Threads.cpp revision 275e8e9de2e11b4b344f5a201f1f0e51fda02d9c
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <media/AudioResamplerPublic.h> 30#include <utils/Log.h> 31#include <utils/Trace.h> 32 33#include <private/media/AudioTrackShared.h> 34#include <hardware/audio.h> 35#include <audio_effects/effect_ns.h> 36#include <audio_effects/effect_aec.h> 37#include <audio_utils/primitives.h> 38#include <audio_utils/format.h> 39#include <audio_utils/minifloat.h> 40 41// NBAIO implementations 42#include <media/nbaio/AudioStreamInSource.h> 43#include <media/nbaio/AudioStreamOutSink.h> 44#include <media/nbaio/MonoPipe.h> 45#include <media/nbaio/MonoPipeReader.h> 46#include <media/nbaio/Pipe.h> 47#include <media/nbaio/PipeReader.h> 48#include <media/nbaio/SourceAudioBufferProvider.h> 49 50#include <powermanager/PowerManager.h> 51 52#include <common_time/cc_helper.h> 53#include <common_time/local_clock.h> 54 55#include "AudioFlinger.h" 56#include "AudioMixer.h" 57#include "FastMixer.h" 58#include "FastCapture.h" 59#include "ServiceUtilities.h" 60#include "SchedulingPolicyService.h" 61 62#ifdef ADD_BATTERY_DATA 63#include <media/IMediaPlayerService.h> 64#include <media/IMediaDeathNotifier.h> 65#endif 66 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72// ---------------------------------------------------------------------------- 73 74// Note: the following macro is used for extremely verbose logging message. In 75// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 76// 0; but one side effect of this is to turn all LOGV's as well. Some messages 77// are so verbose that we want to suppress them even when we have ALOG_ASSERT 78// turned on. Do not uncomment the #def below unless you really know what you 79// are doing and want to see all of the extremely verbose messages. 80//#define VERY_VERY_VERBOSE_LOGGING 81#ifdef VERY_VERY_VERBOSE_LOGGING 82#define ALOGVV ALOGV 83#else 84#define ALOGVV(a...) do { } while(0) 85#endif 86 87#define max(a, b) ((a) > (b) ? (a) : (b)) 88 89namespace android { 90 91// retry counts for buffer fill timeout 92// 50 * ~20msecs = 1 second 93static const int8_t kMaxTrackRetries = 50; 94static const int8_t kMaxTrackStartupRetries = 50; 95// allow less retry attempts on direct output thread. 96// direct outputs can be a scarce resource in audio hardware and should 97// be released as quickly as possible. 98static const int8_t kMaxTrackRetriesDirect = 2; 99 100// don't warn about blocked writes or record buffer overflows more often than this 101static const nsecs_t kWarningThrottleNs = seconds(5); 102 103// RecordThread loop sleep time upon application overrun or audio HAL read error 104static const int kRecordThreadSleepUs = 5000; 105 106// maximum time to wait in sendConfigEvent_l() for a status to be received 107static const nsecs_t kConfigEventTimeoutNs = seconds(2); 108 109// minimum sleep time for the mixer thread loop when tracks are active but in underrun 110static const uint32_t kMinThreadSleepTimeUs = 5000; 111// maximum divider applied to the active sleep time in the mixer thread loop 112static const uint32_t kMaxThreadSleepTimeShift = 2; 113 114// minimum normal sink buffer size, expressed in milliseconds rather than frames 115static const uint32_t kMinNormalSinkBufferSizeMs = 20; 116// maximum normal sink buffer size 117static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 118 119// Offloaded output thread standby delay: allows track transition without going to standby 120static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 121 122// Whether to use fast mixer 123static const enum { 124 FastMixer_Never, // never initialize or use: for debugging only 125 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 126 // normal mixer multiplier is 1 127 FastMixer_Static, // initialize if needed, then use all the time if initialized, 128 // multiplier is calculated based on min & max normal mixer buffer size 129 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 130 // multiplier is calculated based on min & max normal mixer buffer size 131 // FIXME for FastMixer_Dynamic: 132 // Supporting this option will require fixing HALs that can't handle large writes. 133 // For example, one HAL implementation returns an error from a large write, 134 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 135 // We could either fix the HAL implementations, or provide a wrapper that breaks 136 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 137} kUseFastMixer = FastMixer_Static; 138 139// Whether to use fast capture 140static const enum { 141 FastCapture_Never, // never initialize or use: for debugging only 142 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 143 FastCapture_Static, // initialize if needed, then use all the time if initialized 144} kUseFastCapture = FastCapture_Static; 145 146// Priorities for requestPriority 147static const int kPriorityAudioApp = 2; 148static const int kPriorityFastMixer = 3; 149static const int kPriorityFastCapture = 3; 150 151// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 152// for the track. The client then sub-divides this into smaller buffers for its use. 153// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 154// So for now we just assume that client is double-buffered for fast tracks. 155// FIXME It would be better for client to tell AudioFlinger the value of N, 156// so AudioFlinger could allocate the right amount of memory. 157// See the client's minBufCount and mNotificationFramesAct calculations for details. 158 159// This is the default value, if not specified by property. 160static const int kFastTrackMultiplier = 2; 161 162// The minimum and maximum allowed values 163static const int kFastTrackMultiplierMin = 1; 164static const int kFastTrackMultiplierMax = 2; 165 166// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 167static int sFastTrackMultiplier = kFastTrackMultiplier; 168 169// See Thread::readOnlyHeap(). 170// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 171// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 172// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 173static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 174 175// ---------------------------------------------------------------------------- 176 177static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 178 179static void sFastTrackMultiplierInit() 180{ 181 char value[PROPERTY_VALUE_MAX]; 182 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 183 char *endptr; 184 unsigned long ul = strtoul(value, &endptr, 0); 185 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 186 sFastTrackMultiplier = (int) ul; 187 } 188 } 189} 190 191// ---------------------------------------------------------------------------- 192 193#ifdef ADD_BATTERY_DATA 194// To collect the amplifier usage 195static void addBatteryData(uint32_t params) { 196 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 197 if (service == NULL) { 198 // it already logged 199 return; 200 } 201 202 service->addBatteryData(params); 203} 204#endif 205 206 207// ---------------------------------------------------------------------------- 208// CPU Stats 209// ---------------------------------------------------------------------------- 210 211class CpuStats { 212public: 213 CpuStats(); 214 void sample(const String8 &title); 215#ifdef DEBUG_CPU_USAGE 216private: 217 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 218 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 219 220 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 221 222 int mCpuNum; // thread's current CPU number 223 int mCpukHz; // frequency of thread's current CPU in kHz 224#endif 225}; 226 227CpuStats::CpuStats() 228#ifdef DEBUG_CPU_USAGE 229 : mCpuNum(-1), mCpukHz(-1) 230#endif 231{ 232} 233 234void CpuStats::sample(const String8 &title 235#ifndef DEBUG_CPU_USAGE 236 __unused 237#endif 238 ) { 239#ifdef DEBUG_CPU_USAGE 240 // get current thread's delta CPU time in wall clock ns 241 double wcNs; 242 bool valid = mCpuUsage.sampleAndEnable(wcNs); 243 244 // record sample for wall clock statistics 245 if (valid) { 246 mWcStats.sample(wcNs); 247 } 248 249 // get the current CPU number 250 int cpuNum = sched_getcpu(); 251 252 // get the current CPU frequency in kHz 253 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 254 255 // check if either CPU number or frequency changed 256 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 257 mCpuNum = cpuNum; 258 mCpukHz = cpukHz; 259 // ignore sample for purposes of cycles 260 valid = false; 261 } 262 263 // if no change in CPU number or frequency, then record sample for cycle statistics 264 if (valid && mCpukHz > 0) { 265 double cycles = wcNs * cpukHz * 0.000001; 266 mHzStats.sample(cycles); 267 } 268 269 unsigned n = mWcStats.n(); 270 // mCpuUsage.elapsed() is expensive, so don't call it every loop 271 if ((n & 127) == 1) { 272 long long elapsed = mCpuUsage.elapsed(); 273 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 274 double perLoop = elapsed / (double) n; 275 double perLoop100 = perLoop * 0.01; 276 double perLoop1k = perLoop * 0.001; 277 double mean = mWcStats.mean(); 278 double stddev = mWcStats.stddev(); 279 double minimum = mWcStats.minimum(); 280 double maximum = mWcStats.maximum(); 281 double meanCycles = mHzStats.mean(); 282 double stddevCycles = mHzStats.stddev(); 283 double minCycles = mHzStats.minimum(); 284 double maxCycles = mHzStats.maximum(); 285 mCpuUsage.resetElapsed(); 286 mWcStats.reset(); 287 mHzStats.reset(); 288 ALOGD("CPU usage for %s over past %.1f secs\n" 289 " (%u mixer loops at %.1f mean ms per loop):\n" 290 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 291 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 292 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 293 title.string(), 294 elapsed * .000000001, n, perLoop * .000001, 295 mean * .001, 296 stddev * .001, 297 minimum * .001, 298 maximum * .001, 299 mean / perLoop100, 300 stddev / perLoop100, 301 minimum / perLoop100, 302 maximum / perLoop100, 303 meanCycles / perLoop1k, 304 stddevCycles / perLoop1k, 305 minCycles / perLoop1k, 306 maxCycles / perLoop1k); 307 308 } 309 } 310#endif 311}; 312 313// ---------------------------------------------------------------------------- 314// ThreadBase 315// ---------------------------------------------------------------------------- 316 317AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 318 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 319 : Thread(false /*canCallJava*/), 320 mType(type), 321 mAudioFlinger(audioFlinger), 322 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 323 // are set by PlaybackThread::readOutputParameters_l() or 324 // RecordThread::readInputParameters_l() 325 //FIXME: mStandby should be true here. Is this some kind of hack? 326 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 327 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 328 // mName will be set by concrete (non-virtual) subclass 329 mDeathRecipient(new PMDeathRecipient(this)) 330{ 331} 332 333AudioFlinger::ThreadBase::~ThreadBase() 334{ 335 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 336 mConfigEvents.clear(); 337 338 // do not lock the mutex in destructor 339 releaseWakeLock_l(); 340 if (mPowerManager != 0) { 341 sp<IBinder> binder = mPowerManager->asBinder(); 342 binder->unlinkToDeath(mDeathRecipient); 343 } 344} 345 346status_t AudioFlinger::ThreadBase::readyToRun() 347{ 348 status_t status = initCheck(); 349 if (status == NO_ERROR) { 350 ALOGI("AudioFlinger's thread %p ready to run", this); 351 } else { 352 ALOGE("No working audio driver found."); 353 } 354 return status; 355} 356 357void AudioFlinger::ThreadBase::exit() 358{ 359 ALOGV("ThreadBase::exit"); 360 // do any cleanup required for exit to succeed 361 preExit(); 362 { 363 // This lock prevents the following race in thread (uniprocessor for illustration): 364 // if (!exitPending()) { 365 // // context switch from here to exit() 366 // // exit() calls requestExit(), what exitPending() observes 367 // // exit() calls signal(), which is dropped since no waiters 368 // // context switch back from exit() to here 369 // mWaitWorkCV.wait(...); 370 // // now thread is hung 371 // } 372 AutoMutex lock(mLock); 373 requestExit(); 374 mWaitWorkCV.broadcast(); 375 } 376 // When Thread::requestExitAndWait is made virtual and this method is renamed to 377 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 378 requestExitAndWait(); 379} 380 381status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 382{ 383 status_t status; 384 385 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 386 Mutex::Autolock _l(mLock); 387 388 return sendSetParameterConfigEvent_l(keyValuePairs); 389} 390 391// sendConfigEvent_l() must be called with ThreadBase::mLock held 392// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 393status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 394{ 395 status_t status = NO_ERROR; 396 397 mConfigEvents.add(event); 398 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 399 mWaitWorkCV.signal(); 400 mLock.unlock(); 401 { 402 Mutex::Autolock _l(event->mLock); 403 while (event->mWaitStatus) { 404 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 405 event->mStatus = TIMED_OUT; 406 event->mWaitStatus = false; 407 } 408 } 409 status = event->mStatus; 410 } 411 mLock.lock(); 412 return status; 413} 414 415void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 416{ 417 Mutex::Autolock _l(mLock); 418 sendIoConfigEvent_l(event, param); 419} 420 421// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 422void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 423{ 424 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 425 sendConfigEvent_l(configEvent); 426} 427 428// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 429void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 430{ 431 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 432 sendConfigEvent_l(configEvent); 433} 434 435// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 436status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 437{ 438 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 439 return sendConfigEvent_l(configEvent); 440} 441 442status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 443 const struct audio_patch *patch, 444 audio_patch_handle_t *handle) 445{ 446 Mutex::Autolock _l(mLock); 447 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 448 status_t status = sendConfigEvent_l(configEvent); 449 if (status == NO_ERROR) { 450 CreateAudioPatchConfigEventData *data = 451 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 452 *handle = data->mHandle; 453 } 454 return status; 455} 456 457status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 458 const audio_patch_handle_t handle) 459{ 460 Mutex::Autolock _l(mLock); 461 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 462 return sendConfigEvent_l(configEvent); 463} 464 465 466// post condition: mConfigEvents.isEmpty() 467void AudioFlinger::ThreadBase::processConfigEvents_l() 468{ 469 bool configChanged = false; 470 471 while (!mConfigEvents.isEmpty()) { 472 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 473 sp<ConfigEvent> event = mConfigEvents[0]; 474 mConfigEvents.removeAt(0); 475 switch (event->mType) { 476 case CFG_EVENT_PRIO: { 477 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 478 // FIXME Need to understand why this has to be done asynchronously 479 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 480 true /*asynchronous*/); 481 if (err != 0) { 482 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 483 data->mPrio, data->mPid, data->mTid, err); 484 } 485 } break; 486 case CFG_EVENT_IO: { 487 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 488 audioConfigChanged(data->mEvent, data->mParam); 489 } break; 490 case CFG_EVENT_SET_PARAMETER: { 491 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 492 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 493 configChanged = true; 494 } 495 } break; 496 case CFG_EVENT_CREATE_AUDIO_PATCH: { 497 CreateAudioPatchConfigEventData *data = 498 (CreateAudioPatchConfigEventData *)event->mData.get(); 499 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 500 } break; 501 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 502 ReleaseAudioPatchConfigEventData *data = 503 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 504 event->mStatus = releaseAudioPatch_l(data->mHandle); 505 } break; 506 default: 507 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 508 break; 509 } 510 { 511 Mutex::Autolock _l(event->mLock); 512 if (event->mWaitStatus) { 513 event->mWaitStatus = false; 514 event->mCond.signal(); 515 } 516 } 517 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 518 } 519 520 if (configChanged) { 521 cacheParameters_l(); 522 } 523} 524 525String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 526 String8 s; 527 if (output) { 528 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 529 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 530 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 531 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 532 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 533 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 534 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 535 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 536 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 537 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 538 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 539 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 540 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 541 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 542 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 543 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 544 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 545 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 546 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 547 } else { 548 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 549 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 550 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 551 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 552 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 553 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 554 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 555 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 556 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 557 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 558 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 559 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 560 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 561 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 562 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 563 } 564 int len = s.length(); 565 if (s.length() > 2) { 566 char *str = s.lockBuffer(len); 567 s.unlockBuffer(len - 2); 568 } 569 return s; 570} 571 572void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 573{ 574 const size_t SIZE = 256; 575 char buffer[SIZE]; 576 String8 result; 577 578 bool locked = AudioFlinger::dumpTryLock(mLock); 579 if (!locked) { 580 dprintf(fd, "thread %p maybe dead locked\n", this); 581 } 582 583 dprintf(fd, " I/O handle: %d\n", mId); 584 dprintf(fd, " TID: %d\n", getTid()); 585 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 586 dprintf(fd, " Sample rate: %u\n", mSampleRate); 587 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 588 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 589 dprintf(fd, " Channel Count: %u\n", mChannelCount); 590 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 591 channelMaskToString(mChannelMask, mType != RECORD).string()); 592 dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 593 dprintf(fd, " Frame size: %zu\n", mFrameSize); 594 dprintf(fd, " Pending config events:"); 595 size_t numConfig = mConfigEvents.size(); 596 if (numConfig) { 597 for (size_t i = 0; i < numConfig; i++) { 598 mConfigEvents[i]->dump(buffer, SIZE); 599 dprintf(fd, "\n %s", buffer); 600 } 601 dprintf(fd, "\n"); 602 } else { 603 dprintf(fd, " none\n"); 604 } 605 606 if (locked) { 607 mLock.unlock(); 608 } 609} 610 611void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 612{ 613 const size_t SIZE = 256; 614 char buffer[SIZE]; 615 String8 result; 616 617 size_t numEffectChains = mEffectChains.size(); 618 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 619 write(fd, buffer, strlen(buffer)); 620 621 for (size_t i = 0; i < numEffectChains; ++i) { 622 sp<EffectChain> chain = mEffectChains[i]; 623 if (chain != 0) { 624 chain->dump(fd, args); 625 } 626 } 627} 628 629void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 630{ 631 Mutex::Autolock _l(mLock); 632 acquireWakeLock_l(uid); 633} 634 635String16 AudioFlinger::ThreadBase::getWakeLockTag() 636{ 637 switch (mType) { 638 case MIXER: 639 return String16("AudioMix"); 640 case DIRECT: 641 return String16("AudioDirectOut"); 642 case DUPLICATING: 643 return String16("AudioDup"); 644 case RECORD: 645 return String16("AudioIn"); 646 case OFFLOAD: 647 return String16("AudioOffload"); 648 default: 649 ALOG_ASSERT(false); 650 return String16("AudioUnknown"); 651 } 652} 653 654void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 655{ 656 getPowerManager_l(); 657 if (mPowerManager != 0) { 658 sp<IBinder> binder = new BBinder(); 659 status_t status; 660 if (uid >= 0) { 661 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 662 binder, 663 getWakeLockTag(), 664 String16("media"), 665 uid, 666 true /* FIXME force oneway contrary to .aidl */); 667 } else { 668 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 669 binder, 670 getWakeLockTag(), 671 String16("media"), 672 true /* FIXME force oneway contrary to .aidl */); 673 } 674 if (status == NO_ERROR) { 675 mWakeLockToken = binder; 676 } 677 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 678 } 679} 680 681void AudioFlinger::ThreadBase::releaseWakeLock() 682{ 683 Mutex::Autolock _l(mLock); 684 releaseWakeLock_l(); 685} 686 687void AudioFlinger::ThreadBase::releaseWakeLock_l() 688{ 689 if (mWakeLockToken != 0) { 690 ALOGV("releaseWakeLock_l() %s", mName); 691 if (mPowerManager != 0) { 692 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 693 true /* FIXME force oneway contrary to .aidl */); 694 } 695 mWakeLockToken.clear(); 696 } 697} 698 699void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 700 Mutex::Autolock _l(mLock); 701 updateWakeLockUids_l(uids); 702} 703 704void AudioFlinger::ThreadBase::getPowerManager_l() { 705 706 if (mPowerManager == 0) { 707 // use checkService() to avoid blocking if power service is not up yet 708 sp<IBinder> binder = 709 defaultServiceManager()->checkService(String16("power")); 710 if (binder == 0) { 711 ALOGW("Thread %s cannot connect to the power manager service", mName); 712 } else { 713 mPowerManager = interface_cast<IPowerManager>(binder); 714 binder->linkToDeath(mDeathRecipient); 715 } 716 } 717} 718 719void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 720 721 getPowerManager_l(); 722 if (mWakeLockToken == NULL) { 723 ALOGE("no wake lock to update!"); 724 return; 725 } 726 if (mPowerManager != 0) { 727 sp<IBinder> binder = new BBinder(); 728 status_t status; 729 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 730 true /* FIXME force oneway contrary to .aidl */); 731 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 732 } 733} 734 735void AudioFlinger::ThreadBase::clearPowerManager() 736{ 737 Mutex::Autolock _l(mLock); 738 releaseWakeLock_l(); 739 mPowerManager.clear(); 740} 741 742void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 743{ 744 sp<ThreadBase> thread = mThread.promote(); 745 if (thread != 0) { 746 thread->clearPowerManager(); 747 } 748 ALOGW("power manager service died !!!"); 749} 750 751void AudioFlinger::ThreadBase::setEffectSuspended( 752 const effect_uuid_t *type, bool suspend, int sessionId) 753{ 754 Mutex::Autolock _l(mLock); 755 setEffectSuspended_l(type, suspend, sessionId); 756} 757 758void AudioFlinger::ThreadBase::setEffectSuspended_l( 759 const effect_uuid_t *type, bool suspend, int sessionId) 760{ 761 sp<EffectChain> chain = getEffectChain_l(sessionId); 762 if (chain != 0) { 763 if (type != NULL) { 764 chain->setEffectSuspended_l(type, suspend); 765 } else { 766 chain->setEffectSuspendedAll_l(suspend); 767 } 768 } 769 770 updateSuspendedSessions_l(type, suspend, sessionId); 771} 772 773void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 774{ 775 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 776 if (index < 0) { 777 return; 778 } 779 780 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 781 mSuspendedSessions.valueAt(index); 782 783 for (size_t i = 0; i < sessionEffects.size(); i++) { 784 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 785 for (int j = 0; j < desc->mRefCount; j++) { 786 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 787 chain->setEffectSuspendedAll_l(true); 788 } else { 789 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 790 desc->mType.timeLow); 791 chain->setEffectSuspended_l(&desc->mType, true); 792 } 793 } 794 } 795} 796 797void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 798 bool suspend, 799 int sessionId) 800{ 801 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 802 803 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 804 805 if (suspend) { 806 if (index >= 0) { 807 sessionEffects = mSuspendedSessions.valueAt(index); 808 } else { 809 mSuspendedSessions.add(sessionId, sessionEffects); 810 } 811 } else { 812 if (index < 0) { 813 return; 814 } 815 sessionEffects = mSuspendedSessions.valueAt(index); 816 } 817 818 819 int key = EffectChain::kKeyForSuspendAll; 820 if (type != NULL) { 821 key = type->timeLow; 822 } 823 index = sessionEffects.indexOfKey(key); 824 825 sp<SuspendedSessionDesc> desc; 826 if (suspend) { 827 if (index >= 0) { 828 desc = sessionEffects.valueAt(index); 829 } else { 830 desc = new SuspendedSessionDesc(); 831 if (type != NULL) { 832 desc->mType = *type; 833 } 834 sessionEffects.add(key, desc); 835 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 836 } 837 desc->mRefCount++; 838 } else { 839 if (index < 0) { 840 return; 841 } 842 desc = sessionEffects.valueAt(index); 843 if (--desc->mRefCount == 0) { 844 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 845 sessionEffects.removeItemsAt(index); 846 if (sessionEffects.isEmpty()) { 847 ALOGV("updateSuspendedSessions_l() restore removing session %d", 848 sessionId); 849 mSuspendedSessions.removeItem(sessionId); 850 } 851 } 852 } 853 if (!sessionEffects.isEmpty()) { 854 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 855 } 856} 857 858void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 859 bool enabled, 860 int sessionId) 861{ 862 Mutex::Autolock _l(mLock); 863 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 864} 865 866void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 867 bool enabled, 868 int sessionId) 869{ 870 if (mType != RECORD) { 871 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 872 // another session. This gives the priority to well behaved effect control panels 873 // and applications not using global effects. 