Threads.cpp revision 955e24d3a8e218d4711cabc6558781e095011132
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 // mLatchD.mFramesReleased is set in threadloop_mix() 2130 mLatchDValid = true; 2131 } 2132 } 2133 // otherwise use the HAL / AudioStreamOut directly 2134 } else { 2135 // Direct output and offload threads 2136 2137 if (mUseAsyncWrite) { 2138 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2139 mWriteAckSequence += 2; 2140 mWriteAckSequence |= 1; 2141 ALOG_ASSERT(mCallbackThread != 0); 2142 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2143 } 2144 // FIXME We should have an implementation of timestamps for direct output threads. 2145 // They are used e.g for multichannel PCM playback over HDMI. 2146 bytesWritten = mOutput->stream->write(mOutput->stream, 2147 (char *)mSinkBuffer + offset, mBytesRemaining); 2148 if (mUseAsyncWrite && 2149 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2150 // do not wait for async callback in case of error of full write 2151 mWriteAckSequence &= ~1; 2152 ALOG_ASSERT(mCallbackThread != 0); 2153 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2154 } 2155 } 2156 2157 mNumWrites++; 2158 mInWrite = false; 2159 mStandby = false; 2160 return bytesWritten; 2161} 2162 2163void AudioFlinger::PlaybackThread::threadLoop_drain() 2164{ 2165 if (mOutput->stream->drain) { 2166 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2167 if (mUseAsyncWrite) { 2168 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2169 mDrainSequence |= 1; 2170 ALOG_ASSERT(mCallbackThread != 0); 2171 mCallbackThread->setDraining(mDrainSequence); 2172 } 2173 mOutput->stream->drain(mOutput->stream, 2174 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2175 : AUDIO_DRAIN_ALL); 2176 } 2177} 2178 2179void AudioFlinger::PlaybackThread::threadLoop_exit() 2180{ 2181 // Default implementation has nothing to do 2182} 2183 2184/* 2185The derived values that are cached: 2186 - mSinkBufferSize from frame count * frame size 2187 - activeSleepTime from activeSleepTimeUs() 2188 - idleSleepTime from idleSleepTimeUs() 2189 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2190 - maxPeriod from frame count and sample rate (MIXER only) 2191 2192The parameters that affect these derived values are: 2193 - frame count 2194 - frame size 2195 - sample rate 2196 - device type: A2DP or not 2197 - device latency 2198 - format: PCM or not 2199 - active sleep time 2200 - idle sleep time 2201*/ 2202 2203void AudioFlinger::PlaybackThread::cacheParameters_l() 2204{ 2205 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2206 activeSleepTime = activeSleepTimeUs(); 2207 idleSleepTime = idleSleepTimeUs(); 2208} 2209 2210void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2211{ 2212 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2213 this, streamType, mTracks.size()); 2214 Mutex::Autolock _l(mLock); 2215 2216 size_t size = mTracks.size(); 2217 for (size_t i = 0; i < size; i++) { 2218 sp<Track> t = mTracks[i]; 2219 if (t->streamType() == streamType) { 2220 t->invalidate(); 2221 } 2222 } 2223} 2224 2225status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2226{ 2227 int session = chain->sessionId(); 2228 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2229 ? mEffectBuffer : mSinkBuffer); 2230 bool ownsBuffer = false; 2231 2232 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2233 if (session > 0) { 2234 // Only one effect chain can be present in direct output thread and it uses 2235 // the sink buffer as input 2236 if (mType != DIRECT) { 2237 size_t numSamples = mNormalFrameCount * mChannelCount; 2238 buffer = new int16_t[numSamples]; 2239 memset(buffer, 0, numSamples * sizeof(int16_t)); 2240 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2241 ownsBuffer = true; 2242 } 2243 2244 // Attach all tracks with same session ID to this chain. 2245 for (size_t i = 0; i < mTracks.size(); ++i) { 2246 sp<Track> track = mTracks[i]; 2247 if (session == track->sessionId()) { 2248 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2249 buffer); 2250 track->setMainBuffer(buffer); 2251 chain->incTrackCnt(); 2252 } 2253 } 2254 2255 // indicate all active tracks in the chain 2256 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2257 sp<Track> track = mActiveTracks[i].promote(); 2258 if (track == 0) { 2259 continue; 2260 } 2261 if (session == track->sessionId()) { 2262 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2263 chain->incActiveTrackCnt(); 2264 } 2265 } 2266 } 2267 chain->setThread(this); 2268 chain->setInBuffer(buffer, ownsBuffer); 2269 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2270 ? mEffectBuffer : mSinkBuffer)); 2271 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2272 // chains list in order to be processed last as it contains output stage effects 2273 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2274 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2275 // after track specific effects and before output stage 2276 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2277 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2278 // Effect chain for other sessions are inserted at beginning of effect 2279 // chains list to be processed before output mix effects. Relative order between other 2280 // sessions is not important 2281 size_t size = mEffectChains.size(); 2282 size_t i = 0; 2283 for (i = 0; i < size; i++) { 2284 if (mEffectChains[i]->sessionId() < session) { 2285 break; 2286 } 2287 } 2288 mEffectChains.insertAt(chain, i); 2289 checkSuspendOnAddEffectChain_l(chain); 2290 2291 return NO_ERROR; 2292} 2293 2294size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2295{ 2296 int session = chain->sessionId(); 2297 2298 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2299 2300 for (size_t i = 0; i < mEffectChains.size(); i++) { 2301 if (chain == mEffectChains[i]) { 2302 mEffectChains.removeAt(i); 2303 // detach all active tracks from the chain 2304 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2305 sp<Track> track = mActiveTracks[i].promote(); 2306 if (track == 0) { 2307 continue; 2308 } 2309 if (session == track->sessionId()) { 2310 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2311 chain.get(), session); 2312 chain->decActiveTrackCnt(); 2313 } 2314 } 2315 2316 // detach all tracks with same session ID from this chain 2317 for (size_t i = 0; i < mTracks.size(); ++i) { 2318 sp<Track> track = mTracks[i]; 2319 if (session == track->sessionId()) { 2320 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2321 chain->decTrackCnt(); 2322 } 2323 } 2324 break; 2325 } 2326 } 2327 return mEffectChains.size(); 2328} 2329 2330status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2331 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2332{ 2333 Mutex::Autolock _l(mLock); 2334 return attachAuxEffect_l(track, EffectId); 2335} 2336 2337status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2338 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2339{ 2340 status_t status = NO_ERROR; 2341 2342 if (EffectId == 0) { 2343 track->setAuxBuffer(0, NULL); 2344 } else { 2345 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2346 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2347 if (effect != 0) { 2348 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2349 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2350 } else { 2351 status = INVALID_OPERATION; 2352 } 2353 } else { 2354 status = BAD_VALUE; 2355 } 2356 } 2357 return status; 2358} 2359 2360void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2361{ 2362 for (size_t i = 0; i < mTracks.size(); ++i) { 2363 sp<Track> track = mTracks[i]; 2364 if (track->auxEffectId() == effectId) { 2365 attachAuxEffect_l(track, 0); 2366 } 2367 } 2368} 2369 2370bool AudioFlinger::PlaybackThread::threadLoop() 2371{ 2372 Vector< sp<Track> > tracksToRemove; 2373 2374 standbyTime = systemTime(); 2375 2376 // MIXER 2377 nsecs_t lastWarning = 0; 2378 2379 // DUPLICATING 2380 // FIXME could this be made local to while loop? 2381 writeFrames = 0; 2382 2383 int lastGeneration = 0; 2384 2385 cacheParameters_l(); 2386 sleepTime = idleSleepTime; 2387 2388 if (mType == MIXER) { 2389 sleepTimeShift = 0; 2390 } 2391 2392 CpuStats cpuStats; 2393 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2394 2395 acquireWakeLock(); 2396 2397 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2398 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2399 // and then that string will be logged at the next convenient opportunity. 2400 const char *logString = NULL; 2401 2402 checkSilentMode_l(); 2403 2404 while (!exitPending()) 2405 { 2406 cpuStats.sample(myName); 2407 2408 Vector< sp<EffectChain> > effectChains; 2409 2410 { // scope for mLock 2411 2412 Mutex::Autolock _l(mLock); 2413 2414 processConfigEvents_l(); 2415 2416 if (logString != NULL) { 2417 mNBLogWriter->logTimestamp(); 2418 mNBLogWriter->log(logString); 2419 logString = NULL; 2420 } 2421 2422 if (mLatchDValid) { 2423 mLatchQ = mLatchD; 2424 mLatchDValid = false; 2425 mLatchQValid = true; 2426 } 2427 2428 saveOutputTracks(); 2429 if (mSignalPending) { 2430 // A signal was raised while we were unlocked 2431 mSignalPending = false; 2432 } else if (waitingAsyncCallback_l()) { 2433 if (exitPending()) { 2434 break; 2435 } 2436 releaseWakeLock_l(); 2437 mWakeLockUids.clear(); 2438 mActiveTracksGeneration++; 2439 ALOGV("wait async completion"); 2440 mWaitWorkCV.wait(mLock); 2441 ALOGV("async completion/wake"); 2442 acquireWakeLock_l(); 2443 standbyTime = systemTime() + standbyDelay; 2444 sleepTime = 0; 2445 2446 continue; 2447 } 2448 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2449 isSuspended()) { 2450 // put audio hardware into standby after short delay 2451 if (shouldStandby_l()) { 2452 2453 threadLoop_standby(); 2454 2455 mStandby = true; 2456 } 2457 2458 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2459 // we're about to wait, flush the binder command buffer 2460 IPCThreadState::self()->flushCommands(); 2461 2462 clearOutputTracks(); 2463 2464 if (exitPending()) { 2465 break; 2466 } 2467 2468 releaseWakeLock_l(); 2469 mWakeLockUids.clear(); 2470 mActiveTracksGeneration++; 2471 // wait until we have something to do... 2472 ALOGV("%s going to sleep", myName.string()); 2473 mWaitWorkCV.wait(mLock); 2474 ALOGV("%s waking up", myName.string()); 2475 acquireWakeLock_l(); 2476 2477 mMixerStatus = MIXER_IDLE; 2478 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2479 mBytesWritten = 0; 2480 mBytesRemaining = 0; 2481 checkSilentMode_l(); 2482 2483 standbyTime = systemTime() + standbyDelay; 2484 sleepTime = idleSleepTime; 2485 if (mType == MIXER) { 2486 sleepTimeShift = 0; 2487 } 2488 2489 continue; 2490 } 2491 } 2492 // mMixerStatusIgnoringFastTracks is also updated internally 2493 mMixerStatus = prepareTracks_l(&tracksToRemove); 2494 2495 // compare with previously applied list 2496 if (lastGeneration != mActiveTracksGeneration) { 2497 // update wakelock 2498 updateWakeLockUids_l(mWakeLockUids); 2499 lastGeneration = mActiveTracksGeneration; 2500 } 2501 2502 // prevent any changes in effect chain list and in each effect chain 2503 // during mixing and effect process as the audio buffers could be deleted 2504 // or modified if an effect is created or deleted 2505 lockEffectChains_l(effectChains); 2506 } // mLock scope ends 2507 2508 if (mBytesRemaining == 0) { 2509 mCurrentWriteLength = 0; 2510 if (mMixerStatus == MIXER_TRACKS_READY) { 2511 // threadLoop_mix() sets mCurrentWriteLength 2512 threadLoop_mix(); 2513 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2514 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2515 // threadLoop_sleepTime sets sleepTime to 0 if data 2516 // must be written to HAL 2517 threadLoop_sleepTime(); 2518 if (sleepTime == 0) { 2519 mCurrentWriteLength = mSinkBufferSize; 2520 } 2521 } 2522 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2523 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2524 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2525 // or mSinkBuffer (if there are no effects). 2526 // 2527 // This is done pre-effects computation; if effects change to 2528 // support higher precision, this needs to move. 2529 // 2530 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2531 // TODO use sleepTime == 0 as an additional condition. 2532 if (mMixerBufferValid) { 2533 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2534 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2535 2536 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2537 mNormalFrameCount * mChannelCount); 2538 } 2539 2540 mBytesRemaining = mCurrentWriteLength; 2541 if (isSuspended()) { 2542 sleepTime = suspendSleepTimeUs(); 2543 // simulate write to HAL when suspended 2544 mBytesWritten += mSinkBufferSize; 2545 mBytesRemaining = 0; 2546 } 2547 2548 // only process effects if we're going to write 2549 if (sleepTime == 0 && mType != OFFLOAD) { 2550 for (size_t i = 0; i < effectChains.size(); i ++) { 2551 effectChains[i]->process_l(); 2552 } 2553 } 2554 } 2555 // Process effect chains for offloaded thread even if no audio 2556 // was read from audio track: process only updates effect state 2557 // and thus does have to be synchronized with audio writes but may have 2558 // to be called while waiting for async write callback 2559 if (mType == OFFLOAD) { 2560 for (size_t i = 0; i < effectChains.size(); i ++) { 2561 effectChains[i]->process_l(); 2562 } 2563 } 2564 2565 // Only if the Effects buffer is enabled and there is data in the 2566 // Effects buffer (buffer valid), we need to 2567 // copy into the sink buffer. 2568 // TODO use sleepTime == 0 as an additional condition. 2569 if (mEffectBufferValid) { 2570 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2571 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2572 mNormalFrameCount * mChannelCount); 2573 } 2574 2575 // enable changes in effect chain 2576 unlockEffectChains(effectChains); 2577 2578 if (!waitingAsyncCallback()) { 2579 // sleepTime == 0 means we must write to audio hardware 2580 if (sleepTime == 0) { 2581 if (mBytesRemaining) { 2582 ssize_t ret = threadLoop_write(); 2583 if (ret < 0) { 2584 mBytesRemaining = 0; 2585 } else { 2586 mBytesWritten += ret; 2587 mBytesRemaining -= ret; 2588 } 2589 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2590 (mMixerStatus == MIXER_DRAIN_ALL)) { 2591 threadLoop_drain(); 2592 } 2593 if (mType == MIXER) { 2594 // write blocked detection 2595 nsecs_t now = systemTime(); 2596 nsecs_t delta = now - mLastWriteTime; 2597 if (!mStandby && delta > maxPeriod) { 2598 mNumDelayedWrites++; 2599 if ((now - lastWarning) > kWarningThrottleNs) { 2600 ATRACE_NAME("underrun"); 2601 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2602 ns2ms(delta), mNumDelayedWrites, this); 2603 lastWarning = now; 2604 } 2605 } 2606 } 2607 2608 } else { 2609 usleep(sleepTime); 2610 } 2611 } 2612 2613 // Finally let go of removed track(s), without the lock held 2614 // since we can't guarantee the destructors won't acquire that 2615 // same lock. This will also mutate and push a new fast mixer state. 2616 threadLoop_removeTracks(tracksToRemove); 2617 tracksToRemove.clear(); 2618 2619 // FIXME I don't understand the need for this here; 2620 // it was in the original code but maybe the 2621 // assignment in saveOutputTracks() makes this unnecessary? 2622 clearOutputTracks(); 2623 2624 // Effect chains will be actually deleted here if they were removed from 2625 // mEffectChains list during mixing or effects processing 2626 effectChains.clear(); 2627 2628 // FIXME Note that the above .clear() is no longer necessary since effectChains 2629 // is now local to this block, but will keep it for now (at least until merge done). 