Threads.cpp revision 121143d5242a790d0bd01fe1b9cec5d28a1ba1d7
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 = IInterface::asBinder(mPowerManager); 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 // Default implementation has nothing to do 2184} 2185 2186/* 2187The derived values that are cached: 2188 - mSinkBufferSize from frame count * frame size 2189 - activeSleepTime from activeSleepTimeUs() 2190 - idleSleepTime from idleSleepTimeUs() 2191 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2192 - maxPeriod from frame count and sample rate (MIXER only) 2193 2194The parameters that affect these derived values are: 2195 - frame count 2196 - frame size 2197 - sample rate 2198 - device type: A2DP or not 2199 - device latency 2200 - format: PCM or not 2201 - active sleep time 2202 - idle sleep time 2203*/ 2204 2205void AudioFlinger::PlaybackThread::cacheParameters_l() 2206{ 2207 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2208 activeSleepTime = activeSleepTimeUs(); 2209 idleSleepTime = idleSleepTimeUs(); 2210} 2211 2212void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2213{ 2214 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2215 this, streamType, mTracks.size()); 2216 Mutex::Autolock _l(mLock); 2217 2218 size_t size = mTracks.size(); 2219 for (size_t i = 0; i < size; i++) { 2220 sp<Track> t = mTracks[i]; 2221 if (t->streamType() == streamType) { 2222 t->invalidate(); 2223 } 2224 } 2225} 2226 2227status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2228{ 2229 int session = chain->sessionId(); 2230 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2231 ? mEffectBuffer : mSinkBuffer); 2232 bool ownsBuffer = false; 2233 2234 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2235 if (session > 0) { 2236 // Only one effect chain can be present in direct output thread and it uses 2237 // the sink buffer as input 2238 if (mType != DIRECT) { 2239 size_t numSamples = mNormalFrameCount * mChannelCount; 2240 buffer = new int16_t[numSamples]; 2241 memset(buffer, 0, numSamples * sizeof(int16_t)); 2242 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2243 ownsBuffer = true; 2244 } 2245 2246 // Attach all tracks with same session ID to this chain. 2247 for (size_t i = 0; i < mTracks.size(); ++i) { 2248 sp<Track> track = mTracks[i]; 2249 if (session == track->sessionId()) { 2250 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2251 buffer); 2252 track->setMainBuffer(buffer); 2253 chain->incTrackCnt(); 2254 } 2255 } 2256 2257 // indicate all active tracks in the chain 2258 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2259 sp<Track> track = mActiveTracks[i].promote(); 2260 if (track == 0) { 2261 continue; 2262 } 2263 if (session == track->sessionId()) { 2264 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2265 chain->incActiveTrackCnt(); 2266 } 2267 } 2268 } 2269 chain->setThread(this); 2270 chain->setInBuffer(buffer, ownsBuffer); 2271 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2272 ? mEffectBuffer : mSinkBuffer)); 2273 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2274 // chains list in order to be processed last as it contains output stage effects 2275 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2276 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2277 // after track specific effects and before output stage 2278 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2279 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2280 // Effect chain for other sessions are inserted at beginning of effect 2281 // chains list to be processed before output mix effects. Relative order between other 2282 // sessions is not important 2283 size_t size = mEffectChains.size(); 2284 size_t i = 0; 2285 for (i = 0; i < size; i++) { 2286 if (mEffectChains[i]->sessionId() < session) { 2287 break; 2288 } 2289 } 2290 mEffectChains.insertAt(chain, i); 2291 checkSuspendOnAddEffectChain_l(chain); 2292 2293 return NO_ERROR; 2294} 2295 2296size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2297{ 2298 int session = chain->sessionId(); 2299 2300 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2301 2302 for (size_t i = 0; i < mEffectChains.size(); i++) { 2303 if (chain == mEffectChains[i]) { 2304 mEffectChains.removeAt(i); 2305 // detach all active tracks from the chain 2306 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2307 sp<Track> track = mActiveTracks[i].promote(); 2308 if (track == 0) { 2309 continue; 2310 } 2311 if (session == track->sessionId()) { 2312 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2313 chain.get(), session); 2314 chain->decActiveTrackCnt(); 2315 } 2316 } 2317 2318 // detach all tracks with same session ID from this chain 2319 for (size_t i = 0; i < mTracks.size(); ++i) { 2320 sp<Track> track = mTracks[i]; 2321 if (session == track->sessionId()) { 2322 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2323 chain->decTrackCnt(); 2324 } 2325 } 2326 break; 2327 } 2328 } 2329 return mEffectChains.size(); 2330} 2331 2332status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2333 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2334{ 2335 Mutex::Autolock _l(mLock); 2336 return attachAuxEffect_l(track, EffectId); 2337} 2338 2339status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2340 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2341{ 2342 status_t status = NO_ERROR; 2343 2344 if (EffectId == 0) { 2345 track->setAuxBuffer(0, NULL); 2346 } else { 2347 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2348 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2349 if (effect != 0) { 2350 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2351 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2352 } else { 2353 status = INVALID_OPERATION; 2354 } 2355 } else { 2356 status = BAD_VALUE; 2357 } 2358 } 2359 return status; 2360} 2361 2362void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2363{ 2364 for (size_t i = 0; i < mTracks.size(); ++i) { 2365 sp<Track> track = mTracks[i]; 2366 if (track->auxEffectId() == effectId) { 2367 attachAuxEffect_l(track, 0); 2368 } 2369 } 2370} 2371 2372bool AudioFlinger::PlaybackThread::threadLoop() 2373{ 2374 Vector< sp<Track> > tracksToRemove; 2375 2376 standbyTime = systemTime(); 2377 2378 // MIXER 2379 nsecs_t lastWarning = 0; 2380 2381 // DUPLICATING 2382 // FIXME could this be made local to while loop? 2383 writeFrames = 0; 2384 2385 int lastGeneration = 0; 2386 2387 cacheParameters_l(); 2388 sleepTime = idleSleepTime; 2389 2390 if (mType == MIXER) { 2391 sleepTimeShift = 0; 2392 } 2393 2394 CpuStats cpuStats; 2395 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2396 2397 acquireWakeLock(); 2398 2399 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2400 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2401 // and then that string will be logged at the next convenient opportunity. 2402 const char *logString = NULL; 2403 2404 checkSilentMode_l(); 2405 2406 while (!exitPending()) 2407 { 2408 cpuStats.sample(myName); 2409 2410 Vector< sp<EffectChain> > effectChains; 2411 2412 { // scope for mLock 2413 2414 Mutex::Autolock _l(mLock); 2415 2416 processConfigEvents_l(); 2417 2418 if (logString != NULL) { 2419 mNBLogWriter->logTimestamp(); 2420 mNBLogWriter->log(logString); 2421 logString = NULL; 2422 } 2423 2424 // Gather the framesReleased counters for all active tracks, 2425 // and latch them atomically with the timestamp. 2426 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2427 mLatchD.mFramesReleased.clear(); 2428 size_t size = mActiveTracks.size(); 2429 for (size_t i = 0; i < size; i++) { 2430 sp<Track> t = mActiveTracks[i].promote(); 2431 if (t != 0) { 2432 mLatchD.mFramesReleased.add(t.get(), 2433 t->mAudioTrackServerProxy->framesReleased()); 2434 } 2435 } 2436 if (mLatchDValid) { 2437 mLatchQ = mLatchD; 2438 mLatchDValid = false; 2439 mLatchQValid = true; 2440 } 2441 2442 saveOutputTracks(); 2443 if (mSignalPending) { 2444 // A signal was raised while we were unlocked 2445 mSignalPending = false; 2446 } else if (waitingAsyncCallback_l()) { 2447 if (exitPending()) { 2448 break; 2449 } 2450 releaseWakeLock_l(); 2451 mWakeLockUids.clear(); 2452 mActiveTracksGeneration++; 2453 ALOGV("wait async completion"); 2454 mWaitWorkCV.wait(mLock); 2455 ALOGV("async completion/wake"); 2456 acquireWakeLock_l(); 2457 standbyTime = systemTime() + standbyDelay; 2458 sleepTime = 0; 2459 2460 continue; 2461 } 2462 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2463 isSuspended()) { 2464 // put audio hardware into standby after short delay 2465 if (shouldStandby_l()) { 2466 2467 threadLoop_standby(); 2468 2469 mStandby = true; 2470 } 2471 2472 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2473 // we're about to wait, flush the binder command buffer 2474 IPCThreadState::self()->flushCommands(); 2475 2476 clearOutputTracks(); 2477 2478 if (exitPending()) { 2479 break; 2480 } 2481 2482 releaseWakeLock_l(); 2483 mWakeLockUids.clear(); 2484 mActiveTracksGeneration++; 2485 // wait until we have something to do... 2486 ALOGV("%s going to sleep", myName.string()); 2487 mWaitWorkCV.wait(mLock); 2488 ALOGV("%s waking up", myName.string()); 2489 acquireWakeLock_l(); 2490 2491 mMixerStatus = MIXER_IDLE; 2492 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2493 mBytesWritten = 0; 2494 mBytesRemaining = 0; 2495 checkSilentMode_l(); 2496 2497 standbyTime = systemTime() + standbyDelay; 2498 sleepTime = idleSleepTime; 2499 if (mType == MIXER) { 2500 sleepTimeShift = 0; 2501 } 2502 2503 continue; 2504 } 2505 } 2506 // mMixerStatusIgnoringFastTracks is also updated internally 2507 mMixerStatus = prepareTracks_l(&tracksToRemove); 2508 2509 // compare with previously applied list 2510 if (lastGeneration != mActiveTracksGeneration) { 2511 // update wakelock 2512 updateWakeLockUids_l(mWakeLockUids); 2513 lastGeneration = mActiveTracksGeneration; 2514 } 2515 2516 // prevent any changes in effect chain list and in each effect chain 2517 // during mixing and effect process as the audio buffers could be deleted 2518 // or modified if an effect is created or deleted 2519 lockEffectChains_l(effectChains); 2520 } // mLock scope ends 2521 2522 if (mBytesRemaining == 0) { 2523 mCurrentWriteLength = 0; 2524 if (mMixerStatus == MIXER_TRACKS_READY) { 2525 // threadLoop_mix() sets mCurrentWriteLength 2526 threadLoop_mix(); 2527 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2528 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2529 // threadLoop_sleepTime sets sleepTime to 0 if data 2530 // must be written to HAL 2531 threadLoop_sleepTime(); 2532 if (sleepTime == 0) { 2533 mCurrentWriteLength = mSinkBufferSize; 2534 } 2535 } 2536 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2537 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2538 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2539 // or mSinkBuffer (if there are no effects). 2540 // 2541 // This is done pre-effects computation; if effects change to 2542 // support higher precision, this needs to move. 2543 // 2544 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2545 // TODO use sleepTime == 0 as an additional condition. 2546 if (mMixerBufferValid) { 2547 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2548 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2549 2550 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2551 mNormalFrameCount * mChannelCount); 2552 } 2553 2554 mBytesRemaining = mCurrentWriteLength; 2555 if (isSuspended()) { 2556 sleepTime = suspendSleepTimeUs(); 2557 // simulate write to HAL when suspended 2558 mBytesWritten += mSinkBufferSize; 2559 mBytesRemaining = 0; 2560 } 2561 2562 // only process effects if we're going to write 2563 if (sleepTime == 0 && mType != OFFLOAD) { 2564 for (size_t i = 0; i < effectChains.size(); i ++) { 2565 effectChains[i]->process_l(); 2566 } 2567 } 2568 } 2569 // Process effect chains for offloaded thread even if no audio 2570 // was read from audio track: process only updates effect state 2571 // and thus does have to be synchronized with audio writes but may have 2572 // to be called while waiting for async write callback 2573 if (mType == OFFLOAD) { 2574 for (size_t i = 0; i < effectChains.size(); i ++) { 2575 effectChains[i]->process_l(); 2576 } 2577 } 2578 2579 // Only if the Effects buffer is enabled and there is data in the 2580 // Effects buffer (buffer valid), we need to 2581 // copy into the sink buffer. 2582 // TODO use sleepTime == 0 as an additional condition. 2583 if (mEffectBufferValid) { 2584 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2585 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2586 mNormalFrameCount * mChannelCount); 2587 } 2588 2589 // enable changes in effect chain 2590 unlockEffectChains(effectChains); 2591 2592 if (!waitingAsyncCallback()) { 2593 // sleepTime == 0 means we must write to audio hardware 2594 if (sleepTime == 0) { 2595 if (mBytesRemaining) { 2596 ssize_t ret = threadLoop_write(); 2597 if (ret < 0) { 2598 mBytesRemaining = 0; 2599 } else { 2600 mBytesWritten += ret; 2601 mBytesRemaining -= ret; 2602 } 2603 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2604 (mMixerStatus == MIXER_DRAIN_ALL)) { 2605 threadLoop_drain(); 2606 } 2607 if (mType == MIXER) { 2608 // write blocked detection 2609 nsecs_t now = systemTime(); 2610 nsecs_t delta = now - mLastWriteTime; 2611 if (!mStandby && delta > maxPeriod) { 2612 mNumDelayedWrites++; 2613 if ((now - lastWarning) > kWarningThrottleNs) { 2614 ATRACE_NAME("underrun"); 2615 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2616 ns2ms(delta), mNumDelayedWrites, this); 2617 lastWarning = now; 2618 } 2619 } 2620 } 2621 2622 } else { 2623 usleep(sleepTime); 2624 } 2625 } 2626 2627 // Finally let go of removed track(s), without the lock held 2628 // since we can't guarantee the destructors won't acquire that 2629 // same lock. This will also mutate and push a new fast mixer state. 2630 threadLoop_removeTracks(tracksToRemove); 2631 tracksToRemove.clear(); 2632 2633 // FIXME I don't understand the need for this here; 2634 // it was in the original code but maybe the 2635 // assignment in saveOutputTracks() makes this unnecessary? 2636 clearOutputTracks(); 2637 2638 // Effect chains will be actually deleted here if they were removed from 2639 // mEffectChains list during mixing or effects processing 2640 effectChains.clear(); 2641 2642 // FIXME Note that the above .clear() is no longer necessary since effectChains 2643 // is now local to this block, but will keep it for now (at least until merge done). 