Threads.cpp revision cd04484f4837b8ca0041d118286ab6a98e84fc75
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 87namespace android { 88 89// retry counts for buffer fill timeout 90// 50 * ~20msecs = 1 second 91static const int8_t kMaxTrackRetries = 50; 92static const int8_t kMaxTrackStartupRetries = 50; 93// allow less retry attempts on direct output thread. 94// direct outputs can be a scarce resource in audio hardware and should 95// be released as quickly as possible. 96static const int8_t kMaxTrackRetriesDirect = 2; 97 98// don't warn about blocked writes or record buffer overflows more often than this 99static const nsecs_t kWarningThrottleNs = seconds(5); 100 101// RecordThread loop sleep time upon application overrun or audio HAL read error 102static const int kRecordThreadSleepUs = 5000; 103 104// maximum time to wait in sendConfigEvent_l() for a status to be received 105static const nsecs_t kConfigEventTimeoutNs = seconds(2); 106 107// minimum sleep time for the mixer thread loop when tracks are active but in underrun 108static const uint32_t kMinThreadSleepTimeUs = 5000; 109// maximum divider applied to the active sleep time in the mixer thread loop 110static const uint32_t kMaxThreadSleepTimeShift = 2; 111 112// minimum normal sink buffer size, expressed in milliseconds rather than frames 113static const uint32_t kMinNormalSinkBufferSizeMs = 20; 114// maximum normal sink buffer size 115static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 116 117// Offloaded output thread standby delay: allows track transition without going to standby 118static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 119 120// Whether to use fast mixer 121static const enum { 122 FastMixer_Never, // never initialize or use: for debugging only 123 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 124 // normal mixer multiplier is 1 125 FastMixer_Static, // initialize if needed, then use all the time if initialized, 126 // multiplier is calculated based on min & max normal mixer buffer size 127 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 128 // multiplier is calculated based on min & max normal mixer buffer size 129 // FIXME for FastMixer_Dynamic: 130 // Supporting this option will require fixing HALs that can't handle large writes. 131 // For example, one HAL implementation returns an error from a large write, 132 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 133 // We could either fix the HAL implementations, or provide a wrapper that breaks 134 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 135} kUseFastMixer = FastMixer_Static; 136 137// Whether to use fast capture 138static const enum { 139 FastCapture_Never, // never initialize or use: for debugging only 140 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 141 FastCapture_Static, // initialize if needed, then use all the time if initialized 142} kUseFastCapture = FastCapture_Static; 143 144// Priorities for requestPriority 145static const int kPriorityAudioApp = 2; 146static const int kPriorityFastMixer = 3; 147static const int kPriorityFastCapture = 3; 148 149// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 150// for the track. The client then sub-divides this into smaller buffers for its use. 151// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 152// So for now we just assume that client is double-buffered for fast tracks. 153// FIXME It would be better for client to tell AudioFlinger the value of N, 154// so AudioFlinger could allocate the right amount of memory. 155// See the client's minBufCount and mNotificationFramesAct calculations for details. 156 157// This is the default value, if not specified by property. 158static const int kFastTrackMultiplier = 2; 159 160// The minimum and maximum allowed values 161static const int kFastTrackMultiplierMin = 1; 162static const int kFastTrackMultiplierMax = 2; 163 164// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 165static int sFastTrackMultiplier = kFastTrackMultiplier; 166 167// See Thread::readOnlyHeap(). 168// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 169// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 170// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 171static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 172 173// ---------------------------------------------------------------------------- 174 175static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 176 177static void sFastTrackMultiplierInit() 178{ 179 char value[PROPERTY_VALUE_MAX]; 180 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 181 char *endptr; 182 unsigned long ul = strtoul(value, &endptr, 0); 183 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 184 sFastTrackMultiplier = (int) ul; 185 } 186 } 187} 188 189// ---------------------------------------------------------------------------- 190 191#ifdef ADD_BATTERY_DATA 192// To collect the amplifier usage 193static void addBatteryData(uint32_t params) { 194 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 195 if (service == NULL) { 196 // it already logged 197 return; 198 } 199 200 service->addBatteryData(params); 201} 202#endif 203 204 205// ---------------------------------------------------------------------------- 206// CPU Stats 207// ---------------------------------------------------------------------------- 208 209class CpuStats { 210public: 211 CpuStats(); 212 void sample(const String8 &title); 213#ifdef DEBUG_CPU_USAGE 214private: 215 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 216 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 217 218 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 219 220 int mCpuNum; // thread's current CPU number 221 int mCpukHz; // frequency of thread's current CPU in kHz 222#endif 223}; 224 225CpuStats::CpuStats() 226#ifdef DEBUG_CPU_USAGE 227 : mCpuNum(-1), mCpukHz(-1) 228#endif 229{ 230} 231 232void CpuStats::sample(const String8 &title 233#ifndef DEBUG_CPU_USAGE 234 __unused 235#endif 236 ) { 237#ifdef DEBUG_CPU_USAGE 238 // get current thread's delta CPU time in wall clock ns 239 double wcNs; 240 bool valid = mCpuUsage.sampleAndEnable(wcNs); 241 242 // record sample for wall clock statistics 243 if (valid) { 244 mWcStats.sample(wcNs); 245 } 246 247 // get the current CPU number 248 int cpuNum = sched_getcpu(); 249 250 // get the current CPU frequency in kHz 251 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 252 253 // check if either CPU number or frequency changed 254 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 255 mCpuNum = cpuNum; 256 mCpukHz = cpukHz; 257 // ignore sample for purposes of cycles 258 valid = false; 259 } 260 261 // if no change in CPU number or frequency, then record sample for cycle statistics 262 if (valid && mCpukHz > 0) { 263 double cycles = wcNs * cpukHz * 0.000001; 264 mHzStats.sample(cycles); 265 } 266 267 unsigned n = mWcStats.n(); 268 // mCpuUsage.elapsed() is expensive, so don't call it every loop 269 if ((n & 127) == 1) { 270 long long elapsed = mCpuUsage.elapsed(); 271 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 272 double perLoop = elapsed / (double) n; 273 double perLoop100 = perLoop * 0.01; 274 double perLoop1k = perLoop * 0.001; 275 double mean = mWcStats.mean(); 276 double stddev = mWcStats.stddev(); 277 double minimum = mWcStats.minimum(); 278 double maximum = mWcStats.maximum(); 279 double meanCycles = mHzStats.mean(); 280 double stddevCycles = mHzStats.stddev(); 281 double minCycles = mHzStats.minimum(); 282 double maxCycles = mHzStats.maximum(); 283 mCpuUsage.resetElapsed(); 284 mWcStats.reset(); 285 mHzStats.reset(); 286 ALOGD("CPU usage for %s over past %.1f secs\n" 287 " (%u mixer loops at %.1f mean ms per loop):\n" 288 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 289 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 290 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 291 title.string(), 292 elapsed * .000000001, n, perLoop * .000001, 293 mean * .001, 294 stddev * .001, 295 minimum * .001, 296 maximum * .001, 297 mean / perLoop100, 298 stddev / perLoop100, 299 minimum / perLoop100, 300 maximum / perLoop100, 301 meanCycles / perLoop1k, 302 stddevCycles / perLoop1k, 303 minCycles / perLoop1k, 304 maxCycles / perLoop1k); 305 306 } 307 } 308#endif 309}; 310 311// ---------------------------------------------------------------------------- 312// ThreadBase 313// ---------------------------------------------------------------------------- 314 315AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 316 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 317 : Thread(false /*canCallJava*/), 318 mType(type), 319 mAudioFlinger(audioFlinger), 320 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 321 // are set by PlaybackThread::readOutputParameters_l() or 322 // RecordThread::readInputParameters_l() 323 //FIXME: mStandby should be true here. Is this some kind of hack? 324 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 325 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 326 // mName will be set by concrete (non-virtual) subclass 327 mDeathRecipient(new PMDeathRecipient(this)) 328{ 329} 330 331AudioFlinger::ThreadBase::~ThreadBase() 332{ 333 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 334 mConfigEvents.clear(); 335 336 // do not lock the mutex in destructor 337 releaseWakeLock_l(); 338 if (mPowerManager != 0) { 339 sp<IBinder> binder = mPowerManager->asBinder(); 340 binder->unlinkToDeath(mDeathRecipient); 341 } 342} 343 344status_t AudioFlinger::ThreadBase::readyToRun() 345{ 346 status_t status = initCheck(); 347 if (status == NO_ERROR) { 348 ALOGI("AudioFlinger's thread %p ready to run", this); 349 } else { 350 ALOGE("No working audio driver found."); 351 } 352 return status; 353} 354 355void AudioFlinger::ThreadBase::exit() 356{ 357 ALOGV("ThreadBase::exit"); 358 // do any cleanup required for exit to succeed 359 preExit(); 360 { 361 // This lock prevents the following race in thread (uniprocessor for illustration): 362 // if (!exitPending()) { 363 // // context switch from here to exit() 364 // // exit() calls requestExit(), what exitPending() observes 365 // // exit() calls signal(), which is dropped since no waiters 366 // // context switch back from exit() to here 367 // mWaitWorkCV.wait(...); 368 // // now thread is hung 369 // } 370 AutoMutex lock(mLock); 371 requestExit(); 372 mWaitWorkCV.broadcast(); 373 } 374 // When Thread::requestExitAndWait is made virtual and this method is renamed to 375 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 376 requestExitAndWait(); 377} 378 379status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 380{ 381 status_t status; 382 383 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 384 Mutex::Autolock _l(mLock); 385 386 return sendSetParameterConfigEvent_l(keyValuePairs); 387} 388 389// sendConfigEvent_l() must be called with ThreadBase::mLock held 390// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 391status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 392{ 393 status_t status = NO_ERROR; 394 395 mConfigEvents.add(event); 396 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 397 mWaitWorkCV.signal(); 398 mLock.unlock(); 399 { 400 Mutex::Autolock _l(event->mLock); 401 while (event->mWaitStatus) { 402 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 403 event->mStatus = TIMED_OUT; 404 event->mWaitStatus = false; 405 } 406 } 407 status = event->mStatus; 408 } 409 mLock.lock(); 410 return status; 411} 412 413void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 414{ 415 Mutex::Autolock _l(mLock); 416 sendIoConfigEvent_l(event, param); 417} 418 419// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 420void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 421{ 422 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 423 sendConfigEvent_l(configEvent); 424} 425 426// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 427void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 428{ 429 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 430 sendConfigEvent_l(configEvent); 431} 432 433// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 434status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 435{ 436 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 437 return sendConfigEvent_l(configEvent); 438} 439 440status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 441 const struct audio_patch *patch, 442 audio_patch_handle_t *handle) 443{ 444 Mutex::Autolock _l(mLock); 445 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 446 status_t status = sendConfigEvent_l(configEvent); 447 if (status == NO_ERROR) { 448 CreateAudioPatchConfigEventData *data = 449 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 450 *handle = data->mHandle; 451 } 452 return status; 453} 454 455status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 456 const audio_patch_handle_t handle) 457{ 458 Mutex::Autolock _l(mLock); 459 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 460 return sendConfigEvent_l(configEvent); 461} 462 463 464// post condition: mConfigEvents.isEmpty() 465void AudioFlinger::ThreadBase::processConfigEvents_l() 466{ 467 bool configChanged = false; 468 469 while (!mConfigEvents.isEmpty()) { 470 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 471 sp<ConfigEvent> event = mConfigEvents[0]; 472 mConfigEvents.removeAt(0); 473 switch (event->mType) { 474 case CFG_EVENT_PRIO: { 475 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 476 // FIXME Need to understand why this has to be done asynchronously 477 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 478 true /*asynchronous*/); 479 if (err != 0) { 480 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 481 data->mPrio, data->mPid, data->mTid, err); 482 } 483 } break; 484 case CFG_EVENT_IO: { 485 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 486 audioConfigChanged(data->mEvent, data->mParam); 487 } break; 488 case CFG_EVENT_SET_PARAMETER: { 489 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 490 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 491 configChanged = true; 492 } 493 } break; 494 case CFG_EVENT_CREATE_AUDIO_PATCH: { 495 CreateAudioPatchConfigEventData *data = 496 (CreateAudioPatchConfigEventData *)event->mData.get(); 497 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 498 } break; 499 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 500 ReleaseAudioPatchConfigEventData *data = 501 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 502 event->mStatus = releaseAudioPatch_l(data->mHandle); 503 } break; 504 default: 505 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 506 break; 507 } 508 { 509 Mutex::Autolock _l(event->mLock); 510 if (event->mWaitStatus) { 511 event->mWaitStatus = false; 512 event->mCond.signal(); 513 } 514 } 515 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 516 } 517 518 if (configChanged) { 519 cacheParameters_l(); 520 } 521} 522 523String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 524 String8 s; 525 if (output) { 526 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 527 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 528 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 529 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 530 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 531 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 532 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 533 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 534 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 535 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 536 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 537 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 538 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 539 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 540 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 541 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 542 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 543 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 544 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 545 } else { 546 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 547 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 548 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 549 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 550 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 551 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 552 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 553 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 554 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 555 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 556 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 557 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 558 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 559 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 560 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 561 } 562 int len = s.length(); 563 if (s.length() > 2) { 564 char *str = s.lockBuffer(len); 565 s.unlockBuffer(len - 2); 566 } 567 return s; 568} 569 570void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 571{ 572 const size_t SIZE = 256; 573 char buffer[SIZE]; 574 String8 result; 575 576 bool locked = AudioFlinger::dumpTryLock(mLock); 577 if (!locked) { 578 dprintf(fd, "thread %p maybe dead locked\n", this); 579 } 580 581 dprintf(fd, " I/O handle: %d\n", mId); 582 dprintf(fd, " TID: %d\n", getTid()); 583 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 584 dprintf(fd, " Sample rate: %u\n", mSampleRate); 585 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 586 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 587 dprintf(fd, " Channel Count: %u\n", mChannelCount); 588 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 589 channelMaskToString(mChannelMask, mType != RECORD).string()); 590 dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 591 dprintf(fd, " Frame size: %zu\n", mFrameSize); 592 dprintf(fd, " Pending config events:"); 593 size_t numConfig = mConfigEvents.size(); 594 if (numConfig) { 595 for (size_t i = 0; i < numConfig; i++) { 596 mConfigEvents[i]->dump(buffer, SIZE); 597 dprintf(fd, "\n %s", buffer); 598 } 599 dprintf(fd, "\n"); 600 } else { 601 dprintf(fd, " none\n"); 602 } 603 604 if (locked) { 605 mLock.unlock(); 606 } 607} 608 609void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 610{ 611 const size_t SIZE = 256; 612 char buffer[SIZE]; 613 String8 result; 614 615 size_t numEffectChains = mEffectChains.size(); 616 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 617 write(fd, buffer, strlen(buffer)); 618 619 for (size_t i = 0; i < numEffectChains; ++i) { 620 sp<EffectChain> chain = mEffectChains[i]; 621 if (chain != 0) { 622 chain->dump(fd, args); 623 } 624 } 625} 626 627void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 628{ 629 Mutex::Autolock _l(mLock); 630 acquireWakeLock_l(uid); 631} 632 633String16 AudioFlinger::ThreadBase::getWakeLockTag() 634{ 635 switch (mType) { 636 case MIXER: 637 return String16("AudioMix"); 638 case DIRECT: 639 return String16("AudioDirectOut"); 640 case DUPLICATING: 641 return String16("AudioDup"); 642 case RECORD: 643 return String16("AudioIn"); 644 case OFFLOAD: 645 return String16("AudioOffload"); 646 default: 647 ALOG_ASSERT(false); 648 return String16("AudioUnknown"); 649 } 650} 651 652void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 653{ 654 getPowerManager_l(); 655 if (mPowerManager != 0) { 656 sp<IBinder> binder = new BBinder(); 657 status_t status; 658 if (uid >= 0) { 659 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 660 binder, 661 getWakeLockTag(), 662 String16("media"), 663 uid); 664 } else { 665 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 666 binder, 667 getWakeLockTag(), 668 String16("media")); 669 } 670 if (status == NO_ERROR) { 671 mWakeLockToken = binder; 672 } 673 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 674 } 675} 676 677void AudioFlinger::ThreadBase::releaseWakeLock() 678{ 679 Mutex::Autolock _l(mLock); 680 releaseWakeLock_l(); 681} 682 683void AudioFlinger::ThreadBase::releaseWakeLock_l() 684{ 685 if (mWakeLockToken != 0) { 686 ALOGV("releaseWakeLock_l() %s", mName); 687 if (mPowerManager != 0) { 688 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 689 } 690 mWakeLockToken.clear(); 691 } 692} 693 694void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 695 Mutex::Autolock _l(mLock); 696 updateWakeLockUids_l(uids); 697} 698 699void AudioFlinger::ThreadBase::getPowerManager_l() { 700 701 if (mPowerManager == 0) { 702 // use checkService() to avoid blocking if power service is not up yet 703 sp<IBinder> binder = 704 defaultServiceManager()->checkService(String16("power")); 705 if (binder == 0) { 706 ALOGW("Thread %s cannot connect to the power manager service", mName); 707 } else { 708 mPowerManager = interface_cast<IPowerManager>(binder); 709 binder->linkToDeath(mDeathRecipient); 710 } 711 } 712} 713 714void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 715 716 getPowerManager_l(); 717 if (mWakeLockToken == NULL) { 718 ALOGE("no wake lock to update!"); 719 return; 720 } 721 if (mPowerManager != 0) { 722 sp<IBinder> binder = new BBinder(); 723 status_t status; 724 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 725 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 726 } 727} 728 729void AudioFlinger::ThreadBase::clearPowerManager() 730{ 731 Mutex::Autolock _l(mLock); 732 releaseWakeLock_l(); 733 mPowerManager.clear(); 734} 735 736void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 737{ 738 sp<ThreadBase> thread = mThread.promote(); 739 if (thread != 0) { 740 thread->clearPowerManager(); 741 } 742 ALOGW("power manager service died !!!"); 743} 744 745void AudioFlinger::ThreadBase::setEffectSuspended( 746 const effect_uuid_t *type, bool suspend, int sessionId) 747{ 748 Mutex::Autolock _l(mLock); 749 setEffectSuspended_l(type, suspend, sessionId); 750} 751 752void AudioFlinger::ThreadBase::setEffectSuspended_l( 753 const effect_uuid_t *type, bool suspend, int sessionId) 754{ 755 sp<EffectChain> chain = getEffectChain_l(sessionId); 756 if (chain != 0) { 757 if (type != NULL) { 758 chain->setEffectSuspended_l(type, suspend); 759 } else { 760 chain->setEffectSuspendedAll_l(suspend); 761 } 762 } 763 764 updateSuspendedSessions_l(type, suspend, sessionId); 765} 766 767void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 768{ 769 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 770 if (index < 0) { 771 return; 772 } 773 774 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 775 mSuspendedSessions.valueAt(index); 776 777 for (size_t i = 0; i < sessionEffects.size(); i++) { 778 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 779 for (int j = 0; j < desc->mRefCount; j++) { 780 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 781 chain->setEffectSuspendedAll_l(true); 782 } else { 783 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 784 desc->mType.timeLow); 785 chain->setEffectSuspended_l(&desc->mType, true); 786 } 787 } 788 } 789} 790 791void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 792 bool suspend, 793 int sessionId) 794{ 795 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 796 797 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 798 799 if (suspend) { 800 if (index >= 0) { 801 sessionEffects = mSuspendedSessions.valueAt(index); 802 } else { 803 mSuspendedSessions.add(sessionId, sessionEffects); 804 } 805 } else { 806 if (index < 0) { 807 return; 808 } 809 sessionEffects = mSuspendedSessions.valueAt(index); 810 } 811 812 813 int key = EffectChain::kKeyForSuspendAll; 814 if (type != NULL) { 815 key = type->timeLow; 816 } 817 index = sessionEffects.indexOfKey(key); 818 819 sp<SuspendedSessionDesc> desc; 820 if (suspend) { 821 if (index >= 0) { 822 desc = sessionEffects.valueAt(index); 823 } else { 824 desc = new SuspendedSessionDesc(); 825 if (type != NULL) { 826 desc->mType = *type; 827 } 828 sessionEffects.add(key, desc); 829 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 830 } 831 desc->mRefCount++; 832 } else { 833 if (index < 0) { 834 return; 835 } 836 desc = sessionEffects.valueAt(index); 837 if (--desc->mRefCount == 0) { 838 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 839 sessionEffects.removeItemsAt(index); 840 if (sessionEffects.isEmpty()) { 841 ALOGV("updateSuspendedSessions_l() restore removing session %d", 842 sessionId); 843 mSuspendedSessions.removeItem(sessionId); 844 } 845 } 846 } 847 if (!sessionEffects.isEmpty()) { 848 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 849 } 850} 851 852void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 853 bool enabled, 854 int sessionId) 855{ 856 Mutex::Autolock _l(mLock); 857 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 858} 859 860void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 861 bool enabled, 862 int sessionId) 863{ 864 if (mType != RECORD) { 865 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 866 // another session. This gives the priority to well behaved effect control panels 867 // and applications not using global effects. 