Threads.cpp revision 44182c206f7c5584ef2cf504da6be98fab665dbf
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 <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/primitives.h> 40#include <audio_utils/format.h> 41#include <audio_utils/minifloat.h> 42 43// NBAIO implementations 44#include <media/nbaio/AudioStreamInSource.h> 45#include <media/nbaio/AudioStreamOutSink.h> 46#include <media/nbaio/MonoPipe.h> 47#include <media/nbaio/MonoPipeReader.h> 48#include <media/nbaio/Pipe.h> 49#include <media/nbaio/PipeReader.h> 50#include <media/nbaio/SourceAudioBufferProvider.h> 51 52#include <powermanager/PowerManager.h> 53 54#include <common_time/cc_helper.h> 55#include <common_time/local_clock.h> 56 57#include "AudioFlinger.h" 58#include "AudioMixer.h" 59#include "FastMixer.h" 60#include "FastCapture.h" 61#include "ServiceUtilities.h" 62#include "SchedulingPolicyService.h" 63 64#ifdef ADD_BATTERY_DATA 65#include <media/IMediaPlayerService.h> 66#include <media/IMediaDeathNotifier.h> 67#endif 68 69#ifdef DEBUG_CPU_USAGE 70#include <cpustats/CentralTendencyStatistics.h> 71#include <cpustats/ThreadCpuUsage.h> 72#endif 73 74// ---------------------------------------------------------------------------- 75 76// Note: the following macro is used for extremely verbose logging message. In 77// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80// turned on. Do not uncomment the #def below unless you really know what you 81// are doing and want to see all of the extremely verbose messages. 82//#define VERY_VERY_VERBOSE_LOGGING 83#ifdef VERY_VERY_VERBOSE_LOGGING 84#define ALOGVV ALOGV 85#else 86#define ALOGVV(a...) do { } while(0) 87#endif 88 89#define max(a, b) ((a) > (b) ? (a) : (b)) 90 91namespace android { 92 93// retry counts for buffer fill timeout 94// 50 * ~20msecs = 1 second 95static const int8_t kMaxTrackRetries = 50; 96static const int8_t kMaxTrackStartupRetries = 50; 97// allow less retry attempts on direct output thread. 98// direct outputs can be a scarce resource in audio hardware and should 99// be released as quickly as possible. 100static const int8_t kMaxTrackRetriesDirect = 2; 101 102// don't warn about blocked writes or record buffer overflows more often than this 103static const nsecs_t kWarningThrottleNs = seconds(5); 104 105// RecordThread loop sleep time upon application overrun or audio HAL read error 106static const int kRecordThreadSleepUs = 5000; 107 108// maximum time to wait in sendConfigEvent_l() for a status to be received 109static const nsecs_t kConfigEventTimeoutNs = seconds(2); 110 111// minimum sleep time for the mixer thread loop when tracks are active but in underrun 112static const uint32_t kMinThreadSleepTimeUs = 5000; 113// maximum divider applied to the active sleep time in the mixer thread loop 114static const uint32_t kMaxThreadSleepTimeShift = 2; 115 116// minimum normal sink buffer size, expressed in milliseconds rather than frames 117static const uint32_t kMinNormalSinkBufferSizeMs = 20; 118// maximum normal sink buffer size 119static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 120 121// Offloaded output thread standby delay: allows track transition without going to standby 122static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 123 124// Whether to use fast mixer 125static const enum { 126 FastMixer_Never, // never initialize or use: for debugging only 127 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 128 // normal mixer multiplier is 1 129 FastMixer_Static, // initialize if needed, then use all the time if initialized, 130 // multiplier is calculated based on min & max normal mixer buffer size 131 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 132 // multiplier is calculated based on min & max normal mixer buffer size 133 // FIXME for FastMixer_Dynamic: 134 // Supporting this option will require fixing HALs that can't handle large writes. 135 // For example, one HAL implementation returns an error from a large write, 136 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 137 // We could either fix the HAL implementations, or provide a wrapper that breaks 138 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 139} kUseFastMixer = FastMixer_Static; 140 141// Whether to use fast capture 142static const enum { 143 FastCapture_Never, // never initialize or use: for debugging only 144 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 145 FastCapture_Static, // initialize if needed, then use all the time if initialized 146} kUseFastCapture = FastCapture_Static; 147 148// Priorities for requestPriority 149static const int kPriorityAudioApp = 2; 150static const int kPriorityFastMixer = 3; 151static const int kPriorityFastCapture = 3; 152 153// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 154// for the track. The client then sub-divides this into smaller buffers for its use. 155// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 156// So for now we just assume that client is double-buffered for fast tracks. 157// FIXME It would be better for client to tell AudioFlinger the value of N, 158// so AudioFlinger could allocate the right amount of memory. 159// See the client's minBufCount and mNotificationFramesAct calculations for details. 160 161// This is the default value, if not specified by property. 162static const int kFastTrackMultiplier = 2; 163 164// The minimum and maximum allowed values 165static const int kFastTrackMultiplierMin = 1; 166static const int kFastTrackMultiplierMax = 2; 167 168// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 169static int sFastTrackMultiplier = kFastTrackMultiplier; 170 171// See Thread::readOnlyHeap(). 172// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 173// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 174// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 175static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 176 177// ---------------------------------------------------------------------------- 178 179static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 180 181static void sFastTrackMultiplierInit() 182{ 183 char value[PROPERTY_VALUE_MAX]; 184 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 185 char *endptr; 186 unsigned long ul = strtoul(value, &endptr, 0); 187 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 188 sFastTrackMultiplier = (int) ul; 189 } 190 } 191} 192 193// ---------------------------------------------------------------------------- 194 195#ifdef ADD_BATTERY_DATA 196// To collect the amplifier usage 197static void addBatteryData(uint32_t params) { 198 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 199 if (service == NULL) { 200 // it already logged 201 return; 202 } 203 204 service->addBatteryData(params); 205} 206#endif 207 208 209// ---------------------------------------------------------------------------- 210// CPU Stats 211// ---------------------------------------------------------------------------- 212 213class CpuStats { 214public: 215 CpuStats(); 216 void sample(const String8 &title); 217#ifdef DEBUG_CPU_USAGE 218private: 219 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 220 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 221 222 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 223 224 int mCpuNum; // thread's current CPU number 225 int mCpukHz; // frequency of thread's current CPU in kHz 226#endif 227}; 228 229CpuStats::CpuStats() 230#ifdef DEBUG_CPU_USAGE 231 : mCpuNum(-1), mCpukHz(-1) 232#endif 233{ 234} 235 236void CpuStats::sample(const String8 &title 237#ifndef DEBUG_CPU_USAGE 238 __unused 239#endif 240 ) { 241#ifdef DEBUG_CPU_USAGE 242 // get current thread's delta CPU time in wall clock ns 243 double wcNs; 244 bool valid = mCpuUsage.sampleAndEnable(wcNs); 245 246 // record sample for wall clock statistics 247 if (valid) { 248 mWcStats.sample(wcNs); 249 } 250 251 // get the current CPU number 252 int cpuNum = sched_getcpu(); 253 254 // get the current CPU frequency in kHz 255 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 256 257 // check if either CPU number or frequency changed 258 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 259 mCpuNum = cpuNum; 260 mCpukHz = cpukHz; 261 // ignore sample for purposes of cycles 262 valid = false; 263 } 264 265 // if no change in CPU number or frequency, then record sample for cycle statistics 266 if (valid && mCpukHz > 0) { 267 double cycles = wcNs * cpukHz * 0.000001; 268 mHzStats.sample(cycles); 269 } 270 271 unsigned n = mWcStats.n(); 272 // mCpuUsage.elapsed() is expensive, so don't call it every loop 273 if ((n & 127) == 1) { 274 long long elapsed = mCpuUsage.elapsed(); 275 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 276 double perLoop = elapsed / (double) n; 277 double perLoop100 = perLoop * 0.01; 278 double perLoop1k = perLoop * 0.001; 279 double mean = mWcStats.mean(); 280 double stddev = mWcStats.stddev(); 281 double minimum = mWcStats.minimum(); 282 double maximum = mWcStats.maximum(); 283 double meanCycles = mHzStats.mean(); 284 double stddevCycles = mHzStats.stddev(); 285 double minCycles = mHzStats.minimum(); 286 double maxCycles = mHzStats.maximum(); 287 mCpuUsage.resetElapsed(); 288 mWcStats.reset(); 289 mHzStats.reset(); 290 ALOGD("CPU usage for %s over past %.1f secs\n" 291 " (%u mixer loops at %.1f mean ms per loop):\n" 292 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 293 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 294 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 295 title.string(), 296 elapsed * .000000001, n, perLoop * .000001, 297 mean * .001, 298 stddev * .001, 299 minimum * .001, 300 maximum * .001, 301 mean / perLoop100, 302 stddev / perLoop100, 303 minimum / perLoop100, 304 maximum / perLoop100, 305 meanCycles / perLoop1k, 306 stddevCycles / perLoop1k, 307 minCycles / perLoop1k, 308 maxCycles / perLoop1k); 309 310 } 311 } 312#endif 313}; 314 315// ---------------------------------------------------------------------------- 316// ThreadBase 317// ---------------------------------------------------------------------------- 318 319// static 320const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 321{ 322 switch (type) { 323 case MIXER: 324 return "MIXER"; 325 case DIRECT: 326 return "DIRECT"; 327 case DUPLICATING: 328 return "DUPLICATING"; 329 case RECORD: 330 return "RECORD"; 331 case OFFLOAD: 332 return "OFFLOAD"; 333 default: 334 return "unknown"; 335 } 336} 337 338String8 devicesToString(audio_devices_t devices) 339{ 340 static const struct mapping { 341 audio_devices_t mDevices; 342 const char * mString; 343 } mappingsOut[] = { 344 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 345 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 346 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 347 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 348 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 349 AUDIO_DEVICE_NONE, "NONE", // must be last 350 }, mappingsIn[] = { 351 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 352 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 353 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 354 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 355 AUDIO_DEVICE_NONE, "NONE", // must be last 356 }; 357 String8 result; 358 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 359 const mapping *entry; 360 if (devices & AUDIO_DEVICE_BIT_IN) { 361 devices &= ~AUDIO_DEVICE_BIT_IN; 362 entry = mappingsIn; 363 } else { 364 entry = mappingsOut; 365 } 366 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 367 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 368 if (devices & entry->mDevices) { 369 if (!result.isEmpty()) { 370 result.append("|"); 371 } 372 result.append(entry->mString); 373 } 374 } 375 if (devices & ~allDevices) { 376 if (!result.isEmpty()) { 377 result.append("|"); 378 } 379 result.appendFormat("0x%X", devices & ~allDevices); 380 } 381 if (result.isEmpty()) { 382 result.append(entry->mString); 383 } 384 return result; 385} 386 387String8 inputFlagsToString(audio_input_flags_t flags) 388{ 389 static const struct mapping { 390 audio_input_flags_t mFlag; 391 const char * mString; 392 } mappings[] = { 393 AUDIO_INPUT_FLAG_FAST, "FAST", 394 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 395 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 396 }; 397 String8 result; 398 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 399 const mapping *entry; 400 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 401 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 402 if (flags & entry->mFlag) { 403 if (!result.isEmpty()) { 404 result.append("|"); 405 } 406 result.append(entry->mString); 407 } 408 } 409 if (flags & ~allFlags) { 410 if (!result.isEmpty()) { 411 result.append("|"); 412 } 413 result.appendFormat("0x%X", flags & ~allFlags); 414 } 415 if (result.isEmpty()) { 416 result.append(entry->mString); 417 } 418 return result; 419} 420 421String8 outputFlagsToString(audio_output_flags_t flags) 422{ 423 static const struct mapping { 424 audio_output_flags_t mFlag; 425 const char * mString; 426 } mappings[] = { 427 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 428 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 429 AUDIO_OUTPUT_FLAG_FAST, "FAST", 430 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 431 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 432 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 433 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 434 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 435 }; 436 String8 result; 437 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 438 const mapping *entry; 439 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 440 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 441 if (flags & entry->mFlag) { 442 if (!result.isEmpty()) { 443 result.append("|"); 444 } 445 result.append(entry->mString); 446 } 447 } 448 if (flags & ~allFlags) { 449 if (!result.isEmpty()) { 450 result.append("|"); 451 } 452 result.appendFormat("0x%X", flags & ~allFlags); 453 } 454 if (result.isEmpty()) { 455 result.append(entry->mString); 456 } 457 return result; 458} 459 460const char *sourceToString(audio_source_t source) 461{ 462 switch (source) { 463 case AUDIO_SOURCE_DEFAULT: return "default"; 464 case AUDIO_SOURCE_MIC: return "mic"; 465 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 466 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 467 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 468 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 469 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 470 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 471 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 472 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 473 case AUDIO_SOURCE_HOTWORD: return "hotword"; 474 default: return "unknown"; 475 } 476} 477 478AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 479 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 480 : Thread(false /*canCallJava*/), 481 mType(type), 482 mAudioFlinger(audioFlinger), 483 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 484 // are set by PlaybackThread::readOutputParameters_l() or 485 // RecordThread::readInputParameters_l() 486 //FIXME: mStandby should be true here. Is this some kind of hack? 487 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 488 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 489 // mName will be set by concrete (non-virtual) subclass 490 mDeathRecipient(new PMDeathRecipient(this)) 491{ 492} 493 494AudioFlinger::ThreadBase::~ThreadBase() 495{ 496 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 497 mConfigEvents.clear(); 498 499 // do not lock the mutex in destructor 500 releaseWakeLock_l(); 501 if (mPowerManager != 0) { 502 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 503 binder->unlinkToDeath(mDeathRecipient); 504 } 505} 506 507status_t AudioFlinger::ThreadBase::readyToRun() 508{ 509 status_t status = initCheck(); 510 if (status == NO_ERROR) { 511 ALOGI("AudioFlinger's thread %p ready to run", this); 512 } else { 513 ALOGE("No working audio driver found."); 514 } 515 return status; 516} 517 518void AudioFlinger::ThreadBase::exit() 519{ 520 ALOGV("ThreadBase::exit"); 521 // do any cleanup required for exit to succeed 522 preExit(); 523 { 524 // This lock prevents the following race in thread (uniprocessor for illustration): 525 // if (!exitPending()) { 526 // // context switch from here to exit() 527 // // exit() calls requestExit(), what exitPending() observes 528 // // exit() calls signal(), which is dropped since no waiters 529 // // context switch back from exit() to here 530 // mWaitWorkCV.wait(...); 531 // // now thread is hung 532 // } 533 AutoMutex lock(mLock); 534 requestExit(); 535 mWaitWorkCV.broadcast(); 536 } 537 // When Thread::requestExitAndWait is made virtual and this method is renamed to 538 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 539 requestExitAndWait(); 540} 541 542status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 543{ 544 status_t status; 545 546 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 547 Mutex::Autolock _l(mLock); 548 549 return sendSetParameterConfigEvent_l(keyValuePairs); 550} 551 552// sendConfigEvent_l() must be called with ThreadBase::mLock held 553// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 554status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 555{ 556 status_t status = NO_ERROR; 557 558 mConfigEvents.add(event); 559 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 560 mWaitWorkCV.signal(); 561 mLock.unlock(); 562 { 563 Mutex::Autolock _l(event->mLock); 564 while (event->mWaitStatus) { 565 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 566 event->mStatus = TIMED_OUT; 567 event->mWaitStatus = false; 568 } 569 } 570 status = event->mStatus; 571 } 572 mLock.lock(); 573 return status; 574} 575 576void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 577{ 578 Mutex::Autolock _l(mLock); 579 sendIoConfigEvent_l(event, param); 580} 581 582// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 583void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 584{ 585 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 586 sendConfigEvent_l(configEvent); 587} 588 589// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 590void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 591{ 592 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 593 sendConfigEvent_l(configEvent); 594} 595 596// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 597status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 598{ 599 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 600 return sendConfigEvent_l(configEvent); 601} 602 603status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 604 const struct audio_patch *patch, 605 audio_patch_handle_t *handle) 606{ 607 Mutex::Autolock _l(mLock); 608 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 609 status_t status = sendConfigEvent_l(configEvent); 610 if (status == NO_ERROR) { 611 CreateAudioPatchConfigEventData *data = 612 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 613 *handle = data->mHandle; 614 } 615 return status; 616} 617 618status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 619 const audio_patch_handle_t handle) 620{ 621 Mutex::Autolock _l(mLock); 622 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 623 return sendConfigEvent_l(configEvent); 624} 625 626 627// post condition: mConfigEvents.isEmpty() 628void AudioFlinger::ThreadBase::processConfigEvents_l() 629{ 630 bool configChanged = false; 631 632 while (!mConfigEvents.isEmpty()) { 633 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 634 sp<ConfigEvent> event = mConfigEvents[0]; 635 mConfigEvents.removeAt(0); 636 switch (event->mType) { 637 case CFG_EVENT_PRIO: { 638 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 639 // FIXME Need to understand why this has to be done asynchronously 640 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 641 true /*asynchronous*/); 642 if (err != 0) { 643 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 644 data->mPrio, data->mPid, data->mTid, err); 645 } 646 } break; 647 case CFG_EVENT_IO: { 648 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 649 audioConfigChanged(data->mEvent, data->mParam); 650 } break; 651 case CFG_EVENT_SET_PARAMETER: { 652 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 653 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 654 configChanged = true; 655 } 656 } break; 657 case CFG_EVENT_CREATE_AUDIO_PATCH: { 658 CreateAudioPatchConfigEventData *data = 659 (CreateAudioPatchConfigEventData *)event->mData.get(); 660 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 661 } break; 662 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 663 ReleaseAudioPatchConfigEventData *data = 664 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 665 event->mStatus = releaseAudioPatch_l(data->mHandle); 666 } break; 667 default: 668 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 669 break; 670 } 671 { 672 Mutex::Autolock _l(event->mLock); 673 if (event->mWaitStatus) { 674 event->mWaitStatus = false; 675 event->mCond.signal(); 676 } 677 } 678 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 679 } 680 681 if (configChanged) { 682 cacheParameters_l(); 683 } 684} 685 686String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 687 String8 s; 688 if (output) { 689 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 690 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 691 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 692 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 693 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 694 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 695 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 696 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 697 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 698 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 699 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 700 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 701 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 702 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 703 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 704 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 705 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 706 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 707 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 708 } else { 709 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 710 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 711 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 712 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 713 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 714 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 715 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 716 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 717 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 718 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 719 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 