2644 } 2645 2646 threadLoop_exit(); 2647 2648 if (!mStandby) { 2649 threadLoop_standby(); 2650 mStandby = true; 2651 } 2652 2653 releaseWakeLock(); 2654 mWakeLockUids.clear(); 2655 mActiveTracksGeneration++; 2656 2657 ALOGV("Thread %p type %d exiting", this, mType); 2658 return false; 2659} 2660 2661// removeTracks_l() must be called with ThreadBase::mLock held 2662void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2663{ 2664 size_t count = tracksToRemove.size(); 2665 if (count > 0) { 2666 for (size_t i=0 ; i<count ; i++) { 2667 const sp<Track>& track = tracksToRemove.itemAt(i); 2668 mActiveTracks.remove(track); 2669 mWakeLockUids.remove(track->uid()); 2670 mActiveTracksGeneration++; 2671 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2672 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2673 if (chain != 0) { 2674 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2675 track->sessionId()); 2676 chain->decActiveTrackCnt(); 2677 } 2678 if (track->isTerminated()) { 2679 removeTrack_l(track); 2680 } 2681 } 2682 } 2683 2684} 2685 2686status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2687{ 2688 if (mNormalSink != 0) { 2689 return mNormalSink->getTimestamp(timestamp); 2690 } 2691 if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) { 2692 uint64_t position64; 2693 int ret = mOutput->stream->get_presentation_position( 2694 mOutput->stream, &position64, ×tamp.mTime); 2695 if (ret == 0) { 2696 timestamp.mPosition = (uint32_t)position64; 2697 return NO_ERROR; 2698 } 2699 } 2700 return INVALID_OPERATION; 2701} 2702 2703status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2704 audio_patch_handle_t *handle) 2705{ 2706 status_t status = NO_ERROR; 2707 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2708 // store new device and send to effects 2709 audio_devices_t type = AUDIO_DEVICE_NONE; 2710 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2711 type |= patch->sinks[i].ext.device.type; 2712 } 2713 mOutDevice = type; 2714 for (size_t i = 0; i < mEffectChains.size(); i++) { 2715 mEffectChains[i]->setDevice_l(mOutDevice); 2716 } 2717 2718 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2719 status = hwDevice->create_audio_patch(hwDevice, 2720 patch->num_sources, 2721 patch->sources, 2722 patch->num_sinks, 2723 patch->sinks, 2724 handle); 2725 } else { 2726 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2727 } 2728 return status; 2729} 2730 2731status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2732{ 2733 status_t status = NO_ERROR; 2734 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2735 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2736 status = hwDevice->release_audio_patch(hwDevice, handle); 2737 } else { 2738 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2739 } 2740 return status; 2741} 2742 2743void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2744{ 2745 Mutex::Autolock _l(mLock); 2746 mTracks.add(track); 2747} 2748 2749void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2750{ 2751 Mutex::Autolock _l(mLock); 2752 destroyTrack_l(track); 2753} 2754 2755void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2756{ 2757 ThreadBase::getAudioPortConfig(config); 2758 config->role = AUDIO_PORT_ROLE_SOURCE; 2759 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2760 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2761} 2762 2763// ---------------------------------------------------------------------------- 2764 2765AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2766 audio_io_handle_t id, audio_devices_t device, type_t type) 2767 : PlaybackThread(audioFlinger, output, id, device, type), 2768 // mAudioMixer below 2769 // mFastMixer below 2770 mFastMixerFutex(0) 2771 // mOutputSink below 2772 // mPipeSink below 2773 // mNormalSink below 2774{ 2775 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2776 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2777 "mFrameCount=%d, mNormalFrameCount=%d", 2778 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2779 mNormalFrameCount); 2780 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2781 2782 // create an NBAIO sink for the HAL output stream, and negotiate 2783 mOutputSink = new AudioStreamOutSink(output->stream); 2784 size_t numCounterOffers = 0; 2785 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2786 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2787 ALOG_ASSERT(index == 0); 2788 2789 // initialize fast mixer depending on configuration 2790 bool initFastMixer; 2791 switch (kUseFastMixer) { 2792 case FastMixer_Never: 2793 initFastMixer = false; 2794 break; 2795 case FastMixer_Always: 2796 initFastMixer = true; 2797 break; 2798 case FastMixer_Static: 2799 case FastMixer_Dynamic: 2800 initFastMixer = mFrameCount < mNormalFrameCount; 2801 break; 2802 } 2803 if (initFastMixer) { 2804 audio_format_t fastMixerFormat; 2805 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2806 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2807 } else { 2808 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2809 } 2810 if (mFormat != fastMixerFormat) { 2811 // change our Sink format to accept our intermediate precision 2812 mFormat = fastMixerFormat; 2813 free(mSinkBuffer); 2814 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2815 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2816 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2817 } 2818 2819 // create a MonoPipe to connect our submix to FastMixer 2820 NBAIO_Format format = mOutputSink->format(); 2821 NBAIO_Format origformat = format; 2822 // adjust format to match that of the Fast Mixer 2823 format.mFormat = fastMixerFormat; 2824 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2825 2826 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2827 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2828 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2829 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2830 const NBAIO_Format offers[1] = {format}; 2831 size_t numCounterOffers = 0; 2832 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2833 ALOG_ASSERT(index == 0); 2834 monoPipe->setAvgFrames((mScreenState & 1) ? 2835 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2836 mPipeSink = monoPipe; 2837 2838#ifdef TEE_SINK 2839 if (mTeeSinkOutputEnabled) { 2840 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2841 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 2842 const NBAIO_Format offers2[1] = {origformat}; 2843 numCounterOffers = 0; 2844 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 2845 ALOG_ASSERT(index == 0); 2846 mTeeSink = teeSink; 2847 PipeReader *teeSource = new PipeReader(*teeSink); 2848 numCounterOffers = 0; 2849 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 2850 ALOG_ASSERT(index == 0); 2851 mTeeSource = teeSource; 2852 } 2853#endif 2854 2855 // create fast mixer and configure it initially with just one fast track for our submix 2856 mFastMixer = new FastMixer(); 2857 FastMixerStateQueue *sq = mFastMixer->sq(); 2858#ifdef STATE_QUEUE_DUMP 2859 sq->setObserverDump(&mStateQueueObserverDump); 2860 sq->setMutatorDump(&mStateQueueMutatorDump); 2861#endif 2862 FastMixerState *state = sq->begin(); 2863 FastTrack *fastTrack = &state->mFastTracks[0]; 2864 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2865 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2866 fastTrack->mVolumeProvider = NULL; 2867 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2868 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2869 fastTrack->mGeneration++; 2870 state->mFastTracksGen++; 2871 state->mTrackMask = 1; 2872 // fast mixer will use the HAL output sink 2873 state->mOutputSink = mOutputSink.get(); 2874 state->mOutputSinkGen++; 2875 state->mFrameCount = mFrameCount; 2876 state->mCommand = FastMixerState::COLD_IDLE; 2877 // already done in constructor initialization list 2878 //mFastMixerFutex = 0; 2879 state->mColdFutexAddr = &mFastMixerFutex; 2880 state->mColdGen++; 2881 state->mDumpState = &mFastMixerDumpState; 2882#ifdef TEE_SINK 2883 state->mTeeSink = mTeeSink.get(); 2884#endif 2885 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2886 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2887 sq->end(); 2888 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2889 2890 // start the fast mixer 2891 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2892 pid_t tid = mFastMixer->getTid(); 2893 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2894 if (err != 0) { 2895 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2896 kPriorityFastMixer, getpid_cached, tid, err); 2897 } 2898 2899#ifdef AUDIO_WATCHDOG 2900 // create and start the watchdog 2901 mAudioWatchdog = new AudioWatchdog(); 2902 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2903 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2904 tid = mAudioWatchdog->getTid(); 2905 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2906 if (err != 0) { 2907 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2908 kPriorityFastMixer, getpid_cached, tid, err); 2909 } 2910#endif 2911 2912 } 2913 2914 switch (kUseFastMixer) { 2915 case FastMixer_Never: 2916 case FastMixer_Dynamic: 2917 mNormalSink = mOutputSink; 2918 break; 2919 case FastMixer_Always: 2920 mNormalSink = mPipeSink; 2921 break; 2922 case FastMixer_Static: 2923 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2924 break; 2925 } 2926} 2927 2928AudioFlinger::MixerThread::~MixerThread() 2929{ 2930 if (mFastMixer != 0) { 2931 FastMixerStateQueue *sq = mFastMixer->sq(); 2932 FastMixerState *state = sq->begin(); 2933 if (state->mCommand == FastMixerState::COLD_IDLE) { 2934 int32_t old = android_atomic_inc(&mFastMixerFutex); 2935 if (old == -1) { 2936 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2937 } 2938 } 2939 state->mCommand = FastMixerState::EXIT; 2940 sq->end(); 2941 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2942 mFastMixer->join(); 2943 // Though the fast mixer thread has exited, it's state queue is still valid. 2944 // We'll use that extract the final state which contains one remaining fast track 2945 // corresponding to our sub-mix. 2946 state = sq->begin(); 2947 ALOG_ASSERT(state->mTrackMask == 1); 2948 FastTrack *fastTrack = &state->mFastTracks[0]; 2949 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2950 delete fastTrack->mBufferProvider; 2951 sq->end(false /*didModify*/); 2952 mFastMixer.clear(); 2953#ifdef AUDIO_WATCHDOG 2954 if (mAudioWatchdog != 0) { 2955 mAudioWatchdog->requestExit(); 2956 mAudioWatchdog->requestExitAndWait(); 2957 mAudioWatchdog.clear(); 2958 } 2959#endif 2960 } 2961 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2962 delete mAudioMixer; 2963} 2964 2965 2966uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2967{ 2968 if (mFastMixer != 0) { 2969 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2970 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2971 } 2972 return latency; 2973} 2974 2975 2976void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2977{ 2978 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2979} 2980 2981ssize_t AudioFlinger::MixerThread::threadLoop_write() 2982{ 2983 // FIXME we should only do one push per cycle; confirm this is true 2984 // Start the fast mixer if it's not already running 2985 if (mFastMixer != 0) { 2986 FastMixerStateQueue *sq = mFastMixer->sq(); 2987 FastMixerState *state = sq->begin(); 2988 if (state->mCommand != FastMixerState::MIX_WRITE && 2989 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2990 if (state->mCommand == FastMixerState::COLD_IDLE) { 2991 int32_t old = android_atomic_inc(&mFastMixerFutex); 2992 if (old == -1) { 2993 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2994 } 2995#ifdef AUDIO_WATCHDOG 2996 if (mAudioWatchdog != 0) { 2997 mAudioWatchdog->resume(); 2998 } 2999#endif 3000 } 3001 state->mCommand = FastMixerState::MIX_WRITE; 3002 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3003 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 3004 sq->end(); 3005 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3006 if (kUseFastMixer == FastMixer_Dynamic) { 3007 mNormalSink = mPipeSink; 3008 } 3009 } else { 3010 sq->end(false /*didModify*/); 3011 } 3012 } 3013 return PlaybackThread::threadLoop_write(); 3014} 3015 3016void AudioFlinger::MixerThread::threadLoop_standby() 3017{ 3018 // Idle the fast mixer if it's currently running 3019 if (mFastMixer != 0) { 3020 FastMixerStateQueue *sq = mFastMixer->sq(); 3021 FastMixerState *state = sq->begin(); 3022 if (!(state->mCommand & FastMixerState::IDLE)) { 3023 state->mCommand = FastMixerState::COLD_IDLE; 3024 state->mColdFutexAddr = &mFastMixerFutex; 3025 state->mColdGen++; 3026 mFastMixerFutex = 0; 3027 sq->end(); 3028 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3029 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3030 if (kUseFastMixer == FastMixer_Dynamic) { 3031 mNormalSink = mOutputSink; 3032 } 3033#ifdef AUDIO_WATCHDOG 3034 if (mAudioWatchdog != 0) { 3035 mAudioWatchdog->pause(); 3036 } 3037#endif 3038 } else { 3039 sq->end(false /*didModify*/); 3040 } 3041 } 3042 PlaybackThread::threadLoop_standby(); 3043} 3044 3045bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3046{ 3047 return false; 3048} 3049 3050bool AudioFlinger::PlaybackThread::shouldStandby_l() 3051{ 3052 return !mStandby; 3053} 3054 3055bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3056{ 3057 Mutex::Autolock _l(mLock); 3058 return waitingAsyncCallback_l(); 3059} 3060 3061// shared by MIXER and DIRECT, overridden by DUPLICATING 3062void AudioFlinger::PlaybackThread::threadLoop_standby() 3063{ 3064 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3065 mOutput->stream->common.standby(&mOutput->stream->common); 3066 if (mUseAsyncWrite != 0) { 3067 // discard any pending drain or write ack by incrementing sequence 3068 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3069 mDrainSequence = (mDrainSequence + 2) & ~1; 3070 ALOG_ASSERT(mCallbackThread != 0); 3071 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3072 mCallbackThread->setDraining(mDrainSequence); 3073 } 3074} 3075 3076void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3077{ 3078 ALOGV("signal playback thread"); 3079 broadcast_l(); 3080} 3081 3082void AudioFlinger::MixerThread::threadLoop_mix() 3083{ 3084 // obtain the presentation timestamp of the next output buffer 3085 int64_t pts; 3086 status_t status = INVALID_OPERATION; 3087 3088 if (mNormalSink != 0) { 3089 status = mNormalSink->getNextWriteTimestamp(&pts); 3090 } else { 3091 status = mOutputSink->getNextWriteTimestamp(&pts); 3092 } 3093 3094 if (status != NO_ERROR) { 3095 pts = AudioBufferProvider::kInvalidPTS; 3096 } 3097 3098 // mix buffers... 3099 mAudioMixer->process(pts); 3100 mCurrentWriteLength = mSinkBufferSize; 3101 // increase sleep time progressively when application underrun condition clears. 3102 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3103 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3104 // such that we would underrun the audio HAL. 3105 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3106 sleepTimeShift--; 3107 } 3108 sleepTime = 0; 3109 standbyTime = systemTime() + standbyDelay; 3110 //TODO: delay standby when effects have a tail 3111 3112} 3113 3114void AudioFlinger::MixerThread::threadLoop_sleepTime() 3115{ 3116 // If no tracks are ready, sleep once for the duration of an output 3117 // buffer size, then write 0s to the output 3118 if (sleepTime == 0) { 3119 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3120 sleepTime = activeSleepTime >> sleepTimeShift; 3121 if (sleepTime < kMinThreadSleepTimeUs) { 3122 sleepTime = kMinThreadSleepTimeUs; 3123 } 3124 // reduce sleep time in case of consecutive application underruns to avoid 3125 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3126 // duration we would end up writing less data than needed by the audio HAL if 3127 // the condition persists. 3128 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3129 sleepTimeShift++; 3130 } 3131 } else { 3132 sleepTime = idleSleepTime; 3133 } 3134 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3135 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3136 // before effects processing or output. 3137 if (mMixerBufferValid) { 3138 memset(mMixerBuffer, 0, mMixerBufferSize); 3139 } else { 3140 memset(mSinkBuffer, 0, mSinkBufferSize); 3141 } 3142 sleepTime = 0; 3143 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3144 "anticipated start"); 3145 } 3146 // TODO add standby time extension fct of effect tail 3147} 3148 3149// prepareTracks_l() must be called with ThreadBase::mLock held 3150AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3151 Vector< sp<Track> > *tracksToRemove) 3152{ 3153 3154 mixer_state mixerStatus = MIXER_IDLE; 3155 // find out which tracks need to be processed 3156 size_t count = mActiveTracks.size(); 3157 size_t mixedTracks = 0; 3158 size_t tracksWithEffect = 0; 3159 // counts only _active_ fast tracks 3160 size_t fastTracks = 0; 3161 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3162 3163 float masterVolume = mMasterVolume; 3164 bool masterMute = mMasterMute; 3165 3166 if (masterMute) { 3167 masterVolume = 0; 3168 } 3169 // Delegate master volume control to effect in output mix effect chain if needed 3170 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3171 if (chain != 0) { 3172 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3173 chain->setVolume_l(&v, &v); 3174 masterVolume = (float)((v + (1 << 23)) >> 24); 3175 chain.clear(); 3176 } 3177 3178 // prepare a new state to push 3179 FastMixerStateQueue *sq = NULL; 3180 FastMixerState *state = NULL; 3181 bool didModify = false; 3182 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3183 if (mFastMixer != 0) { 3184 sq = mFastMixer->sq(); 3185 state = sq->begin(); 3186 } 3187 3188 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3189 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3190 3191 for (size_t i=0 ; i<count ; i++) { 3192 const sp<Track> t = mActiveTracks[i].promote(); 3193 if (t == 0) { 3194 continue; 3195 } 3196 3197 // this const just means the local variable doesn't change 3198 Track* const track = t.get(); 3199 3200 // process fast tracks 3201 if (track->isFastTrack()) { 3202 3203 // It's theoretically possible (though unlikely) for a fast track to be created 3204 // and then removed within the same normal mix cycle. This is not a problem, as 3205 // the track never becomes active so it's fast mixer slot is never touched. 3206 // The converse, of removing an (active) track and then creating a new track 3207 // at the identical fast mixer slot within the same normal mix cycle, 3208 // is impossible because the slot isn't marked available until the end of each cycle. 3209 int j = track->mFastIndex; 3210 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3211 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3212 FastTrack *fastTrack = &state->mFastTracks[j]; 3213 3214 // Determine whether the track is currently in underrun condition, 3215 // and whether it had a recent underrun. 