874 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 875 // global effects 876 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 877 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 878 } 879 } 880 881 sp<EffectChain> chain = getEffectChain_l(sessionId); 882 if (chain != 0) { 883 chain->checkSuspendOnEffectEnabled(effect, enabled); 884 } 885} 886 887// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 888sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 889 const sp<AudioFlinger::Client>& client, 890 const sp<IEffectClient>& effectClient, 891 int32_t priority, 892 int sessionId, 893 effect_descriptor_t *desc, 894 int *enabled, 895 status_t *status) 896{ 897 sp<EffectModule> effect; 898 sp<EffectHandle> handle; 899 status_t lStatus; 900 sp<EffectChain> chain; 901 bool chainCreated = false; 902 bool effectCreated = false; 903 bool effectRegistered = false; 904 905 lStatus = initCheck(); 906 if (lStatus != NO_ERROR) { 907 ALOGW("createEffect_l() Audio driver not initialized."); 908 goto Exit; 909 } 910 911 // Reject any effect on Direct output threads for now, since the format of 912 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 913 if (mType == DIRECT) { 914 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 915 desc->name, mName); 916 lStatus = BAD_VALUE; 917 goto Exit; 918 } 919 920 // Reject any effect on mixer or duplicating multichannel sinks. 921 // TODO: fix both format and multichannel issues with effects. 922 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 923 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 924 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 925 lStatus = BAD_VALUE; 926 goto Exit; 927 } 928 929 // Allow global effects only on offloaded and mixer threads 930 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 931 switch (mType) { 932 case MIXER: 933 case OFFLOAD: 934 break; 935 case DIRECT: 936 case DUPLICATING: 937 case RECORD: 938 default: 939 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 940 lStatus = BAD_VALUE; 941 goto Exit; 942 } 943 } 944 945 // Only Pre processor effects are allowed on input threads and only on input threads 946 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 947 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 948 desc->name, desc->flags, mType); 949 lStatus = BAD_VALUE; 950 goto Exit; 951 } 952 953 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 954 955 { // scope for mLock 956 Mutex::Autolock _l(mLock); 957 958 // check for existing effect chain with the requested audio session 959 chain = getEffectChain_l(sessionId); 960 if (chain == 0) { 961 // create a new chain for this session 962 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 963 chain = new EffectChain(this, sessionId); 964 addEffectChain_l(chain); 965 chain->setStrategy(getStrategyForSession_l(sessionId)); 966 chainCreated = true; 967 } else { 968 effect = chain->getEffectFromDesc_l(desc); 969 } 970 971 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 972 973 if (effect == 0) { 974 int id = mAudioFlinger->nextUniqueId(); 975 // Check CPU and memory usage 976 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 977 if (lStatus != NO_ERROR) { 978 goto Exit; 979 } 980 effectRegistered = true; 981 // create a new effect module if none present in the chain 982 effect = new EffectModule(this, chain, desc, id, sessionId); 983 lStatus = effect->status(); 984 if (lStatus != NO_ERROR) { 985 goto Exit; 986 } 987 effect->setOffloaded(mType == OFFLOAD, mId); 988 989 lStatus = chain->addEffect_l(effect); 990 if (lStatus != NO_ERROR) { 991 goto Exit; 992 } 993 effectCreated = true; 994 995 effect->setDevice(mOutDevice); 996 effect->setDevice(mInDevice); 997 effect->setMode(mAudioFlinger->getMode()); 998 effect->setAudioSource(mAudioSource); 999 } 1000 // create effect handle and connect it to effect module 1001 handle = new EffectHandle(effect, client, effectClient, priority); 1002 lStatus = handle->initCheck(); 1003 if (lStatus == OK) { 1004 lStatus = effect->addHandle(handle.get()); 1005 } 1006 if (enabled != NULL) { 1007 *enabled = (int)effect->isEnabled(); 1008 } 1009 } 1010 1011Exit: 1012 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1013 Mutex::Autolock _l(mLock); 1014 if (effectCreated) { 1015 chain->removeEffect_l(effect); 1016 } 1017 if (effectRegistered) { 1018 AudioSystem::unregisterEffect(effect->id()); 1019 } 1020 if (chainCreated) { 1021 removeEffectChain_l(chain); 1022 } 1023 handle.clear(); 1024 } 1025 1026 *status = lStatus; 1027 return handle; 1028} 1029 1030sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1031{ 1032 Mutex::Autolock _l(mLock); 1033 return getEffect_l(sessionId, effectId); 1034} 1035 1036sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1037{ 1038 sp<EffectChain> chain = getEffectChain_l(sessionId); 1039 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1040} 1041 1042// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1043// PlaybackThread::mLock held 1044status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1045{ 1046 // check for existing effect chain with the requested audio session 1047 int sessionId = effect->sessionId(); 1048 sp<EffectChain> chain = getEffectChain_l(sessionId); 1049 bool chainCreated = false; 1050 1051 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1052 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1053 this, effect->desc().name, effect->desc().flags); 1054 1055 if (chain == 0) { 1056 // create a new chain for this session 1057 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1058 chain = new EffectChain(this, sessionId); 1059 addEffectChain_l(chain); 1060 chain->setStrategy(getStrategyForSession_l(sessionId)); 1061 chainCreated = true; 1062 } 1063 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1064 1065 if (chain->getEffectFromId_l(effect->id()) != 0) { 1066 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1067 this, effect->desc().name, chain.get()); 1068 return BAD_VALUE; 1069 } 1070 1071 effect->setOffloaded(mType == OFFLOAD, mId); 1072 1073 status_t status = chain->addEffect_l(effect); 1074 if (status != NO_ERROR) { 1075 if (chainCreated) { 1076 removeEffectChain_l(chain); 1077 } 1078 return status; 1079 } 1080 1081 effect->setDevice(mOutDevice); 1082 effect->setDevice(mInDevice); 1083 effect->setMode(mAudioFlinger->getMode()); 1084 effect->setAudioSource(mAudioSource); 1085 return NO_ERROR; 1086} 1087 1088void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1089 1090 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1091 effect_descriptor_t desc = effect->desc(); 1092 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1093 detachAuxEffect_l(effect->id()); 1094 } 1095 1096 sp<EffectChain> chain = effect->chain().promote(); 1097 if (chain != 0) { 1098 // remove effect chain if removing last effect 1099 if (chain->removeEffect_l(effect) == 0) { 1100 removeEffectChain_l(chain); 1101 } 1102 } else { 1103 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1104 } 1105} 1106 1107void AudioFlinger::ThreadBase::lockEffectChains_l( 1108 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1109{ 1110 effectChains = mEffectChains; 1111 for (size_t i = 0; i < mEffectChains.size(); i++) { 1112 mEffectChains[i]->lock(); 1113 } 1114} 1115 1116void AudioFlinger::ThreadBase::unlockEffectChains( 1117 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1118{ 1119 for (size_t i = 0; i < effectChains.size(); i++) { 1120 effectChains[i]->unlock(); 1121 } 1122} 1123 1124sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1125{ 1126 Mutex::Autolock _l(mLock); 1127 return getEffectChain_l(sessionId); 1128} 1129 1130sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1131{ 1132 size_t size = mEffectChains.size(); 1133 for (size_t i = 0; i < size; i++) { 1134 if (mEffectChains[i]->sessionId() == sessionId) { 1135 return mEffectChains[i]; 1136 } 1137 } 1138 return 0; 1139} 1140 1141void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1142{ 1143 Mutex::Autolock _l(mLock); 1144 size_t size = mEffectChains.size(); 1145 for (size_t i = 0; i < size; i++) { 1146 mEffectChains[i]->setMode_l(mode); 1147 } 1148} 1149 1150void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1151{ 1152 config->type = AUDIO_PORT_TYPE_MIX; 1153 config->ext.mix.handle = mId; 1154 config->sample_rate = mSampleRate; 1155 config->format = mFormat; 1156 config->channel_mask = mChannelMask; 1157 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1158 AUDIO_PORT_CONFIG_FORMAT; 1159} 1160 1161 1162// ---------------------------------------------------------------------------- 1163// Playback 1164// ---------------------------------------------------------------------------- 1165 1166AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1167 AudioStreamOut* output, 1168 audio_io_handle_t id, 1169 audio_devices_t device, 1170 type_t type) 1171 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1172 mNormalFrameCount(0), mSinkBuffer(NULL), 1173 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1174 mMixerBuffer(NULL), 1175 mMixerBufferSize(0), 1176 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1177 mMixerBufferValid(false), 1178 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1179 mEffectBuffer(NULL), 1180 mEffectBufferSize(0), 1181 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1182 mEffectBufferValid(false), 1183 mSuspended(0), mBytesWritten(0), 1184 mActiveTracksGeneration(0), 1185 // mStreamTypes[] initialized in constructor body 1186 mOutput(output), 1187 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1188 mMixerStatus(MIXER_IDLE), 1189 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1190 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1191 mBytesRemaining(0), 1192 mCurrentWriteLength(0), 1193 mUseAsyncWrite(false), 1194 mWriteAckSequence(0), 1195 mDrainSequence(0), 1196 mSignalPending(false), 1197 mScreenState(AudioFlinger::mScreenState), 1198 // index 0 is reserved for normal mixer's submix 1199 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1200 // mLatchD, mLatchQ, 1201 mLatchDValid(false), mLatchQValid(false) 1202{ 1203 snprintf(mName, kNameLength, "AudioOut_%X", id); 1204 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1205 1206 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1207 // it would be safer to explicitly pass initial masterVolume/masterMute as 1208 // parameter. 1209 // 1210 // If the HAL we are using has support for master volume or master mute, 1211 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1212 // and the mute set to false). 1213 mMasterVolume = audioFlinger->masterVolume_l(); 1214 mMasterMute = audioFlinger->masterMute_l(); 1215 if (mOutput && mOutput->audioHwDev) { 1216 if (mOutput->audioHwDev->canSetMasterVolume()) { 1217 mMasterVolume = 1.0; 1218 } 1219 1220 if (mOutput->audioHwDev->canSetMasterMute()) { 1221 mMasterMute = false; 1222 } 1223 } 1224 1225 readOutputParameters_l(); 1226 1227 // ++ operator does not compile 1228 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1229 stream = (audio_stream_type_t) (stream + 1)) { 1230 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1231 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1232 } 1233} 1234 1235AudioFlinger::PlaybackThread::~PlaybackThread() 1236{ 1237 mAudioFlinger->unregisterWriter(mNBLogWriter); 1238 free(mSinkBuffer); 1239 free(mMixerBuffer); 1240 free(mEffectBuffer); 1241} 1242 1243void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1244{ 1245 dumpInternals(fd, args); 1246 dumpTracks(fd, args); 1247 dumpEffectChains(fd, args); 1248} 1249 1250void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1251{ 1252 const size_t SIZE = 256; 1253 char buffer[SIZE]; 1254 String8 result; 1255 1256 result.appendFormat(" Stream volumes in dB: "); 1257 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1258 const stream_type_t *st = &mStreamTypes[i]; 1259 if (i > 0) { 1260 result.appendFormat(", "); 1261 } 1262 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1263 if (st->mute) { 1264 result.append("M"); 1265 } 1266 } 1267 result.append("\n"); 1268 write(fd, result.string(), result.length()); 1269 result.clear(); 1270 1271 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1272 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1273 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1274 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1275 1276 size_t numtracks = mTracks.size(); 1277 size_t numactive = mActiveTracks.size(); 1278 dprintf(fd, " %d Tracks", numtracks); 1279 size_t numactiveseen = 0; 1280 if (numtracks) { 1281 dprintf(fd, " of which %d are active\n", numactive); 1282 Track::appendDumpHeader(result); 1283 for (size_t i = 0; i < numtracks; ++i) { 1284 sp<Track> track = mTracks[i]; 1285 if (track != 0) { 1286 bool active = mActiveTracks.indexOf(track) >= 0; 1287 if (active) { 1288 numactiveseen++; 1289 } 1290 track->dump(buffer, SIZE, active); 1291 result.append(buffer); 1292 } 1293 } 1294 } else { 1295 result.append("\n"); 1296 } 1297 if (numactiveseen != numactive) { 1298 // some tracks in the active list were not in the tracks list 1299 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1300 " not in the track list\n"); 1301 result.append(buffer); 1302 Track::appendDumpHeader(result); 1303 for (size_t i = 0; i < numactive; ++i) { 1304 sp<Track> track = mActiveTracks[i].promote(); 1305 if (track != 0 && mTracks.indexOf(track) < 0) { 1306 track->dump(buffer, SIZE, true); 1307 result.append(buffer); 1308 } 1309 } 1310 } 1311 1312 write(fd, result.string(), result.size()); 1313} 1314 1315void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1316{ 1317 dprintf(fd, "\nOutput thread %p:\n", this); 1318 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1319 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1320 dprintf(fd, " Total writes: %d\n", mNumWrites); 1321 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1322 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1323 dprintf(fd, " Suspend count: %d\n", mSuspended); 1324 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1325 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1326 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1327 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1328 1329 dumpBase(fd, args); 1330} 1331 1332// Thread virtuals 1333 1334void AudioFlinger::PlaybackThread::onFirstRef() 1335{ 1336 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1337} 1338 1339// ThreadBase virtuals 1340void AudioFlinger::PlaybackThread::preExit() 1341{ 1342 ALOGV(" preExit()"); 1343 // FIXME this is using hard-coded strings but in the future, this functionality will be 1344 // converted to use audio HAL extensions required to support tunneling 1345 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1346} 1347 1348// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1349sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1350 const sp<AudioFlinger::Client>& client, 1351 audio_stream_type_t streamType, 1352 uint32_t sampleRate, 1353 audio_format_t format, 1354 audio_channel_mask_t channelMask, 1355 size_t *pFrameCount, 1356 const sp<IMemory>& sharedBuffer, 1357 int sessionId, 1358 IAudioFlinger::track_flags_t *flags, 1359 pid_t tid, 1360 int uid, 1361 status_t *status) 1362{ 1363 size_t frameCount = *pFrameCount; 1364 sp<Track> track; 1365 status_t lStatus; 1366 1367 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1368 1369 // client expresses a preference for FAST, but we get the final say 1370 if (*flags & IAudioFlinger::TRACK_FAST) { 1371 if ( 1372 // not timed 1373 (!isTimed) && 1374 // either of these use cases: 1375 ( 1376 // use case 1: shared buffer with any frame count 1377 ( 1378 (sharedBuffer != 0) 1379 ) || 1380 // use case 2: callback handler and frame count is default or at least as large as HAL 1381 ( 1382 (tid != -1) && 1383 ((frameCount == 0) || 1384 (frameCount >= mFrameCount)) 1385 ) 1386 ) && 1387 // PCM data 1388 audio_is_linear_pcm(format) && 1389 // identical channel mask to sink, or mono in and stereo sink 1390 (channelMask == mChannelMask || 1391 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1392 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1393 // hardware sample rate 1394 (sampleRate == mSampleRate) && 1395 // normal mixer has an associated fast mixer 1396 hasFastMixer() && 1397 // there are sufficient fast track slots available 1398 (mFastTrackAvailMask != 0) 1399 // FIXME test that MixerThread for this fast track has a capable output HAL 1400 // FIXME add a permission test also? 1401 ) { 1402 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1403 if (frameCount == 0) { 1404 // read the fast track multiplier property the first time it is needed 1405 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1406 if (ok != 0) { 1407 ALOGE("%s pthread_once failed: %d", __func__, ok); 1408 } 1409 frameCount = mFrameCount * sFastTrackMultiplier; 1410 } 1411 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1412 frameCount, mFrameCount); 1413 } else { 1414 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1415 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1416 "sampleRate=%u mSampleRate=%u " 1417 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1418 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1419 audio_is_linear_pcm(format), 1420 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1421 *flags &= ~IAudioFlinger::TRACK_FAST; 1422 // For compatibility with AudioTrack calculation, buffer depth is forced 1423 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1424 // This is probably too conservative, but legacy application code may depend on it. 1425 // If you change this calculation, also review the start threshold which is related. 1426 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1427 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1428 if (minBufCount < 2) { 1429 minBufCount = 2; 1430 } 1431 size_t minFrameCount = mNormalFrameCount * minBufCount; 1432 if (frameCount < minFrameCount) { 1433 frameCount = minFrameCount; 1434 } 1435 } 1436 } 1437 *pFrameCount = frameCount; 1438 1439 switch (mType) { 1440 1441 case DIRECT: 1442 if (audio_is_linear_pcm(format)) { 1443 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1444 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1445 "for output %p with format %#x", 1446 sampleRate, format, channelMask, mOutput, mFormat); 1447 lStatus = BAD_VALUE; 1448 goto Exit; 1449 } 1450 } 1451 break; 1452 1453 case OFFLOAD: 1454 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1455 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1456 "for output %p with format %#x", 1457 sampleRate, format, channelMask, mOutput, mFormat); 1458 lStatus = BAD_VALUE; 1459 goto Exit; 1460 } 1461 break; 1462 1463 default: 1464 if (!audio_is_linear_pcm(format)) { 1465 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1466 "for output %p with format %#x", 1467 format, mOutput, mFormat); 1468 lStatus = BAD_VALUE; 1469 goto Exit; 1470 } 1471 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1472 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1473 lStatus = BAD_VALUE; 1474 goto Exit; 1475 } 1476 break; 1477 1478 } 1479 1480 lStatus = initCheck(); 1481 if (lStatus != NO_ERROR) { 1482 ALOGE("createTrack_l() audio driver not initialized"); 1483 goto Exit; 1484 } 1485 1486 { // scope for mLock 1487 Mutex::Autolock _l(mLock); 1488 1489 // all tracks in same audio session must share the same routing strategy otherwise 1490 // conflicts will happen when tracks are moved from one output to another by audio policy 1491 // manager 1492 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1493 for (size_t i = 0; i < mTracks.size(); ++i) { 1494 sp<Track> t = mTracks[i]; 1495 if (t != 0 && t->isExternalTrack()) { 1496 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1497 if (sessionId == t->sessionId() && strategy != actual) { 1498 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1499 strategy, actual); 1500 lStatus = BAD_VALUE; 1501 goto Exit; 1502 } 1503 } 1504 } 1505 1506 if (!isTimed) { 1507 track = new Track(this, client, streamType, sampleRate, format, 1508 channelMask, frameCount, NULL, sharedBuffer, 1509 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1510 } else { 1511 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1512 channelMask, frameCount, sharedBuffer, sessionId, uid); 1513 } 1514 1515 // new Track always returns non-NULL, 1516 // but TimedTrack::create() is a factory that could fail by returning NULL 1517 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1518 if (lStatus != NO_ERROR) { 1519 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1520 // track must be cleared from the caller as the caller has the AF lock 1521 goto Exit; 1522 } 1523 mTracks.add(track); 1524 1525 sp<EffectChain> chain = getEffectChain_l(sessionId); 1526 if (chain != 0) { 1527 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1528 track->setMainBuffer(chain->inBuffer()); 1529 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1530 chain->incTrackCnt(); 1531 } 1532 1533 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1534 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1535 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1536 // so ask activity manager to do this on our behalf 1537 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1538 } 1539 } 1540 1541 lStatus = NO_ERROR; 1542 1543Exit: 1544 *status = lStatus; 1545 return track; 1546} 1547 1548uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1549{ 1550 return latency; 1551} 1552 1553uint32_t AudioFlinger::PlaybackThread::latency() const 1554{ 1555 Mutex::Autolock _l(mLock); 1556 return latency_l(); 1557} 1558uint32_t AudioFlinger::PlaybackThread::latency_l() const 1559{ 1560 if (initCheck() == NO_ERROR) { 1561 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1562 } else { 1563 return 0; 1564 } 1565} 1566 1567void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1568{ 1569 Mutex::Autolock _l(mLock); 1570 // Don't apply master volume in SW if our HAL can do it for us. 1571 if (mOutput && mOutput->audioHwDev && 1572 mOutput->audioHwDev->canSetMasterVolume()) { 1573 mMasterVolume = 1.0; 1574 } else { 1575 mMasterVolume = value; 1576 } 1577} 1578 1579void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1580{ 1581 Mutex::Autolock _l(mLock); 1582 // Don't apply master mute in SW if our HAL can do it for us. 1583 if (mOutput && mOutput->audioHwDev && 1584 mOutput->audioHwDev->canSetMasterMute()) { 1585 mMasterMute = false; 1586 } else { 1587 mMasterMute = muted; 1588 } 1589} 1590 1591void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1592{ 1593 Mutex::Autolock _l(mLock); 1594 mStreamTypes[stream].volume = value; 1595 broadcast_l(); 1596} 1597 1598void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1599{ 1600 Mutex::Autolock _l(mLock); 1601 mStreamTypes[stream].mute = muted; 1602 broadcast_l(); 1603} 1604 1605float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1606{ 1607 Mutex::Autolock _l(mLock); 1608 return mStreamTypes[stream].volume; 1609} 1610 1611// addTrack_l() must be called with ThreadBase::mLock held 1612status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1613{ 1614 status_t status = ALREADY_EXISTS; 1615 1616 // set retry count for buffer fill 1617 track->mRetryCount = kMaxTrackStartupRetries; 1618 if (mActiveTracks.indexOf(track) < 0) { 1619 // the track is newly added, make sure it fills up all its 1620 // buffers before playing. This is to ensure the client will 1621 // effectively get the latency it requested. 