720 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 721 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 722 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 723 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 724 } 725 int len = s.length(); 726 if (s.length() > 2) { 727 char *str = s.lockBuffer(len); 728 s.unlockBuffer(len - 2); 729 } 730 return s; 731} 732 733void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 734{ 735 const size_t SIZE = 256; 736 char buffer[SIZE]; 737 String8 result; 738 739 bool locked = AudioFlinger::dumpTryLock(mLock); 740 if (!locked) { 741 dprintf(fd, "thread %p may be deadlocked\n", this); 742 } 743 744 dprintf(fd, " Thread name: %s\n", mThreadName); 745 dprintf(fd, " I/O handle: %d\n", mId); 746 dprintf(fd, " TID: %d\n", getTid()); 747 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 748 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 749 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 750 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 751 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 752 dprintf(fd, " Channel count: %u\n", mChannelCount); 753 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 754 channelMaskToString(mChannelMask, mType != RECORD).string()); 755 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 756 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 757 dprintf(fd, " Pending config events:"); 758 size_t numConfig = mConfigEvents.size(); 759 if (numConfig) { 760 for (size_t i = 0; i < numConfig; i++) { 761 mConfigEvents[i]->dump(buffer, SIZE); 762 dprintf(fd, "\n %s", buffer); 763 } 764 dprintf(fd, "\n"); 765 } else { 766 dprintf(fd, " none\n"); 767 } 768 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 769 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 770 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 771 772 if (locked) { 773 mLock.unlock(); 774 } 775} 776 777void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 778{ 779 const size_t SIZE = 256; 780 char buffer[SIZE]; 781 String8 result; 782 783 size_t numEffectChains = mEffectChains.size(); 784 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 785 write(fd, buffer, strlen(buffer)); 786 787 for (size_t i = 0; i < numEffectChains; ++i) { 788 sp<EffectChain> chain = mEffectChains[i]; 789 if (chain != 0) { 790 chain->dump(fd, args); 791 } 792 } 793} 794 795void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 796{ 797 Mutex::Autolock _l(mLock); 798 acquireWakeLock_l(uid); 799} 800 801String16 AudioFlinger::ThreadBase::getWakeLockTag() 802{ 803 switch (mType) { 804 case MIXER: 805 return String16("AudioMix"); 806 case DIRECT: 807 return String16("AudioDirectOut"); 808 case DUPLICATING: 809 return String16("AudioDup"); 810 case RECORD: 811 return String16("AudioIn"); 812 case OFFLOAD: 813 return String16("AudioOffload"); 814 default: 815 ALOG_ASSERT(false); 816 return String16("AudioUnknown"); 817 } 818} 819 820void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 821{ 822 getPowerManager_l(); 823 if (mPowerManager != 0) { 824 sp<IBinder> binder = new BBinder(); 825 status_t status; 826 if (uid >= 0) { 827 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 828 binder, 829 getWakeLockTag(), 830 String16("media"), 831 uid, 832 true /* FIXME force oneway contrary to .aidl */); 833 } else { 834 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 835 binder, 836 getWakeLockTag(), 837 String16("media"), 838 true /* FIXME force oneway contrary to .aidl */); 839 } 840 if (status == NO_ERROR) { 841 mWakeLockToken = binder; 842 } 843 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 844 } 845} 846 847void AudioFlinger::ThreadBase::releaseWakeLock() 848{ 849 Mutex::Autolock _l(mLock); 850 releaseWakeLock_l(); 851} 852 853void AudioFlinger::ThreadBase::releaseWakeLock_l() 854{ 855 if (mWakeLockToken != 0) { 856 ALOGV("releaseWakeLock_l() %s", mThreadName); 857 if (mPowerManager != 0) { 858 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 859 true /* FIXME force oneway contrary to .aidl */); 860 } 861 mWakeLockToken.clear(); 862 } 863} 864 865void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 866 Mutex::Autolock _l(mLock); 867 updateWakeLockUids_l(uids); 868} 869 870void AudioFlinger::ThreadBase::getPowerManager_l() { 871 872 if (mPowerManager == 0) { 873 // use checkService() to avoid blocking if power service is not up yet 874 sp<IBinder> binder = 875 defaultServiceManager()->checkService(String16("power")); 876 if (binder == 0) { 877 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 878 } else { 879 mPowerManager = interface_cast<IPowerManager>(binder); 880 binder->linkToDeath(mDeathRecipient); 881 } 882 } 883} 884 885void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 886 887 getPowerManager_l(); 888 if (mWakeLockToken == NULL) { 889 ALOGE("no wake lock to update!"); 890 return; 891 } 892 if (mPowerManager != 0) { 893 sp<IBinder> binder = new BBinder(); 894 status_t status; 895 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 896 true /* FIXME force oneway contrary to .aidl */); 897 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 898 } 899} 900 901void AudioFlinger::ThreadBase::clearPowerManager() 902{ 903 Mutex::Autolock _l(mLock); 904 releaseWakeLock_l(); 905 mPowerManager.clear(); 906} 907 908void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 909{ 910 sp<ThreadBase> thread = mThread.promote(); 911 if (thread != 0) { 912 thread->clearPowerManager(); 913 } 914 ALOGW("power manager service died !!!"); 915} 916 917void AudioFlinger::ThreadBase::setEffectSuspended( 918 const effect_uuid_t *type, bool suspend, int sessionId) 919{ 920 Mutex::Autolock _l(mLock); 921 setEffectSuspended_l(type, suspend, sessionId); 922} 923 924void AudioFlinger::ThreadBase::setEffectSuspended_l( 925 const effect_uuid_t *type, bool suspend, int sessionId) 926{ 927 sp<EffectChain> chain = getEffectChain_l(sessionId); 928 if (chain != 0) { 929 if (type != NULL) { 930 chain->setEffectSuspended_l(type, suspend); 931 } else { 932 chain->setEffectSuspendedAll_l(suspend); 933 } 934 } 935 936 updateSuspendedSessions_l(type, suspend, sessionId); 937} 938 939void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 940{ 941 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 942 if (index < 0) { 943 return; 944 } 945 946 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 947 mSuspendedSessions.valueAt(index); 948 949 for (size_t i = 0; i < sessionEffects.size(); i++) { 950 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 951 for (int j = 0; j < desc->mRefCount; j++) { 952 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 953 chain->setEffectSuspendedAll_l(true); 954 } else { 955 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 956 desc->mType.timeLow); 957 chain->setEffectSuspended_l(&desc->mType, true); 958 } 959 } 960 } 961} 962 963void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 964 bool suspend, 965 int sessionId) 966{ 967 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 968 969 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 970 971 if (suspend) { 972 if (index >= 0) { 973 sessionEffects = mSuspendedSessions.valueAt(index); 974 } else { 975 mSuspendedSessions.add(sessionId, sessionEffects); 976 } 977 } else { 978 if (index < 0) { 979 return; 980 } 981 sessionEffects = mSuspendedSessions.valueAt(index); 982 } 983 984 985 int key = EffectChain::kKeyForSuspendAll; 986 if (type != NULL) { 987 key = type->timeLow; 988 } 989 index = sessionEffects.indexOfKey(key); 990 991 sp<SuspendedSessionDesc> desc; 992 if (suspend) { 993 if (index >= 0) { 994 desc = sessionEffects.valueAt(index); 995 } else { 996 desc = new SuspendedSessionDesc(); 997 if (type != NULL) { 998 desc->mType = *type; 999 } 1000 sessionEffects.add(key, desc); 1001 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1002 } 1003 desc->mRefCount++; 1004 } else { 1005 if (index < 0) { 1006 return; 1007 } 1008 desc = sessionEffects.valueAt(index); 1009 if (--desc->mRefCount == 0) { 1010 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1011 sessionEffects.removeItemsAt(index); 1012 if (sessionEffects.isEmpty()) { 1013 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1014 sessionId); 1015 mSuspendedSessions.removeItem(sessionId); 1016 } 1017 } 1018 } 1019 if (!sessionEffects.isEmpty()) { 1020 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1021 } 1022} 1023 1024void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1025 bool enabled, 1026 int sessionId) 1027{ 1028 Mutex::Autolock _l(mLock); 1029 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1030} 1031 1032void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1033 bool enabled, 1034 int sessionId) 1035{ 1036 if (mType != RECORD) { 1037 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1038 // another session. This gives the priority to well behaved effect control panels 1039 // and applications not using global effects. 1040 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1041 // global effects 1042 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1043 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1044 } 1045 } 1046 1047 sp<EffectChain> chain = getEffectChain_l(sessionId); 1048 if (chain != 0) { 1049 chain->checkSuspendOnEffectEnabled(effect, enabled); 1050 } 1051} 1052 1053// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1054sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1055 const sp<AudioFlinger::Client>& client, 1056 const sp<IEffectClient>& effectClient, 1057 int32_t priority, 1058 int sessionId, 1059 effect_descriptor_t *desc, 1060 int *enabled, 1061 status_t *status) 1062{ 1063 sp<EffectModule> effect; 1064 sp<EffectHandle> handle; 1065 status_t lStatus; 1066 sp<EffectChain> chain; 1067 bool chainCreated = false; 1068 bool effectCreated = false; 1069 bool effectRegistered = false; 1070 1071 lStatus = initCheck(); 1072 if (lStatus != NO_ERROR) { 1073 ALOGW("createEffect_l() Audio driver not initialized."); 1074 goto Exit; 1075 } 1076 1077 // Reject any effect on Direct output threads for now, since the format of 1078 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1079 if (mType == DIRECT) { 1080 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1081 desc->name, mThreadName); 1082 lStatus = BAD_VALUE; 1083 goto Exit; 1084 } 1085 1086 // Reject any effect on mixer or duplicating multichannel sinks. 1087 // TODO: fix both format and multichannel issues with effects. 