3216 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3217 FastTrackUnderruns underruns = ftDump->mUnderruns; 3218 uint32_t recentFull = (underruns.mBitFields.mFull - 3219 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3220 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3221 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3222 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3223 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3224 uint32_t recentUnderruns = recentPartial + recentEmpty; 3225 track->mObservedUnderruns = underruns; 3226 // don't count underruns that occur while stopping or pausing 3227 // or stopped which can occur when flush() is called while active 3228 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3229 recentUnderruns > 0) { 3230 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3231 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3232 } 3233 3234 // This is similar to the state machine for normal tracks, 3235 // with a few modifications for fast tracks. 3236 bool isActive = true; 3237 switch (track->mState) { 3238 case TrackBase::STOPPING_1: 3239 // track stays active in STOPPING_1 state until first underrun 3240 if (recentUnderruns > 0 || track->isTerminated()) { 3241 track->mState = TrackBase::STOPPING_2; 3242 } 3243 break; 3244 case TrackBase::PAUSING: 3245 // ramp down is not yet implemented 3246 track->setPaused(); 3247 break; 3248 case TrackBase::RESUMING: 3249 // ramp up is not yet implemented 3250 track->mState = TrackBase::ACTIVE; 3251 break; 3252 case TrackBase::ACTIVE: 3253 if (recentFull > 0 || recentPartial > 0) { 3254 // track has provided at least some frames recently: reset retry count 3255 track->mRetryCount = kMaxTrackRetries; 3256 } 3257 if (recentUnderruns == 0) { 3258 // no recent underruns: stay active 3259 break; 3260 } 3261 // there has recently been an underrun of some kind 3262 if (track->sharedBuffer() == 0) { 3263 // were any of the recent underruns "empty" (no frames available)? 3264 if (recentEmpty == 0) { 3265 // no, then ignore the partial underruns as they are allowed indefinitely 3266 break; 3267 } 3268 // there has recently been an "empty" underrun: decrement the retry counter 3269 if (--(track->mRetryCount) > 0) { 3270 break; 3271 } 3272 // indicate to client process that the track was disabled because of underrun; 3273 // it will then automatically call start() when data is available 3274 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3275 // remove from active list, but state remains ACTIVE [confusing but true] 3276 isActive = false; 3277 break; 3278 } 3279 // fall through 3280 case TrackBase::STOPPING_2: 3281 case TrackBase::PAUSED: 3282 case TrackBase::STOPPED: 3283 case TrackBase::FLUSHED: // flush() while active 3284 // Check for presentation complete if track is inactive 3285 // We have consumed all the buffers of this track. 3286 // This would be incomplete if we auto-paused on underrun 3287 { 3288 size_t audioHALFrames = 3289 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3290 size_t framesWritten = mBytesWritten / mFrameSize; 3291 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3292 // track stays in active list until presentation is complete 3293 break; 3294 } 3295 } 3296 if (track->isStopping_2()) { 3297 track->mState = TrackBase::STOPPED; 3298 } 3299 if (track->isStopped()) { 3300 // Can't reset directly, as fast mixer is still polling this track 3301 // track->reset(); 3302 // So instead mark this track as needing to be reset after push with ack 3303 resetMask |= 1 << i; 3304 } 3305 isActive = false; 3306 break; 3307 case TrackBase::IDLE: 3308 default: 3309 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3310 } 3311 3312 if (isActive) { 3313 // was it previously inactive? 3314 if (!(state->mTrackMask & (1 << j))) { 3315 ExtendedAudioBufferProvider *eabp = track; 3316 VolumeProvider *vp = track; 3317 fastTrack->mBufferProvider = eabp; 3318 fastTrack->mVolumeProvider = vp; 3319 fastTrack->mChannelMask = track->mChannelMask; 3320 fastTrack->mFormat = track->mFormat; 3321 fastTrack->mGeneration++; 3322 state->mTrackMask |= 1 << j; 3323 didModify = true; 3324 // no acknowledgement required for newly active tracks 3325 } 3326 // cache the combined master volume and stream type volume for fast mixer; this 3327 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3328 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3329 ++fastTracks; 3330 } else { 3331 // was it previously active? 3332 if (state->mTrackMask & (1 << j)) { 3333 fastTrack->mBufferProvider = NULL; 3334 fastTrack->mGeneration++; 3335 state->mTrackMask &= ~(1 << j); 3336 didModify = true; 3337 // If any fast tracks were removed, we must wait for acknowledgement 3338 // because we're about to decrement the last sp<> on those tracks. 3339 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3340 } else { 3341 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3342 } 3343 tracksToRemove->add(track); 3344 // Avoids a misleading display in dumpsys 3345 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3346 } 3347 continue; 3348 } 3349 3350 { // local variable scope to avoid goto warning 3351 3352 audio_track_cblk_t* cblk = track->cblk(); 3353 3354 // The first time a track is added we wait 3355 // for all its buffers to be filled before processing it 3356 int name = track->name(); 3357 // make sure that we have enough frames to mix one full buffer. 3358 // enforce this condition only once to enable draining the buffer in case the client 3359 // app does not call stop() and relies on underrun to stop: 3360 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3361 // during last round 3362 size_t desiredFrames; 3363 uint32_t sr = track->sampleRate(); 3364 if (sr == mSampleRate) { 3365 desiredFrames = mNormalFrameCount; 3366 } else { 3367 // +1 for rounding and +1 for additional sample needed for interpolation 3368 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3369 // add frames already consumed but not yet released by the resampler 3370 // because mAudioTrackServerProxy->framesReady() will include these frames 3371 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3372#if 0 3373 // the minimum track buffer size is normally twice the number of frames necessary 3374 // to fill one buffer and the resampler should not leave more than one buffer worth 3375 // of unreleased frames after each pass, but just in case... 3376 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3377#endif 3378 } 3379 uint32_t minFrames = 1; 3380 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3381 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3382 minFrames = desiredFrames; 3383 } 3384 3385 size_t framesReady = track->framesReady(); 3386 if ((framesReady >= minFrames) && track->isReady() && 3387 !track->isPaused() && !track->isTerminated()) 3388 { 3389 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3390 3391 mixedTracks++; 3392 3393 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3394 // there is an effect chain connected to the track 3395 chain.clear(); 3396 if (track->mainBuffer() != mSinkBuffer && 3397 track->mainBuffer() != mMixerBuffer) { 3398 if (mEffectBufferEnabled) { 3399 mEffectBufferValid = true; // Later can set directly. 3400 } 3401 chain = getEffectChain_l(track->sessionId()); 3402 // Delegate volume control to effect in track effect chain if needed 3403 if (chain != 0) { 3404 tracksWithEffect++; 3405 } else { 3406 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3407 "session %d", 3408 name, track->sessionId()); 3409 } 3410 } 3411 3412 3413 int param = AudioMixer::VOLUME; 3414 if (track->mFillingUpStatus == Track::FS_FILLED) { 3415 // no ramp for the first volume setting 3416 track->mFillingUpStatus = Track::FS_ACTIVE; 3417 if (track->mState == TrackBase::RESUMING) { 3418 track->mState = TrackBase::ACTIVE; 3419 param = AudioMixer::RAMP_VOLUME; 3420 } 3421 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3422 // FIXME should not make a decision based on mServer 3423 } else if (cblk->mServer != 0) { 3424 // If the track is stopped before the first frame was mixed, 3425 // do not apply ramp 3426 param = AudioMixer::RAMP_VOLUME; 3427 } 3428 3429 // compute volume for this track 3430 uint32_t vl, vr; // in U8.24 integer format 3431 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3432 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3433 vl = vr = 0; 3434 vlf = vrf = vaf = 0.; 3435 if (track->isPausing()) { 3436 track->setPaused(); 3437 } 3438 } else { 3439 3440 // read original volumes with volume control 3441 float typeVolume = mStreamTypes[track->streamType()].volume; 3442 float v = masterVolume * typeVolume; 3443 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3444 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3445 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3446 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3447 // track volumes come from shared memory, so can't be trusted and must be clamped 3448 if (vlf > GAIN_FLOAT_UNITY) { 3449 ALOGV("Track left volume out of range: %.3g", vlf); 3450 vlf = GAIN_FLOAT_UNITY; 3451 } 3452 if (vrf > GAIN_FLOAT_UNITY) { 3453 ALOGV("Track right volume out of range: %.3g", vrf); 3454 vrf = GAIN_FLOAT_UNITY; 3455 } 3456 // now apply the master volume and stream type volume 3457 vlf *= v; 3458 vrf *= v; 3459 // assuming master volume and stream type volume each go up to 1.0, 3460 // then derive vl and vr as U8.24 versions for the effect chain 3461 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3462 vl = (uint32_t) (scaleto8_24 * vlf); 3463 vr = (uint32_t) (scaleto8_24 * vrf); 3464 // vl and vr are now in U8.24 format 3465 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3466 // send level comes from shared memory and so may be corrupt 3467 if (sendLevel > MAX_GAIN_INT) { 3468 ALOGV("Track send level out of range: %04X", sendLevel); 3469 sendLevel = MAX_GAIN_INT; 3470 } 3471 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3472 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3473 } 3474 3475 // Delegate volume control to effect in track effect chain if needed 3476 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3477 // Do not ramp volume if volume is controlled by effect 3478 param = AudioMixer::VOLUME; 3479 // Update remaining floating point volume levels 3480 vlf = (float)vl / (1 << 24); 3481 vrf = (float)vr / (1 << 24); 3482 track->mHasVolumeController = true; 3483 } else { 3484 // force no volume ramp when volume controller was just disabled or removed 3485 // from effect chain to avoid volume spike 3486 if (track->mHasVolumeController) { 3487 param = AudioMixer::VOLUME; 3488 } 3489 track->mHasVolumeController = false; 3490 } 3491 3492 // XXX: these things DON'T need to be done each time 3493 mAudioMixer->setBufferProvider(name, track); 3494 mAudioMixer->enable(name); 3495 3496 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3497 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3498 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3499 mAudioMixer->setParameter( 3500 name, 3501 AudioMixer::TRACK, 3502 AudioMixer::FORMAT, (void *)track->format()); 3503 mAudioMixer->setParameter( 3504 name, 3505 AudioMixer::TRACK, 3506 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3507 mAudioMixer->setParameter( 3508 name, 3509 AudioMixer::TRACK, 3510 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3511 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3512 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3513 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3514 if (reqSampleRate == 0) { 3515 reqSampleRate = mSampleRate; 3516 } else if (reqSampleRate > maxSampleRate) { 3517 reqSampleRate = maxSampleRate; 3518 } 3519 mAudioMixer->setParameter( 3520 name, 3521 AudioMixer::RESAMPLE, 3522 AudioMixer::SAMPLE_RATE, 3523 (void *)(uintptr_t)reqSampleRate); 3524 /* 3525 * Select the appropriate output buffer for the track. 3526 * 3527 * Tracks with effects go into their own effects chain buffer 3528 * and from there into either mEffectBuffer or mSinkBuffer. 3529 * 3530 * Other tracks can use mMixerBuffer for higher precision 3531 * channel accumulation. If this buffer is enabled 3532 * (mMixerBufferEnabled true), then selected tracks will accumulate 3533 * into it. 3534 * 3535 */ 3536 if (mMixerBufferEnabled 3537 && (track->mainBuffer() == mSinkBuffer 3538 || track->mainBuffer() == mMixerBuffer)) { 3539 mAudioMixer->setParameter( 3540 name, 3541 AudioMixer::TRACK, 3542 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3543 mAudioMixer->setParameter( 3544 name, 3545 AudioMixer::TRACK, 3546 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3547 // TODO: override track->mainBuffer()? 3548 mMixerBufferValid = true; 3549 } else { 3550 mAudioMixer->setParameter( 3551 name, 3552 AudioMixer::TRACK, 3553 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3554 mAudioMixer->setParameter( 3555 name, 3556 AudioMixer::TRACK, 3557 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3558 } 3559 mAudioMixer->setParameter( 3560 name, 3561 AudioMixer::TRACK, 3562 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3563 3564 // reset retry count 3565 track->mRetryCount = kMaxTrackRetries; 3566 3567 // If one track is ready, set the mixer ready if: 3568 // - the mixer was not ready during previous round OR 3569 // - no other track is not ready 3570 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3571 mixerStatus != MIXER_TRACKS_ENABLED) { 3572 mixerStatus = MIXER_TRACKS_READY; 3573 } 3574 } else { 3575 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3576 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3577 } 3578 // clear effect chain input buffer if an active track underruns to avoid sending 3579 // previous audio buffer again to effects 3580 chain = getEffectChain_l(track->sessionId()); 3581 if (chain != 0) { 3582 chain->clearInputBuffer(); 3583 } 3584 3585 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3586 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3587 track->isStopped() || track->isPaused()) { 3588 // We have consumed all the buffers of this track. 3589 // Remove it from the list of active tracks. 3590 // TODO: use actual buffer filling status instead of latency when available from 3591 // audio HAL 3592 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3593 size_t framesWritten = mBytesWritten / mFrameSize; 3594 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3595 if (track->isStopped()) { 3596 track->reset(); 3597 } 3598 tracksToRemove->add(track); 3599 } 3600 } else { 3601 // No buffers for this track. Give it a few chances to 3602 // fill a buffer, then remove it from active list. 3603 if (--(track->mRetryCount) <= 0) { 3604 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3605 tracksToRemove->add(track); 3606 // indicate to client process that the track was disabled because of underrun; 3607 // it will then automatically call start() when data is available 3608 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3609 // If one track is not ready, mark the mixer also not ready if: 3610 // - the mixer was ready during previous round OR 3611 // - no other track is ready 3612 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3613 mixerStatus != MIXER_TRACKS_READY) { 3614 mixerStatus = MIXER_TRACKS_ENABLED; 3615 } 3616 } 3617 mAudioMixer->disable(name); 3618 } 3619 3620 } // local variable scope to avoid goto warning 3621track_is_ready: ; 3622 3623 } 3624 3625 // Push the new FastMixer state if necessary 3626 bool pauseAudioWatchdog = false; 3627 if (didModify) { 3628 state->mFastTracksGen++; 3629 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3630 if (kUseFastMixer == FastMixer_Dynamic && 3631 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3632 state->mCommand = FastMixerState::COLD_IDLE; 3633 state->mColdFutexAddr = &mFastMixerFutex; 3634 state->mColdGen++; 3635 mFastMixerFutex = 0; 3636 if (kUseFastMixer == FastMixer_Dynamic) { 3637 mNormalSink = mOutputSink; 3638 } 3639 // If we go into cold idle, need to wait for acknowledgement 3640 // so that fast mixer stops doing I/O. 3641 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3642 pauseAudioWatchdog = true; 3643 } 3644 } 3645 if (sq != NULL) { 3646 sq->end(didModify); 3647 sq->push(block); 3648 } 3649#ifdef AUDIO_WATCHDOG 3650 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3651 mAudioWatchdog->pause(); 3652 } 3653#endif 3654 3655 // Now perform the deferred reset on fast tracks that have stopped 3656 while (resetMask != 0) { 3657 size_t i = __builtin_ctz(resetMask); 3658 ALOG_ASSERT(i < count); 3659 resetMask &= ~(1 << i); 3660 sp<Track> t = mActiveTracks[i].promote(); 3661 if (t == 0) { 3662 continue; 3663 } 3664 Track* track = t.get(); 3665 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3666 track->reset(); 3667 } 3668 3669 // remove all the tracks that need to be... 3670 removeTracks_l(*tracksToRemove); 3671 3672 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3673 mEffectBufferValid = true; 3674 } 3675 3676 if (mEffectBufferValid) { 3677 // as long as there are effects we should clear the effects buffer, to avoid 3678 // passing a non-clean buffer to the effect chain 3679 memset(mEffectBuffer, 0, mEffectBufferSize); 3680 } 3681 // sink or mix buffer must be cleared if all tracks are connected to an 3682 // effect chain as in this case the mixer will not write to the sink or mix buffer 3683 // and track effects will accumulate into it 3684 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3685 (mixedTracks == 0 && fastTracks > 0))) { 3686 // FIXME as a performance optimization, should remember previous zero status 3687 if (mMixerBufferValid) { 3688 memset(mMixerBuffer, 0, mMixerBufferSize); 3689 // TODO: In testing, mSinkBuffer below need not be cleared because 3690 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3691 // after mixing. 