2630 } 2631 2632 threadLoop_exit(); 2633 2634 if (!mStandby) { 2635 threadLoop_standby(); 2636 mStandby = true; 2637 } 2638 2639 releaseWakeLock(); 2640 mWakeLockUids.clear(); 2641 mActiveTracksGeneration++; 2642 2643 ALOGV("Thread %p type %d exiting", this, mType); 2644 return false; 2645} 2646 2647// removeTracks_l() must be called with ThreadBase::mLock held 2648void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2649{ 2650 size_t count = tracksToRemove.size(); 2651 if (count > 0) { 2652 for (size_t i=0 ; i<count ; i++) { 2653 const sp<Track>& track = tracksToRemove.itemAt(i); 2654 mActiveTracks.remove(track); 2655 mWakeLockUids.remove(track->uid()); 2656 mActiveTracksGeneration++; 2657 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2658 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2659 if (chain != 0) { 2660 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2661 track->sessionId()); 2662 chain->decActiveTrackCnt(); 2663 } 2664 if (track->isTerminated()) { 2665 removeTrack_l(track); 2666 } 2667 } 2668 } 2669 2670} 2671 2672status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2673{ 2674 if (mNormalSink != 0) { 2675 return mNormalSink->getTimestamp(timestamp); 2676 } 2677 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2678 uint64_t position64; 2679 int ret = mOutput->stream->get_presentation_position( 2680 mOutput->stream, &position64, ×tamp.mTime); 2681 if (ret == 0) { 2682 timestamp.mPosition = (uint32_t)position64; 2683 return NO_ERROR; 2684 } 2685 } 2686 return INVALID_OPERATION; 2687} 2688 2689status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2690 audio_patch_handle_t *handle) 2691{ 2692 status_t status = NO_ERROR; 2693 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2694 // store new device and send to effects 2695 audio_devices_t type = AUDIO_DEVICE_NONE; 2696 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2697 type |= patch->sinks[i].ext.device.type; 2698 } 2699 mOutDevice = type; 2700 for (size_t i = 0; i < mEffectChains.size(); i++) { 2701 mEffectChains[i]->setDevice_l(mOutDevice); 2702 } 2703 2704 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2705 status = hwDevice->create_audio_patch(hwDevice, 2706 patch->num_sources, 2707 patch->sources, 2708 patch->num_sinks, 2709 patch->sinks, 2710 handle); 2711 } else { 2712 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2713 } 2714 return status; 2715} 2716 2717status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2718{ 2719 status_t status = NO_ERROR; 2720 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2721 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2722 status = hwDevice->release_audio_patch(hwDevice, handle); 2723 } else { 2724 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2725 } 2726 return status; 2727} 2728 2729void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2730{ 2731 Mutex::Autolock _l(mLock); 2732 mTracks.add(track); 2733} 2734 2735void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2736{ 2737 Mutex::Autolock _l(mLock); 2738 destroyTrack_l(track); 2739} 2740 2741void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2742{ 2743 ThreadBase::getAudioPortConfig(config); 2744 config->role = AUDIO_PORT_ROLE_SOURCE; 2745 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2746 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2747} 2748 2749// ---------------------------------------------------------------------------- 2750 2751AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2752 audio_io_handle_t id, audio_devices_t device, type_t type) 2753 : PlaybackThread(audioFlinger, output, id, device, type), 2754 // mAudioMixer below 2755 // mFastMixer below 2756 mFastMixerFutex(0) 2757 // mOutputSink below 2758 // mPipeSink below 2759 // mNormalSink below 2760{ 2761 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2762 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2763 "mFrameCount=%d, mNormalFrameCount=%d", 2764 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2765 mNormalFrameCount); 2766 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2767 2768 // create an NBAIO sink for the HAL output stream, and negotiate 2769 mOutputSink = new AudioStreamOutSink(output->stream); 2770 size_t numCounterOffers = 0; 2771 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2772 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2773 ALOG_ASSERT(index == 0); 2774 2775 // initialize fast mixer depending on configuration 2776 bool initFastMixer; 2777 switch (kUseFastMixer) { 2778 case FastMixer_Never: 2779 initFastMixer = false; 2780 break; 2781 case FastMixer_Always: 2782 initFastMixer = true; 2783 break; 2784 case FastMixer_Static: 2785 case FastMixer_Dynamic: 2786 initFastMixer = mFrameCount < mNormalFrameCount; 2787 break; 2788 } 2789 if (initFastMixer) { 2790 audio_format_t fastMixerFormat; 2791 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2792 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2793 } else { 2794 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2795 } 2796 if (mFormat != fastMixerFormat) { 2797 // change our Sink format to accept our intermediate precision 2798 mFormat = fastMixerFormat; 2799 free(mSinkBuffer); 2800 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2801 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2802 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2803 } 2804 2805 // create a MonoPipe to connect our submix to FastMixer 2806 NBAIO_Format format = mOutputSink->format(); 2807 NBAIO_Format origformat = format; 2808 // adjust format to match that of the Fast Mixer 2809 format.mFormat = fastMixerFormat; 2810 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2811 2812 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2813 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2814 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2815 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2816 const NBAIO_Format offers[1] = {format}; 2817 size_t numCounterOffers = 0; 2818 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2819 ALOG_ASSERT(index == 0); 2820 monoPipe->setAvgFrames((mScreenState & 1) ? 2821 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2822 mPipeSink = monoPipe; 2823 2824#ifdef TEE_SINK 2825 if (mTeeSinkOutputEnabled) { 2826 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2827 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 2828 const NBAIO_Format offers2[1] = {origformat}; 2829 numCounterOffers = 0; 2830 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 2831 ALOG_ASSERT(index == 0); 2832 mTeeSink = teeSink; 2833 PipeReader *teeSource = new PipeReader(*teeSink); 2834 numCounterOffers = 0; 2835 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 2836 ALOG_ASSERT(index == 0); 2837 mTeeSource = teeSource; 2838 } 2839#endif 2840 2841 // create fast mixer and configure it initially with just one fast track for our submix 2842 mFastMixer = new FastMixer(); 2843 FastMixerStateQueue *sq = mFastMixer->sq(); 2844#ifdef STATE_QUEUE_DUMP 2845 sq->setObserverDump(&mStateQueueObserverDump); 2846 sq->setMutatorDump(&mStateQueueMutatorDump); 2847#endif 2848 FastMixerState *state = sq->begin(); 2849 FastTrack *fastTrack = &state->mFastTracks[0]; 2850 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2851 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2852 fastTrack->mVolumeProvider = NULL; 2853 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2854 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2855 fastTrack->mGeneration++; 2856 state->mFastTracksGen++; 2857 state->mTrackMask = 1; 2858 // fast mixer will use the HAL output sink 2859 state->mOutputSink = mOutputSink.get(); 2860 state->mOutputSinkGen++; 2861 state->mFrameCount = mFrameCount; 2862 state->mCommand = FastMixerState::COLD_IDLE; 2863 // already done in constructor initialization list 2864 //mFastMixerFutex = 0; 2865 state->mColdFutexAddr = &mFastMixerFutex; 2866 state->mColdGen++; 2867 state->mDumpState = &mFastMixerDumpState; 2868#ifdef TEE_SINK 2869 state->mTeeSink = mTeeSink.get(); 2870#endif 2871 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2872 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2873 sq->end(); 2874 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2875 2876 // start the fast mixer 2877 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2878 pid_t tid = mFastMixer->getTid(); 2879 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2880 if (err != 0) { 2881 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2882 kPriorityFastMixer, getpid_cached, tid, err); 2883 } 2884 2885#ifdef AUDIO_WATCHDOG 2886 // create and start the watchdog 2887 mAudioWatchdog = new AudioWatchdog(); 2888 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2889 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2890 tid = mAudioWatchdog->getTid(); 2891 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2892 if (err != 0) { 2893 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2894 kPriorityFastMixer, getpid_cached, tid, err); 2895 } 2896#endif 2897 2898 } 2899 2900 switch (kUseFastMixer) { 2901 case FastMixer_Never: 2902 case FastMixer_Dynamic: 2903 mNormalSink = mOutputSink; 2904 break; 2905 case FastMixer_Always: 2906 mNormalSink = mPipeSink; 2907 break; 2908 case FastMixer_Static: 2909 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2910 break; 2911 } 2912} 2913 2914AudioFlinger::MixerThread::~MixerThread() 2915{ 2916 if (mFastMixer != 0) { 2917 FastMixerStateQueue *sq = mFastMixer->sq(); 2918 FastMixerState *state = sq->begin(); 2919 if (state->mCommand == FastMixerState::COLD_IDLE) { 2920 int32_t old = android_atomic_inc(&mFastMixerFutex); 2921 if (old == -1) { 2922 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2923 } 2924 } 2925 state->mCommand = FastMixerState::EXIT; 2926 sq->end(); 2927 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2928 mFastMixer->join(); 2929 // Though the fast mixer thread has exited, it's state queue is still valid. 2930 // We'll use that extract the final state which contains one remaining fast track 2931 // corresponding to our sub-mix. 2932 state = sq->begin(); 2933 ALOG_ASSERT(state->mTrackMask == 1); 2934 FastTrack *fastTrack = &state->mFastTracks[0]; 2935 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2936 delete fastTrack->mBufferProvider; 2937 sq->end(false /*didModify*/); 2938 mFastMixer.clear(); 2939#ifdef AUDIO_WATCHDOG 2940 if (mAudioWatchdog != 0) { 2941 mAudioWatchdog->requestExit(); 2942 mAudioWatchdog->requestExitAndWait(); 2943 mAudioWatchdog.clear(); 2944 } 2945#endif 2946 } 2947 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2948 delete mAudioMixer; 2949} 2950 2951 2952uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2953{ 2954 if (mFastMixer != 0) { 2955 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2956 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2957 } 2958 return latency; 2959} 2960 2961 2962void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2963{ 2964 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2965} 2966 2967ssize_t AudioFlinger::MixerThread::threadLoop_write() 2968{ 2969 // FIXME we should only do one push per cycle; confirm this is true 2970 // Start the fast mixer if it's not already running 2971 if (mFastMixer != 0) { 2972 FastMixerStateQueue *sq = mFastMixer->sq(); 2973 FastMixerState *state = sq->begin(); 2974 if (state->mCommand != FastMixerState::MIX_WRITE && 2975 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2976 if (state->mCommand == FastMixerState::COLD_IDLE) { 2977 int32_t old = android_atomic_inc(&mFastMixerFutex); 2978 if (old == -1) { 2979 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2980 } 2981#ifdef AUDIO_WATCHDOG 2982 if (mAudioWatchdog != 0) { 2983 mAudioWatchdog->resume(); 2984 } 2985#endif 2986 } 2987 state->mCommand = FastMixerState::MIX_WRITE; 2988 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2989 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2990 sq->end(); 2991 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2992 if (kUseFastMixer == FastMixer_Dynamic) { 2993 mNormalSink = mPipeSink; 2994 } 2995 } else { 2996 sq->end(false /*didModify*/); 2997 } 2998 } 2999 return PlaybackThread::threadLoop_write(); 3000} 3001 3002void AudioFlinger::MixerThread::threadLoop_standby() 3003{ 3004 // Idle the fast mixer if it's currently running 3005 if (mFastMixer != 0) { 3006 FastMixerStateQueue *sq = mFastMixer->sq(); 3007 FastMixerState *state = sq->begin(); 3008 if (!(state->mCommand & FastMixerState::IDLE)) { 3009 state->mCommand = FastMixerState::COLD_IDLE; 3010 state->mColdFutexAddr = &mFastMixerFutex; 3011 state->mColdGen++; 3012 mFastMixerFutex = 0; 3013 sq->end(); 3014 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3015 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3016 if (kUseFastMixer == FastMixer_Dynamic) { 3017 mNormalSink = mOutputSink; 3018 } 3019#ifdef AUDIO_WATCHDOG 3020 if (mAudioWatchdog != 0) { 3021 mAudioWatchdog->pause(); 3022 } 3023#endif 3024 } else { 3025 sq->end(false /*didModify*/); 3026 } 3027 } 3028 PlaybackThread::threadLoop_standby(); 3029} 3030 3031bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3032{ 3033 return false; 3034} 3035 3036bool AudioFlinger::PlaybackThread::shouldStandby_l() 3037{ 3038 return !mStandby; 3039} 3040 3041bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3042{ 3043 Mutex::Autolock _l(mLock); 3044 return waitingAsyncCallback_l(); 3045} 3046 3047// shared by MIXER and DIRECT, overridden by DUPLICATING 3048void AudioFlinger::PlaybackThread::threadLoop_standby() 3049{ 3050 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3051 mOutput->stream->common.standby(&mOutput->stream->common); 3052 if (mUseAsyncWrite != 0) { 3053 // discard any pending drain or write ack by incrementing sequence 3054 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3055 mDrainSequence = (mDrainSequence + 2) & ~1; 3056 ALOG_ASSERT(mCallbackThread != 0); 3057 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3058 mCallbackThread->setDraining(mDrainSequence); 3059 } 3060} 3061 3062void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3063{ 3064 ALOGV("signal playback thread"); 3065 broadcast_l(); 3066} 3067 3068void AudioFlinger::MixerThread::threadLoop_mix() 3069{ 3070 // obtain the presentation timestamp of the next output buffer 3071 int64_t pts; 3072 status_t status = INVALID_OPERATION; 3073 3074 if (mNormalSink != 0) { 3075 status = mNormalSink->getNextWriteTimestamp(&pts); 3076 } else { 3077 status = mOutputSink->getNextWriteTimestamp(&pts); 3078 } 3079 3080 if (status != NO_ERROR) { 3081 pts = AudioBufferProvider::kInvalidPTS; 3082 } 3083 3084 // mix buffers... 3085 mAudioMixer->process(pts); 3086 mCurrentWriteLength = mSinkBufferSize; 3087 // increase sleep time progressively when application underrun condition clears. 3088 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3089 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3090 // such that we would underrun the audio HAL. 3091 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3092 sleepTimeShift--; 3093 } 3094 sleepTime = 0; 3095 standbyTime = systemTime() + standbyDelay; 3096 //TODO: delay standby when effects have a tail 3097 3098 mLatchD.mFramesReleased.clear(); 3099 { 3100 Mutex::Autolock _l(mLock); 3101 size_t size = mActiveTracks.size(); 3102 for (size_t i = 0; i < size; i++) { 3103 sp<Track> t = mActiveTracks[i].promote(); 3104 if (t != 0) { 3105 mLatchD.mFramesReleased.add(t.get(), t->mAudioTrackServerProxy->framesReleased()); 3106 } 3107 } 3108 } 3109} 3110 3111void AudioFlinger::MixerThread::threadLoop_sleepTime() 3112{ 3113 // If no tracks are ready, sleep once for the duration of an output 3114 // buffer size, then write 0s to the output 3115 if (sleepTime == 0) { 3116 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3117 sleepTime = activeSleepTime >> sleepTimeShift; 3118 if (sleepTime < kMinThreadSleepTimeUs) { 3119 sleepTime = kMinThreadSleepTimeUs; 3120 } 3121 // reduce sleep time in case of consecutive application underruns to avoid 3122 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3123 // duration we would end up writing less data than needed by the audio HAL if 3124 // the condition persists. 3125 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3126 sleepTimeShift++; 3127 } 3128 } else { 3129 sleepTime = idleSleepTime; 3130 } 3131 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3132 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3133 // before effects processing or output. 3134 if (mMixerBufferValid) { 3135 memset(mMixerBuffer, 0, mMixerBufferSize); 3136 } else { 3137 memset(mSinkBuffer, 0, mSinkBufferSize); 3138 } 3139 sleepTime = 0; 3140 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3141 "anticipated start"); 3142 } 3143 // TODO add standby time extension fct of effect tail 3144} 3145 3146// prepareTracks_l() must be called with ThreadBase::mLock held 3147AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3148 Vector< sp<Track> > *tracksToRemove) 3149{ 3150 3151 mixer_state mixerStatus = MIXER_IDLE; 3152 // find out which tracks need to be processed 3153 size_t count = mActiveTracks.size(); 3154 size_t mixedTracks = 0; 3155 size_t tracksWithEffect = 0; 3156 // counts only _active_ fast tracks 3157 size_t fastTracks = 0; 3158 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3159 3160 float masterVolume = mMasterVolume; 3161 bool masterMute = mMasterMute; 3162 3163 if (masterMute) { 3164 masterVolume = 0; 3165 } 3166 // Delegate master volume control to effect in output mix effect chain if needed 3167 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3168 if (chain != 0) { 3169 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3170 chain->setVolume_l(&v, &v); 3171 masterVolume = (float)((v + (1 << 23)) >> 24); 3172 chain.clear(); 3173 } 3174 3175 // prepare a new state to push 3176 FastMixerStateQueue *sq = NULL; 3177 FastMixerState *state = NULL; 3178 bool didModify = false; 3179 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3180 if (mFastMixer != 0) { 3181 sq = mFastMixer->sq(); 3182 state = sq->begin(); 3183 } 3184 3185 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3186 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3187 3188 for (size_t i=0 ; i<count ; i++) { 3189 const sp<Track> t = mActiveTracks[i].promote(); 3190 if (t == 0) { 3191 continue; 3192 } 3193 3194 // this const just means the local variable doesn't change 3195 Track* const track = t.get(); 3196 3197 // process fast tracks 3198 if (track->isFastTrack()) { 3199 3200 // It's theoretically possible (though unlikely) for a fast track to be created 3201 // and then removed within the same normal mix cycle. This is not a problem, as 3202 // the track never becomes active so it's fast mixer slot is never touched. 