1088 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1089 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1090 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1091 lStatus = BAD_VALUE; 1092 goto Exit; 1093 } 1094 1095 // Allow global effects only on offloaded and mixer threads 1096 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1097 switch (mType) { 1098 case MIXER: 1099 case OFFLOAD: 1100 break; 1101 case DIRECT: 1102 case DUPLICATING: 1103 case RECORD: 1104 default: 1105 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1106 desc->name, mThreadName); 1107 lStatus = BAD_VALUE; 1108 goto Exit; 1109 } 1110 } 1111 1112 // Only Pre processor effects are allowed on input threads and only on input threads 1113 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1114 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1115 desc->name, desc->flags, mType); 1116 lStatus = BAD_VALUE; 1117 goto Exit; 1118 } 1119 1120 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1121 1122 { // scope for mLock 1123 Mutex::Autolock _l(mLock); 1124 1125 // check for existing effect chain with the requested audio session 1126 chain = getEffectChain_l(sessionId); 1127 if (chain == 0) { 1128 // create a new chain for this session 1129 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1130 chain = new EffectChain(this, sessionId); 1131 addEffectChain_l(chain); 1132 chain->setStrategy(getStrategyForSession_l(sessionId)); 1133 chainCreated = true; 1134 } else { 1135 effect = chain->getEffectFromDesc_l(desc); 1136 } 1137 1138 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1139 1140 if (effect == 0) { 1141 int id = mAudioFlinger->nextUniqueId(); 1142 // Check CPU and memory usage 1143 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1144 if (lStatus != NO_ERROR) { 1145 goto Exit; 1146 } 1147 effectRegistered = true; 1148 // create a new effect module if none present in the chain 1149 effect = new EffectModule(this, chain, desc, id, sessionId); 1150 lStatus = effect->status(); 1151 if (lStatus != NO_ERROR) { 1152 goto Exit; 1153 } 1154 effect->setOffloaded(mType == OFFLOAD, mId); 1155 1156 lStatus = chain->addEffect_l(effect); 1157 if (lStatus != NO_ERROR) { 1158 goto Exit; 1159 } 1160 effectCreated = true; 1161 1162 effect->setDevice(mOutDevice); 1163 effect->setDevice(mInDevice); 1164 effect->setMode(mAudioFlinger->getMode()); 1165 effect->setAudioSource(mAudioSource); 1166 } 1167 // create effect handle and connect it to effect module 1168 handle = new EffectHandle(effect, client, effectClient, priority); 1169 lStatus = handle->initCheck(); 1170 if (lStatus == OK) { 1171 lStatus = effect->addHandle(handle.get()); 1172 } 1173 if (enabled != NULL) { 1174 *enabled = (int)effect->isEnabled(); 1175 } 1176 } 1177 1178Exit: 1179 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1180 Mutex::Autolock _l(mLock); 1181 if (effectCreated) { 1182 chain->removeEffect_l(effect); 1183 } 1184 if (effectRegistered) { 1185 AudioSystem::unregisterEffect(effect->id()); 1186 } 1187 if (chainCreated) { 1188 removeEffectChain_l(chain); 1189 } 1190 handle.clear(); 1191 } 1192 1193 *status = lStatus; 1194 return handle; 1195} 1196 1197sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1198{ 1199 Mutex::Autolock _l(mLock); 1200 return getEffect_l(sessionId, effectId); 1201} 1202 1203sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1204{ 1205 sp<EffectChain> chain = getEffectChain_l(sessionId); 1206 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1207} 1208 1209// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1210// PlaybackThread::mLock held 1211status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1212{ 1213 // check for existing effect chain with the requested audio session 1214 int sessionId = effect->sessionId(); 1215 sp<EffectChain> chain = getEffectChain_l(sessionId); 1216 bool chainCreated = false; 1217 1218 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1219 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1220 this, effect->desc().name, effect->desc().flags); 1221 1222 if (chain == 0) { 1223 // create a new chain for this session 1224 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1225 chain = new EffectChain(this, sessionId); 1226 addEffectChain_l(chain); 1227 chain->setStrategy(getStrategyForSession_l(sessionId)); 1228 chainCreated = true; 1229 } 1230 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1231 1232 if (chain->getEffectFromId_l(effect->id()) != 0) { 1233 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1234 this, effect->desc().name, chain.get()); 1235 return BAD_VALUE; 1236 } 1237 1238 effect->setOffloaded(mType == OFFLOAD, mId); 1239 1240 status_t status = chain->addEffect_l(effect); 1241 if (status != NO_ERROR) { 1242 if (chainCreated) { 1243 removeEffectChain_l(chain); 1244 } 1245 return status; 1246 } 1247 1248 effect->setDevice(mOutDevice); 1249 effect->setDevice(mInDevice); 1250 effect->setMode(mAudioFlinger->getMode()); 1251 effect->setAudioSource(mAudioSource); 1252 return NO_ERROR; 1253} 1254 1255void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1256 1257 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1258 effect_descriptor_t desc = effect->desc(); 1259 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1260 detachAuxEffect_l(effect->id()); 1261 } 1262 1263 sp<EffectChain> chain = effect->chain().promote(); 1264 if (chain != 0) { 1265 // remove effect chain if removing last effect 1266 if (chain->removeEffect_l(effect) == 0) { 1267 removeEffectChain_l(chain); 1268 } 1269 } else { 1270 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1271 } 1272} 1273 1274void AudioFlinger::ThreadBase::lockEffectChains_l( 1275 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1276{ 1277 effectChains = mEffectChains; 1278 for (size_t i = 0; i < mEffectChains.size(); i++) { 1279 mEffectChains[i]->lock(); 1280 } 1281} 1282 1283void AudioFlinger::ThreadBase::unlockEffectChains( 1284 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1285{ 1286 for (size_t i = 0; i < effectChains.size(); i++) { 1287 effectChains[i]->unlock(); 1288 } 1289} 1290 1291sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1292{ 1293 Mutex::Autolock _l(mLock); 1294 return getEffectChain_l(sessionId); 1295} 1296 1297sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1298{ 1299 size_t size = mEffectChains.size(); 1300 for (size_t i = 0; i < size; i++) { 1301 if (mEffectChains[i]->sessionId() == sessionId) { 1302 return mEffectChains[i]; 1303 } 1304 } 1305 return 0; 1306} 1307 1308void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1309{ 1310 Mutex::Autolock _l(mLock); 1311 size_t size = mEffectChains.size(); 1312 for (size_t i = 0; i < size; i++) { 1313 mEffectChains[i]->setMode_l(mode); 1314 } 1315} 1316 1317void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1318{ 1319 config->type = AUDIO_PORT_TYPE_MIX; 1320 config->ext.mix.handle = mId; 1321 config->sample_rate = mSampleRate; 1322 config->format = mFormat; 1323 config->channel_mask = mChannelMask; 1324 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1325 AUDIO_PORT_CONFIG_FORMAT; 1326} 1327 1328 1329// ---------------------------------------------------------------------------- 1330// Playback 1331// ---------------------------------------------------------------------------- 1332 1333AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1334 AudioStreamOut* output, 1335 audio_io_handle_t id, 1336 audio_devices_t device, 1337 type_t type) 1338 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1339 mNormalFrameCount(0), mSinkBuffer(NULL), 1340 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1341 mMixerBuffer(NULL), 1342 mMixerBufferSize(0), 1343 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1344 mMixerBufferValid(false), 1345 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1346 mEffectBuffer(NULL), 1347 mEffectBufferSize(0), 1348 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1349 mEffectBufferValid(false), 1350 mSuspended(0), mBytesWritten(0), 1351 mActiveTracksGeneration(0), 1352 // mStreamTypes[] initialized in constructor body 1353 mOutput(output), 1354 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1355 mMixerStatus(MIXER_IDLE), 1356 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1357 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1358 mBytesRemaining(0), 1359 mCurrentWriteLength(0), 1360 mUseAsyncWrite(false), 1361 mWriteAckSequence(0), 1362 mDrainSequence(0), 1363 mSignalPending(false), 1364 mScreenState(AudioFlinger::mScreenState), 1365 // index 0 is reserved for normal mixer's submix 1366 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1367 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1368 // mLatchD, mLatchQ, 1369 mLatchDValid(false), mLatchQValid(false) 1370{ 1371 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1372 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1373 1374 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1375 // it would be safer to explicitly pass initial masterVolume/masterMute as 1376 // parameter. 1377 // 1378 // If the HAL we are using has support for master volume or master mute, 1379 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1380 // and the mute set to false). 1381 mMasterVolume = audioFlinger->masterVolume_l(); 1382 mMasterMute = audioFlinger->masterMute_l(); 1383 if (mOutput && mOutput->audioHwDev) { 1384 if (mOutput->audioHwDev->canSetMasterVolume()) { 1385 mMasterVolume = 1.0; 1386 } 1387 1388 if (mOutput->audioHwDev->canSetMasterMute()) { 1389 mMasterMute = false; 1390 } 1391 } 1392 1393 readOutputParameters_l(); 1394 1395 // ++ operator does not compile 1396 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1397 stream = (audio_stream_type_t) (stream + 1)) { 1398 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1399 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1400 } 1401} 1402 1403AudioFlinger::PlaybackThread::~PlaybackThread() 1404{ 1405 mAudioFlinger->unregisterWriter(mNBLogWriter); 1406 free(mSinkBuffer); 1407 free(mMixerBuffer); 1408 free(mEffectBuffer); 1409} 1410 1411void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1412{ 1413 dumpInternals(fd, args); 1414 dumpTracks(fd, args); 1415 dumpEffectChains(fd, args); 1416} 1417 1418void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1419{ 1420 const size_t SIZE = 256; 1421 char buffer[SIZE]; 1422 String8 result; 1423 1424 result.appendFormat(" Stream volumes in dB: "); 1425 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1426 const stream_type_t *st = &mStreamTypes[i]; 1427 if (i > 0) { 1428 result.appendFormat(", "); 1429 } 1430 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1431 if (st->mute) { 1432 result.append("M"); 1433 } 1434 } 1435 result.append("\n"); 1436 write(fd, result.string(), result.length()); 1437 result.clear(); 1438 1439 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1440 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1441 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1442 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1443 1444 size_t numtracks = mTracks.size(); 1445 size_t numactive = mActiveTracks.size(); 1446 dprintf(fd, " %d Tracks", numtracks); 1447 size_t numactiveseen = 0; 1448 if (numtracks) { 1449 dprintf(fd, " of which %d are active\n", numactive); 1450 Track::appendDumpHeader(result); 1451 for (size_t i = 0; i < numtracks; ++i) { 1452 sp<Track> track = mTracks[i]; 1453 if (track != 0) { 1454 bool active = mActiveTracks.indexOf(track) >= 0; 1455 if (active) { 1456 numactiveseen++; 1457 } 1458 track->dump(buffer, SIZE, active); 1459 result.append(buffer); 1460 } 1461 } 1462 } else { 1463 result.append("\n"); 1464 } 1465 if (numactiveseen != numactive) { 1466 // some tracks in the active list were not in the tracks list 1467 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1468 " not in the track list\n"); 1469 result.append(buffer); 1470 Track::appendDumpHeader(result); 1471 for (size_t i = 0; i < numactive; ++i) { 1472 sp<Track> track = mActiveTracks[i].promote(); 1473 if (track != 0 && mTracks.indexOf(track) < 0) { 1474 track->dump(buffer, SIZE, true); 1475 result.append(buffer); 1476 } 1477 } 1478 } 1479 1480 write(fd, result.string(), result.size()); 1481} 1482 1483void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1484{ 1485 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1486 1487 dumpBase(fd, args); 1488 1489 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1490 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1491 dprintf(fd, " Total writes: %d\n", mNumWrites); 1492 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1493 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1494 