868 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 869 // global effects 870 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 871 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 872 } 873 } 874 875 sp<EffectChain> chain = getEffectChain_l(sessionId); 876 if (chain != 0) { 877 chain->checkSuspendOnEffectEnabled(effect, enabled); 878 } 879} 880 881// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 882sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 883 const sp<AudioFlinger::Client>& client, 884 const sp<IEffectClient>& effectClient, 885 int32_t priority, 886 int sessionId, 887 effect_descriptor_t *desc, 888 int *enabled, 889 status_t *status) 890{ 891 sp<EffectModule> effect; 892 sp<EffectHandle> handle; 893 status_t lStatus; 894 sp<EffectChain> chain; 895 bool chainCreated = false; 896 bool effectCreated = false; 897 bool effectRegistered = false; 898 899 lStatus = initCheck(); 900 if (lStatus != NO_ERROR) { 901 ALOGW("createEffect_l() Audio driver not initialized."); 902 goto Exit; 903 } 904 905 // Reject any effect on Direct output threads for now, since the format of 906 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 907 if (mType == DIRECT) { 908 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 909 desc->name, mName); 910 lStatus = BAD_VALUE; 911 goto Exit; 912 } 913 914 // Reject any effect on mixer or duplicating multichannel sinks. 915 // TODO: fix both format and multichannel issues with effects. 916 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 917 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 918 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 919 lStatus = BAD_VALUE; 920 goto Exit; 921 } 922 923 // Allow global effects only on offloaded and mixer threads 924 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 925 switch (mType) { 926 case MIXER: 927 case OFFLOAD: 928 break; 929 case DIRECT: 930 case DUPLICATING: 931 case RECORD: 932 default: 933 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 934 lStatus = BAD_VALUE; 935 goto Exit; 936 } 937 } 938 939 // Only Pre processor effects are allowed on input threads and only on input threads 940 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 941 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 942 desc->name, desc->flags, mType); 943 lStatus = BAD_VALUE; 944 goto Exit; 945 } 946 947 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 948 949 { // scope for mLock 950 Mutex::Autolock _l(mLock); 951 952 // check for existing effect chain with the requested audio session 953 chain = getEffectChain_l(sessionId); 954 if (chain == 0) { 955 // create a new chain for this session 956 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 957 chain = new EffectChain(this, sessionId); 958 addEffectChain_l(chain); 959 chain->setStrategy(getStrategyForSession_l(sessionId)); 960 chainCreated = true; 961 } else { 962 effect = chain->getEffectFromDesc_l(desc); 963 } 964 965 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 966 967 if (effect == 0) { 968 int id = mAudioFlinger->nextUniqueId(); 969 // Check CPU and memory usage 970 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 971 if (lStatus != NO_ERROR) { 972 goto Exit; 973 } 974 effectRegistered = true; 975 // create a new effect module if none present in the chain 976 effect = new EffectModule(this, chain, desc, id, sessionId); 977 lStatus = effect->status(); 978 if (lStatus != NO_ERROR) { 979 goto Exit; 980 } 981 effect->setOffloaded(mType == OFFLOAD, mId); 982 983 lStatus = chain->addEffect_l(effect); 984 if (lStatus != NO_ERROR) { 985 goto Exit; 986 } 987 effectCreated = true; 988 989 effect->setDevice(mOutDevice); 990 effect->setDevice(mInDevice); 991 effect->setMode(mAudioFlinger->getMode()); 992 effect->setAudioSource(mAudioSource); 993 } 994 // create effect handle and connect it to effect module 995 handle = new EffectHandle(effect, client, effectClient, priority); 996 lStatus = handle->initCheck(); 997 if (lStatus == OK) { 998 lStatus = effect->addHandle(handle.get()); 999 } 1000 if (enabled != NULL) { 1001 *enabled = (int)effect->isEnabled(); 1002 } 1003 } 1004 1005Exit: 1006 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1007 Mutex::Autolock _l(mLock); 1008 if (effectCreated) { 1009 chain->removeEffect_l(effect); 1010 } 1011 if (effectRegistered) { 1012 AudioSystem::unregisterEffect(effect->id()); 1013 } 1014 if (chainCreated) { 1015 removeEffectChain_l(chain); 1016 } 1017 handle.clear(); 1018 } 1019 1020 *status = lStatus; 1021 return handle; 1022} 1023 1024sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1025{ 1026 Mutex::Autolock _l(mLock); 1027 return getEffect_l(sessionId, effectId); 1028} 1029 1030sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1031{ 1032 sp<EffectChain> chain = getEffectChain_l(sessionId); 1033 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1034} 1035 1036// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1037// PlaybackThread::mLock held 1038status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1039{ 1040 // check for existing effect chain with the requested audio session 1041 int sessionId = effect->sessionId(); 1042 sp<EffectChain> chain = getEffectChain_l(sessionId); 1043 bool chainCreated = false; 1044 1045 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1046 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1047 this, effect->desc().name, effect->desc().flags); 1048 1049 if (chain == 0) { 1050 // create a new chain for this session 1051 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1052 chain = new EffectChain(this, sessionId); 1053 addEffectChain_l(chain); 1054 chain->setStrategy(getStrategyForSession_l(sessionId)); 1055 chainCreated = true; 1056 } 1057 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1058 1059 if (chain->getEffectFromId_l(effect->id()) != 0) { 1060 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1061 this, effect->desc().name, chain.get()); 1062 return BAD_VALUE; 1063 } 1064 1065 effect->setOffloaded(mType == OFFLOAD, mId); 1066 1067 status_t status = chain->addEffect_l(effect); 1068 if (status != NO_ERROR) { 1069 if (chainCreated) { 1070 removeEffectChain_l(chain); 1071 } 1072 return status; 1073 } 1074 1075 effect->setDevice(mOutDevice); 1076 effect->setDevice(mInDevice); 1077 effect->setMode(mAudioFlinger->getMode()); 1078 effect->setAudioSource(mAudioSource); 1079 return NO_ERROR; 1080} 1081 1082void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1083 1084 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1085 effect_descriptor_t desc = effect->desc(); 1086 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1087 detachAuxEffect_l(effect->id()); 1088 } 1089 1090 sp<EffectChain> chain = effect->chain().promote(); 1091 if (chain != 0) { 1092 // remove effect chain if removing last effect 1093 if (chain->removeEffect_l(effect) == 0) { 1094 removeEffectChain_l(chain); 1095 } 1096 } else { 1097 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::lockEffectChains_l( 1102 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1103{ 1104 effectChains = mEffectChains; 1105 for (size_t i = 0; i < mEffectChains.size(); i++) { 1106 mEffectChains[i]->lock(); 1107 } 1108} 1109 1110void AudioFlinger::ThreadBase::unlockEffectChains( 1111 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1112{ 1113 for (size_t i = 0; i < effectChains.size(); i++) { 1114 effectChains[i]->unlock(); 1115 } 1116} 1117 1118sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1119{ 1120 Mutex::Autolock _l(mLock); 1121 return getEffectChain_l(sessionId); 1122} 1123 1124sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1125{ 1126 size_t size = mEffectChains.size(); 1127 for (size_t i = 0; i < size; i++) { 1128 if (mEffectChains[i]->sessionId() == sessionId) { 1129 return mEffectChains[i]; 1130 } 1131 } 1132 return 0; 1133} 1134 1135void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1136{ 1137 Mutex::Autolock _l(mLock); 1138 size_t size = mEffectChains.size(); 1139 for (size_t i = 0; i < size; i++) { 1140 mEffectChains[i]->setMode_l(mode); 1141 } 1142} 1143 1144void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1145 EffectHandle *handle, 1146 bool unpinIfLast) { 1147 1148 Mutex::Autolock _l(mLock); 1149 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1150 // delete the effect module if removing last handle on it 1151 if (effect->removeHandle(handle) == 0) { 1152 if (!effect->isPinned() || unpinIfLast) { 1153 removeEffect_l(effect); 1154 AudioSystem::unregisterEffect(effect->id()); 1155 } 1156 } 1157} 1158 1159void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1160{ 1161 config->type = AUDIO_PORT_TYPE_MIX; 1162 config->ext.mix.handle = mId; 1163 config->sample_rate = mSampleRate; 1164 config->format = mFormat; 1165 config->channel_mask = mChannelMask; 1166 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1167 AUDIO_PORT_CONFIG_FORMAT; 1168} 1169 1170 1171// ---------------------------------------------------------------------------- 1172// Playback 1173// ---------------------------------------------------------------------------- 1174 1175AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1176 AudioStreamOut* output, 1177 audio_io_handle_t id, 1178 audio_devices_t device, 1179 type_t type) 1180 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1181 mNormalFrameCount(0), mSinkBuffer(NULL), 1182 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1183 mMixerBuffer(NULL), 1184 mMixerBufferSize(0), 1185 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1186 mMixerBufferValid(false), 1187 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1188 mEffectBuffer(NULL), 1189 mEffectBufferSize(0), 1190 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1191 mEffectBufferValid(false), 1192 mSuspended(0), mBytesWritten(0), 1193 mActiveTracksGeneration(0), 1194 // mStreamTypes[] initialized in constructor body 1195 mOutput(output), 1196 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1197 mMixerStatus(MIXER_IDLE), 1198 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1199 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1200 mBytesRemaining(0), 1201 mCurrentWriteLength(0), 1202 mUseAsyncWrite(false), 1203 mWriteAckSequence(0), 1204 mDrainSequence(0), 1205 mSignalPending(false), 1206 mScreenState(AudioFlinger::mScreenState), 1207 // index 0 is reserved for normal mixer's submix 1208 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1209 // mLatchD, mLatchQ, 1210 mLatchDValid(false), mLatchQValid(false) 1211{ 1212 snprintf(mName, kNameLength, "AudioOut_%X", id); 1213 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1214 1215 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1216 // it would be safer to explicitly pass initial masterVolume/masterMute as 1217 // parameter. 1218 // 1219 // If the HAL we are using has support for master volume or master mute, 1220 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1221 // and the mute set to false). 1222 mMasterVolume = audioFlinger->masterVolume_l(); 1223 mMasterMute = audioFlinger->masterMute_l(); 1224 if (mOutput && mOutput->audioHwDev) { 1225 if (mOutput->audioHwDev->canSetMasterVolume()) { 1226 mMasterVolume = 1.0; 1227 } 1228 1229 if (mOutput->audioHwDev->canSetMasterMute()) { 1230 mMasterMute = false; 1231 } 1232 } 1233 1234 readOutputParameters_l(); 1235 1236 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1237 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1238 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1239 stream = (audio_stream_type_t) (stream + 1)) { 1240 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1241 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1242 } 1243 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1244 // because mAudioFlinger doesn't have one to copy from 1245} 1246 1247AudioFlinger::PlaybackThread::~PlaybackThread() 1248{ 1249 mAudioFlinger->unregisterWriter(mNBLogWriter); 1250 free(mSinkBuffer); 1251 free(mMixerBuffer); 1252 free(mEffectBuffer); 1253} 1254 1255void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1256{ 1257 dumpInternals(fd, args); 1258 dumpTracks(fd, args); 1259 dumpEffectChains(fd, args); 1260} 1261 1262void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1263{ 1264 const size_t SIZE = 256; 1265 char buffer[SIZE]; 1266 String8 result; 1267 1268 result.appendFormat(" Stream volumes in dB: "); 1269 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1270 const stream_type_t *st = &mStreamTypes[i]; 1271 if (i > 0) { 1272 result.appendFormat(", "); 1273 } 1274 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1275 if (st->mute) { 1276 result.append("M"); 1277 } 1278 } 1279 result.append("\n"); 1280 write(fd, result.string(), result.length()); 1281 result.clear(); 1282 1283 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1284 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1285 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1286 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1287 1288 size_t numtracks = mTracks.size(); 1289 size_t numactive = mActiveTracks.size(); 1290 dprintf(fd, " %d Tracks", numtracks); 1291 size_t numactiveseen = 0; 1292 if (numtracks) { 1293 dprintf(fd, " of which %d are active\n", numactive); 1294 Track::appendDumpHeader(result); 1295 for (size_t i = 0; i < numtracks; ++i) { 1296 sp<Track> track = mTracks[i]; 1297 if (track != 0) { 1298 bool active = mActiveTracks.indexOf(track) >= 0; 1299 if (active) { 1300 numactiveseen++; 1301 } 1302 track->dump(buffer, SIZE, active); 1303 result.append(buffer); 1304 } 1305 } 1306 } else { 1307 result.append("\n"); 1308 } 1309 if (numactiveseen != numactive) { 1310 // some tracks in the active list were not in the tracks list 1311 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1312 " not in the track list\n"); 1313 result.append(buffer); 1314 Track::appendDumpHeader(result); 1315 for (size_t i = 0; i < numactive; ++i) { 1316 sp<Track> track = mActiveTracks[i].promote(); 1317 if (track != 0 && mTracks.indexOf(track) < 0) { 1318 track->dump(buffer, SIZE, true); 1319 result.append(buffer); 1320 } 1321 } 1322 } 1323 1324 write(fd, result.string(), result.size()); 1325} 1326 1327void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1328{ 1329 dprintf(fd, "\nOutput thread %p:\n", this); 1330 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1331 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1332 dprintf(fd, " Total writes: %d\n", mNumWrites); 1333 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1334 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1335 dprintf(fd, " Suspend count: %d\n", mSuspended); 1336 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1337 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1338 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1339 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1340 1341 dumpBase(fd, args); 1342} 1343 1344// Thread virtuals 1345 1346void AudioFlinger::PlaybackThread::onFirstRef() 1347{ 1348 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1349} 1350 1351// ThreadBase virtuals 1352void AudioFlinger::PlaybackThread::preExit() 1353{ 1354 ALOGV(" preExit()"); 1355 // FIXME this is using hard-coded strings but in the future, this functionality will be 1356 // converted to use audio HAL extensions required to support tunneling 1357 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1358} 1359 1360// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1361sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1362 const sp<AudioFlinger::Client>& client, 1363 audio_stream_type_t streamType, 1364 uint32_t sampleRate, 1365 audio_format_t format, 1366 audio_channel_mask_t channelMask, 1367 size_t *pFrameCount, 1368 const sp<IMemory>& sharedBuffer, 1369 int sessionId, 1370 IAudioFlinger::track_flags_t *flags, 1371 pid_t tid, 1372 int uid, 1373 status_t *status) 1374{ 1375 size_t frameCount = *pFrameCount; 1376 sp<Track> track; 1377 status_t lStatus; 1378 1379 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1380 1381 // client expresses a preference for FAST, but we get the final say 1382 if (*flags & IAudioFlinger::TRACK_FAST) { 1383 if ( 1384 // not timed 1385 (!isTimed) && 1386 // either of these use cases: 1387 ( 1388 // use case 1: shared buffer with any frame count 1389 ( 1390 (sharedBuffer != 0) 1391 ) || 1392 // use case 2: callback handler and frame count is default or at least as large as HAL 1393 ( 1394 (tid != -1) && 1395 ((frameCount == 0) || 1396 (frameCount >= mFrameCount)) 1397 ) 1398 ) && 1399 // PCM data 1400 audio_is_linear_pcm(format) && 1401 // identical channel mask to sink, or mono in and stereo sink 1402 (channelMask == mChannelMask || 1403 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1404 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1405 // hardware sample rate 1406 (sampleRate == mSampleRate) && 1407 // normal mixer has an associated fast mixer 1408 hasFastMixer() && 1409 // there are sufficient fast track slots available 1410 (mFastTrackAvailMask != 0) 1411 // FIXME test that MixerThread for this fast track has a capable output HAL 1412 // FIXME add a permission test also? 1413 ) { 1414 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1415 if (frameCount == 0) { 1416 // read the fast track multiplier property the first time it is needed 1417 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1418 if (ok != 0) { 1419 ALOGE("%s pthread_once failed: %d", __func__, ok); 1420 } 1421 frameCount = mFrameCount * sFastTrackMultiplier; 1422 } 1423 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1424 frameCount, mFrameCount); 1425 } else { 1426 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1427 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1428 "sampleRate=%u mSampleRate=%u " 1429 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1430 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1431 audio_is_linear_pcm(format), 1432 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1433 *flags &= ~IAudioFlinger::TRACK_FAST; 1434 // For compatibility with AudioTrack calculation, buffer depth is forced 1435 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1436 // This is probably too conservative, but legacy application code may depend on it. 1437 // If you change this calculation, also review the start threshold which is related. 1438 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1439 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1440 if (minBufCount < 2) { 1441 minBufCount = 2; 1442 } 1443 size_t minFrameCount = mNormalFrameCount * minBufCount; 1444 if (frameCount < minFrameCount) { 1445 frameCount = minFrameCount; 1446 } 1447 } 1448 } 1449 *pFrameCount = frameCount; 1450 1451 switch (mType) { 1452 1453 case DIRECT: 1454 if (audio_is_linear_pcm(format)) { 1455 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1456 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1457 "for output %p with format %#x", 1458 sampleRate, format, channelMask, mOutput, mFormat); 1459 lStatus = BAD_VALUE; 1460 goto Exit; 1461 } 1462 } 1463 break; 1464 1465 case OFFLOAD: 1466 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1467 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1468 "for output %p with format %#x", 1469 sampleRate, format, channelMask, mOutput, mFormat); 1470 lStatus = BAD_VALUE; 1471 goto Exit; 1472 } 1473 break; 1474 1475 default: 1476 if (!audio_is_linear_pcm(format)) { 1477 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1478 "for output %p with format %#x", 1479 format, mOutput, mFormat); 1480 lStatus = BAD_VALUE; 1481 goto Exit; 1482 } 1483 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1484 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1485 lStatus = BAD_VALUE; 1486 goto Exit; 1487 } 1488 break; 1489 1490 } 1491 1492 lStatus = initCheck(); 1493 if (lStatus != NO_ERROR) { 1494 ALOGE("createTrack_l() audio driver not initialized"); 1495 goto Exit; 1496 } 1497 1498 { // scope for mLock 1499 Mutex::Autolock _l(mLock); 1500 1501 // all tracks in same audio session must share the same routing strategy otherwise 1502 // conflicts will happen when tracks are moved from one output to another by audio policy 1503 // manager 1504 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1505 for (size_t i = 0; i < mTracks.size(); ++i) { 1506 sp<Track> t = mTracks[i]; 1507 if (t != 0 && t->isExternalTrack()) { 1508 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1509 if (sessionId == t->sessionId() && strategy != actual) { 1510 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1511 strategy, actual); 1512 lStatus = BAD_VALUE; 1513 goto Exit; 1514 } 1515 } 1516 } 1517 1518 if (!isTimed) { 1519 track = new Track(this, client, streamType, sampleRate, format, 1520 channelMask, frameCount, NULL, sharedBuffer, 1521 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1522 } else { 1523 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1524 channelMask, frameCount, sharedBuffer, sessionId, uid); 1525 } 1526 1527 // new Track always returns non-NULL, 1528 // but TimedTrack::create() is a factory that could fail by returning NULL 1529 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1530 if (lStatus != NO_ERROR) { 1531 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1532 // track must be cleared from the caller as the caller has the AF lock 1533 goto Exit; 1534 } 1535 mTracks.add(track); 1536 1537 sp<EffectChain> chain = getEffectChain_l(sessionId); 1538 if (chain != 0) { 1539 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1540 track->setMainBuffer(chain->inBuffer()); 1541 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1542 chain->incTrackCnt(); 1543 } 1544 1545 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1546 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1547 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1548 // so ask activity manager to do this on our behalf 1549 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1550 } 1551 } 1552 1553 lStatus = NO_ERROR; 1554 1555Exit: 1556 *status = lStatus; 1557 return track; 1558} 1559 1560uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1561{ 1562 return latency; 1563} 1564 1565uint32_t AudioFlinger::PlaybackThread::latency() const 1566{ 1567 Mutex::Autolock _l(mLock); 1568 return latency_l(); 1569} 1570uint32_t AudioFlinger::PlaybackThread::latency_l() const 1571{ 1572 if (initCheck() == NO_ERROR) { 1573 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1574 } else { 1575 return 0; 1576 } 1577} 1578 1579void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1580{ 1581 Mutex::Autolock _l(mLock); 1582 // Don't apply master volume in SW if our HAL can do it for us. 1583 if (mOutput && mOutput->audioHwDev && 1584 mOutput->audioHwDev->canSetMasterVolume()) { 1585 mMasterVolume = 1.0; 1586 } else { 1587 mMasterVolume = value; 1588 } 1589} 1590 1591void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1592{ 1593 Mutex::Autolock _l(mLock); 1594 // Don't apply master mute in SW if our HAL can do it for us. 1595 if (mOutput && mOutput->audioHwDev && 1596 mOutput->audioHwDev->canSetMasterMute()) { 1597 mMasterMute = false; 1598 } else { 1599 mMasterMute = muted; 1600 } 1601} 1602 1603void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1604{ 1605 Mutex::Autolock _l(mLock); 1606 mStreamTypes[stream].volume = value; 1607 broadcast_l(); 1608} 1609 1610void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1611{ 1612 Mutex::Autolock _l(mLock); 1613 mStreamTypes[stream].mute = muted; 1614 broadcast_l(); 1615} 1616 1617float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1618{ 1619 Mutex::Autolock _l(mLock); 1620 return mStreamTypes[stream].volume; 1621} 1622 1623// addTrack_l() must be called with ThreadBase::mLock held 1624status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1625{ 1626 status_t status = ALREADY_EXISTS; 1627 1628 // set retry count for buffer fill 1629 track->mRetryCount = kMaxTrackStartupRetries; 1630 if (mActiveTracks.indexOf(track) < 0) { 1631 // the track is newly added, make sure it fills up all its 1632 // buffers before playing. This is to ensure the client will 1633 // effectively get the latency it requested. 