3203 // The converse, of removing an (active) track and then creating a new track 3204 // at the identical fast mixer slot within the same normal mix cycle, 3205 // is impossible because the slot isn't marked available until the end of each cycle. 3206 int j = track->mFastIndex; 3207 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3208 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3209 FastTrack *fastTrack = &state->mFastTracks[j]; 3210 3211 // Determine whether the track is currently in underrun condition, 3212 // and whether it had a recent underrun. 3213 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3214 FastTrackUnderruns underruns = ftDump->mUnderruns; 3215 uint32_t recentFull = (underruns.mBitFields.mFull - 3216 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3217 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3218 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3219 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3220 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3221 uint32_t recentUnderruns = recentPartial + recentEmpty; 3222 track->mObservedUnderruns = underruns; 3223 // don't count underruns that occur while stopping or pausing 3224 // or stopped which can occur when flush() is called while active 3225 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3226 recentUnderruns > 0) { 3227 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3228 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3229 } 3230 3231 // This is similar to the state machine for normal tracks, 3232 // with a few modifications for fast tracks. 3233 bool isActive = true; 3234 switch (track->mState) { 3235 case TrackBase::STOPPING_1: 3236 // track stays active in STOPPING_1 state until first underrun 3237 if (recentUnderruns > 0 || track->isTerminated()) { 3238 track->mState = TrackBase::STOPPING_2; 3239 } 3240 break; 3241 case TrackBase::PAUSING: 3242 // ramp down is not yet implemented 3243 track->setPaused(); 3244 break; 3245 case TrackBase::RESUMING: 3246 // ramp up is not yet implemented 3247 track->mState = TrackBase::ACTIVE; 3248 break; 3249 case TrackBase::ACTIVE: 3250 if (recentFull > 0 || recentPartial > 0) { 3251 // track has provided at least some frames recently: reset retry count 3252 track->mRetryCount = kMaxTrackRetries; 3253 } 3254 if (recentUnderruns == 0) { 3255 // no recent underruns: stay active 3256 break; 3257 } 3258 // there has recently been an underrun of some kind 3259 if (track->sharedBuffer() == 0) { 3260 // were any of the recent underruns "empty" (no frames available)? 3261 if (recentEmpty == 0) { 3262 // no, then ignore the partial underruns as they are allowed indefinitely 3263 break; 3264 } 3265 // there has recently been an "empty" underrun: decrement the retry counter 3266 if (--(track->mRetryCount) > 0) { 3267 break; 3268 } 3269 // indicate to client process that the track was disabled because of underrun; 3270 // it will then automatically call start() when data is available 3271 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3272 // remove from active list, but state remains ACTIVE [confusing but true] 3273 isActive = false; 3274 break; 3275 } 3276 // fall through 3277 case TrackBase::STOPPING_2: 3278 case TrackBase::PAUSED: 3279 case TrackBase::STOPPED: 3280 case TrackBase::FLUSHED: // flush() while active 3281 // Check for presentation complete if track is inactive 3282 // We have consumed all the buffers of this track. 3283 // This would be incomplete if we auto-paused on underrun 3284 { 3285 size_t audioHALFrames = 3286 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3287 size_t framesWritten = mBytesWritten / mFrameSize; 3288 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3289 // track stays in active list until presentation is complete 3290 break; 3291 } 3292 } 3293 if (track->isStopping_2()) { 3294 track->mState = TrackBase::STOPPED; 3295 } 3296 if (track->isStopped()) { 3297 // Can't reset directly, as fast mixer is still polling this track 3298 // track->reset(); 3299 // So instead mark this track as needing to be reset after push with ack 3300 resetMask |= 1 << i; 3301 } 3302 isActive = false; 3303 break; 3304 case TrackBase::IDLE: 3305 default: 3306 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3307 } 3308 3309 if (isActive) { 3310 // was it previously inactive? 3311 if (!(state->mTrackMask & (1 << j))) { 3312 ExtendedAudioBufferProvider *eabp = track; 3313 VolumeProvider *vp = track; 3314 fastTrack->mBufferProvider = eabp; 3315 fastTrack->mVolumeProvider = vp; 3316 fastTrack->mChannelMask = track->mChannelMask; 3317 fastTrack->mFormat = track->mFormat; 3318 fastTrack->mGeneration++; 3319 state->mTrackMask |= 1 << j; 3320 didModify = true; 3321 // no acknowledgement required for newly active tracks 3322 } 3323 // cache the combined master volume and stream type volume for fast mixer; this 3324 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3325 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3326 ++fastTracks; 3327 } else { 3328 // was it previously active? 3329 if (state->mTrackMask & (1 << j)) { 3330 fastTrack->mBufferProvider = NULL; 3331 fastTrack->mGeneration++; 3332 state->mTrackMask &= ~(1 << j); 3333 didModify = true; 3334 // If any fast tracks were removed, we must wait for acknowledgement 3335 // because we're about to decrement the last sp<> on those tracks. 3336 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3337 } else { 3338 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3339 } 3340 tracksToRemove->add(track); 3341 // Avoids a misleading display in dumpsys 3342 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3343 } 3344 continue; 3345 } 3346 3347 { // local variable scope to avoid goto warning 3348 3349 audio_track_cblk_t* cblk = track->cblk(); 3350 3351 // The first time a track is added we wait 3352 // for all its buffers to be filled before processing it 3353 int name = track->name(); 3354 // make sure that we have enough frames to mix one full buffer. 3355 // enforce this condition only once to enable draining the buffer in case the client 3356 // app does not call stop() and relies on underrun to stop: 3357 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3358 // during last round 3359 size_t desiredFrames; 3360 uint32_t sr = track->sampleRate(); 3361 if (sr == mSampleRate) { 3362 desiredFrames = mNormalFrameCount; 3363 } else { 3364 // +1 for rounding and +1 for additional sample needed for interpolation 3365 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3366 // add frames already consumed but not yet released by the resampler 3367 // because mAudioTrackServerProxy->framesReady() will include these frames 3368 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3369#if 0 3370 // the minimum track buffer size is normally twice the number of frames necessary 3371 // to fill one buffer and the resampler should not leave more than one buffer worth 3372 // of unreleased frames after each pass, but just in case... 3373 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3374#endif 3375 } 3376 uint32_t minFrames = 1; 3377 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3378 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3379 minFrames = desiredFrames; 3380 } 3381 3382 size_t framesReady = track->framesReady(); 3383 if ((framesReady >= minFrames) && track->isReady() && 3384 !track->isPaused() && !track->isTerminated()) 3385 { 3386 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3387 3388 mixedTracks++; 3389 3390 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3391 // there is an effect chain connected to the track 3392 chain.clear(); 3393 if (track->mainBuffer() != mSinkBuffer && 3394 track->mainBuffer() != mMixerBuffer) { 3395 if (mEffectBufferEnabled) { 3396 mEffectBufferValid = true; // Later can set directly. 3397 } 3398 chain = getEffectChain_l(track->sessionId()); 3399 // Delegate volume control to effect in track effect chain if needed 3400 if (chain != 0) { 3401 tracksWithEffect++; 3402 } else { 3403 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3404 "session %d", 3405 name, track->sessionId()); 3406 } 3407 } 3408 3409 3410 int param = AudioMixer::VOLUME; 3411 if (track->mFillingUpStatus == Track::FS_FILLED) { 3412 // no ramp for the first volume setting 3413 track->mFillingUpStatus = Track::FS_ACTIVE; 3414 if (track->mState == TrackBase::RESUMING) { 3415 track->mState = TrackBase::ACTIVE; 3416 param = AudioMixer::RAMP_VOLUME; 3417 } 3418 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3419 // FIXME should not make a decision based on mServer 3420 } else if (cblk->mServer != 0) { 3421 // If the track is stopped before the first frame was mixed, 3422 // do not apply ramp 3423 param = AudioMixer::RAMP_VOLUME; 3424 } 3425 3426 // compute volume for this track 3427 uint32_t vl, vr; // in U8.24 integer format 3428 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3429 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3430 vl = vr = 0; 3431 vlf = vrf = vaf = 0.; 3432 if (track->isPausing()) { 3433 track->setPaused(); 3434 } 3435 } else { 3436 3437 // read original volumes with volume control 3438 float typeVolume = mStreamTypes[track->streamType()].volume; 3439 float v = masterVolume * typeVolume; 3440 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3441 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3442 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3443 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3444 // track volumes come from shared memory, so can't be trusted and must be clamped 3445 if (vlf > GAIN_FLOAT_UNITY) { 3446 ALOGV("Track left volume out of range: %.3g", vlf); 3447 vlf = GAIN_FLOAT_UNITY; 3448 } 3449 if (vrf > GAIN_FLOAT_UNITY) { 3450 ALOGV("Track right volume out of range: %.3g", vrf); 3451 vrf = GAIN_FLOAT_UNITY; 3452 } 3453 // now apply the master volume and stream type volume 3454 vlf *= v; 3455 vrf *= v; 3456 // assuming master volume and stream type volume each go up to 1.0, 3457 // then derive vl and vr as U8.24 versions for the effect chain 3458 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3459 vl = (uint32_t) (scaleto8_24 * vlf); 3460 vr = (uint32_t) (scaleto8_24 * vrf); 3461 // vl and vr are now in U8.24 format 3462 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3463 // send level comes from shared memory and so may be corrupt 3464 if (sendLevel > MAX_GAIN_INT) { 3465 ALOGV("Track send level out of range: %04X", sendLevel); 3466 sendLevel = MAX_GAIN_INT; 3467 } 3468 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3469 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3470 } 3471 3472 // Delegate volume control to effect in track effect chain if needed 3473 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3474 // Do not ramp volume if volume is controlled by effect 3475 param = AudioMixer::VOLUME; 3476 // Update remaining floating point volume levels 3477 vlf = (float)vl / (1 << 24); 3478 vrf = (float)vr / (1 << 24); 3479 track->mHasVolumeController = true; 3480 } else { 3481 // force no volume ramp when volume controller was just disabled or removed 3482 // from effect chain to avoid volume spike 3483 if (track->mHasVolumeController) { 3484 param = AudioMixer::VOLUME; 3485 } 3486 track->mHasVolumeController = false; 3487 } 3488 3489 // XXX: these things DON'T need to be done each time 3490 mAudioMixer->setBufferProvider(name, track); 3491 mAudioMixer->enable(name); 3492 3493 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3494 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3495 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3496 mAudioMixer->setParameter( 3497 name, 3498 AudioMixer::TRACK, 3499 AudioMixer::FORMAT, (void *)track->format()); 3500 mAudioMixer->setParameter( 3501 name, 3502 AudioMixer::TRACK, 3503 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3504 mAudioMixer->setParameter( 3505 name, 3506 AudioMixer::TRACK, 3507 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3508 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3509 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3510 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3511 if (reqSampleRate == 0) { 3512 reqSampleRate = mSampleRate; 3513 } else if (reqSampleRate > maxSampleRate) { 3514 reqSampleRate = maxSampleRate; 3515 } 3516 mAudioMixer->setParameter( 3517 name, 3518 AudioMixer::RESAMPLE, 3519 AudioMixer::SAMPLE_RATE, 3520 (void *)(uintptr_t)reqSampleRate); 3521 /* 3522 * Select the appropriate output buffer for the track. 3523 * 3524 * Tracks with effects go into their own effects chain buffer 3525 * and from there into either mEffectBuffer or mSinkBuffer. 3526 * 3527 * Other tracks can use mMixerBuffer for higher precision 3528 * channel accumulation. If this buffer is enabled 3529 * (mMixerBufferEnabled true), then selected tracks will accumulate 3530 * into it. 3531 * 3532 */ 3533 if (mMixerBufferEnabled 3534 && (track->mainBuffer() == mSinkBuffer 3535 || track->mainBuffer() == mMixerBuffer)) { 3536 mAudioMixer->setParameter( 3537 name, 3538 AudioMixer::TRACK, 3539 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3540 mAudioMixer->setParameter( 3541 name, 3542 AudioMixer::TRACK, 3543 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3544 // TODO: override track->mainBuffer()? 3545 mMixerBufferValid = true; 3546 } else { 3547 mAudioMixer->setParameter( 3548 name, 3549 AudioMixer::TRACK, 3550 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3551 mAudioMixer->setParameter( 3552 name, 3553 AudioMixer::TRACK, 3554 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3555 } 3556 mAudioMixer->setParameter( 3557 name, 3558 AudioMixer::TRACK, 3559 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3560 3561 // reset retry count 3562 track->mRetryCount = kMaxTrackRetries; 3563 3564 // If one track is ready, set the mixer ready if: 3565 // - the mixer was not ready during previous round OR 3566 // - no other track is not ready 3567 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3568 mixerStatus != MIXER_TRACKS_ENABLED) { 3569 mixerStatus = MIXER_TRACKS_READY; 3570 } 3571 } else { 3572 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3573 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3574 } 3575 // clear effect chain input buffer if an active track underruns to avoid sending 3576 // previous audio buffer again to effects 3577 chain = getEffectChain_l(track->sessionId()); 3578 if (chain != 0) { 3579 chain->clearInputBuffer(); 3580 } 3581 3582 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3583 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3584 track->isStopped() || track->isPaused()) { 3585 // We have consumed all the buffers of this track. 3586 // Remove it from the list of active tracks. 3587 // TODO: use actual buffer filling status instead of latency when available from 3588 // audio HAL 3589 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3590 size_t framesWritten = mBytesWritten / mFrameSize; 3591 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3592 if (track->isStopped()) { 3593 track->reset(); 3594 } 3595 tracksToRemove->add(track); 3596 } 3597 } else { 3598 // No buffers for this track. Give it a few chances to 3599 // fill a buffer, then remove it from active list. 3600 if (--(track->mRetryCount) <= 0) { 3601 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3602 tracksToRemove->add(track); 3603 // indicate to client process that the track was disabled because of underrun; 3604 // it will then automatically call start() when data is available 3605 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3606 // If one track is not ready, mark the mixer also not ready if: 3607 // - the mixer was ready during previous round OR 3608 // - no other track is ready 3609 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3610 mixerStatus != MIXER_TRACKS_READY) { 3611 mixerStatus = MIXER_TRACKS_ENABLED; 3612 } 3613 } 3614 mAudioMixer->disable(name); 3615 } 3616 3617 } // local variable scope to avoid goto warning 3618track_is_ready: ; 3619 3620 } 3621 3622 // Push the new FastMixer state if necessary 3623 bool pauseAudioWatchdog = false; 3624 if (didModify) { 3625 state->mFastTracksGen++; 3626 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3627 if (kUseFastMixer == FastMixer_Dynamic && 3628 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3629 state->mCommand = FastMixerState::COLD_IDLE; 3630 state->mColdFutexAddr = &mFastMixerFutex; 3631 state->mColdGen++; 3632 mFastMixerFutex = 0; 3633 if (kUseFastMixer == FastMixer_Dynamic) { 3634 mNormalSink = mOutputSink; 3635 } 3636 // If we go into cold idle, need to wait for acknowledgement 3637 // so that fast mixer stops doing I/O. 3638 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3639 pauseAudioWatchdog = true; 3640 } 3641 } 3642 if (sq != NULL) { 3643 sq->end(didModify); 3644 sq->push(block); 3645 } 3646#ifdef AUDIO_WATCHDOG 3647 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3648 mAudioWatchdog->pause(); 3649 } 3650#endif 3651 3652 // Now perform the deferred reset on fast tracks that have stopped 3653 while (resetMask != 0) { 3654 size_t i = __builtin_ctz(resetMask); 3655 ALOG_ASSERT(i < count); 3656 resetMask &= ~(1 << i); 3657 sp<Track> t = mActiveTracks[i].promote(); 3658 if (t == 0) { 3659 continue; 3660 } 3661 Track* track = t.get(); 3662 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3663 track->reset(); 3664 } 3665 3666 // remove all the tracks that need to be... 3667 removeTracks_l(*tracksToRemove); 3668 3669 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3670 mEffectBufferValid = true; 3671 } 3672 3673 // sink or mix buffer must be cleared if all tracks are connected to an 3674 // effect chain as in this case the mixer will not write to the sink or mix buffer 3675 // and track effects will accumulate into it 3676 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3677 (mixedTracks == 0 && fastTracks > 0))) { 3678 // FIXME as a performance optimization, should remember previous zero status 3679 if (mMixerBufferValid) { 3680 memset(mMixerBuffer, 0, mMixerBufferSize); 3681 // TODO: In testing, mSinkBuffer below need not be cleared because 3682 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3683 // after mixing. 