3692 // 3693 // To enforce this guarantee: 3694 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3695 // (mixedTracks == 0 && fastTracks > 0)) 3696 // must imply MIXER_TRACKS_READY. 3697 // Later, we may clear buffers regardless, and skip much of this logic. 3698 } 3699 // FIXME as a performance optimization, should remember previous zero status 3700 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3701 } 3702 3703 // if any fast tracks, then status is ready 3704 mMixerStatusIgnoringFastTracks = mixerStatus; 3705 if (fastTracks > 0) { 3706 mixerStatus = MIXER_TRACKS_READY; 3707 } 3708 return mixerStatus; 3709} 3710 3711// getTrackName_l() must be called with ThreadBase::mLock held 3712int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3713 audio_format_t format, int sessionId) 3714{ 3715 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3716} 3717 3718// deleteTrackName_l() must be called with ThreadBase::mLock held 3719void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3720{ 3721 ALOGV("remove track (%d) and delete from mixer", name); 3722 mAudioMixer->deleteTrackName(name); 3723} 3724 3725// checkForNewParameter_l() must be called with ThreadBase::mLock held 3726bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3727 status_t& status) 3728{ 3729 bool reconfig = false; 3730 3731 status = NO_ERROR; 3732 3733 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3734 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3735 if (mFastMixer != 0) { 3736 FastMixerStateQueue *sq = mFastMixer->sq(); 3737 FastMixerState *state = sq->begin(); 3738 if (!(state->mCommand & FastMixerState::IDLE)) { 3739 previousCommand = state->mCommand; 3740 state->mCommand = FastMixerState::HOT_IDLE; 3741 sq->end(); 3742 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3743 } else { 3744 sq->end(false /*didModify*/); 3745 } 3746 } 3747 3748 AudioParameter param = AudioParameter(keyValuePair); 3749 int value; 3750 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3751 reconfig = true; 3752 } 3753 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3754 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3755 status = BAD_VALUE; 3756 } else { 3757 // no need to save value, since it's constant 3758 reconfig = true; 3759 } 3760 } 3761 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3762 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3763 status = BAD_VALUE; 3764 } else { 3765 // no need to save value, since it's constant 3766 reconfig = true; 3767 } 3768 } 3769 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3770 // do not accept frame count changes if tracks are open as the track buffer 3771 // size depends on frame count and correct behavior would not be guaranteed 3772 // if frame count is changed after track creation 3773 if (!mTracks.isEmpty()) { 3774 status = INVALID_OPERATION; 3775 } else { 3776 reconfig = true; 3777 } 3778 } 3779 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3780#ifdef ADD_BATTERY_DATA 3781 // when changing the audio output device, call addBatteryData to notify 3782 // the change 3783 if (mOutDevice != value) { 3784 uint32_t params = 0; 3785 // check whether speaker is on 3786 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3787 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3788 } 3789 3790 audio_devices_t deviceWithoutSpeaker 3791 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3792 // check if any other device (except speaker) is on 3793 if (value & deviceWithoutSpeaker ) { 3794 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3795 } 3796 3797 if (params != 0) { 3798 addBatteryData(params); 3799 } 3800 } 3801#endif 3802 3803 // forward device change to effects that have requested to be 3804 // aware of attached audio device. 3805 if (value != AUDIO_DEVICE_NONE) { 3806 mOutDevice = value; 3807 for (size_t i = 0; i < mEffectChains.size(); i++) { 3808 mEffectChains[i]->setDevice_l(mOutDevice); 3809 } 3810 } 3811 } 3812 3813 if (status == NO_ERROR) { 3814 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3815 keyValuePair.string()); 3816 if (!mStandby && status == INVALID_OPERATION) { 3817 mOutput->stream->common.standby(&mOutput->stream->common); 3818 mStandby = true; 3819 mBytesWritten = 0; 3820 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3821 keyValuePair.string()); 3822 } 3823 if (status == NO_ERROR && reconfig) { 3824 readOutputParameters_l(); 3825 delete mAudioMixer; 3826 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3827 for (size_t i = 0; i < mTracks.size() ; i++) { 3828 int name = getTrackName_l(mTracks[i]->mChannelMask, 3829 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3830 if (name < 0) { 3831 break; 3832 } 3833 mTracks[i]->mName = name; 3834 } 3835 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3836 } 3837 } 3838 3839 if (!(previousCommand & FastMixerState::IDLE)) { 3840 ALOG_ASSERT(mFastMixer != 0); 3841 FastMixerStateQueue *sq = mFastMixer->sq(); 3842 FastMixerState *state = sq->begin(); 3843 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3844 state->mCommand = previousCommand; 3845 sq->end(); 3846 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3847 } 3848 3849 return reconfig; 3850} 3851 3852 3853void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3854{ 3855 const size_t SIZE = 256; 3856 char buffer[SIZE]; 3857 String8 result; 3858 3859 PlaybackThread::dumpInternals(fd, args); 3860 3861 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3862 3863 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3864 const FastMixerDumpState copy(mFastMixerDumpState); 3865 copy.dump(fd); 3866 3867#ifdef STATE_QUEUE_DUMP 3868 // Similar for state queue 3869 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3870 observerCopy.dump(fd); 3871 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3872 mutatorCopy.dump(fd); 3873#endif 3874 3875#ifdef TEE_SINK 3876 // Write the tee output to a .wav file 3877 dumpTee(fd, mTeeSource, mId); 3878#endif 3879 3880#ifdef AUDIO_WATCHDOG 3881 if (mAudioWatchdog != 0) { 3882 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3883 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3884 wdCopy.dump(fd); 3885 } 3886#endif 3887} 3888 3889uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3890{ 3891 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3892} 3893 3894uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3895{ 3896 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3897} 3898 3899void AudioFlinger::MixerThread::cacheParameters_l() 3900{ 3901 PlaybackThread::cacheParameters_l(); 3902 3903 // FIXME: Relaxed timing because of a certain device that can't meet latency 3904 // Should be reduced to 2x after the vendor fixes the driver issue 3905 // increase threshold again due to low power audio mode. The way this warning 3906 // threshold is calculated and its usefulness should be reconsidered anyway. 3907 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3908} 3909 3910// ---------------------------------------------------------------------------- 3911 3912AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3913 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3914 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3915 // mLeftVolFloat, mRightVolFloat 3916{ 3917} 3918 3919AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3920 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3921 ThreadBase::type_t type) 3922 : PlaybackThread(audioFlinger, output, id, device, type) 3923 // mLeftVolFloat, mRightVolFloat 3924{ 3925} 3926 3927AudioFlinger::DirectOutputThread::~DirectOutputThread() 3928{ 3929} 3930 3931void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3932{ 3933 audio_track_cblk_t* cblk = track->cblk(); 3934 float left, right; 3935 3936 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3937 left = right = 0; 3938 } else { 3939 float typeVolume = mStreamTypes[track->streamType()].volume; 3940 float v = mMasterVolume * typeVolume; 3941 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3942 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3943 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3944 if (left > GAIN_FLOAT_UNITY) { 3945 left = GAIN_FLOAT_UNITY; 3946 } 3947 left *= v; 3948 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3949 if (right > GAIN_FLOAT_UNITY) { 3950 right = GAIN_FLOAT_UNITY; 3951 } 3952 right *= v; 3953 } 3954 3955 if (lastTrack) { 3956 if (left != mLeftVolFloat || right != mRightVolFloat) { 3957 mLeftVolFloat = left; 3958 mRightVolFloat = right; 3959 3960 // Convert volumes from float to 8.24 3961 uint32_t vl = (uint32_t)(left * (1 << 24)); 3962 uint32_t vr = (uint32_t)(right * (1 << 24)); 3963 3964 // Delegate volume control to effect in track effect chain if needed 3965 // only one effect chain can be present on DirectOutputThread, so if 3966 // there is one, the track is connected to it 3967 if (!mEffectChains.isEmpty()) { 3968 mEffectChains[0]->setVolume_l(&vl, &vr); 3969 left = (float)vl / (1 << 24); 3970 right = (float)vr / (1 << 24); 3971 } 3972 if (mOutput->stream->set_volume) { 3973 mOutput->stream->set_volume(mOutput->stream, left, right); 3974 } 3975 } 3976 } 3977} 3978 3979 3980AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3981 Vector< sp<Track> > *tracksToRemove 3982) 3983{ 3984 size_t count = mActiveTracks.size(); 3985 mixer_state mixerStatus = MIXER_IDLE; 3986 3987 // find out which tracks need to be processed 3988 for (size_t i = 0; i < count; i++) { 3989 sp<Track> t = mActiveTracks[i].promote(); 3990 // The track died recently 3991 if (t == 0) { 3992 continue; 3993 } 3994 3995 Track* const track = t.get(); 3996 audio_track_cblk_t* cblk = track->cblk(); 3997 // Only consider last track started for volume and mixer state control. 3998 // In theory an older track could underrun and restart after the new one starts 3999 // but as we only care about the transition phase between two tracks on a 4000 // direct output, it is not a problem to ignore the underrun case. 4001 sp<Track> l = mLatestActiveTrack.promote(); 4002 bool last = l.get() == track; 4003 4004 // The first time a track is added we wait 4005 // for all its buffers to be filled before processing it 4006 uint32_t minFrames; 4007 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) { 4008 minFrames = mNormalFrameCount; 4009 } else { 4010 minFrames = 1; 4011 } 4012 4013 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4014 !track->isStopping_2() && !track->isStopped()) 4015 { 4016 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4017 4018 if (track->mFillingUpStatus == Track::FS_FILLED) { 4019 track->mFillingUpStatus = Track::FS_ACTIVE; 4020 // make sure processVolume_l() will apply new volume even if 0 4021 mLeftVolFloat = mRightVolFloat = -1.0; 4022 if (track->mState == TrackBase::RESUMING) { 4023 track->mState = TrackBase::ACTIVE; 4024 } 4025 } 4026 4027 // compute volume for this track 4028 processVolume_l(track, last); 4029 if (last) { 4030 // reset retry count 4031 track->mRetryCount = kMaxTrackRetriesDirect; 4032 mActiveTrack = t; 4033 mixerStatus = MIXER_TRACKS_READY; 4034 } 4035 } else { 4036 // clear effect chain input buffer if the last active track started underruns 4037 // to avoid sending previous audio buffer again to effects 4038 if (!mEffectChains.isEmpty() && last) { 4039 mEffectChains[0]->clearInputBuffer(); 4040 } 4041 if (track->isStopping_1()) { 4042 track->mState = TrackBase::STOPPING_2; 4043 } 4044 if ((track->sharedBuffer() != 0) || track->isStopped() || 4045 track->isStopping_2() || track->isPaused()) { 4046 // We have consumed all the buffers of this track. 4047 // Remove it from the list of active tracks. 4048 size_t audioHALFrames; 4049 if (audio_is_linear_pcm(mFormat)) { 4050 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4051 } else { 4052 audioHALFrames = 0; 4053 } 4054 4055 size_t framesWritten = mBytesWritten / mFrameSize; 4056 if (mStandby || !last || 4057 track->presentationComplete(framesWritten, audioHALFrames)) { 4058 if (track->isStopping_2()) { 4059 track->mState = TrackBase::STOPPED; 4060 } 4061 if (track->isStopped()) { 4062 if (track->mState == TrackBase::FLUSHED) { 4063 flushHw_l(); 4064 } 4065 track->reset(); 4066 } 4067 tracksToRemove->add(track); 4068 } 4069 } else { 4070 // No buffers for this track. Give it a few chances to 4071 // fill a buffer, then remove it from active list. 4072 // Only consider last track started for mixer state control 4073 if (--(track->mRetryCount) <= 0) { 4074 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4075 tracksToRemove->add(track); 4076 // indicate to client process that the track was disabled because of underrun; 4077 // it will then automatically call start() when data is available 4078 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4079 } else if (last) { 4080 mixerStatus = MIXER_TRACKS_ENABLED; 4081 } 4082 } 4083 } 4084 } 4085 4086 // remove all the tracks that need to be... 4087 removeTracks_l(*tracksToRemove); 4088 4089 return mixerStatus; 4090} 4091 4092void AudioFlinger::DirectOutputThread::threadLoop_mix() 4093{ 4094 size_t frameCount = mFrameCount; 4095 int8_t *curBuf = (int8_t *)mSinkBuffer; 4096 // output audio to hardware 4097 while (frameCount) { 4098 AudioBufferProvider::Buffer buffer; 4099 buffer.frameCount = frameCount; 4100 mActiveTrack->getNextBuffer(&buffer); 4101 if (buffer.raw == NULL) { 4102 memset(curBuf, 0, frameCount * mFrameSize); 4103 break; 4104 } 4105 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4106 frameCount -= buffer.frameCount; 4107 curBuf += buffer.frameCount * mFrameSize; 4108 mActiveTrack->releaseBuffer(&buffer); 4109 } 4110 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4111 sleepTime = 0; 4112 standbyTime = systemTime() + standbyDelay; 4113 mActiveTrack.clear(); 4114} 4115 4116void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4117{ 4118 if (sleepTime == 0) { 4119 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4120 sleepTime = activeSleepTime; 4121 } else { 4122 sleepTime = idleSleepTime; 4123 } 4124 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4125 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4126 sleepTime = 0; 4127 } 4128} 4129 4130// getTrackName_l() must be called with ThreadBase::mLock held 4131int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4132 audio_format_t format __unused, int sessionId __unused) 4133{ 4134 return 0; 4135} 4136 4137// deleteTrackName_l() must be called with ThreadBase::mLock held 4138void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4139{ 4140} 4141 4142// checkForNewParameter_l() must be called with ThreadBase::mLock held 4143bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4144 status_t& status) 4145{ 4146 bool reconfig = false; 4147 4148 status = NO_ERROR; 4149 4150 AudioParameter param = AudioParameter(keyValuePair); 4151 int value; 4152 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4153 // forward device change to effects that have requested to be 4154 // aware of attached audio device. 