0) { 3072 status = mNormalSink->getNextWriteTimestamp(&pts); 3073 } else { 3074 status = mOutputSink->getNextWriteTimestamp(&pts); 3075 } 3076 3077 if (status != NO_ERROR) { 3078 pts = AudioBufferProvider::kInvalidPTS; 3079 } 3080 3081 // mix buffers... 3082 mAudioMixer->process(pts); 3083 mCurrentWriteLength = mSinkBufferSize; 3084 // increase sleep time progressively when application underrun condition clears. 3085 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3086 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3087 // such that we would underrun the audio HAL. 3088 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3089 sleepTimeShift--; 3090 } 3091 sleepTime = 0; 3092 standbyTime = systemTime() + standbyDelay; 3093 //TODO: delay standby when effects have a tail 3094} 3095 3096void AudioFlinger::MixerThread::threadLoop_sleepTime() 3097{ 3098 // If no tracks are ready, sleep once for the duration of an output 3099 // buffer size, then write 0s to the output 3100 if (sleepTime == 0) { 3101 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3102 sleepTime = activeSleepTime >> sleepTimeShift; 3103 if (sleepTime < kMinThreadSleepTimeUs) { 3104 sleepTime = kMinThreadSleepTimeUs; 3105 } 3106 // reduce sleep time in case of consecutive application underruns to avoid 3107 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3108 // duration we would end up writing less data than needed by the audio HAL if 3109 // the condition persists. 3110 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3111 sleepTimeShift++; 3112 } 3113 } else { 3114 sleepTime = idleSleepTime; 3115 } 3116 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3117 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3118 // before effects processing or output. 3119 if (mMixerBufferValid) { 3120 memset(mMixerBuffer, 0, mMixerBufferSize); 3121 } else { 3122 memset(mSinkBuffer, 0, mSinkBufferSize); 3123 } 3124 sleepTime = 0; 3125 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3126 "anticipated start"); 3127 } 3128 // TODO add standby time extension fct of effect tail 3129} 3130 3131// prepareTracks_l() must be called with ThreadBase::mLock held 3132AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3133 Vector< sp<Track> > *tracksToRemove) 3134{ 3135 3136 mixer_state mixerStatus = MIXER_IDLE; 3137 // find out which tracks need to be processed 3138 size_t count = mActiveTracks.size(); 3139 size_t mixedTracks = 0; 3140 size_t tracksWithEffect = 0; 3141 // counts only _active_ fast tracks 3142 size_t fastTracks = 0; 3143 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3144 3145 float masterVolume = mMasterVolume; 3146 bool masterMute = mMasterMute; 3147 3148 if (masterMute) { 3149 masterVolume = 0; 3150 } 3151 // Delegate master volume control to effect in output mix effect chain if needed 3152 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3153 if (chain != 0) { 3154 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3155 chain->setVolume_l(&v, &v); 3156 masterVolume = (float)((v + (1 << 23)) >> 24); 3157 chain.clear(); 3158 } 3159 3160 // prepare a new state to push 3161 FastMixerStateQueue *sq = NULL; 3162 FastMixerState *state = NULL; 3163 bool didModify = false; 3164 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3165 if (mFastMixer != 0) { 3166 sq = mFastMixer->sq(); 3167 state = sq->begin(); 3168 } 3169 3170 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3171 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3172 3173 for (size_t i=0 ; i<count ; i++) { 3174 const sp<Track> t = mActiveTracks[i].promote(); 3175 if (t == 0) { 3176 continue; 3177 } 3178 3179 // this const just means the local variable doesn't change 3180 Track* const track = t.get(); 3181 3182 // process fast tracks 3183 if (track->isFastTrack()) { 3184 3185 // It's theoretically possible (though unlikely) for a fast track to be created 3186 // and then removed within the same normal mix cycle. This is not a problem, as 3187 // the track never becomes active so it's fast mixer slot is never touched. 3188 // The converse, of removing an (active) track and then creating a new track 3189 // at the identical fast mixer slot within the same normal mix cycle, 3190 // is impossible because the slot isn't marked available until the end of each cycle. 3191 int j = track->mFastIndex; 3192 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3193 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3194 FastTrack *fastTrack = &state->mFastTracks[j]; 3195 3196 // Determine whether the track is currently in underrun condition, 3197 // and whether it had a recent underrun. 3198 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3199 FastTrackUnderruns underruns = ftDump->mUnderruns; 3200 uint32_t recentFull = (underruns.mBitFields.mFull - 3201 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3202 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3203 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3204 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3205 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3206 uint32_t recentUnderruns = recentPartial + recentEmpty; 3207 track->mObservedUnderruns = underruns; 3208 // don't count underruns that occur while stopping or pausing 3209 // or stopped which can occur when flush() is called while active 3210 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3211 recentUnderruns > 0) { 3212 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3213 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3214 } 3215 3216 // This is similar to the state machine for normal tracks, 3217 // with a few modifications for fast tracks. 3218 bool isActive = true; 3219 switch (track->mState) { 3220 case TrackBase::STOPPING_1: 3221 // track stays active in STOPPING_1 state until first underrun 3222 if (recentUnderruns > 0 || track->isTerminated()) { 3223 track->mState = TrackBase::STOPPING_2; 3224 } 3225 break; 3226 case TrackBase::PAUSING: 3227 // ramp down is not yet implemented 3228 track->setPaused(); 3229 break; 3230 case TrackBase::RESUMING: 3231 // ramp up is not yet implemented 3232 track->mState = TrackBase::ACTIVE; 3233 break; 3234 case TrackBase::ACTIVE: 3235 if (recentFull > 0 || recentPartial > 0) { 3236 // track has provided at least some frames recently: reset retry count 3237 track->mRetryCount = kMaxTrackRetries; 3238 } 3239 if (recentUnderruns == 0) { 3240 // no recent underruns: stay active 3241 break; 3242 } 3243 // there has recently been an underrun of some kind 3244 if (track->sharedBuffer() == 0) { 3245 // were any of the recent underruns "empty" (no frames available)? 3246 if (recentEmpty == 0) { 3247 // no, then ignore the partial underruns as they are allowed indefinitely 3248 break; 3249 } 3250 // there has recently been an "empty" underrun: decrement the retry counter 3251 if (--(track->mRetryCount) > 0) { 3252 break; 3253 } 3254 // indicate to client process that the track was disabled because of underrun; 3255 // it will then automatically call start() when data is available 3256 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3257 // remove from active list, but state remains ACTIVE [confusing but true] 3258 isActive = false; 3259 break; 3260 } 3261 // fall through 3262 case TrackBase::STOPPING_2: 3263 case TrackBase::PAUSED: 3264 case TrackBase::STOPPED: 3265 case TrackBase::FLUSHED: // flush() while active 3266 // Check for presentation complete if track is inactive 3267 // We have consumed all the buffers of this track. 3268 // This would be incomplete if we auto-paused on underrun 3269 { 3270 size_t audioHALFrames = 3271 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3272 size_t framesWritten = mBytesWritten / mFrameSize; 3273 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3274 // track stays in active list until presentation is complete 3275 break; 3276 } 3277 } 3278 if (track->isStopping_2()) { 3279 track->mState = TrackBase::STOPPED; 3280 } 3281 if (track->isStopped()) { 3282 // Can't reset directly, as fast mixer is still polling this track 3283 // track->reset(); 3284 // So instead mark this track as needing to be reset after push with ack 3285 resetMask |= 1 << i; 3286 } 3287 isActive = false; 3288 break; 3289 case TrackBase::IDLE: 3290 default: 3291 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3292 } 3293 3294 if (isActive) { 3295 // was it previously inactive? 3296 if (!(state->mTrackMask & (1 << j))) { 3297 ExtendedAudioBufferProvider *eabp = track; 3298 VolumeProvider *vp = track; 3299 fastTrack->mBufferProvider = eabp; 3300 fastTrack->mVolumeProvider = vp; 3301 fastTrack->mChannelMask = track->mChannelMask; 3302 fastTrack->mFormat = track->mFormat; 3303 fastTrack->mGeneration++; 3304 state->mTrackMask |= 1 << j; 3305 didModify = true; 3306 // no acknowledgement required for newly active tracks 3307 } 3308 // cache the combined master volume and stream type volume for fast mixer; this 3309 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3310 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3311 ++fastTracks; 3312 } else { 3313 // was it previously active? 3314 if (state->mTrackMask & (1 << j)) { 3315 fastTrack->mBufferProvider = NULL; 3316 fastTrack->mGeneration++; 3317 state->mTrackMask &= ~(1 << j); 3318 didModify = true; 3319 // If any fast tracks were removed, we must wait for acknowledgement 3320 // because we're about to decrement the last sp<> on those tracks. 3321 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3322 } else { 3323 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3324 } 3325 tracksToRemove->add(track); 3326 // Avoids a misleading display in dumpsys 3327 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3328 } 3329 continue; 3330 } 3331 3332 { // local variable scope to avoid goto warning 3333 3334 audio_track_cblk_t* cblk = track->cblk(); 3335 3336 // The first time a track is added we wait 3337 // for all its buffers to be filled before processing it 3338 int name = track->name(); 3339 // make sure that we have enough frames to mix one full buffer. 3340 // enforce this condition only once to enable draining the buffer in case the client 3341 // app does not call stop() and relies on underrun to stop: 3342 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3343 // during last round 3344 size_t desiredFrames; 3345 uint32_t sr = track->sampleRate(); 3346 if (sr == mSampleRate) { 3347 desiredFrames = mNormalFrameCount; 3348 } else { 3349 // +1 for rounding and +1 for additional sample needed for interpolation 3350 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3351 // add frames already consumed but not yet released by the resampler 3352 // because mAudioTrackServerProxy->framesReady() will include these frames 3353 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3354#if 0 3355 // the minimum track buffer size is normally twice the number of frames necessary 3356 // to fill one buffer and the resampler should not leave more than one buffer worth 3357 // of unreleased frames after each pass, but just in case... 3358 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3359#endif 3360 } 3361 uint32_t minFrames = 1; 3362 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3363 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3364 minFrames = desiredFrames; 3365 } 3366 3367 size_t framesReady = track->framesReady(); 3368 if ((framesReady >= minFrames) && track->isReady() && 3369 !track->isPaused() && !track->isTerminated()) 3370 { 3371 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3372 3373 mixedTracks++; 3374 3375 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3376 // there is an effect chain connected to the track 3377 chain.clear(); 3378 if (track->mainBuffer() != mSinkBuffer && 3379 track->mainBuffer() != mMixerBuffer) { 3380 if (mEffectBufferEnabled) { 3381 mEffectBufferValid = true; // Later can set directly. 