3684 // 3685 // To enforce this guarantee: 3686 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3687 // (mixedTracks == 0 && fastTracks > 0)) 3688 // must imply MIXER_TRACKS_READY. 3689 // Later, we may clear buffers regardless, and skip much of this logic. 3690 } 3691 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3692 if (mEffectBufferValid) { 3693 memset(mEffectBuffer, 0, mEffectBufferSize); 3694 } 3695 // FIXME as a performance optimization, should remember previous zero status 3696 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3697 } 3698 3699 // if any fast tracks, then status is ready 3700 mMixerStatusIgnoringFastTracks = mixerStatus; 3701 if (fastTracks > 0) { 3702 mixerStatus = MIXER_TRACKS_READY; 3703 } 3704 return mixerStatus; 3705} 3706 3707// getTrackName_l() must be called with ThreadBase::mLock held 3708int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3709 audio_format_t format, int sessionId) 3710{ 3711 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3712} 3713 3714// deleteTrackName_l() must be called with ThreadBase::mLock held 3715void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3716{ 3717 ALOGV("remove track (%d) and delete from mixer", name); 3718 mAudioMixer->deleteTrackName(name); 3719} 3720 3721// checkForNewParameter_l() must be called with ThreadBase::mLock held 3722bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3723 status_t& status) 3724{ 3725 bool reconfig = false; 3726 3727 status = NO_ERROR; 3728 3729 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3730 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3731 if (mFastMixer != 0) { 3732 FastMixerStateQueue *sq = mFastMixer->sq(); 3733 FastMixerState *state = sq->begin(); 3734 if (!(state->mCommand & FastMixerState::IDLE)) { 3735 previousCommand = state->mCommand; 3736 state->mCommand = FastMixerState::HOT_IDLE; 3737 sq->end(); 3738 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3739 } else { 3740 sq->end(false /*didModify*/); 3741 } 3742 } 3743 3744 AudioParameter param = AudioParameter(keyValuePair); 3745 int value; 3746 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3747 reconfig = true; 3748 } 3749 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3750 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3751 status = BAD_VALUE; 3752 } else { 3753 // no need to save value, since it's constant 3754 reconfig = true; 3755 } 3756 } 3757 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3758 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3759 status = BAD_VALUE; 3760 } else { 3761 // no need to save value, since it's constant 3762 reconfig = true; 3763 } 3764 } 3765 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3766 // do not accept frame count changes if tracks are open as the track buffer 3767 // size depends on frame count and correct behavior would not be guaranteed 3768 // if frame count is changed after track creation 3769 if (!mTracks.isEmpty()) { 3770 status = INVALID_OPERATION; 3771 } else { 3772 reconfig = true; 3773 } 3774 } 3775 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3776#ifdef ADD_BATTERY_DATA 3777 // when changing the audio output device, call addBatteryData to notify 3778 // the change 3779 if (mOutDevice != value) { 3780 uint32_t params = 0; 3781 // check whether speaker is on 3782 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3783 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3784 } 3785 3786 audio_devices_t deviceWithoutSpeaker 3787 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3788 // check if any other device (except speaker) is on 3789 if (value & deviceWithoutSpeaker ) { 3790 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3791 } 3792 3793 if (params != 0) { 3794 addBatteryData(params); 3795 } 3796 } 3797#endif 3798 3799 // forward device change to effects that have requested to be 3800 // aware of attached audio device. 3801 if (value != AUDIO_DEVICE_NONE) { 3802 mOutDevice = value; 3803 for (size_t i = 0; i < mEffectChains.size(); i++) { 3804 mEffectChains[i]->setDevice_l(mOutDevice); 3805 } 3806 } 3807 } 3808 3809 if (status == NO_ERROR) { 3810 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3811 keyValuePair.string()); 3812 if (!mStandby && status == INVALID_OPERATION) { 3813 mOutput->stream->common.standby(&mOutput->stream->common); 3814 mStandby = true; 3815 mBytesWritten = 0; 3816 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3817 keyValuePair.string()); 3818 } 3819 if (status == NO_ERROR && reconfig) { 3820 readOutputParameters_l(); 3821 delete mAudioMixer; 3822 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3823 for (size_t i = 0; i < mTracks.size() ; i++) { 3824 int name = getTrackName_l(mTracks[i]->mChannelMask, 3825 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3826 if (name < 0) { 3827 break; 3828 } 3829 mTracks[i]->mName = name; 3830 } 3831 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3832 } 3833 } 3834 3835 if (!(previousCommand & FastMixerState::IDLE)) { 3836 ALOG_ASSERT(mFastMixer != 0); 3837 FastMixerStateQueue *sq = mFastMixer->sq(); 3838 FastMixerState *state = sq->begin(); 3839 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3840 state->mCommand = previousCommand; 3841 sq->end(); 3842 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3843 } 3844 3845 return reconfig; 3846} 3847 3848 3849void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3850{ 3851 const size_t SIZE = 256; 3852 char buffer[SIZE]; 3853 String8 result; 3854 3855 PlaybackThread::dumpInternals(fd, args); 3856 3857 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3858 3859 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3860 const FastMixerDumpState copy(mFastMixerDumpState); 3861 copy.dump(fd); 3862 3863#ifdef STATE_QUEUE_DUMP 3864 // Similar for state queue 3865 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3866 observerCopy.dump(fd); 3867 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3868 mutatorCopy.dump(fd); 3869#endif 3870 3871#ifdef TEE_SINK 3872 // Write the tee output to a .wav file 3873 dumpTee(fd, mTeeSource, mId); 3874#endif 3875 3876#ifdef AUDIO_WATCHDOG 3877 if (mAudioWatchdog != 0) { 3878 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3879 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3880 wdCopy.dump(fd); 3881 } 3882#endif 3883} 3884 3885uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3886{ 3887 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3888} 3889 3890uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3891{ 3892 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3893} 3894 3895void AudioFlinger::MixerThread::cacheParameters_l() 3896{ 3897 PlaybackThread::cacheParameters_l(); 3898 3899 // FIXME: Relaxed timing because of a certain device that can't meet latency 3900 // Should be reduced to 2x after the vendor fixes the driver issue 3901 // increase threshold again due to low power audio mode. The way this warning 3902 // threshold is calculated and its usefulness should be reconsidered anyway. 3903 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3904} 3905 3906// ---------------------------------------------------------------------------- 3907 3908AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3909 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3910 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3911 // mLeftVolFloat, mRightVolFloat 3912{ 3913} 3914 3915AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3916 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3917 ThreadBase::type_t type) 3918 : PlaybackThread(audioFlinger, output, id, device, type) 3919 // mLeftVolFloat, mRightVolFloat 3920{ 3921} 3922 3923AudioFlinger::DirectOutputThread::~DirectOutputThread() 3924{ 3925} 3926 3927void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3928{ 3929 audio_track_cblk_t* cblk = track->cblk(); 3930 float left, right; 3931 3932 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3933 left = right = 0; 3934 } else { 3935 float typeVolume = mStreamTypes[track->streamType()].volume; 3936 float v = mMasterVolume * typeVolume; 3937 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3938 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3939 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3940 if (left > GAIN_FLOAT_UNITY) { 3941 left = GAIN_FLOAT_UNITY; 3942 } 3943 left *= v; 3944 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3945 if (right > GAIN_FLOAT_UNITY) { 3946 right = GAIN_FLOAT_UNITY; 3947 } 3948 right *= v; 3949 } 3950 3951 if (lastTrack) { 3952 if (left != mLeftVolFloat || right != mRightVolFloat) { 3953 mLeftVolFloat = left; 3954 mRightVolFloat = right; 3955 3956 // Convert volumes from float to 8.24 3957 uint32_t vl = (uint32_t)(left * (1 << 24)); 3958 uint32_t vr = (uint32_t)(right * (1 << 24)); 3959 3960 // Delegate volume control to effect in track effect chain if needed 3961 // only one effect chain can be present on DirectOutputThread, so if 3962 // there is one, the track is connected to it 3963 if (!mEffectChains.isEmpty()) { 3964 mEffectChains[0]->setVolume_l(&vl, &vr); 3965 left = (float)vl / (1 << 24); 3966 right = (float)vr / (1 << 24); 3967 } 3968 if (mOutput->stream->set_volume) { 3969 mOutput->stream->set_volume(mOutput->stream, left, right); 3970 } 3971 } 3972 } 3973} 3974 3975 3976AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3977 Vector< sp<Track> > *tracksToRemove 3978) 3979{ 3980 size_t count = mActiveTracks.size(); 3981 mixer_state mixerStatus = MIXER_IDLE; 3982 3983 // find out which tracks need to be processed 3984 for (size_t i = 0; i < count; i++) { 3985 sp<Track> t = mActiveTracks[i].promote(); 3986 // The track died recently 3987 if (t == 0) { 3988 continue; 3989 } 3990 3991 Track* const track = t.get(); 3992 audio_track_cblk_t* cblk = track->cblk(); 3993 // Only consider last track started for volume and mixer state control. 3994 // In theory an older track could underrun and restart after the new one starts 3995 // but as we only care about the transition phase between two tracks on a 3996 // direct output, it is not a problem to ignore the underrun case. 3997 sp<Track> l = mLatestActiveTrack.promote(); 3998 bool last = l.get() == track; 3999 4000 // The first time a track is added we wait 4001 // for all its buffers to be filled before processing it 4002 uint32_t minFrames; 4003 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 4004 minFrames = mNormalFrameCount; 4005 } else { 4006 minFrames = 1; 4007 } 4008 4009 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4010 !track->isStopping_2() && !track->isStopped()) 4011 { 4012 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4013 4014 if (track->mFillingUpStatus == Track::FS_FILLED) { 4015 track->mFillingUpStatus = Track::FS_ACTIVE; 4016 // make sure processVolume_l() will apply new volume even if 0 4017 mLeftVolFloat = mRightVolFloat = -1.0; 4018 if (track->mState == TrackBase::RESUMING) { 4019 track->mState = TrackBase::ACTIVE; 4020 } 4021 } 4022 4023 // compute volume for this track 4024 processVolume_l(track, last); 4025 if (last) { 4026 // reset retry count 4027 track->mRetryCount = kMaxTrackRetriesDirect; 4028 mActiveTrack = t; 4029 mixerStatus = MIXER_TRACKS_READY; 4030 } 4031 } else { 4032 // clear effect chain input buffer if the last active track started underruns 4033 // to avoid sending previous audio buffer again to effects 4034 if (!mEffectChains.isEmpty() && last) { 4035 mEffectChains[0]->clearInputBuffer(); 4036 } 4037 if (track->isStopping_1()) { 4038 track->mState = TrackBase::STOPPING_2; 4039 } 4040 if ((track->sharedBuffer() != 0) || track->isStopped() || 4041 track->isStopping_2() || track->isPaused()) { 4042 // We have consumed all the buffers of this track. 4043 // Remove it from the list of active tracks. 4044 size_t audioHALFrames; 4045 if (audio_is_linear_pcm(mFormat)) { 4046 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4047 } else { 4048 audioHALFrames = 0; 4049 } 4050 4051 size_t framesWritten = mBytesWritten / mFrameSize; 4052 if (mStandby || !last || 4053 track->presentationComplete(framesWritten, audioHALFrames)) { 4054 if (track->isStopping_2()) { 4055 track->mState = TrackBase::STOPPED; 4056 } 4057 if (track->isStopped()) { 4058 if (track->mState == TrackBase::FLUSHED) { 4059 flushHw_l(); 4060 } 4061 track->reset(); 4062 } 4063 tracksToRemove->add(track); 4064 } 4065 } else { 4066 // No buffers for this track. Give it a few chances to 4067 // fill a buffer, then remove it from active list. 4068 // Only consider last track started for mixer state control 4069 if (--(track->mRetryCount) <= 0) { 4070 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4071 tracksToRemove->add(track); 4072 // indicate to client process that the track was disabled because of underrun; 4073 // it will then automatically call start() when data is available 4074 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4075 } else if (last) { 4076 mixerStatus = MIXER_TRACKS_ENABLED; 4077 } 4078 } 4079 } 4080 } 4081 4082 // remove all the tracks that need to be... 4083 removeTracks_l(*tracksToRemove); 4084 4085 return mixerStatus; 4086} 4087 4088void AudioFlinger::DirectOutputThread::threadLoop_mix() 4089{ 4090 size_t frameCount = mFrameCount; 4091 int8_t *curBuf = (int8_t *)mSinkBuffer; 4092 // output audio to hardware 4093 while (frameCount) { 4094 AudioBufferProvider::Buffer buffer; 4095 buffer.frameCount = frameCount; 4096 mActiveTrack->getNextBuffer(&buffer); 4097 if (buffer.raw == NULL) { 4098 memset(curBuf, 0, frameCount * mFrameSize); 4099 break; 4100 } 4101 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4102 frameCount -= buffer.frameCount; 4103 curBuf += buffer.frameCount * mFrameSize; 4104 mActiveTrack->releaseBuffer(&buffer); 4105 } 4106 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4107 sleepTime = 0; 4108 standbyTime = systemTime() + standbyDelay; 4109 mActiveTrack.clear(); 4110} 4111 4112void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4113{ 4114 if (sleepTime == 0) { 4115 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4116 sleepTime = activeSleepTime; 4117 } else { 4118 sleepTime = idleSleepTime; 4119 } 4120 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4121 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4122 sleepTime = 0; 4123 } 4124} 4125 4126// getTrackName_l() must be called with ThreadBase::mLock held 4127int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4128 audio_format_t format __unused, int sessionId __unused) 4129{ 4130 return 0; 4131} 4132 4133// deleteTrackName_l() must be called with ThreadBase::mLock held 4134void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4135{ 4136} 4137 4138// checkForNewParameter_l() must be called with ThreadBase::mLock held 4139bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4140 status_t& status) 4141{ 4142 bool reconfig = false; 4143 4144 status = NO_ERROR; 4145 4146 AudioParameter param = AudioParameter(keyValuePair); 4147 int value; 4148 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4149 // forward device change to effects that have requested to be 4150 // aware of attached audio device. 