4155 if (value != AUDIO_DEVICE_NONE) { 4156 mOutDevice = value; 4157 for (size_t i = 0; i < mEffectChains.size(); i++) { 4158 mEffectChains[i]->setDevice_l(mOutDevice); 4159 } 4160 } 4161 } 4162 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4163 // do not accept frame count changes if tracks are open as the track buffer 4164 // size depends on frame count and correct behavior would not be garantied 4165 // if frame count is changed after track creation 4166 if (!mTracks.isEmpty()) { 4167 status = INVALID_OPERATION; 4168 } else { 4169 reconfig = true; 4170 } 4171 } 4172 if (status == NO_ERROR) { 4173 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4174 keyValuePair.string()); 4175 if (!mStandby && status == INVALID_OPERATION) { 4176 mOutput->stream->common.standby(&mOutput->stream->common); 4177 mStandby = true; 4178 mBytesWritten = 0; 4179 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4180 keyValuePair.string()); 4181 } 4182 if (status == NO_ERROR && reconfig) { 4183 readOutputParameters_l(); 4184 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4185 } 4186 } 4187 4188 return reconfig; 4189} 4190 4191uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4192{ 4193 uint32_t time; 4194 if (audio_is_linear_pcm(mFormat)) { 4195 time = PlaybackThread::activeSleepTimeUs(); 4196 } else { 4197 time = 10000; 4198 } 4199 return time; 4200} 4201 4202uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4203{ 4204 uint32_t time; 4205 if (audio_is_linear_pcm(mFormat)) { 4206 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4207 } else { 4208 time = 10000; 4209 } 4210 return time; 4211} 4212 4213uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4214{ 4215 uint32_t time; 4216 if (audio_is_linear_pcm(mFormat)) { 4217 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4218 } else { 4219 time = 10000; 4220 } 4221 return time; 4222} 4223 4224void AudioFlinger::DirectOutputThread::cacheParameters_l() 4225{ 4226 PlaybackThread::cacheParameters_l(); 4227 4228 // use shorter standby delay as on normal output to release 4229 // hardware resources as soon as possible 4230 if (audio_is_linear_pcm(mFormat)) { 4231 standbyDelay = microseconds(activeSleepTime*2); 4232 } else { 4233 standbyDelay = kOffloadStandbyDelayNs; 4234 } 4235} 4236 4237void AudioFlinger::DirectOutputThread::flushHw_l() 4238{ 4239 if (mOutput->stream->flush != NULL) 4240 mOutput->stream->flush(mOutput->stream); 4241} 4242 4243// ---------------------------------------------------------------------------- 4244 4245AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4246 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4247 : Thread(false /*canCallJava*/), 4248 mPlaybackThread(playbackThread), 4249 mWriteAckSequence(0), 4250 mDrainSequence(0) 4251{ 4252} 4253 4254AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4255{ 4256} 4257 4258void AudioFlinger::AsyncCallbackThread::onFirstRef() 4259{ 4260 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4261} 4262 4263bool AudioFlinger::AsyncCallbackThread::threadLoop() 4264{ 4265 while (!exitPending()) { 4266 uint32_t writeAckSequence; 4267 uint32_t drainSequence; 4268 4269 { 4270 Mutex::Autolock _l(mLock); 4271 while (!((mWriteAckSequence & 1) || 4272 (mDrainSequence & 1) || 4273 exitPending())) { 4274 mWaitWorkCV.wait(mLock); 4275 } 4276 4277 if (exitPending()) { 4278 break; 4279 } 4280 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4281 mWriteAckSequence, mDrainSequence); 4282 writeAckSequence = mWriteAckSequence; 4283 mWriteAckSequence &= ~1; 4284 drainSequence = mDrainSequence; 4285 mDrainSequence &= ~1; 4286 } 4287 { 4288 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4289 if (playbackThread != 0) { 4290 if (writeAckSequence & 1) { 4291 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4292 } 4293 if (drainSequence & 1) { 4294 playbackThread->resetDraining(drainSequence >> 1); 4295 } 4296 } 4297 } 4298 } 4299 return false; 4300} 4301 4302void AudioFlinger::AsyncCallbackThread::exit() 4303{ 4304 ALOGV("AsyncCallbackThread::exit"); 4305 Mutex::Autolock _l(mLock); 4306 requestExit(); 4307 mWaitWorkCV.broadcast(); 4308} 4309 4310void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4311{ 4312 Mutex::Autolock _l(mLock); 4313 // bit 0 is cleared 4314 mWriteAckSequence = sequence << 1; 4315} 4316 4317void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4318{ 4319 Mutex::Autolock _l(mLock); 4320 // ignore unexpected callbacks 4321 if (mWriteAckSequence & 2) { 4322 mWriteAckSequence |= 1; 4323 mWaitWorkCV.signal(); 4324 } 4325} 4326 4327void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4328{ 4329 Mutex::Autolock _l(mLock); 4330 // bit 0 is cleared 4331 mDrainSequence = sequence << 1; 4332} 4333 4334void AudioFlinger::AsyncCallbackThread::resetDraining() 4335{ 4336 Mutex::Autolock _l(mLock); 4337 // ignore unexpected callbacks 4338 if (mDrainSequence & 2) { 4339 mDrainSequence |= 1; 4340 mWaitWorkCV.signal(); 4341 } 4342} 4343 4344 4345// ---------------------------------------------------------------------------- 4346AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4347 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4348 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4349 mHwPaused(false), 4350 mFlushPending(false), 4351 mPausedBytesRemaining(0) 4352{ 4353 //FIXME: mStandby should be set to true by ThreadBase constructor 4354 mStandby = true; 4355} 4356 4357void AudioFlinger::OffloadThread::threadLoop_exit() 4358{ 4359 if (mFlushPending || mHwPaused) { 4360 // If a flush is pending or track was paused, just discard buffered data 4361 flushHw_l(); 4362 } else { 4363 mMixerStatus = MIXER_DRAIN_ALL; 4364 threadLoop_drain(); 4365 } 4366 if (mUseAsyncWrite) { 4367 ALOG_ASSERT(mCallbackThread != 0); 4368 mCallbackThread->exit(); 4369 } 4370 PlaybackThread::threadLoop_exit(); 4371} 4372 4373AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4374 Vector< sp<Track> > *tracksToRemove 4375) 4376{ 4377 size_t count = mActiveTracks.size(); 4378 4379 mixer_state mixerStatus = MIXER_IDLE; 4380 bool doHwPause = false; 4381 bool doHwResume = false; 4382 4383 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4384 4385 // find out which tracks need to be processed 4386 for (size_t i = 0; i < count; i++) { 4387 sp<Track> t = mActiveTracks[i].promote(); 4388 // The track died recently 4389 if (t == 0) { 4390 continue; 4391 } 4392 Track* const track = t.get(); 4393 audio_track_cblk_t* cblk = track->cblk(); 4394 // Only consider last track started for volume and mixer state control. 4395 // In theory an older track could underrun and restart after the new one starts 4396 // but as we only care about the transition phase between two tracks on a 4397 // direct output, it is not a problem to ignore the underrun case. 4398 sp<Track> l = mLatestActiveTrack.promote(); 4399 bool last = l.get() == track; 4400 4401 if (track->isInvalid()) { 4402 ALOGW("An invalidated track shouldn't be in active list"); 4403 tracksToRemove->add(track); 4404 continue; 4405 } 4406 4407 if (track->mState == TrackBase::IDLE) { 4408 ALOGW("An idle track shouldn't be in active list"); 4409 continue; 4410 } 4411 4412 if (track->isPausing()) { 4413 track->setPaused(); 4414 if (last) { 4415 if (!mHwPaused) { 4416 doHwPause = true; 4417 mHwPaused = true; 4418 } 4419 // If we were part way through writing the mixbuffer to 4420 // the HAL we must save this until we resume 4421 // BUG - this will be wrong if a different track is made active, 4422 // in that case we want to discard the pending data in the 4423 // mixbuffer and tell the client to present it again when the 4424 // track is resumed 4425 mPausedWriteLength = mCurrentWriteLength; 4426 mPausedBytesRemaining = mBytesRemaining; 4427 mBytesRemaining = 0; // stop writing 4428 } 4429 tracksToRemove->add(track); 4430 } else if (track->isFlushPending()) { 4431 track->flushAck(); 4432 if (last) { 4433 mFlushPending = true; 4434 } 4435 } else if (track->isResumePending()){ 4436 track->resumeAck(); 4437 if (last) { 4438 if (mPausedBytesRemaining) { 4439 // Need to continue write that was interrupted 4440 mCurrentWriteLength = mPausedWriteLength; 4441 mBytesRemaining = mPausedBytesRemaining; 4442 mPausedBytesRemaining = 0; 4443 } 4444 if (mHwPaused) { 4445 doHwResume = true; 4446 mHwPaused = false; 4447 // threadLoop_mix() will handle the case that we need to 4448 // resume an interrupted write 4449 } 4450 // enable write to audio HAL 4451 sleepTime = 0; 4452 4453 // Do not handle new data in this iteration even if track->framesReady() 4454 mixerStatus = MIXER_TRACKS_ENABLED; 4455 } 4456 } else if (track->framesReady() && track->isReady() && 4457 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4458 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4459 if (track->mFillingUpStatus == Track::FS_FILLED) { 4460 track->mFillingUpStatus = Track::FS_ACTIVE; 4461 // make sure processVolume_l() will apply new volume even if 0 4462 mLeftVolFloat = mRightVolFloat = -1.0; 4463 } 4464 4465 if (last) { 4466 sp<Track> previousTrack = mPreviousTrack.promote(); 4467 if (previousTrack != 0) { 4468 if (track != previousTrack.get()) { 4469 // Flush any data still being written from last track 4470 mBytesRemaining = 0; 4471 if (mPausedBytesRemaining) { 4472 // Last track was paused so we also need to flush saved 4473 // mixbuffer state and invalidate track so that it will 4474 // re-submit that unwritten data when it is next resumed 4475 mPausedBytesRemaining = 0; 4476 // Invalidate is a bit drastic - would be more efficient 4477 // to have a flag to tell client that some of the 4478 // previously written data was lost 4479 previousTrack->invalidate(); 4480 } 4481 // flush data already sent to the DSP if changing audio session as audio 4482 // comes from a different source. Also invalidate previous track to force a 4483 // seek when resuming. 4484 if (previousTrack->sessionId() != track->sessionId()) { 4485 previousTrack->invalidate(); 4486 } 4487 } 4488 } 4489 mPreviousTrack = track; 4490 // reset retry count 4491 track->mRetryCount = kMaxTrackRetriesOffload; 4492 mActiveTrack = t; 4493 mixerStatus = MIXER_TRACKS_READY; 4494 } 4495 } else { 4496 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4497 if (track->isStopping_1()) { 4498 // Hardware buffer can hold a large amount of audio so we must 4499 // wait for all current track's data to drain before we say 4500 // that the track is stopped. 4501 if (mBytesRemaining == 0) { 4502 // Only start draining when all data in mixbuffer 4503 // has been written 4504 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4505 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4506 // do not drain if no data was ever sent to HAL (mStandby == true) 4507 if (last && !mStandby) { 4508 // do not modify drain sequence if we are already draining. This happens 4509 // when resuming from pause after drain. 4510 if ((mDrainSequence & 1) == 0) { 4511 sleepTime = 0; 4512 standbyTime = systemTime() + standbyDelay; 4513 mixerStatus = MIXER_DRAIN_TRACK; 4514 mDrainSequence += 2; 4515 } 4516 if (mHwPaused) { 4517 // It is possible to move from PAUSED to STOPPING_1 without 4518 // a resume so we must ensure hardware is running 4519 doHwResume = true; 4520 mHwPaused = false; 4521 } 4522 } 4523 } 4524 } else if (track->isStopping_2()) { 4525 // Drain has completed or we are in standby, signal presentation complete 4526 if (!(mDrainSequence & 1) || !last || mStandby) { 4527 track->mState = TrackBase::STOPPED; 4528 size_t audioHALFrames = 4529 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4530 size_t framesWritten = 4531 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4532 track->presentationComplete(framesWritten, audioHALFrames); 4533 track->reset(); 4534 tracksToRemove->add(track); 4535 } 4536 } else { 4537 // No buffers for this track. Give it a few chances to 4538 // fill a buffer, then remove it from active list. 4539 if (--(track->mRetryCount) <= 0) { 4540 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4541 track->name()); 4542 tracksToRemove->add(track); 4543 // indicate to client process that the track was disabled because of underrun; 4544 // it will then automatically call start() when data is available 4545 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4546 } else if (last){ 4547 mixerStatus = MIXER_TRACKS_ENABLED; 4548 } 4549 } 4550 } 4551 // compute volume for this track 4552 processVolume_l(track, last); 4553 } 4554 4555 // make sure the pause/flush/resume sequence is executed in the right order. 4556 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4557 // before flush and then resume HW. This can happen in case of pause/flush/resume 4558 // if resume is received before pause is executed. 4559 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4560 mOutput->stream->pause(mOutput->stream); 4561 } 4562 if (mFlushPending) { 4563 flushHw_l(); 4564 mFlushPending = false; 4565 } 4566 if (!mStandby && doHwResume) { 4567 mOutput->stream->resume(mOutput->stream); 4568 } 4569 4570 // remove all the tracks that need to be... 4571 removeTracks_l(*tracksToRemove); 4572 4573 return mixerStatus; 4574} 4575 4576// must be called with thread mutex locked 4577bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4578{ 4579 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4580 mWriteAckSequence, mDrainSequence); 4581 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4582 return true; 4583 } 4584 return false; 4585} 4586 4587// must be called with thread mutex locked 4588bool AudioFlinger::OffloadThread::shouldStandby_l() 4589{ 4590 bool trackPaused = false; 4591 4592 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4593 // after a timeout and we will enter standby then. 4594 if (mTracks.size() > 0) { 4595 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4596 } 4597 4598 return !mStandby && !trackPaused; 4599} 4600 4601 4602bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4603{ 4604 Mutex::Autolock _l(mLock); 4605 return waitingAsyncCallback_l(); 4606} 4607 4608void AudioFlinger::OffloadThread::flushHw_l() 4609{ 4610 DirectOutputThread::flushHw_l(); 4611 // Flush anything still waiting in the mixbuffer 4612 mCurrentWriteLength = 0; 4613 mBytesRemaining = 0; 4614 mPausedWriteLength = 0; 4615 mPausedBytesRemaining = 0; 4616 mHwPaused = false; 4617 4618 if (mUseAsyncWrite) { 4619 // discard any pending drain or write ack by incrementing sequence 4620 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4621 mDrainSequence = (mDrainSequence + 2) & ~1; 4622 ALOG_ASSERT(mCallbackThread != 0); 4623 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4624 mCallbackThread->setDraining(mDrainSequence); 4625 } 4626} 4627 4628void AudioFlinger::OffloadThread::onAddNewTrack_l() 4629{ 4630 sp<Track> previousTrack = mPreviousTrack.promote(); 4631 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4632 4633 if (previousTrack != 0 && latestTrack != 0 && 4634 (previousTrack->sessionId() != latestTrack->sessionId())) { 4635 mFlushPending = true; 4636 } 4637 PlaybackThread::onAddNewTrack_l(); 4638} 4639 4640// ---------------------------------------------------------------------------- 4641 4642AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4643 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4644 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4645 DUPLICATING), 4646 mWaitTimeMs(UINT_MAX) 4647{ 4648 addOutputTrack(mainThread); 4649} 4650 4651AudioFlinger::DuplicatingThread::~DuplicatingThread() 4652{ 4653 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4654 mOutputTracks[i]->destroy(); 4655 } 4656} 4657 4658void AudioFlinger::DuplicatingThread::threadLoop_mix() 4659{ 4660 // mix buffers... 4661 if (outputsReady(outputTracks)) { 4662 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4663 } else { 4664 if (mMixerBufferValid) { 4665 memset(mMixerBuffer, 0, mMixerBufferSize); 4666 } else { 4667 memset(mSinkBuffer, 0, mSinkBufferSize); 4668 } 4669 } 4670 sleepTime = 0; 4671 writeFrames = mNormalFrameCount; 4672 mCurrentWriteLength = mSinkBufferSize; 4673 standbyTime = systemTime() + standbyDelay; 4674} 4675 4676void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4677{ 4678 if (sleepTime == 0) { 4679 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4680 sleepTime = activeSleepTime; 4681 } else { 4682 sleepTime = idleSleepTime; 4683 } 4684 } else if (mBytesWritten != 0) { 4685 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4686 writeFrames = mNormalFrameCount; 4687 memset(mSinkBuffer, 0, mSinkBufferSize); 4688 } else { 4689 // flush remaining overflow buffers in output tracks 4690 writeFrames = 0; 4691 } 4692 sleepTime = 0; 4693 } 4694} 4695 4696ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4697{ 4698 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4699 // for delivery downstream as needed. This in-place conversion is safe as 4700 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4701 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4702 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4703 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4704 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4705 } 4706 for (size_t i = 0; i < outputTracks.size(); i++) { 4707 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4708 } 4709 mStandby = false; 4710 return (ssize_t)mSinkBufferSize; 4711} 4712 4713void AudioFlinger::DuplicatingThread::threadLoop_standby() 4714{ 4715 // DuplicatingThread implements standby by stopping all tracks 4716 for (size_t i = 0; i < outputTracks.size(); i++) { 4717 outputTracks[i]->stop(); 4718 } 4719} 4720 4721void AudioFlinger::DuplicatingThread::saveOutputTracks() 4722{ 4723 outputTracks = mOutputTracks; 4724} 4725 4726void AudioFlinger::DuplicatingThread::clearOutputTracks() 4727{ 4728 outputTracks.clear(); 4729} 4730 4731void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4732{ 4733 Mutex::Autolock _l(mLock); 4734 // FIXME explain this formula 4735 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4736 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4737 // due to current usage case and restrictions on the AudioBufferProvider. 