3382 } 3383 chain = getEffectChain_l(track->sessionId()); 3384 // Delegate volume control to effect in track effect chain if needed 3385 if (chain != 0) { 3386 tracksWithEffect++; 3387 } else { 3388 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3389 "session %d", 3390 name, track->sessionId()); 3391 } 3392 } 3393 3394 3395 int param = AudioMixer::VOLUME; 3396 if (track->mFillingUpStatus == Track::FS_FILLED) { 3397 // no ramp for the first volume setting 3398 track->mFillingUpStatus = Track::FS_ACTIVE; 3399 if (track->mState == TrackBase::RESUMING) { 3400 track->mState = TrackBase::ACTIVE; 3401 param = AudioMixer::RAMP_VOLUME; 3402 } 3403 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3404 // FIXME should not make a decision based on mServer 3405 } else if (cblk->mServer != 0) { 3406 // If the track is stopped before the first frame was mixed, 3407 // do not apply ramp 3408 param = AudioMixer::RAMP_VOLUME; 3409 } 3410 3411 // compute volume for this track 3412 uint32_t vl, vr; // in U8.24 integer format 3413 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3414 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3415 vl = vr = 0; 3416 vlf = vrf = vaf = 0.; 3417 if (track->isPausing()) { 3418 track->setPaused(); 3419 } 3420 } else { 3421 3422 // read original volumes with volume control 3423 float typeVolume = mStreamTypes[track->streamType()].volume; 3424 float v = masterVolume * typeVolume; 3425 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3426 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3427 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3428 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3429 // track volumes come from shared memory, so can't be trusted and must be clamped 3430 if (vlf > GAIN_FLOAT_UNITY) { 3431 ALOGV("Track left volume out of range: %.3g", vlf); 3432 vlf = GAIN_FLOAT_UNITY; 3433 } 3434 if (vrf > GAIN_FLOAT_UNITY) { 3435 ALOGV("Track right volume out of range: %.3g", vrf); 3436 vrf = GAIN_FLOAT_UNITY; 3437 } 3438 // now apply the master volume and stream type volume 3439 vlf *= v; 3440 vrf *= v; 3441 // assuming master volume and stream type volume each go up to 1.0, 3442 // then derive vl and vr as U8.24 versions for the effect chain 3443 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3444 vl = (uint32_t) (scaleto8_24 * vlf); 3445 vr = (uint32_t) (scaleto8_24 * vrf); 3446 // vl and vr are now in U8.24 format 3447 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3448 // send level comes from shared memory and so may be corrupt 3449 if (sendLevel > MAX_GAIN_INT) { 3450 ALOGV("Track send level out of range: %04X", sendLevel); 3451 sendLevel = MAX_GAIN_INT; 3452 } 3453 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3454 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3455 } 3456 3457 // Delegate volume control to effect in track effect chain if needed 3458 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3459 // Do not ramp volume if volume is controlled by effect 3460 param = AudioMixer::VOLUME; 3461 // Update remaining floating point volume levels 3462 vlf = (float)vl / (1 << 24); 3463 vrf = (float)vr / (1 << 24); 3464 track->mHasVolumeController = true; 3465 } else { 3466 // force no volume ramp when volume controller was just disabled or removed 3467 // from effect chain to avoid volume spike 3468 if (track->mHasVolumeController) { 3469 param = AudioMixer::VOLUME; 3470 } 3471 track->mHasVolumeController = false; 3472 } 3473 3474 // XXX: these things DON'T need to be done each time 3475 mAudioMixer->setBufferProvider(name, track); 3476 mAudioMixer->enable(name); 3477 3478 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3479 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3480 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3481 mAudioMixer->setParameter( 3482 name, 3483 AudioMixer::TRACK, 3484 AudioMixer::FORMAT, (void *)track->format()); 3485 mAudioMixer->setParameter( 3486 name, 3487 AudioMixer::TRACK, 3488 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3489 mAudioMixer->setParameter( 3490 name, 3491 AudioMixer::TRACK, 3492 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3493 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3494 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3495 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3496 if (reqSampleRate == 0) { 3497 reqSampleRate = mSampleRate; 3498 } else if (reqSampleRate > maxSampleRate) { 3499 reqSampleRate = maxSampleRate; 3500 } 3501 mAudioMixer->setParameter( 3502 name, 3503 AudioMixer::RESAMPLE, 3504 AudioMixer::SAMPLE_RATE, 3505 (void *)(uintptr_t)reqSampleRate); 3506 /* 3507 * Select the appropriate output buffer for the track. 3508 * 3509 * Tracks with effects go into their own effects chain buffer 3510 * and from there into either mEffectBuffer or mSinkBuffer. 3511 * 3512 * Other tracks can use mMixerBuffer for higher precision 3513 * channel accumulation. If this buffer is enabled 3514 * (mMixerBufferEnabled true), then selected tracks will accumulate 3515 * into it. 3516 * 3517 */ 3518 if (mMixerBufferEnabled 3519 && (track->mainBuffer() == mSinkBuffer 3520 || track->mainBuffer() == mMixerBuffer)) { 3521 mAudioMixer->setParameter( 3522 name, 3523 AudioMixer::TRACK, 3524 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3525 mAudioMixer->setParameter( 3526 name, 3527 AudioMixer::TRACK, 3528 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3529 // TODO: override track->mainBuffer()? 3530 mMixerBufferValid = true; 3531 } else { 3532 mAudioMixer->setParameter( 3533 name, 3534 AudioMixer::TRACK, 3535 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3536 mAudioMixer->setParameter( 3537 name, 3538 AudioMixer::TRACK, 3539 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3540 } 3541 mAudioMixer->setParameter( 3542 name, 3543 AudioMixer::TRACK, 3544 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3545 3546 // reset retry count 3547 track->mRetryCount = kMaxTrackRetries; 3548 3549 // If one track is ready, set the mixer ready if: 3550 // - the mixer was not ready during previous round OR 3551 // - no other track is not ready 3552 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3553 mixerStatus != MIXER_TRACKS_ENABLED) { 3554 mixerStatus = MIXER_TRACKS_READY; 3555 } 3556 } else { 3557 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3558 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3559 } 3560 // clear effect chain input buffer if an active track underruns to avoid sending 3561 // previous audio buffer again to effects 3562 chain = getEffectChain_l(track->sessionId()); 3563 if (chain != 0) { 3564 chain->clearInputBuffer(); 3565 } 3566 3567 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3568 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3569 track->isStopped() || track->isPaused()) { 3570 // We have consumed all the buffers of this track. 3571 // Remove it from the list of active tracks. 3572 // TODO: use actual buffer filling status instead of latency when available from 3573 // audio HAL 3574 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3575 size_t framesWritten = mBytesWritten / mFrameSize; 3576 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3577 if (track->isStopped()) { 3578 track->reset(); 3579 } 3580 tracksToRemove->add(track); 3581 } 3582 } else { 3583 // No buffers for this track. Give it a few chances to 3584 // fill a buffer, then remove it from active list. 3585 if (--(track->mRetryCount) <= 0) { 3586 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3587 tracksToRemove->add(track); 3588 // indicate to client process that the track was disabled because of underrun; 3589 // it will then automatically call start() when data is available 3590 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3591 // If one track is not ready, mark the mixer also not ready if: 3592 // - the mixer was ready during previous round OR 3593 // - no other track is ready 3594 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3595 mixerStatus != MIXER_TRACKS_READY) { 3596 mixerStatus = MIXER_TRACKS_ENABLED; 3597 } 3598 } 3599 mAudioMixer->disable(name); 3600 } 3601 3602 } // local variable scope to avoid goto warning 3603track_is_ready: ; 3604 3605 } 3606 3607 // Push the new FastMixer state if necessary 3608 bool pauseAudioWatchdog = false; 3609 if (didModify) { 3610 state->mFastTracksGen++; 3611 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3612 if (kUseFastMixer == FastMixer_Dynamic && 3613 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3614 state->mCommand = FastMixerState::COLD_IDLE; 3615 state->mColdFutexAddr = &mFastMixerFutex; 3616 state->mColdGen++; 3617 mFastMixerFutex = 0; 3618 if (kUseFastMixer == FastMixer_Dynamic) { 3619 mNormalSink = mOutputSink; 3620 } 3621 // If we go into cold idle, need to wait for acknowledgement 3622 // so that fast mixer stops doing I/O. 3623 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3624 pauseAudioWatchdog = true; 3625 } 3626 } 3627 if (sq != NULL) { 3628 sq->end(didModify); 3629 sq->push(block); 3630 } 3631#ifdef AUDIO_WATCHDOG 3632 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3633 mAudioWatchdog->pause(); 3634 } 3635#endif 3636 3637 // Now perform the deferred reset on fast tracks that have stopped 3638 while (resetMask != 0) { 3639 size_t i = __builtin_ctz(resetMask); 3640 ALOG_ASSERT(i < count); 3641 resetMask &= ~(1 << i); 3642 sp<Track> t = mActiveTracks[i].promote(); 3643 if (t == 0) { 3644 continue; 3645 } 3646 Track* track = t.get(); 3647 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3648 track->reset(); 3649 } 3650 3651 // remove all the tracks that need to be... 3652 removeTracks_l(*tracksToRemove); 3653 3654 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3655 mEffectBufferValid = true; 3656 } 3657 3658 // sink or mix buffer must be cleared if all tracks are connected to an 3659 // effect chain as in this case the mixer will not write to the sink or mix buffer 3660 // and track effects will accumulate into it 3661 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3662 (mixedTracks == 0 && fastTracks > 0))) { 3663 // FIXME as a performance optimization, should remember previous zero status 3664 if (mMixerBufferValid) { 3665 memset(mMixerBuffer, 0, mMixerBufferSize); 3666 // TODO: In testing, mSinkBuffer below need not be cleared because 3667 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3668 // after mixing. 3669 // 3670 // To enforce this guarantee: 3671 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3672 // (mixedTracks == 0 && fastTracks > 0)) 3673 // must imply MIXER_TRACKS_READY. 3674 // Later, we may clear buffers regardless, and skip much of this logic. 3675 } 3676 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3677 if (mEffectBufferValid) { 3678 memset(mEffectBuffer, 0, mEffectBufferSize); 3679 } 3680 // FIXME as a performance optimization, should remember previous zero status 3681 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3682 } 3683 3684 // if any fast tracks, then status is ready 3685 mMixerStatusIgnoringFastTracks = mixerStatus; 3686 if (fastTracks > 0) { 3687 mixerStatus = MIXER_TRACKS_READY; 3688 } 3689 return mixerStatus; 3690} 3691 3692// getTrackName_l() must be called with ThreadBase::mLock held 3693int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3694 audio_format_t format, int sessionId) 3695{ 3696 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3697} 3698 3699// deleteTrackName_l() must be called with ThreadBase::mLock held 3700void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3701{ 3702 ALOGV("remove track (%d) and delete from mixer", name); 3703 mAudioMixer->deleteTrackName(name); 3704} 3705 3706// checkForNewParameter_l() must be called with ThreadBase::mLock held 3707bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3708 status_t& status) 3709{ 3710 bool reconfig = false; 3711 3712 status = NO_ERROR; 3713 3714 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3715 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3716 if (mFastMixer != 0) { 3717 FastMixerStateQueue *sq = mFastMixer->sq(); 3718 FastMixerState *state = sq->begin(); 3719 if (!(state->mCommand & FastMixerState::IDLE)) { 3720 previousCommand = state->mCommand; 3721 state->mCommand = FastMixerState::HOT_IDLE; 3722 sq->end(); 3723 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3724 } else { 3725 sq->end(false /*didModify*/); 3726 } 3727 } 3728 3729 AudioParameter param = AudioParameter(keyValuePair); 3730 int value; 3731 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3732 reconfig = true; 3733 } 3734 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3735 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3736 status = BAD_VALUE; 3737 } else { 3738 // no need to save value, since it's constant 3739 reconfig = true; 3740 } 3741 } 3742 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3743 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3744 status = BAD_VALUE; 3745 } else { 3746 // no need to save value, since it's constant 3747 reconfig = true; 3748 } 3749 } 3750 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3751 // do not accept frame count changes if tracks are open as the track buffer 3752 // size depends on frame count and correct behavior would not be guaranteed 3753 // if frame count is changed after track creation 3754 if (!mTracks.isEmpty()) { 3755 status = INVALID_OPERATION; 3756 } else { 3757 reconfig = true; 3758 } 3759 } 3760 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3761#ifdef ADD_BATTERY_DATA 3762 // when changing the audio output device, call addBatteryData to notify 3763 // the change 3764 if (mOutDevice != value) { 3765 uint32_t params = 0; 3766 // check whether speaker is on 3767 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3768 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3769 } 3770 3771 audio_devices_t deviceWithoutSpeaker 3772 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3773 // check if any other device (except speaker) is on 3774 if (value & deviceWithoutSpeaker ) { 3775 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3776 } 3777 3778 if (params != 0) { 3779 addBatteryData(params); 3780 } 3781 } 3782#endif 3783 3784 // forward device change to effects that have requested to be 3785 // aware of attached audio device. 