4151 if (value != AUDIO_DEVICE_NONE) { 4152 mOutDevice = value; 4153 for (size_t i = 0; i < mEffectChains.size(); i++) { 4154 mEffectChains[i]->setDevice_l(mOutDevice); 4155 } 4156 } 4157 } 4158 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4159 // do not accept frame count changes if tracks are open as the track buffer 4160 // size depends on frame count and correct behavior would not be garantied 4161 // if frame count is changed after track creation 4162 if (!mTracks.isEmpty()) { 4163 status = INVALID_OPERATION; 4164 } else { 4165 reconfig = true; 4166 } 4167 } 4168 if (status == NO_ERROR) { 4169 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4170 keyValuePair.string()); 4171 if (!mStandby && status == INVALID_OPERATION) { 4172 mOutput->stream->common.standby(&mOutput->stream->common); 4173 mStandby = true; 4174 mBytesWritten = 0; 4175 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4176 keyValuePair.string()); 4177 } 4178 if (status == NO_ERROR && reconfig) { 4179 readOutputParameters_l(); 4180 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4181 } 4182 } 4183 4184 return reconfig; 4185} 4186 4187uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4188{ 4189 uint32_t time; 4190 if (audio_is_linear_pcm(mFormat)) { 4191 time = PlaybackThread::activeSleepTimeUs(); 4192 } else { 4193 time = 10000; 4194 } 4195 return time; 4196} 4197 4198uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4199{ 4200 uint32_t time; 4201 if (audio_is_linear_pcm(mFormat)) { 4202 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4203 } else { 4204 time = 10000; 4205 } 4206 return time; 4207} 4208 4209uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4210{ 4211 uint32_t time; 4212 if (audio_is_linear_pcm(mFormat)) { 4213 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4214 } else { 4215 time = 10000; 4216 } 4217 return time; 4218} 4219 4220void AudioFlinger::DirectOutputThread::cacheParameters_l() 4221{ 4222 PlaybackThread::cacheParameters_l(); 4223 4224 // use shorter standby delay as on normal output to release 4225 // hardware resources as soon as possible 4226 if (audio_is_linear_pcm(mFormat)) { 4227 standbyDelay = microseconds(activeSleepTime*2); 4228 } else { 4229 standbyDelay = kOffloadStandbyDelayNs; 4230 } 4231} 4232 4233void AudioFlinger::DirectOutputThread::flushHw_l() 4234{ 4235 if (mOutput->stream->flush != NULL) 4236 mOutput->stream->flush(mOutput->stream); 4237} 4238 4239// ---------------------------------------------------------------------------- 4240 4241AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4242 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4243 : Thread(false /*canCallJava*/), 4244 mPlaybackThread(playbackThread), 4245 mWriteAckSequence(0), 4246 mDrainSequence(0) 4247{ 4248} 4249 4250AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4251{ 4252} 4253 4254void AudioFlinger::AsyncCallbackThread::onFirstRef() 4255{ 4256 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4257} 4258 4259bool AudioFlinger::AsyncCallbackThread::threadLoop() 4260{ 4261 while (!exitPending()) { 4262 uint32_t writeAckSequence; 4263 uint32_t drainSequence; 4264 4265 { 4266 Mutex::Autolock _l(mLock); 4267 while (!((mWriteAckSequence & 1) || 4268 (mDrainSequence & 1) || 4269 exitPending())) { 4270 mWaitWorkCV.wait(mLock); 4271 } 4272 4273 if (exitPending()) { 4274 break; 4275 } 4276 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4277 mWriteAckSequence, mDrainSequence); 4278 writeAckSequence = mWriteAckSequence; 4279 mWriteAckSequence &= ~1; 4280 drainSequence = mDrainSequence; 4281 mDrainSequence &= ~1; 4282 } 4283 { 4284 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4285 if (playbackThread != 0) { 4286 if (writeAckSequence & 1) { 4287 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4288 } 4289 if (drainSequence & 1) { 4290 playbackThread->resetDraining(drainSequence >> 1); 4291 } 4292 } 4293 } 4294 } 4295 return false; 4296} 4297 4298void AudioFlinger::AsyncCallbackThread::exit() 4299{ 4300 ALOGV("AsyncCallbackThread::exit"); 4301 Mutex::Autolock _l(mLock); 4302 requestExit(); 4303 mWaitWorkCV.broadcast(); 4304} 4305 4306void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4307{ 4308 Mutex::Autolock _l(mLock); 4309 // bit 0 is cleared 4310 mWriteAckSequence = sequence << 1; 4311} 4312 4313void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4314{ 4315 Mutex::Autolock _l(mLock); 4316 // ignore unexpected callbacks 4317 if (mWriteAckSequence & 2) { 4318 mWriteAckSequence |= 1; 4319 mWaitWorkCV.signal(); 4320 } 4321} 4322 4323void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4324{ 4325 Mutex::Autolock _l(mLock); 4326 // bit 0 is cleared 4327 mDrainSequence = sequence << 1; 4328} 4329 4330void AudioFlinger::AsyncCallbackThread::resetDraining() 4331{ 4332 Mutex::Autolock _l(mLock); 4333 // ignore unexpected callbacks 4334 if (mDrainSequence & 2) { 4335 mDrainSequence |= 1; 4336 mWaitWorkCV.signal(); 4337 } 4338} 4339 4340 4341// ---------------------------------------------------------------------------- 4342AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4343 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4344 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4345 mHwPaused(false), 4346 mFlushPending(false), 4347 mPausedBytesRemaining(0) 4348{ 4349 //FIXME: mStandby should be set to true by ThreadBase constructor 4350 mStandby = true; 4351} 4352 4353void AudioFlinger::OffloadThread::threadLoop_exit() 4354{ 4355 if (mFlushPending || mHwPaused) { 4356 // If a flush is pending or track was paused, just discard buffered data 4357 flushHw_l(); 4358 } else { 4359 mMixerStatus = MIXER_DRAIN_ALL; 4360 threadLoop_drain(); 4361 } 4362 if (mUseAsyncWrite) { 4363 ALOG_ASSERT(mCallbackThread != 0); 4364 mCallbackThread->exit(); 4365 } 4366 PlaybackThread::threadLoop_exit(); 4367} 4368 4369AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4370 Vector< sp<Track> > *tracksToRemove 4371) 4372{ 4373 size_t count = mActiveTracks.size(); 4374 4375 mixer_state mixerStatus = MIXER_IDLE; 4376 bool doHwPause = false; 4377 bool doHwResume = false; 4378 4379 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4380 4381 // find out which tracks need to be processed 4382 for (size_t i = 0; i < count; i++) { 4383 sp<Track> t = mActiveTracks[i].promote(); 4384 // The track died recently 4385 if (t == 0) { 4386 continue; 4387 } 4388 Track* const track = t.get(); 4389 audio_track_cblk_t* cblk = track->cblk(); 4390 // Only consider last track started for volume and mixer state control. 4391 // In theory an older track could underrun and restart after the new one starts 4392 // but as we only care about the transition phase between two tracks on a 4393 // direct output, it is not a problem to ignore the underrun case. 4394 sp<Track> l = mLatestActiveTrack.promote(); 4395 bool last = l.get() == track; 4396 4397 if (track->isInvalid()) { 4398 ALOGW("An invalidated track shouldn't be in active list"); 4399 tracksToRemove->add(track); 4400 continue; 4401 } 4402 4403 if (track->mState == TrackBase::IDLE) { 4404 ALOGW("An idle track shouldn't be in active list"); 4405 continue; 4406 } 4407 4408 if (track->isPausing()) { 4409 track->setPaused(); 4410 if (last) { 4411 if (!mHwPaused) { 4412 doHwPause = true; 4413 mHwPaused = true; 4414 } 4415 // If we were part way through writing the mixbuffer to 4416 // the HAL we must save this until we resume 4417 // BUG - this will be wrong if a different track is made active, 4418 // in that case we want to discard the pending data in the 4419 // mixbuffer and tell the client to present it again when the 4420 // track is resumed 4421 mPausedWriteLength = mCurrentWriteLength; 4422 mPausedBytesRemaining = mBytesRemaining; 4423 mBytesRemaining = 0; // stop writing 4424 } 4425 tracksToRemove->add(track); 4426 } else if (track->isFlushPending()) { 4427 track->flushAck(); 4428 if (last) { 4429 mFlushPending = true; 4430 } 4431 } else if (track->isResumePending()){ 4432 track->resumeAck(); 4433 if (last) { 4434 if (mPausedBytesRemaining) { 4435 // Need to continue write that was interrupted 4436 mCurrentWriteLength = mPausedWriteLength; 4437 mBytesRemaining = mPausedBytesRemaining; 4438 mPausedBytesRemaining = 0; 4439 } 4440 if (mHwPaused) { 4441 doHwResume = true; 4442 mHwPaused = false; 4443 // threadLoop_mix() will handle the case that we need to 4444 // resume an interrupted write 4445 } 4446 // enable write to audio HAL 4447 sleepTime = 0; 4448 4449 // Do not handle new data in this iteration even if track->framesReady() 4450 mixerStatus = MIXER_TRACKS_ENABLED; 4451 } 4452 } else if (track->framesReady() && track->isReady() && 4453 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4454 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4455 if (track->mFillingUpStatus == Track::FS_FILLED) { 4456 track->mFillingUpStatus = Track::FS_ACTIVE; 4457 // make sure processVolume_l() will apply new volume even if 0 4458 mLeftVolFloat = mRightVolFloat = -1.0; 4459 } 4460 4461 if (last) { 4462 sp<Track> previousTrack = mPreviousTrack.promote(); 4463 if (previousTrack != 0) { 4464 if (track != previousTrack.get()) { 4465 // Flush any data still being written from last track 4466 mBytesRemaining = 0; 4467 if (mPausedBytesRemaining) { 4468 // Last track was paused so we also need to flush saved 4469 // mixbuffer state and invalidate track so that it will 4470 // re-submit that unwritten data when it is next resumed 4471 mPausedBytesRemaining = 0; 4472 // Invalidate is a bit drastic - would be more efficient 4473 // to have a flag to tell client that some of the 4474 // previously written data was lost 4475 previousTrack->invalidate(); 4476 } 4477 // flush data already sent to the DSP if changing audio session as audio 4478 // comes from a different source. Also invalidate previous track to force a 4479 // seek when resuming. 4480 if (previousTrack->sessionId() != track->sessionId()) { 4481 previousTrack->invalidate(); 4482 } 4483 } 4484 } 4485 mPreviousTrack = track; 4486 // reset retry count 4487 track->mRetryCount = kMaxTrackRetriesOffload; 4488 mActiveTrack = t; 4489 mixerStatus = MIXER_TRACKS_READY; 4490 } 4491 } else { 4492 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4493 if (track->isStopping_1()) { 4494 // Hardware buffer can hold a large amount of audio so we must 4495 // wait for all current track's data to drain before we say 4496 // that the track is stopped. 4497 if (mBytesRemaining == 0) { 4498 // Only start draining when all data in mixbuffer 4499 // has been written 4500 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4501 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4502 // do not drain if no data was ever sent to HAL (mStandby == true) 4503 if (last && !mStandby) { 4504 // do not modify drain sequence if we are already draining. This happens 4505 // when resuming from pause after drain. 4506 if ((mDrainSequence & 1) == 0) { 4507 sleepTime = 0; 4508 standbyTime = systemTime() + standbyDelay; 4509 mixerStatus = MIXER_DRAIN_TRACK; 4510 mDrainSequence += 2; 4511 } 4512 if (mHwPaused) { 4513 // It is possible to move from PAUSED to STOPPING_1 without 4514 // a resume so we must ensure hardware is running 4515 doHwResume = true; 4516 mHwPaused = false; 4517 } 4518 } 4519 } 4520 } else if (track->isStopping_2()) { 4521 // Drain has completed or we are in standby, signal presentation complete 4522 if (!(mDrainSequence & 1) || !last || mStandby) { 4523 track->mState = TrackBase::STOPPED; 4524 size_t audioHALFrames = 4525 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4526 size_t framesWritten = 4527 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4528 track->presentationComplete(framesWritten, audioHALFrames); 4529 track->reset(); 4530 tracksToRemove->add(track); 4531 } 4532 } else { 4533 // No buffers for this track. Give it a few chances to 4534 // fill a buffer, then remove it from active list. 4535 if (--(track->mRetryCount) <= 0) { 4536 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4537 track->name()); 4538 tracksToRemove->add(track); 4539 // indicate to client process that the track was disabled because of underrun; 4540 // it will then automatically call start() when data is available 4541 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4542 } else if (last){ 4543 mixerStatus = MIXER_TRACKS_ENABLED; 4544 } 4545 } 4546 } 4547 // compute volume for this track 4548 processVolume_l(track, last); 4549 } 4550 4551 // make sure the pause/flush/resume sequence is executed in the right order. 4552 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4553 // before flush and then resume HW. This can happen in case of pause/flush/resume 4554 // if resume is received before pause is executed. 4555 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4556 mOutput->stream->pause(mOutput->stream); 4557 } 4558 if (mFlushPending) { 4559 flushHw_l(); 4560 mFlushPending = false; 4561 } 4562 if (!mStandby && doHwResume) { 4563 mOutput->stream->resume(mOutput->stream); 4564 } 4565 4566 // remove all the tracks that need to be... 4567 removeTracks_l(*tracksToRemove); 4568 4569 return mixerStatus; 4570} 4571 4572// must be called with thread mutex locked 4573bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4574{ 4575 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4576 mWriteAckSequence, mDrainSequence); 4577 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4578 return true; 4579 } 4580 return false; 4581} 4582 4583// must be called with thread mutex locked 4584bool AudioFlinger::OffloadThread::shouldStandby_l() 4585{ 4586 bool trackPaused = false; 4587 4588 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4589 // after a timeout and we will enter standby then. 4590 if (mTracks.size() > 0) { 4591 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4592 } 4593 4594 return !mStandby && !trackPaused; 4595} 4596 4597 4598bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4599{ 4600 Mutex::Autolock _l(mLock); 4601 return waitingAsyncCallback_l(); 4602} 4603 4604void AudioFlinger::OffloadThread::flushHw_l() 4605{ 4606 DirectOutputThread::flushHw_l(); 4607 // Flush anything still waiting in the mixbuffer 4608 mCurrentWriteLength = 0; 4609 mBytesRemaining = 0; 4610 mPausedWriteLength = 0; 4611 mPausedBytesRemaining = 0; 4612 mHwPaused = false; 4613 4614 if (mUseAsyncWrite) { 4615 // discard any pending drain or write ack by incrementing sequence 4616 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4617 mDrainSequence = (mDrainSequence + 2) & ~1; 4618 ALOG_ASSERT(mCallbackThread != 0); 4619 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4620 mCallbackThread->setDraining(mDrainSequence); 4621 } 4622} 4623 4624void AudioFlinger::OffloadThread::onAddNewTrack_l() 4625{ 4626 sp<Track> previousTrack = mPreviousTrack.promote(); 4627 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4628 4629 if (previousTrack != 0 && latestTrack != 0 && 4630 (previousTrack->sessionId() != latestTrack->sessionId())) { 4631 mFlushPending = true; 4632 } 4633 PlaybackThread::onAddNewTrack_l(); 4634} 4635 4636// ---------------------------------------------------------------------------- 4637 4638AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4639 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4640 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4641 DUPLICATING), 4642 mWaitTimeMs(UINT_MAX) 4643{ 4644 addOutputTrack(mainThread); 4645} 4646 4647AudioFlinger::DuplicatingThread::~DuplicatingThread() 4648{ 4649 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4650 mOutputTracks[i]->destroy(); 4651 } 4652} 4653 4654void AudioFlinger::DuplicatingThread::threadLoop_mix() 4655{ 4656 // mix buffers... 4657 if (outputsReady(outputTracks)) { 4658 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4659 } else { 4660 memset(mSinkBuffer, 0, mSinkBufferSize); 4661 } 4662 sleepTime = 0; 4663 writeFrames = mNormalFrameCount; 4664 mCurrentWriteLength = mSinkBufferSize; 4665 standbyTime = systemTime() + standbyDelay; 4666} 4667 4668void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4669{ 4670 if (sleepTime == 0) { 4671 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4672 sleepTime = activeSleepTime; 4673 } else { 4674 sleepTime = idleSleepTime; 4675 } 4676 } else if (mBytesWritten != 0) { 4677 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4678 writeFrames = mNormalFrameCount; 4679 memset(mSinkBuffer, 0, mSinkBufferSize); 4680 } else { 4681 // flush remaining overflow buffers in output tracks 4682 writeFrames = 0; 4683 } 4684 sleepTime = 0; 4685 } 4686} 4687 4688ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4689{ 4690 for (size_t i = 0; i < outputTracks.size(); i++) { 4691 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4692 // for delivery downstream as needed. This in-place conversion is safe as 4693 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4694 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4695 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4696 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4697 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4698 } 4699 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4700 } 4701 mStandby = false; 4702 return (ssize_t)mSinkBufferSize; 4703} 4704 4705void AudioFlinger::DuplicatingThread::threadLoop_standby() 4706{ 4707 // DuplicatingThread implements standby by stopping all tracks 4708 for (size_t i = 0; i < outputTracks.size(); i++) { 4709 outputTracks[i]->stop(); 4710 } 4711} 4712 4713void AudioFlinger::DuplicatingThread::saveOutputTracks() 4714{ 4715 outputTracks = mOutputTracks; 4716} 4717 4718void AudioFlinger::DuplicatingThread::clearOutputTracks() 4719{ 4720 outputTracks.clear(); 4721} 4722 4723void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4724{ 4725 Mutex::Autolock _l(mLock); 4726 // FIXME explain this formula 4727 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4728 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4729 // due to current usage case and restrictions on the AudioBufferProvider. 