1622 if (track->isExternalTrack()) { 1623 TrackBase::track_state state = track->mState; 1624 mLock.unlock(); 1625 status = AudioSystem::startOutput(mId, track->streamType(), 1626 (audio_session_t)track->sessionId()); 1627 mLock.lock(); 1628 // abort track was stopped/paused while we released the lock 1629 if (state != track->mState) { 1630 if (status == NO_ERROR) { 1631 mLock.unlock(); 1632 AudioSystem::stopOutput(mId, track->streamType(), 1633 (audio_session_t)track->sessionId()); 1634 mLock.lock(); 1635 } 1636 return INVALID_OPERATION; 1637 } 1638 // abort if start is rejected by audio policy manager 1639 if (status != NO_ERROR) { 1640 return PERMISSION_DENIED; 1641 } 1642#ifdef ADD_BATTERY_DATA 1643 // to track the speaker usage 1644 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1645#endif 1646 } 1647 1648 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1649 track->mResetDone = false; 1650 track->mPresentationCompleteFrames = 0; 1651 mActiveTracks.add(track); 1652 mWakeLockUids.add(track->uid()); 1653 mActiveTracksGeneration++; 1654 mLatestActiveTrack = track; 1655 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1656 if (chain != 0) { 1657 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1658 track->sessionId()); 1659 chain->incActiveTrackCnt(); 1660 } 1661 1662 status = NO_ERROR; 1663 } 1664 1665 onAddNewTrack_l(); 1666 return status; 1667} 1668 1669bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1670{ 1671 track->terminate(); 1672 // active tracks are removed by threadLoop() 1673 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1674 track->mState = TrackBase::STOPPED; 1675 if (!trackActive) { 1676 removeTrack_l(track); 1677 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1678 track->mState = TrackBase::STOPPING_1; 1679 } 1680 1681 return trackActive; 1682} 1683 1684void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1685{ 1686 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1687 mTracks.remove(track); 1688 deleteTrackName_l(track->name()); 1689 // redundant as track is about to be destroyed, for dumpsys only 1690 track->mName = -1; 1691 if (track->isFastTrack()) { 1692 int index = track->mFastIndex; 1693 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1694 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1695 mFastTrackAvailMask |= 1 << index; 1696 // redundant as track is about to be destroyed, for dumpsys only 1697 track->mFastIndex = -1; 1698 } 1699 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1700 if (chain != 0) { 1701 chain->decTrackCnt(); 1702 } 1703} 1704 1705void AudioFlinger::PlaybackThread::broadcast_l() 1706{ 1707 // Thread could be blocked waiting for async 1708 // so signal it to handle state changes immediately 1709 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1710 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1711 mSignalPending = true; 1712 mWaitWorkCV.broadcast(); 1713} 1714 1715String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1716{ 1717 Mutex::Autolock _l(mLock); 1718 if (initCheck() != NO_ERROR) { 1719 return String8(); 1720 } 1721 1722 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1723 const String8 out_s8(s); 1724 free(s); 1725 return out_s8; 1726} 1727 1728void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1729 AudioSystem::OutputDescriptor desc; 1730 void *param2 = NULL; 1731 1732 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1733 param); 1734 1735 switch (event) { 1736 case AudioSystem::OUTPUT_OPENED: 1737 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1738 desc.channelMask = mChannelMask; 1739 desc.samplingRate = mSampleRate; 1740 desc.format = mFormat; 1741 desc.frameCount = mNormalFrameCount; // FIXME see 1742 // AudioFlinger::frameCount(audio_io_handle_t) 1743 desc.latency = latency_l(); 1744 param2 = &desc; 1745 break; 1746 1747 case AudioSystem::STREAM_CONFIG_CHANGED: 1748 param2 = ¶m; 1749 case AudioSystem::OUTPUT_CLOSED: 1750 default: 1751 break; 1752 } 1753 mAudioFlinger->audioConfigChanged(event, mId, param2); 1754} 1755 1756void AudioFlinger::PlaybackThread::writeCallback() 1757{ 1758 ALOG_ASSERT(mCallbackThread != 0); 1759 mCallbackThread->resetWriteBlocked(); 1760} 1761 1762void AudioFlinger::PlaybackThread::drainCallback() 1763{ 1764 ALOG_ASSERT(mCallbackThread != 0); 1765 mCallbackThread->resetDraining(); 1766} 1767 1768void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1769{ 1770 Mutex::Autolock _l(mLock); 1771 // reject out of sequence requests 1772 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1773 mWriteAckSequence &= ~1; 1774 mWaitWorkCV.signal(); 1775 } 1776} 1777 1778void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1779{ 1780 Mutex::Autolock _l(mLock); 1781 // reject out of sequence requests 1782 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1783 mDrainSequence &= ~1; 1784 mWaitWorkCV.signal(); 1785 } 1786} 1787 1788// static 1789int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1790 void *param __unused, 1791 void *cookie) 1792{ 1793 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1794 ALOGV("asyncCallback() event %d", event); 1795 switch (event) { 1796 case STREAM_CBK_EVENT_WRITE_READY: 1797 me->writeCallback(); 1798 break; 1799 case STREAM_CBK_EVENT_DRAIN_READY: 1800 me->drainCallback(); 1801 break; 1802 default: 1803 ALOGW("asyncCallback() unknown event %d", event); 1804 break; 1805 } 1806 return 0; 1807} 1808 1809void AudioFlinger::PlaybackThread::readOutputParameters_l() 1810{ 1811 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1812 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1813 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1814 if (!audio_is_output_channel(mChannelMask)) { 1815 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1816 } 1817 if ((mType == MIXER || mType == DUPLICATING) 1818 && !isValidPcmSinkChannelMask(mChannelMask)) { 1819 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1820 mChannelMask); 1821 } 1822 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1823 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1824 mFormat = mHALFormat; 1825 if (!audio_is_valid_format(mFormat)) { 1826 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1827 } 1828 if ((mType == MIXER || mType == DUPLICATING) 1829 && !isValidPcmSinkFormat(mFormat)) { 1830 LOG_FATAL("HAL format %#x not supported for mixed output", 1831 mFormat); 1832 } 1833 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1834 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1835 mFrameCount = mBufferSize / mFrameSize; 1836 if (mFrameCount & 15) { 1837 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1838 mFrameCount); 1839 } 1840 1841 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1842 (mOutput->stream->set_callback != NULL)) { 1843 if (mOutput->stream->set_callback(mOutput->stream, 1844 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1845 mUseAsyncWrite = true; 1846 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1847 } 1848 } 1849 1850 // Calculate size of normal sink buffer relative to the HAL output buffer size 1851 double multiplier = 1.0; 1852 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1853 kUseFastMixer == FastMixer_Dynamic)) { 1854 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1855 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1856 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1857 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1858 maxNormalFrameCount = maxNormalFrameCount & ~15; 1859 if (maxNormalFrameCount < minNormalFrameCount) { 1860 maxNormalFrameCount = minNormalFrameCount; 1861 } 1862 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1863 if (multiplier <= 1.0) { 1864 multiplier = 1.0; 1865 } else if (multiplier <= 2.0) { 1866 if (2 * mFrameCount <= maxNormalFrameCount) { 1867 multiplier = 2.0; 1868 } else { 1869 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1870 } 1871 } else { 1872 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1873 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1874 // track, but we sometimes have to do this to satisfy the maximum frame count 1875 // constraint) 1876 // FIXME this rounding up should not be done if no HAL SRC 1877 uint32_t truncMult = (uint32_t) multiplier; 1878 if ((truncMult & 1)) { 1879 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1880 ++truncMult; 1881 } 1882 } 1883 multiplier = (double) truncMult; 1884 } 1885 } 1886 mNormalFrameCount = multiplier * mFrameCount; 1887 // round up to nearest 16 frames to satisfy AudioMixer 1888 if (mType == MIXER || mType == DUPLICATING) { 1889 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1890 } 1891 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1892 mNormalFrameCount); 1893 1894 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1895 // Originally this was int16_t[] array, need to remove legacy implications. 1896 free(mSinkBuffer); 1897 mSinkBuffer = NULL; 1898 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1899 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1900 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1901 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1902 1903 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1904 // drives the output. 1905 free(mMixerBuffer); 1906 mMixerBuffer = NULL; 1907 if (mMixerBufferEnabled) { 1908 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1909 mMixerBufferSize = mNormalFrameCount * mChannelCount 1910 * audio_bytes_per_sample(mMixerBufferFormat); 1911 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1912 } 1913 free(mEffectBuffer); 1914 mEffectBuffer = NULL; 1915 if (mEffectBufferEnabled) { 1916 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1917 mEffectBufferSize = mNormalFrameCount * mChannelCount 1918 * audio_bytes_per_sample(mEffectBufferFormat); 1919 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1920 } 1921 1922 // force reconfiguration of effect chains and engines to take new buffer size and audio 1923 // parameters into account 1924 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1925 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1926 // matter. 1927 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1928 Vector< sp<EffectChain> > effectChains = mEffectChains; 1929 for (size_t i = 0; i < effectChains.size(); i ++) { 1930 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1931 } 1932} 1933 1934 1935status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1936{ 1937 if (halFrames == NULL || dspFrames == NULL) { 1938 return BAD_VALUE; 1939 } 1940 Mutex::Autolock _l(mLock); 1941 if (initCheck() != NO_ERROR) { 1942 return INVALID_OPERATION; 1943 } 1944 size_t framesWritten = mBytesWritten / mFrameSize; 1945 *halFrames = framesWritten; 1946 1947 if (isSuspended()) { 1948 // return an estimation of rendered frames when the output is suspended 1949 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1950 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1951 return NO_ERROR; 1952 } else { 1953 status_t status; 1954 uint32_t frames; 1955 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1956 *dspFrames = (size_t)frames; 1957 return status; 1958 } 1959} 1960 1961uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1962{ 1963 Mutex::Autolock _l(mLock); 1964 uint32_t result = 0; 1965 if (getEffectChain_l(sessionId) != 0) { 1966 result = EFFECT_SESSION; 1967 } 1968 1969 for (size_t i = 0; i < mTracks.size(); ++i) { 1970 sp<Track> track = mTracks[i]; 1971 if (sessionId == track->sessionId() && !track->isInvalid()) { 1972 result |= TRACK_SESSION; 1973 break; 1974 } 1975 } 1976 1977 return result; 1978} 1979 1980uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1981{ 1982 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1983 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1984 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1985 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1986 } 1987 for (size_t i = 0; i < mTracks.size(); i++) { 1988 sp<Track> track = mTracks[i]; 1989 if (sessionId == track->sessionId() && !track->isInvalid()) { 1990 return AudioSystem::getStrategyForStream(track->streamType()); 1991 } 1992 } 1993 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1994} 1995 1996 1997AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1998{ 1999 Mutex::Autolock _l(mLock); 2000 return mOutput; 2001} 2002 2003AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2004{ 2005 Mutex::Autolock _l(mLock); 2006 AudioStreamOut *output = mOutput; 2007 mOutput = NULL; 2008 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2009 // must push a NULL and wait for ack 2010 mOutputSink.clear(); 2011 mPipeSink.clear(); 2012 mNormalSink.clear(); 2013 return output; 2014} 2015 2016// this method must always be called either with ThreadBase mLock held or inside the thread loop 2017audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2018{ 2019 if (mOutput == NULL) { 2020 return NULL; 2021 } 2022 return &mOutput->stream->common; 2023} 2024 2025uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2026{ 2027 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2028} 2029 2030status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2031{ 2032 if (!isValidSyncEvent(event)) { 2033 return BAD_VALUE; 2034 } 2035 2036 Mutex::Autolock _l(mLock); 2037 2038 for (size_t i = 0; i < mTracks.size(); ++i) { 2039 sp<Track> track = mTracks[i]; 2040 if (event->triggerSession() == track->sessionId()) { 2041 (void) track->setSyncEvent(event); 2042 return NO_ERROR; 2043 } 2044 } 2045 2046 return NAME_NOT_FOUND; 2047} 2048 2049bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2050{ 2051 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2052} 2053 2054void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2055 const Vector< sp<Track> >& tracksToRemove) 2056{ 2057 size_t count = tracksToRemove.size(); 2058 if (count > 0) { 2059 for (size_t i = 0 ; i < count ; i++) { 2060 const sp<Track>& track = tracksToRemove.itemAt(i); 2061 if (track->isExternalTrack()) { 2062 AudioSystem::stopOutput(mId, track->streamType(), 2063 (audio_session_t)track->sessionId()); 2064#ifdef ADD_BATTERY_DATA 2065 // to track the speaker usage 2066 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2067#endif 2068 if (track->isTerminated()) { 2069 AudioSystem::releaseOutput(mId, track->streamType(), 2070 (audio_session_t)track->sessionId()); 2071 } 2072 } 2073 } 2074 } 2075} 2076 2077void AudioFlinger::PlaybackThread::checkSilentMode_l() 2078{ 2079 if (!mMasterMute) { 2080 char value[PROPERTY_VALUE_MAX]; 2081 if (property_get("ro.audio.silent", value, "0") > 0) { 2082 char *endptr; 2083 unsigned long ul = strtoul(value, &endptr, 0); 2084 if (*endptr == '\0' && ul != 0) { 2085 ALOGD("Silence is golden"); 2086 // The setprop command will not allow a property to be changed after 2087 // the first time it is set, so we don't have to worry about un-muting. 2088 setMasterMute_l(true); 2089 } 2090 } 2091 } 2092} 2093 2094// shared by MIXER and DIRECT, overridden by DUPLICATING 2095ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2096{ 2097 // FIXME rewrite to reduce number of system calls 2098 mLastWriteTime = systemTime(); 2099 mInWrite = true; 2100 ssize_t bytesWritten; 2101 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2102 2103 // If an NBAIO sink is present, use it to write the normal mixer's submix 2104 if (mNormalSink != 0) { 2105 2106 const size_t count = mBytesRemaining / mFrameSize; 2107 2108 ATRACE_BEGIN("write"); 2109 // update the setpoint when AudioFlinger::mScreenState changes 2110 uint32_t screenState = AudioFlinger::mScreenState; 2111 if (screenState != mScreenState) { 2112 mScreenState = screenState; 2113 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2114 if (pipe != NULL) { 2115 pipe->setAvgFrames((mScreenState & 1) ? 2116 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2117 } 2118 } 2119 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2120 ATRACE_END(); 2121 if (framesWritten > 0) { 2122 bytesWritten = framesWritten * mFrameSize; 2123 } else { 2124 bytesWritten = framesWritten; 2125 } 2126 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2127 if (status == NO_ERROR) { 2128 size_t totalFramesWritten = mNormalSink->framesWritten(); 2129 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2130 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2131 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2132 mLatchDValid = true; 2133 } 2134 } 2135 // otherwise use the HAL / AudioStreamOut directly 2136 } else { 2137 // Direct output and offload threads 2138 2139 if (mUseAsyncWrite) { 2140 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2141 mWriteAckSequence += 2; 2142 mWriteAckSequence |= 1; 2143 ALOG_ASSERT(mCallbackThread != 0); 2144 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2145 } 2146 // FIXME We should have an implementation of timestamps for direct output threads. 2147 // They are used e.g for multichannel PCM playback over HDMI. 2148 bytesWritten = mOutput->stream->write(mOutput->stream, 2149 (char *)mSinkBuffer + offset, mBytesRemaining); 2150 if (mUseAsyncWrite && 2151 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2152 // do not wait for async callback in case of error of full write 2153 mWriteAckSequence &= ~1; 2154 ALOG_ASSERT(mCallbackThread != 0); 2155 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2156 } 2157 } 2158 2159 mNumWrites++; 2160 mInWrite = false; 2161 mStandby = false; 2162 return bytesWritten; 2163} 2164 2165void AudioFlinger::PlaybackThread::threadLoop_drain() 2166{ 2167 if (mOutput->stream->drain) { 2168 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2169 if (mUseAsyncWrite) { 2170 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2171 mDrainSequence |= 1; 2172 ALOG_ASSERT(mCallbackThread != 0); 2173 mCallbackThread->setDraining(mDrainSequence); 2174 } 2175 mOutput->stream->drain(mOutput->stream, 2176 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2177 : AUDIO_DRAIN_ALL); 2178 } 2179} 2180 2181void AudioFlinger::PlaybackThread::threadLoop_exit() 2182{ 2183 { 2184 Mutex::Autolock _l(mLock); 2185 for (size_t i = 0; i < mTracks.size(); i++) { 2186 sp<Track> track = mTracks[i]; 2187 track->invalidate(); 2188 } 2189 } 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 chain->setThread(this); 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 // Gather the framesReleased counters for all active tracks, 2431 // and latch them atomically with the timestamp. 2432 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2433 mLatchD.mFramesReleased.clear(); 2434 size_t size = mActiveTracks.size(); 2435 for (size_t i = 0; i < size; i++) { 2436 sp<Track> t = mActiveTracks[i].promote(); 2437 if (t != 0) { 2438 mLatchD.mFramesReleased.add(t.get(), 2439 t->mAudioTrackServerProxy->framesReleased()); 2440 } 2441 } 2442 if (mLatchDValid) { 2443 mLatchQ = mLatchD; 2444 mLatchDValid = false; 2445 mLatchQValid = true; 2446 } 2447 2448 saveOutputTracks(); 2449 if (mSignalPending) { 2450 // A signal was raised while we were unlocked 2451 mSignalPending = false; 2452 } else if (waitingAsyncCallback_l()) { 2453 if (exitPending()) { 2454 break; 2455 } 2456 releaseWakeLock_l(); 2457 mWakeLockUids.clear(); 2458 mActiveTracksGeneration++; 2459 ALOGV("wait async completion"); 2460 mWaitWorkCV.wait(mLock); 2461 ALOGV("async completion/wake"); 2462 acquireWakeLock_l(); 2463 standbyTime = systemTime() + standbyDelay; 2464 sleepTime = 0; 2465 2466 continue; 2467 } 2468 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2469 isSuspended()) { 2470 // put audio hardware into standby after short delay 2471 if (shouldStandby_l()) { 2472 2473 threadLoop_standby(); 2474 2475 mStandby = true; 2476 } 2477 2478 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2479 // we're about to wait, flush the binder command buffer 2480 IPCThreadState::self()->flushCommands(); 2481 2482 clearOutputTracks(); 2483 2484 if (exitPending()) { 2485 break; 2486 } 2487 2488 releaseWakeLock_l(); 2489 mWakeLockUids.clear(); 2490 mActiveTracksGeneration++; 2491 // wait until we have something to do... 2492 ALOGV("%s going to sleep", myName.string()); 2493 mWaitWorkCV.wait(mLock); 2494 ALOGV("%s waking up", myName.string()); 2495 acquireWakeLock_l(); 2496 2497 mMixerStatus = MIXER_IDLE; 2498 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2499 mBytesWritten = 0; 2500 mBytesRemaining = 0; 2501 checkSilentMode_l(); 2502 2503 standbyTime = systemTime() + standbyDelay; 2504 sleepTime = idleSleepTime; 2505 if (mType == MIXER) { 2506 sleepTimeShift = 0; 2507 } 2508 2509 continue; 2510 } 2511 } 2512 // mMixerStatusIgnoringFastTracks is also updated internally 2513 mMixerStatus = prepareTracks_l(&tracksToRemove); 2514 2515 // compare with previously applied list 2516 if (lastGeneration != mActiveTracksGeneration) { 2517 // update wakelock 2518 updateWakeLockUids_l(mWakeLockUids); 2519 lastGeneration = mActiveTracksGeneration; 2520 } 2521 2522 // prevent any changes in effect chain list and in each effect chain 2523 // during mixing and effect process as the audio buffers could be deleted 2524 // or modified if an effect is created or deleted 2525 lockEffectChains_l(effectChains); 2526 } // mLock scope ends 2527 2528 if (mBytesRemaining == 0) { 2529 mCurrentWriteLength = 0; 2530 if (mMixerStatus == MIXER_TRACKS_READY) { 2531 // threadLoop_mix() sets mCurrentWriteLength 2532 threadLoop_mix(); 2533 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2534 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2535 // threadLoop_sleepTime sets sleepTime to 0 if data 2536 // must be written to HAL 2537 threadLoop_sleepTime(); 2538 if (sleepTime == 0) { 2539 mCurrentWriteLength = mSinkBufferSize; 2540 } 2541 } 2542 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2543 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2544 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2545 // or mSinkBuffer (if there are no effects). 2546 // 2547 // This is done pre-effects computation; if effects change to 2548 // support higher precision, this needs to move. 2549 // 2550 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2551 // TODO use sleepTime == 0 as an additional condition. 2552 if (mMixerBufferValid) { 2553 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2554 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2555 2556 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2557 mNormalFrameCount * mChannelCount); 2558 } 2559 2560 mBytesRemaining = mCurrentWriteLength; 2561 if (isSuspended()) { 2562 sleepTime = suspendSleepTimeUs(); 2563 // simulate write to HAL when suspended 2564 mBytesWritten += mSinkBufferSize; 2565 mBytesRemaining = 0; 2566 } 2567 2568 // only process effects if we're going to write 2569 if (sleepTime == 0 && mType != OFFLOAD) { 2570 for (size_t i = 0; i < effectChains.size(); i ++) { 2571 effectChains[i]->process_l(); 2572 } 2573 } 2574 } 2575 // Process effect chains for offloaded thread even if no audio 2576 // was read from audio track: process only updates effect state 2577 // and thus does have to be synchronized with audio writes but may have 2578 // to be called while waiting for async write callback 2579 if (mType == OFFLOAD) { 2580 for (size_t i = 0; i < effectChains.size(); i ++) { 2581 effectChains[i]->process_l(); 2582 } 2583 } 2584 2585 // Only if the Effects buffer is enabled and there is data in the 2586 // Effects buffer (buffer valid), we need to 2587 // copy into the sink buffer. 