1634 if (track->isExternalTrack()) { 1635 TrackBase::track_state state = track->mState; 1636 mLock.unlock(); 1637 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1638 mLock.lock(); 1639 // abort track was stopped/paused while we released the lock 1640 if (state != track->mState) { 1641 if (status == NO_ERROR) { 1642 mLock.unlock(); 1643 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1644 mLock.lock(); 1645 } 1646 return INVALID_OPERATION; 1647 } 1648 // abort if start is rejected by audio policy manager 1649 if (status != NO_ERROR) { 1650 return PERMISSION_DENIED; 1651 } 1652#ifdef ADD_BATTERY_DATA 1653 // to track the speaker usage 1654 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1655#endif 1656 } 1657 1658 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1659 track->mResetDone = false; 1660 track->mPresentationCompleteFrames = 0; 1661 mActiveTracks.add(track); 1662 mWakeLockUids.add(track->uid()); 1663 mActiveTracksGeneration++; 1664 mLatestActiveTrack = track; 1665 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1666 if (chain != 0) { 1667 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1668 track->sessionId()); 1669 chain->incActiveTrackCnt(); 1670 } 1671 1672 status = NO_ERROR; 1673 } 1674 1675 onAddNewTrack_l(); 1676 return status; 1677} 1678 1679bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1680{ 1681 track->terminate(); 1682 // active tracks are removed by threadLoop() 1683 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1684 track->mState = TrackBase::STOPPED; 1685 if (!trackActive) { 1686 removeTrack_l(track); 1687 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1688 track->mState = TrackBase::STOPPING_1; 1689 } 1690 1691 return trackActive; 1692} 1693 1694void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1695{ 1696 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1697 mTracks.remove(track); 1698 deleteTrackName_l(track->name()); 1699 // redundant as track is about to be destroyed, for dumpsys only 1700 track->mName = -1; 1701 if (track->isFastTrack()) { 1702 int index = track->mFastIndex; 1703 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1704 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1705 mFastTrackAvailMask |= 1 << index; 1706 // redundant as track is about to be destroyed, for dumpsys only 1707 track->mFastIndex = -1; 1708 } 1709 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1710 if (chain != 0) { 1711 chain->decTrackCnt(); 1712 } 1713} 1714 1715void AudioFlinger::PlaybackThread::broadcast_l() 1716{ 1717 // Thread could be blocked waiting for async 1718 // so signal it to handle state changes immediately 1719 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1720 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1721 mSignalPending = true; 1722 mWaitWorkCV.broadcast(); 1723} 1724 1725String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1726{ 1727 Mutex::Autolock _l(mLock); 1728 if (initCheck() != NO_ERROR) { 1729 return String8(); 1730 } 1731 1732 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1733 const String8 out_s8(s); 1734 free(s); 1735 return out_s8; 1736} 1737 1738void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1739 AudioSystem::OutputDescriptor desc; 1740 void *param2 = NULL; 1741 1742 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1743 param); 1744 1745 switch (event) { 1746 case AudioSystem::OUTPUT_OPENED: 1747 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1748 desc.channelMask = mChannelMask; 1749 desc.samplingRate = mSampleRate; 1750 desc.format = mFormat; 1751 desc.frameCount = mNormalFrameCount; // FIXME see 1752 // AudioFlinger::frameCount(audio_io_handle_t) 1753 desc.latency = latency_l(); 1754 param2 = &desc; 1755 break; 1756 1757 case AudioSystem::STREAM_CONFIG_CHANGED: 1758 param2 = ¶m; 1759 case AudioSystem::OUTPUT_CLOSED: 1760 default: 1761 break; 1762 } 1763 mAudioFlinger->audioConfigChanged(event, mId, param2); 1764} 1765 1766void AudioFlinger::PlaybackThread::writeCallback() 1767{ 1768 ALOG_ASSERT(mCallbackThread != 0); 1769 mCallbackThread->resetWriteBlocked(); 1770} 1771 1772void AudioFlinger::PlaybackThread::drainCallback() 1773{ 1774 ALOG_ASSERT(mCallbackThread != 0); 1775 mCallbackThread->resetDraining(); 1776} 1777 1778void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1779{ 1780 Mutex::Autolock _l(mLock); 1781 // reject out of sequence requests 1782 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1783 mWriteAckSequence &= ~1; 1784 mWaitWorkCV.signal(); 1785 } 1786} 1787 1788void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1789{ 1790 Mutex::Autolock _l(mLock); 1791 // reject out of sequence requests 1792 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1793 mDrainSequence &= ~1; 1794 mWaitWorkCV.signal(); 1795 } 1796} 1797 1798// static 1799int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1800 void *param __unused, 1801 void *cookie) 1802{ 1803 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1804 ALOGV("asyncCallback() event %d", event); 1805 switch (event) { 1806 case STREAM_CBK_EVENT_WRITE_READY: 1807 me->writeCallback(); 1808 break; 1809 case STREAM_CBK_EVENT_DRAIN_READY: 1810 me->drainCallback(); 1811 break; 1812 default: 1813 ALOGW("asyncCallback() unknown event %d", event); 1814 break; 1815 } 1816 return 0; 1817} 1818 1819void AudioFlinger::PlaybackThread::readOutputParameters_l() 1820{ 1821 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1822 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1823 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1824 if (!audio_is_output_channel(mChannelMask)) { 1825 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1826 } 1827 if ((mType == MIXER || mType == DUPLICATING) 1828 && !isValidPcmSinkChannelMask(mChannelMask)) { 1829 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1830 mChannelMask); 1831 } 1832 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1833 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1834 mFormat = mHALFormat; 1835 if (!audio_is_valid_format(mFormat)) { 1836 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1837 } 1838 if ((mType == MIXER || mType == DUPLICATING) 1839 && !isValidPcmSinkFormat(mFormat)) { 1840 LOG_FATAL("HAL format %#x not supported for mixed output", 1841 mFormat); 1842 } 1843 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1844 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1845 mFrameCount = mBufferSize / mFrameSize; 1846 if (mFrameCount & 15) { 1847 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1848 mFrameCount); 1849 } 1850 1851 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1852 (mOutput->stream->set_callback != NULL)) { 1853 if (mOutput->stream->set_callback(mOutput->stream, 1854 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1855 mUseAsyncWrite = true; 1856 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1857 } 1858 } 1859 1860 // Calculate size of normal sink buffer relative to the HAL output buffer size 1861 double multiplier = 1.0; 1862 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1863 kUseFastMixer == FastMixer_Dynamic)) { 1864 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1865 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1866 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1867 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1868 maxNormalFrameCount = maxNormalFrameCount & ~15; 1869 if (maxNormalFrameCount < minNormalFrameCount) { 1870 maxNormalFrameCount = minNormalFrameCount; 1871 } 1872 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1873 if (multiplier <= 1.0) { 1874 multiplier = 1.0; 1875 } else if (multiplier <= 2.0) { 1876 if (2 * mFrameCount <= maxNormalFrameCount) { 1877 multiplier = 2.0; 1878 } else { 1879 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1880 } 1881 } else { 1882 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1883 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1884 // track, but we sometimes have to do this to satisfy the maximum frame count 1885 // constraint) 1886 // FIXME this rounding up should not be done if no HAL SRC 1887 uint32_t truncMult = (uint32_t) multiplier; 1888 if ((truncMult & 1)) { 1889 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1890 ++truncMult; 1891 } 1892 } 1893 multiplier = (double) truncMult; 1894 } 1895 } 1896 mNormalFrameCount = multiplier * mFrameCount; 1897 // round up to nearest 16 frames to satisfy AudioMixer 1898 if (mType == MIXER || mType == DUPLICATING) { 1899 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1900 } 1901 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1902 mNormalFrameCount); 1903 1904 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1905 // Originally this was int16_t[] array, need to remove legacy implications. 1906 free(mSinkBuffer); 1907 mSinkBuffer = NULL; 1908 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1909 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1910 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1911 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1912 1913 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1914 // drives the output. 1915 free(mMixerBuffer); 1916 mMixerBuffer = NULL; 1917 if (mMixerBufferEnabled) { 1918 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1919 mMixerBufferSize = mNormalFrameCount * mChannelCount 1920 * audio_bytes_per_sample(mMixerBufferFormat); 1921 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1922 } 1923 free(mEffectBuffer); 1924 mEffectBuffer = NULL; 1925 if (mEffectBufferEnabled) { 1926 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1927 mEffectBufferSize = mNormalFrameCount * mChannelCount 1928 * audio_bytes_per_sample(mEffectBufferFormat); 1929 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1930 } 1931 1932 // force reconfiguration of effect chains and engines to take new buffer size and audio 1933 // parameters into account 1934 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1935 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1936 // matter. 1937 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1938 Vector< sp<EffectChain> > effectChains = mEffectChains; 1939 for (size_t i = 0; i < effectChains.size(); i ++) { 1940 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1941 } 1942} 1943 1944 1945status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1946{ 1947 if (halFrames == NULL || dspFrames == NULL) { 1948 return BAD_VALUE; 1949 } 1950 Mutex::Autolock _l(mLock); 1951 if (initCheck() != NO_ERROR) { 1952 return INVALID_OPERATION; 1953 } 1954 size_t framesWritten = mBytesWritten / mFrameSize; 1955 *halFrames = framesWritten; 1956 1957 if (isSuspended()) { 1958 // return an estimation of rendered frames when the output is suspended 1959 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1960 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1961 return NO_ERROR; 1962 } else { 1963 status_t status; 1964 uint32_t frames; 1965 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1966 *dspFrames = (size_t)frames; 1967 return status; 1968 } 1969} 1970 1971uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1972{ 1973 Mutex::Autolock _l(mLock); 1974 uint32_t result = 0; 1975 if (getEffectChain_l(sessionId) != 0) { 1976 result = EFFECT_SESSION; 1977 } 1978 1979 for (size_t i = 0; i < mTracks.size(); ++i) { 1980 sp<Track> track = mTracks[i]; 1981 if (sessionId == track->sessionId() && !track->isInvalid()) { 1982 result |= TRACK_SESSION; 1983 break; 1984 } 1985 } 1986 1987 return result; 1988} 1989 1990uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1991{ 1992 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1993 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1994 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1995 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1996 } 1997 for (size_t i = 0; i < mTracks.size(); i++) { 1998 sp<Track> track = mTracks[i]; 1999 if (sessionId == track->sessionId() && !track->isInvalid()) { 2000 return AudioSystem::getStrategyForStream(track->streamType()); 2001 } 2002 } 2003 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2004} 2005 2006 2007AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2008{ 2009 Mutex::Autolock _l(mLock); 2010 return mOutput; 2011} 2012 2013AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2014{ 2015 Mutex::Autolock _l(mLock); 2016 AudioStreamOut *output = mOutput; 2017 mOutput = NULL; 2018 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2019 // must push a NULL and wait for ack 2020 mOutputSink.clear(); 2021 mPipeSink.clear(); 2022 mNormalSink.clear(); 2023 return output; 2024} 2025 2026// this method must always be called either with ThreadBase mLock held or inside the thread loop 2027audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2028{ 2029 if (mOutput == NULL) { 2030 return NULL; 2031 } 2032 return &mOutput->stream->common; 2033} 2034 2035uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2036{ 2037 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2038} 2039 2040status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2041{ 2042 if (!isValidSyncEvent(event)) { 2043 return BAD_VALUE; 2044 } 2045 2046 Mutex::Autolock _l(mLock); 2047 2048 for (size_t i = 0; i < mTracks.size(); ++i) { 2049 sp<Track> track = mTracks[i]; 2050 if (event->triggerSession() == track->sessionId()) { 2051 (void) track->setSyncEvent(event); 2052 return NO_ERROR; 2053 } 2054 } 2055 2056 return NAME_NOT_FOUND; 2057} 2058 2059bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2060{ 2061 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2062} 2063 2064void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2065 const Vector< sp<Track> >& tracksToRemove) 2066{ 2067 size_t count = tracksToRemove.size(); 2068 if (count > 0) { 2069 for (size_t i = 0 ; i < count ; i++) { 2070 const sp<Track>& track = tracksToRemove.itemAt(i); 2071 if (track->isExternalTrack()) { 2072 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2073#ifdef ADD_BATTERY_DATA 2074 // to track the speaker usage 2075 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2076#endif 2077 if (track->isTerminated()) { 2078 AudioSystem::releaseOutput(mId); 2079 } 2080 } 2081 } 2082 } 2083} 2084 2085void AudioFlinger::PlaybackThread::checkSilentMode_l() 2086{ 2087 if (!mMasterMute) { 2088 char value[PROPERTY_VALUE_MAX]; 2089 if (property_get("ro.audio.silent", value, "0") > 0) { 2090 char *endptr; 2091 unsigned long ul = strtoul(value, &endptr, 0); 2092 if (*endptr == '\0' && ul != 0) { 2093 ALOGD("Silence is golden"); 2094 // The setprop command will not allow a property to be changed after 2095 // the first time it is set, so we don't have to worry about un-muting. 2096 setMasterMute_l(true); 2097 } 2098 } 2099 } 2100} 2101 2102// shared by MIXER and DIRECT, overridden by DUPLICATING 2103ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2104{ 2105 // FIXME rewrite to reduce number of system calls 2106 mLastWriteTime = systemTime(); 2107 mInWrite = true; 2108 ssize_t bytesWritten; 2109 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2110 2111 // If an NBAIO sink is present, use it to write the normal mixer's submix 2112 if (mNormalSink != 0) { 2113 const size_t count = mBytesRemaining / mFrameSize; 2114 2115 ATRACE_BEGIN("write"); 2116 // update the setpoint when AudioFlinger::mScreenState changes 2117 uint32_t screenState = AudioFlinger::mScreenState; 2118 if (screenState != mScreenState) { 2119 mScreenState = screenState; 2120 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2121 if (pipe != NULL) { 2122 pipe->setAvgFrames((mScreenState & 1) ? 2123 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2124 } 2125 } 2126 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2127 ATRACE_END(); 2128 if (framesWritten > 0) { 2129 bytesWritten = framesWritten * mFrameSize; 2130 } else { 2131 bytesWritten = framesWritten; 2132 } 2133 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2134 if (status == NO_ERROR) { 2135 size_t totalFramesWritten = mNormalSink->framesWritten(); 2136 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2137 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2138 mLatchDValid = true; 2139 } 2140 } 2141 // otherwise use the HAL / AudioStreamOut directly 2142 } else { 2143 // Direct output and offload threads 2144 2145 if (mUseAsyncWrite) { 2146 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2147 mWriteAckSequence += 2; 2148 mWriteAckSequence |= 1; 2149 ALOG_ASSERT(mCallbackThread != 0); 2150 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2151 } 2152 // FIXME We should have an implementation of timestamps for direct output threads. 2153 // They are used e.g for multichannel PCM playback over HDMI. 2154 bytesWritten = mOutput->stream->write(mOutput->stream, 2155 (char *)mSinkBuffer + offset, mBytesRemaining); 2156 if (mUseAsyncWrite && 2157 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2158 // do not wait for async callback in case of error of full write 2159 mWriteAckSequence &= ~1; 2160 ALOG_ASSERT(mCallbackThread != 0); 2161 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2162 } 2163 } 2164 2165 mNumWrites++; 2166 mInWrite = false; 2167 mStandby = false; 2168 return bytesWritten; 2169} 2170 2171void AudioFlinger::PlaybackThread::threadLoop_drain() 2172{ 2173 if (mOutput->stream->drain) { 2174 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2175 if (mUseAsyncWrite) { 2176 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2177 mDrainSequence |= 1; 2178 ALOG_ASSERT(mCallbackThread != 0); 2179 mCallbackThread->setDraining(mDrainSequence); 2180 } 2181 mOutput->stream->drain(mOutput->stream, 2182 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2183 : AUDIO_DRAIN_ALL); 2184 } 2185} 2186 2187void AudioFlinger::PlaybackThread::threadLoop_exit() 2188{ 2189 // Default implementation has nothing to do 2190} 2191 2192/* 2193The derived values that are cached: 2194 - mSinkBufferSize from frame count * frame size 2195 - activeSleepTime from activeSleepTimeUs() 2196 - idleSleepTime from idleSleepTimeUs() 2197 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2198 - maxPeriod from frame count and sample rate (MIXER only) 2199 2200The parameters that affect these derived values are: 2201 - frame count 2202 - frame size 2203 - sample rate 2204 - device type: A2DP or not 2205 - device latency 2206 - format: PCM or not 2207 - active sleep time 2208 - idle sleep time 2209*/ 2210 2211void AudioFlinger::PlaybackThread::cacheParameters_l() 2212{ 2213 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2214 activeSleepTime = activeSleepTimeUs(); 2215 idleSleepTime = idleSleepTimeUs(); 2216} 2217 2218void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2219{ 2220 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2221 this, streamType, mTracks.size()); 2222 Mutex::Autolock _l(mLock); 2223 2224 size_t size = mTracks.size(); 2225 for (size_t i = 0; i < size; i++) { 2226 sp<Track> t = mTracks[i]; 2227 if (t->streamType() == streamType) { 2228 t->invalidate(); 2229 } 2230 } 2231} 2232 2233status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2234{ 2235 int session = chain->sessionId(); 2236 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2237 ? mEffectBuffer : mSinkBuffer); 2238 bool ownsBuffer = false; 2239 2240 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2241 if (session > 0) { 2242 // Only one effect chain can be present in direct output thread and it uses 2243 // the sink buffer as input 2244 if (mType != DIRECT) { 2245 size_t numSamples = mNormalFrameCount * mChannelCount; 2246 buffer = new int16_t[numSamples]; 2247 memset(buffer, 0, numSamples * sizeof(int16_t)); 2248 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2249 ownsBuffer = true; 2250 } 2251 2252 // Attach all tracks with same session ID to this chain. 2253 for (size_t i = 0; i < mTracks.size(); ++i) { 2254 sp<Track> track = mTracks[i]; 2255 if (session == track->sessionId()) { 2256 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2257 buffer); 2258 track->setMainBuffer(buffer); 2259 chain->incTrackCnt(); 2260 } 2261 } 2262 2263 // indicate all active tracks in the chain 2264 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2265 sp<Track> track = mActiveTracks[i].promote(); 2266 if (track == 0) { 2267 continue; 2268 } 2269 if (session == track->sessionId()) { 2270 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2271 chain->incActiveTrackCnt(); 2272 } 2273 } 2274 } 2275 2276 chain->setInBuffer(buffer, ownsBuffer); 2277 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2278 ? mEffectBuffer : mSinkBuffer)); 2279 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2280 // chains list in order to be processed last as it contains output stage effects 2281 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2282 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2283 // after track specific effects and before output stage 2284 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2285 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2286 // Effect chain for other sessions are inserted at beginning of effect 2287 // chains list to be processed before output mix effects. Relative order between other 2288 // sessions is not important 2289 size_t size = mEffectChains.size(); 2290 size_t i = 0; 2291 for (i = 0; i < size; i++) { 2292 if (mEffectChains[i]->sessionId() < session) { 2293 break; 2294 } 2295 } 2296 mEffectChains.insertAt(chain, i); 2297 checkSuspendOnAddEffectChain_l(chain); 2298 2299 return NO_ERROR; 2300} 2301 2302size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2303{ 2304 int session = chain->sessionId(); 2305 2306 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2307 2308 for (size_t i = 0; i < mEffectChains.size(); i++) { 2309 if (chain == mEffectChains[i]) { 2310 mEffectChains.removeAt(i); 2311 // detach all active tracks from the chain 2312 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2313 sp<Track> track = mActiveTracks[i].promote(); 2314 if (track == 0) { 2315 continue; 2316 } 2317 if (session == track->sessionId()) { 2318 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2319 chain.get(), session); 2320 chain->decActiveTrackCnt(); 2321 } 2322 } 2323 2324 // detach all tracks with same session ID from this chain 2325 for (size_t i = 0; i < mTracks.size(); ++i) { 2326 sp<Track> track = mTracks[i]; 2327 if (session == track->sessionId()) { 2328 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2329 chain->decTrackCnt(); 2330 } 2331 } 2332 break; 2333 } 2334 } 2335 return mEffectChains.size(); 2336} 2337 2338status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2339 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2340{ 2341 Mutex::Autolock _l(mLock); 2342 return attachAuxEffect_l(track, EffectId); 2343} 2344 2345status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2346 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2347{ 2348 status_t status = NO_ERROR; 2349 2350 if (EffectId == 0) { 2351 track->setAuxBuffer(0, NULL); 2352 } else { 2353 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2354 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2355 if (effect != 0) { 2356 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2357 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2358 } else { 2359 status = INVALID_OPERATION; 2360 } 2361 } else { 2362 status = BAD_VALUE; 2363 } 2364 } 2365 return status; 2366} 2367 2368void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2369{ 2370 for (size_t i = 0; i < mTracks.size(); ++i) { 2371 sp<Track> track = mTracks[i]; 2372 if (track->auxEffectId() == effectId) { 2373 attachAuxEffect_l(track, 0); 2374 } 2375 } 2376} 2377 2378bool AudioFlinger::PlaybackThread::threadLoop() 2379{ 2380 Vector< sp<Track> > tracksToRemove; 2381 2382 standbyTime = systemTime(); 2383 2384 // MIXER 2385 nsecs_t lastWarning = 0; 2386 2387 // DUPLICATING 2388 // FIXME could this be made local to while loop? 2389 writeFrames = 0; 2390 2391 int lastGeneration = 0; 2392 2393 cacheParameters_l(); 2394 sleepTime = idleSleepTime; 2395 2396 if (mType == MIXER) { 2397 sleepTimeShift = 0; 2398 } 2399 2400 CpuStats cpuStats; 2401 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2402 2403 acquireWakeLock(); 2404 2405 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2406 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2407 // and then that string will be logged at the next convenient opportunity. 