4738 // Actual buffer conversion is done in threadLoop_write(). 4739 // 4740 // TODO: This may change in the future, depending on multichannel 4741 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4742 OutputTrack *outputTrack = new OutputTrack(thread, 4743 this, 4744 mSampleRate, 4745 AUDIO_FORMAT_PCM_16_BIT, 4746 mChannelMask, 4747 frameCount, 4748 IPCThreadState::self()->getCallingUid()); 4749 if (outputTrack->cblk() != NULL) { 4750 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 4751 mOutputTracks.add(outputTrack); 4752 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4753 updateWaitTime_l(); 4754 } 4755} 4756 4757void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4758{ 4759 Mutex::Autolock _l(mLock); 4760 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4761 if (mOutputTracks[i]->thread() == thread) { 4762 mOutputTracks[i]->destroy(); 4763 mOutputTracks.removeAt(i); 4764 updateWaitTime_l(); 4765 return; 4766 } 4767 } 4768 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4769} 4770 4771// caller must hold mLock 4772void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4773{ 4774 mWaitTimeMs = UINT_MAX; 4775 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4776 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4777 if (strong != 0) { 4778 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4779 if (waitTimeMs < mWaitTimeMs) { 4780 mWaitTimeMs = waitTimeMs; 4781 } 4782 } 4783 } 4784} 4785 4786 4787bool AudioFlinger::DuplicatingThread::outputsReady( 4788 const SortedVector< sp<OutputTrack> > &outputTracks) 4789{ 4790 for (size_t i = 0; i < outputTracks.size(); i++) { 4791 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4792 if (thread == 0) { 4793 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4794 outputTracks[i].get()); 4795 return false; 4796 } 4797 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4798 // see note at standby() declaration 4799 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4800 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4801 thread.get()); 4802 return false; 4803 } 4804 } 4805 return true; 4806} 4807 4808uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4809{ 4810 return (mWaitTimeMs * 1000) / 2; 4811} 4812 4813void AudioFlinger::DuplicatingThread::cacheParameters_l() 4814{ 4815 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4816 updateWaitTime_l(); 4817 4818 MixerThread::cacheParameters_l(); 4819} 4820 4821// ---------------------------------------------------------------------------- 4822// Record 4823// ---------------------------------------------------------------------------- 4824 4825AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4826 AudioStreamIn *input, 4827 audio_io_handle_t id, 4828 audio_devices_t outDevice, 4829 audio_devices_t inDevice 4830#ifdef TEE_SINK 4831 , const sp<NBAIO_Sink>& teeSink 4832#endif 4833 ) : 4834 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4835 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4836 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4837 mRsmpInRear(0) 4838#ifdef TEE_SINK 4839 , mTeeSink(teeSink) 4840#endif 4841 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4842 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4843 // mFastCapture below 4844 , mFastCaptureFutex(0) 4845 // mInputSource 4846 // mPipeSink 4847 // mPipeSource 4848 , mPipeFramesP2(0) 4849 // mPipeMemory 4850 // mFastCaptureNBLogWriter 4851 , mFastTrackAvail(false) 4852{ 4853 snprintf(mName, kNameLength, "AudioIn_%X", id); 4854 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4855 4856 readInputParameters_l(); 4857 4858 // create an NBAIO source for the HAL input stream, and negotiate 4859 mInputSource = new AudioStreamInSource(input->stream); 4860 size_t numCounterOffers = 0; 4861 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 4862 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 4863 ALOG_ASSERT(index == 0); 4864 4865 // initialize fast capture depending on configuration 4866 bool initFastCapture; 4867 switch (kUseFastCapture) { 4868 case FastCapture_Never: 4869 initFastCapture = false; 4870 break; 4871 case FastCapture_Always: 4872 initFastCapture = true; 4873 break; 4874 case FastCapture_Static: 4875 uint32_t primaryOutputSampleRate; 4876 { 4877 AutoMutex _l(audioFlinger->mHardwareLock); 4878 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 4879 } 4880 initFastCapture = 4881 // either capture sample rate is same as (a reasonable) primary output sample rate 4882 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 4883 (mSampleRate == primaryOutputSampleRate)) || 4884 // or primary output sample rate is unknown, and capture sample rate is reasonable 4885 ((primaryOutputSampleRate == 0) && 4886 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 4887 // and the buffer size is < 12 ms 4888 (mFrameCount * 1000) / mSampleRate < 12; 4889 break; 4890 // case FastCapture_Dynamic: 4891 } 4892 4893 if (initFastCapture) { 4894 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 4895 NBAIO_Format format = mInputSource->format(); 4896 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 4897 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 4898 void *pipeBuffer; 4899 const sp<MemoryDealer> roHeap(readOnlyHeap()); 4900 sp<IMemory> pipeMemory; 4901 if ((roHeap == 0) || 4902 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 4903 (pipeBuffer = pipeMemory->pointer()) == NULL) { 4904 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 4905 goto failed; 4906 } 4907 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 4908 memset(pipeBuffer, 0, pipeSize); 4909 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 4910 const NBAIO_Format offers[1] = {format}; 4911 size_t numCounterOffers = 0; 4912 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 4913 ALOG_ASSERT(index == 0); 4914 mPipeSink = pipe; 4915 PipeReader *pipeReader = new PipeReader(*pipe); 4916 numCounterOffers = 0; 4917 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 4918 ALOG_ASSERT(index == 0); 4919 mPipeSource = pipeReader; 4920 mPipeFramesP2 = pipeFramesP2; 4921 mPipeMemory = pipeMemory; 4922 4923 // create fast capture 4924 mFastCapture = new FastCapture(); 4925 FastCaptureStateQueue *sq = mFastCapture->sq(); 4926#ifdef STATE_QUEUE_DUMP 4927 // FIXME 4928#endif 4929 FastCaptureState *state = sq->begin(); 4930 state->mCblk = NULL; 4931 state->mInputSource = mInputSource.get(); 4932 state->mInputSourceGen++; 4933 state->mPipeSink = pipe; 4934 state->mPipeSinkGen++; 4935 state->mFrameCount = mFrameCount; 4936 state->mCommand = FastCaptureState::COLD_IDLE; 4937 // already done in constructor initialization list 4938 //mFastCaptureFutex = 0; 4939 state->mColdFutexAddr = &mFastCaptureFutex; 4940 state->mColdGen++; 4941 state->mDumpState = &mFastCaptureDumpState; 4942#ifdef TEE_SINK 4943 // FIXME 4944#endif 4945 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 4946 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 4947 sq->end(); 4948 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4949 4950 // start the fast capture 4951 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 4952 pid_t tid = mFastCapture->getTid(); 4953 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 4954 if (err != 0) { 4955 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 4956 kPriorityFastCapture, getpid_cached, tid, err); 4957 } 4958 4959#ifdef AUDIO_WATCHDOG 4960 // FIXME 4961#endif 4962 4963 mFastTrackAvail = true; 4964 } 4965failed: ; 4966 4967 // FIXME mNormalSource 4968} 4969 4970 4971AudioFlinger::RecordThread::~RecordThread() 4972{ 4973 if (mFastCapture != 0) { 4974 FastCaptureStateQueue *sq = mFastCapture->sq(); 4975 FastCaptureState *state = sq->begin(); 4976 if (state->mCommand == FastCaptureState::COLD_IDLE) { 4977 int32_t old = android_atomic_inc(&mFastCaptureFutex); 4978 if (old == -1) { 4979 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 4980 } 4981 } 4982 state->mCommand = FastCaptureState::EXIT; 4983 sq->end(); 4984 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 4985 mFastCapture->join(); 4986 mFastCapture.clear(); 4987 } 4988 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 4989 mAudioFlinger->unregisterWriter(mNBLogWriter); 4990 delete[] mRsmpInBuffer; 4991} 4992 4993void AudioFlinger::RecordThread::onFirstRef() 4994{ 4995 run(mName, PRIORITY_URGENT_AUDIO); 4996} 4997 4998bool AudioFlinger::RecordThread::threadLoop() 4999{ 5000 nsecs_t lastWarning = 0; 5001 5002 inputStandBy(); 5003 5004reacquire_wakelock: 5005 sp<RecordTrack> activeTrack; 5006 int activeTracksGen; 5007 { 5008 Mutex::Autolock _l(mLock); 5009 size_t size = mActiveTracks.size(); 5010 activeTracksGen = mActiveTracksGen; 5011 if (size > 0) { 5012 // FIXME an arbitrary choice 5013 activeTrack = mActiveTracks[0]; 5014 acquireWakeLock_l(activeTrack->uid()); 5015 if (size > 1) { 5016 SortedVector<int> tmp; 5017 for (size_t i = 0; i < size; i++) { 5018 tmp.add(mActiveTracks[i]->uid()); 5019 } 5020 updateWakeLockUids_l(tmp); 5021 } 5022 } else { 5023 acquireWakeLock_l(-1); 5024 } 5025 } 5026 5027 // used to request a deferred sleep, to be executed later while mutex is unlocked 5028 uint32_t sleepUs = 0; 5029 5030 // loop while there is work to do 5031 for (;;) { 5032 Vector< sp<EffectChain> > effectChains; 5033 5034 // sleep with mutex unlocked 5035 if (sleepUs > 0) { 5036 usleep(sleepUs); 5037 sleepUs = 0; 5038 } 5039 5040 // activeTracks accumulates a copy of a subset of mActiveTracks 5041 Vector< sp<RecordTrack> > activeTracks; 5042 5043 // reference to the (first and only) active fast track 5044 sp<RecordTrack> fastTrack; 5045 5046 // reference to a fast track which is about to be removed 5047 sp<RecordTrack> fastTrackToRemove; 5048 5049 { // scope for mLock 5050 Mutex::Autolock _l(mLock); 5051 5052 processConfigEvents_l(); 5053 5054 // check exitPending here because checkForNewParameters_l() and 5055 // checkForNewParameters_l() can temporarily release mLock 5056 if (exitPending()) { 5057 break; 5058 } 5059 5060 // if no active track(s), then standby and release wakelock 5061 size_t size = mActiveTracks.size(); 5062 if (size == 0) { 5063 standbyIfNotAlreadyInStandby(); 5064 // exitPending() can't become true here 5065 releaseWakeLock_l(); 5066 ALOGV("RecordThread: loop stopping"); 5067 // go to sleep 5068 mWaitWorkCV.wait(mLock); 5069 ALOGV("RecordThread: loop starting"); 5070 goto reacquire_wakelock; 5071 } 5072 5073 if (mActiveTracksGen != activeTracksGen) { 5074 activeTracksGen = mActiveTracksGen; 5075 SortedVector<int> tmp; 5076 for (size_t i = 0; i < size; i++) { 5077 tmp.add(mActiveTracks[i]->uid()); 5078 } 5079 updateWakeLockUids_l(tmp); 5080 } 5081 5082 bool doBroadcast = false; 5083 for (size_t i = 0; i < size; ) { 5084 5085 activeTrack = mActiveTracks[i]; 5086 if (activeTrack->isTerminated()) { 5087 if (activeTrack->isFastTrack()) { 5088 ALOG_ASSERT(fastTrackToRemove == 0); 5089 fastTrackToRemove = activeTrack; 5090 } 5091 removeTrack_l(activeTrack); 5092 mActiveTracks.remove(activeTrack); 5093 mActiveTracksGen++; 5094 size--; 5095 continue; 5096 } 5097 5098 TrackBase::track_state activeTrackState = activeTrack->mState; 5099 switch (activeTrackState) { 5100 5101 case TrackBase::PAUSING: 5102 mActiveTracks.remove(activeTrack); 5103 mActiveTracksGen++; 5104 doBroadcast = true; 5105 size--; 5106 continue; 5107 5108 case TrackBase::STARTING_1: 5109 sleepUs = 10000; 5110 i++; 5111 continue; 5112 5113 case TrackBase::STARTING_2: 5114 doBroadcast = true; 5115 mStandby = false; 5116 activeTrack->mState = TrackBase::ACTIVE; 5117 break; 5118 5119 case TrackBase::ACTIVE: 5120 break; 5121 5122 case TrackBase::IDLE: 5123 i++; 5124 continue; 5125 5126 default: 5127 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5128 } 5129 5130 activeTracks.add(activeTrack); 5131 i++; 5132 5133 if (activeTrack->isFastTrack()) { 5134 ALOG_ASSERT(!mFastTrackAvail); 5135 ALOG_ASSERT(fastTrack == 0); 5136 fastTrack = activeTrack; 5137 } 5138 } 5139 if (doBroadcast) { 5140 mStartStopCond.broadcast(); 5141 } 5142 5143 // sleep if there are no active tracks to process 5144 if (activeTracks.size() == 0) { 5145 if (sleepUs == 0) { 5146 sleepUs = kRecordThreadSleepUs; 5147 } 5148 continue; 5149 } 5150 sleepUs = 0; 5151 5152 lockEffectChains_l(effectChains); 5153 } 5154 5155 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5156 5157 size_t size = effectChains.size(); 5158 for (size_t i = 0; i < size; i++) { 5159 // thread mutex is not locked, but effect chain is locked 5160 effectChains[i]->process_l(); 5161 } 5162 5163 // Push a new fast capture state if fast capture is not already running, or cblk change 5164 if (mFastCapture != 0) { 5165 FastCaptureStateQueue *sq = mFastCapture->sq(); 5166 FastCaptureState *state = sq->begin(); 5167 bool didModify = false; 5168 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5169 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5170 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5171 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5172 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5173 if (old == -1) { 5174 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5175 } 5176 } 5177 state->mCommand = FastCaptureState::READ_WRITE; 5178#if 0 // FIXME 5179 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5180 FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 5181#endif 5182 didModify = true; 5183 } 5184 audio_track_cblk_t *cblkOld = state->mCblk; 5185 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5186 if (cblkNew != cblkOld) { 5187 state->mCblk = cblkNew; 5188 // block until acked if removing a fast track 5189 if (cblkOld != NULL) { 5190 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5191 } 5192 didModify = true; 5193 } 5194 sq->end(didModify); 5195 if (didModify) { 5196 sq->push(block); 5197#if 0 5198 if (kUseFastCapture == FastCapture_Dynamic) { 5199 mNormalSource = mPipeSource; 5200 } 5201#endif 5202 } 5203 } 5204 5205 // now run the fast track destructor with thread mutex unlocked 5206 fastTrackToRemove.clear(); 5207 5208 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5209 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5210 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5211 // If destination is non-contiguous, first read past the nominal end of buffer, then 5212 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5213 5214 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5215 ssize_t framesRead; 5216 5217 // If an NBAIO source is present, use it to read the normal capture's data 5218 if (mPipeSource != 0) { 5219 size_t framesToRead = mBufferSize / mFrameSize; 5220 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5221 framesToRead, AudioBufferProvider::kInvalidPTS); 5222 if (framesRead == 0) { 5223 // since pipe is non-blocking, simulate blocking input 5224 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5225 } 5226 // otherwise use the HAL / AudioStreamIn directly 5227 } else { 5228 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5229 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5230 if (bytesRead < 0) { 5231 framesRead = bytesRead; 5232 } else { 5233 framesRead = bytesRead / mFrameSize; 5234 } 5235 } 5236 5237 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5238 ALOGE("read failed: framesRead=%d", framesRead); 5239 // Force input into standby so that it tries to recover at next read attempt 5240 inputStandBy(); 5241 sleepUs = kRecordThreadSleepUs; 5242 } 5243 if (framesRead <= 0) { 5244 goto unlock; 5245 } 5246 ALOG_ASSERT(framesRead > 0); 5247 5248 if (mTeeSink != 0) { 5249 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5250 } 5251 // If destination is non-contiguous, we now correct for reading past end of buffer. 