2588 // TODO use sleepTime == 0 as an additional condition. 2589 if (mEffectBufferValid) { 2590 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2591 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2592 mNormalFrameCount * mChannelCount); 2593 } 2594 2595 // enable changes in effect chain 2596 unlockEffectChains(effectChains); 2597 2598 if (!waitingAsyncCallback()) { 2599 // sleepTime == 0 means we must write to audio hardware 2600 if (sleepTime == 0) { 2601 if (mBytesRemaining) { 2602 ssize_t ret = threadLoop_write(); 2603 if (ret < 0) { 2604 mBytesRemaining = 0; 2605 } else { 2606 mBytesWritten += ret; 2607 mBytesRemaining -= ret; 2608 } 2609 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2610 (mMixerStatus == MIXER_DRAIN_ALL)) { 2611 threadLoop_drain(); 2612 } 2613 if (mType == MIXER) { 2614 // write blocked detection 2615 nsecs_t now = systemTime(); 2616 nsecs_t delta = now - mLastWriteTime; 2617 if (!mStandby && delta > maxPeriod) { 2618 mNumDelayedWrites++; 2619 if ((now - lastWarning) > kWarningThrottleNs) { 2620 ATRACE_NAME("underrun"); 2621 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2622 ns2ms(delta), mNumDelayedWrites, this); 2623 lastWarning = now; 2624 } 2625 } 2626 } 2627 2628 } else { 2629 usleep(sleepTime); 2630 } 2631 } 2632 2633 // Finally let go of removed track(s), without the lock held 2634 // since we can't guarantee the destructors won't acquire that 2635 // same lock. This will also mutate and push a new fast mixer state. 2636 threadLoop_removeTracks(tracksToRemove); 2637 tracksToRemove.clear(); 2638 2639 // FIXME I don't understand the need for this here; 2640 // it was in the original code but maybe the 2641 // assignment in saveOutputTracks() makes this unnecessary? 2642 clearOutputTracks(); 2643 2644 // Effect chains will be actually deleted here if they were removed from 2645 // mEffectChains list during mixing or effects processing 2646 effectChains.clear(); 2647 2648 // FIXME Note that the above .clear() is no longer necessary since effectChains 2649 // is now local to this block, but will keep it for now (at least until merge done). 2650 } 2651 2652 threadLoop_exit(); 2653 2654 if (!mStandby) { 2655 threadLoop_standby(); 2656 mStandby = true; 2657 } 2658 2659 releaseWakeLock(); 2660 mWakeLockUids.clear(); 2661 mActiveTracksGeneration++; 2662 2663 ALOGV("Thread %p type %d exiting", this, mType); 2664 return false; 2665} 2666 2667// removeTracks_l() must be called with ThreadBase::mLock held 2668void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2669{ 2670 size_t count = tracksToRemove.size(); 2671 if (count > 0) { 2672 for (size_t i=0 ; i<count ; i++) { 2673 const sp<Track>& track = tracksToRemove.itemAt(i); 2674 mActiveTracks.remove(track); 2675 mWakeLockUids.remove(track->uid()); 2676 mActiveTracksGeneration++; 2677 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2678 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2679 if (chain != 0) { 2680 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2681 track->sessionId()); 2682 chain->decActiveTrackCnt(); 2683 } 2684 if (track->isTerminated()) { 2685 removeTrack_l(track); 2686 } 2687 } 2688 } 2689 2690} 2691 2692status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2693{ 2694 if (mNormalSink != 0) { 2695 return mNormalSink->getTimestamp(timestamp); 2696 } 2697 if ((mType == OFFLOAD || mType == DIRECT) 2698 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2699 uint64_t position64; 2700 int ret = mOutput->stream->get_presentation_position( 2701 mOutput->stream, &position64, ×tamp.mTime); 2702 if (ret == 0) { 2703 timestamp.mPosition = (uint32_t)position64; 2704 return NO_ERROR; 2705 } 2706 } 2707 return INVALID_OPERATION; 2708} 2709 2710status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2711 audio_patch_handle_t *handle) 2712{ 2713 status_t status = NO_ERROR; 2714 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2715 // store new device and send to effects 2716 audio_devices_t type = AUDIO_DEVICE_NONE; 2717 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2718 type |= patch->sinks[i].ext.device.type; 2719 } 2720 mOutDevice = type; 2721 for (size_t i = 0; i < mEffectChains.size(); i++) { 2722 mEffectChains[i]->setDevice_l(mOutDevice); 2723 } 2724 2725 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2726 status = hwDevice->create_audio_patch(hwDevice, 2727 patch->num_sources, 2728 patch->sources, 2729 patch->num_sinks, 2730 patch->sinks, 2731 handle); 2732 } else { 2733 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2734 } 2735 return status; 2736} 2737 2738status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2739{ 2740 status_t status = NO_ERROR; 2741 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2742 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2743 status = hwDevice->release_audio_patch(hwDevice, handle); 2744 } else { 2745 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2746 } 2747 return status; 2748} 2749 2750void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2751{ 2752 Mutex::Autolock _l(mLock); 2753 mTracks.add(track); 2754} 2755 2756void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2757{ 2758 Mutex::Autolock _l(mLock); 2759 destroyTrack_l(track); 2760} 2761 2762void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2763{ 2764 ThreadBase::getAudioPortConfig(config); 2765 config->role = AUDIO_PORT_ROLE_SOURCE; 2766 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2767 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2768} 2769 2770// ---------------------------------------------------------------------------- 2771 2772AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2773 audio_io_handle_t id, audio_devices_t device, type_t type) 2774 : PlaybackThread(audioFlinger, output, id, device, type), 2775 // mAudioMixer below 2776 // mFastMixer below 2777 mFastMixerFutex(0) 2778 // mOutputSink below 2779 // mPipeSink below 2780 // mNormalSink below 2781{ 2782 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2783 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2784 "mFrameCount=%d, mNormalFrameCount=%d", 2785 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2786 mNormalFrameCount); 2787 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2788 2789 // create an NBAIO sink for the HAL output stream, and negotiate 2790 mOutputSink = new AudioStreamOutSink(output->stream); 2791 size_t numCounterOffers = 0; 2792 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2793 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2794 ALOG_ASSERT(index == 0); 2795 2796 // initialize fast mixer depending on configuration 2797 bool initFastMixer; 2798 switch (kUseFastMixer) { 2799 case FastMixer_Never: 2800 initFastMixer = false; 2801 break; 2802 case FastMixer_Always: 2803 initFastMixer = true; 2804 break; 2805 case FastMixer_Static: 2806 case FastMixer_Dynamic: 2807 initFastMixer = mFrameCount < mNormalFrameCount; 2808 break; 2809 } 2810 if (initFastMixer) { 2811 audio_format_t fastMixerFormat; 2812 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2813 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2814 } else { 2815 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2816 } 2817 if (mFormat != fastMixerFormat) { 2818 // change our Sink format to accept our intermediate precision 2819 mFormat = fastMixerFormat; 2820 free(mSinkBuffer); 2821 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2822 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2823 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2824 } 2825 2826 // create a MonoPipe to connect our submix to FastMixer 2827 NBAIO_Format format = mOutputSink->format(); 2828 NBAIO_Format origformat = format; 2829 // adjust format to match that of the Fast Mixer 2830 format.mFormat = fastMixerFormat; 2831 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2832 2833 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2834 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2835 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2836 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2837 const NBAIO_Format offers[1] = {format}; 2838 size_t numCounterOffers = 0; 2839 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2840 ALOG_ASSERT(index == 0); 2841 monoPipe->setAvgFrames((mScreenState & 1) ? 2842 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2843 mPipeSink = monoPipe; 2844 2845#ifdef TEE_SINK 2846 if (mTeeSinkOutputEnabled) { 2847 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2848 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 2849 const NBAIO_Format offers2[1] = {origformat}; 2850 numCounterOffers = 0; 2851 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 2852 ALOG_ASSERT(index == 0); 2853 mTeeSink = teeSink; 2854 PipeReader *teeSource = new PipeReader(*teeSink); 2855 numCounterOffers = 0; 2856 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 2857 ALOG_ASSERT(index == 0); 2858 mTeeSource = teeSource; 2859 } 2860#endif 2861 2862 // create fast mixer and configure it initially with just one fast track for our submix 2863 mFastMixer = new FastMixer(); 2864 FastMixerStateQueue *sq = mFastMixer->sq(); 2865#ifdef STATE_QUEUE_DUMP 2866 sq->setObserverDump(&mStateQueueObserverDump); 2867 sq->setMutatorDump(&mStateQueueMutatorDump); 2868#endif 2869 FastMixerState *state = sq->begin(); 2870 FastTrack *fastTrack = &state->mFastTracks[0]; 2871 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2872 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2873 fastTrack->mVolumeProvider = NULL; 2874 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2875 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2876 fastTrack->mGeneration++; 2877 state->mFastTracksGen++; 2878 state->mTrackMask = 1; 2879 // fast mixer will use the HAL output sink 2880 state->mOutputSink = mOutputSink.get(); 2881 state->mOutputSinkGen++; 2882 state->mFrameCount = mFrameCount; 2883 state->mCommand = FastMixerState::COLD_IDLE; 2884 // already done in constructor initialization list 2885 //mFastMixerFutex = 0; 2886 state->mColdFutexAddr = &mFastMixerFutex; 2887 state->mColdGen++; 2888 state->mDumpState = &mFastMixerDumpState; 2889#ifdef TEE_SINK 2890 state->mTeeSink = mTeeSink.get(); 2891#endif 2892 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2893 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2894 sq->end(); 2895 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2896 2897 // start the fast mixer 2898 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2899 pid_t tid = mFastMixer->getTid(); 2900 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2901 if (err != 0) { 2902 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2903 kPriorityFastMixer, getpid_cached, tid, err); 2904 } 2905 2906#ifdef AUDIO_WATCHDOG 2907 // create and start the watchdog 2908 mAudioWatchdog = new AudioWatchdog(); 2909 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2910 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2911 tid = mAudioWatchdog->getTid(); 2912 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2913 if (err != 0) { 2914 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2915 kPriorityFastMixer, getpid_cached, tid, err); 2916 } 2917#endif 2918 2919 } 2920 2921 switch (kUseFastMixer) { 2922 case FastMixer_Never: 2923 case FastMixer_Dynamic: 2924 mNormalSink = mOutputSink; 2925 break; 2926 case FastMixer_Always: 2927 mNormalSink = mPipeSink; 2928 break; 2929 case FastMixer_Static: 2930 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2931 break; 2932 } 2933} 2934 2935AudioFlinger::MixerThread::~MixerThread() 2936{ 2937 if (mFastMixer != 0) { 2938 FastMixerStateQueue *sq = mFastMixer->sq(); 2939 FastMixerState *state = sq->begin(); 2940 if (state->mCommand == FastMixerState::COLD_IDLE) { 2941 int32_t old = android_atomic_inc(&mFastMixerFutex); 2942 if (old == -1) { 2943 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2944 } 2945 } 2946 state->mCommand = FastMixerState::EXIT; 2947 sq->end(); 2948 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2949 mFastMixer->join(); 2950 // Though the fast mixer thread has exited, it's state queue is still valid. 2951 // We'll use that extract the final state which contains one remaining fast track 2952 // corresponding to our sub-mix. 2953 state = sq->begin(); 2954 ALOG_ASSERT(state->mTrackMask == 1); 2955 FastTrack *fastTrack = &state->mFastTracks[0]; 2956 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2957 delete fastTrack->mBufferProvider; 2958 sq->end(false /*didModify*/); 2959 mFastMixer.clear(); 2960#ifdef AUDIO_WATCHDOG 2961 if (mAudioWatchdog != 0) { 2962 mAudioWatchdog->requestExit(); 2963 mAudioWatchdog->requestExitAndWait(); 2964 mAudioWatchdog.clear(); 2965 } 2966#endif 2967 } 2968 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2969 delete mAudioMixer; 2970} 2971 2972 2973uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2974{ 2975 if (mFastMixer != 0) { 2976 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2977 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2978 } 2979 return latency; 2980} 2981 2982 2983void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2984{ 2985 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2986} 2987 2988ssize_t AudioFlinger::MixerThread::threadLoop_write() 2989{ 2990 // FIXME we should only do one push per cycle; confirm this is true 2991 // Start the fast mixer if it's not already running 2992 if (mFastMixer != 0) { 2993 FastMixerStateQueue *sq = mFastMixer->sq(); 2994 FastMixerState *state = sq->begin(); 2995 if (state->mCommand != FastMixerState::MIX_WRITE && 2996 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2997 if (state->mCommand == FastMixerState::COLD_IDLE) { 2998 int32_t old = android_atomic_inc(&mFastMixerFutex); 2999 if (old == -1) { 3000 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3001 } 3002#ifdef AUDIO_WATCHDOG 3003 if (mAudioWatchdog != 0) { 3004 mAudioWatchdog->resume(); 3005 } 3006#endif 3007 } 3008 state->mCommand = FastMixerState::MIX_WRITE; 3009 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3010 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 3011 sq->end(); 3012 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3013 if (kUseFastMixer == FastMixer_Dynamic) { 3014 mNormalSink = mPipeSink; 3015 } 3016 } else { 3017 sq->end(false /*didModify*/); 3018 } 3019 } 3020 return PlaybackThread::threadLoop_write(); 3021} 3022 3023void AudioFlinger::MixerThread::threadLoop_standby() 3024{ 3025 // Idle the fast mixer if it's currently running 3026 if (mFastMixer != 0) { 3027 FastMixerStateQueue *sq = mFastMixer->sq(); 3028 FastMixerState *state = sq->begin(); 3029 if (!(state->mCommand & FastMixerState::IDLE)) { 3030 state->mCommand = FastMixerState::COLD_IDLE; 3031 state->mColdFutexAddr = &mFastMixerFutex; 3032 state->mColdGen++; 3033 mFastMixerFutex = 0; 3034 sq->end(); 3035 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3036 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3037 if (kUseFastMixer == FastMixer_Dynamic) { 3038 mNormalSink = mOutputSink; 3039 } 3040#ifdef AUDIO_WATCHDOG 3041 if (mAudioWatchdog != 0) { 3042 mAudioWatchdog->pause(); 3043 } 3044#endif 3045 } else { 3046 sq->end(false /*didModify*/); 3047 } 3048 } 3049 PlaybackThread::threadLoop_standby(); 3050} 3051 3052bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3053{ 3054 return false; 3055} 3056 3057bool AudioFlinger::PlaybackThread::shouldStandby_l() 3058{ 3059 return !mStandby; 3060} 3061 3062bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3063{ 3064 Mutex::Autolock _l(mLock); 3065 return waitingAsyncCallback_l(); 3066} 3067 3068// shared by MIXER and DIRECT, overridden by DUPLICATING 3069void AudioFlinger::PlaybackThread::threadLoop_standby() 3070{ 3071 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3072 mOutput->stream->common.standby(&mOutput->stream->common); 3073 if (mUseAsyncWrite != 0) { 3074 // discard any pending drain or write ack by incrementing sequence 3075 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3076 mDrainSequence = (mDrainSequence + 2) & ~1; 3077 ALOG_ASSERT(mCallbackThread != 0); 3078 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3079 mCallbackThread->setDraining(mDrainSequence); 3080 } 3081} 3082 3083void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3084{ 3085 ALOGV("signal playback thread"); 3086 broadcast_l(); 3087} 3088 3089void AudioFlinger::MixerThread::threadLoop_mix() 3090{ 3091 // obtain the presentation timestamp of the next output buffer 3092 int64_t pts; 3093 status_t status = INVALID_OPERATION; 3094 3095 if (mNormalSink != 0) { 3096 status = mNormalSink->getNextWriteTimestamp(&pts); 3097 } else { 3098 status = mOutputSink->getNextWriteTimestamp(&pts); 3099 } 3100 3101 if (status != NO_ERROR) { 3102 pts = AudioBufferProvider::kInvalidPTS; 3103 } 3104 3105 // mix buffers... 3106 mAudioMixer->process(pts); 3107 mCurrentWriteLength = mSinkBufferSize; 3108 // increase sleep time progressively when application underrun condition clears. 3109 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3110 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3111 // such that we would underrun the audio HAL. 3112 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3113 sleepTimeShift--; 3114 } 3115 sleepTime = 0; 3116 standbyTime = systemTime() + standbyDelay; 3117 //TODO: delay standby when effects have a tail 3118 3119} 3120 3121void AudioFlinger::MixerThread::threadLoop_sleepTime() 3122{ 3123 // If no tracks are ready, sleep once for the duration of an output 3124 // buffer size, then write 0s to the output 3125 if (sleepTime == 0) { 3126 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3127 sleepTime = activeSleepTime >> sleepTimeShift; 3128 if (sleepTime < kMinThreadSleepTimeUs) { 3129 sleepTime = kMinThreadSleepTimeUs; 3130 } 3131 // reduce sleep time in case of consecutive application underruns to avoid 3132 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3133 // duration we would end up writing less data than needed by the audio HAL if 3134 // the condition persists. 3135 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3136 sleepTimeShift++; 3137 } 3138 } else { 3139 sleepTime = idleSleepTime; 3140 } 3141 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3142 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3143 // before effects processing or output. 3144 if (mMixerBufferValid) { 3145 memset(mMixerBuffer, 0, mMixerBufferSize); 3146 } else { 3147 memset(mSinkBuffer, 0, mSinkBufferSize); 3148 } 3149 sleepTime = 0; 3150 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3151 "anticipated start"); 3152 } 3153 // TODO add standby time extension fct of effect tail 3154} 3155 3156// prepareTracks_l() must be called with ThreadBase::mLock held 3157AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3158 Vector< sp<Track> > *tracksToRemove) 3159{ 3160 3161 mixer_state mixerStatus = MIXER_IDLE; 3162 // find out which tracks need to be processed 3163 size_t count = mActiveTracks.size(); 3164 size_t mixedTracks = 0; 3165 size_t tracksWithEffect = 0; 3166 // counts only _active_ fast tracks 3167 size_t fastTracks = 0; 3168 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3169 3170 float masterVolume = mMasterVolume; 3171 bool masterMute = mMasterMute; 3172 3173 if (masterMute) { 3174 masterVolume = 0; 3175 } 3176 // Delegate master volume control to effect in output mix effect chain if needed 3177 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3178 if (chain != 0) { 3179 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3180 chain->setVolume_l(&v, &v); 3181 masterVolume = (float)((v + (1 << 23)) >> 24); 3182 chain.clear(); 3183 } 3184 3185 // prepare a new state to push 3186 FastMixerStateQueue *sq = NULL; 3187 FastMixerState *state = NULL; 3188 bool didModify = false; 3189 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3190 if (mFastMixer != 0) { 3191 sq = mFastMixer->sq(); 3192 state = sq->begin(); 3193 } 3194 3195 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3196 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3197 3198 for (size_t i=0 ; i<count ; i++) { 3199 const sp<Track> t = mActiveTracks[i].promote(); 3200 if (t == 0) { 3201 continue; 3202 } 3203 3204 // this const just means the local variable doesn't change 3205 Track* const track = t.get(); 3206 3207 // process fast tracks 3208 if (track->isFastTrack()) { 3209 3210 // It's theoretically possible (though unlikely) for a fast track to be created 3211 // and then removed within the same normal mix cycle. This is not a problem, as 3212 // the track never becomes active so it's fast mixer slot is never touched. 3213 // The converse, of removing an (active) track and then creating a new track 3214 // at the identical fast mixer slot within the same normal mix cycle, 3215 // is impossible because the slot isn't marked available until the end of each cycle. 3216 int j = track->mFastIndex; 3217 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3218 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3219 FastTrack *fastTrack = &state->mFastTracks[j]; 3220 3221 // Determine whether the track is currently in underrun condition, 3222 // and whether it had a recent underrun. 3223 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3224 FastTrackUnderruns underruns = ftDump->mUnderruns; 3225 uint32_t recentFull = (underruns.mBitFields.mFull - 3226 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3227 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3228 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3229 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3230 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3231 uint32_t recentUnderruns = recentPartial + recentEmpty; 3232 track->mObservedUnderruns = underruns; 3233 // don't count underruns that occur while stopping or pausing 3234 // or stopped which can occur when flush() is called while active 3235 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3236 recentUnderruns > 0) { 3237 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3238 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3239 } 3240 3241 // This is similar to the state machine for normal tracks, 3242 // with a few modifications for fast tracks. 3243 bool isActive = true; 3244 switch (track->mState) { 3245 case TrackBase::STOPPING_1: 3246 // track stays active in STOPPING_1 state until first underrun 3247 if (recentUnderruns > 0 || track->isTerminated()) { 3248 track->mState = TrackBase::STOPPING_2; 3249 } 3250 break; 3251 case TrackBase::PAUSING: 3252 // ramp down is not yet implemented 3253 track->setPaused(); 3254 break; 3255 case TrackBase::RESUMING: 3256 // ramp up is not yet implemented 3257 track->mState = TrackBase::ACTIVE; 3258 break; 3259 case TrackBase::ACTIVE: 3260 if (recentFull > 0 || recentPartial > 0) { 3261 // track has provided at least some frames recently: reset retry count 3262 track->mRetryCount = kMaxTrackRetries; 3263 } 3264 if (recentUnderruns == 0) { 3265 // no recent underruns: stay active 3266 break; 3267 } 3268 // there has recently been an underrun of some kind 3269 if (track->sharedBuffer() == 0) { 3270 // were any of the recent underruns "empty" (no frames available)? 3271 if (recentEmpty == 0) { 3272 // no, then ignore the partial underruns as they are allowed indefinitely 3273 break; 3274 } 3275 // there has recently been an "empty" underrun: decrement the retry counter 3276 if (--(track->mRetryCount) > 0) { 3277 break; 3278 } 3279 // indicate to client process that the track was disabled because of underrun; 3280 // it will then automatically call start() when data is available 3281 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3282 // remove from active list, but state remains ACTIVE [confusing but true] 3283 isActive = false; 3284 break; 3285 } 3286 // fall through 3287 case TrackBase::STOPPING_2: 3288 case TrackBase::PAUSED: 3289 case TrackBase::STOPPED: 3290 case TrackBase::FLUSHED: // flush() while active 3291 // Check for presentation complete if track is inactive 3292 // We have consumed all the buffers of this track. 