2408 const char *logString = NULL; 2409 2410 checkSilentMode_l(); 2411 2412 while (!exitPending()) 2413 { 2414 cpuStats.sample(myName); 2415 2416 Vector< sp<EffectChain> > effectChains; 2417 2418 { // scope for mLock 2419 2420 Mutex::Autolock _l(mLock); 2421 2422 processConfigEvents_l(); 2423 2424 if (logString != NULL) { 2425 mNBLogWriter->logTimestamp(); 2426 mNBLogWriter->log(logString); 2427 logString = NULL; 2428 } 2429 2430 if (mLatchDValid) { 2431 mLatchQ = mLatchD; 2432 mLatchDValid = false; 2433 mLatchQValid = true; 2434 } 2435 2436 saveOutputTracks(); 2437 if (mSignalPending) { 2438 // A signal was raised while we were unlocked 2439 mSignalPending = false; 2440 } else if (waitingAsyncCallback_l()) { 2441 if (exitPending()) { 2442 break; 2443 } 2444 releaseWakeLock_l(); 2445 mWakeLockUids.clear(); 2446 mActiveTracksGeneration++; 2447 ALOGV("wait async completion"); 2448 mWaitWorkCV.wait(mLock); 2449 ALOGV("async completion/wake"); 2450 acquireWakeLock_l(); 2451 standbyTime = systemTime() + standbyDelay; 2452 sleepTime = 0; 2453 2454 continue; 2455 } 2456 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2457 isSuspended()) { 2458 // put audio hardware into standby after short delay 2459 if (shouldStandby_l()) { 2460 2461 threadLoop_standby(); 2462 2463 mStandby = true; 2464 } 2465 2466 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2467 // we're about to wait, flush the binder command buffer 2468 IPCThreadState::self()->flushCommands(); 2469 2470 clearOutputTracks(); 2471 2472 if (exitPending()) { 2473 break; 2474 } 2475 2476 releaseWakeLock_l(); 2477 mWakeLockUids.clear(); 2478 mActiveTracksGeneration++; 2479 // wait until we have something to do... 2480 ALOGV("%s going to sleep", myName.string()); 2481 mWaitWorkCV.wait(mLock); 2482 ALOGV("%s waking up", myName.string()); 2483 acquireWakeLock_l(); 2484 2485 mMixerStatus = MIXER_IDLE; 2486 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2487 mBytesWritten = 0; 2488 mBytesRemaining = 0; 2489 checkSilentMode_l(); 2490 2491 standbyTime = systemTime() + standbyDelay; 2492 sleepTime = idleSleepTime; 2493 if (mType == MIXER) { 2494 sleepTimeShift = 0; 2495 } 2496 2497 continue; 2498 } 2499 } 2500 // mMixerStatusIgnoringFastTracks is also updated internally 2501 mMixerStatus = prepareTracks_l(&tracksToRemove); 2502 2503 // compare with previously applied list 2504 if (lastGeneration != mActiveTracksGeneration) { 2505 // update wakelock 2506 updateWakeLockUids_l(mWakeLockUids); 2507 lastGeneration = mActiveTracksGeneration; 2508 } 2509 2510 // prevent any changes in effect chain list and in each effect chain 2511 // during mixing and effect process as the audio buffers could be deleted 2512 // or modified if an effect is created or deleted 2513 lockEffectChains_l(effectChains); 2514 } // mLock scope ends 2515 2516 if (mBytesRemaining == 0) { 2517 mCurrentWriteLength = 0; 2518 if (mMixerStatus == MIXER_TRACKS_READY) { 2519 // threadLoop_mix() sets mCurrentWriteLength 2520 threadLoop_mix(); 2521 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2522 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2523 // threadLoop_sleepTime sets sleepTime to 0 if data 2524 // must be written to HAL 2525 threadLoop_sleepTime(); 2526 if (sleepTime == 0) { 2527 mCurrentWriteLength = mSinkBufferSize; 2528 } 2529 } 2530 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2531 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2532 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2533 // or mSinkBuffer (if there are no effects). 2534 // 2535 // This is done pre-effects computation; if effects change to 2536 // support higher precision, this needs to move. 2537 // 2538 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2539 // TODO use sleepTime == 0 as an additional condition. 2540 if (mMixerBufferValid) { 2541 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2542 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2543 2544 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2545 mNormalFrameCount * mChannelCount); 2546 } 2547 2548 mBytesRemaining = mCurrentWriteLength; 2549 if (isSuspended()) { 2550 sleepTime = suspendSleepTimeUs(); 2551 // simulate write to HAL when suspended 2552 mBytesWritten += mSinkBufferSize; 2553 mBytesRemaining = 0; 2554 } 2555 2556 // only process effects if we're going to write 2557 if (sleepTime == 0 && mType != OFFLOAD) { 2558 for (size_t i = 0; i < effectChains.size(); i ++) { 2559 effectChains[i]->process_l(); 2560 } 2561 } 2562 } 2563 // Process effect chains for offloaded thread even if no audio 2564 // was read from audio track: process only updates effect state 2565 // and thus does have to be synchronized with audio writes but may have 2566 // to be called while waiting for async write callback 2567 if (mType == OFFLOAD) { 2568 for (size_t i = 0; i < effectChains.size(); i ++) { 2569 effectChains[i]->process_l(); 2570 } 2571 } 2572 2573 // Only if the Effects buffer is enabled and there is data in the 2574 // Effects buffer (buffer valid), we need to 2575 // copy into the sink buffer. 2576 // TODO use sleepTime == 0 as an additional condition. 2577 if (mEffectBufferValid) { 2578 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2579 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2580 mNormalFrameCount * mChannelCount); 2581 } 2582 2583 // enable changes in effect chain 2584 unlockEffectChains(effectChains); 2585 2586 if (!waitingAsyncCallback()) { 2587 // sleepTime == 0 means we must write to audio hardware 2588 if (sleepTime == 0) { 2589 if (mBytesRemaining) { 2590 ssize_t ret = threadLoop_write(); 2591 if (ret < 0) { 2592 mBytesRemaining = 0; 2593 } else { 2594 mBytesWritten += ret; 2595 mBytesRemaining -= ret; 2596 } 2597 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2598 (mMixerStatus == MIXER_DRAIN_ALL)) { 2599 threadLoop_drain(); 2600 } 2601 if (mType == MIXER) { 2602 // write blocked detection 2603 nsecs_t now = systemTime(); 2604 nsecs_t delta = now - mLastWriteTime; 2605 if (!mStandby && delta > maxPeriod) { 2606 mNumDelayedWrites++; 2607 if ((now - lastWarning) > kWarningThrottleNs) { 2608 ATRACE_NAME("underrun"); 2609 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2610 ns2ms(delta), mNumDelayedWrites, this); 2611 lastWarning = now; 2612 } 2613 } 2614 } 2615 2616 } else { 2617 usleep(sleepTime); 2618 } 2619 } 2620 2621 // Finally let go of removed track(s), without the lock held 2622 // since we can't guarantee the destructors won't acquire that 2623 // same lock. This will also mutate and push a new fast mixer state. 2624 threadLoop_removeTracks(tracksToRemove); 2625 tracksToRemove.clear(); 2626 2627 // FIXME I don't understand the need for this here; 2628 // it was in the original code but maybe the 2629 // assignment in saveOutputTracks() makes this unnecessary? 2630 clearOutputTracks(); 2631 2632 // Effect chains will be actually deleted here if they were removed from 2633 // mEffectChains list during mixing or effects processing 2634 effectChains.clear(); 2635 2636 // FIXME Note that the above .clear() is no longer necessary since effectChains 2637 // is now local to this block, but will keep it for now (at least until merge done). 2638 } 2639 2640 threadLoop_exit(); 2641 2642 if (!mStandby) { 2643 threadLoop_standby(); 2644 mStandby = true; 2645 } 2646 2647 releaseWakeLock(); 2648 mWakeLockUids.clear(); 2649 mActiveTracksGeneration++; 2650 2651 ALOGV("Thread %p type %d exiting", this, mType); 2652 return false; 2653} 2654 2655// removeTracks_l() must be called with ThreadBase::mLock held 2656void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2657{ 2658 size_t count = tracksToRemove.size(); 2659 if (count > 0) { 2660 for (size_t i=0 ; i<count ; i++) { 2661 const sp<Track>& track = tracksToRemove.itemAt(i); 2662 mActiveTracks.remove(track); 2663 mWakeLockUids.remove(track->uid()); 2664 mActiveTracksGeneration++; 2665 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2666 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2667 if (chain != 0) { 2668 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2669 track->sessionId()); 2670 chain->decActiveTrackCnt(); 2671 } 2672 if (track->isTerminated()) { 2673 removeTrack_l(track); 2674 } 2675 } 2676 } 2677 2678} 2679 2680status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2681{ 2682 if (mNormalSink != 0) { 2683 return mNormalSink->getTimestamp(timestamp); 2684 } 2685 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2686 uint64_t position64; 2687 int ret = mOutput->stream->get_presentation_position( 2688 mOutput->stream, &position64, ×tamp.mTime); 2689 if (ret == 0) { 2690 timestamp.mPosition = (uint32_t)position64; 2691 return NO_ERROR; 2692 } 2693 } 2694 return INVALID_OPERATION; 2695} 2696 2697status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2698 audio_patch_handle_t *handle) 2699{ 2700 status_t status = NO_ERROR; 2701 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2702 // store new device and send to effects 2703 audio_devices_t type = AUDIO_DEVICE_NONE; 2704 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2705 type |= patch->sinks[i].ext.device.type; 2706 } 2707 mOutDevice = type; 2708 for (size_t i = 0; i < mEffectChains.size(); i++) { 2709 mEffectChains[i]->setDevice_l(mOutDevice); 2710 } 2711 2712 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2713 status = hwDevice->create_audio_patch(hwDevice, 2714 patch->num_sources, 2715 patch->sources, 2716 patch->num_sinks, 2717 patch->sinks, 2718 handle); 2719 } else { 2720 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2721 } 2722 return status; 2723} 2724 2725status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2726{ 2727 status_t status = NO_ERROR; 2728 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2729 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2730 status = hwDevice->release_audio_patch(hwDevice, handle); 2731 } else { 2732 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2733 } 2734 return status; 2735} 2736 2737void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2738{ 2739 Mutex::Autolock _l(mLock); 2740 mTracks.add(track); 2741} 2742 2743void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2744{ 2745 Mutex::Autolock _l(mLock); 2746 destroyTrack_l(track); 2747} 2748 2749void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2750{ 2751 ThreadBase::getAudioPortConfig(config); 2752 config->role = AUDIO_PORT_ROLE_SOURCE; 2753 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2754 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2755} 2756 2757// ---------------------------------------------------------------------------- 2758 2759AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2760 audio_io_handle_t id, audio_devices_t device, type_t type) 2761 : PlaybackThread(audioFlinger, output, id, device, type), 2762 // mAudioMixer below 2763 // mFastMixer below 2764 mFastMixerFutex(0) 2765 // mOutputSink below 2766 // mPipeSink below 2767 // mNormalSink below 2768{ 2769 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2770 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2771 "mFrameCount=%d, mNormalFrameCount=%d", 2772 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2773 mNormalFrameCount); 2774 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2775 2776 // create an NBAIO sink for the HAL output stream, and negotiate 2777 mOutputSink = new AudioStreamOutSink(output->stream); 2778 size_t numCounterOffers = 0; 2779 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2780 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2781 ALOG_ASSERT(index == 0); 2782 2783 // initialize fast mixer depending on configuration 2784 bool initFastMixer; 2785 switch (kUseFastMixer) { 2786 case FastMixer_Never: 2787 initFastMixer = false; 2788 break; 2789 case FastMixer_Always: 2790 initFastMixer = true; 2791 break; 2792 case FastMixer_Static: 2793 case FastMixer_Dynamic: 2794 initFastMixer = mFrameCount < mNormalFrameCount; 2795 break; 2796 } 2797 if (initFastMixer) { 2798 audio_format_t fastMixerFormat; 2799 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2800 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2801 } else { 2802 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2803 } 2804 if (mFormat != fastMixerFormat) { 2805 // change our Sink format to accept our intermediate precision 2806 mFormat = fastMixerFormat; 2807 free(mSinkBuffer); 2808 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2809 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2810 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2811 } 2812 2813 // create a MonoPipe to connect our submix to FastMixer 2814 NBAIO_Format format = mOutputSink->format(); 2815 // adjust format to match that of the Fast Mixer 2816 format.mFormat = fastMixerFormat; 2817 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2818 2819 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2820 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2821 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2822 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2823 const NBAIO_Format offers[1] = {format}; 2824 size_t numCounterOffers = 0; 2825 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2826 ALOG_ASSERT(index == 0); 2827 monoPipe->setAvgFrames((mScreenState & 1) ? 2828 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2829 mPipeSink = monoPipe; 2830 2831#ifdef TEE_SINK 2832 if (mTeeSinkOutputEnabled) { 2833 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2834 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2835 numCounterOffers = 0; 2836 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2837 ALOG_ASSERT(index == 0); 2838 mTeeSink = teeSink; 2839 PipeReader *teeSource = new PipeReader(*teeSink); 2840 numCounterOffers = 0; 2841 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2842 ALOG_ASSERT(index == 0); 2843 mTeeSource = teeSource; 2844 } 2845#endif 2846 2847 // create fast mixer and configure it initially with just one fast track for our submix 2848 mFastMixer = new FastMixer(); 2849 FastMixerStateQueue *sq = mFastMixer->sq(); 2850#ifdef STATE_QUEUE_DUMP 2851 sq->setObserverDump(&mStateQueueObserverDump); 2852 sq->setMutatorDump(&mStateQueueMutatorDump); 2853#endif 2854 FastMixerState *state = sq->begin(); 2855 FastTrack *fastTrack = &state->mFastTracks[0]; 2856 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2857 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2858 fastTrack->mVolumeProvider = NULL; 2859 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2860 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2861 fastTrack->mGeneration++; 2862 state->mFastTracksGen++; 2863 state->mTrackMask = 1; 2864 // fast mixer will use the HAL output sink 2865 state->mOutputSink = mOutputSink.get(); 2866 state->mOutputSinkGen++; 2867 state->mFrameCount = mFrameCount; 2868 state->mCommand = FastMixerState::COLD_IDLE; 2869 // already done in constructor initialization list 2870 //mFastMixerFutex = 0; 2871 state->mColdFutexAddr = &mFastMixerFutex; 2872 state->mColdGen++; 2873 state->mDumpState = &mFastMixerDumpState; 2874#ifdef TEE_SINK 2875 state->mTeeSink = mTeeSink.get(); 2876#endif 2877 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2878 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2879 sq->end(); 2880 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2881 2882 // start the fast mixer 2883 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2884 pid_t tid = mFastMixer->getTid(); 2885 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2886 if (err != 0) { 2887 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2888 kPriorityFastMixer, getpid_cached, tid, err); 2889 } 2890 2891#ifdef AUDIO_WATCHDOG 2892 // create and start the watchdog 2893 mAudioWatchdog = new AudioWatchdog(); 2894 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2895 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2896 tid = mAudioWatchdog->getTid(); 2897 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2898 if (err != 0) { 2899 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2900 kPriorityFastMixer, getpid_cached, tid, err); 2901 } 2902#endif 2903 2904 } 2905 2906 switch (kUseFastMixer) { 2907 case FastMixer_Never: 2908 case FastMixer_Dynamic: 2909 mNormalSink = mOutputSink; 2910 break; 2911 case FastMixer_Always: 2912 mNormalSink = mPipeSink; 2913 break; 2914 case FastMixer_Static: 2915 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2916 break; 2917 } 2918} 2919 2920AudioFlinger::MixerThread::~MixerThread() 2921{ 2922 if (mFastMixer != 0) { 2923 FastMixerStateQueue *sq = mFastMixer->sq(); 2924 FastMixerState *state = sq->begin(); 2925 if (state->mCommand == FastMixerState::COLD_IDLE) { 2926 int32_t old = android_atomic_inc(&mFastMixerFutex); 2927 if (old == -1) { 2928 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2929 } 2930 } 2931 state->mCommand = FastMixerState::EXIT; 2932 sq->end(); 2933 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2934 mFastMixer->join(); 2935 // Though the fast mixer thread has exited, it's state queue is still valid. 2936 // We'll use that extract the final state which contains one remaining fast track 2937 // corresponding to our sub-mix. 2938 state = sq->begin(); 2939 ALOG_ASSERT(state->mTrackMask == 1); 2940 FastTrack *fastTrack = &state->mFastTracks[0]; 2941 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2942 delete fastTrack->mBufferProvider; 2943 sq->end(false /*didModify*/); 2944 mFastMixer.clear(); 2945#ifdef AUDIO_WATCHDOG 2946 if (mAudioWatchdog != 0) { 2947 mAudioWatchdog->requestExit(); 2948 mAudioWatchdog->requestExitAndWait(); 2949 mAudioWatchdog.clear(); 2950 } 2951#endif 2952 } 2953 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2954 delete mAudioMixer; 2955} 2956 2957 2958uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2959{ 2960 if (mFastMixer != 0) { 2961 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2962 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2963 } 2964 return latency; 2965} 2966 2967 2968void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2969{ 2970 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2971} 2972 2973ssize_t AudioFlinger::MixerThread::threadLoop_write() 2974{ 2975 // FIXME we should only do one push per cycle; confirm this is true 2976 // Start the fast mixer if it's not already running 2977 if (mFastMixer != 0) { 2978 FastMixerStateQueue *sq = mFastMixer->sq(); 2979 FastMixerState *state = sq->begin(); 2980 if (state->mCommand != FastMixerState::MIX_WRITE && 2981 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2982 if (state->mCommand == FastMixerState::COLD_IDLE) { 2983 int32_t old = android_atomic_inc(&mFastMixerFutex); 2984 if (old == -1) { 2985 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2986 } 2987#ifdef AUDIO_WATCHDOG 2988 if (mAudioWatchdog != 0) { 2989 mAudioWatchdog->resume(); 2990 } 2991#endif 2992 } 2993 state->mCommand = FastMixerState::MIX_WRITE; 2994 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2995 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2996 sq->end(); 2997 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2998 if (kUseFastMixer == FastMixer_Dynamic) { 2999 mNormalSink = mPipeSink; 3000 } 3001 } else { 3002 sq->end(false /*didModify*/); 3003 } 3004 } 3005 return PlaybackThread::threadLoop_write(); 3006} 3007 3008void AudioFlinger::MixerThread::threadLoop_standby() 3009{ 3010 // Idle the fast mixer if it's currently running 3011 if (mFastMixer != 0) { 3012 FastMixerStateQueue *sq = mFastMixer->sq(); 3013 FastMixerState *state = sq->begin(); 3014 if (!(state->mCommand & FastMixerState::IDLE)) { 3015 state->mCommand = FastMixerState::COLD_IDLE; 3016 state->mColdFutexAddr = &mFastMixerFutex; 3017 state->mColdGen++; 3018 mFastMixerFutex = 0; 3019 sq->end(); 3020 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3021 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3022 if (kUseFastMixer == FastMixer_Dynamic) { 3023 mNormalSink = mOutputSink; 3024 } 3025#ifdef AUDIO_WATCHDOG 3026 if (mAudioWatchdog != 0) { 3027 mAudioWatchdog->pause(); 3028 } 3029#endif 3030 } else { 3031 sq->end(false /*didModify*/); 3032 } 3033 } 3034 PlaybackThread::threadLoop_standby(); 3035} 3036 3037bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3038{ 3039 return false; 3040} 3041 3042bool AudioFlinger::PlaybackThread::shouldStandby_l() 3043{ 3044 return !mStandby; 3045} 3046 3047bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3048{ 3049 Mutex::Autolock _l(mLock); 3050 return waitingAsyncCallback_l(); 3051} 3052 3053// shared by MIXER and DIRECT, overridden by DUPLICATING 3054void AudioFlinger::PlaybackThread::threadLoop_standby() 3055{ 3056 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3057 mOutput->stream->common.standby(&mOutput->stream->common); 3058 if (mUseAsyncWrite != 0) { 3059 // discard any pending drain or write ack by incrementing sequence 3060 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3061 mDrainSequence = (mDrainSequence + 2) & ~1; 3062 ALOG_ASSERT(mCallbackThread != 0); 3063 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3064 mCallbackThread->setDraining(mDrainSequence); 3065 } 3066} 3067 3068void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3069{ 3070 ALOGV("signal playback thread"); 3071 broadcast_l(); 3072} 3073 3074void AudioFlinger::MixerThread::threadLoop_mix() 3075{ 3076 // obtain the presentation timestamp of the next output buffer 3077 int64_t pts; 3078 status_t status = INVALID_OPERATION; 3079 3080 if (mNormalSink != 0) { 3081 status = mNormalSink->getNextWriteTimestamp(&pts); 3082 } else { 3083 status = mOutputSink->getNextWriteTimestamp(&pts); 3084 } 3085 3086 if (status != NO_ERROR) { 3087 pts = AudioBufferProvider::kInvalidPTS; 3088 } 3089 3090 // mix buffers... 3091 mAudioMixer->process(pts); 3092 mCurrentWriteLength = mSinkBufferSize; 3093 // increase sleep time progressively when application underrun condition clears. 3094 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3095 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3096 // such that we would underrun the audio HAL. 3097 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3098 sleepTimeShift--; 3099 } 3100 sleepTime = 0; 3101 standbyTime = systemTime() + standbyDelay; 3102 //TODO: delay standby when effects have a tail 3103} 3104 3105void AudioFlinger::MixerThread::threadLoop_sleepTime() 3106{ 3107 // If no tracks are ready, sleep once for the duration of an output 3108 // buffer size, then write 0s to the output 3109 if (sleepTime == 0) { 3110 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3111 sleepTime = activeSleepTime >> sleepTimeShift; 3112 if (sleepTime < kMinThreadSleepTimeUs) { 3113 sleepTime = kMinThreadSleepTimeUs; 3114 } 3115 // reduce sleep time in case of consecutive application underruns to avoid 3116 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3117 // duration we would end up writing less data than needed by the audio HAL if 3118 // the condition persists. 3119 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3120 sleepTimeShift++; 3121 } 3122 } else { 3123 sleepTime = idleSleepTime; 3124 } 3125 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3126 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3127 // before effects processing or output. 3128 if (mMixerBufferValid) { 3129 memset(mMixerBuffer, 0, mMixerBufferSize); 3130 } else { 3131 memset(mSinkBuffer, 0, mSinkBufferSize); 3132 } 3133 sleepTime = 0; 3134 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3135 "anticipated start"); 3136 } 3137 // TODO add standby time extension fct of effect tail 3138} 3139 3140// prepareTracks_l() must be called with ThreadBase::mLock held 3141AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3142 Vector< sp<Track> > *tracksToRemove) 3143{ 3144 3145 mixer_state mixerStatus = MIXER_IDLE; 3146 // find out which tracks need to be processed 3147 size_t count = mActiveTracks.size(); 3148 size_t mixedTracks = 0; 3149 size_t tracksWithEffect = 0; 3150 // counts only _active_ fast tracks 3151 size_t fastTracks = 0; 3152 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3153 3154 float masterVolume = mMasterVolume; 3155 bool masterMute = mMasterMute; 3156 3157 if (masterMute) { 3158 masterVolume = 0; 3159 } 3160 // Delegate master volume control to effect in output mix effect chain if needed 3161 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3162 if (chain != 0) { 3163 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3164 chain->setVolume_l(&v, &v); 3165 masterVolume = (float)((v + (1 << 23)) >> 24); 3166 chain.clear(); 3167 } 3168 3169 // prepare a new state to push 3170 FastMixerStateQueue *sq = NULL; 3171 FastMixerState *state = NULL; 3172 bool didModify = false; 3173 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3174 if (mFastMixer != 0) { 3175 sq = mFastMixer->sq(); 3176 state = sq->begin(); 3177 } 3178 3179 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3180 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3181 3182 for (size_t i=0 ; i<count ; i++) { 3183 const sp<Track> t = mActiveTracks[i].promote(); 3184 if (t == 0) { 3185 continue; 3186 } 3187 3188 // this const just means the local variable doesn't change 3189 Track* const track = t.get(); 3190 3191 // process fast tracks 3192 if (track->isFastTrack()) { 3193 3194 // It's theoretically possible (though unlikely) for a fast track to be created 3195 // and then removed within the same normal mix cycle. This is not a problem, as 3196 // the track never becomes active so it's fast mixer slot is never touched. 3197 // The converse, of removing an (active) track and then creating a new track 3198 // at the identical fast mixer slot within the same normal mix cycle, 3199 // is impossible because the slot isn't marked available until the end of each cycle. 