5252 { 5253 size_t part1 = mRsmpInFramesP2 - rear; 5254 if ((size_t) framesRead > part1) { 5255 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5256 (framesRead - part1) * mFrameSize); 5257 } 5258 } 5259 rear = mRsmpInRear += framesRead; 5260 5261 size = activeTracks.size(); 5262 // loop over each active track 5263 for (size_t i = 0; i < size; i++) { 5264 activeTrack = activeTracks[i]; 5265 5266 // skip fast tracks, as those are handled directly by FastCapture 5267 if (activeTrack->isFastTrack()) { 5268 continue; 5269 } 5270 5271 enum { 5272 OVERRUN_UNKNOWN, 5273 OVERRUN_TRUE, 5274 OVERRUN_FALSE 5275 } overrun = OVERRUN_UNKNOWN; 5276 5277 // loop over getNextBuffer to handle circular sink 5278 for (;;) { 5279 5280 activeTrack->mSink.frameCount = ~0; 5281 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5282 size_t framesOut = activeTrack->mSink.frameCount; 5283 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5284 5285 int32_t front = activeTrack->mRsmpInFront; 5286 ssize_t filled = rear - front; 5287 size_t framesIn; 5288 5289 if (filled < 0) { 5290 // should not happen, but treat like a massive overrun and re-sync 5291 framesIn = 0; 5292 activeTrack->mRsmpInFront = rear; 5293 overrun = OVERRUN_TRUE; 5294 } else if ((size_t) filled <= mRsmpInFrames) { 5295 framesIn = (size_t) filled; 5296 } else { 5297 // client is not keeping up with server, but give it latest data 5298 framesIn = mRsmpInFrames; 5299 activeTrack->mRsmpInFront = front = rear - framesIn; 5300 overrun = OVERRUN_TRUE; 5301 } 5302 5303 if (framesOut == 0 || framesIn == 0) { 5304 break; 5305 } 5306 5307 if (activeTrack->mResampler == NULL) { 5308 // no resampling 5309 if (framesIn > framesOut) { 5310 framesIn = framesOut; 5311 } else { 5312 framesOut = framesIn; 5313 } 5314 int8_t *dst = activeTrack->mSink.i8; 5315 while (framesIn > 0) { 5316 front &= mRsmpInFramesP2 - 1; 5317 size_t part1 = mRsmpInFramesP2 - front; 5318 if (part1 > framesIn) { 5319 part1 = framesIn; 5320 } 5321 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5322 if (mChannelCount == activeTrack->mChannelCount) { 5323 memcpy(dst, src, part1 * mFrameSize); 5324 } else if (mChannelCount == 1) { 5325 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5326 part1); 5327 } else { 5328 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src, 5329 part1); 5330 } 5331 dst += part1 * activeTrack->mFrameSize; 5332 front += part1; 5333 framesIn -= part1; 5334 } 5335 activeTrack->mRsmpInFront += framesOut; 5336 5337 } else { 5338 // resampling 5339 // FIXME framesInNeeded should really be part of resampler API, and should 5340 // depend on the SRC ratio 5341 // to keep mRsmpInBuffer full so resampler always has sufficient input 5342 size_t framesInNeeded; 5343 // FIXME only re-calculate when it changes, and optimize for common ratios 5344 // Do not precompute in/out because floating point is not associative 5345 // e.g. a*b/c != a*(b/c). 5346 const double in(mSampleRate); 5347 const double out(activeTrack->mSampleRate); 5348 framesInNeeded = ceil(framesOut * in / out) + 1; 5349 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5350 framesInNeeded, framesOut, in / out); 5351 // Although we theoretically have framesIn in circular buffer, some of those are 5352 // unreleased frames, and thus must be discounted for purpose of budgeting. 5353 size_t unreleased = activeTrack->mRsmpInUnrel; 5354 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5355 if (framesIn < framesInNeeded) { 5356 ALOGV("not enough to resample: have %u frames in but need %u in to " 5357 "produce %u out given in/out ratio of %.4g", 5358 framesIn, framesInNeeded, framesOut, in / out); 5359 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5360 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5361 if (newFramesOut == 0) { 5362 break; 5363 } 5364 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5365 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5366 framesInNeeded, newFramesOut, out / in); 5367 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5368 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5369 "given in/out ratio of %.4g", 5370 framesIn, framesInNeeded, newFramesOut, in / out); 5371 framesOut = newFramesOut; 5372 } else { 5373 ALOGV("success 1: have %u in and need %u in to produce %u out " 5374 "given in/out ratio of %.4g", 5375 framesIn, framesInNeeded, framesOut, in / out); 5376 } 5377 5378 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5379 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5380 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5381 delete[] activeTrack->mRsmpOutBuffer; 5382 // resampler always outputs stereo 5383 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5384 activeTrack->mRsmpOutFrameCount = framesOut; 5385 } 5386 5387 // resampler accumulates, but we only have one source track 5388 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5389 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5390 // FIXME how about having activeTrack implement this interface itself? 5391 activeTrack->mResamplerBufferProvider 5392 /*this*/ /* AudioBufferProvider* */); 5393 // ditherAndClamp() works as long as all buffers returned by 5394 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5395 if (activeTrack->mChannelCount == 1) { 5396 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5397 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5398 framesOut); 5399 // the resampler always outputs stereo samples: 5400 // do post stereo to mono conversion 5401 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5402 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5403 } else { 5404 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5405 activeTrack->mRsmpOutBuffer, framesOut); 5406 } 5407 // now done with mRsmpOutBuffer 5408 5409 } 5410 5411 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5412 overrun = OVERRUN_FALSE; 5413 } 5414 5415 if (activeTrack->mFramesToDrop == 0) { 5416 if (framesOut > 0) { 5417 activeTrack->mSink.frameCount = framesOut; 5418 activeTrack->releaseBuffer(&activeTrack->mSink); 5419 } 5420 } else { 5421 // FIXME could do a partial drop of framesOut 5422 if (activeTrack->mFramesToDrop > 0) { 5423 activeTrack->mFramesToDrop -= framesOut; 5424 if (activeTrack->mFramesToDrop <= 0) { 5425 activeTrack->clearSyncStartEvent(); 5426 } 5427 } else { 5428 activeTrack->mFramesToDrop += framesOut; 5429 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5430 activeTrack->mSyncStartEvent->isCancelled()) { 5431 ALOGW("Synced record %s, session %d, trigger session %d", 5432 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5433 activeTrack->sessionId(), 5434 (activeTrack->mSyncStartEvent != 0) ? 5435 activeTrack->mSyncStartEvent->triggerSession() : 0); 5436 activeTrack->clearSyncStartEvent(); 5437 } 5438 } 5439 } 5440 5441 if (framesOut == 0) { 5442 break; 5443 } 5444 } 5445 5446 switch (overrun) { 5447 case OVERRUN_TRUE: 5448 // client isn't retrieving buffers fast enough 5449 if (!activeTrack->setOverflow()) { 5450 nsecs_t now = systemTime(); 5451 // FIXME should lastWarning per track? 5452 if ((now - lastWarning) > kWarningThrottleNs) { 5453 ALOGW("RecordThread: buffer overflow"); 5454 lastWarning = now; 5455 } 5456 } 5457 break; 5458 case OVERRUN_FALSE: 5459 activeTrack->clearOverflow(); 5460 break; 5461 case OVERRUN_UNKNOWN: 5462 break; 5463 } 5464 5465 } 5466 5467unlock: 5468 // enable changes in effect chain 5469 unlockEffectChains(effectChains); 5470 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5471 } 5472 5473 standbyIfNotAlreadyInStandby(); 5474 5475 { 5476 Mutex::Autolock _l(mLock); 5477 for (size_t i = 0; i < mTracks.size(); i++) { 5478 sp<RecordTrack> track = mTracks[i]; 5479 track->invalidate(); 5480 } 5481 mActiveTracks.clear(); 5482 mActiveTracksGen++; 5483 mStartStopCond.broadcast(); 5484 } 5485 5486 releaseWakeLock(); 5487 5488 ALOGV("RecordThread %p exiting", this); 5489 return false; 5490} 5491 5492void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5493{ 5494 if (!mStandby) { 5495 inputStandBy(); 5496 mStandby = true; 5497 } 5498} 5499 5500void AudioFlinger::RecordThread::inputStandBy() 5501{ 5502 // Idle the fast capture if it's currently running 5503 if (mFastCapture != 0) { 5504 FastCaptureStateQueue *sq = mFastCapture->sq(); 5505 FastCaptureState *state = sq->begin(); 5506 if (!(state->mCommand & FastCaptureState::IDLE)) { 5507 state->mCommand = FastCaptureState::COLD_IDLE; 5508 state->mColdFutexAddr = &mFastCaptureFutex; 5509 state->mColdGen++; 5510 mFastCaptureFutex = 0; 5511 sq->end(); 5512 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5513 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5514#if 0 5515 if (kUseFastCapture == FastCapture_Dynamic) { 5516 // FIXME 5517 } 5518#endif 5519#ifdef AUDIO_WATCHDOG 5520 // FIXME 5521#endif 5522 } else { 5523 sq->end(false /*didModify*/); 5524 } 5525 } 5526 mInput->stream->common.standby(&mInput->stream->common); 5527} 5528 5529// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5530sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5531 const sp<AudioFlinger::Client>& client, 5532 uint32_t sampleRate, 5533 audio_format_t format, 5534 audio_channel_mask_t channelMask, 5535 size_t *pFrameCount, 5536 int sessionId, 5537 size_t *notificationFrames, 5538 int uid, 5539 IAudioFlinger::track_flags_t *flags, 5540 pid_t tid, 5541 status_t *status) 5542{ 5543 size_t frameCount = *pFrameCount; 5544 sp<RecordTrack> track; 5545 status_t lStatus; 5546 5547 // client expresses a preference for FAST, but we get the final say 5548 if (*flags & IAudioFlinger::TRACK_FAST) { 5549 if ( 5550 // use case: callback handler 5551 (tid != -1) && 5552 // frame count is not specified, or is exactly the pipe depth 5553 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5554 // PCM data 5555 audio_is_linear_pcm(format) && 5556 // native format 5557 (format == mFormat) && 5558 // native channel mask 5559 (channelMask == mChannelMask) && 5560 // native hardware sample rate 5561 (sampleRate == mSampleRate) && 5562 // record thread has an associated fast capture 5563 hasFastCapture() && 5564 // there are sufficient fast track slots available 5565 mFastTrackAvail 5566 ) { 5567 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5568 frameCount, mFrameCount); 5569 } else { 5570 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5571 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5572 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5573 frameCount, mFrameCount, mPipeFramesP2, 5574 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5575 hasFastCapture(), tid, mFastTrackAvail); 5576 *flags &= ~IAudioFlinger::TRACK_FAST; 5577 } 5578 } 5579 5580 // compute track buffer size in frames, and suggest the notification frame count 5581 if (*flags & IAudioFlinger::TRACK_FAST) { 5582 // fast track: frame count is exactly the pipe depth 5583 frameCount = mPipeFramesP2; 5584 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5585 *notificationFrames = mFrameCount; 5586 } else { 5587 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5588 // or 20 ms if there is a fast capture 5589 // TODO This could be a roundupRatio inline, and const 5590 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5591 * sampleRate + mSampleRate - 1) / mSampleRate; 5592 // minimum number of notification periods is at least kMinNotifications, 5593 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5594 static const size_t kMinNotifications = 3; 5595 static const uint32_t kMinMs = 30; 5596 // TODO This could be a roundupRatio inline 5597 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5598 // TODO This could be a roundupRatio inline 5599 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5600 maxNotificationFrames; 5601 const size_t minFrameCount = maxNotificationFrames * 5602 max(kMinNotifications, minNotificationsByMs); 5603 frameCount = max(frameCount, minFrameCount); 5604 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5605 *notificationFrames = maxNotificationFrames; 5606 } 5607 } 5608 *pFrameCount = frameCount; 5609 5610 lStatus = initCheck(); 5611 if (lStatus != NO_ERROR) { 5612 ALOGE("createRecordTrack_l() audio driver not initialized"); 5613 goto Exit; 5614 } 5615 5616 { // scope for mLock 5617 Mutex::Autolock _l(mLock); 5618 5619 track = new RecordTrack(this, client, sampleRate, 5620 format, channelMask, frameCount, NULL, sessionId, uid, 5621 *flags, TrackBase::TYPE_DEFAULT); 5622 5623 lStatus = track->initCheck(); 5624 if (lStatus != NO_ERROR) { 5625 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5626 // track must be cleared from the caller as the caller has the AF lock 5627 goto Exit; 5628 } 5629 mTracks.add(track); 5630 5631 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5632 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5633 mAudioFlinger->btNrecIsOff(); 5634 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5635 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5636 5637 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5638 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5639 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5640 // so ask activity manager to do this on our behalf 5641 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5642 } 5643 } 5644 5645 lStatus = NO_ERROR; 5646 5647Exit: 5648 *status = lStatus; 5649 return track; 5650} 5651 5652status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5653 AudioSystem::sync_event_t event, 5654 int triggerSession) 5655{ 5656 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5657 sp<ThreadBase> strongMe = this; 5658 status_t status = NO_ERROR; 5659 5660 if (event == AudioSystem::SYNC_EVENT_NONE) { 5661 recordTrack->clearSyncStartEvent(); 5662 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5663 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5664 triggerSession, 5665 recordTrack->sessionId(), 5666 syncStartEventCallback, 5667 recordTrack); 5668 // Sync event can be cancelled by the trigger session if the track is not in a 5669 // compatible state in which case we start record immediately 5670 if (recordTrack->mSyncStartEvent->isCancelled()) { 5671 recordTrack->clearSyncStartEvent(); 5672 } else { 5673 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5674 recordTrack->mFramesToDrop = - 5675 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5676 } 5677 } 5678 5679 { 5680 // This section is a rendezvous between binder thread executing start() and RecordThread 5681 AutoMutex lock(mLock); 5682 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5683 if (recordTrack->mState == TrackBase::PAUSING) { 5684 ALOGV("active record track PAUSING -> ACTIVE"); 5685 recordTrack->mState = TrackBase::ACTIVE; 5686 } else { 5687 ALOGV("active record track state %d", recordTrack->mState); 5688 } 5689 return status; 5690 } 5691 5692 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5693 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5694 // or using a separate command thread 5695 recordTrack->mState = TrackBase::STARTING_1; 5696 mActiveTracks.add(recordTrack); 5697 mActiveTracksGen++; 5698 status_t status = NO_ERROR; 5699 if (recordTrack->isExternalTrack()) { 5700 mLock.unlock(); 5701 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5702 mLock.lock(); 5703 // FIXME should verify that recordTrack is still in mActiveTracks 5704 if (status != NO_ERROR) { 5705 mActiveTracks.remove(recordTrack); 5706 mActiveTracksGen++; 5707 recordTrack->clearSyncStartEvent(); 5708 ALOGV("RecordThread::start error %d", status); 5709 return status; 5710 } 5711 } 5712 // Catch up with current buffer indices if thread is already running. 5713 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5714 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5715 // see previously buffered data before it called start(), but with greater risk of overrun. 5716 5717 recordTrack->mRsmpInFront = mRsmpInRear; 5718 recordTrack->mRsmpInUnrel = 0; 5719 // FIXME why reset? 