3786 if (value != AUDIO_DEVICE_NONE) { 3787 mOutDevice = value; 3788 for (size_t i = 0; i < mEffectChains.size(); i++) { 3789 mEffectChains[i]->setDevice_l(mOutDevice); 3790 } 3791 } 3792 } 3793 3794 if (status == NO_ERROR) { 3795 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3796 keyValuePair.string()); 3797 if (!mStandby && status == INVALID_OPERATION) { 3798 mOutput->stream->common.standby(&mOutput->stream->common); 3799 mStandby = true; 3800 mBytesWritten = 0; 3801 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3802 keyValuePair.string()); 3803 } 3804 if (status == NO_ERROR && reconfig) { 3805 readOutputParameters_l(); 3806 delete mAudioMixer; 3807 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3808 for (size_t i = 0; i < mTracks.size() ; i++) { 3809 int name = getTrackName_l(mTracks[i]->mChannelMask, 3810 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3811 if (name < 0) { 3812 break; 3813 } 3814 mTracks[i]->mName = name; 3815 } 3816 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3817 } 3818 } 3819 3820 if (!(previousCommand & FastMixerState::IDLE)) { 3821 ALOG_ASSERT(mFastMixer != 0); 3822 FastMixerStateQueue *sq = mFastMixer->sq(); 3823 FastMixerState *state = sq->begin(); 3824 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3825 state->mCommand = previousCommand; 3826 sq->end(); 3827 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3828 } 3829 3830 return reconfig; 3831} 3832 3833 3834void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3835{ 3836 const size_t SIZE = 256; 3837 char buffer[SIZE]; 3838 String8 result; 3839 3840 PlaybackThread::dumpInternals(fd, args); 3841 3842 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3843 3844 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3845 const FastMixerDumpState copy(mFastMixerDumpState); 3846 copy.dump(fd); 3847 3848#ifdef STATE_QUEUE_DUMP 3849 // Similar for state queue 3850 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3851 observerCopy.dump(fd); 3852 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3853 mutatorCopy.dump(fd); 3854#endif 3855 3856#ifdef TEE_SINK 3857 // Write the tee output to a .wav file 3858 dumpTee(fd, mTeeSource, mId); 3859#endif 3860 3861#ifdef AUDIO_WATCHDOG 3862 if (mAudioWatchdog != 0) { 3863 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3864 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3865 wdCopy.dump(fd); 3866 } 3867#endif 3868} 3869 3870uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3871{ 3872 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3873} 3874 3875uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3876{ 3877 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3878} 3879 3880void AudioFlinger::MixerThread::cacheParameters_l() 3881{ 3882 PlaybackThread::cacheParameters_l(); 3883 3884 // FIXME: Relaxed timing because of a certain device that can't meet latency 3885 // Should be reduced to 2x after the vendor fixes the driver issue 3886 // increase threshold again due to low power audio mode. The way this warning 3887 // threshold is calculated and its usefulness should be reconsidered anyway. 3888 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3889} 3890 3891// ---------------------------------------------------------------------------- 3892 3893AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3894 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3895 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3896 // mLeftVolFloat, mRightVolFloat 3897{ 3898} 3899 3900AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3901 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3902 ThreadBase::type_t type) 3903 : PlaybackThread(audioFlinger, output, id, device, type) 3904 // mLeftVolFloat, mRightVolFloat 3905{ 3906} 3907 3908AudioFlinger::DirectOutputThread::~DirectOutputThread() 3909{ 3910} 3911 3912void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3913{ 3914 audio_track_cblk_t* cblk = track->cblk(); 3915 float left, right; 3916 3917 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3918 left = right = 0; 3919 } else { 3920 float typeVolume = mStreamTypes[track->streamType()].volume; 3921 float v = mMasterVolume * typeVolume; 3922 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3923 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3924 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3925 if (left > GAIN_FLOAT_UNITY) { 3926 left = GAIN_FLOAT_UNITY; 3927 } 3928 left *= v; 3929 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3930 if (right > GAIN_FLOAT_UNITY) { 3931 right = GAIN_FLOAT_UNITY; 3932 } 3933 right *= v; 3934 } 3935 3936 if (lastTrack) { 3937 if (left != mLeftVolFloat || right != mRightVolFloat) { 3938 mLeftVolFloat = left; 3939 mRightVolFloat = right; 3940 3941 // Convert volumes from float to 8.24 3942 uint32_t vl = (uint32_t)(left * (1 << 24)); 3943 uint32_t vr = (uint32_t)(right * (1 << 24)); 3944 3945 // Delegate volume control to effect in track effect chain if needed 3946 // only one effect chain can be present on DirectOutputThread, so if 3947 // there is one, the track is connected to it 3948 if (!mEffectChains.isEmpty()) { 3949 mEffectChains[0]->setVolume_l(&vl, &vr); 3950 left = (float)vl / (1 << 24); 3951 right = (float)vr / (1 << 24); 3952 } 3953 if (mOutput->stream->set_volume) { 3954 mOutput->stream->set_volume(mOutput->stream, left, right); 3955 } 3956 } 3957 } 3958} 3959 3960 3961AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3962 Vector< sp<Track> > *tracksToRemove 3963) 3964{ 3965 size_t count = mActiveTracks.size(); 3966 mixer_state mixerStatus = MIXER_IDLE; 3967 3968 // find out which tracks need to be processed 3969 for (size_t i = 0; i < count; i++) { 3970 sp<Track> t = mActiveTracks[i].promote(); 3971 // The track died recently 3972 if (t == 0) { 3973 continue; 3974 } 3975 3976 Track* const track = t.get(); 3977 audio_track_cblk_t* cblk = track->cblk(); 3978 // Only consider last track started for volume and mixer state control. 3979 // In theory an older track could underrun and restart after the new one starts 3980 // but as we only care about the transition phase between two tracks on a 3981 // direct output, it is not a problem to ignore the underrun case. 3982 sp<Track> l = mLatestActiveTrack.promote(); 3983 bool last = l.get() == track; 3984 3985 // The first time a track is added we wait 3986 // for all its buffers to be filled before processing it 3987 uint32_t minFrames; 3988 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 3989 minFrames = mNormalFrameCount; 3990 } else { 3991 minFrames = 1; 3992 } 3993 3994 ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ", 3995 minFrames, track->mState, track->framesReady()); 3996 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 3997 !track->isStopping_2() && !track->isStopped()) 3998 { 3999 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4000 4001 if (track->mFillingUpStatus == Track::FS_FILLED) { 4002 track->mFillingUpStatus = Track::FS_ACTIVE; 4003 // make sure processVolume_l() will apply new volume even if 0 4004 mLeftVolFloat = mRightVolFloat = -1.0; 4005 if (track->mState == TrackBase::RESUMING) { 4006 track->mState = TrackBase::ACTIVE; 4007 } 4008 } 4009 4010 // compute volume for this track 4011 processVolume_l(track, last); 4012 if (last) { 4013 // reset retry count 4014 track->mRetryCount = kMaxTrackRetriesDirect; 4015 mActiveTrack = t; 4016 mixerStatus = MIXER_TRACKS_READY; 4017 } 4018 } else { 4019 // clear effect chain input buffer if the last active track started underruns 4020 // to avoid sending previous audio buffer again to effects 4021 if (!mEffectChains.isEmpty() && last) { 4022 mEffectChains[0]->clearInputBuffer(); 4023 } 4024 if (track->isStopping_1()) { 4025 track->mState = TrackBase::STOPPING_2; 4026 } 4027 if ((track->sharedBuffer() != 0) || track->isStopped() || 4028 track->isStopping_2() || track->isPaused()) { 4029 // We have consumed all the buffers of this track. 4030 // Remove it from the list of active tracks. 4031 size_t audioHALFrames; 4032 if (audio_is_linear_pcm(mFormat)) { 4033 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4034 } else { 4035 audioHALFrames = 0; 4036 } 4037 4038 size_t framesWritten = mBytesWritten / mFrameSize; 4039 if (mStandby || !last || 4040 track->presentationComplete(framesWritten, audioHALFrames)) { 4041 if (track->isStopping_2()) { 4042 track->mState = TrackBase::STOPPED; 4043 } 4044 if (track->isStopped()) { 4045 track->reset(); 4046 } 4047 tracksToRemove->add(track); 4048 } 4049 } else { 4050 // No buffers for this track. Give it a few chances to 4051 // fill a buffer, then remove it from active list. 4052 // Only consider last track started for mixer state control 4053 if (--(track->mRetryCount) <= 0) { 4054 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4055 tracksToRemove->add(track); 4056 // indicate to client process that the track was disabled because of underrun; 4057 // it will then automatically call start() when data is available 4058 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4059 } else if (last) { 4060 mixerStatus = MIXER_TRACKS_ENABLED; 4061 } 4062 } 4063 } 4064 } 4065 4066 // remove all the tracks that need to be... 4067 removeTracks_l(*tracksToRemove); 4068 4069 return mixerStatus; 4070} 4071 4072void AudioFlinger::DirectOutputThread::threadLoop_mix() 4073{ 4074 size_t frameCount = mFrameCount; 4075 int8_t *curBuf = (int8_t *)mSinkBuffer; 4076 // output audio to hardware 4077 while (frameCount) { 4078 AudioBufferProvider::Buffer buffer; 4079 buffer.frameCount = frameCount; 4080 mActiveTrack->getNextBuffer(&buffer); 4081 if (buffer.raw == NULL) { 4082 memset(curBuf, 0, frameCount * mFrameSize); 4083 break; 4084 } 4085 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4086 frameCount -= buffer.frameCount; 4087 curBuf += buffer.frameCount * mFrameSize; 4088 mActiveTrack->releaseBuffer(&buffer); 4089 } 4090 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4091 sleepTime = 0; 4092 standbyTime = systemTime() + standbyDelay; 4093 mActiveTrack.clear(); 4094} 4095 4096void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4097{ 4098 if (sleepTime == 0) { 4099 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4100 sleepTime = activeSleepTime; 4101 } else { 4102 sleepTime = idleSleepTime; 4103 } 4104 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4105 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4106 sleepTime = 0; 4107 } 4108} 4109 4110// getTrackName_l() must be called with ThreadBase::mLock held 4111int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4112 audio_format_t format __unused, int sessionId __unused) 4113{ 4114 return 0; 4115} 4116 4117// deleteTrackName_l() must be called with ThreadBase::mLock held 4118void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4119{ 4120} 4121 4122// checkForNewParameter_l() must be called with ThreadBase::mLock held 4123bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4124 status_t& status) 4125{ 4126 bool reconfig = false; 4127 4128 status = NO_ERROR; 4129 4130 AudioParameter param = AudioParameter(keyValuePair); 4131 int value; 4132 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4133 // forward device change to effects that have requested to be 4134 // aware of attached audio device. 4135 if (value != AUDIO_DEVICE_NONE) { 4136 mOutDevice = value; 4137 for (size_t i = 0; i < mEffectChains.size(); i++) { 4138 mEffectChains[i]->setDevice_l(mOutDevice); 4139 } 4140 } 4141 } 4142 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4143 // do not accept frame count changes if tracks are open as the track buffer 4144 // size depends on frame count and correct behavior would not be garantied 4145 // if frame count is changed after track creation 4146 if (!mTracks.isEmpty()) { 4147 status = INVALID_OPERATION; 4148 } else { 4149 reconfig = true; 4150 } 4151 } 4152 if (status == NO_ERROR) { 4153 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4154 keyValuePair.string()); 4155 if (!mStandby && status == INVALID_OPERATION) { 4156 mOutput->stream->common.standby(&mOutput->stream->common); 4157 mStandby = true; 4158 mBytesWritten = 0; 4159 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4160 keyValuePair.string()); 4161 } 4162 if (status == NO_ERROR && reconfig) { 4163 readOutputParameters_l(); 4164 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4165 } 4166 } 4167 4168 return reconfig; 4169} 4170 4171uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4172{ 4173 uint32_t time; 4174 if (audio_is_linear_pcm(mFormat)) { 4175 time = PlaybackThread::activeSleepTimeUs(); 4176 } else { 4177 time = 10000; 4178 } 4179 return time; 4180} 4181 4182uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4183{ 4184 uint32_t time; 4185 if (audio_is_linear_pcm(mFormat)) { 4186 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4187 } else { 4188 time = 10000; 4189 } 4190 return time; 4191} 4192 4193uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4194{ 4195 uint32_t time; 4196 if (audio_is_linear_pcm(mFormat)) { 4197 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4198 } else { 4199 time = 10000; 4200 } 4201 return time; 4202} 4203 4204void AudioFlinger::DirectOutputThread::cacheParameters_l() 4205{ 4206 PlaybackThread::cacheParameters_l(); 4207 4208 // use shorter standby delay as on normal output to release 4209 // hardware resources as soon as possible 4210 if (audio_is_linear_pcm(mFormat)) { 4211 standbyDelay = microseconds(activeSleepTime*2); 4212 } else { 4213 standbyDelay = kOffloadStandbyDelayNs; 4214 } 4215} 4216 4217// ---------------------------------------------------------------------------- 4218 4219AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4220 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4221 : Thread(false /*canCallJava*/), 4222 mPlaybackThread(playbackThread), 4223 mWriteAckSequence(0), 4224 mDrainSequence(0) 4225{ 4226} 4227 4228AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4229{ 4230} 4231 4232void AudioFlinger::AsyncCallbackThread::onFirstRef() 4233{ 4234 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4235} 4236 4237bool AudioFlinger::AsyncCallbackThread::threadLoop() 4238{ 4239 while (!exitPending()) { 4240 uint32_t writeAckSequence; 4241 uint32_t drainSequence; 4242 4243 { 4244 Mutex::Autolock _l(mLock); 4245 while (!