4730 // Actual buffer conversion is done in threadLoop_write(). 4731 // 4732 // TODO: This may change in the future, depending on multichannel 4733 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4734 OutputTrack *outputTrack = new OutputTrack(thread, 4735 this, 4736 mSampleRate, 4737 AUDIO_FORMAT_PCM_16_BIT, 4738 mChannelMask, 4739 frameCount, 4740 IPCThreadState::self()->getCallingUid()); 4741 if (outputTrack->cblk() != NULL) { 4742 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4743 mOutputTracks.add(outputTrack); 4744 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4745 updateWaitTime_l(); 4746 } 4747} 4748 4749void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4750{ 4751 Mutex::Autolock _l(mLock); 4752 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4753 if (mOutputTracks[i]->thread() == thread) { 4754 mOutputTracks[i]->destroy(); 4755 mOutputTracks.removeAt(i); 4756 updateWaitTime_l(); 4757 return; 4758 } 4759 } 4760 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4761} 4762 4763// caller must hold mLock 4764void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4765{ 4766 mWaitTimeMs = UINT_MAX; 4767 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4768 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4769 if (strong != 0) { 4770 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4771 if (waitTimeMs < mWaitTimeMs) { 4772 mWaitTimeMs = waitTimeMs; 4773 } 4774 } 4775 } 4776} 4777 4778 4779bool AudioFlinger::DuplicatingThread::outputsReady( 4780 const SortedVector< sp<OutputTrack> > &outputTracks) 4781{ 4782 for (size_t i = 0; i < outputTracks.size(); i++) { 4783 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4784 if (thread == 0) { 4785 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4786 outputTracks[i].get()); 4787 return false; 4788 } 4789 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4790 // see note at standby() declaration 4791 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4792 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4793 thread.get()); 4794 return false; 4795 } 4796 } 4797 return true; 4798} 4799 4800uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4801{ 4802 return (mWaitTimeMs * 1000) / 2; 4803} 4804 4805void AudioFlinger::DuplicatingThread::cacheParameters_l() 4806{ 4807 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4808 updateWaitTime_l(); 4809 4810 MixerThread::cacheParameters_l(); 4811} 4812 4813// ---------------------------------------------------------------------------- 4814// Record 4815// ---------------------------------------------------------------------------- 4816 4817AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4818 AudioStreamIn *input, 4819 audio_io_handle_t id, 4820 audio_devices_t outDevice, 4821 audio_devices_t inDevice 4822#ifdef TEE_SINK 4823 , const sp<NBAIO_Sink>& teeSink 4824#endif 4825 ) : 4826 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4827 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4828 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4829 mRsmpInRear(0) 4830#ifdef TEE_SINK 4831 , mTeeSink(teeSink) 4832#endif 4833 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4834 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4835 // mFastCapture below 4836 , mFastCaptureFutex(0) 4837 // mInputSource 4838 // mPipeSink 4839 // mPipeSource 4840 , mPipeFramesP2(0) 4841 // mPipeMemory 4842 // mFastCaptureNBLogWriter 4843 , mFastTrackAvail(false) 4844{ 4845 snprintf(mName, kNameLength, "AudioIn_%X", id); 4846 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4847 4848 readInputParameters_l(); 4849 4850 // create an NBAIO source for the HAL input stream, and negotiate 4851 mInputSource = new AudioStreamInSource(input->stream); 4852 size_t numCounterOffers = 0; 4853 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4854 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4855 ALOG_ASSERT(index == 0); 4856 4857 // initialize fast capture depending on configuration 4858 bool initFastCapture; 4859 switch (kUseFastCapture) { 4860 case FastCapture_Never: 4861 initFastCapture = false; 4862 break; 4863 case FastCapture_Always: 4864 initFastCapture = true; 4865 break; 4866 case FastCapture_Static: 4867 uint32_t primaryOutputSampleRate; 4868 { 4869 AutoMutex _l(audioFlinger->mHardwareLock); 4870 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4871 } 4872 initFastCapture = 4873 // either capture sample rate is same as (a reasonable) primary output sample rate 4874 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4875 (mSampleRate == primaryOutputSampleRate)) || 4876 // or primary output sample rate is unknown, and capture sample rate is reasonable 4877 ((primaryOutputSampleRate == 0) && 4878 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4879 // and the buffer size is < 12 ms 4880 (mFrameCount * 1000) / mSampleRate < 12; 4881 break; 4882 // case FastCapture_Dynamic: 4883 } 4884 4885 if (initFastCapture) { 4886 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4887 NBAIO_Format format = mInputSource->format(); 4888 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4889 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4890 void *pipeBuffer; 4891 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4892 sp<IMemory> pipeMemory; 4893 if ((roHeap == 0) || 4894 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4895 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4896 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4897 goto failed; 4898 } 4899 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4900 memset(pipeBuffer, 0, pipeSize); 4901 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4902 const NBAIO_Format offers[1] = {format}; 4903 size_t numCounterOffers = 0; 4904 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4905 ALOG_ASSERT(index == 0); 4906 mPipeSink = pipe; 4907 PipeReader *pipeReader = new PipeReader(*pipe); 4908 numCounterOffers = 0; 4909 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4910 ALOG_ASSERT(index == 0); 4911 mPipeSource = pipeReader; 4912 mPipeFramesP2 = pipeFramesP2; 4913 mPipeMemory = pipeMemory; 4914 4915 // create fast capture 4916 mFastCapture = new FastCapture(); 4917 FastCaptureStateQueue *sq = mFastCapture->sq(); 4918#ifdef STATE_QUEUE_DUMP 4919 // FIXME 4920#endif 4921 FastCaptureState *state = sq->begin(); 4922 state->mCblk = NULL; 4923 state->mInputSource = mInputSource.get(); 4924 state->mInputSourceGen++; 4925 state->mPipeSink = pipe; 4926 state->mPipeSinkGen++; 4927 state->mFrameCount = mFrameCount; 4928 state->mCommand = FastCaptureState::COLD_IDLE; 4929 // already done in constructor initialization list 4930 //mFastCaptureFutex = 0; 4931 state->mColdFutexAddr = &mFastCaptureFutex; 4932 state->mColdGen++; 4933 state->mDumpState = &mFastCaptureDumpState; 4934#ifdef TEE_SINK 4935 // FIXME 4936#endif 4937 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4938 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4939 sq->end(); 4940 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4941 4942 // start the fast capture 4943 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4944 pid_t tid = mFastCapture->getTid(); 4945 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4946 if (err != 0) { 4947 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4948 kPriorityFastCapture, getpid_cached, tid, err); 4949 } 4950 4951#ifdef AUDIO_WATCHDOG 4952 // FIXME 4953#endif 4954 4955 mFastTrackAvail = true; 4956 } 4957failed: ; 4958 4959 // FIXME mNormalSource 4960} 4961 4962 4963AudioFlinger::RecordThread::~RecordThread() 4964{ 4965 if (mFastCapture != 0) { 4966 FastCaptureStateQueue *sq = mFastCapture->sq(); 4967 FastCaptureState *state = sq->begin(); 4968 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4969 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4970 if (old == -1) { 4971 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4972 } 4973 } 4974 state->mCommand = FastCaptureState::EXIT; 4975 sq->end(); 4976 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4977 mFastCapture->join(); 4978 mFastCapture.clear(); 4979 } 4980 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4981 mAudioFlinger->unregisterWriter(mNBLogWriter); 4982 delete[] mRsmpInBuffer; 4983} 4984 4985void AudioFlinger::RecordThread::onFirstRef() 4986{ 4987 run(mName, PRIORITY_URGENT_AUDIO); 4988} 4989 4990bool AudioFlinger::RecordThread::threadLoop() 4991{ 4992 nsecs_t lastWarning = 0; 4993 4994 inputStandBy(); 4995 4996reacquire_wakelock: 4997 sp<RecordTrack> activeTrack; 4998 int activeTracksGen; 4999 { 5000 Mutex::Autolock _l(mLock); 5001 size_t size = mActiveTracks.size(); 5002 activeTracksGen = mActiveTracksGen; 5003 if (size > 0) { 5004 // FIXME an arbitrary choice 5005 activeTrack = mActiveTracks[0]; 5006 acquireWakeLock_l(activeTrack->uid()); 5007 if (size > 1) { 5008 SortedVector<int> tmp; 5009 for (size_t i = 0; i < size; i++) { 5010 tmp.add(mActiveTracks[i]->uid()); 5011 } 5012 updateWakeLockUids_l(tmp); 5013 } 5014 } else { 5015 acquireWakeLock_l(-1); 5016 } 5017 } 5018 5019 // used to request a deferred sleep, to be executed later while mutex is unlocked 5020 uint32_t sleepUs = 0; 5021 5022 // loop while there is work to do 5023 for (;;) { 5024 Vector< sp<EffectChain> > effectChains; 5025 5026 // sleep with mutex unlocked 5027 if (sleepUs > 0) { 5028 usleep(sleepUs); 5029 sleepUs = 0; 5030 } 5031 5032 // activeTracks accumulates a copy of a subset of mActiveTracks 5033 Vector< sp<RecordTrack> > activeTracks; 5034 5035 // reference to the (first and only) active fast track 5036 sp<RecordTrack> fastTrack; 5037 5038 // reference to a fast track which is about to be removed 5039 sp<RecordTrack> fastTrackToRemove; 5040 5041 { // scope for mLock 5042 Mutex::Autolock _l(mLock); 5043 5044 processConfigEvents_l(); 5045 5046 // check exitPending here because checkForNewParameters_l() and 5047 // checkForNewParameters_l() can temporarily release mLock 5048 if (exitPending()) { 5049 break; 5050 } 5051 5052 // if no active track(s), then standby and release wakelock 5053 size_t size = mActiveTracks.size(); 5054 if (size == 0) { 5055 standbyIfNotAlreadyInStandby(); 5056 // exitPending() can't become true here 5057 releaseWakeLock_l(); 5058 ALOGV("RecordThread: loop stopping"); 5059 // go to sleep 5060 mWaitWorkCV.wait(mLock); 5061 ALOGV("RecordThread: loop starting"); 5062 goto reacquire_wakelock; 5063 } 5064 5065 if (mActiveTracksGen != activeTracksGen) { 5066 activeTracksGen = mActiveTracksGen; 5067 SortedVector<int> tmp; 5068 for (size_t i = 0; i < size; i++) { 5069 tmp.add(mActiveTracks[i]->uid()); 5070 } 5071 updateWakeLockUids_l(tmp); 5072 } 5073 5074 bool doBroadcast = false; 5075 for (size_t i = 0; i < size; ) { 5076 5077 activeTrack = mActiveTracks[i]; 5078 if (activeTrack->isTerminated()) { 5079 if (activeTrack->isFastTrack()) { 5080 ALOG_ASSERT(fastTrackToRemove == 0); 5081 fastTrackToRemove = activeTrack; 5082 } 5083 removeTrack_l(activeTrack); 5084 mActiveTracks.remove(activeTrack); 5085 mActiveTracksGen++; 5086 size--; 5087 continue; 5088 } 5089 5090 TrackBase::track_state activeTrackState = activeTrack->mState; 5091 switch (activeTrackState) { 5092 5093 case TrackBase::PAUSING: 5094 mActiveTracks.remove(activeTrack); 5095 mActiveTracksGen++; 5096 doBroadcast = true; 5097 size--; 5098 continue; 5099 5100 case TrackBase::STARTING_1: 5101 sleepUs = 10000; 5102 i++; 5103 continue; 5104 5105 case TrackBase::STARTING_2: 5106 doBroadcast = true; 5107 mStandby = false; 5108 activeTrack->mState = TrackBase::ACTIVE; 5109 break; 5110 5111 case TrackBase::ACTIVE: 5112 break; 5113 5114 case TrackBase::IDLE: 5115 i++; 5116 continue; 5117 5118 default: 5119 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5120 } 5121 5122 activeTracks.add(activeTrack); 5123 i++; 5124 5125 if (activeTrack->isFastTrack()) { 5126 ALOG_ASSERT(!mFastTrackAvail); 5127 ALOG_ASSERT(fastTrack == 0); 5128 fastTrack = activeTrack; 5129 } 5130 } 5131 if (doBroadcast) { 5132 mStartStopCond.broadcast(); 5133 } 5134 5135 // sleep if there are no active tracks to process 5136 if (activeTracks.size() == 0) { 5137 if (sleepUs == 0) { 5138 sleepUs = kRecordThreadSleepUs; 5139 } 5140 continue; 5141 } 5142 sleepUs = 0; 5143 5144 lockEffectChains_l(effectChains); 5145 } 5146 5147 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5148 5149 size_t size = effectChains.size(); 5150 for (size_t i = 0; i < size; i++) { 5151 // thread mutex is not locked, but effect chain is locked 5152 effectChains[i]->process_l(); 5153 } 5154 5155 // Push a new fast capture state if fast capture is not already running, or cblk change 5156 if (mFastCapture != 0) { 5157 FastCaptureStateQueue *sq = mFastCapture->sq(); 5158 FastCaptureState *state = sq->begin(); 5159 bool didModify = false; 5160 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5161 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5162 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5163 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5164 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5165 if (old == -1) { 5166 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5167 } 5168 } 5169 state->mCommand = FastCaptureState::READ_WRITE; 5170#if 0 // FIXME 5171 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5172 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5173#endif 5174 didModify = true; 5175 } 5176 audio_track_cblk_t *cblkOld = state->mCblk; 5177 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5178 if (cblkNew != cblkOld) { 5179 state->mCblk = cblkNew; 5180 // block until acked if removing a fast track 5181 if (cblkOld != NULL) { 5182 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5183 } 5184 didModify = true; 5185 } 5186 sq->end(didModify); 5187 if (didModify) { 5188 sq->push(block); 5189#if 0 5190 if (kUseFastCapture == FastCapture_Dynamic) { 5191 mNormalSource = mPipeSource; 5192 } 5193#endif 5194 } 5195 } 5196 5197 // now run the fast track destructor with thread mutex unlocked 5198 fastTrackToRemove.clear(); 5199 5200 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5201 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5202 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5203 // If destination is non-contiguous, first read past the nominal end of buffer, then 5204 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5205 5206 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5207 ssize_t framesRead; 5208 5209 // If an NBAIO source is present, use it to read the normal capture's data 5210 if (mPipeSource != 0) { 5211 size_t framesToRead = mBufferSize / mFrameSize; 5212 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5213 framesToRead, AudioBufferProvider::kInvalidPTS); 5214 if (framesRead == 0) { 5215 // since pipe is non-blocking, simulate blocking input 5216 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5217 } 5218 // otherwise use the HAL / AudioStreamIn directly 5219 } else { 5220 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5221 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5222 if (bytesRead < 0) { 5223 framesRead = bytesRead; 5224 } else { 5225 framesRead = bytesRead / mFrameSize; 5226 } 5227 } 5228 5229 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5230 ALOGE("read failed: framesRead=%d", framesRead); 5231 // Force input into standby so that it tries to recover at next read attempt 5232 inputStandBy(); 5233 sleepUs = kRecordThreadSleepUs; 5234 } 5235 if (framesRead <= 0) { 5236 goto unlock; 5237 } 5238 ALOG_ASSERT(framesRead > 0); 5239 5240 if (mTeeSink != 0) { 5241 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5242 } 5243 // If destination is non-contiguous, we now correct for reading past end of buffer. 