dprintf(fd, " Suspend count: %d\n", mSuspended); 1495 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1496 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1497 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1498 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1499 AudioStreamOut *output = mOutput; 1500 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1501 String8 flagsAsString = outputFlagsToString(flags); 1502 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1503} 1504 1505// Thread virtuals 1506 1507void AudioFlinger::PlaybackThread::onFirstRef() 1508{ 1509 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1510} 1511 1512// ThreadBase virtuals 1513void AudioFlinger::PlaybackThread::preExit() 1514{ 1515 ALOGV(" preExit()"); 1516 // FIXME this is using hard-coded strings but in the future, this functionality will be 1517 // converted to use audio HAL extensions required to support tunneling 1518 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1519} 1520 1521// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1522sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1523 const sp<AudioFlinger::Client>& client, 1524 audio_stream_type_t streamType, 1525 uint32_t sampleRate, 1526 audio_format_t format, 1527 audio_channel_mask_t channelMask, 1528 size_t *pFrameCount, 1529 const sp<IMemory>& sharedBuffer, 1530 int sessionId, 1531 IAudioFlinger::track_flags_t *flags, 1532 pid_t tid, 1533 int uid, 1534 status_t *status) 1535{ 1536 size_t frameCount = *pFrameCount; 1537 sp<Track> track; 1538 status_t lStatus; 1539 1540 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1541 1542 // client expresses a preference for FAST, but we get the final say 1543 if (*flags & IAudioFlinger::TRACK_FAST) { 1544 if ( 1545 // not timed 1546 (!isTimed) && 1547 // either of these use cases: 1548 ( 1549 // use case 1: shared buffer with any frame count 1550 ( 1551 (sharedBuffer != 0) 1552 ) || 1553 // use case 2: callback handler and frame count is default or at least as large as HAL 1554 ( 1555 (tid != -1) && 1556 ((frameCount == 0) || 1557 (frameCount >= mFrameCount)) 1558 ) 1559 ) && 1560 // PCM data 1561 audio_is_linear_pcm(format) && 1562 // identical channel mask to sink, or mono in and stereo sink 1563 (channelMask == mChannelMask || 1564 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1565 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1566 // hardware sample rate 1567 (sampleRate == mSampleRate) && 1568 // normal mixer has an associated fast mixer 1569 hasFastMixer() && 1570 // there are sufficient fast track slots available 1571 (mFastTrackAvailMask != 0) 1572 // FIXME test that MixerThread for this fast track has a capable output HAL 1573 // FIXME add a permission test also? 1574 ) { 1575 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1576 if (frameCount == 0) { 1577 // read the fast track multiplier property the first time it is needed 1578 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1579 if (ok != 0) { 1580 ALOGE("%s pthread_once failed: %d", __func__, ok); 1581 } 1582 frameCount = mFrameCount * sFastTrackMultiplier; 1583 } 1584 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1585 frameCount, mFrameCount); 1586 } else { 1587 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1588 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1589 "sampleRate=%u mSampleRate=%u " 1590 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1591 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1592 audio_is_linear_pcm(format), 1593 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1594 *flags &= ~IAudioFlinger::TRACK_FAST; 1595 } 1596 } 1597 // For normal PCM streaming tracks, update minimum frame count. 1598 // For compatibility with AudioTrack calculation, buffer depth is forced 1599 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1600 // This is probably too conservative, but legacy application code may depend on it. 1601 // If you change this calculation, also review the start threshold which is related. 1602 if (!(*flags & IAudioFlinger::TRACK_FAST) 1603 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1604 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1605 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1606 if (minBufCount < 2) { 1607 minBufCount = 2; 1608 } 1609 size_t minFrameCount = 1610 minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate); 1611 if (frameCount < minFrameCount) { // including frameCount == 0 1612 frameCount = minFrameCount; 1613 } 1614 } 1615 *pFrameCount = frameCount; 1616 1617 switch (mType) { 1618 1619 case DIRECT: 1620 if (audio_is_linear_pcm(format)) { 1621 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1622 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1623 "for output %p with format %#x", 1624 sampleRate, format, channelMask, mOutput, mFormat); 1625 lStatus = BAD_VALUE; 1626 goto Exit; 1627 } 1628 } 1629 break; 1630 1631 case OFFLOAD: 1632 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1633 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1634 "for output %p with format %#x", 1635 sampleRate, format, channelMask, mOutput, mFormat); 1636 lStatus = BAD_VALUE; 1637 goto Exit; 1638 } 1639 break; 1640 1641 default: 1642 if (!audio_is_linear_pcm(format)) { 1643 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1644 "for output %p with format %#x", 1645 format, mOutput, mFormat); 1646 lStatus = BAD_VALUE; 1647 goto Exit; 1648 } 1649 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1650 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1651 lStatus = BAD_VALUE; 1652 goto Exit; 1653 } 1654 break; 1655 1656 } 1657 1658 lStatus = initCheck(); 1659 if (lStatus != NO_ERROR) { 1660 ALOGE("createTrack_l() audio driver not initialized"); 1661 goto Exit; 1662 } 1663 1664 { // scope for mLock 1665 Mutex::Autolock _l(mLock); 1666 1667 // all tracks in same audio session must share the same routing strategy otherwise 1668 // conflicts will happen when tracks are moved from one output to another by audio policy 1669 // manager 1670 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1671 for (size_t i = 0; i < mTracks.size(); ++i) { 1672 sp<Track> t = mTracks[i]; 1673 if (t != 0 && t->isExternalTrack()) { 1674 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1675 if (sessionId == t->sessionId() && strategy != actual) { 1676 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1677 strategy, actual); 1678 lStatus = BAD_VALUE; 1679 goto Exit; 1680 } 1681 } 1682 } 1683 1684 if (!isTimed) { 1685 track = new Track(this, client, streamType, sampleRate, format, 1686 channelMask, frameCount, NULL, sharedBuffer, 1687 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1688 } else { 1689 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1690 channelMask, frameCount, sharedBuffer, sessionId, uid); 1691 } 1692 1693 // new Track always returns non-NULL, 1694 // but TimedTrack::create() is a factory that could fail by returning NULL 1695 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1696 if (lStatus != NO_ERROR) { 1697 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1698 // track must be cleared from the caller as the caller has the AF lock 1699 goto Exit; 1700 } 1701 mTracks.add(track); 1702 1703 sp<EffectChain> chain = getEffectChain_l(sessionId); 1704 if (chain != 0) { 1705 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1706 track->setMainBuffer(chain->inBuffer()); 1707 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1708 chain->incTrackCnt(); 1709 } 1710 1711 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1712 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1713 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1714 // so ask activity manager to do this on our behalf 1715 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1716 } 1717 } 1718 1719 lStatus = NO_ERROR; 1720 1721Exit: 1722 *status = lStatus; 1723 return track; 1724} 1725 1726uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1727{ 1728 return latency; 1729} 1730 1731uint32_t AudioFlinger::PlaybackThread::latency() const 1732{ 1733 Mutex::Autolock _l(mLock); 1734 return latency_l(); 1735} 1736uint32_t AudioFlinger::PlaybackThread::latency_l() const 1737{ 1738 if (initCheck() == NO_ERROR) { 1739 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1740 } else { 1741 return 0; 1742 } 1743} 1744 1745void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1746{ 1747 Mutex::Autolock _l(mLock); 1748 // Don't apply master volume in SW if our HAL can do it for us. 1749 if (mOutput && mOutput->audioHwDev && 1750 mOutput->audioHwDev->canSetMasterVolume()) { 1751 mMasterVolume = 1.0; 1752 } else { 1753 mMasterVolume = value; 1754 } 1755} 1756 1757void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1758{ 1759 Mutex::Autolock _l(mLock); 1760 // Don't apply master mute in SW if our HAL can do it for us. 1761 if (mOutput && mOutput->audioHwDev && 1762 mOutput->audioHwDev->canSetMasterMute()) { 1763 mMasterMute = false; 1764 } else { 1765 mMasterMute = muted; 1766 } 1767} 1768 1769void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1770{ 1771 Mutex::Autolock _l(mLock); 1772 mStreamTypes[stream].volume = value; 1773 broadcast_l(); 1774} 1775 1776void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1777{ 1778 Mutex::Autolock _l(mLock); 1779 mStreamTypes[stream].mute = muted; 1780 broadcast_l(); 1781} 1782 1783float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1784{ 1785 Mutex::Autolock _l(mLock); 1786 return mStreamTypes[stream].volume; 1787} 1788 1789// addTrack_l() must be called with ThreadBase::mLock held 1790status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1791{ 1792 status_t status = ALREADY_EXISTS; 1793 1794 // set retry count for buffer fill 1795 track->mRetryCount = kMaxTrackStartupRetries; 1796 if (mActiveTracks.indexOf(track) < 0) { 1797 // the track is newly added, make sure it fills up all its 1798 // buffers before playing. This is to ensure the client will 1799 // effectively get the latency it requested. 