((mWriteAckSequence & 1) || 4246 (mDrainSequence & 1) || 4247 exitPending())) { 4248 mWaitWorkCV.wait(mLock); 4249 } 4250 4251 if (exitPending()) { 4252 break; 4253 } 4254 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4255 mWriteAckSequence, mDrainSequence); 4256 writeAckSequence = mWriteAckSequence; 4257 mWriteAckSequence &= ~1; 4258 drainSequence = mDrainSequence; 4259 mDrainSequence &= ~1; 4260 } 4261 { 4262 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4263 if (playbackThread != 0) { 4264 if (writeAckSequence & 1) { 4265 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4266 } 4267 if (drainSequence & 1) { 4268 playbackThread->resetDraining(drainSequence >> 1); 4269 } 4270 } 4271 } 4272 } 4273 return false; 4274} 4275 4276void AudioFlinger::AsyncCallbackThread::exit() 4277{ 4278 ALOGV("AsyncCallbackThread::exit"); 4279 Mutex::Autolock _l(mLock); 4280 requestExit(); 4281 mWaitWorkCV.broadcast(); 4282} 4283 4284void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4285{ 4286 Mutex::Autolock _l(mLock); 4287 // bit 0 is cleared 4288 mWriteAckSequence = sequence << 1; 4289} 4290 4291void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4292{ 4293 Mutex::Autolock _l(mLock); 4294 // ignore unexpected callbacks 4295 if (mWriteAckSequence & 2) { 4296 mWriteAckSequence |= 1; 4297 mWaitWorkCV.signal(); 4298 } 4299} 4300 4301void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4302{ 4303 Mutex::Autolock _l(mLock); 4304 // bit 0 is cleared 4305 mDrainSequence = sequence << 1; 4306} 4307 4308void AudioFlinger::AsyncCallbackThread::resetDraining() 4309{ 4310 Mutex::Autolock _l(mLock); 4311 // ignore unexpected callbacks 4312 if (mDrainSequence & 2) { 4313 mDrainSequence |= 1; 4314 mWaitWorkCV.signal(); 4315 } 4316} 4317 4318 4319// ---------------------------------------------------------------------------- 4320AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4321 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4322 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4323 mHwPaused(false), 4324 mFlushPending(false), 4325 mPausedBytesRemaining(0) 4326{ 4327 //FIXME: mStandby should be set to true by ThreadBase constructor 4328 mStandby = true; 4329} 4330 4331void AudioFlinger::OffloadThread::threadLoop_exit() 4332{ 4333 if (mFlushPending || mHwPaused) { 4334 // If a flush is pending or track was paused, just discard buffered data 4335 flushHw_l(); 4336 } else { 4337 mMixerStatus = MIXER_DRAIN_ALL; 4338 threadLoop_drain(); 4339 } 4340 if (mUseAsyncWrite) { 4341 ALOG_ASSERT(mCallbackThread != 0); 4342 mCallbackThread->exit(); 4343 } 4344 PlaybackThread::threadLoop_exit(); 4345} 4346 4347AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4348 Vector< sp<Track> > *tracksToRemove 4349) 4350{ 4351 size_t count = mActiveTracks.size(); 4352 4353 mixer_state mixerStatus = MIXER_IDLE; 4354 bool doHwPause = false; 4355 bool doHwResume = false; 4356 4357 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4358 4359 // find out which tracks need to be processed 4360 for (size_t i = 0; i < count; i++) { 4361 sp<Track> t = mActiveTracks[i].promote(); 4362 // The track died recently 4363 if (t == 0) { 4364 continue; 4365 } 4366 Track* const track = t.get(); 4367 audio_track_cblk_t* cblk = track->cblk(); 4368 // Only consider last track started for volume and mixer state control. 4369 // In theory an older track could underrun and restart after the new one starts 4370 // but as we only care about the transition phase between two tracks on a 4371 // direct output, it is not a problem to ignore the underrun case. 4372 sp<Track> l = mLatestActiveTrack.promote(); 4373 bool last = l.get() == track; 4374 4375 if (track->isInvalid()) { 4376 ALOGW("An invalidated track shouldn't be in active list"); 4377 tracksToRemove->add(track); 4378 continue; 4379 } 4380 4381 if (track->mState == TrackBase::IDLE) { 4382 ALOGW("An idle track shouldn't be in active list"); 4383 continue; 4384 } 4385 4386 if (track->isPausing()) { 4387 track->setPaused(); 4388 if (last) { 4389 if (!mHwPaused) { 4390 doHwPause = true; 4391 mHwPaused = true; 4392 } 4393 // If we were part way through writing the mixbuffer to 4394 // the HAL we must save this until we resume 4395 // BUG - this will be wrong if a different track is made active, 4396 // in that case we want to discard the pending data in the 4397 // mixbuffer and tell the client to present it again when the 4398 // track is resumed 4399 mPausedWriteLength = mCurrentWriteLength; 4400 mPausedBytesRemaining = mBytesRemaining; 4401 mBytesRemaining = 0; // stop writing 4402 } 4403 tracksToRemove->add(track); 4404 } else if (track->isFlushPending()) { 4405 track->flushAck(); 4406 if (last) { 4407 mFlushPending = true; 4408 } 4409 } else if (track->isResumePending()){ 4410 track->resumeAck(); 4411 if (last) { 4412 if (mPausedBytesRemaining) { 4413 // Need to continue write that was interrupted 4414 mCurrentWriteLength = mPausedWriteLength; 4415 mBytesRemaining = mPausedBytesRemaining; 4416 mPausedBytesRemaining = 0; 4417 } 4418 if (mHwPaused) { 4419 doHwResume = true; 4420 mHwPaused = false; 4421 // threadLoop_mix() will handle the case that we need to 4422 // resume an interrupted write 4423 } 4424 // enable write to audio HAL 4425 sleepTime = 0; 4426 4427 // Do not handle new data in this iteration even if track->framesReady() 4428 mixerStatus = MIXER_TRACKS_ENABLED; 4429 } 4430 } else if (track->framesReady() && track->isReady() && 4431 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4432 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4433 if (track->mFillingUpStatus == Track::FS_FILLED) { 4434 track->mFillingUpStatus = Track::FS_ACTIVE; 4435 // make sure processVolume_l() will apply new volume even if 0 4436 mLeftVolFloat = mRightVolFloat = -1.0; 4437 } 4438 4439 if (last) { 4440 sp<Track> previousTrack = mPreviousTrack.promote(); 4441 if (previousTrack != 0) { 4442 if (track != previousTrack.get()) { 4443 // Flush any data still being written from last track 4444 mBytesRemaining = 0; 4445 if (mPausedBytesRemaining) { 4446 // Last track was paused so we also need to flush saved 4447 // mixbuffer state and invalidate track so that it will 4448 // re-submit that unwritten data when it is next resumed 4449 mPausedBytesRemaining = 0; 4450 // Invalidate is a bit drastic - would be more efficient 4451 // to have a flag to tell client that some of the 4452 // previously written data was lost 4453 previousTrack->invalidate(); 4454 } 4455 // flush data already sent to the DSP if changing audio session as audio 4456 // comes from a different source. Also invalidate previous track to force a 4457 // seek when resuming. 4458 if (previousTrack->sessionId() != track->sessionId()) { 4459 previousTrack->invalidate(); 4460 } 4461 } 4462 } 4463 mPreviousTrack = track; 4464 // reset retry count 4465 track->mRetryCount = kMaxTrackRetriesOffload; 4466 mActiveTrack = t; 4467 mixerStatus = MIXER_TRACKS_READY; 4468 } 4469 } else { 4470 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4471 if (track->isStopping_1()) { 4472 // Hardware buffer can hold a large amount of audio so we must 4473 // wait for all current track's data to drain before we say 4474 // that the track is stopped. 4475 if (mBytesRemaining == 0) { 4476 // Only start draining when all data in mixbuffer 4477 // has been written 4478 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4479 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4480 // do not drain if no data was ever sent to HAL (mStandby == true) 4481 if (last && !mStandby) { 4482 // do not modify drain sequence if we are already draining. This happens 4483 // when resuming from pause after drain. 4484 if ((mDrainSequence & 1) == 0) { 4485 sleepTime = 0; 4486 standbyTime = systemTime() + standbyDelay; 4487 mixerStatus = MIXER_DRAIN_TRACK; 4488 mDrainSequence += 2; 4489 } 4490 if (mHwPaused) { 4491 // It is possible to move from PAUSED to STOPPING_1 without 4492 // a resume so we must ensure hardware is running 4493 doHwResume = true; 4494 mHwPaused = false; 4495 } 4496 } 4497 } 4498 } else if (track->isStopping_2()) { 4499 // Drain has completed or we are in standby, signal presentation complete 4500 if (!(mDrainSequence & 1) || !last || mStandby) { 4501 track->mState = TrackBase::STOPPED; 4502 size_t audioHALFrames = 4503 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4504 size_t framesWritten = 4505 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4506 track->presentationComplete(framesWritten, audioHALFrames); 4507 track->reset(); 4508 tracksToRemove->add(track); 4509 } 4510 } else { 4511 // No buffers for this track. Give it a few chances to 4512 // fill a buffer, then remove it from active list. 4513 if (--(track->mRetryCount) <= 0) { 4514 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4515 track->name()); 4516 tracksToRemove->add(track); 4517 // indicate to client process that the track was disabled because of underrun; 4518 // it will then automatically call start() when data is available 4519 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4520 } else if (last){ 4521 mixerStatus = MIXER_TRACKS_ENABLED; 4522 } 4523 } 4524 } 4525 // compute volume for this track 4526 processVolume_l(track, last); 4527 } 4528 4529 // make sure the pause/flush/resume sequence is executed in the right order. 4530 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4531 // before flush and then resume HW. This can happen in case of pause/flush/resume 4532 // if resume is received before pause is executed. 4533 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4534 mOutput->stream->pause(mOutput->stream); 4535 } 4536 if (mFlushPending) { 4537 flushHw_l(); 4538 mFlushPending = false; 4539 } 4540 if (!mStandby && doHwResume) { 4541 mOutput->stream->resume(mOutput->stream); 4542 } 4543 4544 // remove all the tracks that need to be... 4545 removeTracks_l(*tracksToRemove); 4546 4547 return mixerStatus; 4548} 4549 4550// must be called with thread mutex locked 4551bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4552{ 4553 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4554 mWriteAckSequence, mDrainSequence); 4555 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4556 return true; 4557 } 4558 return false; 4559} 4560 4561// must be called with thread mutex locked 4562bool AudioFlinger::OffloadThread::shouldStandby_l() 4563{ 4564 bool trackPaused = false; 4565 4566 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4567 // after a timeout and we will enter standby then. 4568 if (mTracks.size() > 0) { 4569 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4570 } 4571 4572 return !mStandby && !trackPaused; 4573} 4574 4575 4576bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4577{ 4578 Mutex::Autolock _l(mLock); 4579 return waitingAsyncCallback_l(); 4580} 4581 4582void AudioFlinger::OffloadThread::flushHw_l() 4583{ 4584 mOutput->stream->flush(mOutput->stream); 4585 // Flush anything still waiting in the mixbuffer 4586 mCurrentWriteLength = 0; 4587 mBytesRemaining = 0; 4588 mPausedWriteLength = 0; 4589 mPausedBytesRemaining = 0; 4590 mHwPaused = false; 4591 4592 if (mUseAsyncWrite) { 4593 // discard any pending drain or write ack by incrementing sequence 4594 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4595 mDrainSequence = (mDrainSequence + 2) & ~1; 4596 ALOG_ASSERT(mCallbackThread != 0); 4597 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4598 mCallbackThread->setDraining(mDrainSequence); 4599 } 4600} 4601 4602void AudioFlinger::OffloadThread::onAddNewTrack_l() 4603{ 4604 sp<Track> previousTrack = mPreviousTrack.promote(); 4605 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4606 4607 if (previousTrack != 0 && latestTrack != 0 && 4608 (previousTrack->sessionId() != latestTrack->sessionId())) { 4609 mFlushPending = true; 4610 } 4611 PlaybackThread::onAddNewTrack_l(); 4612} 4613 4614// ---------------------------------------------------------------------------- 4615 4616AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4617 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4618 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4619 DUPLICATING), 4620 mWaitTimeMs(UINT_MAX) 4621{ 4622 addOutputTrack(mainThread); 4623} 4624 4625AudioFlinger::DuplicatingThread::~DuplicatingThread() 4626{ 4627 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4628 mOutputTracks[i]->destroy(); 4629 } 4630} 4631 4632void AudioFlinger::DuplicatingThread::threadLoop_mix() 4633{ 4634 // mix buffers... 