3293 // This would be incomplete if we auto-paused on underrun 3294 { 3295 size_t audioHALFrames = 3296 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3297 size_t framesWritten = mBytesWritten / mFrameSize; 3298 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3299 // track stays in active list until presentation is complete 3300 break; 3301 } 3302 } 3303 if (track->isStopping_2()) { 3304 track->mState = TrackBase::STOPPED; 3305 } 3306 if (track->isStopped()) { 3307 // Can't reset directly, as fast mixer is still polling this track 3308 // track->reset(); 3309 // So instead mark this track as needing to be reset after push with ack 3310 resetMask |= 1 << i; 3311 } 3312 isActive = false; 3313 break; 3314 case TrackBase::IDLE: 3315 default: 3316 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3317 } 3318 3319 if (isActive) { 3320 // was it previously inactive? 3321 if (!(state->mTrackMask & (1 << j))) { 3322 ExtendedAudioBufferProvider *eabp = track; 3323 VolumeProvider *vp = track; 3324 fastTrack->mBufferProvider = eabp; 3325 fastTrack->mVolumeProvider = vp; 3326 fastTrack->mChannelMask = track->mChannelMask; 3327 fastTrack->mFormat = track->mFormat; 3328 fastTrack->mGeneration++; 3329 state->mTrackMask |= 1 << j; 3330 didModify = true; 3331 // no acknowledgement required for newly active tracks 3332 } 3333 // cache the combined master volume and stream type volume for fast mixer; this 3334 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3335 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3336 ++fastTracks; 3337 } else { 3338 // was it previously active? 3339 if (state->mTrackMask & (1 << j)) { 3340 fastTrack->mBufferProvider = NULL; 3341 fastTrack->mGeneration++; 3342 state->mTrackMask &= ~(1 << j); 3343 didModify = true; 3344 // If any fast tracks were removed, we must wait for acknowledgement 3345 // because we're about to decrement the last sp<> on those tracks. 3346 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3347 } else { 3348 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3349 } 3350 tracksToRemove->add(track); 3351 // Avoids a misleading display in dumpsys 3352 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3353 } 3354 continue; 3355 } 3356 3357 { // local variable scope to avoid goto warning 3358 3359 audio_track_cblk_t* cblk = track->cblk(); 3360 3361 // The first time a track is added we wait 3362 // for all its buffers to be filled before processing it 3363 int name = track->name(); 3364 // make sure that we have enough frames to mix one full buffer. 3365 // enforce this condition only once to enable draining the buffer in case the client 3366 // app does not call stop() and relies on underrun to stop: 3367 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3368 // during last round 3369 size_t desiredFrames; 3370 uint32_t sr = track->sampleRate(); 3371 if (sr == mSampleRate) { 3372 desiredFrames = mNormalFrameCount; 3373 } else { 3374 // +1 for rounding and +1 for additional sample needed for interpolation 3375 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3376 // add frames already consumed but not yet released by the resampler 3377 // because mAudioTrackServerProxy->framesReady() will include these frames 3378 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3379#if 0 3380 // the minimum track buffer size is normally twice the number of frames necessary 3381 // to fill one buffer and the resampler should not leave more than one buffer worth 3382 // of unreleased frames after each pass, but just in case... 3383 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3384#endif 3385 } 3386 uint32_t minFrames = 1; 3387 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3388 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3389 minFrames = desiredFrames; 3390 } 3391 3392 size_t framesReady = track->framesReady(); 3393 if ((framesReady >= minFrames) && track->isReady() && 3394 !track->isPaused() && !track->isTerminated()) 3395 { 3396 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3397 3398 mixedTracks++; 3399 3400 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3401 // there is an effect chain connected to the track 3402 chain.clear(); 3403 if (track->mainBuffer() != mSinkBuffer && 3404 track->mainBuffer() != mMixerBuffer) { 3405 if (mEffectBufferEnabled) { 3406 mEffectBufferValid = true; // Later can set directly. 3407 } 3408 chain = getEffectChain_l(track->sessionId()); 3409 // Delegate volume control to effect in track effect chain if needed 3410 if (chain != 0) { 3411 tracksWithEffect++; 3412 } else { 3413 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3414 "session %d", 3415 name, track->sessionId()); 3416 } 3417 } 3418 3419 3420 int param = AudioMixer::VOLUME; 3421 if (track->mFillingUpStatus == Track::FS_FILLED) { 3422 // no ramp for the first volume setting 3423 track->mFillingUpStatus = Track::FS_ACTIVE; 3424 if (track->mState == TrackBase::RESUMING) { 3425 track->mState = TrackBase::ACTIVE; 3426 param = AudioMixer::RAMP_VOLUME; 3427 } 3428 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3429 // FIXME should not make a decision based on mServer 3430 } else if (cblk->mServer != 0) { 3431 // If the track is stopped before the first frame was mixed, 3432 // do not apply ramp 3433 param = AudioMixer::RAMP_VOLUME; 3434 } 3435 3436 // compute volume for this track 3437 uint32_t vl, vr; // in U8.24 integer format 3438 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3439 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3440 vl = vr = 0; 3441 vlf = vrf = vaf = 0.; 3442 if (track->isPausing()) { 3443 track->setPaused(); 3444 } 3445 } else { 3446 3447 // read original volumes with volume control 3448 float typeVolume = mStreamTypes[track->streamType()].volume; 3449 float v = masterVolume * typeVolume; 3450 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3451 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3452 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3453 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3454 // track volumes come from shared memory, so can't be trusted and must be clamped 3455 if (vlf > GAIN_FLOAT_UNITY) { 3456 ALOGV("Track left volume out of range: %.3g", vlf); 3457 vlf = GAIN_FLOAT_UNITY; 3458 } 3459 if (vrf > GAIN_FLOAT_UNITY) { 3460 ALOGV("Track right volume out of range: %.3g", vrf); 3461 vrf = GAIN_FLOAT_UNITY; 3462 } 3463 // now apply the master volume and stream type volume 3464 vlf *= v; 3465 vrf *= v; 3466 // assuming master volume and stream type volume each go up to 1.0, 3467 // then derive vl and vr as U8.24 versions for the effect chain 3468 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3469 vl = (uint32_t) (scaleto8_24 * vlf); 3470 vr = (uint32_t) (scaleto8_24 * vrf); 3471 // vl and vr are now in U8.24 format 3472 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3473 // send level comes from shared memory and so may be corrupt 3474 if (sendLevel > MAX_GAIN_INT) { 3475 ALOGV("Track send level out of range: %04X", sendLevel); 3476 sendLevel = MAX_GAIN_INT; 3477 } 3478 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3479 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3480 } 3481 3482 // Delegate volume control to effect in track effect chain if needed 3483 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3484 // Do not ramp volume if volume is controlled by effect 3485 param = AudioMixer::VOLUME; 3486 // Update remaining floating point volume levels 3487 vlf = (float)vl / (1 << 24); 3488 vrf = (float)vr / (1 << 24); 3489 track->mHasVolumeController = true; 3490 } else { 3491 // force no volume ramp when volume controller was just disabled or removed 3492 // from effect chain to avoid volume spike 3493 if (track->mHasVolumeController) { 3494 param = AudioMixer::VOLUME; 3495 } 3496 track->mHasVolumeController = false; 3497 } 3498 3499 // XXX: these things DON'T need to be done each time 3500 mAudioMixer->setBufferProvider(name, track); 3501 mAudioMixer->enable(name); 3502 3503 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3504 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3505 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3506 mAudioMixer->setParameter( 3507 name, 3508 AudioMixer::TRACK, 3509 AudioMixer::FORMAT, (void *)track->format()); 3510 mAudioMixer->setParameter( 3511 name, 3512 AudioMixer::TRACK, 3513 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3514 mAudioMixer->setParameter( 3515 name, 3516 AudioMixer::TRACK, 3517 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3518 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3519 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3520 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3521 if (reqSampleRate == 0) { 3522 reqSampleRate = mSampleRate; 3523 } else if (reqSampleRate > maxSampleRate) { 3524 reqSampleRate = maxSampleRate; 3525 } 3526 mAudioMixer->setParameter( 3527 name, 3528 AudioMixer::RESAMPLE, 3529 AudioMixer::SAMPLE_RATE, 3530 (void *)(uintptr_t)reqSampleRate); 3531 /* 3532 * Select the appropriate output buffer for the track. 3533 * 3534 * Tracks with effects go into their own effects chain buffer 3535 * and from there into either mEffectBuffer or mSinkBuffer. 3536 * 3537 * Other tracks can use mMixerBuffer for higher precision 3538 * channel accumulation. If this buffer is enabled 3539 * (mMixerBufferEnabled true), then selected tracks will accumulate 3540 * into it. 3541 * 3542 */ 3543 if (mMixerBufferEnabled 3544 && (track->mainBuffer() == mSinkBuffer 3545 || track->mainBuffer() == mMixerBuffer)) { 3546 mAudioMixer->setParameter( 3547 name, 3548 AudioMixer::TRACK, 3549 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3550 mAudioMixer->setParameter( 3551 name, 3552 AudioMixer::TRACK, 3553 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3554 // TODO: override track->mainBuffer()? 3555 mMixerBufferValid = true; 3556 } else { 3557 mAudioMixer->setParameter( 3558 name, 3559 AudioMixer::TRACK, 3560 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3561 mAudioMixer->setParameter( 3562 name, 3563 AudioMixer::TRACK, 3564 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3565 } 3566 mAudioMixer->setParameter( 3567 name, 3568 AudioMixer::TRACK, 3569 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3570 3571 // reset retry count 3572 track->mRetryCount = kMaxTrackRetries; 3573 3574 // If one track is ready, set the mixer ready if: 3575 // - the mixer was not ready during previous round OR 3576 // - no other track is not ready 3577 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3578 mixerStatus != MIXER_TRACKS_ENABLED) { 3579 mixerStatus = MIXER_TRACKS_READY; 3580 } 3581 } else { 3582 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3583 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3584 } 3585 // clear effect chain input buffer if an active track underruns to avoid sending 3586 // previous audio buffer again to effects 3587 chain = getEffectChain_l(track->sessionId()); 3588 if (chain != 0) { 3589 chain->clearInputBuffer(); 3590 } 3591 3592 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3593 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3594 track->isStopped() || track->isPaused()) { 3595 // We have consumed all the buffers of this track. 3596 // Remove it from the list of active tracks. 3597 // TODO: use actual buffer filling status instead of latency when available from 3598 // audio HAL 3599 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3600 size_t framesWritten = mBytesWritten / mFrameSize; 3601 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3602 if (track->isStopped()) { 3603 track->reset(); 3604 } 3605 tracksToRemove->add(track); 3606 } 3607 } else { 3608 // No buffers for this track. Give it a few chances to 3609 // fill a buffer, then remove it from active list. 3610 if (--(track->mRetryCount) <= 0) { 3611 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3612 tracksToRemove->add(track); 3613 // indicate to client process that the track was disabled because of underrun; 3614 // it will then automatically call start() when data is available 3615 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3616 // If one track is not ready, mark the mixer also not ready if: 3617 // - the mixer was ready during previous round OR 3618 // - no other track is ready 3619 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3620 mixerStatus != MIXER_TRACKS_READY) { 3621 mixerStatus = MIXER_TRACKS_ENABLED; 3622 } 3623 } 3624 mAudioMixer->disable(name); 3625 } 3626 3627 } // local variable scope to avoid goto warning 3628track_is_ready: ; 3629 3630 } 3631 3632 // Push the new FastMixer state if necessary 3633 bool pauseAudioWatchdog = false; 3634 if (didModify) { 3635 state->mFastTracksGen++; 3636 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3637 if (kUseFastMixer == FastMixer_Dynamic && 3638 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3639 state->mCommand = FastMixerState::COLD_IDLE; 3640 state->mColdFutexAddr = &mFastMixerFutex; 3641 state->mColdGen++; 3642 mFastMixerFutex = 0; 3643 if (kUseFastMixer == FastMixer_Dynamic) { 3644 mNormalSink = mOutputSink; 3645 } 3646 // If we go into cold idle, need to wait for acknowledgement 3647 // so that fast mixer stops doing I/O. 3648 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3649 pauseAudioWatchdog = true; 3650 } 3651 } 3652 if (sq != NULL) { 3653 sq->end(didModify); 3654 sq->push(block); 3655 } 3656#ifdef AUDIO_WATCHDOG 3657 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3658 mAudioWatchdog->pause(); 3659 } 3660#endif 3661 3662 // Now perform the deferred reset on fast tracks that have stopped 3663 while (resetMask != 0) { 3664 size_t i = __builtin_ctz(resetMask); 3665 ALOG_ASSERT(i < count); 3666 resetMask &= ~(1 << i); 3667 sp<Track> t = mActiveTracks[i].promote(); 3668 if (t == 0) { 3669 continue; 3670 } 3671 Track* track = t.get(); 3672 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3673 track->reset(); 3674 } 3675 3676 // remove all the tracks that need to be... 3677 removeTracks_l(*tracksToRemove); 3678 3679 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3680 mEffectBufferValid = true; 3681 } 3682 3683 if (mEffectBufferValid) { 3684 // as long as there are effects we should clear the effects buffer, to avoid 3685 // passing a non-clean buffer to the effect chain 3686 memset(mEffectBuffer, 0, mEffectBufferSize); 3687 } 3688 // sink or mix buffer must be cleared if all tracks are connected to an 3689 // effect chain as in this case the mixer will not write to the sink or mix buffer 3690 // and track effects will accumulate into it 3691 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3692 (mixedTracks == 0 && fastTracks > 0))) { 3693 // FIXME as a performance optimization, should remember previous zero status 3694 if (mMixerBufferValid) { 3695 memset(mMixerBuffer, 0, mMixerBufferSize); 3696 // TODO: In testing, mSinkBuffer below need not be cleared because 3697 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3698 // after mixing. 3699 // 3700 // To enforce this guarantee: 3701 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3702 // (mixedTracks == 0 && fastTracks > 0)) 3703 // must imply MIXER_TRACKS_READY. 3704 // Later, we may clear buffers regardless, and skip much of this logic. 3705 } 3706 // FIXME as a performance optimization, should remember previous zero status 3707 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3708 } 3709 3710 // if any fast tracks, then status is ready 3711 mMixerStatusIgnoringFastTracks = mixerStatus; 3712 if (fastTracks > 0) { 3713 mixerStatus = MIXER_TRACKS_READY; 3714 } 3715 return mixerStatus; 3716} 3717 3718// getTrackName_l() must be called with ThreadBase::mLock held 3719int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3720 audio_format_t format, int sessionId) 3721{ 3722 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3723} 3724 3725// deleteTrackName_l() must be called with ThreadBase::mLock held 3726void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3727{ 3728 ALOGV("remove track (%d) and delete from mixer", name); 3729 mAudioMixer->deleteTrackName(name); 3730} 3731 3732// checkForNewParameter_l() must be called with ThreadBase::mLock held 3733bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3734 status_t& status) 3735{ 3736 bool reconfig = false; 3737 3738 status = NO_ERROR; 3739 3740 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3741 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3742 if (mFastMixer != 0) { 3743 FastMixerStateQueue *sq = mFastMixer->sq(); 3744 FastMixerState *state = sq->begin(); 3745 if (!(state->mCommand & FastMixerState::IDLE)) { 3746 previousCommand = state->mCommand; 3747 state->mCommand = FastMixerState::HOT_IDLE; 3748 sq->end(); 3749 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3750 } else { 3751 sq->end(false /*didModify*/); 3752 } 3753 } 3754 3755 AudioParameter param = AudioParameter(keyValuePair); 3756 int value; 3757 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3758 reconfig = true; 3759 } 3760 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3761 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3762 status = BAD_VALUE; 3763 } else { 3764 // no need to save value, since it's constant 3765 reconfig = true; 3766 } 3767 } 3768 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3769 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3770 status = BAD_VALUE; 3771 } else { 3772 // no need to save value, since it's constant 3773 reconfig = true; 3774 } 3775 } 3776 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3777 // do not accept frame count changes if tracks are open as the track buffer 3778 // size depends on frame count and correct behavior would not be guaranteed 3779 // if frame count is changed after track creation 3780 if (!mTracks.isEmpty()) { 3781 status = INVALID_OPERATION; 3782 } else { 3783 reconfig = true; 3784 } 3785 } 3786 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3787#ifdef ADD_BATTERY_DATA 3788 // when changing the audio output device, call addBatteryData to notify 3789 // the change 3790 if (mOutDevice != value) { 3791 uint32_t params = 0; 3792 // check whether speaker is on 3793 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3794 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3795 } 3796 3797 audio_devices_t deviceWithoutSpeaker 3798 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3799 // check if any other device (except speaker) is on 3800 if (value & deviceWithoutSpeaker ) { 3801 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3802 } 3803 3804 if (params != 0) { 3805 addBatteryData(params); 3806 } 3807 } 3808#endif 3809 3810 // forward device change to effects that have requested to be 3811 // aware of attached audio device. 3812 if (value != AUDIO_DEVICE_NONE) { 3813 mOutDevice = value; 3814 for (size_t i = 0; i < mEffectChains.size(); i++) { 3815 mEffectChains[i]->setDevice_l(mOutDevice); 3816 } 3817 } 3818 } 3819 3820 if (status == NO_ERROR) { 3821 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3822 keyValuePair.string()); 3823 if (!mStandby && status == INVALID_OPERATION) { 3824 mOutput->stream->common.standby(&mOutput->stream->common); 3825 mStandby = true; 3826 mBytesWritten = 0; 3827 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3828 keyValuePair.string()); 3829 } 3830 if (status == NO_ERROR && reconfig) { 3831 readOutputParameters_l(); 3832 delete mAudioMixer; 3833 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3834 for (size_t i = 0; i < mTracks.size() ; i++) { 3835 int name = getTrackName_l(mTracks[i]->mChannelMask, 3836 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3837 if (name < 0) { 3838 break; 3839 } 3840 mTracks[i]->mName = name; 3841 } 3842 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3843 } 3844 } 3845 3846 if (!(previousCommand & FastMixerState::IDLE)) { 3847 ALOG_ASSERT(mFastMixer != 0); 3848 FastMixerStateQueue *sq = mFastMixer->sq(); 3849 FastMixerState *state = sq->begin(); 3850 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3851 state->mCommand = previousCommand; 3852 sq->end(); 3853 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3854 } 3855 3856 return reconfig; 3857} 3858 3859 3860void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3861{ 3862 const size_t SIZE = 256; 3863 char buffer[SIZE]; 3864 String8 result; 3865 3866 PlaybackThread::dumpInternals(fd, args); 3867 3868 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3869 3870 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3871 const FastMixerDumpState copy(mFastMixerDumpState); 3872 copy.dump(fd); 3873 3874#ifdef STATE_QUEUE_DUMP 3875 // Similar for state queue 3876 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3877 observerCopy.dump(fd); 3878 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3879 mutatorCopy.dump(fd); 3880#endif 3881 3882#ifdef TEE_SINK 3883 // Write the tee output to a .wav file 3884 dumpTee(fd, mTeeSource, mId); 3885#endif 3886 3887#ifdef AUDIO_WATCHDOG 3888 if (mAudioWatchdog != 0) { 3889 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3890 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3891 wdCopy.dump(fd); 3892 } 3893#endif 3894} 3895 3896uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3897{ 3898 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3899} 3900 3901uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3902{ 3903 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3904} 3905 3906void AudioFlinger::MixerThread::cacheParameters_l() 3907{ 3908 PlaybackThread::cacheParameters_l(); 3909 3910 // FIXME: Relaxed timing because of a certain device that can't meet latency 3911 // Should be reduced to 2x after the vendor fixes the driver issue 3912 // increase threshold again due to low power audio mode. The way this warning 3913 // threshold is calculated and its usefulness should be reconsidered anyway. 3914 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3915} 3916 3917// ---------------------------------------------------------------------------- 3918 3919AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3920 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3921 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3922 // mLeftVolFloat, mRightVolFloat 3923{ 3924} 3925 3926AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3927 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3928 ThreadBase::type_t type) 3929 : PlaybackThread(audioFlinger, output, id, device, type) 3930 // mLeftVolFloat, mRightVolFloat 3931{ 3932} 3933 3934AudioFlinger::DirectOutputThread::~DirectOutputThread() 3935{ 3936} 3937 3938void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3939{ 3940 audio_track_cblk_t* cblk = track->cblk(); 3941 float left, right; 3942 3943 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3944 left = right = 0; 3945 } else { 3946 float typeVolume = mStreamTypes[track->streamType()].volume; 3947 float v = mMasterVolume * typeVolume; 3948 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3949 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3950 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3951 if (left > GAIN_FLOAT_UNITY) { 3952 left = GAIN_FLOAT_UNITY; 3953 } 3954 left *= v; 3955 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3956 if (right > GAIN_FLOAT_UNITY) { 3957 right = GAIN_FLOAT_UNITY; 3958 } 3959 right *= v; 3960 } 3961 3962 if (lastTrack) { 3963 if (left != mLeftVolFloat || right != mRightVolFloat) { 3964 mLeftVolFloat = left; 3965 mRightVolFloat = right; 3966 3967 // Convert volumes from float to 8.24 3968 uint32_t vl = (uint32_t)(left * (1 << 24)); 3969 uint32_t vr = (uint32_t)(right * (1 << 24)); 3970 3971 // Delegate volume control to effect in track effect chain if needed 3972 // only one effect chain can be present on DirectOutputThread, so if 3973 // there is one, the track is connected to it 3974 if (!mEffectChains.isEmpty()) { 3975 mEffectChains[0]->setVolume_l(&vl, &vr); 3976 left = (float)vl / (1 << 24); 3977 right = (float)vr / (1 << 24); 3978 } 3979 if (mOutput->stream->set_volume) { 3980 mOutput->stream->set_volume(mOutput->stream, left, right); 3981 } 3982 } 3983 } 3984} 3985 3986 3987AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3988 Vector< sp<Track> > *tracksToRemove 3989) 3990{ 3991 size_t count = mActiveTracks.size(); 3992 mixer_state mixerStatus = MIXER_IDLE; 3993 3994 // find out which tracks need to be processed 3995 for (size_t i = 0; i < count; i++) { 3996 sp<Track> t = mActiveTracks[i].promote(); 3997 // The track died recently 3998 if (t == 0) { 3999 continue; 4000 } 4001 4002 Track* const track = t.get(); 4003 audio_track_cblk_t* cblk = track->cblk(); 4004 // Only consider last track started for volume and mixer state control. 