3200 int j = track->mFastIndex; 3201 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3202 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3203 FastTrack *fastTrack = &state->mFastTracks[j]; 3204 3205 // Determine whether the track is currently in underrun condition, 3206 // and whether it had a recent underrun. 3207 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3208 FastTrackUnderruns underruns = ftDump->mUnderruns; 3209 uint32_t recentFull = (underruns.mBitFields.mFull - 3210 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3211 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3212 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3213 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3214 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3215 uint32_t recentUnderruns = recentPartial + recentEmpty; 3216 track->mObservedUnderruns = underruns; 3217 // don't count underruns that occur while stopping or pausing 3218 // or stopped which can occur when flush() is called while active 3219 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3220 recentUnderruns > 0) { 3221 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3222 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3223 } 3224 3225 // This is similar to the state machine for normal tracks, 3226 // with a few modifications for fast tracks. 3227 bool isActive = true; 3228 switch (track->mState) { 3229 case TrackBase::STOPPING_1: 3230 // track stays active in STOPPING_1 state until first underrun 3231 if (recentUnderruns > 0 || track->isTerminated()) { 3232 track->mState = TrackBase::STOPPING_2; 3233 } 3234 break; 3235 case TrackBase::PAUSING: 3236 // ramp down is not yet implemented 3237 track->setPaused(); 3238 break; 3239 case TrackBase::RESUMING: 3240 // ramp up is not yet implemented 3241 track->mState = TrackBase::ACTIVE; 3242 break; 3243 case TrackBase::ACTIVE: 3244 if (recentFull > 0 || recentPartial > 0) { 3245 // track has provided at least some frames recently: reset retry count 3246 track->mRetryCount = kMaxTrackRetries; 3247 } 3248 if (recentUnderruns == 0) { 3249 // no recent underruns: stay active 3250 break; 3251 } 3252 // there has recently been an underrun of some kind 3253 if (track->sharedBuffer() == 0) { 3254 // were any of the recent underruns "empty" (no frames available)? 3255 if (recentEmpty == 0) { 3256 // no, then ignore the partial underruns as they are allowed indefinitely 3257 break; 3258 } 3259 // there has recently been an "empty" underrun: decrement the retry counter 3260 if (--(track->mRetryCount) > 0) { 3261 break; 3262 } 3263 // indicate to client process that the track was disabled because of underrun; 3264 // it will then automatically call start() when data is available 3265 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3266 // remove from active list, but state remains ACTIVE [confusing but true] 3267 isActive = false; 3268 break; 3269 } 3270 // fall through 3271 case TrackBase::STOPPING_2: 3272 case TrackBase::PAUSED: 3273 case TrackBase::STOPPED: 3274 case TrackBase::FLUSHED: // flush() while active 3275 // Check for presentation complete if track is inactive 3276 // We have consumed all the buffers of this track. 3277 // This would be incomplete if we auto-paused on underrun 3278 { 3279 size_t audioHALFrames = 3280 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3281 size_t framesWritten = mBytesWritten / mFrameSize; 3282 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3283 // track stays in active list until presentation is complete 3284 break; 3285 } 3286 } 3287 if (track->isStopping_2()) { 3288 track->mState = TrackBase::STOPPED; 3289 } 3290 if (track->isStopped()) { 3291 // Can't reset directly, as fast mixer is still polling this track 3292 // track->reset(); 3293 // So instead mark this track as needing to be reset after push with ack 3294 resetMask |= 1 << i; 3295 } 3296 isActive = false; 3297 break; 3298 case TrackBase::IDLE: 3299 default: 3300 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3301 } 3302 3303 if (isActive) { 3304 // was it previously inactive? 3305 if (!(state->mTrackMask & (1 << j))) { 3306 ExtendedAudioBufferProvider *eabp = track; 3307 VolumeProvider *vp = track; 3308 fastTrack->mBufferProvider = eabp; 3309 fastTrack->mVolumeProvider = vp; 3310 fastTrack->mChannelMask = track->mChannelMask; 3311 fastTrack->mFormat = track->mFormat; 3312 fastTrack->mGeneration++; 3313 state->mTrackMask |= 1 << j; 3314 didModify = true; 3315 // no acknowledgement required for newly active tracks 3316 } 3317 // cache the combined master volume and stream type volume for fast mixer; this 3318 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3319 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3320 ++fastTracks; 3321 } else { 3322 // was it previously active? 3323 if (state->mTrackMask & (1 << j)) { 3324 fastTrack->mBufferProvider = NULL; 3325 fastTrack->mGeneration++; 3326 state->mTrackMask &= ~(1 << j); 3327 didModify = true; 3328 // If any fast tracks were removed, we must wait for acknowledgement 3329 // because we're about to decrement the last sp<> on those tracks. 3330 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3331 } else { 3332 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3333 } 3334 tracksToRemove->add(track); 3335 // Avoids a misleading display in dumpsys 3336 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3337 } 3338 continue; 3339 } 3340 3341 { // local variable scope to avoid goto warning 3342 3343 audio_track_cblk_t* cblk = track->cblk(); 3344 3345 // The first time a track is added we wait 3346 // for all its buffers to be filled before processing it 3347 int name = track->name(); 3348 // make sure that we have enough frames to mix one full buffer. 3349 // enforce this condition only once to enable draining the buffer in case the client 3350 // app does not call stop() and relies on underrun to stop: 3351 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3352 // during last round 3353 size_t desiredFrames; 3354 uint32_t sr = track->sampleRate(); 3355 if (sr == mSampleRate) { 3356 desiredFrames = mNormalFrameCount; 3357 } else { 3358 // +1 for rounding and +1 for additional sample needed for interpolation 3359 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3360 // add frames already consumed but not yet released by the resampler 3361 // because mAudioTrackServerProxy->framesReady() will include these frames 3362 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3363#if 0 3364 // the minimum track buffer size is normally twice the number of frames necessary 3365 // to fill one buffer and the resampler should not leave more than one buffer worth 3366 // of unreleased frames after each pass, but just in case... 3367 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3368#endif 3369 } 3370 uint32_t minFrames = 1; 3371 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3372 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3373 minFrames = desiredFrames; 3374 } 3375 3376 size_t framesReady = track->framesReady(); 3377 if ((framesReady >= minFrames) && track->isReady() && 3378 !track->isPaused() && !track->isTerminated()) 3379 { 3380 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3381 3382 mixedTracks++; 3383 3384 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3385 // there is an effect chain connected to the track 3386 chain.clear(); 3387 if (track->mainBuffer() != mSinkBuffer && 3388 track->mainBuffer() != mMixerBuffer) { 3389 if (mEffectBufferEnabled) { 3390 mEffectBufferValid = true; // Later can set directly. 3391 } 3392 chain = getEffectChain_l(track->sessionId()); 3393 // Delegate volume control to effect in track effect chain if needed 3394 if (chain != 0) { 3395 tracksWithEffect++; 3396 } else { 3397 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3398 "session %d", 3399 name, track->sessionId()); 3400 } 3401 } 3402 3403 3404 int param = AudioMixer::VOLUME; 3405 if (track->mFillingUpStatus == Track::FS_FILLED) { 3406 // no ramp for the first volume setting 3407 track->mFillingUpStatus = Track::FS_ACTIVE; 3408 if (track->mState == TrackBase::RESUMING) { 3409 track->mState = TrackBase::ACTIVE; 3410 param = AudioMixer::RAMP_VOLUME; 3411 } 3412 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3413 // FIXME should not make a decision based on mServer 3414 } else if (cblk->mServer != 0) { 3415 // If the track is stopped before the first frame was mixed, 3416 // do not apply ramp 3417 param = AudioMixer::RAMP_VOLUME; 3418 } 3419 3420 // compute volume for this track 3421 uint32_t vl, vr; // in U8.24 integer format 3422 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3423 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3424 vl = vr = 0; 3425 vlf = vrf = vaf = 0.; 3426 if (track->isPausing()) { 3427 track->setPaused(); 3428 } 3429 } else { 3430 3431 // read original volumes with volume control 3432 float typeVolume = mStreamTypes[track->streamType()].volume; 3433 float v = masterVolume * typeVolume; 3434 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3435 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3436 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3437 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3438 // track volumes come from shared memory, so can't be trusted and must be clamped 3439 if (vlf > GAIN_FLOAT_UNITY) { 3440 ALOGV("Track left volume out of range: %.3g", vlf); 3441 vlf = GAIN_FLOAT_UNITY; 3442 } 3443 if (vrf > GAIN_FLOAT_UNITY) { 3444 ALOGV("Track right volume out of range: %.3g", vrf); 3445 vrf = GAIN_FLOAT_UNITY; 3446 } 3447 // now apply the master volume and stream type volume 3448 vlf *= v; 3449 vrf *= v; 3450 // assuming master volume and stream type volume each go up to 1.0, 3451 // then derive vl and vr as U8.24 versions for the effect chain 3452 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3453 vl = (uint32_t) (scaleto8_24 * vlf); 3454 vr = (uint32_t) (scaleto8_24 * vrf); 3455 // vl and vr are now in U8.24 format 3456 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3457 // send level comes from shared memory and so may be corrupt 3458 if (sendLevel > MAX_GAIN_INT) { 3459 ALOGV("Track send level out of range: %04X", sendLevel); 3460 sendLevel = MAX_GAIN_INT; 3461 } 3462 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3463 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3464 } 3465 3466 // Delegate volume control to effect in track effect chain if needed 3467 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3468 // Do not ramp volume if volume is controlled by effect 3469 param = AudioMixer::VOLUME; 3470 // Update remaining floating point volume levels 3471 vlf = (float)vl / (1 << 24); 3472 vrf = (float)vr / (1 << 24); 3473 track->mHasVolumeController = true; 3474 } else { 3475 // force no volume ramp when volume controller was just disabled or removed 3476 // from effect chain to avoid volume spike 3477 if (track->mHasVolumeController) { 3478 param = AudioMixer::VOLUME; 3479 } 3480 track->mHasVolumeController = false; 3481 } 3482 3483 // XXX: these things DON'T need to be done each time 3484 mAudioMixer->setBufferProvider(name, track); 3485 mAudioMixer->enable(name); 3486 3487 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3488 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3489 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3490 mAudioMixer->setParameter( 3491 name, 3492 AudioMixer::TRACK, 3493 AudioMixer::FORMAT, (void *)track->format()); 3494 mAudioMixer->setParameter( 3495 name, 3496 AudioMixer::TRACK, 3497 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3498 mAudioMixer->setParameter( 3499 name, 3500 AudioMixer::TRACK, 3501 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3502 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3503 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3504 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3505 if (reqSampleRate == 0) { 3506 reqSampleRate = mSampleRate; 3507 } else if (reqSampleRate > maxSampleRate) { 3508 reqSampleRate = maxSampleRate; 3509 } 3510 mAudioMixer->setParameter( 3511 name, 3512 AudioMixer::RESAMPLE, 3513 AudioMixer::SAMPLE_RATE, 3514 (void *)(uintptr_t)reqSampleRate); 3515 /* 3516 * Select the appropriate output buffer for the track. 3517 * 3518 * Tracks with effects go into their own effects chain buffer 3519 * and from there into either mEffectBuffer or mSinkBuffer. 3520 * 3521 * Other tracks can use mMixerBuffer for higher precision 3522 * channel accumulation. If this buffer is enabled 3523 * (mMixerBufferEnabled true), then selected tracks will accumulate 3524 * into it. 3525 * 3526 */ 3527 if (mMixerBufferEnabled 3528 && (track->mainBuffer() == mSinkBuffer 3529 || track->mainBuffer() == mMixerBuffer)) { 3530 mAudioMixer->setParameter( 3531 name, 3532 AudioMixer::TRACK, 3533 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3534 mAudioMixer->setParameter( 3535 name, 3536 AudioMixer::TRACK, 3537 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3538 // TODO: override track->mainBuffer()? 3539 mMixerBufferValid = true; 3540 } else { 3541 mAudioMixer->setParameter( 3542 name, 3543 AudioMixer::TRACK, 3544 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3545 mAudioMixer->setParameter( 3546 name, 3547 AudioMixer::TRACK, 3548 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3549 } 3550 mAudioMixer->setParameter( 3551 name, 3552 AudioMixer::TRACK, 3553 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3554 3555 // reset retry count 3556 track->mRetryCount = kMaxTrackRetries; 3557 3558 // If one track is ready, set the mixer ready if: 3559 // - the mixer was not ready during previous round OR 3560 // - no other track is not ready 3561 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3562 mixerStatus != MIXER_TRACKS_ENABLED) { 3563 mixerStatus = MIXER_TRACKS_READY; 3564 } 3565 } else { 3566 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3567 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3568 } 3569 // clear effect chain input buffer if an active track underruns to avoid sending 3570 // previous audio buffer again to effects 3571 chain = getEffectChain_l(track->sessionId()); 3572 if (chain != 0) { 3573 chain->clearInputBuffer(); 3574 } 3575 3576 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3577 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3578 track->isStopped() || track->isPaused()) { 3579 // We have consumed all the buffers of this track. 3580 // Remove it from the list of active tracks. 3581 // TODO: use actual buffer filling status instead of latency when available from 3582 // audio HAL 3583 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3584 size_t framesWritten = mBytesWritten / mFrameSize; 3585 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3586 if (track->isStopped()) { 3587 track->reset(); 3588 } 3589 tracksToRemove->add(track); 3590 } 3591 } else { 3592 // No buffers for this track. Give it a few chances to 3593 // fill a buffer, then remove it from active list. 3594 if (--(track->mRetryCount) <= 0) { 3595 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3596 tracksToRemove->add(track); 3597 // indicate to client process that the track was disabled because of underrun; 3598 // it will then automatically call start() when data is available 3599 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3600 // If one track is not ready, mark the mixer also not ready if: 3601 // - the mixer was ready during previous round OR 3602 // - no other track is ready 3603 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3604 mixerStatus != MIXER_TRACKS_READY) { 3605 mixerStatus = MIXER_TRACKS_ENABLED; 3606 } 3607 } 3608 mAudioMixer->disable(name); 3609 } 3610 3611 } // local variable scope to avoid goto warning 3612track_is_ready: ; 3613 3614 } 3615 3616 // Push the new FastMixer state if necessary 3617 bool pauseAudioWatchdog = false; 3618 if (didModify) { 3619 state->mFastTracksGen++; 3620 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3621 if (kUseFastMixer == FastMixer_Dynamic && 3622 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3623 state->mCommand = FastMixerState::COLD_IDLE; 3624 state->mColdFutexAddr = &mFastMixerFutex; 3625 state->mColdGen++; 3626 mFastMixerFutex = 0; 3627 if (kUseFastMixer == FastMixer_Dynamic) { 3628 mNormalSink = mOutputSink; 3629 } 3630 // If we go into cold idle, need to wait for acknowledgement 3631 // so that fast mixer stops doing I/O. 3632 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3633 pauseAudioWatchdog = true; 3634 } 3635 } 3636 if (sq != NULL) { 3637 sq->end(didModify); 3638 sq->push(block); 3639 } 3640#ifdef AUDIO_WATCHDOG 3641 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3642 mAudioWatchdog->pause(); 3643 } 3644#endif 3645 3646 // Now perform the deferred reset on fast tracks that have stopped 3647 while (resetMask != 0) { 3648 size_t i = __builtin_ctz(resetMask); 3649 ALOG_ASSERT(i < count); 3650 resetMask &= ~(1 << i); 3651 sp<Track> t = mActiveTracks[i].promote(); 3652 if (t == 0) { 3653 continue; 3654 } 3655 Track* track = t.get(); 3656 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3657 track->reset(); 3658 } 3659 3660 // remove all the tracks that need to be... 3661 removeTracks_l(*tracksToRemove); 3662 3663 // sink or mix buffer must be cleared if all tracks are connected to an 3664 // effect chain as in this case the mixer will not write to the sink or mix buffer 3665 // and track effects will accumulate into it 3666 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3667 (mixedTracks == 0 && fastTracks > 0))) { 3668 // FIXME as a performance optimization, should remember previous zero status 3669 if (mMixerBufferValid) { 3670 memset(mMixerBuffer, 0, mMixerBufferSize); 3671 // TODO: In testing, mSinkBuffer below need not be cleared because 3672 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3673 // after mixing. 3674 // 3675 // To enforce this guarantee: 3676 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3677 // (mixedTracks == 0 && fastTracks > 0)) 3678 // must imply MIXER_TRACKS_READY. 3679 // Later, we may clear buffers regardless, and skip much of this logic. 3680 } 3681 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3682 if (mEffectBufferValid) { 3683 memset(mEffectBuffer, 0, mEffectBufferSize); 3684 } 3685 // FIXME as a performance optimization, should remember previous zero status 3686 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3687 } 3688 3689 // if any fast tracks, then status is ready 3690 mMixerStatusIgnoringFastTracks = mixerStatus; 3691 if (fastTracks > 0) { 3692 mixerStatus = MIXER_TRACKS_READY; 3693 } 3694 return mixerStatus; 3695} 3696 3697// getTrackName_l() must be called with ThreadBase::mLock held 3698int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3699 audio_format_t format, int sessionId) 3700{ 3701 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3702} 3703 3704// deleteTrackName_l() must be called with ThreadBase::mLock held 3705void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3706{ 3707 ALOGV("remove track (%d) and delete from mixer", name); 3708 mAudioMixer->deleteTrackName(name); 3709} 3710 3711// checkForNewParameter_l() must be called with ThreadBase::mLock held 3712bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3713 status_t& status) 3714{ 3715 bool reconfig = false; 3716 3717 status = NO_ERROR; 3718 3719 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3720 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3721 if (mFastMixer != 0) { 3722 FastMixerStateQueue *sq = mFastMixer->sq(); 3723 FastMixerState *state = sq->begin(); 3724 if (!(state->mCommand & FastMixerState::IDLE)) { 3725 previousCommand = state->mCommand; 3726 state->mCommand = FastMixerState::HOT_IDLE; 3727 sq->end(); 3728 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3729 } else { 3730 sq->end(false /*didModify*/); 3731 } 3732 } 3733 3734 AudioParameter param = AudioParameter(keyValuePair); 3735 int value; 3736 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3737 reconfig = true; 3738 } 3739 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3740 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3741 status = BAD_VALUE; 3742 } else { 3743 // no need to save value, since it's constant 3744 reconfig = true; 3745 } 3746 } 3747 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3748 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3749 status = BAD_VALUE; 3750 } else { 3751 // no need to save value, since it's constant 3752 reconfig = true; 3753 } 3754 } 3755 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3756 // do not accept frame count changes if tracks are open as the track buffer 3757 // size depends on frame count and correct behavior would not be guaranteed 3758 // if frame count is changed after track creation 3759 if (!mTracks.isEmpty()) { 3760 status = INVALID_OPERATION; 3761 } else { 3762 reconfig = true; 3763 } 3764 } 3765 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3766#ifdef ADD_BATTERY_DATA 3767 // when changing the audio output device, call addBatteryData to notify 3768 // the change 3769 if (mOutDevice != value) { 3770 uint32_t params = 0; 3771 // check whether speaker is on 3772 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3773 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3774 } 3775 3776 audio_devices_t deviceWithoutSpeaker 3777 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3778 // check if any other device (except speaker) is on 3779 if (value & deviceWithoutSpeaker ) { 3780 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3781 } 3782 3783 if (params != 0) { 3784 addBatteryData(params); 3785 } 3786 } 3787#endif 3788 3789 // forward device change to effects that have requested to be 3790 // aware of attached audio device. 3791 if (value != AUDIO_DEVICE_NONE) { 3792 mOutDevice = value; 3793 for (size_t i = 0; i < mEffectChains.size(); i++) { 3794 mEffectChains[i]->setDevice_l(mOutDevice); 3795 } 3796 } 3797 } 3798 3799 if (status == NO_ERROR) { 3800 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3801 keyValuePair.string()); 3802 if (!mStandby && status == INVALID_OPERATION) { 3803 mOutput->stream->common.standby(&mOutput->stream->common); 3804 mStandby = true; 3805 mBytesWritten = 0; 3806 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3807 keyValuePair.string()); 3808 } 3809 if (status == NO_ERROR && reconfig) { 3810 readOutputParameters_l(); 3811 delete mAudioMixer; 3812 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3813 for (size_t i = 0; i < mTracks.size() ; i++) { 3814 int name = getTrackName_l(mTracks[i]->mChannelMask, 3815 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3816 if (name < 0) { 3817 break; 3818 } 3819 mTracks[i]->mName = name; 3820 } 3821 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3822 } 3823 } 3824 3825 if (!(previousCommand & FastMixerState::IDLE)) { 3826 ALOG_ASSERT(mFastMixer != 0); 3827 FastMixerStateQueue *sq = mFastMixer->sq(); 3828 FastMixerState *state = sq->begin(); 3829 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3830 state->mCommand = previousCommand; 3831 sq->end(); 3832 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3833 } 3834 3835 return reconfig; 3836} 3837 3838 3839void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3840{ 3841 const size_t SIZE = 256; 3842 char buffer[SIZE]; 3843 String8 result; 3844 3845 PlaybackThread::dumpInternals(fd, args); 3846 3847 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3848 3849 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3850 const FastMixerDumpState copy(mFastMixerDumpState); 3851 copy.dump(fd); 3852 3853#ifdef STATE_QUEUE_DUMP 3854 // Similar for state queue 3855 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3856 observerCopy.dump(fd); 3857 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3858 mutatorCopy.dump(fd); 3859#endif 3860 3861#ifdef TEE_SINK 3862 // Write the tee output to a .wav file 3863 dumpTee(fd, mTeeSource, mId); 3864#endif 3865 3866#ifdef AUDIO_WATCHDOG 3867 if (mAudioWatchdog != 0) { 3868 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3869 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3870 wdCopy.dump(fd); 3871 } 3872#endif 3873} 3874 3875uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3876{ 3877 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3878} 3879 3880uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3881{ 3882 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3883} 3884 3885void AudioFlinger::MixerThread::cacheParameters_l() 3886{ 3887 PlaybackThread::cacheParameters_l(); 3888 3889 // FIXME: Relaxed timing because of a certain device that can't meet latency 3890 // Should be reduced to 2x after the vendor fixes the driver issue 3891 // increase threshold again due to low power audio mode. The way this warning 3892 // threshold is calculated and its usefulness should be reconsidered anyway. 