1800 if (track->isExternalTrack()) { 1801 TrackBase::track_state state = track->mState; 1802 mLock.unlock(); 1803 status = AudioSystem::startOutput(mId, track->streamType(), 1804 (audio_session_t)track->sessionId()); 1805 mLock.lock(); 1806 // abort track was stopped/paused while we released the lock 1807 if (state != track->mState) { 1808 if (status == NO_ERROR) { 1809 mLock.unlock(); 1810 AudioSystem::stopOutput(mId, track->streamType(), 1811 (audio_session_t)track->sessionId()); 1812 mLock.lock(); 1813 } 1814 return INVALID_OPERATION; 1815 } 1816 // abort if start is rejected by audio policy manager 1817 if (status != NO_ERROR) { 1818 return PERMISSION_DENIED; 1819 } 1820#ifdef ADD_BATTERY_DATA 1821 // to track the speaker usage 1822 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1823#endif 1824 } 1825 1826 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1827 track->mResetDone = false; 1828 track->mPresentationCompleteFrames = 0; 1829 mActiveTracks.add(track); 1830 mWakeLockUids.add(track->uid()); 1831 mActiveTracksGeneration++; 1832 mLatestActiveTrack = track; 1833 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1834 if (chain != 0) { 1835 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1836 track->sessionId()); 1837 chain->incActiveTrackCnt(); 1838 } 1839 1840 status = NO_ERROR; 1841 } 1842 1843 onAddNewTrack_l(); 1844 return status; 1845} 1846 1847bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1848{ 1849 track->terminate(); 1850 // active tracks are removed by threadLoop() 1851 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1852 track->mState = TrackBase::STOPPED; 1853 if (!trackActive) { 1854 removeTrack_l(track); 1855 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1856 track->mState = TrackBase::STOPPING_1; 1857 } 1858 1859 return trackActive; 1860} 1861 1862void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1863{ 1864 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1865 mTracks.remove(track); 1866 deleteTrackName_l(track->name()); 1867 // redundant as track is about to be destroyed, for dumpsys only 1868 track->mName = -1; 1869 if (track->isFastTrack()) { 1870 int index = track->mFastIndex; 1871 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1872 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1873 mFastTrackAvailMask |= 1 << index; 1874 // redundant as track is about to be destroyed, for dumpsys only 1875 track->mFastIndex = -1; 1876 } 1877 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1878 if (chain != 0) { 1879 chain->decTrackCnt(); 1880 } 1881} 1882 1883void AudioFlinger::PlaybackThread::broadcast_l() 1884{ 1885 // Thread could be blocked waiting for async 1886 // so signal it to handle state changes immediately 1887 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1888 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1889 mSignalPending = true; 1890 mWaitWorkCV.broadcast(); 1891} 1892 1893String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1894{ 1895 Mutex::Autolock _l(mLock); 1896 if (initCheck() != NO_ERROR) { 1897 return String8(); 1898 } 1899 1900 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1901 const String8 out_s8(s); 1902 free(s); 1903 return out_s8; 1904} 1905 1906void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1907 AudioSystem::OutputDescriptor desc; 1908 void *param2 = NULL; 1909 1910 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1911 param); 1912 1913 switch (event) { 1914 case AudioSystem::OUTPUT_OPENED: 1915 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1916 desc.channelMask = mChannelMask; 1917 desc.samplingRate = mSampleRate; 1918 desc.format = mFormat; 1919 desc.frameCount = mNormalFrameCount; // FIXME see 1920 // AudioFlinger::frameCount(audio_io_handle_t) 1921 desc.latency = latency_l(); 1922 param2 = &desc; 1923 break; 1924 1925 case AudioSystem::STREAM_CONFIG_CHANGED: 1926 param2 = ¶m; 1927 case AudioSystem::OUTPUT_CLOSED: 1928 default: 1929 break; 1930 } 1931 mAudioFlinger->audioConfigChanged(event, mId, param2); 1932} 1933 1934void AudioFlinger::PlaybackThread::writeCallback() 1935{ 1936 ALOG_ASSERT(mCallbackThread != 0); 1937 mCallbackThread->resetWriteBlocked(); 1938} 1939 1940void AudioFlinger::PlaybackThread::drainCallback() 1941{ 1942 ALOG_ASSERT(mCallbackThread != 0); 1943 mCallbackThread->resetDraining(); 1944} 1945 1946void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1947{ 1948 Mutex::Autolock _l(mLock); 1949 // reject out of sequence requests 1950 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1951 mWriteAckSequence &= ~1; 1952 mWaitWorkCV.signal(); 1953 } 1954} 1955 1956void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1957{ 1958 Mutex::Autolock _l(mLock); 1959 // reject out of sequence requests 1960 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1961 mDrainSequence &= ~1; 1962 mWaitWorkCV.signal(); 1963 } 1964} 1965 1966// static 1967int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1968 void *param __unused, 1969 void *cookie) 1970{ 1971 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1972 ALOGV("asyncCallback() event %d", event); 1973 switch (event) { 1974 case STREAM_CBK_EVENT_WRITE_READY: 1975 me->writeCallback(); 1976 break; 1977 case STREAM_CBK_EVENT_DRAIN_READY: 1978 me->drainCallback(); 1979 break; 1980 default: 1981 ALOGW("asyncCallback() unknown event %d", event); 1982 break; 1983 } 1984 return 0; 1985} 1986 1987void AudioFlinger::PlaybackThread::readOutputParameters_l() 1988{ 1989 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1990 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1991 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1992 if (!audio_is_output_channel(mChannelMask)) { 1993 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1994 } 1995 if ((mType == MIXER || mType == DUPLICATING) 1996 && !isValidPcmSinkChannelMask(mChannelMask)) { 1997 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1998 mChannelMask); 1999 } 2000 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2001 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2002 mFormat = mHALFormat; 2003 if (!audio_is_valid_format(mFormat)) { 2004 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2005 } 2006 if ((mType == MIXER || mType == DUPLICATING) 2007 && !isValidPcmSinkFormat(mFormat)) { 2008 LOG_FATAL("HAL format %#x not supported for mixed output", 2009 mFormat); 2010 } 2011 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 2012 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2013 mFrameCount = mBufferSize / mFrameSize; 2014 if (mFrameCount & 15) { 2015 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2016 mFrameCount); 2017 } 2018 2019 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2020 (mOutput->stream->set_callback != NULL)) { 2021 if (mOutput->stream->set_callback(mOutput->stream, 2022 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2023 mUseAsyncWrite = true; 2024 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2025 } 2026 } 2027 2028 mHwSupportsPause = false; 2029 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2030 if (mOutput->stream->pause != NULL) { 2031 if (mOutput->stream->resume != NULL) { 2032 mHwSupportsPause = true; 2033 } else { 2034 ALOGW("direct output implements pause but not resume"); 2035 } 2036 } else if (mOutput->stream->resume != NULL) { 2037 ALOGW("direct output implements resume but not pause"); 2038 } 2039 } 2040 2041 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2042 // For best precision, we use float instead of the associated output 2043 // device format (typically PCM 16 bit). 2044 2045 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2046 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2047 mBufferSize = mFrameSize * mFrameCount; 2048 2049 // TODO: We currently use the associated output device channel mask and sample rate. 2050 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2051 // (if a valid mask) to avoid premature downmix. 2052 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2053 // instead of the output device sample rate to avoid loss of high frequency information. 2054 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2055 } 2056 2057 // Calculate size of normal sink buffer relative to the HAL output buffer size 2058 double multiplier = 1.0; 2059 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2060 kUseFastMixer == FastMixer_Dynamic)) { 2061 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2062 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2063 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2064 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2065 maxNormalFrameCount = maxNormalFrameCount & ~15; 2066 if (maxNormalFrameCount < minNormalFrameCount) { 2067 maxNormalFrameCount = minNormalFrameCount; 2068 } 2069 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2070 if (multiplier <= 1.0) { 2071 multiplier = 1.0; 2072 } else if (multiplier <= 2.0) { 2073 if (2 * mFrameCount <= maxNormalFrameCount) { 2074 multiplier = 2.0; 2075 } else { 2076 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2077 } 2078 } else { 2079 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2080 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2081 // track, but we sometimes have to do this to satisfy the maximum frame count 2082 // constraint) 2083 // FIXME this rounding up should not be done if no HAL SRC 2084 uint32_t truncMult = (uint32_t) multiplier; 2085 if ((truncMult & 1)) { 2086 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2087 ++truncMult; 2088 } 2089 } 2090 multiplier = (double) truncMult; 2091 } 2092 } 2093 mNormalFrameCount = multiplier * mFrameCount; 2094 // round up to nearest 16 frames to satisfy AudioMixer 2095 if (mType == MIXER || mType == DUPLICATING) { 2096 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2097 } 2098 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2099 mNormalFrameCount); 2100 2101 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2102 // Originally this was int16_t[] array, need to remove legacy implications. 2103 free(mSinkBuffer); 2104 mSinkBuffer = NULL; 2105 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2106 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2107 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2108 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2109 2110 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2111 // drives the output. 2112 free(mMixerBuffer); 2113 mMixerBuffer = NULL; 2114 if (mMixerBufferEnabled) { 2115 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2116 mMixerBufferSize = mNormalFrameCount * mChannelCount 2117 * audio_bytes_per_sample(mMixerBufferFormat); 2118 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2119 } 2120 free(mEffectBuffer); 2121 mEffectBuffer = NULL; 2122 if (mEffectBufferEnabled) { 2123 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2124 mEffectBufferSize = mNormalFrameCount * mChannelCount 2125 * audio_bytes_per_sample(mEffectBufferFormat); 2126 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2127 } 2128 2129 // force reconfiguration of effect chains and engines to take new buffer size and audio 2130 // parameters into account 2131 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2132 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2133 // matter. 2134 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2135 Vector< sp<EffectChain> > effectChains = mEffectChains; 2136 for (size_t i = 0; i < effectChains.size(); i ++) { 2137 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2138 } 2139} 2140 2141 2142status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2143{ 2144 if (halFrames == NULL || dspFrames == NULL) { 2145 return BAD_VALUE; 2146 } 2147 Mutex::Autolock _l(mLock); 2148 if (initCheck() != NO_ERROR) { 2149 return INVALID_OPERATION; 2150 } 2151 size_t framesWritten = mBytesWritten / mFrameSize; 2152 *halFrames = framesWritten; 2153 2154 if (isSuspended()) { 2155 // return an estimation of rendered frames when the output is suspended 2156 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2157 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2158 return NO_ERROR; 2159 } else { 2160 status_t status; 2161 uint32_t frames; 2162 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 2163 *dspFrames = (size_t)frames; 2164 return status; 2165 } 2166} 2167 2168uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2169{ 2170 Mutex::Autolock _l(mLock); 2171 uint32_t result = 0; 2172 if (getEffectChain_l(sessionId) != 0) { 2173 result = EFFECT_SESSION; 2174 } 2175 2176 for (size_t i = 0; i < mTracks.size(); ++i) { 2177 sp<Track> track = mTracks[i]; 2178 if (sessionId == track->sessionId() && !track->isInvalid()) { 2179 result |= TRACK_SESSION; 2180 break; 2181 } 2182 } 2183 2184 return result; 2185} 2186 2187uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2188{ 2189 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2190 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2191 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2192 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2193 } 2194 for (size_t i = 0; i < mTracks.size(); i++) { 