4005 // In theory an older track could underrun and restart after the new one starts 4006 // but as we only care about the transition phase between two tracks on a 4007 // direct output, it is not a problem to ignore the underrun case. 4008 sp<Track> l = mLatestActiveTrack.promote(); 4009 bool last = l.get() == track; 4010 4011 // The first time a track is added we wait 4012 // for all its buffers to be filled before processing it 4013 uint32_t minFrames; 4014 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 4015 minFrames = mNormalFrameCount; 4016 } else { 4017 minFrames = 1; 4018 } 4019 4020 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4021 !track->isStopping_2() && !track->isStopped()) 4022 { 4023 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4024 4025 if (track->mFillingUpStatus == Track::FS_FILLED) { 4026 track->mFillingUpStatus = Track::FS_ACTIVE; 4027 // make sure processVolume_l() will apply new volume even if 0 4028 mLeftVolFloat = mRightVolFloat = -1.0; 4029 if (track->mState == TrackBase::RESUMING) { 4030 track->mState = TrackBase::ACTIVE; 4031 } 4032 } 4033 4034 // compute volume for this track 4035 processVolume_l(track, last); 4036 if (last) { 4037 // reset retry count 4038 track->mRetryCount = kMaxTrackRetriesDirect; 4039 mActiveTrack = t; 4040 mixerStatus = MIXER_TRACKS_READY; 4041 } 4042 } else { 4043 // clear effect chain input buffer if the last active track started underruns 4044 // to avoid sending previous audio buffer again to effects 4045 if (!mEffectChains.isEmpty() && last) { 4046 mEffectChains[0]->clearInputBuffer(); 4047 } 4048 if (track->isStopping_1()) { 4049 track->mState = TrackBase::STOPPING_2; 4050 } 4051 if ((track->sharedBuffer() != 0) || track->isStopped() || 4052 track->isStopping_2() || track->isPaused()) { 4053 // We have consumed all the buffers of this track. 4054 // Remove it from the list of active tracks. 4055 size_t audioHALFrames; 4056 if (audio_is_linear_pcm(mFormat)) { 4057 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4058 } else { 4059 audioHALFrames = 0; 4060 } 4061 4062 size_t framesWritten = mBytesWritten / mFrameSize; 4063 if (mStandby || !last || 4064 track->presentationComplete(framesWritten, audioHALFrames)) { 4065 if (track->isStopping_2()) { 4066 track->mState = TrackBase::STOPPED; 4067 } 4068 if (track->isStopped()) { 4069 if (track->mState == TrackBase::FLUSHED) { 4070 flushHw_l(); 4071 } 4072 track->reset(); 4073 } 4074 tracksToRemove->add(track); 4075 } 4076 } else { 4077 // No buffers for this track. Give it a few chances to 4078 // fill a buffer, then remove it from active list. 4079 // Only consider last track started for mixer state control 4080 if (--(track->mRetryCount) <= 0) { 4081 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4082 tracksToRemove->add(track); 4083 // indicate to client process that the track was disabled because of underrun; 4084 // it will then automatically call start() when data is available 4085 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4086 } else if (last) { 4087 mixerStatus = MIXER_TRACKS_ENABLED; 4088 } 4089 } 4090 } 4091 } 4092 4093 // remove all the tracks that need to be... 4094 removeTracks_l(*tracksToRemove); 4095 4096 return mixerStatus; 4097} 4098 4099void AudioFlinger::DirectOutputThread::threadLoop_mix() 4100{ 4101 size_t frameCount = mFrameCount; 4102 int8_t *curBuf = (int8_t *)mSinkBuffer; 4103 // output audio to hardware 4104 while (frameCount) { 4105 AudioBufferProvider::Buffer buffer; 4106 buffer.frameCount = frameCount; 4107 mActiveTrack->getNextBuffer(&buffer); 4108 if (buffer.raw == NULL) { 4109 memset(curBuf, 0, frameCount * mFrameSize); 4110 break; 4111 } 4112 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4113 frameCount -= buffer.frameCount; 4114 curBuf += buffer.frameCount * mFrameSize; 4115 mActiveTrack->releaseBuffer(&buffer); 4116 } 4117 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4118 sleepTime = 0; 4119 standbyTime = systemTime() + standbyDelay; 4120 mActiveTrack.clear(); 4121} 4122 4123void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4124{ 4125 if (sleepTime == 0) { 4126 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4127 sleepTime = activeSleepTime; 4128 } else { 4129 sleepTime = idleSleepTime; 4130 } 4131 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4132 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4133 sleepTime = 0; 4134 } 4135} 4136 4137// getTrackName_l() must be called with ThreadBase::mLock held 4138int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4139 audio_format_t format __unused, int sessionId __unused) 4140{ 4141 return 0; 4142} 4143 4144// deleteTrackName_l() must be called with ThreadBase::mLock held 4145void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4146{ 4147} 4148 4149// checkForNewParameter_l() must be called with ThreadBase::mLock held 4150bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4151 status_t& status) 4152{ 4153 bool reconfig = false; 4154 4155 status = NO_ERROR; 4156 4157 AudioParameter param = AudioParameter(keyValuePair); 4158 int value; 4159 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4160 // forward device change to effects that have requested to be 4161 // aware of attached audio device. 4162 if (value != AUDIO_DEVICE_NONE) { 4163 mOutDevice = value; 4164 for (size_t i = 0; i < mEffectChains.size(); i++) { 4165 mEffectChains[i]->setDevice_l(mOutDevice); 4166 } 4167 } 4168 } 4169 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4170 // do not accept frame count changes if tracks are open as the track buffer 4171 // size depends on frame count and correct behavior would not be garantied 4172 // if frame count is changed after track creation 4173 if (!mTracks.isEmpty()) { 4174 status = INVALID_OPERATION; 4175 } else { 4176 reconfig = true; 4177 } 4178 } 4179 if (status == NO_ERROR) { 4180 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4181 keyValuePair.string()); 4182 if (!mStandby && status == INVALID_OPERATION) { 4183 mOutput->stream->common.standby(&mOutput->stream->common); 4184 mStandby = true; 4185 mBytesWritten = 0; 4186 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4187 keyValuePair.string()); 4188 } 4189 if (status == NO_ERROR && reconfig) { 4190 readOutputParameters_l(); 4191 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4192 } 4193 } 4194 4195 return reconfig; 4196} 4197 4198uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4199{ 4200 uint32_t time; 4201 if (audio_is_linear_pcm(mFormat)) { 4202 time = PlaybackThread::activeSleepTimeUs(); 4203 } else { 4204 time = 10000; 4205 } 4206 return time; 4207} 4208 4209uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4210{ 4211 uint32_t time; 4212 if (audio_is_linear_pcm(mFormat)) { 4213 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4214 } else { 4215 time = 10000; 4216 } 4217 return time; 4218} 4219 4220uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4221{ 4222 uint32_t time; 4223 if (audio_is_linear_pcm(mFormat)) { 4224 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4225 } else { 4226 time = 10000; 4227 } 4228 return time; 4229} 4230 4231void AudioFlinger::DirectOutputThread::cacheParameters_l() 4232{ 4233 PlaybackThread::cacheParameters_l(); 4234 4235 // use shorter standby delay as on normal output to release 4236 // hardware resources as soon as possible 4237 if (audio_is_linear_pcm(mFormat)) { 4238 standbyDelay = microseconds(activeSleepTime*2); 4239 } else { 4240 standbyDelay = kOffloadStandbyDelayNs; 4241 } 4242} 4243 4244void AudioFlinger::DirectOutputThread::flushHw_l() 4245{ 4246 if (mOutput->stream->flush != NULL) 4247 mOutput->stream->flush(mOutput->stream); 4248} 4249 4250// ---------------------------------------------------------------------------- 4251 4252AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4253 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4254 : Thread(false /*canCallJava*/), 4255 mPlaybackThread(playbackThread), 4256 mWriteAckSequence(0), 4257 mDrainSequence(0) 4258{ 4259} 4260 4261AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4262{ 4263} 4264 4265void AudioFlinger::AsyncCallbackThread::onFirstRef() 4266{ 4267 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4268} 4269 4270bool AudioFlinger::AsyncCallbackThread::threadLoop() 4271{ 4272 while (!exitPending()) { 4273 uint32_t writeAckSequence; 4274 uint32_t drainSequence; 4275 4276 { 4277 Mutex::Autolock _l(mLock); 4278 while (!((mWriteAckSequence & 1) || 4279 (mDrainSequence & 1) || 4280 exitPending())) { 4281 mWaitWorkCV.wait(mLock); 4282 } 4283 4284 if (exitPending()) { 4285 break; 4286 } 4287 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4288 mWriteAckSequence, mDrainSequence); 4289 writeAckSequence = mWriteAckSequence; 4290 mWriteAckSequence &= ~1; 4291 drainSequence = mDrainSequence; 4292 mDrainSequence &= ~1; 4293 } 4294 { 4295 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4296 if (playbackThread != 0) { 4297 if (writeAckSequence & 1) { 4298 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4299 } 4300 if (drainSequence & 1) { 4301 playbackThread->resetDraining(drainSequence >> 1); 4302 } 4303 } 4304 } 4305 } 4306 return false; 4307} 4308 4309void AudioFlinger::AsyncCallbackThread::exit() 4310{ 4311 ALOGV("AsyncCallbackThread::exit"); 4312 Mutex::Autolock _l(mLock); 4313 requestExit(); 4314 mWaitWorkCV.broadcast(); 4315} 4316 4317void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4318{ 4319 Mutex::Autolock _l(mLock); 4320 // bit 0 is cleared 4321 mWriteAckSequence = sequence << 1; 4322} 4323 4324void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4325{ 4326 Mutex::Autolock _l(mLock); 4327 // ignore unexpected callbacks 4328 if (mWriteAckSequence & 2) { 4329 mWriteAckSequence |= 1; 4330 mWaitWorkCV.signal(); 4331 } 4332} 4333 4334void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4335{ 4336 Mutex::Autolock _l(mLock); 4337 // bit 0 is cleared 4338 mDrainSequence = sequence << 1; 4339} 4340 4341void AudioFlinger::AsyncCallbackThread::resetDraining() 4342{ 4343 Mutex::Autolock _l(mLock); 4344 // ignore unexpected callbacks 4345 if (mDrainSequence & 2) { 4346 mDrainSequence |= 1; 4347 mWaitWorkCV.signal(); 4348 } 4349} 4350 4351 4352// ---------------------------------------------------------------------------- 4353AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4354 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4355 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4356 mHwPaused(false), 4357 mFlushPending(false), 4358 mPausedBytesRemaining(0) 4359{ 4360 //FIXME: mStandby should be set to true by ThreadBase constructor 4361 mStandby = true; 4362} 4363 4364void AudioFlinger::OffloadThread::threadLoop_exit() 4365{ 4366 if (mFlushPending || mHwPaused) { 4367 // If a flush is pending or track was paused, just discard buffered data 4368 flushHw_l(); 4369 } else { 4370 mMixerStatus = MIXER_DRAIN_ALL; 4371 threadLoop_drain(); 4372 } 4373 if (mUseAsyncWrite) { 4374 ALOG_ASSERT(mCallbackThread != 0); 4375 mCallbackThread->exit(); 4376 } 4377 PlaybackThread::threadLoop_exit(); 4378} 4379 4380AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4381 Vector< sp<Track> > *tracksToRemove 4382) 4383{ 4384 size_t count = mActiveTracks.size(); 4385 4386 mixer_state mixerStatus = MIXER_IDLE; 4387 bool doHwPause = false; 4388 bool doHwResume = false; 4389 4390 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4391 4392 // find out which tracks need to be processed 4393 for (size_t i = 0; i < count; i++) { 4394 sp<Track> t = mActiveTracks[i].promote(); 4395 // The track died recently 4396 if (t == 0) { 4397 continue; 4398 } 4399 Track* const track = t.get(); 4400 audio_track_cblk_t* cblk = track->cblk(); 4401 // Only consider last track started for volume and mixer state control. 4402 // In theory an older track could underrun and restart after the new one starts 4403 // but as we only care about the transition phase between two tracks on a 4404 // direct output, it is not a problem to ignore the underrun case. 4405 sp<Track> l = mLatestActiveTrack.promote(); 4406 bool last = l.get() == track; 4407 4408 if (track->isInvalid()) { 4409 ALOGW("An invalidated track shouldn't be in active list"); 4410 tracksToRemove->add(track); 4411 continue; 4412 } 4413 4414 if (track->mState == TrackBase::IDLE) { 4415 ALOGW("An idle track shouldn't be in active list"); 4416 continue; 4417 } 4418 4419 if (track->isPausing()) { 4420 track->setPaused(); 4421 if (last) { 4422 if (!mHwPaused) { 4423 doHwPause = true; 4424 mHwPaused = true; 4425 } 4426 // If we were part way through writing the mixbuffer to 4427 // the HAL we must save this until we resume 4428 // BUG - this will be wrong if a different track is made active, 4429 // in that case we want to discard the pending data in the 4430 // mixbuffer and tell the client to present it again when the 4431 // track is resumed 4432 mPausedWriteLength = mCurrentWriteLength; 4433 mPausedBytesRemaining = mBytesRemaining; 4434 mBytesRemaining = 0; // stop writing 4435 } 4436 tracksToRemove->add(track); 4437 } else if (track->isFlushPending()) { 4438 track->flushAck(); 4439 if (last) { 4440 mFlushPending = true; 4441 } 4442 } else if (track->isResumePending()){ 4443 track->resumeAck(); 4444 if (last) { 4445 if (mPausedBytesRemaining) { 4446 // Need to continue write that was interrupted 4447 mCurrentWriteLength = mPausedWriteLength; 4448 mBytesRemaining = mPausedBytesRemaining; 4449 mPausedBytesRemaining = 0; 4450 } 4451 if (mHwPaused) { 4452 doHwResume = true; 4453 mHwPaused = false; 4454 // threadLoop_mix() will handle the case that we need to 4455 // resume an interrupted write 4456 } 4457 // enable write to audio HAL 4458 sleepTime = 0; 4459 4460 // Do not handle new data in this iteration even if track->framesReady() 4461 mixerStatus = MIXER_TRACKS_ENABLED; 4462 } 4463 } else if (track->framesReady() && track->isReady() && 4464 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4465 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4466 if (track->mFillingUpStatus == Track::FS_FILLED) { 4467 track->mFillingUpStatus = Track::FS_ACTIVE; 4468 // make sure processVolume_l() will apply new volume even if 0 4469 mLeftVolFloat = mRightVolFloat = -1.0; 4470 } 4471 4472 if (last) { 4473 sp<Track> previousTrack = mPreviousTrack.promote(); 4474 if (previousTrack != 0) { 4475 if (track != previousTrack.get()) { 4476 // Flush any data still being written from last track 4477 mBytesRemaining = 0; 4478 if (mPausedBytesRemaining) { 4479 // Last track was paused so we also need to flush saved 4480 // mixbuffer state and invalidate track so that it will 4481 // re-submit that unwritten data when it is next resumed 4482 mPausedBytesRemaining = 0; 4483 // Invalidate is a bit drastic - would be more efficient 4484 // to have a flag to tell client that some of the 4485 // previously written data was lost 4486 previousTrack->invalidate(); 4487 } 4488 // flush data already sent to the DSP if changing audio session as audio 4489 // comes from a different source. Also invalidate previous track to force a 4490 // seek when resuming. 4491 if (previousTrack->sessionId() != track->sessionId()) { 4492 previousTrack->invalidate(); 4493 } 4494 } 4495 } 4496 mPreviousTrack = track; 4497 // reset retry count 4498 track->mRetryCount = kMaxTrackRetriesOffload; 4499 mActiveTrack = t; 4500 mixerStatus = MIXER_TRACKS_READY; 4501 } 4502 } else { 4503 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4504 if (track->isStopping_1()) { 4505 // Hardware buffer can hold a large amount of audio so we must 4506 // wait for all current track's data to drain before we say 4507 // that the track is stopped. 4508 if (mBytesRemaining == 0) { 4509 // Only start draining when all data in mixbuffer 4510 // has been written 4511 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4512 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4513 // do not drain if no data was ever sent to HAL (mStandby == true) 4514 if (last && !mStandby) { 4515 // do not modify drain sequence if we are already draining. This happens 4516 // when resuming from pause after drain. 4517 if ((mDrainSequence & 1) == 0) { 4518 sleepTime = 0; 4519 standbyTime = systemTime() + standbyDelay; 4520 mixerStatus = MIXER_DRAIN_TRACK; 4521 mDrainSequence += 2; 4522 } 4523 if (mHwPaused) { 4524 // It is possible to move from PAUSED to STOPPING_1 without 4525 // a resume so we must ensure hardware is running 4526 doHwResume = true; 4527 mHwPaused = false; 4528 } 4529 } 4530 } 4531 } else if (track->isStopping_2()) { 4532 // Drain has completed or we are in standby, signal presentation complete 4533 if (!(mDrainSequence & 1) || !last || mStandby) { 4534 track->mState = TrackBase::STOPPED; 4535 size_t audioHALFrames = 4536 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4537 size_t framesWritten = 4538 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4539 track->presentationComplete(framesWritten, audioHALFrames); 4540 track->reset(); 4541 tracksToRemove->add(track); 4542 } 4543 } else { 4544 // No buffers for this track. Give it a few chances to 4545 // fill a buffer, then remove it from active list. 4546 if (--(track->mRetryCount) <= 0) { 4547 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4548 track->name()); 4549 tracksToRemove->add(track); 4550 // indicate to client process that the track was disabled because of underrun; 4551 // it will then automatically call start() when data is available 4552 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4553 } else if (last){ 4554 mixerStatus = MIXER_TRACKS_ENABLED; 4555 } 4556 } 4557 } 4558 // compute volume for this track 4559 processVolume_l(track, last); 4560 } 4561 4562 // make sure the pause/flush/resume sequence is executed in the right order. 4563 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4564 // before flush and then resume HW. This can happen in case of pause/flush/resume 4565 // if resume is received before pause is executed. 4566 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4567 mOutput->stream->pause(mOutput->stream); 4568 } 4569 if (mFlushPending) { 4570 flushHw_l(); 4571 mFlushPending = false; 4572 } 4573 if (!mStandby && doHwResume) { 4574 mOutput->stream->resume(mOutput->stream); 4575 } 4576 4577 // remove all the tracks that need to be... 4578 removeTracks_l(*tracksToRemove); 4579 4580 return mixerStatus; 4581} 4582 4583// must be called with thread mutex locked 4584bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4585{ 4586 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4587 mWriteAckSequence, mDrainSequence); 4588 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4589 return true; 4590 } 4591 return false; 4592} 4593 4594// must be called with thread mutex locked 4595bool AudioFlinger::OffloadThread::shouldStandby_l() 4596{ 4597 bool trackPaused = false; 4598 4599 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4600 // after a timeout and we will enter standby then. 4601 if (mTracks.size() > 0) { 4602 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4603 } 4604 4605 return !mStandby && !trackPaused; 4606} 4607 4608 4609bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4610{ 4611 Mutex::Autolock _l(mLock); 4612 return waitingAsyncCallback_l(); 4613} 4614 4615void AudioFlinger::OffloadThread::flushHw_l() 4616{ 4617 DirectOutputThread::flushHw_l(); 4618 // Flush anything still waiting in the mixbuffer 4619 mCurrentWriteLength = 0; 4620 mBytesRemaining = 0; 4621 mPausedWriteLength = 0; 4622 mPausedBytesRemaining = 0; 4623 mHwPaused = false; 4624 4625 if (mUseAsyncWrite) { 4626 // discard any pending drain or write ack by incrementing sequence 4627 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4628 mDrainSequence = (mDrainSequence + 2) & ~1; 4629 ALOG_ASSERT(mCallbackThread != 0); 4630 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4631 mCallbackThread->setDraining(mDrainSequence); 4632 } 4633} 4634 4635void AudioFlinger::OffloadThread::onAddNewTrack_l() 4636{ 4637 sp<Track> previousTrack = mPreviousTrack.promote(); 4638 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4639 4640 if (previousTrack != 0 && latestTrack != 0 && 4641 (previousTrack->sessionId() != latestTrack->sessionId())) { 4642 mFlushPending = true; 4643 } 4644 PlaybackThread::onAddNewTrack_l(); 4645} 4646 4647// ---------------------------------------------------------------------------- 4648 4649AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4650 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4651 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4652 DUPLICATING), 4653 mWaitTimeMs(UINT_MAX) 4654{ 4655 addOutputTrack(mainThread); 4656} 4657 4658AudioFlinger::DuplicatingThread::~DuplicatingThread() 4659{ 4660 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4661 mOutputTracks[i]->destroy(); 4662 } 4663} 4664 4665void AudioFlinger::DuplicatingThread::threadLoop_mix() 4666{ 4667 // mix buffers... 4668 if (outputsReady(outputTracks)) { 4669 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4670 } else { 4671 if (mMixerBufferValid) { 4672 memset(mMixerBuffer, 0, mMixerBufferSize); 4673 } else { 4674 memset(mSinkBuffer, 0, mSinkBufferSize); 4675 } 4676 } 4677 sleepTime = 0; 4678 writeFrames = mNormalFrameCount; 4679 mCurrentWriteLength = mSinkBufferSize; 4680 standbyTime = systemTime() + standbyDelay; 4681} 4682 4683void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4684{ 4685 if (sleepTime == 0) { 4686 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4687 sleepTime = activeSleepTime; 4688 } else { 4689 sleepTime = idleSleepTime; 4690 } 4691 } else if (mBytesWritten != 0) { 4692 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4693 writeFrames = mNormalFrameCount; 4694 memset(mSinkBuffer, 0, mSinkBufferSize); 4695 } else { 4696 // flush remaining overflow buffers in output tracks 4697 writeFrames = 0; 4698 } 4699 sleepTime = 0; 4700 } 4701} 4702 4703ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4704{ 4705 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4706 // for delivery downstream as needed. This in-place conversion is safe as 4707 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4708 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4709 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4710 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4711 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4712 } 4713 for (size_t i = 0; i < outputTracks.size(); i++) { 4714 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4715 } 4716 mStandby = false; 4717 return (ssize_t)mSinkBufferSize; 4718} 4719 4720void AudioFlinger::DuplicatingThread::threadLoop_standby() 4721{ 4722 // DuplicatingThread implements standby by stopping all tracks 4723 for (size_t i = 0; i < outputTracks.size(); i++) { 4724 outputTracks[i]->stop(); 4725 } 4726} 4727 4728void AudioFlinger::DuplicatingThread::saveOutputTracks() 4729{ 4730 outputTracks = mOutputTracks; 4731} 4732 4733void AudioFlinger::DuplicatingThread::clearOutputTracks() 4734{ 4735 outputTracks.clear(); 4736} 4737 4738void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4739{ 4740 Mutex::Autolock _l(mLock); 4741 // FIXME explain this formula 4742 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4743 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4744 // due to current usage case and restrictions on the AudioBufferProvider. 4745 // Actual buffer conversion is done in threadLoop_write(). 