5720 if (recordTrack->mResampler != NULL) { 5721 recordTrack->mResampler->reset(); 5722 } 5723 recordTrack->mState = TrackBase::STARTING_2; 5724 // signal thread to start 5725 mWaitWorkCV.broadcast(); 5726 if (mActiveTracks.indexOf(recordTrack) < 0) { 5727 ALOGV("Record failed to start"); 5728 status = BAD_VALUE; 5729 goto startError; 5730 } 5731 return status; 5732 } 5733 5734startError: 5735 if (recordTrack->isExternalTrack()) { 5736 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5737 } 5738 recordTrack->clearSyncStartEvent(); 5739 // FIXME I wonder why we do not reset the state here? 5740 return status; 5741} 5742 5743void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5744{ 5745 sp<SyncEvent> strongEvent = event.promote(); 5746 5747 if (strongEvent != 0) { 5748 sp<RefBase> ptr = strongEvent->cookie().promote(); 5749 if (ptr != 0) { 5750 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5751 recordTrack->handleSyncStartEvent(strongEvent); 5752 } 5753 } 5754} 5755 5756bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5757 ALOGV("RecordThread::stop"); 5758 AutoMutex _l(mLock); 5759 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5760 return false; 5761 } 5762 // note that threadLoop may still be processing the track at this point [without lock] 5763 recordTrack->mState = TrackBase::PAUSING; 5764 // do not wait for mStartStopCond if exiting 5765 if (exitPending()) { 5766 return true; 5767 } 5768 // FIXME incorrect usage of wait: no explicit predicate or loop 5769 mStartStopCond.wait(mLock); 5770 // if we have been restarted, recordTrack is in mActiveTracks here 5771 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5772 ALOGV("Record stopped OK"); 5773 return true; 5774 } 5775 return false; 5776} 5777 5778bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5779{ 5780 return false; 5781} 5782 5783status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5784{ 5785#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5786 if (!isValidSyncEvent(event)) { 5787 return BAD_VALUE; 5788 } 5789 5790 int eventSession = event->triggerSession(); 5791 status_t ret = NAME_NOT_FOUND; 5792 5793 Mutex::Autolock _l(mLock); 5794 5795 for (size_t i = 0; i < mTracks.size(); i++) { 5796 sp<RecordTrack> track = mTracks[i]; 5797 if (eventSession == track->sessionId()) { 5798 (void) track->setSyncEvent(event); 5799 ret = NO_ERROR; 5800 } 5801 } 5802 return ret; 5803#else 5804 return BAD_VALUE; 5805#endif 5806} 5807 5808// destroyTrack_l() must be called with ThreadBase::mLock held 5809void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5810{ 5811 track->terminate(); 5812 track->mState = TrackBase::STOPPED; 5813 // active tracks are removed by threadLoop() 5814 if (mActiveTracks.indexOf(track) < 0) { 5815 removeTrack_l(track); 5816 } 5817} 5818 5819void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5820{ 5821 mTracks.remove(track); 5822 // need anything related to effects here? 5823 if (track->isFastTrack()) { 5824 ALOG_ASSERT(!mFastTrackAvail); 5825 mFastTrackAvail = true; 5826 } 5827} 5828 5829void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5830{ 5831 dumpInternals(fd, args); 5832 dumpTracks(fd, args); 5833 dumpEffectChains(fd, args); 5834} 5835 5836void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5837{ 5838 dprintf(fd, "\nInput thread %p:\n", this); 5839 5840 if (mActiveTracks.size() > 0) { 5841 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5842 } else { 5843 dprintf(fd, " No active record clients\n"); 5844 } 5845 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 5846 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 5847 5848 dumpBase(fd, args); 5849} 5850 5851void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5852{ 5853 const size_t SIZE = 256; 5854 char buffer[SIZE]; 5855 String8 result; 5856 5857 size_t numtracks = mTracks.size(); 5858 size_t numactive = mActiveTracks.size(); 5859 size_t numactiveseen = 0; 5860 dprintf(fd, " %d Tracks", numtracks); 5861 if (numtracks) { 5862 dprintf(fd, " of which %d are active\n", numactive); 5863 RecordTrack::appendDumpHeader(result); 5864 for (size_t i = 0; i < numtracks ; ++i) { 5865 sp<RecordTrack> track = mTracks[i]; 5866 if (track != 0) { 5867 bool active = mActiveTracks.indexOf(track) >= 0; 5868 if (active) { 5869 numactiveseen++; 5870 } 5871 track->dump(buffer, SIZE, active); 5872 result.append(buffer); 5873 } 5874 } 5875 } else { 5876 dprintf(fd, "\n"); 5877 } 5878 5879 if (numactiveseen != numactive) { 5880 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5881 " not in the track list\n"); 5882 result.append(buffer); 5883 RecordTrack::appendDumpHeader(result); 5884 for (size_t i = 0; i < numactive; ++i) { 5885 sp<RecordTrack> track = mActiveTracks[i]; 5886 if (mTracks.indexOf(track) < 0) { 5887 track->dump(buffer, SIZE, true); 5888 result.append(buffer); 5889 } 5890 } 5891 5892 } 5893 write(fd, result.string(), result.size()); 5894} 5895 5896// AudioBufferProvider interface 5897status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5898 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5899{ 5900 RecordTrack *activeTrack = mRecordTrack; 5901 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5902 if (threadBase == 0) { 5903 buffer->frameCount = 0; 5904 buffer->raw = NULL; 5905 return NOT_ENOUGH_DATA; 5906 } 5907 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5908 int32_t rear = recordThread->mRsmpInRear; 5909 int32_t front = activeTrack->mRsmpInFront; 5910 ssize_t filled = rear - front; 5911 // FIXME should not be P2 (don't want to increase latency) 5912 // FIXME if client not keeping up, discard 5913 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5914 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5915 front &= recordThread->mRsmpInFramesP2 - 1; 5916 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5917 if (part1 > (size_t) filled) { 5918 part1 = filled; 5919 } 5920 size_t ask = buffer->frameCount; 5921 ALOG_ASSERT(ask > 0); 5922 if (part1 > ask) { 5923 part1 = ask; 5924 } 5925 if (part1 == 0) { 5926 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5927 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5928 buffer->raw = NULL; 5929 buffer->frameCount = 0; 5930 activeTrack->mRsmpInUnrel = 0; 5931 return NOT_ENOUGH_DATA; 5932 } 5933 5934 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5935 buffer->frameCount = part1; 5936 activeTrack->mRsmpInUnrel = part1; 5937 return NO_ERROR; 5938} 5939 5940// AudioBufferProvider interface 5941void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5942 AudioBufferProvider::Buffer* buffer) 5943{ 5944 RecordTrack *activeTrack = mRecordTrack; 5945 size_t stepCount = buffer->frameCount; 5946 if (stepCount == 0) { 5947 return; 5948 } 5949 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5950 activeTrack->mRsmpInUnrel -= stepCount; 5951 activeTrack->mRsmpInFront += stepCount; 5952 buffer->raw = NULL; 5953 buffer->frameCount = 0; 5954} 5955 5956bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5957 status_t& status) 5958{ 5959 bool reconfig = false; 5960 5961 status = NO_ERROR; 5962 5963 audio_format_t reqFormat = mFormat; 5964 uint32_t samplingRate = mSampleRate; 5965 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5966 5967 AudioParameter param = AudioParameter(keyValuePair); 5968 int value; 5969 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5970 // channel count change can be requested. Do we mandate the first client defines the 5971 // HAL sampling rate and channel count or do we allow changes on the fly? 5972 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5973 samplingRate = value; 5974 reconfig = true; 5975 } 5976 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5977 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5978 status = BAD_VALUE; 5979 } else { 5980 reqFormat = (audio_format_t) value; 5981 reconfig = true; 5982 } 5983 } 5984 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5985 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5986 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5987 status = BAD_VALUE; 5988 } else { 5989 channelMask = mask; 5990 reconfig = true; 5991 } 5992 } 5993 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5994 // do not accept frame count changes if tracks are open as the track buffer 5995 // size depends on frame count and correct behavior would not be guaranteed 5996 // if frame count is changed after track creation 5997 if (mActiveTracks.size() > 0) { 5998 status = INVALID_OPERATION; 5999 } else { 6000 reconfig = true; 6001 } 6002 } 6003 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6004 // forward device change to effects that have requested to be 6005 // aware of attached audio device. 6006 for (size_t i = 0; i < mEffectChains.size(); i++) { 6007 mEffectChains[i]->setDevice_l(value); 6008 } 6009 6010 // store input device and output device but do not forward output device to audio HAL. 6011 // Note that status is ignored by the caller for output device 6012 // (see AudioFlinger::setParameters() 6013 if (audio_is_output_devices(value)) { 6014 mOutDevice = value; 6015 status = BAD_VALUE; 6016 } else { 6017 mInDevice = value; 6018 // disable AEC and NS if the device is a BT SCO headset supporting those 6019 // pre processings 6020 if (mTracks.size() > 0) { 6021 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6022 mAudioFlinger->btNrecIsOff(); 6023 for (size_t i = 0; i < mTracks.size(); i++) { 6024 sp<RecordTrack> track = mTracks[i]; 6025 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6026 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6027 } 6028 } 6029 } 6030 } 6031 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6032 mAudioSource != (audio_source_t)value) { 6033 // forward device change to effects that have requested to be 6034 // aware of attached audio device. 6035 for (size_t i = 0; i < mEffectChains.size(); i++) { 6036 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6037 } 6038 mAudioSource = (audio_source_t)value; 6039 } 6040 6041 if (status == NO_ERROR) { 6042 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6043 keyValuePair.string()); 6044 if (status == INVALID_OPERATION) { 6045 inputStandBy(); 6046 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6047 keyValuePair.string()); 6048 } 6049 if (reconfig) { 6050 if (status == BAD_VALUE && 6051 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6052 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6053 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6054 <= (2 * samplingRate)) && 6055 audio_channel_count_from_in_mask( 6056 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6057 (channelMask == AUDIO_CHANNEL_IN_MONO || 6058 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6059 status = NO_ERROR; 6060 } 6061 if (status == NO_ERROR) { 6062 readInputParameters_l(); 6063 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6064 } 6065 } 6066 } 6067 6068 return reconfig; 6069} 6070 6071String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6072{ 6073 Mutex::Autolock _l(mLock); 6074 if (initCheck() != NO_ERROR) { 6075 return String8(); 6076 } 6077 6078 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6079 const String8 out_s8(s); 6080 free(s); 6081 return out_s8; 6082} 6083 6084void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6085 AudioSystem::OutputDescriptor desc; 6086 const void *param2 = NULL; 6087 6088 switch (event) { 6089 case AudioSystem::INPUT_OPENED: 6090 case AudioSystem::INPUT_CONFIG_CHANGED: 6091 desc.channelMask = mChannelMask; 6092 desc.samplingRate = mSampleRate; 6093 desc.format = mFormat; 6094 desc.frameCount = mFrameCount; 6095 desc.latency = 0; 6096 param2 = &desc; 6097 break; 6098 6099 case AudioSystem::INPUT_CLOSED: 6100 default: 6101 break; 6102 } 6103 mAudioFlinger->audioConfigChanged(event, mId, param2); 6104} 6105 6106void AudioFlinger::RecordThread::readInputParameters_l() 6107{ 6108 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6109 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6110 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6111 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6112 mFormat = mHALFormat; 6113 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6114 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6115 } 6116 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6117 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6118 mFrameCount = mBufferSize / mFrameSize; 6119 // This is the formula for calculating the temporary buffer size. 6120 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6121 // 1 full output buffer, regardless of the alignment of the available input. 6122 // The value is somewhat arbitrary, and could probably be even larger. 6123 // A larger value should allow more old data to be read after a track calls start(), 6124 // without increasing latency. 6125 mRsmpInFrames = mFrameCount * 7; 6126 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6127 delete[] mRsmpInBuffer; 6128 6129 // TODO optimize audio capture buffer sizes ... 6130 // Here we calculate the size of the sliding buffer used as a source 6131 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6132 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6133 // be better to have it derived from the pipe depth in the long term. 6134 // The current value is higher than necessary. However it should not add to latency. 6135 6136 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6137 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6138 6139 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6140 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6141} 6142 6143uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6144{ 6145 Mutex::Autolock _l(mLock); 6146 if (initCheck() != NO_ERROR) { 6147 return 0; 6148 } 6149 6150 return mInput->stream->get_input_frames_lost(mInput->stream); 6151} 6152 6153uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6154{ 6155 Mutex::Autolock _l(mLock); 6156 uint32_t result = 0; 6157 if (getEffectChain_l(sessionId) != 0) { 6158 result = EFFECT_SESSION; 6159 } 6160 6161 for (size_t i = 0; i < mTracks.size(); ++i) { 6162 if (sessionId == mTracks[i]->sessionId()) { 6163 result |= TRACK_SESSION; 6164 break; 6165 } 6166 } 6167 6168 return result; 6169} 6170 6171KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6172{ 6173 KeyedVector<int, bool> ids; 6174 Mutex::Autolock _l(mLock); 6175 for (size_t j = 0; j < mTracks.size(); ++j) { 6176 sp<RecordThread::RecordTrack> track = mTracks[j]; 6177 int sessionId = track->sessionId(); 6178 if (ids.indexOfKey(sessionId) < 0) { 6179 ids.add(sessionId, true); 6180 } 6181 } 6182 return ids; 6183} 6184 6185AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6186{ 6187 Mutex::Autolock _l(mLock); 6188 AudioStreamIn *input = mInput; 6189 mInput = NULL; 6190 return input; 6191} 6192 6193// this method must always be called either with ThreadBase mLock held or inside the thread loop 6194audio_stream_t* AudioFlinger::RecordThread::stream() const 6195{ 6196 if (mInput == NULL) { 6197 return NULL; 6198 } 6199 return &mInput->stream->common; 6200} 6201 6202status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6203{ 6204 // only one chain per input thread 6205 if (mEffectChains.size() != 0) { 6206 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6207 return INVALID_OPERATION; 6208 } 6209 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6210 chain->setThread(this); 6211 chain->setInBuffer(NULL); 6212 chain->setOutBuffer(NULL); 6213 6214 checkSuspendOnAddEffectChain_l(chain); 6215 6216 // make sure enabled pre processing effects state is communicated to the HAL as we 6217 // just moved them to a new input stream. 6218 chain->syncHalEffectsState(); 6219 6220 mEffectChains.add(chain); 6221 6222 return NO_ERROR; 6223} 6224 6225size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6226{ 6227 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6228 ALOGW_IF(mEffectChains.size() != 1, 6229 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6230 chain.get(), mEffectChains.size(), this); 6231 if (mEffectChains.size() == 1) { 6232 mEffectChains.removeAt(0); 6233 } 6234 return 0; 6235} 6236 6237status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6238 audio_patch_handle_t *handle) 6239{ 6240 status_t status = NO_ERROR; 6241 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6242 // store new device and send to effects 6243 mInDevice = patch->sources[0].ext.device.type; 6244 for (size_t i = 0; i < mEffectChains.size(); i++) { 6245 mEffectChains[i]->setDevice_l(mInDevice); 6246 } 6247 6248 // disable AEC and NS if the device is a BT SCO headset supporting those 6249 // pre processings 6250 if (mTracks.size() > 0) { 6251 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6252 mAudioFlinger->btNrecIsOff(); 6253 for (size_t i = 0; i < mTracks.size(); i++) { 6254 sp<RecordTrack> track = mTracks[i]; 6255 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6256 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6257 } 6258 } 6259 6260 // store new source and send to effects 6261 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6262 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6263 for (size_t i = 0; i < mEffectChains.size(); i++) { 6264 mEffectChains[i]->setAudioSource_l(mAudioSource); 6265 } 6266 } 6267 6268 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6269 status = hwDevice->create_audio_patch(hwDevice, 6270 patch->num_sources, 6271 patch->sources, 6272 patch->num_sinks, 6273 patch->sinks, 6274 handle); 6275 } else { 6276 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6277 } 6278 return status; 6279} 6280 6281status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6282{ 6283 status_t status = NO_ERROR; 6284 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6285 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6286 status = hwDevice->release_audio_patch(hwDevice, handle); 6287 } else { 6288 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6289 } 6290 return status; 6291} 6292 6293void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6294{ 6295 Mutex::Autolock _l(mLock); 6296 mTracks.add(record); 6297} 6298 6299void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6300{ 6301 Mutex::Autolock _l(mLock); 6302 destroyTrack_l(record); 6303} 6304 6305void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6306{ 6307 ThreadBase::getAudioPortConfig(config); 6308 config->role = AUDIO_PORT_ROLE_SINK; 6309 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6310 config->ext.mix.usecase.source = mAudioSource; 6311} 6312 6313}; // namespace android 6314