2195 sp<Track> track = mTracks[i]; 2196 if (sessionId == track->sessionId() && !track->isInvalid()) { 2197 return AudioSystem::getStrategyForStream(track->streamType()); 2198 } 2199 } 2200 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2201} 2202 2203 2204AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2205{ 2206 Mutex::Autolock _l(mLock); 2207 return mOutput; 2208} 2209 2210AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2211{ 2212 Mutex::Autolock _l(mLock); 2213 AudioStreamOut *output = mOutput; 2214 mOutput = NULL; 2215 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2216 // must push a NULL and wait for ack 2217 mOutputSink.clear(); 2218 mPipeSink.clear(); 2219 mNormalSink.clear(); 2220 return output; 2221} 2222 2223// this method must always be called either with ThreadBase mLock held or inside the thread loop 2224audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2225{ 2226 if (mOutput == NULL) { 2227 return NULL; 2228 } 2229 return &mOutput->stream->common; 2230} 2231 2232uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2233{ 2234 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2235} 2236 2237status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2238{ 2239 if (!isValidSyncEvent(event)) { 2240 return BAD_VALUE; 2241 } 2242 2243 Mutex::Autolock _l(mLock); 2244 2245 for (size_t i = 0; i < mTracks.size(); ++i) { 2246 sp<Track> track = mTracks[i]; 2247 if (event->triggerSession() == track->sessionId()) { 2248 (void) track->setSyncEvent(event); 2249 return NO_ERROR; 2250 } 2251 } 2252 2253 return NAME_NOT_FOUND; 2254} 2255 2256bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2257{ 2258 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2259} 2260 2261void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2262 const Vector< sp<Track> >& tracksToRemove) 2263{ 2264 size_t count = tracksToRemove.size(); 2265 if (count > 0) { 2266 for (size_t i = 0 ; i < count ; i++) { 2267 const sp<Track>& track = tracksToRemove.itemAt(i); 2268 if (track->isExternalTrack()) { 2269 AudioSystem::stopOutput(mId, track->streamType(), 2270 (audio_session_t)track->sessionId()); 2271#ifdef ADD_BATTERY_DATA 2272 // to track the speaker usage 2273 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2274#endif 2275 if (track->isTerminated()) { 2276 AudioSystem::releaseOutput(mId, track->streamType(), 2277 (audio_session_t)track->sessionId()); 2278 } 2279 } 2280 } 2281 } 2282} 2283 2284void AudioFlinger::PlaybackThread::checkSilentMode_l() 2285{ 2286 if (!mMasterMute) { 2287 char value[PROPERTY_VALUE_MAX]; 2288 if (property_get("ro.audio.silent", value, "0") > 0) { 2289 char *endptr; 2290 unsigned long ul = strtoul(value, &endptr, 0); 2291 if (*endptr == '\0' && ul != 0) { 2292 ALOGD("Silence is golden"); 2293 // The setprop command will not allow a property to be changed after 2294 // the first time it is set, so we don't have to worry about un-muting. 2295 setMasterMute_l(true); 2296 } 2297 } 2298 } 2299} 2300 2301// shared by MIXER and DIRECT, overridden by DUPLICATING 2302ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2303{ 2304 // FIXME rewrite to reduce number of system calls 2305 mLastWriteTime = systemTime(); 2306 mInWrite = true; 2307 ssize_t bytesWritten; 2308 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2309 2310 // If an NBAIO sink is present, use it to write the normal mixer's submix 2311 if (mNormalSink != 0) { 2312 2313 const size_t count = mBytesRemaining / mFrameSize; 2314 2315 ATRACE_BEGIN("write"); 2316 // update the setpoint when AudioFlinger::mScreenState changes 2317 uint32_t screenState = AudioFlinger::mScreenState; 2318 if (screenState != mScreenState) { 2319 mScreenState = screenState; 2320 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2321 if (pipe != NULL) { 2322 pipe->setAvgFrames((mScreenState & 1) ? 2323 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2324 } 2325 } 2326 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2327 ATRACE_END(); 2328 if (framesWritten > 0) { 2329 bytesWritten = framesWritten * mFrameSize; 2330 } else { 2331 bytesWritten = framesWritten; 2332 } 2333 mLatchDValid = false; 2334 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2335 if (status == NO_ERROR) { 2336 size_t totalFramesWritten = mNormalSink->framesWritten(); 2337 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2338 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2339 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2340 mLatchDValid = true; 2341 } 2342 } 2343 // otherwise use the HAL / AudioStreamOut directly 2344 } else { 2345 // Direct output and offload threads 2346 2347 if (mUseAsyncWrite) { 2348 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2349 mWriteAckSequence += 2; 2350 mWriteAckSequence |= 1; 2351 ALOG_ASSERT(mCallbackThread != 0); 2352 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2353 } 2354 // FIXME We should have an implementation of timestamps for direct output threads. 2355 // They are used e.g for multichannel PCM playback over HDMI. 2356 bytesWritten = mOutput->stream->write(mOutput->stream, 2357 (char *)mSinkBuffer + offset, mBytesRemaining); 2358 if (mUseAsyncWrite && 2359 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2360 // do not wait for async callback in case of error of full write 2361 mWriteAckSequence &= ~1; 2362 ALOG_ASSERT(mCallbackThread != 0); 2363 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2364 } 2365 } 2366 2367 mNumWrites++; 2368 mInWrite = false; 2369 mStandby = false; 2370 return bytesWritten; 2371} 2372 2373void AudioFlinger::PlaybackThread::threadLoop_drain() 2374{ 2375 if (mOutput->stream->drain) { 2376 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2377 if (mUseAsyncWrite) { 2378 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2379 mDrainSequence |= 1; 2380 ALOG_ASSERT(mCallbackThread != 0); 2381 mCallbackThread->setDraining(mDrainSequence); 2382 } 2383 mOutput->stream->drain(mOutput->stream, 2384 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2385 : AUDIO_DRAIN_ALL); 2386 } 2387} 2388 2389void AudioFlinger::PlaybackThread::threadLoop_exit() 2390{ 2391 { 2392 Mutex::Autolock _l(mLock); 2393 for (size_t i = 0; i < mTracks.size(); i++) { 2394 sp<Track> track = mTracks[i]; 2395 track->invalidate(); 2396 } 2397 } 2398} 2399 2400/* 2401The derived values that are cached: 2402 - mSinkBufferSize from frame count * frame size 2403 - activeSleepTime from activeSleepTimeUs() 2404 - idleSleepTime from idleSleepTimeUs() 2405 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2406 - maxPeriod from frame count and sample rate (MIXER only) 2407 2408The parameters that affect these derived values are: 2409 - frame count 2410 - frame size 2411 - sample rate 2412 - device type: A2DP or not 2413 - device latency 2414 - format: PCM or not 2415 - active sleep time 2416 - idle sleep time 2417*/ 2418 2419void AudioFlinger::PlaybackThread::cacheParameters_l() 2420{ 2421 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2422 activeSleepTime = activeSleepTimeUs(); 2423 idleSleepTime = idleSleepTimeUs(); 2424} 2425 2426void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2427{ 2428 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2429 this, streamType, mTracks.size()); 2430 Mutex::Autolock _l(mLock); 2431 2432 size_t size = mTracks.size(); 2433 for (size_t i = 0; i < size; i++) { 2434 sp<Track> t = mTracks[i]; 2435 if (t->streamType() == streamType) { 2436 t->invalidate(); 2437 } 2438 } 2439} 2440 2441status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2442{ 2443 int session = chain->sessionId(); 2444 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2445 ? mEffectBuffer : mSinkBuffer); 2446 bool ownsBuffer = false; 2447 2448 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2449 if (session > 0) { 2450 // Only one effect chain can be present in direct output thread and it uses 2451 // the sink buffer as input 2452 if (mType != DIRECT) { 2453 size_t numSamples = mNormalFrameCount * mChannelCount; 2454 buffer = new int16_t[numSamples]; 2455 memset(buffer, 0, numSamples * sizeof(int16_t)); 2456 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2457 ownsBuffer = true; 2458 } 2459 2460 // Attach all tracks with same session ID to this chain. 2461 for (size_t i = 0; i < mTracks.size(); ++i) { 2462 sp<Track> track = mTracks[i]; 2463 if (session == track->sessionId()) { 2464 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2465 buffer); 2466 track->setMainBuffer(buffer); 2467 chain->incTrackCnt(); 2468 } 2469 } 2470 2471 // indicate all active tracks in the chain 2472 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2473 sp<Track> track = mActiveTracks[i].promote(); 2474 if (track == 0) { 2475 continue; 2476 } 2477 if (session == track->sessionId()) { 2478 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2479 chain->incActiveTrackCnt(); 2480 } 2481 } 2482 } 2483 chain->setThread(this); 2484 chain->setInBuffer(buffer, ownsBuffer); 2485 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2486 ? mEffectBuffer : mSinkBuffer)); 2487 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2488 // chains list in order to be processed last as it contains output stage effects 2489 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2490 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2491 // after track specific effects and before output stage 2492 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2493 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2494 // Effect chain for other sessions are inserted at beginning of effect 2495 // chains list to be processed before output mix effects. Relative order between other 2496 // sessions is not important 2497 size_t size = mEffectChains.size(); 2498 size_t i = 0; 2499 for (i = 0; i < size; i++) { 2500 if (mEffectChains[i]->sessionId() < session) { 2501 break; 2502 } 2503 } 2504 mEffectChains.insertAt(chain, i); 2505 checkSuspendOnAddEffectChain_l(chain); 2506 2507 return NO_ERROR; 2508} 2509 2510size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2511{ 2512 int session = chain->sessionId(); 2513 2514 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2515 2516 for (size_t i = 0; i < mEffectChains.size(); i++) { 2517 if (chain == mEffectChains[i]) { 2518 mEffectChains.removeAt(i); 2519 // detach all active tracks from the chain 2520 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2521 sp<Track> track = mActiveTracks[i].promote(); 2522 if (track == 0) { 2523 continue; 2524 } 2525 if (session == track->sessionId()) { 2526 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2527 chain.get(), session); 2528 chain->decActiveTrackCnt(); 2529 } 2530 } 2531 2532 // detach all tracks with same session ID from this chain 2533 for (size_t i = 0; i < mTracks.size(); ++i) { 2534 sp<Track> track = mTracks[i]; 2535 if (session == track->sessionId()) { 2536 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2537 chain->decTrackCnt(); 2538 } 2539 } 2540 break; 2541 } 2542 } 2543 return mEffectChains.size(); 2544} 2545 2546status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2547 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2548{ 2549 Mutex::Autolock _l(mLock); 2550 return attachAuxEffect_l(track, EffectId); 2551} 2552 2553status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2554 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2555{ 2556 status_t status = NO_ERROR; 2557 2558 if (EffectId == 0) { 2559 track->setAuxBuffer(0, NULL); 2560 } else { 2561 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2562 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2563 if (effect != 0) { 2564 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2565 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2566 } else { 2567 status = INVALID_OPERATION; 2568 } 2569 } else { 2570 status = BAD_VALUE; 2571 } 2572 } 2573 return status; 2574} 2575 2576void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2577{ 2578 for (size_t i = 0; i < mTracks.size(); ++i) { 2579 sp<Track> track = mTracks[i]; 2580 if (track->auxEffectId() == effectId) { 2581 attachAuxEffect_l(track, 0); 2582 } 2583 } 2584} 2585 2586bool AudioFlinger::PlaybackThread::threadLoop() 2587{ 2588 Vector< sp<Track> > tracksToRemove; 2589 2590 standbyTime = systemTime(); 2591 2592 // MIXER 2593 nsecs_t lastWarning = 0; 2594 2595 // DUPLICATING 2596 // FIXME could this be made local to while loop? 2597 writeFrames = 0; 2598 2599 int lastGeneration = 0; 2600 2601 cacheParameters_l(); 2602 sleepTime = idleSleepTime; 2603 2604 if (mType == MIXER) { 2605 sleepTimeShift = 0; 2606 } 2607 2608 CpuStats cpuStats; 2609 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2610 2611 acquireWakeLock(); 2612 2613 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2614 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2615 // and then that string will be logged at the next convenient opportunity. 