3893 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3894} 3895 3896// ---------------------------------------------------------------------------- 3897 3898AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3899 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3900 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3901 // mLeftVolFloat, mRightVolFloat 3902{ 3903} 3904 3905AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3906 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3907 ThreadBase::type_t type) 3908 : PlaybackThread(audioFlinger, output, id, device, type) 3909 // mLeftVolFloat, mRightVolFloat 3910{ 3911} 3912 3913AudioFlinger::DirectOutputThread::~DirectOutputThread() 3914{ 3915} 3916 3917void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3918{ 3919 audio_track_cblk_t* cblk = track->cblk(); 3920 float left, right; 3921 3922 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3923 left = right = 0; 3924 } else { 3925 float typeVolume = mStreamTypes[track->streamType()].volume; 3926 float v = mMasterVolume * typeVolume; 3927 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3928 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3929 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3930 if (left > GAIN_FLOAT_UNITY) { 3931 left = GAIN_FLOAT_UNITY; 3932 } 3933 left *= v; 3934 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3935 if (right > GAIN_FLOAT_UNITY) { 3936 right = GAIN_FLOAT_UNITY; 3937 } 3938 right *= v; 3939 } 3940 3941 if (lastTrack) { 3942 if (left != mLeftVolFloat || right != mRightVolFloat) { 3943 mLeftVolFloat = left; 3944 mRightVolFloat = right; 3945 3946 // Convert volumes from float to 8.24 3947 uint32_t vl = (uint32_t)(left * (1 << 24)); 3948 uint32_t vr = (uint32_t)(right * (1 << 24)); 3949 3950 // Delegate volume control to effect in track effect chain if needed 3951 // only one effect chain can be present on DirectOutputThread, so if 3952 // there is one, the track is connected to it 3953 if (!mEffectChains.isEmpty()) { 3954 mEffectChains[0]->setVolume_l(&vl, &vr); 3955 left = (float)vl / (1 << 24); 3956 right = (float)vr / (1 << 24); 3957 } 3958 if (mOutput->stream->set_volume) { 3959 mOutput->stream->set_volume(mOutput->stream, left, right); 3960 } 3961 } 3962 } 3963} 3964 3965 3966AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3967 Vector< sp<Track> > *tracksToRemove 3968) 3969{ 3970 size_t count = mActiveTracks.size(); 3971 mixer_state mixerStatus = MIXER_IDLE; 3972 3973 // find out which tracks need to be processed 3974 for (size_t i = 0; i < count; i++) { 3975 sp<Track> t = mActiveTracks[i].promote(); 3976 // The track died recently 3977 if (t == 0) { 3978 continue; 3979 } 3980 3981 Track* const track = t.get(); 3982 audio_track_cblk_t* cblk = track->cblk(); 3983 // Only consider last track started for volume and mixer state control. 3984 // In theory an older track could underrun and restart after the new one starts 3985 // but as we only care about the transition phase between two tracks on a 3986 // direct output, it is not a problem to ignore the underrun case. 3987 sp<Track> l = mLatestActiveTrack.promote(); 3988 bool last = l.get() == track; 3989 3990 // The first time a track is added we wait 3991 // for all its buffers to be filled before processing it 3992 uint32_t minFrames; 3993 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 3994 minFrames = mNormalFrameCount; 3995 } else { 3996 minFrames = 1; 3997 } 3998 3999 ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ", 4000 minFrames, track->mState, track->framesReady()); 4001 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4002 !track->isStopping_2() && !track->isStopped()) 4003 { 4004 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4005 4006 if (track->mFillingUpStatus == Track::FS_FILLED) { 4007 track->mFillingUpStatus = Track::FS_ACTIVE; 4008 // make sure processVolume_l() will apply new volume even if 0 4009 mLeftVolFloat = mRightVolFloat = -1.0; 4010 if (track->mState == TrackBase::RESUMING) { 4011 track->mState = TrackBase::ACTIVE; 4012 } 4013 } 4014 4015 // compute volume for this track 4016 processVolume_l(track, last); 4017 if (last) { 4018 // reset retry count 4019 track->mRetryCount = kMaxTrackRetriesDirect; 4020 mActiveTrack = t; 4021 mixerStatus = MIXER_TRACKS_READY; 4022 } 4023 } else { 4024 // clear effect chain input buffer if the last active track started underruns 4025 // to avoid sending previous audio buffer again to effects 4026 if (!mEffectChains.isEmpty() && last) { 4027 mEffectChains[0]->clearInputBuffer(); 4028 } 4029 if (track->isStopping_1()) { 4030 track->mState = TrackBase::STOPPING_2; 4031 } 4032 if ((track->sharedBuffer() != 0) || track->isStopped() || 4033 track->isStopping_2() || track->isPaused()) { 4034 // We have consumed all the buffers of this track. 4035 // Remove it from the list of active tracks. 4036 size_t audioHALFrames; 4037 if (audio_is_linear_pcm(mFormat)) { 4038 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4039 } else { 4040 audioHALFrames = 0; 4041 } 4042 4043 size_t framesWritten = mBytesWritten / mFrameSize; 4044 if (mStandby || !last || 4045 track->presentationComplete(framesWritten, audioHALFrames)) { 4046 if (track->isStopping_2()) { 4047 track->mState = TrackBase::STOPPED; 4048 } 4049 if (track->isStopped()) { 4050 track->reset(); 4051 } 4052 tracksToRemove->add(track); 4053 } 4054 } else { 4055 // No buffers for this track. Give it a few chances to 4056 // fill a buffer, then remove it from active list. 4057 // Only consider last track started for mixer state control 4058 if (--(track->mRetryCount) <= 0) { 4059 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4060 tracksToRemove->add(track); 4061 // indicate to client process that the track was disabled because of underrun; 4062 // it will then automatically call start() when data is available 4063 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4064 } else if (last) { 4065 mixerStatus = MIXER_TRACKS_ENABLED; 4066 } 4067 } 4068 } 4069 } 4070 4071 // remove all the tracks that need to be... 4072 removeTracks_l(*tracksToRemove); 4073 4074 return mixerStatus; 4075} 4076 4077void AudioFlinger::DirectOutputThread::threadLoop_mix() 4078{ 4079 size_t frameCount = mFrameCount; 4080 int8_t *curBuf = (int8_t *)mSinkBuffer; 4081 // output audio to hardware 4082 while (frameCount) { 4083 AudioBufferProvider::Buffer buffer; 4084 buffer.frameCount = frameCount; 4085 mActiveTrack->getNextBuffer(&buffer); 4086 if (buffer.raw == NULL) { 4087 memset(curBuf, 0, frameCount * mFrameSize); 4088 break; 4089 } 4090 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4091 frameCount -= buffer.frameCount; 4092 curBuf += buffer.frameCount * mFrameSize; 4093 mActiveTrack->releaseBuffer(&buffer); 4094 } 4095 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4096 sleepTime = 0; 4097 standbyTime = systemTime() + standbyDelay; 4098 mActiveTrack.clear(); 4099} 4100 4101void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4102{ 4103 if (sleepTime == 0) { 4104 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4105 sleepTime = activeSleepTime; 4106 } else { 4107 sleepTime = idleSleepTime; 4108 } 4109 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4110 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4111 sleepTime = 0; 4112 } 4113} 4114 4115// getTrackName_l() must be called with ThreadBase::mLock held 4116int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4117 audio_format_t format __unused, int sessionId __unused) 4118{ 4119 return 0; 4120} 4121 4122// deleteTrackName_l() must be called with ThreadBase::mLock held 4123void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4124{ 4125} 4126 4127// checkForNewParameter_l() must be called with ThreadBase::mLock held 4128bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4129 status_t& status) 4130{ 4131 bool reconfig = false; 4132 4133 status = NO_ERROR; 4134 4135 AudioParameter param = AudioParameter(keyValuePair); 4136 int value; 4137 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4138 // forward device change to effects that have requested to be 4139 // aware of attached audio device. 4140 if (value != AUDIO_DEVICE_NONE) { 4141 mOutDevice = value; 4142 for (size_t i = 0; i < mEffectChains.size(); i++) { 4143 mEffectChains[i]->setDevice_l(mOutDevice); 4144 } 4145 } 4146 } 4147 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4148 // do not accept frame count changes if tracks are open as the track buffer 4149 // size depends on frame count and correct behavior would not be garantied 4150 // if frame count is changed after track creation 4151 if (!mTracks.isEmpty()) { 4152 status = INVALID_OPERATION; 4153 } else { 4154 reconfig = true; 4155 } 4156 } 4157 if (status == NO_ERROR) { 4158 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4159 keyValuePair.string()); 4160 if (!mStandby && status == INVALID_OPERATION) { 4161 mOutput->stream->common.standby(&mOutput->stream->common); 4162 mStandby = true; 4163 mBytesWritten = 0; 4164 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4165 keyValuePair.string()); 4166 } 4167 if (status == NO_ERROR && reconfig) { 4168 readOutputParameters_l(); 4169 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4170 } 4171 } 4172 4173 return reconfig; 4174} 4175 4176uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4177{ 4178 uint32_t time; 4179 if (audio_is_linear_pcm(mFormat)) { 4180 time = PlaybackThread::activeSleepTimeUs(); 4181 } else { 4182 time = 10000; 4183 } 4184 return time; 4185} 4186 4187uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4188{ 4189 uint32_t time; 4190 if (audio_is_linear_pcm(mFormat)) { 4191 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4192 } else { 4193 time = 10000; 4194 } 4195 return time; 4196} 4197 4198uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4199{ 4200 uint32_t time; 4201 if (audio_is_linear_pcm(mFormat)) { 4202 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4203 } else { 4204 time = 10000; 4205 } 4206 return time; 4207} 4208 4209void AudioFlinger::DirectOutputThread::cacheParameters_l() 4210{ 4211 PlaybackThread::cacheParameters_l(); 4212 4213 // use shorter standby delay as on normal output to release 4214 // hardware resources as soon as possible 4215 if (audio_is_linear_pcm(mFormat)) { 4216 standbyDelay = microseconds(activeSleepTime*2); 4217 } else { 4218 standbyDelay = kOffloadStandbyDelayNs; 4219 } 4220} 4221 4222// ---------------------------------------------------------------------------- 4223 4224AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4225 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4226 : Thread(false /*canCallJava*/), 4227 mPlaybackThread(playbackThread), 4228 mWriteAckSequence(0), 4229 mDrainSequence(0) 4230{ 4231} 4232 4233AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4234{ 4235} 4236 4237void AudioFlinger::AsyncCallbackThread::onFirstRef() 4238{ 4239 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4240} 4241 4242bool AudioFlinger::AsyncCallbackThread::threadLoop() 4243{ 4244 while (!exitPending()) { 4245 uint32_t writeAckSequence; 4246 uint32_t drainSequence; 4247 4248 { 4249 Mutex::Autolock _l(mLock); 4250 while (!((mWriteAckSequence & 1) || 4251 (mDrainSequence & 1) || 4252 exitPending())) { 4253 mWaitWorkCV.wait(mLock); 4254 } 4255 4256 if (exitPending()) { 4257 break; 4258 } 4259 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4260 mWriteAckSequence, mDrainSequence); 4261 writeAckSequence = mWriteAckSequence; 4262 mWriteAckSequence &= ~1; 4263 drainSequence = mDrainSequence; 4264 mDrainSequence &= ~1; 4265 } 4266 { 4267 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4268 if (playbackThread != 0) { 4269 if (writeAckSequence & 1) { 4270 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4271 } 4272 if (drainSequence & 1) { 4273 playbackThread->resetDraining(drainSequence >> 1); 4274 } 4275 } 4276 } 4277 } 4278 return false; 4279} 4280 4281void AudioFlinger::AsyncCallbackThread::exit() 4282{ 4283 ALOGV("AsyncCallbackThread::exit"); 4284 Mutex::Autolock _l(mLock); 4285 requestExit(); 4286 mWaitWorkCV.broadcast(); 4287} 4288 4289void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4290{ 4291 Mutex::Autolock _l(mLock); 4292 // bit 0 is cleared 4293 mWriteAckSequence = sequence << 1; 4294} 4295 4296void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4297{ 4298 Mutex::Autolock _l(mLock); 4299 // ignore unexpected callbacks 4300 if (mWriteAckSequence & 2) { 4301 mWriteAckSequence |= 1; 4302 mWaitWorkCV.signal(); 4303 } 4304} 4305 4306void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4307{ 4308 Mutex::Autolock _l(mLock); 4309 // bit 0 is cleared 4310 mDrainSequence = sequence << 1; 4311} 4312 4313void AudioFlinger::AsyncCallbackThread::resetDraining() 4314{ 4315 Mutex::Autolock _l(mLock); 4316 // ignore unexpected callbacks 4317 if (mDrainSequence & 2) { 4318 mDrainSequence |= 1; 4319 mWaitWorkCV.signal(); 4320 } 4321} 4322 4323 4324// ---------------------------------------------------------------------------- 4325AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4326 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4327 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4328 mHwPaused(false), 4329 mFlushPending(false), 4330 mPausedBytesRemaining(0) 4331{ 4332 //FIXME: mStandby should be set to true by ThreadBase constructor 4333 mStandby = true; 4334} 4335 4336void AudioFlinger::OffloadThread::threadLoop_exit() 4337{ 4338 if (mFlushPending || mHwPaused) { 4339 // If a flush is pending or track was paused, just discard buffered data 4340 flushHw_l(); 4341 } else { 4342 mMixerStatus = MIXER_DRAIN_ALL; 4343 threadLoop_drain(); 4344 } 4345 if (mUseAsyncWrite) { 4346 ALOG_ASSERT(mCallbackThread != 0); 4347 mCallbackThread->exit(); 4348 } 4349 PlaybackThread::threadLoop_exit(); 4350} 4351 4352AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4353 Vector< sp<Track> > *tracksToRemove 4354) 4355{ 4356 size_t count = mActiveTracks.size(); 4357 4358 mixer_state mixerStatus = MIXER_IDLE; 4359 bool doHwPause = false; 4360 bool doHwResume = false; 4361 4362 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4363 4364 // find out which tracks need to be processed 4365 for (size_t i = 0; i < count; i++) { 4366 sp<Track> t = mActiveTracks[i].promote(); 4367 // The track died recently 4368 if (t == 0) { 4369 continue; 4370 } 4371 Track* const track = t.get(); 4372 audio_track_cblk_t* cblk = track->cblk(); 4373 // Only consider last track started for volume and mixer state control. 4374 // In theory an older track could underrun and restart after the new one starts 4375 // but as we only care about the transition phase between two tracks on a 4376 // direct output, it is not a problem to ignore the underrun case. 4377 sp<Track> l = mLatestActiveTrack.promote(); 4378 bool last = l.get() == track; 4379 4380 if (track->isInvalid()) { 4381 ALOGW("An invalidated track shouldn't be in active list"); 4382 tracksToRemove->add(track); 4383 continue; 4384 } 4385 4386 if (track->mState == TrackBase::IDLE) { 4387 ALOGW("An idle track shouldn't be in active list"); 4388 continue; 4389 } 4390 4391 if (track->isPausing()) { 4392 track->setPaused(); 4393 if (last) { 4394 if (!mHwPaused) { 4395 doHwPause = true; 4396 mHwPaused = true; 4397 } 4398 // If we were part way through writing the mixbuffer to 4399 // the HAL we must save this until we resume 4400 // BUG - this will be wrong if a different track is made active, 4401 // in that case we want to discard the pending data in the 4402 // mixbuffer and tell the client to present it again when the 4403 // track is resumed 4404 mPausedWriteLength = mCurrentWriteLength; 4405 mPausedBytesRemaining = mBytesRemaining; 4406 mBytesRemaining = 0; // stop writing 4407 } 4408 tracksToRemove->add(track); 4409 } else if (track->isFlushPending()) { 4410 track->flushAck(); 4411 if (last) { 4412 mFlushPending = true; 4413 } 4414 } else if (track->isResumePending()){ 4415 track->resumeAck(); 4416 if (last) { 4417 if (mPausedBytesRemaining) { 4418 // Need to continue write that was interrupted 4419 mCurrentWriteLength = mPausedWriteLength; 4420 mBytesRemaining = mPausedBytesRemaining; 4421 mPausedBytesRemaining = 0; 4422 } 4423 if (mHwPaused) { 4424 doHwResume = true; 4425 mHwPaused = false; 4426 // threadLoop_mix() will handle the case that we need to 4427 // resume an interrupted write 4428 } 4429 // enable write to audio HAL 4430 sleepTime = 0; 4431 4432 // Do not handle new data in this iteration even if track->framesReady() 4433 mixerStatus = MIXER_TRACKS_ENABLED; 4434 } 4435 } else if (track->framesReady() && track->isReady() && 4436 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4437 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4438 if (track->mFillingUpStatus == Track::FS_FILLED) { 4439 track->mFillingUpStatus = Track::FS_ACTIVE; 4440 // make sure processVolume_l() will apply new volume even if 0 4441 mLeftVolFloat = mRightVolFloat = -1.0; 4442 } 4443 4444 if (last) { 4445 sp<Track> previousTrack = mPreviousTrack.promote(); 4446 if (previousTrack != 0) { 4447 if (track != previousTrack.get()) { 4448 // Flush any data still being written from last track 4449 mBytesRemaining = 0; 4450 if (mPausedBytesRemaining) { 4451 // Last track was paused so we also need to flush saved 4452 // mixbuffer state and invalidate track so that it will 4453 // re-submit that unwritten data when it is next resumed 4454 mPausedBytesRemaining = 0; 4455 // Invalidate is a bit drastic - would be more efficient 4456 // to have a flag to tell client that some of the 4457 // previously written data was lost 4458 previousTrack->invalidate(); 4459 } 4460 // flush data already sent to the DSP if changing audio session as audio 4461 // comes from a different source. Also invalidate previous track to force a 4462 // seek when resuming. 4463 if (previousTrack->sessionId() != track->sessionId()) { 4464 previousTrack->invalidate(); 4465 } 4466 } 4467 } 4468 mPreviousTrack = track; 4469 // reset retry count 4470 track->mRetryCount = kMaxTrackRetriesOffload; 4471 mActiveTrack = t; 4472 mixerStatus = MIXER_TRACKS_READY; 4473 } 4474 } else { 4475 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4476 if (track->isStopping_1()) { 4477 // Hardware buffer can hold a large amount of audio so we must 4478 // wait for all current track's data to drain before we say 4479 // that the track is stopped. 4480 if (mBytesRemaining == 0) { 4481 // Only start draining when all data in mixbuffer 4482 // has been written 4483 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4484 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4485 // do not drain if no data was ever sent to HAL (mStandby == true) 4486 if (last && !mStandby) { 4487 // do not modify drain sequence if we are already draining. This happens 4488 // when resuming from pause after drain. 4489 if ((mDrainSequence & 1) == 0) { 4490 sleepTime = 0; 4491 standbyTime = systemTime() + standbyDelay; 4492 mixerStatus = MIXER_DRAIN_TRACK; 4493 mDrainSequence += 2; 4494 } 4495 if (mHwPaused) { 4496 // It is possible to move from PAUSED to STOPPING_1 without 4497 // a resume so we must ensure hardware is running 4498 doHwResume = true; 4499 mHwPaused = false; 4500 } 4501 } 4502 } 4503 } else if (track->isStopping_2()) { 4504 // Drain has completed or we are in standby, signal presentation complete 4505 if (!(mDrainSequence & 1) || !last || mStandby) { 4506 track->mState = TrackBase::STOPPED; 4507 size_t audioHALFrames = 4508 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4509 size_t framesWritten = 4510 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4511 track->presentationComplete(framesWritten, audioHALFrames); 4512 track->reset(); 4513 tracksToRemove->add(track); 4514 } 4515 } else { 4516 // No buffers for this track. Give it a few chances to 4517 // fill a buffer, then remove it from active list. 4518 if (--(track->mRetryCount) <= 0) { 4519 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4520 track->name()); 4521 tracksToRemove->add(track); 4522 // indicate to client process that the track was disabled because of underrun; 4523 // it will then automatically call start() when data is available 4524 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4525 } else if (last){ 4526 mixerStatus = MIXER_TRACKS_ENABLED; 4527 } 4528 } 4529 } 4530 // compute volume for this track 4531 processVolume_l(track, last); 4532 } 4533 4534 // make sure the pause/flush/resume sequence is executed in the right order. 4535 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4536 // before flush and then resume HW. This can happen in case of pause/flush/resume 4537 // if resume is received before pause is executed. 4538 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4539 mOutput->stream->pause(mOutput->stream); 4540 } 4541 if (mFlushPending) { 4542 flushHw_l(); 4543 mFlushPending = false; 4544 } 4545 if (!mStandby && doHwResume) { 4546 mOutput->stream->resume(mOutput->stream); 4547 } 4548 4549 // remove all the tracks that need to be... 4550 removeTracks_l(*tracksToRemove); 4551 4552 return mixerStatus; 4553} 4554 4555// must be called with thread mutex locked 4556bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4557{ 4558 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4559 mWriteAckSequence, mDrainSequence); 4560 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4561 return true; 4562 } 4563 return false; 4564} 4565 4566// must be called with thread mutex locked 4567bool AudioFlinger::OffloadThread::shouldStandby_l() 4568{ 4569 bool trackPaused = false; 4570 4571 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4572 // after a timeout and we will enter standby then. 4573 if (mTracks.size() > 0) { 4574 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4575 } 4576 4577 return !mStandby && !trackPaused; 4578} 4579 4580 4581bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4582{ 4583 Mutex::Autolock _l(mLock); 4584 return waitingAsyncCallback_l(); 4585} 4586 4587void AudioFlinger::OffloadThread::flushHw_l() 4588{ 4589 mOutput->stream->flush(mOutput->stream); 4590 // Flush anything still waiting in the mixbuffer 4591 mCurrentWriteLength = 0; 4592 mBytesRemaining = 0; 4593 mPausedWriteLength = 0; 4594 mPausedBytesRemaining = 0; 4595 mHwPaused = false; 4596 4597 if (mUseAsyncWrite) { 4598 // discard any pending drain or write ack by incrementing sequence 4599 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4600 mDrainSequence = (mDrainSequence + 2) & ~1; 4601 ALOG_ASSERT(mCallbackThread != 0); 4602 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4603 mCallbackThread->setDraining(mDrainSequence); 4604 } 4605} 4606 4607void AudioFlinger::OffloadThread::onAddNewTrack_l() 4608{ 4609 sp<Track> previousTrack = mPreviousTrack.promote(); 4610 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4611 4612 if (previousTrack != 0 && latestTrack != 0 && 4613 (previousTrack->sessionId() != latestTrack->sessionId())) { 4614 mFlushPending = true; 4615 } 4616 PlaybackThread::onAddNewTrack_l(); 4617} 4618 4619// ---------------------------------------------------------------------------- 4620 4621AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4622 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4623 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4624 DUPLICATING), 4625 mWaitTimeMs(UINT_MAX) 4626{ 4627 addOutputTrack(mainThread); 4628} 4629 4630AudioFlinger::DuplicatingThread::~DuplicatingThread() 4631{ 4632 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4633 mOutputTracks[i]->destroy(); 4634 } 4635} 4636 4637void AudioFlinger::DuplicatingThread::threadLoop_mix() 4638{ 4639 // mix buffers... 