4746 // 4747 // TODO: This may change in the future, depending on multichannel 4748 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4749 OutputTrack *outputTrack = new OutputTrack(thread, 4750 this, 4751 mSampleRate, 4752 AUDIO_FORMAT_PCM_16_BIT, 4753 mChannelMask, 4754 frameCount, 4755 IPCThreadState::self()->getCallingUid()); 4756 if (outputTrack->cblk() != NULL) { 4757 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 4758 mOutputTracks.add(outputTrack); 4759 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4760 updateWaitTime_l(); 4761 } 4762} 4763 4764void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4765{ 4766 Mutex::Autolock _l(mLock); 4767 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4768 if (mOutputTracks[i]->thread() == thread) { 4769 mOutputTracks[i]->destroy(); 4770 mOutputTracks.removeAt(i); 4771 updateWaitTime_l(); 4772 return; 4773 } 4774 } 4775 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4776} 4777 4778// caller must hold mLock 4779void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4780{ 4781 mWaitTimeMs = UINT_MAX; 4782 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4783 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4784 if (strong != 0) { 4785 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4786 if (waitTimeMs < mWaitTimeMs) { 4787 mWaitTimeMs = waitTimeMs; 4788 } 4789 } 4790 } 4791} 4792 4793 4794bool AudioFlinger::DuplicatingThread::outputsReady( 4795 const SortedVector< sp<OutputTrack> > &outputTracks) 4796{ 4797 for (size_t i = 0; i < outputTracks.size(); i++) { 4798 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4799 if (thread == 0) { 4800 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4801 outputTracks[i].get()); 4802 return false; 4803 } 4804 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4805 // see note at standby() declaration 4806 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4807 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4808 thread.get()); 4809 return false; 4810 } 4811 } 4812 return true; 4813} 4814 4815uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4816{ 4817 return (mWaitTimeMs * 1000) / 2; 4818} 4819 4820void AudioFlinger::DuplicatingThread::cacheParameters_l() 4821{ 4822 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4823 updateWaitTime_l(); 4824 4825 MixerThread::cacheParameters_l(); 4826} 4827 4828// ---------------------------------------------------------------------------- 4829// Record 4830// ---------------------------------------------------------------------------- 4831 4832AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4833 AudioStreamIn *input, 4834 audio_io_handle_t id, 4835 audio_devices_t outDevice, 4836 audio_devices_t inDevice 4837#ifdef TEE_SINK 4838 , const sp<NBAIO_Sink>& teeSink 4839#endif 4840 ) : 4841 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4842 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4843 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4844 mRsmpInRear(0) 4845#ifdef TEE_SINK 4846 , mTeeSink(teeSink) 4847#endif 4848 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4849 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4850 // mFastCapture below 4851 , mFastCaptureFutex(0) 4852 // mInputSource 4853 // mPipeSink 4854 // mPipeSource 4855 , mPipeFramesP2(0) 4856 // mPipeMemory 4857 // mFastCaptureNBLogWriter 4858 , mFastTrackAvail(false) 4859{ 4860 snprintf(mName, kNameLength, "AudioIn_%X", id); 4861 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4862 4863 readInputParameters_l(); 4864 4865 // create an NBAIO source for the HAL input stream, and negotiate 4866 mInputSource = new AudioStreamInSource(input->stream); 4867 size_t numCounterOffers = 0; 4868 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4869 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4870 ALOG_ASSERT(index == 0); 4871 4872 // initialize fast capture depending on configuration 4873 bool initFastCapture; 4874 switch (kUseFastCapture) { 4875 case FastCapture_Never: 4876 initFastCapture = false; 4877 break; 4878 case FastCapture_Always: 4879 initFastCapture = true; 4880 break; 4881 case FastCapture_Static: 4882 uint32_t primaryOutputSampleRate; 4883 { 4884 AutoMutex _l(audioFlinger->mHardwareLock); 4885 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4886 } 4887 initFastCapture = 4888 // either capture sample rate is same as (a reasonable) primary output sample rate 4889 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4890 (mSampleRate == primaryOutputSampleRate)) || 4891 // or primary output sample rate is unknown, and capture sample rate is reasonable 4892 ((primaryOutputSampleRate == 0) && 4893 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4894 // and the buffer size is < 12 ms 4895 (mFrameCount * 1000) / mSampleRate < 12; 4896 break; 4897 // case FastCapture_Dynamic: 4898 } 4899 4900 if (initFastCapture) { 4901 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4902 NBAIO_Format format = mInputSource->format(); 4903 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4904 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4905 void *pipeBuffer; 4906 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4907 sp<IMemory> pipeMemory; 4908 if ((roHeap == 0) || 4909 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4910 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4911 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4912 goto failed; 4913 } 4914 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4915 memset(pipeBuffer, 0, pipeSize); 4916 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4917 const NBAIO_Format offers[1] = {format}; 4918 size_t numCounterOffers = 0; 4919 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4920 ALOG_ASSERT(index == 0); 4921 mPipeSink = pipe; 4922 PipeReader *pipeReader = new PipeReader(*pipe); 4923 numCounterOffers = 0; 4924 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4925 ALOG_ASSERT(index == 0); 4926 mPipeSource = pipeReader; 4927 mPipeFramesP2 = pipeFramesP2; 4928 mPipeMemory = pipeMemory; 4929 4930 // create fast capture 4931 mFastCapture = new FastCapture(); 4932 FastCaptureStateQueue *sq = mFastCapture->sq(); 4933#ifdef STATE_QUEUE_DUMP 4934 // FIXME 4935#endif 4936 FastCaptureState *state = sq->begin(); 4937 state->mCblk = NULL; 4938 state->mInputSource = mInputSource.get(); 4939 state->mInputSourceGen++; 4940 state->mPipeSink = pipe; 4941 state->mPipeSinkGen++; 4942 state->mFrameCount = mFrameCount; 4943 state->mCommand = FastCaptureState::COLD_IDLE; 4944 // already done in constructor initialization list 4945 //mFastCaptureFutex = 0; 4946 state->mColdFutexAddr = &mFastCaptureFutex; 4947 state->mColdGen++; 4948 state->mDumpState = &mFastCaptureDumpState; 4949#ifdef TEE_SINK 4950 // FIXME 4951#endif 4952 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4953 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4954 sq->end(); 4955 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4956 4957 // start the fast capture 4958 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4959 pid_t tid = mFastCapture->getTid(); 4960 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4961 if (err != 0) { 4962 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4963 kPriorityFastCapture, getpid_cached, tid, err); 4964 } 4965 4966#ifdef AUDIO_WATCHDOG 4967 // FIXME 4968#endif 4969 4970 mFastTrackAvail = true; 4971 } 4972failed: ; 4973 4974 // FIXME mNormalSource 4975} 4976 4977 4978AudioFlinger::RecordThread::~RecordThread() 4979{ 4980 if (mFastCapture != 0) { 4981 FastCaptureStateQueue *sq = mFastCapture->sq(); 4982 FastCaptureState *state = sq->begin(); 4983 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4984 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4985 if (old == -1) { 4986 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4987 } 4988 } 4989 state->mCommand = FastCaptureState::EXIT; 4990 sq->end(); 4991 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4992 mFastCapture->join(); 4993 mFastCapture.clear(); 4994 } 4995 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4996 mAudioFlinger->unregisterWriter(mNBLogWriter); 4997 delete[] mRsmpInBuffer; 4998} 4999 5000void AudioFlinger::RecordThread::onFirstRef() 5001{ 5002 run(mName, PRIORITY_URGENT_AUDIO); 5003} 5004 5005bool AudioFlinger::RecordThread::threadLoop() 5006{ 5007 nsecs_t lastWarning = 0; 5008 5009 inputStandBy(); 5010 5011reacquire_wakelock: 5012 sp<RecordTrack> activeTrack; 5013 int activeTracksGen; 5014 { 5015 Mutex::Autolock _l(mLock); 5016 size_t size = mActiveTracks.size(); 5017 activeTracksGen = mActiveTracksGen; 5018 if (size > 0) { 5019 // FIXME an arbitrary choice 5020 activeTrack = mActiveTracks[0]; 5021 acquireWakeLock_l(activeTrack->uid()); 5022 if (size > 1) { 5023 SortedVector<int> tmp; 5024 for (size_t i = 0; i < size; i++) { 5025 tmp.add(mActiveTracks[i]->uid()); 5026 } 5027 updateWakeLockUids_l(tmp); 5028 } 5029 } else { 5030 acquireWakeLock_l(-1); 5031 } 5032 } 5033 5034 // used to request a deferred sleep, to be executed later while mutex is unlocked 5035 uint32_t sleepUs = 0; 5036 5037 // loop while there is work to do 5038 for (;;) { 5039 Vector< sp<EffectChain> > effectChains; 5040 5041 // sleep with mutex unlocked 5042 if (sleepUs > 0) { 5043 usleep(sleepUs); 5044 sleepUs = 0; 5045 } 5046 5047 // activeTracks accumulates a copy of a subset of mActiveTracks 5048 Vector< sp<RecordTrack> > activeTracks; 5049 5050 // reference to the (first and only) active fast track 5051 sp<RecordTrack> fastTrack; 5052 5053 // reference to a fast track which is about to be removed 5054 sp<RecordTrack> fastTrackToRemove; 5055 5056 { // scope for mLock 5057 Mutex::Autolock _l(mLock); 5058 5059 processConfigEvents_l(); 5060 5061 // check exitPending here because checkForNewParameters_l() and 5062 // checkForNewParameters_l() can temporarily release mLock 5063 if (exitPending()) { 5064 break; 5065 } 5066 5067 // if no active track(s), then standby and release wakelock 5068 size_t size = mActiveTracks.size(); 5069 if (size == 0) { 5070 standbyIfNotAlreadyInStandby(); 5071 // exitPending() can't become true here 5072 releaseWakeLock_l(); 5073 ALOGV("RecordThread: loop stopping"); 5074 // go to sleep 5075 mWaitWorkCV.wait(mLock); 5076 ALOGV("RecordThread: loop starting"); 5077 goto reacquire_wakelock; 5078 } 5079 5080 if (mActiveTracksGen != activeTracksGen) { 5081 activeTracksGen = mActiveTracksGen; 5082 SortedVector<int> tmp; 5083 for (size_t i = 0; i < size; i++) { 5084 tmp.add(mActiveTracks[i]->uid()); 5085 } 5086 updateWakeLockUids_l(tmp); 5087 } 5088 5089 bool doBroadcast = false; 5090 for (size_t i = 0; i < size; ) { 5091 5092 activeTrack = mActiveTracks[i]; 5093 if (activeTrack->isTerminated()) { 5094 if (activeTrack->isFastTrack()) { 5095 ALOG_ASSERT(fastTrackToRemove == 0); 5096 fastTrackToRemove = activeTrack; 5097 } 5098 removeTrack_l(activeTrack); 5099 mActiveTracks.remove(activeTrack); 5100 mActiveTracksGen++; 5101 size--; 5102 continue; 5103 } 5104 5105 TrackBase::track_state activeTrackState = activeTrack->mState; 5106 switch (activeTrackState) { 5107 5108 case TrackBase::PAUSING: 5109 mActiveTracks.remove(activeTrack); 5110 mActiveTracksGen++; 5111 doBroadcast = true; 5112 size--; 5113 continue; 5114 5115 case TrackBase::STARTING_1: 5116 sleepUs = 10000; 5117 i++; 5118 continue; 5119 5120 case TrackBase::STARTING_2: 5121 doBroadcast = true; 5122 mStandby = false; 5123 activeTrack->mState = TrackBase::ACTIVE; 5124 break; 5125 5126 case TrackBase::ACTIVE: 5127 break; 5128 5129 case TrackBase::IDLE: 5130 i++; 5131 continue; 5132 5133 default: 5134 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5135 } 5136 5137 activeTracks.add(activeTrack); 5138 i++; 5139 5140 if (activeTrack->isFastTrack()) { 5141 ALOG_ASSERT(!mFastTrackAvail); 5142 ALOG_ASSERT(fastTrack == 0); 5143 fastTrack = activeTrack; 5144 } 5145 } 5146 if (doBroadcast) { 5147 mStartStopCond.broadcast(); 5148 } 5149 5150 // sleep if there are no active tracks to process 5151 if (activeTracks.size() == 0) { 5152 if (sleepUs == 0) { 5153 sleepUs = kRecordThreadSleepUs; 5154 } 5155 continue; 5156 } 5157 sleepUs = 0; 5158 5159 lockEffectChains_l(effectChains); 5160 } 5161 5162 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5163 5164 size_t size = effectChains.size(); 5165 for (size_t i = 0; i < size; i++) { 5166 // thread mutex is not locked, but effect chain is locked 5167 effectChains[i]->process_l(); 5168 } 5169 5170 // Push a new fast capture state if fast capture is not already running, or cblk change 5171 if (mFastCapture != 0) { 5172 FastCaptureStateQueue *sq = mFastCapture->sq(); 5173 FastCaptureState *state = sq->begin(); 5174 bool didModify = false; 5175 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5176 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5177 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5178 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5179 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5180 if (old == -1) { 5181 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5182 } 5183 } 5184 state->mCommand = FastCaptureState::READ_WRITE; 5185#if 0 // FIXME 5186 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5187 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5188#endif 5189 didModify = true; 5190 } 5191 audio_track_cblk_t *cblkOld = state->mCblk; 5192 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5193 if (cblkNew != cblkOld) { 5194 state->mCblk = cblkNew; 5195 // block until acked if removing a fast track 5196 if (cblkOld != NULL) { 5197 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5198 } 5199 didModify = true; 5200 } 5201 sq->end(didModify); 5202 if (didModify) { 5203 sq->push(block); 5204#if 0 5205 if (kUseFastCapture == FastCapture_Dynamic) { 5206 mNormalSource = mPipeSource; 5207 } 5208#endif 5209 } 5210 } 5211 5212 // now run the fast track destructor with thread mutex unlocked 5213 fastTrackToRemove.clear(); 5214 5215 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5216 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5217 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5218 // If destination is non-contiguous, first read past the nominal end of buffer, then 5219 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5220 5221 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5222 ssize_t framesRead; 5223 5224 // If an NBAIO source is present, use it to read the normal capture's data 5225 if (mPipeSource != 0) { 5226 size_t framesToRead = mBufferSize / mFrameSize; 5227 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5228 framesToRead, AudioBufferProvider::kInvalidPTS); 5229 if (framesRead == 0) { 5230 // since pipe is non-blocking, simulate blocking input 5231 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5232 } 5233 // otherwise use the HAL / AudioStreamIn directly 5234 } else { 5235 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5236 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5237 if (bytesRead < 0) { 5238 framesRead = bytesRead; 5239 } else { 5240 framesRead = bytesRead / mFrameSize; 5241 } 5242 } 5243 5244 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5245 ALOGE("read failed: framesRead=%d", framesRead); 5246 // Force input into standby so that it tries to recover at next read attempt 5247 inputStandBy(); 5248 sleepUs = kRecordThreadSleepUs; 5249 } 5250 if (framesRead <= 0) { 5251 goto unlock; 5252 } 5253 ALOG_ASSERT(framesRead > 0); 5254 5255 if (mTeeSink != 0) { 5256 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5257 } 5258 // If destination is non-contiguous, we now correct for reading past end of buffer. 5259 { 5260 size_t part1 = mRsmpInFramesP2 - rear; 5261 if ((size_t) framesRead > part1) { 5262 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5263 (framesRead - part1) * mFrameSize); 5264 } 5265 } 5266 rear = mRsmpInRear += framesRead; 5267 5268 size = activeTracks.size(); 5269 // loop over each active track 5270 for (size_t i = 0; i < size; i++) { 5271 activeTrack = activeTracks[i]; 5272 5273 // skip fast tracks, as those are handled directly by FastCapture 5274 if (activeTrack->isFastTrack()) { 5275 continue; 5276 } 5277 5278 enum { 5279 OVERRUN_UNKNOWN, 5280 OVERRUN_TRUE, 5281 OVERRUN_FALSE 5282 } overrun = OVERRUN_UNKNOWN; 5283 5284 // loop over getNextBuffer to handle circular sink 5285 for (;;) { 5286 5287 activeTrack->mSink.frameCount = ~0; 5288 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5289 size_t framesOut = activeTrack->mSink.frameCount; 5290 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5291 5292 int32_t front = activeTrack->mRsmpInFront; 5293 ssize_t filled = rear - front; 5294 size_t framesIn; 5295 5296 if (filled < 0) { 5297 // should not happen, but treat like a massive overrun and re-sync 5298 framesIn = 0; 5299 activeTrack->mRsmpInFront = rear; 5300 overrun = OVERRUN_TRUE; 5301 } else if ((size_t) filled <= mRsmpInFrames) { 5302 framesIn = (size_t) filled; 5303 } else { 5304 // client is not keeping up with server, but give it latest data 5305 framesIn = mRsmpInFrames; 5306 activeTrack->mRsmpInFront = front = rear - framesIn; 5307 overrun = OVERRUN_TRUE; 5308 } 5309 5310 if (framesOut == 0 || framesIn == 0) { 5311 break; 5312 } 5313 5314 if (activeTrack->mResampler == NULL) { 5315 // no resampling 5316 if (framesIn > framesOut) { 5317 framesIn = framesOut; 5318 } else { 5319 framesOut = framesIn; 5320 } 5321 int8_t *dst = activeTrack->mSink.i8; 5322 while (framesIn > 0) { 5323 front &= mRsmpInFramesP2 - 1; 5324 size_t part1 = mRsmpInFramesP2 - front; 5325 if (part1 > framesIn) { 5326 part1 = framesIn; 5327 } 5328 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5329 if (mChannelCount == activeTrack->mChannelCount) { 5330 memcpy(dst, src, part1 * mFrameSize); 5331 } else if (mChannelCount == 1) { 5332 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5333 part1); 5334 } else { 5335 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5336 part1); 5337 } 5338 dst += part1 * activeTrack->mFrameSize; 5339 front += part1; 5340 framesIn -= part1; 5341 } 5342 activeTrack->mRsmpInFront += framesOut; 5343 5344 } else { 5345 // resampling 5346 // FIXME framesInNeeded should really be part of resampler API, and should 5347 // depend on the SRC ratio 5348 // to keep mRsmpInBuffer full so resampler always has sufficient input 5349 size_t framesInNeeded; 5350 // FIXME only re-calculate when it changes, and optimize for common ratios 5351 // Do not precompute in/out because floating point is not associative 5352 // e.g. a*b/c != a*(b/c). 5353 const double in(mSampleRate); 5354 const double out(activeTrack->mSampleRate); 5355 framesInNeeded = ceil(framesOut * in / out) + 1; 5356 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5357 framesInNeeded, framesOut, in / out); 5358 // Although we theoretically have framesIn in circular buffer, some of those are 5359 // unreleased frames, and thus must be discounted for purpose of budgeting. 5360 size_t unreleased = activeTrack->mRsmpInUnrel; 5361 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5362 if (framesIn < framesInNeeded) { 5363 ALOGV("not enough to resample: have %u frames in but need %u in to " 5364 "produce %u out given in/out ratio of %.4g", 5365 framesIn, framesInNeeded, framesOut, in / out); 5366 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5367 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5368 if (newFramesOut == 0) { 5369 break; 5370 } 5371 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5372 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5373 framesInNeeded, newFramesOut, out / in); 5374 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5375 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5376 "given in/out ratio of %.4g", 5377 framesIn, framesInNeeded, newFramesOut, in / out); 5378 framesOut = newFramesOut; 5379 } else { 5380 ALOGV("success 1: have %u in and need %u in to produce %u out " 5381 "given in/out ratio of %.4g", 5382 framesIn, framesInNeeded, framesOut, in / out); 5383 } 5384 5385 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5386 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5387 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5388 delete[] activeTrack->mRsmpOutBuffer; 5389 // resampler always outputs stereo 5390 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5391 activeTrack->mRsmpOutFrameCount = framesOut; 5392 } 5393 5394 // resampler accumulates, but we only have one source track 5395 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5396 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5397 // FIXME how about having activeTrack implement this interface itself? 5398 activeTrack->mResamplerBufferProvider 5399 /*this*/ /* AudioBufferProvider* */); 5400 // ditherAndClamp() works as long as all buffers returned by 5401 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5402 if (activeTrack->mChannelCount == 1) { 5403 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5404 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5405 framesOut); 5406 // the resampler always outputs stereo samples: 5407 // do post stereo to mono conversion 5408 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5409 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5410 } else { 5411 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5412 activeTrack->mRsmpOutBuffer, framesOut); 5413 } 5414 // now done with mRsmpOutBuffer 5415 5416 } 5417 5418 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5419 overrun = OVERRUN_FALSE; 5420 } 5421 5422 if (activeTrack->mFramesToDrop == 0) { 5423 if (framesOut > 0) { 5424 activeTrack->mSink.frameCount = framesOut; 5425 activeTrack->releaseBuffer(&activeTrack->mSink); 5426 } 5427 } else { 5428 // FIXME could do a partial drop of framesOut 5429 if (activeTrack->mFramesToDrop > 0) { 5430 activeTrack->mFramesToDrop -= framesOut; 5431 if (activeTrack->mFramesToDrop <= 0) { 5432 activeTrack->clearSyncStartEvent(); 5433 } 5434 } else { 5435 activeTrack->mFramesToDrop += framesOut; 5436 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5437 activeTrack->mSyncStartEvent->isCancelled()) { 5438 ALOGW("Synced record %s, session %d, trigger session %d", 5439 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5440 activeTrack->sessionId(), 5441 (activeTrack->mSyncStartEvent != 0) ? 5442 activeTrack->mSyncStartEvent->triggerSession() : 0); 5443 activeTrack->clearSyncStartEvent(); 5444 } 5445 } 5446 } 5447 5448 if (framesOut == 0) { 5449 break; 5450 } 5451 } 5452 5453 switch (overrun) { 5454 case OVERRUN_TRUE: 5455 // client isn't retrieving buffers fast enough 5456 if (!activeTrack->setOverflow()) { 5457 nsecs_t now = systemTime(); 5458 // FIXME should lastWarning per track? 5459 if ((now - lastWarning) > kWarningThrottleNs) { 5460 ALOGW("RecordThread: buffer overflow"); 5461 lastWarning = now; 5462 } 5463 } 5464 break; 5465 case OVERRUN_FALSE: 5466 activeTrack->clearOverflow(); 5467 break; 5468 case OVERRUN_UNKNOWN: 5469 break; 5470 } 5471 5472 } 5473 5474unlock: 5475 // enable changes in effect chain 5476 unlockEffectChains(effectChains); 5477 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5478 } 5479 5480 standbyIfNotAlreadyInStandby(); 5481 5482 { 5483 Mutex::Autolock _l(mLock); 5484 for (size_t i = 0; i < mTracks.size(); i++) { 5485 sp<RecordTrack> track = mTracks[i]; 5486 track->invalidate(); 5487 } 5488 mActiveTracks.clear(); 5489 mActiveTracksGen++; 5490 mStartStopCond.broadcast(); 5491 } 5492 5493 releaseWakeLock(); 5494 5495 ALOGV("RecordThread %p exiting", this); 5496 return false; 5497} 5498 5499void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5500{ 5501 if (!mStandby) { 5502 inputStandBy(); 5503 mStandby = true; 5504 } 5505} 5506 5507void AudioFlinger::RecordThread::inputStandBy() 5508{ 5509 // Idle the fast capture if it's currently running 5510 if (mFastCapture != 0) { 5511 FastCaptureStateQueue *sq = mFastCapture->sq(); 5512 FastCaptureState *state = sq->begin(); 5513 if (!