5244 { 5245 size_t part1 = mRsmpInFramesP2 - rear; 5246 if ((size_t) framesRead > part1) { 5247 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5248 (framesRead - part1) * mFrameSize); 5249 } 5250 } 5251 rear = mRsmpInRear += framesRead; 5252 5253 size = activeTracks.size(); 5254 // loop over each active track 5255 for (size_t i = 0; i < size; i++) { 5256 activeTrack = activeTracks[i]; 5257 5258 // skip fast tracks, as those are handled directly by FastCapture 5259 if (activeTrack->isFastTrack()) { 5260 continue; 5261 } 5262 5263 enum { 5264 OVERRUN_UNKNOWN, 5265 OVERRUN_TRUE, 5266 OVERRUN_FALSE 5267 } overrun = OVERRUN_UNKNOWN; 5268 5269 // loop over getNextBuffer to handle circular sink 5270 for (;;) { 5271 5272 activeTrack->mSink.frameCount = ~0; 5273 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5274 size_t framesOut = activeTrack->mSink.frameCount; 5275 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5276 5277 int32_t front = activeTrack->mRsmpInFront; 5278 ssize_t filled = rear - front; 5279 size_t framesIn; 5280 5281 if (filled < 0) { 5282 // should not happen, but treat like a massive overrun and re-sync 5283 framesIn = 0; 5284 activeTrack->mRsmpInFront = rear; 5285 overrun = OVERRUN_TRUE; 5286 } else if ((size_t) filled <= mRsmpInFrames) { 5287 framesIn = (size_t) filled; 5288 } else { 5289 // client is not keeping up with server, but give it latest data 5290 framesIn = mRsmpInFrames; 5291 activeTrack->mRsmpInFront = front = rear - framesIn; 5292 overrun = OVERRUN_TRUE; 5293 } 5294 5295 if (framesOut == 0 || framesIn == 0) { 5296 break; 5297 } 5298 5299 if (activeTrack->mResampler == NULL) { 5300 // no resampling 5301 if (framesIn > framesOut) { 5302 framesIn = framesOut; 5303 } else { 5304 framesOut = framesIn; 5305 } 5306 int8_t *dst = activeTrack->mSink.i8; 5307 while (framesIn > 0) { 5308 front &= mRsmpInFramesP2 - 1; 5309 size_t part1 = mRsmpInFramesP2 - front; 5310 if (part1 > framesIn) { 5311 part1 = framesIn; 5312 } 5313 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5314 if (mChannelCount == activeTrack->mChannelCount) { 5315 memcpy(dst, src, part1 * mFrameSize); 5316 } else if (mChannelCount == 1) { 5317 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5318 part1); 5319 } else { 5320 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5321 part1); 5322 } 5323 dst += part1 * activeTrack->mFrameSize; 5324 front += part1; 5325 framesIn -= part1; 5326 } 5327 activeTrack->mRsmpInFront += framesOut; 5328 5329 } else { 5330 // resampling 5331 // FIXME framesInNeeded should really be part of resampler API, and should 5332 // depend on the SRC ratio 5333 // to keep mRsmpInBuffer full so resampler always has sufficient input 5334 size_t framesInNeeded; 5335 // FIXME only re-calculate when it changes, and optimize for common ratios 5336 // Do not precompute in/out because floating point is not associative 5337 // e.g. a*b/c != a*(b/c). 5338 const double in(mSampleRate); 5339 const double out(activeTrack->mSampleRate); 5340 framesInNeeded = ceil(framesOut * in / out) + 1; 5341 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5342 framesInNeeded, framesOut, in / out); 5343 // Although we theoretically have framesIn in circular buffer, some of those are 5344 // unreleased frames, and thus must be discounted for purpose of budgeting. 5345 size_t unreleased = activeTrack->mRsmpInUnrel; 5346 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5347 if (framesIn < framesInNeeded) { 5348 ALOGV("not enough to resample: have %u frames in but need %u in to " 5349 "produce %u out given in/out ratio of %.4g", 5350 framesIn, framesInNeeded, framesOut, in / out); 5351 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5352 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5353 if (newFramesOut == 0) { 5354 break; 5355 } 5356 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5357 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5358 framesInNeeded, newFramesOut, out / in); 5359 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5360 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5361 "given in/out ratio of %.4g", 5362 framesIn, framesInNeeded, newFramesOut, in / out); 5363 framesOut = newFramesOut; 5364 } else { 5365 ALOGV("success 1: have %u in and need %u in to produce %u out " 5366 "given in/out ratio of %.4g", 5367 framesIn, framesInNeeded, framesOut, in / out); 5368 } 5369 5370 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5371 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5372 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5373 delete[] activeTrack->mRsmpOutBuffer; 5374 // resampler always outputs stereo 5375 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5376 activeTrack->mRsmpOutFrameCount = framesOut; 5377 } 5378 5379 // resampler accumulates, but we only have one source track 5380 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5381 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5382 // FIXME how about having activeTrack implement this interface itself? 5383 activeTrack->mResamplerBufferProvider 5384 /*this*/ /* AudioBufferProvider* */); 5385 // ditherAndClamp() works as long as all buffers returned by 5386 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5387 if (activeTrack->mChannelCount == 1) { 5388 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5389 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5390 framesOut); 5391 // the resampler always outputs stereo samples: 5392 // do post stereo to mono conversion 5393 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5394 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5395 } else { 5396 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5397 activeTrack->mRsmpOutBuffer, framesOut); 5398 } 5399 // now done with mRsmpOutBuffer 5400 5401 } 5402 5403 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5404 overrun = OVERRUN_FALSE; 5405 } 5406 5407 if (activeTrack->mFramesToDrop == 0) { 5408 if (framesOut > 0) { 5409 activeTrack->mSink.frameCount = framesOut; 5410 activeTrack->releaseBuffer(&activeTrack->mSink); 5411 } 5412 } else { 5413 // FIXME could do a partial drop of framesOut 5414 if (activeTrack->mFramesToDrop > 0) { 5415 activeTrack->mFramesToDrop -= framesOut; 5416 if (activeTrack->mFramesToDrop <= 0) { 5417 activeTrack->clearSyncStartEvent(); 5418 } 5419 } else { 5420 activeTrack->mFramesToDrop += framesOut; 5421 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5422 activeTrack->mSyncStartEvent->isCancelled()) { 5423 ALOGW("Synced record %s, session %d, trigger session %d", 5424 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5425 activeTrack->sessionId(), 5426 (activeTrack->mSyncStartEvent != 0) ? 5427 activeTrack->mSyncStartEvent->triggerSession() : 0); 5428 activeTrack->clearSyncStartEvent(); 5429 } 5430 } 5431 } 5432 5433 if (framesOut == 0) { 5434 break; 5435 } 5436 } 5437 5438 switch (overrun) { 5439 case OVERRUN_TRUE: 5440 // client isn't retrieving buffers fast enough 5441 if (!activeTrack->setOverflow()) { 5442 nsecs_t now = systemTime(); 5443 // FIXME should lastWarning per track? 5444 if ((now - lastWarning) > kWarningThrottleNs) { 5445 ALOGW("RecordThread: buffer overflow"); 5446 lastWarning = now; 5447 } 5448 } 5449 break; 5450 case OVERRUN_FALSE: 5451 activeTrack->clearOverflow(); 5452 break; 5453 case OVERRUN_UNKNOWN: 5454 break; 5455 } 5456 5457 } 5458 5459unlock: 5460 // enable changes in effect chain 5461 unlockEffectChains(effectChains); 5462 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5463 } 5464 5465 standbyIfNotAlreadyInStandby(); 5466 5467 { 5468 Mutex::Autolock _l(mLock); 5469 for (size_t i = 0; i < mTracks.size(); i++) { 5470 sp<RecordTrack> track = mTracks[i]; 5471 track->invalidate(); 5472 } 5473 mActiveTracks.clear(); 5474 mActiveTracksGen++; 5475 mStartStopCond.broadcast(); 5476 } 5477 5478 releaseWakeLock(); 5479 5480 ALOGV("RecordThread %p exiting", this); 5481 return false; 5482} 5483 5484void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5485{ 5486 if (!mStandby) { 5487 inputStandBy(); 5488 mStandby = true; 5489 } 5490} 5491 5492void AudioFlinger::RecordThread::inputStandBy() 5493{ 5494 // Idle the fast capture if it's currently running 5495 if (mFastCapture != 0) { 5496 FastCaptureStateQueue *sq = mFastCapture->sq(); 5497 FastCaptureState *state = sq->begin(); 5498 if (!(state->mCommand & FastCaptureState::IDLE)) { 5499 state->mCommand = FastCaptureState::COLD_IDLE; 5500 state->mColdFutexAddr = &mFastCaptureFutex; 5501 state->mColdGen++; 5502 mFastCaptureFutex = 0; 5503 sq->end(); 5504 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5505 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5506#if 0 5507 if (kUseFastCapture == FastCapture_Dynamic) { 5508 // FIXME 5509 } 5510#endif 5511#ifdef AUDIO_WATCHDOG 5512 // FIXME 5513#endif 5514 } else { 5515 sq->end(false /*didModify*/); 5516 } 5517 } 5518 mInput->stream->common.standby(&mInput->stream->common); 5519} 5520 5521// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5522sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5523 const sp<AudioFlinger::Client>& client, 5524 uint32_t sampleRate, 5525 audio_format_t format, 5526 audio_channel_mask_t channelMask, 5527 size_t *pFrameCount, 5528 int sessionId, 5529 size_t *notificationFrames, 5530 int uid, 5531 IAudioFlinger::track_flags_t *flags, 5532 pid_t tid, 5533 status_t *status) 5534{ 5535 size_t frameCount = *pFrameCount; 5536 sp<RecordTrack> track; 5537 status_t lStatus; 5538 5539 // client expresses a preference for FAST, but we get the final say 5540 if (*flags & IAudioFlinger::TRACK_FAST) { 5541 if ( 5542 // use case: callback handler 5543 (tid != -1) && 5544 // frame count is not specified, or is exactly the pipe depth 5545 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5546 // PCM data 5547 audio_is_linear_pcm(format) && 5548 // native format 5549 (format == mFormat) && 5550 // native channel mask 5551 (channelMask == mChannelMask) && 5552 // native hardware sample rate 5553 (sampleRate == mSampleRate) && 5554 // record thread has an associated fast capture 5555 hasFastCapture() && 5556 // there are sufficient fast track slots available 5557 mFastTrackAvail 5558 ) { 5559 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5560 frameCount, mFrameCount); 5561 } else { 5562 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5563 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5564 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5565 frameCount, mFrameCount, mPipeFramesP2, 5566 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5567 hasFastCapture(), tid, mFastTrackAvail); 5568 *flags &= ~IAudioFlinger::TRACK_FAST; 5569 } 5570 } 5571 5572 // compute track buffer size in frames, and suggest the notification frame count 5573 if (*flags & IAudioFlinger::TRACK_FAST) { 5574 // fast track: frame count is exactly the pipe depth 5575 frameCount = mPipeFramesP2; 5576 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5577 *notificationFrames = mFrameCount; 5578 } else { 5579 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5580 // or 20 ms if there is a fast capture 5581 // TODO This could be a roundupRatio inline, and const 5582 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5583 * sampleRate + mSampleRate - 1) / mSampleRate; 5584 // minimum number of notification periods is at least kMinNotifications, 5585 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5586 static const size_t kMinNotifications = 3; 5587 static const uint32_t kMinMs = 30; 5588 // TODO This could be a roundupRatio inline 5589 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5590 // TODO This could be a roundupRatio inline 5591 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5592 maxNotificationFrames; 5593 const size_t minFrameCount = maxNotificationFrames * 5594 max(kMinNotifications, minNotificationsByMs); 5595 frameCount = max(frameCount, minFrameCount); 5596 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5597 *notificationFrames = maxNotificationFrames; 5598 } 5599 } 5600 *pFrameCount = frameCount; 5601 5602 lStatus = initCheck(); 5603 if (lStatus != NO_ERROR) { 5604 ALOGE("createRecordTrack_l() audio driver not initialized"); 5605 goto Exit; 5606 } 5607 5608 { // scope for mLock 5609 Mutex::Autolock _l(mLock); 5610 5611 track = new RecordTrack(this, client, sampleRate, 5612 format, channelMask, frameCount, NULL, sessionId, uid, 5613 *flags, TrackBase::TYPE_DEFAULT); 5614 5615 lStatus = track->initCheck(); 5616 if (lStatus != NO_ERROR) { 5617 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5618 // track must be cleared from the caller as the caller has the AF lock 5619 goto Exit; 5620 } 5621 mTracks.add(track); 5622 5623 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5624 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5625 mAudioFlinger->btNrecIsOff(); 5626 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5627 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5628 5629 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5630 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5631 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5632 // so ask activity manager to do this on our behalf 5633 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5634 } 5635 } 5636 5637 lStatus = NO_ERROR; 5638 5639Exit: 5640 *status = lStatus; 5641 return track; 5642} 5643 5644status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5645 AudioSystem::sync_event_t event, 5646 int triggerSession) 5647{ 5648 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5649 sp<ThreadBase> strongMe = this; 5650 status_t status = NO_ERROR; 5651 5652 if (event == AudioSystem::SYNC_EVENT_NONE) { 5653 recordTrack->clearSyncStartEvent(); 5654 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5655 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5656 triggerSession, 5657 recordTrack->sessionId(), 5658 syncStartEventCallback, 5659 recordTrack); 5660 // Sync event can be cancelled by the trigger session if the track is not in a 5661 // compatible state in which case we start record immediately 5662 if (recordTrack->mSyncStartEvent->isCancelled()) { 5663 recordTrack->clearSyncStartEvent(); 5664 } else { 5665 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5666 recordTrack->mFramesToDrop = - 5667 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5668 } 5669 } 5670 5671 { 5672 // This section is a rendezvous between binder thread executing start() and RecordThread 5673 AutoMutex lock(mLock); 5674 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5675 if (recordTrack->mState == TrackBase::PAUSING) { 5676 ALOGV("active record track PAUSING -> ACTIVE"); 5677 recordTrack->mState = TrackBase::ACTIVE; 5678 } else { 5679 ALOGV("active record track state %d", recordTrack->mState); 5680 } 5681 return status; 5682 } 5683 5684 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5685 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5686 // or using a separate command thread 5687 recordTrack->mState = TrackBase::STARTING_1; 5688 mActiveTracks.add(recordTrack); 5689 mActiveTracksGen++; 5690 status_t status = NO_ERROR; 5691 if (recordTrack->isExternalTrack()) { 5692 mLock.unlock(); 5693 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5694 mLock.lock(); 5695 // FIXME should verify that recordTrack is still in mActiveTracks 5696 if (status != NO_ERROR) { 5697 mActiveTracks.remove(recordTrack); 5698 mActiveTracksGen++; 5699 recordTrack->clearSyncStartEvent(); 5700 ALOGV("RecordThread::start error %d", status); 5701 return status; 5702 } 5703 } 5704 // Catch up with current buffer indices if thread is already running. 5705 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5706 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5707 // see previously buffered data before it called start(), but with greater risk of overrun. 5708 5709 recordTrack->mRsmpInFront = mRsmpInRear; 5710 recordTrack->mRsmpInUnrel = 0; 5711 // FIXME why reset? 