4640 if (outputsReady(outputTracks)) { 4641 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4642 } else { 4643 memset(mSinkBuffer, 0, mSinkBufferSize); 4644 } 4645 sleepTime = 0; 4646 writeFrames = mNormalFrameCount; 4647 mCurrentWriteLength = mSinkBufferSize; 4648 standbyTime = systemTime() + standbyDelay; 4649} 4650 4651void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4652{ 4653 if (sleepTime == 0) { 4654 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4655 sleepTime = activeSleepTime; 4656 } else { 4657 sleepTime = idleSleepTime; 4658 } 4659 } else if (mBytesWritten != 0) { 4660 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4661 writeFrames = mNormalFrameCount; 4662 memset(mSinkBuffer, 0, mSinkBufferSize); 4663 } else { 4664 // flush remaining overflow buffers in output tracks 4665 writeFrames = 0; 4666 } 4667 sleepTime = 0; 4668 } 4669} 4670 4671ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4672{ 4673 for (size_t i = 0; i < outputTracks.size(); i++) { 4674 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4675 // for delivery downstream as needed. This in-place conversion is safe as 4676 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4677 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4678 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4679 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4680 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4681 } 4682 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4683 } 4684 mStandby = false; 4685 return (ssize_t)mSinkBufferSize; 4686} 4687 4688void AudioFlinger::DuplicatingThread::threadLoop_standby() 4689{ 4690 // DuplicatingThread implements standby by stopping all tracks 4691 for (size_t i = 0; i < outputTracks.size(); i++) { 4692 outputTracks[i]->stop(); 4693 } 4694} 4695 4696void AudioFlinger::DuplicatingThread::saveOutputTracks() 4697{ 4698 outputTracks = mOutputTracks; 4699} 4700 4701void AudioFlinger::DuplicatingThread::clearOutputTracks() 4702{ 4703 outputTracks.clear(); 4704} 4705 4706void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4707{ 4708 Mutex::Autolock _l(mLock); 4709 // FIXME explain this formula 4710 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4711 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4712 // due to current usage case and restrictions on the AudioBufferProvider. 4713 // Actual buffer conversion is done in threadLoop_write(). 4714 // 4715 // TODO: This may change in the future, depending on multichannel 4716 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4717 OutputTrack *outputTrack = new OutputTrack(thread, 4718 this, 4719 mSampleRate, 4720 AUDIO_FORMAT_PCM_16_BIT, 4721 mChannelMask, 4722 frameCount, 4723 IPCThreadState::self()->getCallingUid()); 4724 if (outputTrack->cblk() != NULL) { 4725 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4726 mOutputTracks.add(outputTrack); 4727 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4728 updateWaitTime_l(); 4729 } 4730} 4731 4732void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4733{ 4734 Mutex::Autolock _l(mLock); 4735 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4736 if (mOutputTracks[i]->thread() == thread) { 4737 mOutputTracks[i]->destroy(); 4738 mOutputTracks.removeAt(i); 4739 updateWaitTime_l(); 4740 return; 4741 } 4742 } 4743 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4744} 4745 4746// caller must hold mLock 4747void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4748{ 4749 mWaitTimeMs = UINT_MAX; 4750 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4751 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4752 if (strong != 0) { 4753 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4754 if (waitTimeMs < mWaitTimeMs) { 4755 mWaitTimeMs = waitTimeMs; 4756 } 4757 } 4758 } 4759} 4760 4761 4762bool AudioFlinger::DuplicatingThread::outputsReady( 4763 const SortedVector< sp<OutputTrack> > &outputTracks) 4764{ 4765 for (size_t i = 0; i < outputTracks.size(); i++) { 4766 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4767 if (thread == 0) { 4768 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4769 outputTracks[i].get()); 4770 return false; 4771 } 4772 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4773 // see note at standby() declaration 4774 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4775 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4776 thread.get()); 4777 return false; 4778 } 4779 } 4780 return true; 4781} 4782 4783uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4784{ 4785 return (mWaitTimeMs * 1000) / 2; 4786} 4787 4788void AudioFlinger::DuplicatingThread::cacheParameters_l() 4789{ 4790 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4791 updateWaitTime_l(); 4792 4793 MixerThread::cacheParameters_l(); 4794} 4795 4796// ---------------------------------------------------------------------------- 4797// Record 4798// ---------------------------------------------------------------------------- 4799 4800AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4801 AudioStreamIn *input, 4802 audio_io_handle_t id, 4803 audio_devices_t outDevice, 4804 audio_devices_t inDevice 4805#ifdef TEE_SINK 4806 , const sp<NBAIO_Sink>& teeSink 4807#endif 4808 ) : 4809 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4810 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4811 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4812 mRsmpInRear(0) 4813#ifdef TEE_SINK 4814 , mTeeSink(teeSink) 4815#endif 4816 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4817 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4818 // mFastCapture below 4819 , mFastCaptureFutex(0) 4820 // mInputSource 4821 // mPipeSink 4822 // mPipeSource 4823 , mPipeFramesP2(0) 4824 // mPipeMemory 4825 // mFastCaptureNBLogWriter 4826 , mFastTrackAvail(false) 4827{ 4828 snprintf(mName, kNameLength, "AudioIn_%X", id); 4829 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4830 4831 readInputParameters_l(); 4832 4833 // create an NBAIO source for the HAL input stream, and negotiate 4834 mInputSource = new AudioStreamInSource(input->stream); 4835 size_t numCounterOffers = 0; 4836 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4837 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4838 ALOG_ASSERT(index == 0); 4839 4840 // initialize fast capture depending on configuration 4841 bool initFastCapture; 4842 switch (kUseFastCapture) { 4843 case FastCapture_Never: 4844 initFastCapture = false; 4845 break; 4846 case FastCapture_Always: 4847 initFastCapture = true; 4848 break; 4849 case FastCapture_Static: 4850 uint32_t primaryOutputSampleRate; 4851 { 4852 AutoMutex _l(audioFlinger->mHardwareLock); 4853 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4854 } 4855 initFastCapture = 4856 // either capture sample rate is same as (a reasonable) primary output sample rate 4857 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4858 (mSampleRate == primaryOutputSampleRate)) || 4859 // or primary output sample rate is unknown, and capture sample rate is reasonable 4860 ((primaryOutputSampleRate == 0) && 4861 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4862 // and the buffer size is < 12 ms 4863 (mFrameCount * 1000) / mSampleRate < 12; 4864 break; 4865 // case FastCapture_Dynamic: 4866 } 4867 4868 if (initFastCapture) { 4869 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4870 NBAIO_Format format = mInputSource->format(); 4871 size_t pipeFramesP2 = roundup(mFrameCount * 8); 4872 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4873 void *pipeBuffer; 4874 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4875 sp<IMemory> pipeMemory; 4876 if ((roHeap == 0) || 4877 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4878 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4879 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4880 goto failed; 4881 } 4882 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4883 memset(pipeBuffer, 0, pipeSize); 4884 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4885 const NBAIO_Format offers[1] = {format}; 4886 size_t numCounterOffers = 0; 4887 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4888 ALOG_ASSERT(index == 0); 4889 mPipeSink = pipe; 4890 PipeReader *pipeReader = new PipeReader(*pipe); 4891 numCounterOffers = 0; 4892 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4893 ALOG_ASSERT(index == 0); 4894 mPipeSource = pipeReader; 4895 mPipeFramesP2 = pipeFramesP2; 4896 mPipeMemory = pipeMemory; 4897 4898 // create fast capture 4899 mFastCapture = new FastCapture(); 4900 FastCaptureStateQueue *sq = mFastCapture->sq(); 4901#ifdef STATE_QUEUE_DUMP 4902 // FIXME 4903#endif 4904 FastCaptureState *state = sq->begin(); 4905 state->mCblk = NULL; 4906 state->mInputSource = mInputSource.get(); 4907 state->mInputSourceGen++; 4908 state->mPipeSink = pipe; 4909 state->mPipeSinkGen++; 4910 state->mFrameCount = mFrameCount; 4911 state->mCommand = FastCaptureState::COLD_IDLE; 4912 // already done in constructor initialization list 4913 //mFastCaptureFutex = 0; 4914 state->mColdFutexAddr = &mFastCaptureFutex; 4915 state->mColdGen++; 4916 state->mDumpState = &mFastCaptureDumpState; 4917#ifdef TEE_SINK 4918 // FIXME 4919#endif 4920 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4921 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4922 sq->end(); 4923 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4924 4925 // start the fast capture 4926 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4927 pid_t tid = mFastCapture->getTid(); 4928 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4929 if (err != 0) { 4930 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4931 kPriorityFastCapture, getpid_cached, tid, err); 4932 } 4933 4934#ifdef AUDIO_WATCHDOG 4935 // FIXME 4936#endif 4937 4938 mFastTrackAvail = true; 4939 } 4940failed: ; 4941 4942 // FIXME mNormalSource 4943} 4944 4945 4946AudioFlinger::RecordThread::~RecordThread() 4947{ 4948 if (mFastCapture != 0) { 4949 FastCaptureStateQueue *sq = mFastCapture->sq(); 4950 FastCaptureState *state = sq->begin(); 4951 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4952 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4953 if (old == -1) { 4954 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4955 } 4956 } 4957 state->mCommand = FastCaptureState::EXIT; 4958 sq->end(); 4959 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4960 mFastCapture->join(); 4961 mFastCapture.clear(); 4962 } 4963 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4964 mAudioFlinger->unregisterWriter(mNBLogWriter); 4965 delete[] mRsmpInBuffer; 4966} 4967 4968void AudioFlinger::RecordThread::onFirstRef() 4969{ 4970 run(mName, PRIORITY_URGENT_AUDIO); 4971} 4972 4973bool AudioFlinger::RecordThread::threadLoop() 4974{ 4975 nsecs_t lastWarning = 0; 4976 4977 inputStandBy(); 4978 4979reacquire_wakelock: 4980 sp<RecordTrack> activeTrack; 4981 int activeTracksGen; 4982 { 4983 Mutex::Autolock _l(mLock); 4984 size_t size = mActiveTracks.size(); 4985 activeTracksGen = mActiveTracksGen; 4986 if (size > 0) { 4987 // FIXME an arbitrary choice 4988 activeTrack = mActiveTracks[0]; 4989 acquireWakeLock_l(activeTrack->uid()); 4990 if (size > 1) { 4991 SortedVector<int> tmp; 4992 for (size_t i = 0; i < size; i++) { 4993 tmp.add(mActiveTracks[i]->uid()); 4994 } 4995 updateWakeLockUids_l(tmp); 4996 } 4997 } else { 4998 acquireWakeLock_l(-1); 4999 } 5000 } 5001 5002 // used to request a deferred sleep, to be executed later while mutex is unlocked 5003 uint32_t sleepUs = 0; 5004 5005 // loop while there is work to do 5006 for (;;) { 5007 Vector< sp<EffectChain> > effectChains; 5008 5009 // sleep with mutex unlocked 5010 if (sleepUs > 0) { 5011 usleep(sleepUs); 5012 sleepUs = 0; 5013 } 5014 5015 // activeTracks accumulates a copy of a subset of mActiveTracks 5016 Vector< sp<RecordTrack> > activeTracks; 5017 5018 // reference to the (first and only) fast track 5019 sp<RecordTrack> fastTrack; 5020 5021 { // scope for mLock 5022 Mutex::Autolock _l(mLock); 5023 5024 processConfigEvents_l(); 5025 5026 // check exitPending here because checkForNewParameters_l() and 5027 // checkForNewParameters_l() can temporarily release mLock 5028 if (exitPending()) { 5029 break; 5030 } 5031 5032 // if no active track(s), then standby and release wakelock 5033 size_t size = mActiveTracks.size(); 5034 if (size == 0) { 5035 standbyIfNotAlreadyInStandby(); 5036 // exitPending() can't become true here 5037 releaseWakeLock_l(); 5038 ALOGV("RecordThread: loop stopping"); 5039 // go to sleep 5040 mWaitWorkCV.wait(mLock); 5041 ALOGV("RecordThread: loop starting"); 5042 goto reacquire_wakelock; 5043 } 5044 5045 if (mActiveTracksGen != activeTracksGen) { 5046 activeTracksGen = mActiveTracksGen; 5047 SortedVector<int> tmp; 5048 for (size_t i = 0; i < size; i++) { 5049 tmp.add(mActiveTracks[i]->uid()); 5050 } 5051 updateWakeLockUids_l(tmp); 5052 } 5053 5054 bool doBroadcast = false; 5055 for (size_t i = 0; i < size; ) { 5056 5057 activeTrack = mActiveTracks[i]; 5058 if (activeTrack->isTerminated()) { 5059 removeTrack_l(activeTrack); 5060 mActiveTracks.remove(activeTrack); 5061 mActiveTracksGen++; 5062 size--; 5063 continue; 5064 } 5065 5066 TrackBase::track_state activeTrackState = activeTrack->mState; 5067 switch (activeTrackState) { 5068 5069 case TrackBase::PAUSING: 5070 mActiveTracks.remove(activeTrack); 5071 mActiveTracksGen++; 5072 doBroadcast = true; 5073 size--; 5074 continue; 5075 5076 case TrackBase::STARTING_1: 5077 sleepUs = 10000; 5078 i++; 5079 continue; 5080 5081 case TrackBase::STARTING_2: 5082 doBroadcast = true; 5083 mStandby = false; 5084 activeTrack->mState = TrackBase::ACTIVE; 5085 break; 5086 5087 case TrackBase::ACTIVE: 5088 break; 5089 5090 case TrackBase::IDLE: 5091 i++; 5092 continue; 5093 5094 default: 5095 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5096 } 5097 5098 activeTracks.add(activeTrack); 5099 i++; 5100 5101 if (activeTrack->isFastTrack()) { 5102 ALOG_ASSERT(!mFastTrackAvail); 5103 ALOG_ASSERT(fastTrack == 0); 5104 fastTrack = activeTrack; 5105 } 5106 } 5107 if (doBroadcast) { 5108 mStartStopCond.broadcast(); 5109 } 5110 5111 // sleep if there are no active tracks to process 5112 if (activeTracks.size() == 0) { 5113 if (sleepUs == 0) { 5114 sleepUs = kRecordThreadSleepUs; 5115 } 5116 continue; 5117 } 5118 sleepUs = 0; 5119 5120 lockEffectChains_l(effectChains); 5121 } 5122 5123 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5124 5125 size_t size = effectChains.size(); 5126 for (size_t i = 0; i < size; i++) { 5127 // thread mutex is not locked, but effect chain is locked 5128 effectChains[i]->process_l(); 5129 } 5130 5131 // Start the fast capture if it's not already running 5132 if (mFastCapture != 0) { 5133 FastCaptureStateQueue *sq = mFastCapture->sq(); 5134 FastCaptureState *state = sq->begin(); 5135 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5136 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5137 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5138 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5139 if (old == -1) { 5140 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5141 } 5142 } 5143 state->mCommand = FastCaptureState::READ_WRITE; 5144#if 0 // FIXME 5145 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5146 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5147#endif 5148 state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL; 5149 sq->end(); 5150 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5151#if 0 5152 if (kUseFastCapture == FastCapture_Dynamic) { 5153 mNormalSource = mPipeSource; 5154 } 5155#endif 5156 } else { 5157 sq->end(false /*didModify*/); 5158 } 5159 } 5160 5161 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5162 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5163 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5164 // If destination is non-contiguous, first read past the nominal end of buffer, then 5165 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5166 5167 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5168 ssize_t framesRead; 5169 5170 // If an NBAIO source is present, use it to read the normal capture's data 5171 if (mPipeSource != 0) { 5172 size_t framesToRead = mBufferSize / mFrameSize; 5173 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5174 framesToRead, AudioBufferProvider::kInvalidPTS); 5175 if (framesRead == 0) { 5176 // since pipe is non-blocking, simulate blocking input 5177 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5178 } 5179 // otherwise use the HAL / AudioStreamIn directly 5180 } else { 5181 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5182 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5183 if (bytesRead < 0) { 5184 framesRead = bytesRead; 5185 } else { 5186 framesRead = bytesRead / mFrameSize; 5187 } 5188 } 5189 5190 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5191 ALOGE("read failed: framesRead=%d", framesRead); 5192 // Force input into standby so that it tries to recover at next read attempt 5193 inputStandBy(); 5194 sleepUs = kRecordThreadSleepUs; 5195 } 5196 if (framesRead <= 0) { 5197 goto unlock; 5198 } 5199 ALOG_ASSERT(framesRead > 0); 5200 5201 if (mTeeSink != 0) { 5202 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5203 } 5204 // If destination is non-contiguous, we now correct for reading past end of buffer. 5205 { 5206 size_t part1 = mRsmpInFramesP2 - rear; 5207 if ((size_t) framesRead > part1) { 5208 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5209 (framesRead - part1) * mFrameSize); 5210 } 5211 } 5212 rear = mRsmpInRear += framesRead; 5213 5214 size = activeTracks.size(); 5215 // loop over each active track 5216 for (size_t i = 0; i < size; i++) { 5217 activeTrack = activeTracks[i]; 5218 5219 // skip fast tracks, as those are handled directly by FastCapture 5220 if (activeTrack->isFastTrack()) { 5221 continue; 5222 } 5223 5224 enum { 5225 OVERRUN_UNKNOWN, 5226 OVERRUN_TRUE, 5227 OVERRUN_FALSE 5228 } overrun = OVERRUN_UNKNOWN; 5229 5230 // loop over getNextBuffer to handle circular sink 5231 for (;;) { 5232 5233 activeTrack->mSink.frameCount = ~0; 5234 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5235 size_t framesOut = activeTrack->mSink.frameCount; 5236 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5237 5238 int32_t front = activeTrack->mRsmpInFront; 5239 ssize_t filled = rear - front; 5240 size_t framesIn; 5241 5242 if (filled < 0) { 5243 // should not happen, but treat like a massive overrun and re-sync 5244 framesIn = 0; 5245 activeTrack->mRsmpInFront = rear; 5246 overrun = OVERRUN_TRUE; 5247 } else if ((size_t) filled <= mRsmpInFrames) { 5248 framesIn = (size_t) filled; 5249 } else { 5250 // client is not keeping up with server, but give it latest data 5251 framesIn = mRsmpInFrames; 5252 activeTrack->mRsmpInFront = front = rear - framesIn; 5253 overrun = OVERRUN_TRUE; 5254 } 5255 5256 if (framesOut == 0 || framesIn == 0) { 5257 break; 5258 } 5259 5260 if (activeTrack->mResampler == NULL) { 5261 // no resampling 5262 if (framesIn > framesOut) { 5263 framesIn = framesOut; 5264 } else { 5265 framesOut = framesIn; 5266 } 5267 int8_t *dst = activeTrack->mSink.i8; 5268 while (framesIn > 0) { 5269 front &= mRsmpInFramesP2 - 1; 5270 size_t part1 = mRsmpInFramesP2 - front; 5271 if (part1 > framesIn) { 5272 part1 = framesIn; 5273 } 5274 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5275 if (mChannelCount == activeTrack->mChannelCount) { 5276 memcpy(dst, src, part1 * mFrameSize); 5277 } else if (mChannelCount == 1) { 5278 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5279 part1); 5280 } else { 5281 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5282 part1); 5283 } 5284 dst += part1 * activeTrack->mFrameSize; 5285 front += part1; 5286 framesIn -= part1; 5287 } 5288 activeTrack->mRsmpInFront += framesOut; 5289 5290 } else { 5291 // resampling 5292 // FIXME framesInNeeded should really be part of resampler API, and should 5293 // depend on the SRC ratio 5294 // to keep mRsmpInBuffer full so resampler always has sufficient input 5295 size_t framesInNeeded; 5296 // FIXME only re-calculate when it changes, and optimize for common ratios 5297 // Do not precompute in/out because floating point is not associative 5298 // e.g. a*b/c != a*(b/c). 5299 const double in(mSampleRate); 5300 const double out(activeTrack->mSampleRate); 5301 framesInNeeded = ceil(framesOut * in / out) + 1; 5302 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5303 framesInNeeded, framesOut, in / out); 5304 // Although we theoretically have framesIn in circular buffer, some of those are 5305 // unreleased frames, and thus must be discounted for purpose of budgeting. 5306 size_t unreleased = activeTrack->mRsmpInUnrel; 5307 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5308 if (framesIn < framesInNeeded) { 5309 ALOGV("not enough to resample: have %u frames in but need %u in to " 5310 "produce %u out given in/out ratio of %.4g", 5311 framesIn, framesInNeeded, framesOut, in / out); 5312 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5313 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5314 if (newFramesOut == 0) { 5315 break; 5316 } 5317 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5318 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5319 framesInNeeded, newFramesOut, out / in); 5320 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5321 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5322 "given in/out ratio of %.4g", 5323 framesIn, framesInNeeded, newFramesOut, in / out); 5324 framesOut = newFramesOut; 5325 } else { 5326 ALOGV("success 1: have %u in and need %u in to produce %u out " 5327 "given in/out ratio of %.4g", 5328 framesIn, framesInNeeded, framesOut, in / out); 5329 } 5330 5331 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5332 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5333 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5334 delete[] activeTrack->mRsmpOutBuffer; 5335 // resampler always outputs stereo 5336 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5337 activeTrack->mRsmpOutFrameCount = framesOut; 5338 } 5339 5340 // resampler accumulates, but we only have one source track 5341 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5342 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5343 // FIXME how about having activeTrack implement this interface itself? 5344 activeTrack->mResamplerBufferProvider 5345 /*this*/ /* AudioBufferProvider* */); 5346 // ditherAndClamp() works as long as all buffers returned by 5347 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5348 if (activeTrack->mChannelCount == 1) { 5349 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5350 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5351 framesOut); 5352 // the resampler always outputs stereo samples: 5353 // do post stereo to mono conversion 5354 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5355 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5356 } else { 5357 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5358 activeTrack->mRsmpOutBuffer, framesOut); 5359 } 5360 // now done with mRsmpOutBuffer 5361 5362 } 5363 5364 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5365 overrun = OVERRUN_FALSE; 5366 } 5367 5368 if (activeTrack->mFramesToDrop == 0) { 5369 if (framesOut > 0) { 5370 activeTrack->mSink.frameCount = framesOut; 5371 activeTrack->releaseBuffer(&activeTrack->mSink); 5372 } 5373 } else { 5374 // FIXME could do a partial drop of framesOut 5375 if (activeTrack->mFramesToDrop > 0) { 5376 activeTrack->mFramesToDrop -= framesOut; 5377 if (activeTrack->mFramesToDrop <= 0) { 5378 activeTrack->clearSyncStartEvent(); 5379 } 5380 } else { 5381 activeTrack->mFramesToDrop += framesOut; 5382 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5383 activeTrack->mSyncStartEvent->isCancelled()) { 5384 ALOGW("Synced record %s, session %d, trigger session %d", 5385 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5386 activeTrack->sessionId(), 5387 (activeTrack->mSyncStartEvent != 0) ? 5388 activeTrack->mSyncStartEvent->triggerSession() : 0); 5389 activeTrack->clearSyncStartEvent(); 5390 } 5391 } 5392 } 5393 5394 if (framesOut == 0) { 5395 break; 5396 } 5397 } 5398 5399 switch (overrun) { 5400 case OVERRUN_TRUE: 5401 // client isn't retrieving buffers fast enough 5402 if (!activeTrack->setOverflow()) { 5403 nsecs_t now = systemTime(); 5404 // FIXME should lastWarning per track? 5405 if ((now - lastWarning) > kWarningThrottleNs) { 5406 ALOGW("RecordThread: buffer overflow"); 5407 lastWarning = now; 5408 } 5409 } 5410 break; 5411 case OVERRUN_FALSE: 5412 activeTrack->clearOverflow(); 5413 break; 5414 case OVERRUN_UNKNOWN: 5415 break; 5416 } 5417 5418 } 5419 5420unlock: 5421 // enable changes in effect chain 5422 unlockEffectChains(effectChains); 5423 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5424 } 5425 5426 standbyIfNotAlreadyInStandby(); 5427 5428 { 5429 Mutex::Autolock _l(mLock); 5430 for (size_t i = 0; i < mTracks.size(); i++) { 5431 sp<RecordTrack> track = mTracks[i]; 5432 track->invalidate(); 5433 } 5434 mActiveTracks.clear(); 5435 mActiveTracksGen++; 5436 mStartStopCond.broadcast(); 5437 } 5438 5439 releaseWakeLock(); 5440 5441 ALOGV("RecordThread %p exiting", this); 5442 return false; 5443} 5444 5445void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5446{ 5447 if (!mStandby) { 5448 inputStandBy(); 5449 mStandby = true; 5450 } 5451} 5452 5453void AudioFlinger::RecordThread::inputStandBy() 5454{ 5455 // Idle the fast capture if it's currently running 5456 if (mFastCapture != 0) { 5457 FastCaptureStateQueue *sq = mFastCapture->sq(); 5458 FastCaptureState *state = sq->begin(); 5459 if (!