4635 if (outputsReady(outputTracks)) { 4636 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4637 } else { 4638 memset(mSinkBuffer, 0, mSinkBufferSize); 4639 } 4640 sleepTime = 0; 4641 writeFrames = mNormalFrameCount; 4642 mCurrentWriteLength = mSinkBufferSize; 4643 standbyTime = systemTime() + standbyDelay; 4644} 4645 4646void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4647{ 4648 if (sleepTime == 0) { 4649 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4650 sleepTime = activeSleepTime; 4651 } else { 4652 sleepTime = idleSleepTime; 4653 } 4654 } else if (mBytesWritten != 0) { 4655 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4656 writeFrames = mNormalFrameCount; 4657 memset(mSinkBuffer, 0, mSinkBufferSize); 4658 } else { 4659 // flush remaining overflow buffers in output tracks 4660 writeFrames = 0; 4661 } 4662 sleepTime = 0; 4663 } 4664} 4665 4666ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4667{ 4668 for (size_t i = 0; i < outputTracks.size(); i++) { 4669 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4670 // for delivery downstream as needed. This in-place conversion is safe as 4671 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4672 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4673 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4674 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4675 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4676 } 4677 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4678 } 4679 mStandby = false; 4680 return (ssize_t)mSinkBufferSize; 4681} 4682 4683void AudioFlinger::DuplicatingThread::threadLoop_standby() 4684{ 4685 // DuplicatingThread implements standby by stopping all tracks 4686 for (size_t i = 0; i < outputTracks.size(); i++) { 4687 outputTracks[i]->stop(); 4688 } 4689} 4690 4691void AudioFlinger::DuplicatingThread::saveOutputTracks() 4692{ 4693 outputTracks = mOutputTracks; 4694} 4695 4696void AudioFlinger::DuplicatingThread::clearOutputTracks() 4697{ 4698 outputTracks.clear(); 4699} 4700 4701void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4702{ 4703 Mutex::Autolock _l(mLock); 4704 // FIXME explain this formula 4705 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4706 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4707 // due to current usage case and restrictions on the AudioBufferProvider. 4708 // Actual buffer conversion is done in threadLoop_write(). 4709 // 4710 // TODO: This may change in the future, depending on multichannel 4711 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4712 OutputTrack *outputTrack = new OutputTrack(thread, 4713 this, 4714 mSampleRate, 4715 AUDIO_FORMAT_PCM_16_BIT, 4716 mChannelMask, 4717 frameCount, 4718 IPCThreadState::self()->getCallingUid()); 4719 if (outputTrack->cblk() != NULL) { 4720 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4721 mOutputTracks.add(outputTrack); 4722 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4723 updateWaitTime_l(); 4724 } 4725} 4726 4727void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4728{ 4729 Mutex::Autolock _l(mLock); 4730 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4731 if (mOutputTracks[i]->thread() == thread) { 4732 mOutputTracks[i]->destroy(); 4733 mOutputTracks.removeAt(i); 4734 updateWaitTime_l(); 4735 return; 4736 } 4737 } 4738 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4739} 4740 4741// caller must hold mLock 4742void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4743{ 4744 mWaitTimeMs = UINT_MAX; 4745 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4746 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4747 if (strong != 0) { 4748 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4749 if (waitTimeMs < mWaitTimeMs) { 4750 mWaitTimeMs = waitTimeMs; 4751 } 4752 } 4753 } 4754} 4755 4756 4757bool AudioFlinger::DuplicatingThread::outputsReady( 4758 const SortedVector< sp<OutputTrack> > &outputTracks) 4759{ 4760 for (size_t i = 0; i < outputTracks.size(); i++) { 4761 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4762 if (thread == 0) { 4763 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4764 outputTracks[i].get()); 4765 return false; 4766 } 4767 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4768 // see note at standby() declaration 4769 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4770 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4771 thread.get()); 4772 return false; 4773 } 4774 } 4775 return true; 4776} 4777 4778uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4779{ 4780 return (mWaitTimeMs * 1000) / 2; 4781} 4782 4783void AudioFlinger::DuplicatingThread::cacheParameters_l() 4784{ 4785 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4786 updateWaitTime_l(); 4787 4788 MixerThread::cacheParameters_l(); 4789} 4790 4791// ---------------------------------------------------------------------------- 4792// Record 4793// ---------------------------------------------------------------------------- 4794 4795AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4796 AudioStreamIn *input, 4797 audio_io_handle_t id, 4798 audio_devices_t outDevice, 4799 audio_devices_t inDevice 4800#ifdef TEE_SINK 4801 , const sp<NBAIO_Sink>& teeSink 4802#endif 4803 ) : 4804 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4805 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4806 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4807 mRsmpInRear(0) 4808#ifdef TEE_SINK 4809 , mTeeSink(teeSink) 4810#endif 4811 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4812 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4813 // mFastCapture below 4814 , mFastCaptureFutex(0) 4815 // mInputSource 4816 // mPipeSink 4817 // mPipeSource 4818 , mPipeFramesP2(0) 4819 // mPipeMemory 4820 // mFastCaptureNBLogWriter 4821 , mFastTrackAvail(false) 4822{ 4823 snprintf(mName, kNameLength, "AudioIn_%X", id); 4824 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4825 4826 readInputParameters_l(); 4827 4828 // create an NBAIO source for the HAL input stream, and negotiate 4829 mInputSource = new AudioStreamInSource(input->stream); 4830 size_t numCounterOffers = 0; 4831 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4832 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4833 ALOG_ASSERT(index == 0); 4834 4835 // initialize fast capture depending on configuration 4836 bool initFastCapture; 4837 switch (kUseFastCapture) { 4838 case FastCapture_Never: 4839 initFastCapture = false; 4840 break; 4841 case FastCapture_Always: 4842 initFastCapture = true; 4843 break; 4844 case FastCapture_Static: 4845 uint32_t primaryOutputSampleRate; 4846 { 4847 AutoMutex _l(audioFlinger->mHardwareLock); 4848 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4849 } 4850 initFastCapture = 4851 // either capture sample rate is same as (a reasonable) primary output sample rate 4852 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4853 (mSampleRate == primaryOutputSampleRate)) || 4854 // or primary output sample rate is unknown, and capture sample rate is reasonable 4855 ((primaryOutputSampleRate == 0) && 4856 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4857 // and the buffer size is < 12 ms 4858 (mFrameCount * 1000) / mSampleRate < 12; 4859 break; 4860 // case FastCapture_Dynamic: 4861 } 4862 4863 if (initFastCapture) { 4864 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4865 NBAIO_Format format = mInputSource->format(); 4866 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4867 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4868 void *pipeBuffer; 4869 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4870 sp<IMemory> pipeMemory; 4871 if ((roHeap == 0) || 4872 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4873 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4874 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4875 goto failed; 4876 } 4877 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4878 memset(pipeBuffer, 0, pipeSize); 4879 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4880 const NBAIO_Format offers[1] = {format}; 4881 size_t numCounterOffers = 0; 4882 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4883 ALOG_ASSERT(index == 0); 4884 mPipeSink = pipe; 4885 PipeReader *pipeReader = new PipeReader(*pipe); 4886 numCounterOffers = 0; 4887 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4888 ALOG_ASSERT(index == 0); 4889 mPipeSource = pipeReader; 4890 mPipeFramesP2 = pipeFramesP2; 4891 mPipeMemory = pipeMemory; 4892 4893 // create fast capture 4894 mFastCapture = new FastCapture(); 4895 FastCaptureStateQueue *sq = mFastCapture->sq(); 4896#ifdef STATE_QUEUE_DUMP 4897 // FIXME 4898#endif 4899 FastCaptureState *state = sq->begin(); 4900 state->mCblk = NULL; 4901 state->mInputSource = mInputSource.get(); 4902 state->mInputSourceGen++; 4903 state->mPipeSink = pipe; 4904 state->mPipeSinkGen++; 4905 state->mFrameCount = mFrameCount; 4906 state->mCommand = FastCaptureState::COLD_IDLE; 4907 // already done in constructor initialization list 4908 //mFastCaptureFutex = 0; 4909 state->mColdFutexAddr = &mFastCaptureFutex; 4910 state->mColdGen++; 4911 state->mDumpState = &mFastCaptureDumpState; 4912#ifdef TEE_SINK 4913 // FIXME 4914#endif 4915 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4916 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4917 sq->end(); 4918 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4919 4920 // start the fast capture 4921 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4922 pid_t tid = mFastCapture->getTid(); 4923 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4924 if (err != 0) { 4925 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4926 kPriorityFastCapture, getpid_cached, tid, err); 4927 } 4928 4929#ifdef AUDIO_WATCHDOG 4930 // FIXME 4931#endif 4932 4933 mFastTrackAvail = true; 4934 } 4935failed: ; 4936 4937 // FIXME mNormalSource 4938} 4939 4940 4941AudioFlinger::RecordThread::~RecordThread() 4942{ 4943 if (mFastCapture != 0) { 4944 FastCaptureStateQueue *sq = mFastCapture->sq(); 4945 FastCaptureState *state = sq->begin(); 4946 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4947 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4948 if (old == -1) { 4949 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4950 } 4951 } 4952 state->mCommand = FastCaptureState::EXIT; 4953 sq->end(); 4954 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4955 mFastCapture->join(); 4956 mFastCapture.clear(); 4957 } 4958 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4959 mAudioFlinger->unregisterWriter(mNBLogWriter); 4960 delete[] mRsmpInBuffer; 4961} 4962 4963void AudioFlinger::RecordThread::onFirstRef() 4964{ 4965 run(mName, PRIORITY_URGENT_AUDIO); 4966} 4967 4968bool AudioFlinger::RecordThread::threadLoop() 4969{ 4970 nsecs_t lastWarning = 0; 4971 4972 inputStandBy(); 4973 4974reacquire_wakelock: 4975 sp<RecordTrack> activeTrack; 4976 int activeTracksGen; 4977 { 4978 Mutex::Autolock _l(mLock); 4979 size_t size = mActiveTracks.size(); 4980 activeTracksGen = mActiveTracksGen; 4981 if (size > 0) { 4982 // FIXME an arbitrary choice 4983 activeTrack = mActiveTracks[0]; 4984 acquireWakeLock_l(activeTrack->uid()); 4985 if (size > 1) { 4986 SortedVector<int> tmp; 4987 for (size_t i = 0; i < size; i++) { 4988 tmp.add(mActiveTracks[i]->uid()); 4989 } 4990 updateWakeLockUids_l(tmp); 4991 } 4992 } else { 4993 acquireWakeLock_l(-1); 4994 } 4995 } 4996 4997 // used to request a deferred sleep, to be executed later while mutex is unlocked 4998 uint32_t sleepUs = 0; 4999 5000 // loop while there is work to do 5001 for (;;) { 5002 Vector< sp<EffectChain> > effectChains; 5003 5004 // sleep with mutex unlocked 5005 if (sleepUs > 0) { 5006 usleep(sleepUs); 5007 sleepUs = 0; 5008 } 5009 5010 // activeTracks accumulates a copy of a subset of mActiveTracks 5011 Vector< sp<RecordTrack> > activeTracks; 5012 5013 // reference to the (first and only) active fast track 5014 sp<RecordTrack> fastTrack; 5015 5016 // reference to a fast track which is about to be removed 5017 sp<RecordTrack> fastTrackToRemove; 5018 5019 { // scope for mLock 5020 Mutex::Autolock _l(mLock); 5021 5022 processConfigEvents_l(); 5023 5024 // check exitPending here because checkForNewParameters_l() and 5025 // checkForNewParameters_l() can temporarily release mLock 5026 if (exitPending()) { 5027 break; 5028 } 5029 5030 // if no active track(s), then standby and release wakelock 5031 size_t size = mActiveTracks.size(); 5032 if (size == 0) { 5033 standbyIfNotAlreadyInStandby(); 5034 // exitPending() can't become true here 5035 releaseWakeLock_l(); 5036 ALOGV("RecordThread: loop stopping"); 5037 // go to sleep 5038 mWaitWorkCV.wait(mLock); 5039 ALOGV("RecordThread: loop starting"); 5040 goto reacquire_wakelock; 5041 } 5042 5043 if (mActiveTracksGen != activeTracksGen) { 5044 activeTracksGen = mActiveTracksGen; 5045 SortedVector<int> tmp; 5046 for (size_t i = 0; i < size; i++) { 