5712 if (recordTrack->mResampler != NULL) { 5713 recordTrack->mResampler->reset(); 5714 } 5715 recordTrack->mState = TrackBase::STARTING_2; 5716 // signal thread to start 5717 mWaitWorkCV.broadcast(); 5718 if (mActiveTracks.indexOf(recordTrack) < 0) { 5719 ALOGV("Record failed to start"); 5720 status = BAD_VALUE; 5721 goto startError; 5722 } 5723 return status; 5724 } 5725 5726startError: 5727 if (recordTrack->isExternalTrack()) { 5728 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5729 } 5730 recordTrack->clearSyncStartEvent(); 5731 // FIXME I wonder why we do not reset the state here? 5732 return status; 5733} 5734 5735void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5736{ 5737 sp<SyncEvent> strongEvent = event.promote(); 5738 5739 if (strongEvent != 0) { 5740 sp<RefBase> ptr = strongEvent->cookie().promote(); 5741 if (ptr != 0) { 5742 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5743 recordTrack->handleSyncStartEvent(strongEvent); 5744 } 5745 } 5746} 5747 5748bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5749 ALOGV("RecordThread::stop"); 5750 AutoMutex _l(mLock); 5751 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5752 return false; 5753 } 5754 // note that threadLoop may still be processing the track at this point [without lock] 5755 recordTrack->mState = TrackBase::PAUSING; 5756 // do not wait for mStartStopCond if exiting 5757 if (exitPending()) { 5758 return true; 5759 } 5760 // FIXME incorrect usage of wait: no explicit predicate or loop 5761 mStartStopCond.wait(mLock); 5762 // if we have been restarted, recordTrack is in mActiveTracks here 5763 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5764 ALOGV("Record stopped OK"); 5765 return true; 5766 } 5767 return false; 5768} 5769 5770bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5771{ 5772 return false; 5773} 5774 5775status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5776{ 5777#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5778 if (!isValidSyncEvent(event)) { 5779 return BAD_VALUE; 5780 } 5781 5782 int eventSession = event->triggerSession(); 5783 status_t ret = NAME_NOT_FOUND; 5784 5785 Mutex::Autolock _l(mLock); 5786 5787 for (size_t i = 0; i < mTracks.size(); i++) { 5788 sp<RecordTrack> track = mTracks[i]; 5789 if (eventSession == track->sessionId()) { 5790 (void) track->setSyncEvent(event); 5791 ret = NO_ERROR; 5792 } 5793 } 5794 return ret; 5795#else 5796 return BAD_VALUE; 5797#endif 5798} 5799 5800// destroyTrack_l() must be called with ThreadBase::mLock held 5801void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5802{ 5803 track->terminate(); 5804 track->mState = TrackBase::STOPPED; 5805 // active tracks are removed by threadLoop() 5806 if (mActiveTracks.indexOf(track) < 0) { 5807 removeTrack_l(track); 5808 } 5809} 5810 5811void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5812{ 5813 mTracks.remove(track); 5814 // need anything related to effects here? 5815 if (track->isFastTrack()) { 5816 ALOG_ASSERT(!mFastTrackAvail); 5817 mFastTrackAvail = true; 5818 } 5819} 5820 5821void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5822{ 5823 dumpInternals(fd, args); 5824 dumpTracks(fd, args); 5825 dumpEffectChains(fd, args); 5826} 5827 5828void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5829{ 5830 dprintf(fd, "\nInput thread %p:\n", this); 5831 5832 if (mActiveTracks.size() > 0) { 5833 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5834 } else { 5835 dprintf(fd, " No active record clients\n"); 5836 } 5837 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5838 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5839 5840 dumpBase(fd, args); 5841} 5842 5843void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5844{ 5845 const size_t SIZE = 256; 5846 char buffer[SIZE]; 5847 String8 result; 5848 5849 size_t numtracks = mTracks.size(); 5850 size_t numactive = mActiveTracks.size(); 5851 size_t numactiveseen = 0; 5852 dprintf(fd, " %d Tracks", numtracks); 5853 if (numtracks) { 5854 dprintf(fd, " of which %d are active\n", numactive); 5855 RecordTrack::appendDumpHeader(result); 5856 for (size_t i = 0; i < numtracks ; ++i) { 5857 sp<RecordTrack> track = mTracks[i]; 5858 if (track != 0) { 5859 bool active = mActiveTracks.indexOf(track) >= 0; 5860 if (active) { 5861 numactiveseen++; 5862 } 5863 track->dump(buffer, SIZE, active); 5864 result.append(buffer); 5865 } 5866 } 5867 } else { 5868 dprintf(fd, "\n"); 5869 } 5870 5871 if (numactiveseen != numactive) { 5872 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5873 " not in the track list\n"); 5874 result.append(buffer); 5875 RecordTrack::appendDumpHeader(result); 5876 for (size_t i = 0; i < numactive; ++i) { 5877 sp<RecordTrack> track = mActiveTracks[i]; 5878 if (mTracks.indexOf(track) < 0) { 5879 track->dump(buffer, SIZE, true); 5880 result.append(buffer); 5881 } 5882 } 5883 5884 } 5885 write(fd, result.string(), result.size()); 5886} 5887 5888// AudioBufferProvider interface 5889status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5890 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5891{ 5892 RecordTrack *activeTrack = mRecordTrack; 5893 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5894 if (threadBase == 0) { 5895 buffer->frameCount = 0; 5896 buffer->raw = NULL; 5897 return NOT_ENOUGH_DATA; 5898 } 5899 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5900 int32_t rear = recordThread->mRsmpInRear; 5901 int32_t front = activeTrack->mRsmpInFront; 5902 ssize_t filled = rear - front; 5903 // FIXME should not be P2 (don't want to increase latency) 5904 // FIXME if client not keeping up, discard 5905 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5906 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5907 front &= recordThread->mRsmpInFramesP2 - 1; 5908 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5909 if (part1 > (size_t) filled) { 5910 part1 = filled; 5911 } 5912 size_t ask = buffer->frameCount; 5913 ALOG_ASSERT(ask > 0); 5914 if (part1 > ask) { 5915 part1 = ask; 5916 } 5917 if (part1 == 0) { 5918 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5919 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5920 buffer->raw = NULL; 5921 buffer->frameCount = 0; 5922 activeTrack->mRsmpInUnrel = 0; 5923 return NOT_ENOUGH_DATA; 5924 } 5925 5926 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5927 buffer->frameCount = part1; 5928 activeTrack->mRsmpInUnrel = part1; 5929 return NO_ERROR; 5930} 5931 5932// AudioBufferProvider interface 5933void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5934 AudioBufferProvider::Buffer* buffer) 5935{ 5936 RecordTrack *activeTrack = mRecordTrack; 5937 size_t stepCount = buffer->frameCount; 5938 if (stepCount == 0) { 5939 return; 5940 } 5941 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5942 activeTrack->mRsmpInUnrel -= stepCount; 5943 activeTrack->mRsmpInFront += stepCount; 5944 buffer->raw = NULL; 5945 buffer->frameCount = 0; 5946} 5947 5948bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5949 status_t& status) 5950{ 5951 bool reconfig = false; 5952 5953 status = NO_ERROR; 5954 5955 audio_format_t reqFormat = mFormat; 5956 uint32_t samplingRate = mSampleRate; 5957 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5958 5959 AudioParameter param = AudioParameter(keyValuePair); 5960 int value; 5961 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5962 // channel count change can be requested. Do we mandate the first client defines the 5963 // HAL sampling rate and channel count or do we allow changes on the fly? 5964 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5965 samplingRate = value; 5966 reconfig = true; 5967 } 5968 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5969 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5970 status = BAD_VALUE; 5971 } else { 5972 reqFormat = (audio_format_t) value; 5973 reconfig = true; 5974 } 5975 } 5976 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5977 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5978 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5979 status = BAD_VALUE; 5980 } else { 5981 channelMask = mask; 5982 reconfig = true; 5983 } 5984 } 5985 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5986 // do not accept frame count changes if tracks are open as the track buffer 5987 // size depends on frame count and correct behavior would not be guaranteed 5988 // if frame count is changed after track creation 5989 if (mActiveTracks.size() > 0) { 5990 status = INVALID_OPERATION; 5991 } else { 5992 reconfig = true; 5993 } 5994 } 5995 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5996 // forward device change to effects that have requested to be 5997 // aware of attached audio device. 5998 for (size_t i = 0; i < mEffectChains.size(); i++) { 5999 mEffectChains[i]->setDevice_l(value); 6000 } 6001 6002 // store input device and output device but do not forward output device to audio HAL. 6003 // Note that status is ignored by the caller for output device 6004 // (see AudioFlinger::setParameters() 6005 if (audio_is_output_devices(value)) { 6006 mOutDevice = value; 6007 status = BAD_VALUE; 6008 } else { 6009 mInDevice = value; 6010 // disable AEC and NS if the device is a BT SCO headset supporting those 6011 // pre processings 6012 if (mTracks.size() > 0) { 6013 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6014 mAudioFlinger->btNrecIsOff(); 6015 for (size_t i = 0; i < mTracks.size(); i++) { 6016 sp<RecordTrack> track = mTracks[i]; 6017 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6018 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6019 } 6020 } 6021 } 6022 } 6023 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6024 mAudioSource != (audio_source_t)value) { 6025 // forward device change to effects that have requested to be 6026 // aware of attached audio device. 6027 for (size_t i = 0; i < mEffectChains.size(); i++) { 6028 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6029 } 6030 mAudioSource = (audio_source_t)value; 6031 } 6032 6033 if (status == NO_ERROR) { 6034 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6035 keyValuePair.string()); 6036 if (status == INVALID_OPERATION) { 6037 inputStandBy(); 6038 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6039 keyValuePair.string()); 6040 } 6041 if (reconfig) { 6042 if (status == BAD_VALUE && 6043 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6044 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6045 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6046 <= (2 * samplingRate)) && 6047 audio_channel_count_from_in_mask( 6048 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6049 (channelMask == AUDIO_CHANNEL_IN_MONO || 6050 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6051 status = NO_ERROR; 6052 } 6053 if (status == NO_ERROR) { 6054 readInputParameters_l(); 6055 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6056 } 6057 } 6058 } 6059 6060 return reconfig; 6061} 6062 6063String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6064{ 6065 Mutex::Autolock _l(mLock); 6066 if (initCheck() != NO_ERROR) { 6067 return String8(); 6068 } 6069 6070 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6071 const String8 out_s8(s); 6072 free(s); 6073 return out_s8; 6074} 6075 6076void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6077 AudioSystem::OutputDescriptor desc; 6078 const void *param2 = NULL; 6079 6080 switch (event) { 6081 case AudioSystem::INPUT_OPENED: 6082 case AudioSystem::INPUT_CONFIG_CHANGED: 6083 desc.channelMask = mChannelMask; 6084 desc.samplingRate = mSampleRate; 6085 desc.format = mFormat; 6086 desc.frameCount = mFrameCount; 6087 desc.latency = 0; 6088 param2 = &desc; 6089 break; 6090 6091 case AudioSystem::INPUT_CLOSED: 6092 default: 6093 break; 6094 } 6095 mAudioFlinger->audioConfigChanged(event, mId, param2); 6096} 6097 6098void AudioFlinger::RecordThread::readInputParameters_l() 6099{ 6100 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6101 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6102 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6103 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6104 mFormat = mHALFormat; 6105 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6106 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6107 } 6108 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6109 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6110 mFrameCount = mBufferSize / mFrameSize; 6111 // This is the formula for calculating the temporary buffer size. 6112 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6113 // 1 full output buffer, regardless of the alignment of the available input. 6114 // The value is somewhat arbitrary, and could probably be even larger. 6115 // A larger value should allow more old data to be read after a track calls start(), 6116 // without increasing latency. 6117 mRsmpInFrames = mFrameCount * 7; 6118 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6119 delete[] mRsmpInBuffer; 6120 6121 // TODO optimize audio capture buffer sizes ... 6122 // Here we calculate the size of the sliding buffer used as a source 6123 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6124 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6125 // be better to have it derived from the pipe depth in the long term. 6126 // The current value is higher than necessary. However it should not add to latency. 6127 6128 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6129 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6130 6131 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6132 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6133} 6134 6135uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6136{ 6137 Mutex::Autolock _l(mLock); 6138 if (initCheck() != NO_ERROR) { 6139 return 0; 6140 } 6141 6142 return mInput->stream->get_input_frames_lost(mInput->stream); 6143} 6144 6145uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6146{ 6147 Mutex::Autolock _l(mLock); 6148 uint32_t result = 0; 6149 if (getEffectChain_l(sessionId) != 0) { 6150 result = EFFECT_SESSION; 6151 } 6152 6153 for (size_t i = 0; i < mTracks.size(); ++i) { 6154 if (sessionId == mTracks[i]->sessionId()) { 6155 result |= TRACK_SESSION; 6156 break; 6157 } 6158 } 6159 6160 return result; 6161} 6162 6163KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6164{ 6165 KeyedVector<int, bool> ids; 6166 Mutex::Autolock _l(mLock); 6167 for (size_t j = 0; j < mTracks.size(); ++j) { 6168 sp<RecordThread::RecordTrack> track = mTracks[j]; 6169 int sessionId = track->sessionId(); 6170 if (ids.indexOfKey(sessionId) < 0) { 6171 ids.add(sessionId, true); 6172 } 6173 } 6174 return ids; 6175} 6176 6177AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6178{ 6179 Mutex::Autolock _l(mLock); 6180 AudioStreamIn *input = mInput; 6181 mInput = NULL; 6182 return input; 6183} 6184 6185// this method must always be called either with ThreadBase mLock held or inside the thread loop 6186audio_stream_t* AudioFlinger::RecordThread::stream() const 6187{ 6188 if (mInput == NULL) { 6189 return NULL; 6190 } 6191 return &mInput->stream->common; 6192} 6193 6194status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6195{ 6196 // only one chain per input thread 6197 if (mEffectChains.size() != 0) { 6198 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6199 return INVALID_OPERATION; 6200 } 6201 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6202 chain->setThread(this); 6203 chain->setInBuffer(NULL); 6204 chain->setOutBuffer(NULL); 6205 6206 checkSuspendOnAddEffectChain_l(chain); 6207 6208 mEffectChains.add(chain); 6209 6210 return NO_ERROR; 6211} 6212 6213size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6214{ 6215 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6216 ALOGW_IF(mEffectChains.size() != 1, 6217 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6218 chain.get(), mEffectChains.size(), this); 6219 if (mEffectChains.size() == 1) { 6220 mEffectChains.removeAt(0); 6221 } 6222 return 0; 6223} 6224 6225status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6226 audio_patch_handle_t *handle) 6227{ 6228 status_t status = NO_ERROR; 6229 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6230 // store new device and send to effects 6231 mInDevice = patch->sources[0].ext.device.type; 6232 for (size_t i = 0; i < mEffectChains.size(); i++) { 6233 mEffectChains[i]->setDevice_l(mInDevice); 6234 } 6235 6236 // disable AEC and NS if the device is a BT SCO headset supporting those 6237 // pre processings 6238 if (mTracks.size() > 0) { 6239 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6240 mAudioFlinger->btNrecIsOff(); 6241 for (size_t i = 0; i < mTracks.size(); i++) { 6242 sp<RecordTrack> track = mTracks[i]; 6243 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6244 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6245 } 6246 } 6247 6248 // store new source and send to effects 6249 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6250 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6251 for (size_t i = 0; i < mEffectChains.size(); i++) { 6252 mEffectChains[i]->setAudioSource_l(mAudioSource); 6253 } 6254 } 6255 6256 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6257 status = hwDevice->create_audio_patch(hwDevice, 6258 patch->num_sources, 6259 patch->sources, 6260 patch->num_sinks, 6261 patch->sinks, 6262 handle); 6263 } else { 6264 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6265 } 6266 return status; 6267} 6268 6269status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6270{ 6271 status_t status = NO_ERROR; 6272 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6273 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6274 status = hwDevice->release_audio_patch(hwDevice, handle); 6275 } else { 6276 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6277 } 6278 return status; 6279} 6280 6281void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6282{ 6283 Mutex::Autolock _l(mLock); 6284 mTracks.add(record); 6285} 6286 6287void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6288{ 6289 Mutex::Autolock _l(mLock); 6290 destroyTrack_l(record); 6291} 6292 6293void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6294{ 6295 ThreadBase::getAudioPortConfig(config); 6296 config->role = AUDIO_PORT_ROLE_SINK; 6297 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6298 config->ext.mix.usecase.source = mAudioSource; 6299} 6300 6301}; // namespace android 6302