(state->mCommand & FastCaptureState::IDLE)) { 5514 state->mCommand = FastCaptureState::COLD_IDLE; 5515 state->mColdFutexAddr = &mFastCaptureFutex; 5516 state->mColdGen++; 5517 mFastCaptureFutex = 0; 5518 sq->end(); 5519 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5520 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5521#if 0 5522 if (kUseFastCapture == FastCapture_Dynamic) { 5523 // FIXME 5524 } 5525#endif 5526#ifdef AUDIO_WATCHDOG 5527 // FIXME 5528#endif 5529 } else { 5530 sq->end(false /*didModify*/); 5531 } 5532 } 5533 mInput->stream->common.standby(&mInput->stream->common); 5534} 5535 5536// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5537sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5538 const sp<AudioFlinger::Client>& client, 5539 uint32_t sampleRate, 5540 audio_format_t format, 5541 audio_channel_mask_t channelMask, 5542 size_t *pFrameCount, 5543 int sessionId, 5544 size_t *notificationFrames, 5545 int uid, 5546 IAudioFlinger::track_flags_t *flags, 5547 pid_t tid, 5548 status_t *status) 5549{ 5550 size_t frameCount = *pFrameCount; 5551 sp<RecordTrack> track; 5552 status_t lStatus; 5553 5554 // client expresses a preference for FAST, but we get the final say 5555 if (*flags & IAudioFlinger::TRACK_FAST) { 5556 if ( 5557 // use case: callback handler 5558 (tid != -1) && 5559 // frame count is not specified, or is exactly the pipe depth 5560 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5561 // PCM data 5562 audio_is_linear_pcm(format) && 5563 // native format 5564 (format == mFormat) && 5565 // native channel mask 5566 (channelMask == mChannelMask) && 5567 // native hardware sample rate 5568 (sampleRate == mSampleRate) && 5569 // record thread has an associated fast capture 5570 hasFastCapture() && 5571 // there are sufficient fast track slots available 5572 mFastTrackAvail 5573 ) { 5574 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5575 frameCount, mFrameCount); 5576 } else { 5577 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5578 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5579 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5580 frameCount, mFrameCount, mPipeFramesP2, 5581 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5582 hasFastCapture(), tid, mFastTrackAvail); 5583 *flags &= ~IAudioFlinger::TRACK_FAST; 5584 } 5585 } 5586 5587 // compute track buffer size in frames, and suggest the notification frame count 5588 if (*flags & IAudioFlinger::TRACK_FAST) { 5589 // fast track: frame count is exactly the pipe depth 5590 frameCount = mPipeFramesP2; 5591 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5592 *notificationFrames = mFrameCount; 5593 } else { 5594 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5595 // or 20 ms if there is a fast capture 5596 // TODO This could be a roundupRatio inline, and const 5597 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5598 * sampleRate + mSampleRate - 1) / mSampleRate; 5599 // minimum number of notification periods is at least kMinNotifications, 5600 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5601 static const size_t kMinNotifications = 3; 5602 static const uint32_t kMinMs = 30; 5603 // TODO This could be a roundupRatio inline 5604 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5605 // TODO This could be a roundupRatio inline 5606 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5607 maxNotificationFrames; 5608 const size_t minFrameCount = maxNotificationFrames * 5609 max(kMinNotifications, minNotificationsByMs); 5610 frameCount = max(frameCount, minFrameCount); 5611 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5612 *notificationFrames = maxNotificationFrames; 5613 } 5614 } 5615 *pFrameCount = frameCount; 5616 5617 lStatus = initCheck(); 5618 if (lStatus != NO_ERROR) { 5619 ALOGE("createRecordTrack_l() audio driver not initialized"); 5620 goto Exit; 5621 } 5622 5623 { // scope for mLock 5624 Mutex::Autolock _l(mLock); 5625 5626 track = new RecordTrack(this, client, sampleRate, 5627 format, channelMask, frameCount, NULL, sessionId, uid, 5628 *flags, TrackBase::TYPE_DEFAULT); 5629 5630 lStatus = track->initCheck(); 5631 if (lStatus != NO_ERROR) { 5632 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5633 // track must be cleared from the caller as the caller has the AF lock 5634 goto Exit; 5635 } 5636 mTracks.add(track); 5637 5638 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5639 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5640 mAudioFlinger->btNrecIsOff(); 5641 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5642 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5643 5644 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5645 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5646 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5647 // so ask activity manager to do this on our behalf 5648 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5649 } 5650 } 5651 5652 lStatus = NO_ERROR; 5653 5654Exit: 5655 *status = lStatus; 5656 return track; 5657} 5658 5659status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5660 AudioSystem::sync_event_t event, 5661 int triggerSession) 5662{ 5663 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5664 sp<ThreadBase> strongMe = this; 5665 status_t status = NO_ERROR; 5666 5667 if (event == AudioSystem::SYNC_EVENT_NONE) { 5668 recordTrack->clearSyncStartEvent(); 5669 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5670 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5671 triggerSession, 5672 recordTrack->sessionId(), 5673 syncStartEventCallback, 5674 recordTrack); 5675 // Sync event can be cancelled by the trigger session if the track is not in a 5676 // compatible state in which case we start record immediately 5677 if (recordTrack->mSyncStartEvent->isCancelled()) { 5678 recordTrack->clearSyncStartEvent(); 5679 } else { 5680 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5681 recordTrack->mFramesToDrop = - 5682 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5683 } 5684 } 5685 5686 { 5687 // This section is a rendezvous between binder thread executing start() and RecordThread 5688 AutoMutex lock(mLock); 5689 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5690 if (recordTrack->mState == TrackBase::PAUSING) { 5691 ALOGV("active record track PAUSING -> ACTIVE"); 5692 recordTrack->mState = TrackBase::ACTIVE; 5693 } else { 5694 ALOGV("active record track state %d", recordTrack->mState); 5695 } 5696 return status; 5697 } 5698 5699 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5700 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5701 // or using a separate command thread 5702 recordTrack->mState = TrackBase::STARTING_1; 5703 mActiveTracks.add(recordTrack); 5704 mActiveTracksGen++; 5705 status_t status = NO_ERROR; 5706 if (recordTrack->isExternalTrack()) { 5707 mLock.unlock(); 5708 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5709 mLock.lock(); 5710 // FIXME should verify that recordTrack is still in mActiveTracks 5711 if (status != NO_ERROR) { 5712 mActiveTracks.remove(recordTrack); 5713 mActiveTracksGen++; 5714 recordTrack->clearSyncStartEvent(); 5715 ALOGV("RecordThread::start error %d", status); 5716 return status; 5717 } 5718 } 5719 // Catch up with current buffer indices if thread is already running. 5720 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5721 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5722 // see previously buffered data before it called start(), but with greater risk of overrun. 5723 5724 recordTrack->mRsmpInFront = mRsmpInRear; 5725 recordTrack->mRsmpInUnrel = 0; 5726 // FIXME why reset? 5727 if (recordTrack->mResampler != NULL) { 5728 recordTrack->mResampler->reset(); 5729 } 5730 recordTrack->mState = TrackBase::STARTING_2; 5731 // signal thread to start 5732 mWaitWorkCV.broadcast(); 5733 if (mActiveTracks.indexOf(recordTrack) < 0) { 5734 ALOGV("Record failed to start"); 5735 status = BAD_VALUE; 5736 goto startError; 5737 } 5738 return status; 5739 } 5740 5741startError: 5742 if (recordTrack->isExternalTrack()) { 5743 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5744 } 5745 recordTrack->clearSyncStartEvent(); 5746 // FIXME I wonder why we do not reset the state here? 5747 return status; 5748} 5749 5750void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5751{ 5752 sp<SyncEvent> strongEvent = event.promote(); 5753 5754 if (strongEvent != 0) { 5755 sp<RefBase> ptr = strongEvent->cookie().promote(); 5756 if (ptr != 0) { 5757 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5758 recordTrack->handleSyncStartEvent(strongEvent); 5759 } 5760 } 5761} 5762 5763bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5764 ALOGV("RecordThread::stop"); 5765 AutoMutex _l(mLock); 5766 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5767 return false; 5768 } 5769 // note that threadLoop may still be processing the track at this point [without lock] 5770 recordTrack->mState = TrackBase::PAUSING; 5771 // do not wait for mStartStopCond if exiting 5772 if (exitPending()) { 5773 return true; 5774 } 5775 // FIXME incorrect usage of wait: no explicit predicate or loop 5776 mStartStopCond.wait(mLock); 5777 // if we have been restarted, recordTrack is in mActiveTracks here 5778 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5779 ALOGV("Record stopped OK"); 5780 return true; 5781 } 5782 return false; 5783} 5784 5785bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5786{ 5787 return false; 5788} 5789 5790status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5791{ 5792#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5793 if (!isValidSyncEvent(event)) { 5794 return BAD_VALUE; 5795 } 5796 5797 int eventSession = event->triggerSession(); 5798 status_t ret = NAME_NOT_FOUND; 5799 5800 Mutex::Autolock _l(mLock); 5801 5802 for (size_t i = 0; i < mTracks.size(); i++) { 5803 sp<RecordTrack> track = mTracks[i]; 5804 if (eventSession == track->sessionId()) { 5805 (void) track->setSyncEvent(event); 5806 ret = NO_ERROR; 5807 } 5808 } 5809 return ret; 5810#else 5811 return BAD_VALUE; 5812#endif 5813} 5814 5815// destroyTrack_l() must be called with ThreadBase::mLock held 5816void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5817{ 5818 track->terminate(); 5819 track->mState = TrackBase::STOPPED; 5820 // active tracks are removed by threadLoop() 5821 if (mActiveTracks.indexOf(track) < 0) { 5822 removeTrack_l(track); 5823 } 5824} 5825 5826void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5827{ 5828 mTracks.remove(track); 5829 // need anything related to effects here? 5830 if (track->isFastTrack()) { 5831 ALOG_ASSERT(!mFastTrackAvail); 5832 mFastTrackAvail = true; 5833 } 5834} 5835 5836void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5837{ 5838 dumpInternals(fd, args); 5839 dumpTracks(fd, args); 5840 dumpEffectChains(fd, args); 5841} 5842 5843void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5844{ 5845 dprintf(fd, "\nInput thread %p:\n", this); 5846 5847 if (mActiveTracks.size() > 0) { 5848 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5849 } else { 5850 dprintf(fd, " No active record clients\n"); 5851 } 5852 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5853 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5854 5855 dumpBase(fd, args); 5856} 5857 5858void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5859{ 5860 const size_t SIZE = 256; 5861 char buffer[SIZE]; 5862 String8 result; 5863 5864 size_t numtracks = mTracks.size(); 5865 size_t numactive = mActiveTracks.size(); 5866 size_t numactiveseen = 0; 5867 dprintf(fd, " %d Tracks", numtracks); 5868 if (numtracks) { 5869 dprintf(fd, " of which %d are active\n", numactive); 5870 RecordTrack::appendDumpHeader(result); 5871 for (size_t i = 0; i < numtracks ; ++i) { 5872 sp<RecordTrack> track = mTracks[i]; 5873 if (track != 0) { 5874 bool active = mActiveTracks.indexOf(track) >= 0; 5875 if (active) { 5876 numactiveseen++; 5877 } 5878 track->dump(buffer, SIZE, active); 5879 result.append(buffer); 5880 } 5881 } 5882 } else { 5883 dprintf(fd, "\n"); 5884 } 5885 5886 if (numactiveseen != numactive) { 5887 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5888 " not in the track list\n"); 5889 result.append(buffer); 5890 RecordTrack::appendDumpHeader(result); 5891 for (size_t i = 0; i < numactive; ++i) { 5892 sp<RecordTrack> track = mActiveTracks[i]; 5893 if (mTracks.indexOf(track) < 0) { 5894 track->dump(buffer, SIZE, true); 5895 result.append(buffer); 5896 } 5897 } 5898 5899 } 5900 write(fd, result.string(), result.size()); 5901} 5902 5903// AudioBufferProvider interface 5904status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5905 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5906{ 5907 RecordTrack *activeTrack = mRecordTrack; 5908 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5909 if (threadBase == 0) { 5910 buffer->frameCount = 0; 5911 buffer->raw = NULL; 5912 return NOT_ENOUGH_DATA; 5913 } 5914 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5915 int32_t rear = recordThread->mRsmpInRear; 5916 int32_t front = activeTrack->mRsmpInFront; 5917 ssize_t filled = rear - front; 5918 // FIXME should not be P2 (don't want to increase latency) 5919 // FIXME if client not keeping up, discard 5920 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5921 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5922 front &= recordThread->mRsmpInFramesP2 - 1; 5923 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5924 if (part1 > (size_t) filled) { 5925 part1 = filled; 5926 } 5927 size_t ask = buffer->frameCount; 5928 ALOG_ASSERT(ask > 0); 5929 if (part1 > ask) { 5930 part1 = ask; 5931 } 5932 if (part1 == 0) { 5933 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5934 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5935 buffer->raw = NULL; 5936 buffer->frameCount = 0; 5937 activeTrack->mRsmpInUnrel = 0; 5938 return NOT_ENOUGH_DATA; 5939 } 5940 5941 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5942 buffer->frameCount = part1; 5943 activeTrack->mRsmpInUnrel = part1; 5944 return NO_ERROR; 5945} 5946 5947// AudioBufferProvider interface 5948void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5949 AudioBufferProvider::Buffer* buffer) 5950{ 5951 RecordTrack *activeTrack = mRecordTrack; 5952 size_t stepCount = buffer->frameCount; 5953 if (stepCount == 0) { 5954 return; 5955 } 5956 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5957 activeTrack->mRsmpInUnrel -= stepCount; 5958 activeTrack->mRsmpInFront += stepCount; 5959 buffer->raw = NULL; 5960 buffer->frameCount = 0; 5961} 5962 5963bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5964 status_t& status) 5965{ 5966 bool reconfig = false; 5967 5968 status = NO_ERROR; 5969 5970 audio_format_t reqFormat = mFormat; 5971 uint32_t samplingRate = mSampleRate; 5972 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5973 5974 AudioParameter param = AudioParameter(keyValuePair); 5975 int value; 5976 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5977 // channel count change can be requested. Do we mandate the first client defines the 5978 // HAL sampling rate and channel count or do we allow changes on the fly? 5979 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5980 samplingRate = value; 5981 reconfig = true; 5982 } 5983 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5984 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5985 status = BAD_VALUE; 5986 } else { 5987 reqFormat = (audio_format_t) value; 5988 reconfig = true; 5989 } 5990 } 5991 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5992 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5993 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5994 status = BAD_VALUE; 5995 } else { 5996 channelMask = mask; 5997 reconfig = true; 5998 } 5999 } 6000 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6001 // do not accept frame count changes if tracks are open as the track buffer 6002 // size depends on frame count and correct behavior would not be guaranteed 6003 // if frame count is changed after track creation 6004 if (mActiveTracks.size() > 0) { 6005 status = INVALID_OPERATION; 6006 } else { 6007 reconfig = true; 6008 } 6009 } 6010 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6011 // forward device change to effects that have requested to be 6012 // aware of attached audio device. 6013 for (size_t i = 0; i < mEffectChains.size(); i++) { 6014 mEffectChains[i]->setDevice_l(value); 6015 } 6016 6017 // store input device and output device but do not forward output device to audio HAL. 6018 // Note that status is ignored by the caller for output device 6019 // (see AudioFlinger::setParameters() 6020 if (audio_is_output_devices(value)) { 6021 mOutDevice = value; 6022 status = BAD_VALUE; 6023 } else { 6024 mInDevice = value; 6025 // disable AEC and NS if the device is a BT SCO headset supporting those 6026 // pre processings 6027 if (mTracks.size() > 0) { 6028 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6029 mAudioFlinger->btNrecIsOff(); 6030 for (size_t i = 0; i < mTracks.size(); i++) { 6031 sp<RecordTrack> track = mTracks[i]; 6032 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6033 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6034 } 6035 } 6036 } 6037 } 6038 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6039 mAudioSource != (audio_source_t)value) { 6040 // forward device change to effects that have requested to be 6041 // aware of attached audio device. 6042 for (size_t i = 0; i < mEffectChains.size(); i++) { 6043 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6044 } 6045 mAudioSource = (audio_source_t)value; 6046 } 6047 6048 if (status == NO_ERROR) { 6049 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6050 keyValuePair.string()); 6051 if (status == INVALID_OPERATION) { 6052 inputStandBy(); 6053 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6054 keyValuePair.string()); 6055 } 6056 if (reconfig) { 6057 if (status == BAD_VALUE && 6058 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6059 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6060 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6061 <= (2 * samplingRate)) && 6062 audio_channel_count_from_in_mask( 6063 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6064 (channelMask == AUDIO_CHANNEL_IN_MONO || 6065 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6066 status = NO_ERROR; 6067 } 6068 if (status == NO_ERROR) { 6069 readInputParameters_l(); 6070 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6071 } 6072 } 6073 } 6074 6075 return reconfig; 6076} 6077 6078String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6079{ 6080 Mutex::Autolock _l(mLock); 6081 if (initCheck() != NO_ERROR) { 6082 return String8(); 6083 } 6084 6085 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6086 const String8 out_s8(s); 6087 free(s); 6088 return out_s8; 6089} 6090 6091void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6092 AudioSystem::OutputDescriptor desc; 6093 const void *param2 = NULL; 6094 6095 switch (event) { 6096 case AudioSystem::INPUT_OPENED: 6097 case AudioSystem::INPUT_CONFIG_CHANGED: 6098 desc.channelMask = mChannelMask; 6099 desc.samplingRate = mSampleRate; 6100 desc.format = mFormat; 6101 desc.frameCount = mFrameCount; 6102 desc.latency = 0; 6103 param2 = &desc; 6104 break; 6105 6106 case AudioSystem::INPUT_CLOSED: 6107 default: 6108 break; 6109 } 6110 mAudioFlinger->audioConfigChanged(event, mId, param2); 6111} 6112 6113void AudioFlinger::RecordThread::readInputParameters_l() 6114{ 6115 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6116 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6117 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6118 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6119 mFormat = mHALFormat; 6120 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6121 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6122 } 6123 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6124 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6125 mFrameCount = mBufferSize / mFrameSize; 6126 // This is the formula for calculating the temporary buffer size. 6127 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6128 // 1 full output buffer, regardless of the alignment of the available input. 6129 // The value is somewhat arbitrary, and could probably be even larger. 6130 // A larger value should allow more old data to be read after a track calls start(), 6131 // without increasing latency. 6132 mRsmpInFrames = mFrameCount * 7; 6133 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6134 delete[] mRsmpInBuffer; 6135 6136 // TODO optimize audio capture buffer sizes ... 6137 // Here we calculate the size of the sliding buffer used as a source 6138 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6139 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6140 // be better to have it derived from the pipe depth in the long term. 6141 // The current value is higher than necessary. However it should not add to latency. 6142 6143 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6144 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6145 6146 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6147 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6148} 6149 6150uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6151{ 6152 Mutex::Autolock _l(mLock); 6153 if (initCheck() != NO_ERROR) { 6154 return 0; 6155 } 6156 6157 return mInput->stream->get_input_frames_lost(mInput->stream); 6158} 6159 6160uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6161{ 6162 Mutex::Autolock _l(mLock); 6163 uint32_t result = 0; 6164 if (getEffectChain_l(sessionId) != 0) { 6165 result = EFFECT_SESSION; 6166 } 6167 6168 for (size_t i = 0; i < mTracks.size(); ++i) { 6169 if (sessionId == mTracks[i]->sessionId()) { 6170 result |= TRACK_SESSION; 6171 break; 6172 } 6173 } 6174 6175 return result; 6176} 6177 6178KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6179{ 6180 KeyedVector<int, bool> ids; 6181 Mutex::Autolock _l(mLock); 6182 for (size_t j = 0; j < mTracks.size(); ++j) { 6183 sp<RecordThread::RecordTrack> track = mTracks[j]; 6184 int sessionId = track->sessionId(); 6185 if (ids.indexOfKey(sessionId) < 0) { 6186 ids.add(sessionId, true); 6187 } 6188 } 6189 return ids; 6190} 6191 6192AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6193{ 6194 Mutex::Autolock _l(mLock); 6195 AudioStreamIn *input = mInput; 6196 mInput = NULL; 6197 return input; 6198} 6199 6200// this method must always be called either with ThreadBase mLock held or inside the thread loop 6201audio_stream_t* AudioFlinger::RecordThread::stream() const 6202{ 6203 if (mInput == NULL) { 6204 return NULL; 6205 } 6206 return &mInput->stream->common; 6207} 6208 6209status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6210{ 6211 // only one chain per input thread 6212 if (mEffectChains.size() != 0) { 6213 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6214 return INVALID_OPERATION; 6215 } 6216 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6217 chain->setThread(this); 6218 chain->setInBuffer(NULL); 6219 chain->setOutBuffer(NULL); 6220 6221 checkSuspendOnAddEffectChain_l(chain); 6222 6223 // make sure enabled pre processing effects state is communicated to the HAL as we 6224 // just moved them to a new input stream. 6225 chain->syncHalEffectsState(); 6226 6227 mEffectChains.add(chain); 6228 6229 return NO_ERROR; 6230} 6231 6232size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6233{ 6234 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6235 ALOGW_IF(mEffectChains.size() != 1, 6236 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6237 chain.get(), mEffectChains.size(), this); 6238 if (mEffectChains.size() == 1) { 6239 mEffectChains.removeAt(0); 6240 } 6241 return 0; 6242} 6243 6244status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6245 audio_patch_handle_t *handle) 6246{ 6247 status_t status = NO_ERROR; 6248 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6249 // store new device and send to effects 6250 mInDevice = patch->sources[0].ext.device.type; 6251 for (size_t i = 0; i < mEffectChains.size(); i++) { 6252 mEffectChains[i]->setDevice_l(mInDevice); 6253 } 6254 6255 // disable AEC and NS if the device is a BT SCO headset supporting those 6256 // pre processings 6257 if (mTracks.size() > 0) { 6258 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6259 mAudioFlinger->btNrecIsOff(); 6260 for (size_t i = 0; i < mTracks.size(); i++) { 6261 sp<RecordTrack> track = mTracks[i]; 6262 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6263 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6264 } 6265 } 6266 6267 // store new source and send to effects 6268 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6269 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6270 for (size_t i = 0; i < mEffectChains.size(); i++) { 6271 mEffectChains[i]->setAudioSource_l(mAudioSource); 6272 } 6273 } 6274 6275 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6276 status = hwDevice->create_audio_patch(hwDevice, 6277 patch->num_sources, 6278 patch->sources, 6279 patch->num_sinks, 6280 patch->sinks, 6281 handle); 6282 } else { 6283 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6284 } 6285 return status; 6286} 6287 6288status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6289{ 6290 status_t status = NO_ERROR; 6291 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6292 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6293 status = hwDevice->release_audio_patch(hwDevice, handle); 6294 } else { 6295 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6296 } 6297 return status; 6298} 6299 6300void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6301{ 6302 Mutex::Autolock _l(mLock); 6303 mTracks.add(record); 6304} 6305 6306void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6307{ 6308 Mutex::Autolock _l(mLock); 6309 destroyTrack_l(record); 6310} 6311 6312void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6313{ 6314 ThreadBase::getAudioPortConfig(config); 6315 config->role = AUDIO_PORT_ROLE_SINK; 6316 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6317 config->ext.mix.usecase.source = mAudioSource; 6318} 6319 6320}; // namespace android 6321