5047 tmp.add(mActiveTracks[i]->uid()); 5048 } 5049 updateWakeLockUids_l(tmp); 5050 } 5051 5052 bool doBroadcast = false; 5053 for (size_t i = 0; i < size; ) { 5054 5055 activeTrack = mActiveTracks[i]; 5056 if (activeTrack->isTerminated()) { 5057 if (activeTrack->isFastTrack()) { 5058 ALOG_ASSERT(fastTrackToRemove == 0); 5059 fastTrackToRemove = activeTrack; 5060 } 5061 removeTrack_l(activeTrack); 5062 mActiveTracks.remove(activeTrack); 5063 mActiveTracksGen++; 5064 size--; 5065 continue; 5066 } 5067 5068 TrackBase::track_state activeTrackState = activeTrack->mState; 5069 switch (activeTrackState) { 5070 5071 case TrackBase::PAUSING: 5072 mActiveTracks.remove(activeTrack); 5073 mActiveTracksGen++; 5074 doBroadcast = true; 5075 size--; 5076 continue; 5077 5078 case TrackBase::STARTING_1: 5079 sleepUs = 10000; 5080 i++; 5081 continue; 5082 5083 case TrackBase::STARTING_2: 5084 doBroadcast = true; 5085 mStandby = false; 5086 activeTrack->mState = TrackBase::ACTIVE; 5087 break; 5088 5089 case TrackBase::ACTIVE: 5090 break; 5091 5092 case TrackBase::IDLE: 5093 i++; 5094 continue; 5095 5096 default: 5097 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5098 } 5099 5100 activeTracks.add(activeTrack); 5101 i++; 5102 5103 if (activeTrack->isFastTrack()) { 5104 ALOG_ASSERT(!mFastTrackAvail); 5105 ALOG_ASSERT(fastTrack == 0); 5106 fastTrack = activeTrack; 5107 } 5108 } 5109 if (doBroadcast) { 5110 mStartStopCond.broadcast(); 5111 } 5112 5113 // sleep if there are no active tracks to process 5114 if (activeTracks.size() == 0) { 5115 if (sleepUs == 0) { 5116 sleepUs = kRecordThreadSleepUs; 5117 } 5118 continue; 5119 } 5120 sleepUs = 0; 5121 5122 lockEffectChains_l(effectChains); 5123 } 5124 5125 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5126 5127 size_t size = effectChains.size(); 5128 for (size_t i = 0; i < size; i++) { 5129 // thread mutex is not locked, but effect chain is locked 5130 effectChains[i]->process_l(); 5131 } 5132 5133 // Push a new fast capture state if fast capture is not already running, or cblk change 5134 if (mFastCapture != 0) { 5135 FastCaptureStateQueue *sq = mFastCapture->sq(); 5136 FastCaptureState *state = sq->begin(); 5137 bool didModify = false; 5138 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5139 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5140 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5141 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5142 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5143 if (old == -1) { 5144 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5145 } 5146 } 5147 state->mCommand = FastCaptureState::READ_WRITE; 5148#if 0 // FIXME 5149 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5150 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5151#endif 5152 didModify = true; 5153 } 5154 audio_track_cblk_t *cblkOld = state->mCblk; 5155 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5156 if (cblkNew != cblkOld) { 5157 state->mCblk = cblkNew; 5158 // block until acked if removing a fast track 5159 if (cblkOld != NULL) { 5160 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5161 } 5162 didModify = true; 5163 } 5164 sq->end(didModify); 5165 if (didModify) { 5166 sq->push(block); 5167#if 0 5168 if (kUseFastCapture == FastCapture_Dynamic) { 5169 mNormalSource = mPipeSource; 5170 } 5171#endif 5172 } 5173 } 5174 5175 // now run the fast track destructor with thread mutex unlocked 5176 fastTrackToRemove.clear(); 5177 5178 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5179 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5180 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5181 // If destination is non-contiguous, first read past the nominal end of buffer, then 5182 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5183 5184 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5185 ssize_t framesRead; 5186 5187 // If an NBAIO source is present, use it to read the normal capture's data 5188 if (mPipeSource != 0) { 5189 size_t framesToRead = mBufferSize / mFrameSize; 5190 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5191 framesToRead, AudioBufferProvider::kInvalidPTS); 5192 if (framesRead == 0) { 5193 // since pipe is non-blocking, simulate blocking input 5194 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5195 } 5196 // otherwise use the HAL / AudioStreamIn directly 5197 } else { 5198 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5199 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5200 if (bytesRead < 0) { 5201 framesRead = bytesRead; 5202 } else { 5203 framesRead = bytesRead / mFrameSize; 5204 } 5205 } 5206 5207 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5208 ALOGE("read failed: framesRead=%d", framesRead); 5209 // Force input into standby so that it tries to recover at next read attempt 5210 inputStandBy(); 5211 sleepUs = kRecordThreadSleepUs; 5212 } 5213 if (framesRead <= 0) { 5214 goto unlock; 5215 } 5216 ALOG_ASSERT(framesRead > 0); 5217 5218 if (mTeeSink != 0) { 5219 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5220 } 5221 // If destination is non-contiguous, we now correct for reading past end of buffer. 5222 { 5223 size_t part1 = mRsmpInFramesP2 - rear; 5224 if ((size_t) framesRead > part1) { 5225 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5226 (framesRead - part1) * mFrameSize); 5227 } 5228 } 5229 rear = mRsmpInRear += framesRead; 5230 5231 size = activeTracks.size(); 5232 // loop over each active track 5233 for (size_t i = 0; i < size; i++) { 5234 activeTrack = activeTracks[i]; 5235 5236 // skip fast tracks, as those are handled directly by FastCapture 5237 if (activeTrack->isFastTrack()) { 5238 continue; 5239 } 5240 5241 enum { 5242 OVERRUN_UNKNOWN, 5243 OVERRUN_TRUE, 5244 OVERRUN_FALSE 5245 } overrun = OVERRUN_UNKNOWN; 5246 5247 // loop over getNextBuffer to handle circular sink 5248 for (;;) { 5249 5250 activeTrack->mSink.frameCount = ~0; 5251 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5252 size_t framesOut = activeTrack->mSink.frameCount; 5253 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5254 5255 int32_t front = activeTrack->mRsmpInFront; 5256 ssize_t filled = rear - front; 5257 size_t framesIn; 5258 5259 if (filled < 0) { 5260 // should not happen, but treat like a massive overrun and re-sync 5261 framesIn = 0; 5262 activeTrack->mRsmpInFront = rear; 5263 overrun = OVERRUN_TRUE; 5264 } else if ((size_t) filled <= mRsmpInFrames) { 5265 framesIn = (size_t) filled; 5266 } else { 5267 // client is not keeping up with server, but give it latest data 5268 framesIn = mRsmpInFrames; 5269 activeTrack->mRsmpInFront = front = rear - framesIn; 5270 overrun = OVERRUN_TRUE; 5271 } 5272 5273 if (framesOut == 0 || framesIn == 0) { 5274 break; 5275 } 5276 5277 if (activeTrack->mResampler == NULL) { 5278 // no resampling 5279 if (framesIn > framesOut) { 5280 framesIn = framesOut; 5281 } else { 5282 framesOut = framesIn; 5283 } 5284 int8_t *dst = activeTrack->mSink.i8; 5285 while (framesIn > 0) { 5286 front &= mRsmpInFramesP2 - 1; 5287 size_t part1 = mRsmpInFramesP2 - front; 5288 if (part1 > framesIn) { 5289 part1 = framesIn; 5290 } 5291 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5292 if (mChannelCount == activeTrack->mChannelCount) { 5293 memcpy(dst, src, part1 * mFrameSize); 5294 } else if (mChannelCount == 1) { 5295 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5296 part1); 5297 } else { 5298 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5299 part1); 5300 } 5301 dst += part1 * activeTrack->mFrameSize; 5302 front += part1; 5303 framesIn -= part1; 5304 } 5305 activeTrack->mRsmpInFront += framesOut; 5306 5307 } else { 5308 // resampling 5309 // FIXME framesInNeeded should really be part of resampler API, and should 5310 // depend on the SRC ratio 5311 // to keep mRsmpInBuffer full so resampler always has sufficient input 5312 size_t framesInNeeded; 5313 // FIXME only re-calculate when it changes, and optimize for common ratios 5314 // Do not precompute in/out because floating point is not associative 5315 // e.g. a*b/c != a*(b/c). 5316 const double in(mSampleRate); 5317 const double out(activeTrack->mSampleRate); 5318 framesInNeeded = ceil(framesOut * in / out) + 1; 5319 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5320 framesInNeeded, framesOut, in / out); 5321 // Although we theoretically have framesIn in circular buffer, some of those are 5322 // unreleased frames, and thus must be discounted for purpose of budgeting. 5323 size_t unreleased = activeTrack->mRsmpInUnrel; 5324 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5325 if (framesIn < framesInNeeded) { 5326 ALOGV("not enough to resample: have %u frames in but need %u in to " 5327 "produce %u out given in/out ratio of %.4g", 5328 framesIn, framesInNeeded, framesOut, in / out); 5329 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5330 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5331 if (newFramesOut == 0) { 5332 break; 5333 } 5334 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5335 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5336 framesInNeeded, newFramesOut, out / in); 5337 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5338 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5339 "given in/out ratio of %.4g", 5340 framesIn, framesInNeeded, newFramesOut, in / out); 5341 framesOut = newFramesOut; 5342 } else { 5343 ALOGV("success 1: have %u in and need %u in to produce %u out " 5344 "given in/out ratio of %.4g", 5345 framesIn, framesInNeeded, framesOut, in / out); 5346 } 5347 5348 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5349 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5350 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5351 delete[] activeTrack->mRsmpOutBuffer; 5352 // resampler always outputs stereo 5353 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5354 activeTrack->mRsmpOutFrameCount = framesOut; 5355 } 5356 5357 // resampler accumulates, but we only have one source track 5358 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5359 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5360 // FIXME how about having activeTrack implement this interface itself? 5361 activeTrack->mResamplerBufferProvider 5362 /*this*/ /* AudioBufferProvider* */); 5363 // ditherAndClamp() works as long as all buffers returned by 5364 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5365 if (activeTrack->mChannelCount == 1) { 5366 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5367 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5368 framesOut); 5369 // the resampler always outputs stereo samples: 5370 // do post stereo to mono conversion 5371 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5372 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5373 } else { 5374 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5375 activeTrack->mRsmpOutBuffer, framesOut); 5376 } 5377 // now done with mRsmpOutBuffer 5378 5379 } 5380 5381 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5382 overrun = OVERRUN_FALSE; 5383 } 5384 5385 if (activeTrack->mFramesToDrop == 0) { 5386 if (framesOut > 0) { 5387 activeTrack->mSink.frameCount = framesOut; 5388 activeTrack->releaseBuffer(&activeTrack->mSink); 5389 } 5390 } else { 5391 // FIXME could do a partial drop of framesOut 5392 if (activeTrack->mFramesToDrop > 0) { 5393 activeTrack->mFramesToDrop -= framesOut; 5394 if (activeTrack->mFramesToDrop <= 0) { 5395 activeTrack->clearSyncStartEvent(); 5396 } 5397 } else { 5398 activeTrack->mFramesToDrop += framesOut; 5399 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5400 activeTrack->mSyncStartEvent->isCancelled()) { 5401 ALOGW("Synced record %s, session %d, trigger session %d", 5402 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5403 activeTrack->sessionId(), 5404 (activeTrack->mSyncStartEvent != 0) ? 5405 activeTrack->mSyncStartEvent->triggerSession() : 0); 5406 activeTrack->clearSyncStartEvent(); 5407 } 5408 } 5409 } 5410 5411 if (framesOut == 0) { 5412 break; 5413 } 5414 } 5415 5416 switch (overrun) { 5417 case OVERRUN_TRUE: 5418 // client isn't retrieving buffers fast enough 5419 if (!activeTrack->setOverflow()) { 5420 nsecs_t now = systemTime(); 5421 // FIXME should lastWarning per track? 5422 if ((now - lastWarning) > kWarningThrottleNs) { 5423 ALOGW("RecordThread: buffer overflow"); 5424 lastWarning = now; 5425 } 5426 } 5427 break; 5428 case OVERRUN_FALSE: 5429 activeTrack->clearOverflow(); 5430 break; 5431 case OVERRUN_UNKNOWN: 5432 break; 5433 } 5434 5435 } 5436 5437unlock: 5438 // enable changes in effect chain 5439 unlockEffectChains(effectChains); 5440 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5441 } 5442 5443 standbyIfNotAlreadyInStandby(); 5444 5445 { 5446 Mutex::Autolock _l(mLock); 5447 for (size_t i = 0; i < mTracks.size(); i++) { 5448 sp<RecordTrack> track = mTracks[i]; 5449 track->invalidate(); 5450 } 5451 mActiveTracks.clear(); 5452 mActiveTracksGen++; 5453 mStartStopCond.broadcast(); 5454 } 5455 5456 releaseWakeLock(); 5457 5458 ALOGV("RecordThread %p exiting", this); 5459 return false; 5460} 5461 5462void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5463{ 5464 if (!mStandby) { 5465 inputStandBy(); 5466 mStandby = true; 5467 } 5468} 5469 5470void AudioFlinger::RecordThread::inputStandBy() 5471{ 5472 // Idle the fast capture if it's currently running 5473 if (mFastCapture != 0) { 5474 FastCaptureStateQueue *sq = mFastCapture->sq(); 5475 FastCaptureState *state = sq->begin(); 5476 if (!