(state->mCommand & FastCaptureState::IDLE)) { 5460 state->mCommand = FastCaptureState::COLD_IDLE; 5461 state->mColdFutexAddr = &mFastCaptureFutex; 5462 state->mColdGen++; 5463 mFastCaptureFutex = 0; 5464 sq->end(); 5465 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5466 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5467#if 0 5468 if (kUseFastCapture == FastCapture_Dynamic) { 5469 // FIXME 5470 } 5471#endif 5472#ifdef AUDIO_WATCHDOG 5473 // FIXME 5474#endif 5475 } else { 5476 sq->end(false /*didModify*/); 5477 } 5478 } 5479 mInput->stream->common.standby(&mInput->stream->common); 5480} 5481 5482// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5483sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5484 const sp<AudioFlinger::Client>& client, 5485 uint32_t sampleRate, 5486 audio_format_t format, 5487 audio_channel_mask_t channelMask, 5488 size_t *pFrameCount, 5489 int sessionId, 5490 size_t *notificationFrames, 5491 int uid, 5492 IAudioFlinger::track_flags_t *flags, 5493 pid_t tid, 5494 status_t *status) 5495{ 5496 size_t frameCount = *pFrameCount; 5497 sp<RecordTrack> track; 5498 status_t lStatus; 5499 5500 // client expresses a preference for FAST, but we get the final say 5501 if (*flags & IAudioFlinger::TRACK_FAST) { 5502 if ( 5503 // use case: callback handler 5504 (tid != -1) && 5505 // frame count is not specified, or is exactly the pipe depth 5506 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5507 // PCM data 5508 audio_is_linear_pcm(format) && 5509 // native format 5510 (format == mFormat) && 5511 // native channel mask 5512 (channelMask == mChannelMask) && 5513 // native hardware sample rate 5514 (sampleRate == mSampleRate) && 5515 // record thread has an associated fast capture 5516 hasFastCapture() && 5517 // there are sufficient fast track slots available 5518 mFastTrackAvail 5519 ) { 5520 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5521 frameCount, mFrameCount); 5522 } else { 5523 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5524 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5525 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5526 frameCount, mFrameCount, mPipeFramesP2, 5527 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5528 hasFastCapture(), tid, mFastTrackAvail); 5529 *flags &= ~IAudioFlinger::TRACK_FAST; 5530 } 5531 } 5532 5533 // compute track buffer size in frames, and suggest the notification frame count 5534 if (*flags & IAudioFlinger::TRACK_FAST) { 5535 // fast track: frame count is exactly the pipe depth 5536 frameCount = mPipeFramesP2; 5537 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5538 *notificationFrames = mFrameCount; 5539 } else { 5540 // not fast track: frame count is at least 2 HAL buffers and at least 20 ms 5541 size_t minFrameCount = ((int64_t) mFrameCount * 2 * sampleRate + mSampleRate - 1) / 5542 mSampleRate; 5543 if (frameCount < minFrameCount) { 5544 frameCount = minFrameCount; 5545 } 5546 minFrameCount = (sampleRate * 20 / 1000 + 1) & ~1; 5547 if (frameCount < minFrameCount) { 5548 frameCount = minFrameCount; 5549 } 5550 // notification is forced to be at least double-buffering 5551 size_t maxNotification = frameCount / 2; 5552 if (*notificationFrames == 0 || *notificationFrames > maxNotification) { 5553 *notificationFrames = maxNotification; 5554 } 5555 } 5556 *pFrameCount = frameCount; 5557 5558 lStatus = initCheck(); 5559 if (lStatus != NO_ERROR) { 5560 ALOGE("createRecordTrack_l() audio driver not initialized"); 5561 goto Exit; 5562 } 5563 5564 { // scope for mLock 5565 Mutex::Autolock _l(mLock); 5566 5567 track = new RecordTrack(this, client, sampleRate, 5568 format, channelMask, frameCount, NULL, sessionId, uid, 5569 *flags, TrackBase::TYPE_DEFAULT); 5570 5571 lStatus = track->initCheck(); 5572 if (lStatus != NO_ERROR) { 5573 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5574 // track must be cleared from the caller as the caller has the AF lock 5575 goto Exit; 5576 } 5577 mTracks.add(track); 5578 5579 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5580 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5581 mAudioFlinger->btNrecIsOff(); 5582 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5583 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5584 5585 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5586 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5587 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5588 // so ask activity manager to do this on our behalf 5589 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5590 } 5591 } 5592 5593 lStatus = NO_ERROR; 5594 5595Exit: 5596 *status = lStatus; 5597 return track; 5598} 5599 5600status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5601 AudioSystem::sync_event_t event, 5602 int triggerSession) 5603{ 5604 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5605 sp<ThreadBase> strongMe = this; 5606 status_t status = NO_ERROR; 5607 5608 if (event == AudioSystem::SYNC_EVENT_NONE) { 5609 recordTrack->clearSyncStartEvent(); 5610 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5611 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5612 triggerSession, 5613 recordTrack->sessionId(), 5614 syncStartEventCallback, 5615 recordTrack); 5616 // Sync event can be cancelled by the trigger session if the track is not in a 5617 // compatible state in which case we start record immediately 5618 if (recordTrack->mSyncStartEvent->isCancelled()) { 5619 recordTrack->clearSyncStartEvent(); 5620 } else { 5621 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5622 recordTrack->mFramesToDrop = - 5623 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5624 } 5625 } 5626 5627 { 5628 // This section is a rendezvous between binder thread executing start() and RecordThread 5629 AutoMutex lock(mLock); 5630 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5631 if (recordTrack->mState == TrackBase::PAUSING) { 5632 ALOGV("active record track PAUSING -> ACTIVE"); 5633 recordTrack->mState = TrackBase::ACTIVE; 5634 } else { 5635 ALOGV("active record track state %d", recordTrack->mState); 5636 } 5637 return status; 5638 } 5639 5640 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5641 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5642 // or using a separate command thread 5643 recordTrack->mState = TrackBase::STARTING_1; 5644 mActiveTracks.add(recordTrack); 5645 mActiveTracksGen++; 5646 status_t status = NO_ERROR; 5647 if (recordTrack->isExternalTrack()) { 5648 mLock.unlock(); 5649 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5650 mLock.lock(); 5651 // FIXME should verify that recordTrack is still in mActiveTracks 5652 if (status != NO_ERROR) { 5653 mActiveTracks.remove(recordTrack); 5654 mActiveTracksGen++; 5655 recordTrack->clearSyncStartEvent(); 5656 ALOGV("RecordThread::start error %d", status); 5657 return status; 5658 } 5659 } 5660 // Catch up with current buffer indices if thread is already running. 5661 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5662 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5663 // see previously buffered data before it called start(), but with greater risk of overrun. 5664 5665 recordTrack->mRsmpInFront = mRsmpInRear; 5666 recordTrack->mRsmpInUnrel = 0; 5667 // FIXME why reset? 5668 if (recordTrack->mResampler != NULL) { 5669 recordTrack->mResampler->reset(); 5670 } 5671 recordTrack->mState = TrackBase::STARTING_2; 5672 // signal thread to start 5673 mWaitWorkCV.broadcast(); 5674 if (mActiveTracks.indexOf(recordTrack) < 0) { 5675 ALOGV("Record failed to start"); 5676 status = BAD_VALUE; 5677 goto startError; 5678 } 5679 return status; 5680 } 5681 5682startError: 5683 if (recordTrack->isExternalTrack()) { 5684 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5685 } 5686 recordTrack->clearSyncStartEvent(); 5687 // FIXME I wonder why we do not reset the state here? 5688 return status; 5689} 5690 5691void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5692{ 5693 sp<SyncEvent> strongEvent = event.promote(); 5694 5695 if (strongEvent != 0) { 5696 sp<RefBase> ptr = strongEvent->cookie().promote(); 5697 if (ptr != 0) { 5698 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5699 recordTrack->handleSyncStartEvent(strongEvent); 5700 } 5701 } 5702} 5703 5704bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5705 ALOGV("RecordThread::stop"); 5706 AutoMutex _l(mLock); 5707 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5708 return false; 5709 } 5710 // note that threadLoop may still be processing the track at this point [without lock] 5711 recordTrack->mState = TrackBase::PAUSING; 5712 // do not wait for mStartStopCond if exiting 5713 if (exitPending()) { 5714 return true; 5715 } 5716 // FIXME incorrect usage of wait: no explicit predicate or loop 5717 mStartStopCond.wait(mLock); 5718 // if we have been restarted, recordTrack is in mActiveTracks here 5719 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5720 ALOGV("Record stopped OK"); 5721 return true; 5722 } 5723 return false; 5724} 5725 5726bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5727{ 5728 return false; 5729} 5730 5731status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5732{ 5733#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5734 if (!isValidSyncEvent(event)) { 5735 return BAD_VALUE; 5736 } 5737 5738 int eventSession = event->triggerSession(); 5739 status_t ret = NAME_NOT_FOUND; 5740 5741 Mutex::Autolock _l(mLock); 5742 5743 for (size_t i = 0; i < mTracks.size(); i++) { 5744 sp<RecordTrack> track = mTracks[i]; 5745 if (eventSession == track->sessionId()) { 5746 (void) track->setSyncEvent(event); 5747 ret = NO_ERROR; 5748 } 5749 } 5750 return ret; 5751#else 5752 return BAD_VALUE; 5753#endif 5754} 5755 5756// destroyTrack_l() must be called with ThreadBase::mLock held 5757void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5758{ 5759 track->terminate(); 5760 track->mState = TrackBase::STOPPED; 5761 // active tracks are removed by threadLoop() 5762 if (mActiveTracks.indexOf(track) < 0) { 5763 removeTrack_l(track); 5764 } 5765} 5766 5767void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5768{ 5769 mTracks.remove(track); 5770 // need anything related to effects here? 5771 if (track->isFastTrack()) { 5772 ALOG_ASSERT(!mFastTrackAvail); 5773 mFastTrackAvail = true; 5774 } 5775} 5776 5777void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5778{ 5779 dumpInternals(fd, args); 5780 dumpTracks(fd, args); 5781 dumpEffectChains(fd, args); 5782} 5783 5784void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5785{ 5786 dprintf(fd, "\nInput thread %p:\n", this); 5787 5788 if (mActiveTracks.size() > 0) { 5789 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5790 } else { 5791 dprintf(fd, " No active record clients\n"); 5792 } 5793 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5794 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5795 5796 dumpBase(fd, args); 5797} 5798 5799void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5800{ 5801 const size_t SIZE = 256; 5802 char buffer[SIZE]; 5803 String8 result; 5804 5805 size_t numtracks = mTracks.size(); 5806 size_t numactive = mActiveTracks.size(); 5807 size_t numactiveseen = 0; 5808 dprintf(fd, " %d Tracks", numtracks); 5809 if (numtracks) { 5810 dprintf(fd, " of which %d are active\n", numactive); 5811 RecordTrack::appendDumpHeader(result); 5812 for (size_t i = 0; i < numtracks ; ++i) { 5813 sp<RecordTrack> track = mTracks[i]; 5814 if (track != 0) { 5815 bool active = mActiveTracks.indexOf(track) >= 0; 5816 if (active) { 5817 numactiveseen++; 5818 } 5819 track->dump(buffer, SIZE, active); 5820 result.append(buffer); 5821 } 5822 } 5823 } else { 5824 dprintf(fd, "\n"); 5825 } 5826 5827 if (numactiveseen != numactive) { 5828 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5829 " not in the track list\n"); 5830 result.append(buffer); 5831 RecordTrack::appendDumpHeader(result); 5832 for (size_t i = 0; i < numactive; ++i) { 5833 sp<RecordTrack> track = mActiveTracks[i]; 5834 if (mTracks.indexOf(track) < 0) { 5835 track->dump(buffer, SIZE, true); 5836 result.append(buffer); 5837 } 5838 } 5839 5840 } 5841 write(fd, result.string(), result.size()); 5842} 5843 5844// AudioBufferProvider interface 5845status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5846 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5847{ 5848 RecordTrack *activeTrack = mRecordTrack; 5849 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5850 if (threadBase == 0) { 5851 buffer->frameCount = 0; 5852 buffer->raw = NULL; 5853 return NOT_ENOUGH_DATA; 5854 } 5855 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5856 int32_t rear = recordThread->mRsmpInRear; 5857 int32_t front = activeTrack->mRsmpInFront; 5858 ssize_t filled = rear - front; 5859 // FIXME should not be P2 (don't want to increase latency) 5860 // FIXME if client not keeping up, discard 5861 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5862 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5863 front &= recordThread->mRsmpInFramesP2 - 1; 5864 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5865 if (part1 > (size_t) filled) { 5866 part1 = filled; 5867 } 5868 size_t ask = buffer->frameCount; 5869 ALOG_ASSERT(ask > 0); 5870 if (part1 > ask) { 5871 part1 = ask; 5872 } 5873 if (part1 == 0) { 5874 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5875 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5876 buffer->raw = NULL; 5877 buffer->frameCount = 0; 5878 activeTrack->mRsmpInUnrel = 0; 5879 return NOT_ENOUGH_DATA; 5880 } 5881 5882 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5883 buffer->frameCount = part1; 5884 activeTrack->mRsmpInUnrel = part1; 5885 return NO_ERROR; 5886} 5887 5888// AudioBufferProvider interface 5889void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5890 AudioBufferProvider::Buffer* buffer) 5891{ 5892 RecordTrack *activeTrack = mRecordTrack; 5893 size_t stepCount = buffer->frameCount; 5894 if (stepCount == 0) { 5895 return; 5896 } 5897 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5898 activeTrack->mRsmpInUnrel -= stepCount; 5899 activeTrack->mRsmpInFront += stepCount; 5900 buffer->raw = NULL; 5901 buffer->frameCount = 0; 5902} 5903 5904bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5905 status_t& status) 5906{ 5907 bool reconfig = false; 5908 5909 status = NO_ERROR; 5910 5911 audio_format_t reqFormat = mFormat; 5912 uint32_t samplingRate = mSampleRate; 5913 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5914 5915 AudioParameter param = AudioParameter(keyValuePair); 5916 int value; 5917 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5918 // channel count change can be requested. Do we mandate the first client defines the 5919 // HAL sampling rate and channel count or do we allow changes on the fly? 5920 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5921 samplingRate = value; 5922 reconfig = true; 5923 } 5924 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5925 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5926 status = BAD_VALUE; 5927 } else { 5928 reqFormat = (audio_format_t) value; 5929 reconfig = true; 5930 } 5931 } 5932 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5933 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5934 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5935 status = BAD_VALUE; 5936 } else { 5937 channelMask = mask; 5938 reconfig = true; 5939 } 5940 } 5941 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5942 // do not accept frame count changes if tracks are open as the track buffer 5943 // size depends on frame count and correct behavior would not be guaranteed 5944 // if frame count is changed after track creation 5945 if (mActiveTracks.size() > 0) { 5946 status = INVALID_OPERATION; 5947 } else { 5948 reconfig = true; 5949 } 5950 } 5951 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5952 // forward device change to effects that have requested to be 5953 // aware of attached audio device. 5954 for (size_t i = 0; i < mEffectChains.size(); i++) { 5955 mEffectChains[i]->setDevice_l(value); 5956 } 5957 5958 // store input device and output device but do not forward output device to audio HAL. 5959 // Note that status is ignored by the caller for output device 5960 // (see AudioFlinger::setParameters() 5961 if (audio_is_output_devices(value)) { 5962 mOutDevice = value; 5963 status = BAD_VALUE; 5964 } else { 5965 mInDevice = value; 5966 // disable AEC and NS if the device is a BT SCO headset supporting those 5967 // pre processings 5968 if (mTracks.size() > 0) { 5969 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5970 mAudioFlinger->btNrecIsOff(); 5971 for (size_t i = 0; i < mTracks.size(); i++) { 5972 sp<RecordTrack> track = mTracks[i]; 5973 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5974 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5975 } 5976 } 5977 } 5978 } 5979 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5980 mAudioSource != (audio_source_t)value) { 5981 // forward device change to effects that have requested to be 5982 // aware of attached audio device. 5983 for (size_t i = 0; i < mEffectChains.size(); i++) { 5984 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5985 } 5986 mAudioSource = (audio_source_t)value; 5987 } 5988 5989 if (status == NO_ERROR) { 5990 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5991 keyValuePair.string()); 5992 if (status == INVALID_OPERATION) { 5993 inputStandBy(); 5994 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5995 keyValuePair.string()); 5996 } 5997 if (reconfig) { 5998 if (status == BAD_VALUE && 5999 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6000 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6001 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6002 <= (2 * samplingRate)) && 6003 audio_channel_count_from_in_mask( 6004 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6005 (channelMask == AUDIO_CHANNEL_IN_MONO || 6006 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6007 status = NO_ERROR; 6008 } 6009 if (status == NO_ERROR) { 6010 readInputParameters_l(); 6011 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6012 } 6013 } 6014 } 6015 6016 return reconfig; 6017} 6018 6019String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6020{ 6021 Mutex::Autolock _l(mLock); 6022 if (initCheck() != NO_ERROR) { 6023 return String8(); 6024 } 6025 6026 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6027 const String8 out_s8(s); 6028 free(s); 6029 return out_s8; 6030} 6031 6032void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6033 AudioSystem::OutputDescriptor desc; 6034 const void *param2 = NULL; 6035 6036 switch (event) { 6037 case AudioSystem::INPUT_OPENED: 6038 case AudioSystem::INPUT_CONFIG_CHANGED: 6039 desc.channelMask = mChannelMask; 6040 desc.samplingRate = mSampleRate; 6041 desc.format = mFormat; 6042 desc.frameCount = mFrameCount; 6043 desc.latency = 0; 6044 param2 = &desc; 6045 break; 6046 6047 case AudioSystem::INPUT_CLOSED: 6048 default: 6049 break; 6050 } 6051 mAudioFlinger->audioConfigChanged(event, mId, param2); 6052} 6053 6054void AudioFlinger::RecordThread::readInputParameters_l() 6055{ 6056 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6057 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6058 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6059 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6060 mFormat = mHALFormat; 6061 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6062 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6063 } 6064 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6065 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6066 mFrameCount = mBufferSize / mFrameSize; 6067 // This is the formula for calculating the temporary buffer size. 6068 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6069 // 1 full output buffer, regardless of the alignment of the available input. 6070 // The value is somewhat arbitrary, and could probably be even larger. 6071 // A larger value should allow more old data to be read after a track calls start(), 6072 // without increasing latency. 6073 mRsmpInFrames = mFrameCount * 7; 6074 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6075 delete[] mRsmpInBuffer; 6076 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6077 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6078 6079 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6080 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6081} 6082 6083uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6084{ 6085 Mutex::Autolock _l(mLock); 6086 if (initCheck() != NO_ERROR) { 6087 return 0; 6088 } 6089 6090 return mInput->stream->get_input_frames_lost(mInput->stream); 6091} 6092 6093uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6094{ 6095 Mutex::Autolock _l(mLock); 6096 uint32_t result = 0; 6097 if (getEffectChain_l(sessionId) != 0) { 6098 result = EFFECT_SESSION; 6099 } 6100 6101 for (size_t i = 0; i < mTracks.size(); ++i) { 6102 if (sessionId == mTracks[i]->sessionId()) { 6103 result |= TRACK_SESSION; 6104 break; 6105 } 6106 } 6107 6108 return result; 6109} 6110 6111KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6112{ 6113 KeyedVector<int, bool> ids; 6114 Mutex::Autolock _l(mLock); 6115 for (size_t j = 0; j < mTracks.size(); ++j) { 6116 sp<RecordThread::RecordTrack> track = mTracks[j]; 6117 int sessionId = track->sessionId(); 6118 if (ids.indexOfKey(sessionId) < 0) { 6119 ids.add(sessionId, true); 6120 } 6121 } 6122 return ids; 6123} 6124 6125AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6126{ 6127 Mutex::Autolock _l(mLock); 6128 AudioStreamIn *input = mInput; 6129 mInput = NULL; 6130 return input; 6131} 6132 6133// this method must always be called either with ThreadBase mLock held or inside the thread loop 6134audio_stream_t* AudioFlinger::RecordThread::stream() const 6135{ 6136 if (mInput == NULL) { 6137 return NULL; 6138 } 6139 return &mInput->stream->common; 6140} 6141 6142status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6143{ 6144 // only one chain per input thread 6145 if (mEffectChains.size() != 0) { 6146 return INVALID_OPERATION; 6147 } 6148 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6149 6150 chain->setInBuffer(NULL); 6151 chain->setOutBuffer(NULL); 6152 6153 checkSuspendOnAddEffectChain_l(chain); 6154 6155 mEffectChains.add(chain); 6156 6157 return NO_ERROR; 6158} 6159 6160size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6161{ 6162 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6163 ALOGW_IF(mEffectChains.size() != 1, 6164 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6165 chain.get(), mEffectChains.size(), this); 6166 if (mEffectChains.size() == 1) { 6167 mEffectChains.removeAt(0); 6168 } 6169 return 0; 6170} 6171 6172status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6173 audio_patch_handle_t *handle) 6174{ 6175 status_t status = NO_ERROR; 6176 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6177 // store new device and send to effects 6178 mInDevice = patch->sources[0].ext.device.type; 6179 for (size_t i = 0; i < mEffectChains.size(); i++) { 6180 mEffectChains[i]->setDevice_l(mInDevice); 6181 } 6182 6183 // disable AEC and NS if the device is a BT SCO headset supporting those 6184 // pre processings 6185 if (mTracks.size() > 0) { 6186 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6187 mAudioFlinger->btNrecIsOff(); 6188 for (size_t i = 0; i < mTracks.size(); i++) { 6189 sp<RecordTrack> track = mTracks[i]; 6190 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6191 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6192 } 6193 } 6194 6195 // store new source and send to effects 6196 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6197 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6198 for (size_t i = 0; i < mEffectChains.size(); i++) { 6199 mEffectChains[i]->setAudioSource_l(mAudioSource); 6200 } 6201 } 6202 6203 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6204 status = hwDevice->create_audio_patch(hwDevice, 6205 patch->num_sources, 6206 patch->sources, 6207 patch->num_sinks, 6208 patch->sinks, 6209 handle); 6210 } else { 6211 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6212 } 6213 return status; 6214} 6215 6216status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6217{ 6218 status_t status = NO_ERROR; 6219 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6220 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6221 status = hwDevice->release_audio_patch(hwDevice, handle); 6222 } else { 6223 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6224 } 6225 return status; 6226} 6227 6228void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6229{ 6230 Mutex::Autolock _l(mLock); 6231 mTracks.add(record); 6232} 6233 6234void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6235{ 6236 Mutex::Autolock _l(mLock); 6237 destroyTrack_l(record); 6238} 6239 6240void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6241{ 6242 ThreadBase::getAudioPortConfig(config); 6243 config->role = AUDIO_PORT_ROLE_SINK; 6244 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6245 config->ext.mix.usecase.source = mAudioSource; 6246} 6247 6248}; // namespace android 6249