(state->mCommand & FastCaptureState::IDLE)) { 5477 state->mCommand = FastCaptureState::COLD_IDLE; 5478 state->mColdFutexAddr = &mFastCaptureFutex; 5479 state->mColdGen++; 5480 mFastCaptureFutex = 0; 5481 sq->end(); 5482 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5483 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5484#if 0 5485 if (kUseFastCapture == FastCapture_Dynamic) { 5486 // FIXME 5487 } 5488#endif 5489#ifdef AUDIO_WATCHDOG 5490 // FIXME 5491#endif 5492 } else { 5493 sq->end(false /*didModify*/); 5494 } 5495 } 5496 mInput->stream->common.standby(&mInput->stream->common); 5497} 5498 5499// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5500sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5501 const sp<AudioFlinger::Client>& client, 5502 uint32_t sampleRate, 5503 audio_format_t format, 5504 audio_channel_mask_t channelMask, 5505 size_t *pFrameCount, 5506 int sessionId, 5507 size_t *notificationFrames, 5508 int uid, 5509 IAudioFlinger::track_flags_t *flags, 5510 pid_t tid, 5511 status_t *status) 5512{ 5513 size_t frameCount = *pFrameCount; 5514 sp<RecordTrack> track; 5515 status_t lStatus; 5516 5517 // client expresses a preference for FAST, but we get the final say 5518 if (*flags & IAudioFlinger::TRACK_FAST) { 5519 if ( 5520 // use case: callback handler 5521 (tid != -1) && 5522 // frame count is not specified, or is exactly the pipe depth 5523 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5524 // PCM data 5525 audio_is_linear_pcm(format) && 5526 // native format 5527 (format == mFormat) && 5528 // native channel mask 5529 (channelMask == mChannelMask) && 5530 // native hardware sample rate 5531 (sampleRate == mSampleRate) && 5532 // record thread has an associated fast capture 5533 hasFastCapture() && 5534 // there are sufficient fast track slots available 5535 mFastTrackAvail 5536 ) { 5537 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5538 frameCount, mFrameCount); 5539 } else { 5540 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5541 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5542 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5543 frameCount, mFrameCount, mPipeFramesP2, 5544 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5545 hasFastCapture(), tid, mFastTrackAvail); 5546 *flags &= ~IAudioFlinger::TRACK_FAST; 5547 } 5548 } 5549 5550 // compute track buffer size in frames, and suggest the notification frame count 5551 if (*flags & IAudioFlinger::TRACK_FAST) { 5552 // fast track: frame count is exactly the pipe depth 5553 frameCount = mPipeFramesP2; 5554 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5555 *notificationFrames = mFrameCount; 5556 } else { 5557 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5558 // or 20 ms if there is a fast capture 5559 // TODO This could be a roundupRatio inline, and const 5560 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5561 * sampleRate + mSampleRate - 1) / mSampleRate; 5562 // minimum number of notification periods is at least kMinNotifications, 5563 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5564 static const size_t kMinNotifications = 3; 5565 static const uint32_t kMinMs = 30; 5566 // TODO This could be a roundupRatio inline 5567 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5568 // TODO This could be a roundupRatio inline 5569 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5570 maxNotificationFrames; 5571 const size_t minFrameCount = maxNotificationFrames * 5572 max(kMinNotifications, minNotificationsByMs); 5573 frameCount = max(frameCount, minFrameCount); 5574 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5575 *notificationFrames = maxNotificationFrames; 5576 } 5577 } 5578 *pFrameCount = frameCount; 5579 5580 lStatus = initCheck(); 5581 if (lStatus != NO_ERROR) { 5582 ALOGE("createRecordTrack_l() audio driver not initialized"); 5583 goto Exit; 5584 } 5585 5586 { // scope for mLock 5587 Mutex::Autolock _l(mLock); 5588 5589 track = new RecordTrack(this, client, sampleRate, 5590 format, channelMask, frameCount, NULL, sessionId, uid, 5591 *flags, TrackBase::TYPE_DEFAULT); 5592 5593 lStatus = track->initCheck(); 5594 if (lStatus != NO_ERROR) { 5595 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5596 // track must be cleared from the caller as the caller has the AF lock 5597 goto Exit; 5598 } 5599 mTracks.add(track); 5600 5601 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5602 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5603 mAudioFlinger->btNrecIsOff(); 5604 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5605 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5606 5607 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5608 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5609 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5610 // so ask activity manager to do this on our behalf 5611 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5612 } 5613 } 5614 5615 lStatus = NO_ERROR; 5616 5617Exit: 5618 *status = lStatus; 5619 return track; 5620} 5621 5622status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5623 AudioSystem::sync_event_t event, 5624 int triggerSession) 5625{ 5626 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5627 sp<ThreadBase> strongMe = this; 5628 status_t status = NO_ERROR; 5629 5630 if (event == AudioSystem::SYNC_EVENT_NONE) { 5631 recordTrack->clearSyncStartEvent(); 5632 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5633 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5634 triggerSession, 5635 recordTrack->sessionId(), 5636 syncStartEventCallback, 5637 recordTrack); 5638 // Sync event can be cancelled by the trigger session if the track is not in a 5639 // compatible state in which case we start record immediately 5640 if (recordTrack->mSyncStartEvent->isCancelled()) { 5641 recordTrack->clearSyncStartEvent(); 5642 } else { 5643 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5644 recordTrack->mFramesToDrop = - 5645 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5646 } 5647 } 5648 5649 { 5650 // This section is a rendezvous between binder thread executing start() and RecordThread 5651 AutoMutex lock(mLock); 5652 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5653 if (recordTrack->mState == TrackBase::PAUSING) { 5654 ALOGV("active record track PAUSING -> ACTIVE"); 5655 recordTrack->mState = TrackBase::ACTIVE; 5656 } else { 5657 ALOGV("active record track state %d", recordTrack->mState); 5658 } 5659 return status; 5660 } 5661 5662 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5663 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5664 // or using a separate command thread 5665 recordTrack->mState = TrackBase::STARTING_1; 5666 mActiveTracks.add(recordTrack); 5667 mActiveTracksGen++; 5668 status_t status = NO_ERROR; 5669 if (recordTrack->isExternalTrack()) { 5670 mLock.unlock(); 5671 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5672 mLock.lock(); 5673 // FIXME should verify that recordTrack is still in mActiveTracks 5674 if (status != NO_ERROR) { 5675 mActiveTracks.remove(recordTrack); 5676 mActiveTracksGen++; 5677 recordTrack->clearSyncStartEvent(); 5678 ALOGV("RecordThread::start error %d", status); 5679 return status; 5680 } 5681 } 5682 // Catch up with current buffer indices if thread is already running. 5683 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5684 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5685 // see previously buffered data before it called start(), but with greater risk of overrun. 5686 5687 recordTrack->mRsmpInFront = mRsmpInRear; 5688 recordTrack->mRsmpInUnrel = 0; 5689 // FIXME why reset? 5690 if (recordTrack->mResampler != NULL) { 5691 recordTrack->mResampler->reset(); 5692 } 5693 recordTrack->mState = TrackBase::STARTING_2; 5694 // signal thread to start 5695 mWaitWorkCV.broadcast(); 5696 if (mActiveTracks.indexOf(recordTrack) < 0) { 5697 ALOGV("Record failed to start"); 5698 status = BAD_VALUE; 5699 goto startError; 5700 } 5701 return status; 5702 } 5703 5704startError: 5705 if (recordTrack->isExternalTrack()) { 5706 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5707 } 5708 recordTrack->clearSyncStartEvent(); 5709 // FIXME I wonder why we do not reset the state here? 5710 return status; 5711} 5712 5713void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5714{ 5715 sp<SyncEvent> strongEvent = event.promote(); 5716 5717 if (strongEvent != 0) { 5718 sp<RefBase> ptr = strongEvent->cookie().promote(); 5719 if (ptr != 0) { 5720 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5721 recordTrack->handleSyncStartEvent(strongEvent); 5722 } 5723 } 5724} 5725 5726bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5727 ALOGV("RecordThread::stop"); 5728 AutoMutex _l(mLock); 5729 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5730 return false; 5731 } 5732 // note that threadLoop may still be processing the track at this point [without lock] 5733 recordTrack->mState = TrackBase::PAUSING; 5734 // do not wait for mStartStopCond if exiting 5735 if (exitPending()) { 5736 return true; 5737 } 5738 // FIXME incorrect usage of wait: no explicit predicate or loop 5739 mStartStopCond.wait(mLock); 5740 // if we have been restarted, recordTrack is in mActiveTracks here 5741 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5742 ALOGV("Record stopped OK"); 5743 return true; 5744 } 5745 return false; 5746} 5747 5748bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5749{ 5750 return false; 5751} 5752 5753status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5754{ 5755#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5756 if (!isValidSyncEvent(event)) { 5757 return BAD_VALUE; 5758 } 5759 5760 int eventSession = event->triggerSession(); 5761 status_t ret = NAME_NOT_FOUND; 5762 5763 Mutex::Autolock _l(mLock); 5764 5765 for (size_t i = 0; i < mTracks.size(); i++) { 5766 sp<RecordTrack> track = mTracks[i]; 5767 if (eventSession == track->sessionId()) { 5768 (void) track->setSyncEvent(event); 5769 ret = NO_ERROR; 5770 } 5771 } 5772 return ret; 5773#else 5774 return BAD_VALUE; 5775#endif 5776} 5777 5778// destroyTrack_l() must be called with ThreadBase::mLock held 5779void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5780{ 5781 track->terminate(); 5782 track->mState = TrackBase::STOPPED; 5783 // active tracks are removed by threadLoop() 5784 if (mActiveTracks.indexOf(track) < 0) { 5785 removeTrack_l(track); 5786 } 5787} 5788 5789void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5790{ 5791 mTracks.remove(track); 5792 // need anything related to effects here? 5793 if (track->isFastTrack()) { 5794 ALOG_ASSERT(!mFastTrackAvail); 5795 mFastTrackAvail = true; 5796 } 5797} 5798 5799void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5800{ 5801 dumpInternals(fd, args); 5802 dumpTracks(fd, args); 5803 dumpEffectChains(fd, args); 5804} 5805 5806void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5807{ 5808 dprintf(fd, "\nInput thread %p:\n", this); 5809 5810 if (mActiveTracks.size() > 0) { 5811 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5812 } else { 5813 dprintf(fd, " No active record clients\n"); 5814 } 5815 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5816 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5817 5818 dumpBase(fd, args); 5819} 5820 5821void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5822{ 5823 const size_t SIZE = 256; 5824 char buffer[SIZE]; 5825 String8 result; 5826 5827 size_t numtracks = mTracks.size(); 5828 size_t numactive = mActiveTracks.size(); 5829 size_t numactiveseen = 0; 5830 dprintf(fd, " %d Tracks", numtracks); 5831 if (numtracks) { 5832 dprintf(fd, " of which %d are active\n", numactive); 5833 RecordTrack::appendDumpHeader(result); 5834 for (size_t i = 0; i < numtracks ; ++i) { 5835 sp<RecordTrack> track = mTracks[i]; 5836 if (track != 0) { 5837 bool active = mActiveTracks.indexOf(track) >= 0; 5838 if (active) { 5839 numactiveseen++; 5840 } 5841 track->dump(buffer, SIZE, active); 5842 result.append(buffer); 5843 } 5844 } 5845 } else { 5846 dprintf(fd, "\n"); 5847 } 5848 5849 if (numactiveseen != numactive) { 5850 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5851 " not in the track list\n"); 5852 result.append(buffer); 5853 RecordTrack::appendDumpHeader(result); 5854 for (size_t i = 0; i < numactive; ++i) { 5855 sp<RecordTrack> track = mActiveTracks[i]; 5856 if (mTracks.indexOf(track) < 0) { 5857 track->dump(buffer, SIZE, true); 5858 result.append(buffer); 5859 } 5860 } 5861 5862 } 5863 write(fd, result.string(), result.size()); 5864} 5865 5866// AudioBufferProvider interface 5867status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5868 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5869{ 5870 RecordTrack *activeTrack = mRecordTrack; 5871 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5872 if (threadBase == 0) { 5873 buffer->frameCount = 0; 5874 buffer->raw = NULL; 5875 return NOT_ENOUGH_DATA; 5876 } 5877 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5878 int32_t rear = recordThread->mRsmpInRear; 5879 int32_t front = activeTrack->mRsmpInFront; 5880 ssize_t filled = rear - front; 5881 // FIXME should not be P2 (don't want to increase latency) 5882 // FIXME if client not keeping up, discard 5883 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5884 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5885 front &= recordThread->mRsmpInFramesP2 - 1; 5886 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5887 if (part1 > (size_t) filled) { 5888 part1 = filled; 5889 } 5890 size_t ask = buffer->frameCount; 5891 ALOG_ASSERT(ask > 0); 5892 if (part1 > ask) { 5893 part1 = ask; 5894 } 5895 if (part1 == 0) { 5896 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5897 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5898 buffer->raw = NULL; 5899 buffer->frameCount = 0; 5900 activeTrack->mRsmpInUnrel = 0; 5901 return NOT_ENOUGH_DATA; 5902 } 5903 5904 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5905 buffer->frameCount = part1; 5906 activeTrack->mRsmpInUnrel = part1; 5907 return NO_ERROR; 5908} 5909 5910// AudioBufferProvider interface 5911void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5912 AudioBufferProvider::Buffer* buffer) 5913{ 5914 RecordTrack *activeTrack = mRecordTrack; 5915 size_t stepCount = buffer->frameCount; 5916 if (stepCount == 0) { 5917 return; 5918 } 5919 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5920 activeTrack->mRsmpInUnrel -= stepCount; 5921 activeTrack->mRsmpInFront += stepCount; 5922 buffer->raw = NULL; 5923 buffer->frameCount = 0; 5924} 5925 5926bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5927 status_t& status) 5928{ 5929 bool reconfig = false; 5930 5931 status = NO_ERROR; 5932 5933 audio_format_t reqFormat = mFormat; 5934 uint32_t samplingRate = mSampleRate; 5935 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5936 5937 AudioParameter param = AudioParameter(keyValuePair); 5938 int value; 5939 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5940 // channel count change can be requested. Do we mandate the first client defines the 5941 // HAL sampling rate and channel count or do we allow changes on the fly? 5942 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5943 samplingRate = value; 5944 reconfig = true; 5945 } 5946 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5947 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5948 status = BAD_VALUE; 5949 } else { 5950 reqFormat = (audio_format_t) value; 5951 reconfig = true; 5952 } 5953 } 5954 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5955 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5956 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5957 status = BAD_VALUE; 5958 } else { 5959 channelMask = mask; 5960 reconfig = true; 5961 } 5962 } 5963 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5964 // do not accept frame count changes if tracks are open as the track buffer 5965 // size depends on frame count and correct behavior would not be guaranteed 5966 // if frame count is changed after track creation 5967 if (mActiveTracks.size() > 0) { 5968 status = INVALID_OPERATION; 5969 } else { 5970 reconfig = true; 5971 } 5972 } 5973 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5974 // forward device change to effects that have requested to be 5975 // aware of attached audio device. 5976 for (size_t i = 0; i < mEffectChains.size(); i++) { 5977 mEffectChains[i]->setDevice_l(value); 5978 } 5979 5980 // store input device and output device but do not forward output device to audio HAL. 5981 // Note that status is ignored by the caller for output device 5982 // (see AudioFlinger::setParameters() 5983 if (audio_is_output_devices(value)) { 5984 mOutDevice = value; 5985 status = BAD_VALUE; 5986 } else { 5987 mInDevice = value; 5988 // disable AEC and NS if the device is a BT SCO headset supporting those 5989 // pre processings 5990 if (mTracks.size() > 0) { 5991 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5992 mAudioFlinger->btNrecIsOff(); 5993 for (size_t i = 0; i < mTracks.size(); i++) { 5994 sp<RecordTrack> track = mTracks[i]; 5995 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5996 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5997 } 5998 } 5999 } 6000 } 6001 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6002 mAudioSource != (audio_source_t)value) { 6003 // forward device change to effects that have requested to be 6004 // aware of attached audio device. 6005 for (size_t i = 0; i < mEffectChains.size(); i++) { 6006 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6007 } 6008 mAudioSource = (audio_source_t)value; 6009 } 6010 6011 if (status == NO_ERROR) { 6012 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6013 keyValuePair.string()); 6014 if (status == INVALID_OPERATION) { 6015 inputStandBy(); 6016 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6017 keyValuePair.string()); 6018 } 6019 if (reconfig) { 6020 if (status == BAD_VALUE && 6021 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6022 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6023 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6024 <= (2 * samplingRate)) && 6025 audio_channel_count_from_in_mask( 6026 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6027 (channelMask == AUDIO_CHANNEL_IN_MONO || 6028 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6029 status = NO_ERROR; 6030 } 6031 if (status == NO_ERROR) { 6032 readInputParameters_l(); 6033 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6034 } 6035 } 6036 } 6037 6038 return reconfig; 6039} 6040 6041String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6042{ 6043 Mutex::Autolock _l(mLock); 6044 if (initCheck() != NO_ERROR) { 6045 return String8(); 6046 } 6047 6048 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6049 const String8 out_s8(s); 6050 free(s); 6051 return out_s8; 6052} 6053 6054void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6055 AudioSystem::OutputDescriptor desc; 6056 const void *param2 = NULL; 6057 6058 switch (event) { 6059 case AudioSystem::INPUT_OPENED: 6060 case AudioSystem::INPUT_CONFIG_CHANGED: 6061 desc.channelMask = mChannelMask; 6062 desc.samplingRate = mSampleRate; 6063 desc.format = mFormat; 6064 desc.frameCount = mFrameCount; 6065 desc.latency = 0; 6066 param2 = &desc; 6067 break; 6068 6069 case AudioSystem::INPUT_CLOSED: 6070 default: 6071 break; 6072 } 6073 mAudioFlinger->audioConfigChanged(event, mId, param2); 6074} 6075 6076void AudioFlinger::RecordThread::readInputParameters_l() 6077{ 6078 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6079 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6080 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6081 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6082 mFormat = mHALFormat; 6083 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6084 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6085 } 6086 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6087 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6088 mFrameCount = mBufferSize / mFrameSize; 6089 // This is the formula for calculating the temporary buffer size. 6090 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6091 // 1 full output buffer, regardless of the alignment of the available input. 6092 // The value is somewhat arbitrary, and could probably be even larger. 6093 // A larger value should allow more old data to be read after a track calls start(), 6094 // without increasing latency. 6095 mRsmpInFrames = mFrameCount * 7; 6096 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6097 delete[] mRsmpInBuffer; 6098 6099 // TODO optimize audio capture buffer sizes ... 6100 // Here we calculate the size of the sliding buffer used as a source 6101 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6102 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6103 // be better to have it derived from the pipe depth in the long term. 6104 // The current value is higher than necessary. However it should not add to latency. 6105 6106 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6107 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6108 6109 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6110 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6111} 6112 6113uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6114{ 6115 Mutex::Autolock _l(mLock); 6116 if (initCheck() != NO_ERROR) { 6117 return 0; 6118 } 6119 6120 return mInput->stream->get_input_frames_lost(mInput->stream); 6121} 6122 6123uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6124{ 6125 Mutex::Autolock _l(mLock); 6126 uint32_t result = 0; 6127 if (getEffectChain_l(sessionId) != 0) { 6128 result = EFFECT_SESSION; 6129 } 6130 6131 for (size_t i = 0; i < mTracks.size(); ++i) { 6132 if (sessionId == mTracks[i]->sessionId()) { 6133 result |= TRACK_SESSION; 6134 break; 6135 } 6136 } 6137 6138 return result; 6139} 6140 6141KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6142{ 6143 KeyedVector<int, bool> ids; 6144 Mutex::Autolock _l(mLock); 6145 for (size_t j = 0; j < mTracks.size(); ++j) { 6146 sp<RecordThread::RecordTrack> track = mTracks[j]; 6147 int sessionId = track->sessionId(); 6148 if (ids.indexOfKey(sessionId) < 0) { 6149 ids.add(sessionId, true); 6150 } 6151 } 6152 return ids; 6153} 6154 6155AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6156{ 6157 Mutex::Autolock _l(mLock); 6158 AudioStreamIn *input = mInput; 6159 mInput = NULL; 6160 return input; 6161} 6162 6163// this method must always be called either with ThreadBase mLock held or inside the thread loop 6164audio_stream_t* AudioFlinger::RecordThread::stream() const 6165{ 6166 if (mInput == NULL) { 6167 return NULL; 6168 } 6169 return &mInput->stream->common; 6170} 6171 6172status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6173{ 6174 // only one chain per input thread 6175 if (mEffectChains.size() != 0) { 6176 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6177 return INVALID_OPERATION; 6178 } 6179 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6180 chain->setThread(this); 6181 chain->setInBuffer(NULL); 6182 chain->setOutBuffer(NULL); 6183 6184 checkSuspendOnAddEffectChain_l(chain); 6185 6186 mEffectChains.add(chain); 6187 6188 return NO_ERROR; 6189} 6190 6191size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6192{ 6193 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6194 ALOGW_IF(mEffectChains.size() != 1, 6195 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6196 chain.get(), mEffectChains.size(), this); 6197 if (mEffectChains.size() == 1) { 6198 mEffectChains.removeAt(0); 6199 } 6200 return 0; 6201} 6202 6203status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6204 audio_patch_handle_t *handle) 6205{ 6206 status_t status = NO_ERROR; 6207 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6208 // store new device and send to effects 6209 mInDevice = patch->sources[0].ext.device.type; 6210 for (size_t i = 0; i < mEffectChains.size(); i++) { 6211 mEffectChains[i]->setDevice_l(mInDevice); 6212 } 6213 6214 // disable AEC and NS if the device is a BT SCO headset supporting those 6215 // pre processings 6216 if (mTracks.size() > 0) { 6217 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6218 mAudioFlinger->btNrecIsOff(); 6219 for (size_t i = 0; i < mTracks.size(); i++) { 6220 sp<RecordTrack> track = mTracks[i]; 6221 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6222 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6223 } 6224 } 6225 6226 // store new source and send to effects 6227 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6228 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6229 for (size_t i = 0; i < mEffectChains.size(); i++) { 6230 mEffectChains[i]->setAudioSource_l(mAudioSource); 6231 } 6232 } 6233 6234 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6235 status = hwDevice->create_audio_patch(hwDevice, 6236 patch->num_sources, 6237 patch->sources, 6238 patch->num_sinks, 6239 patch->sinks, 6240 handle); 6241 } else { 6242 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6243 } 6244 return status; 6245} 6246 6247status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6248{ 6249 status_t status = NO_ERROR; 6250 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6251 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6252 status = hwDevice->release_audio_patch(hwDevice, handle); 6253 } else { 6254 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6255 } 6256 return status; 6257} 6258 6259void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6260{ 6261 Mutex::Autolock _l(mLock); 6262 mTracks.add(record); 6263} 6264 6265void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6266{ 6267 Mutex::Autolock _l(mLock); 6268 destroyTrack_l(record); 6269} 6270 6271void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6272{ 6273 ThreadBase::getAudioPortConfig(config); 6274 config->role = AUDIO_PORT_ROLE_SINK; 6275 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6276 config->ext.mix.usecase.source = mAudioSource; 6277} 6278 6279}; // namespace android 6280