2616 const char *logString = NULL; 2617 2618 checkSilentMode_l(); 2619 2620 while (!exitPending()) 2621 { 2622 cpuStats.sample(myName); 2623 2624 Vector< sp<EffectChain> > effectChains; 2625 2626 { // scope for mLock 2627 2628 Mutex::Autolock _l(mLock); 2629 2630 processConfigEvents_l(); 2631 2632 if (logString != NULL) { 2633 mNBLogWriter->logTimestamp(); 2634 mNBLogWriter->log(logString); 2635 logString = NULL; 2636 } 2637 2638 // Gather the framesReleased counters for all active tracks, 2639 // and latch them atomically with the timestamp. 2640 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2641 mLatchD.mFramesReleased.clear(); 2642 size_t size = mActiveTracks.size(); 2643 for (size_t i = 0; i < size; i++) { 2644 sp<Track> t = mActiveTracks[i].promote(); 2645 if (t != 0) { 2646 mLatchD.mFramesReleased.add(t.get(), 2647 t->mAudioTrackServerProxy->framesReleased()); 2648 } 2649 } 2650 if (mLatchDValid) { 2651 mLatchQ = mLatchD; 2652 mLatchDValid = false; 2653 mLatchQValid = true; 2654 } 2655 2656 saveOutputTracks(); 2657 if (mSignalPending) { 2658 // A signal was raised while we were unlocked 2659 mSignalPending = false; 2660 } else if (waitingAsyncCallback_l()) { 2661 if (exitPending()) { 2662 break; 2663 } 2664 releaseWakeLock_l(); 2665 mWakeLockUids.clear(); 2666 mActiveTracksGeneration++; 2667 ALOGV("wait async completion"); 2668 mWaitWorkCV.wait(mLock); 2669 ALOGV("async completion/wake"); 2670 acquireWakeLock_l(); 2671 standbyTime = systemTime() + standbyDelay; 2672 sleepTime = 0; 2673 2674 continue; 2675 } 2676 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2677 isSuspended()) { 2678 // put audio hardware into standby after short delay 2679 if (shouldStandby_l()) { 2680 2681 threadLoop_standby(); 2682 2683 mStandby = true; 2684 } 2685 2686 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2687 // we're about to wait, flush the binder command buffer 2688 IPCThreadState::self()->flushCommands(); 2689 2690 clearOutputTracks(); 2691 2692 if (exitPending()) { 2693 break; 2694 } 2695 2696 releaseWakeLock_l(); 2697 mWakeLockUids.clear(); 2698 mActiveTracksGeneration++; 2699 // wait until we have something to do... 2700 ALOGV("%s going to sleep", myName.string()); 2701 mWaitWorkCV.wait(mLock); 2702 ALOGV("%s waking up", myName.string()); 2703 acquireWakeLock_l(); 2704 2705 mMixerStatus = MIXER_IDLE; 2706 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2707 mBytesWritten = 0; 2708 mBytesRemaining = 0; 2709 checkSilentMode_l(); 2710 2711 standbyTime = systemTime() + standbyDelay; 2712 sleepTime = idleSleepTime; 2713 if (mType == MIXER) { 2714 sleepTimeShift = 0; 2715 } 2716 2717 continue; 2718 } 2719 } 2720 // mMixerStatusIgnoringFastTracks is also updated internally 2721 mMixerStatus = prepareTracks_l(&tracksToRemove); 2722 2723 // compare with previously applied list 2724 if (lastGeneration != mActiveTracksGeneration) { 2725 // update wakelock 2726 updateWakeLockUids_l(mWakeLockUids); 2727 lastGeneration = mActiveTracksGeneration; 2728 } 2729 2730 // prevent any changes in effect chain list and in each effect chain 2731 // during mixing and effect process as the audio buffers could be deleted 2732 // or modified if an effect is created or deleted 2733 lockEffectChains_l(effectChains); 2734 } // mLock scope ends 2735 2736 if (mBytesRemaining == 0) { 2737 mCurrentWriteLength = 0; 2738 if (mMixerStatus == MIXER_TRACKS_READY) { 2739 // threadLoop_mix() sets mCurrentWriteLength 2740 threadLoop_mix(); 2741 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2742 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2743 // threadLoop_sleepTime sets sleepTime to 0 if data 2744 // must be written to HAL 2745 threadLoop_sleepTime(); 2746 if (sleepTime == 0) { 2747 mCurrentWriteLength = mSinkBufferSize; 2748 } 2749 } 2750 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2751 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2752 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2753 // or mSinkBuffer (if there are no effects). 2754 // 2755 // This is done pre-effects computation; if effects change to 2756 // support higher precision, this needs to move. 2757 // 2758 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2759 // TODO use sleepTime == 0 as an additional condition. 2760 if (mMixerBufferValid) { 2761 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2762 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2763 2764 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2765 mNormalFrameCount * mChannelCount); 2766 } 2767 2768 mBytesRemaining = mCurrentWriteLength; 2769 if (isSuspended()) { 2770 sleepTime = suspendSleepTimeUs(); 2771 // simulate write to HAL when suspended 2772 mBytesWritten += mSinkBufferSize; 2773 mBytesRemaining = 0; 2774 } 2775 2776 // only process effects if we're going to write 2777 if (sleepTime == 0 && mType != OFFLOAD) { 2778 for (size_t i = 0; i < effectChains.size(); i ++) { 2779 effectChains[i]->process_l(); 2780 } 2781 } 2782 } 2783 // Process effect chains for offloaded thread even if no audio 2784 // was read from audio track: process only updates effect state 2785 // and thus does have to be synchronized with audio writes but may have 2786 // to be called while waiting for async write callback 2787 if (mType == OFFLOAD) { 2788 for (size_t i = 0; i < effectChains.size(); i ++) { 2789 effectChains[i]->process_l(); 2790 } 2791 } 2792 2793 // Only if the Effects buffer is enabled and there is data in the 2794 // Effects buffer (buffer valid), we need to 2795 // copy into the sink buffer. 2796 // TODO use sleepTime == 0 as an additional condition. 2797 if (mEffectBufferValid) { 2798 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2799 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2800 mNormalFrameCount * mChannelCount); 2801 } 2802 2803 // enable changes in effect chain 2804 unlockEffectChains(effectChains); 2805 2806 if (!waitingAsyncCallback()) { 2807 // sleepTime == 0 means we must write to audio hardware 2808 if (sleepTime == 0) { 2809 if (mBytesRemaining) { 2810 ssize_t ret = threadLoop_write(); 2811 if (ret < 0) { 2812 mBytesRemaining = 0; 2813 } else { 2814 mBytesWritten += ret; 2815 mBytesRemaining -= ret; 2816 } 2817 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2818 (mMixerStatus == MIXER_DRAIN_ALL)) { 2819 threadLoop_drain(); 2820 } 2821 if (mType == MIXER) { 2822 // write blocked detection 2823 nsecs_t now = systemTime(); 2824 nsecs_t delta = now - mLastWriteTime; 2825 if (!mStandby && delta > maxPeriod) { 2826 mNumDelayedWrites++; 2827 if ((now - lastWarning) > kWarningThrottleNs) { 2828 ATRACE_NAME("underrun"); 2829 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2830 ns2ms(delta), mNumDelayedWrites, this); 2831 lastWarning = now; 2832 } 2833 } 2834 } 2835 2836 } else { 2837 ATRACE_BEGIN("sleep"); 2838 usleep(sleepTime); 2839 ATRACE_END(); 2840 } 2841 } 2842 2843 // Finally let go of removed track(s), without the lock held 2844 // since we can't guarantee the destructors won't acquire that 2845 // same lock. This will also mutate and push a new fast mixer state. 2846 threadLoop_removeTracks(tracksToRemove); 2847 tracksToRemove.clear(); 2848 2849 // FIXME I don't understand the need for this here; 2850 // it was in the original code but maybe the 2851 // assignment in saveOutputTracks() makes this unnecessary? 2852 clearOutputTracks(); 2853 2854 // Effect chains will be actually deleted here if they were removed from 2855 // mEffectChains list during mixing or effects processing 2856 effectChains.clear(); 2857 2858 // FIXME Note that the above .clear() is no longer necessary since effectChains 2859 // is now local to this block, but will keep it for now (at least until merge done). 2860 } 2861 2862 threadLoop_exit(); 2863 2864 if (!mStandby) { 2865 threadLoop_standby(); 2866 mStandby = true; 2867 } 2868 2869 releaseWakeLock(); 2870 mWakeLockUids.clear(); 2871 mActiveTracksGeneration++; 2872 2873 ALOGV("Thread %p type %d exiting", this, mType); 2874 return false; 2875} 2876 2877// removeTracks_l() must be called with ThreadBase::mLock held 2878void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2879{ 2880 size_t count = tracksToRemove.size(); 2881 if (count > 0) { 2882 for (size_t i=0 ; i<count ; i++) { 2883 const sp<Track>& track = tracksToRemove.itemAt(i); 2884 mActiveTracks.remove(track); 2885 mWakeLockUids.remove(track->uid()); 2886 mActiveTracksGeneration++; 2887 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2888 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2889 if (chain != 0) { 2890 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2891 track->sessionId()); 2892 chain->decActiveTrackCnt(); 2893 } 2894 if (track->isTerminated()) { 2895 removeTrack_l(track); 2896 } 2897 } 2898 } 2899 2900} 2901 2902status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2903{ 2904 if (mNormalSink != 0) { 2905 return mNormalSink->getTimestamp(timestamp); 2906 } 2907 if ((mType == OFFLOAD || mType == DIRECT) 2908 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2909 uint64_t position64; 2910 int ret = mOutput->stream->get_presentation_position( 2911 mOutput->stream, &position64, ×tamp.mTime); 2912 if (ret == 0) { 2913 timestamp.mPosition = (uint32_t)position64; 2914 return NO_ERROR; 2915 } 2916 } 2917 return INVALID_OPERATION; 2918} 2919 2920status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2921 audio_patch_handle_t *handle) 2922{ 2923 status_t status = NO_ERROR; 2924 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2925 // store new device and send to effects 2926 audio_devices_t type = AUDIO_DEVICE_NONE; 2927 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2928 type |= patch->sinks[i].ext.device.type; 2929 } 2930 mOutDevice = type; 2931 for (size_t i = 0; i < mEffectChains.size(); i++) { 2932 mEffectChains[i]->setDevice_l(mOutDevice); 2933 } 2934 2935 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2936 status = hwDevice->create_audio_patch(hwDevice, 2937 patch->num_sources, 2938 patch->sources, 2939 patch->num_sinks, 2940 patch->sinks, 2941 handle); 2942 } else { 2943 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2944 } 2945 return status; 2946} 2947 2948status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2949{ 2950 status_t status = NO_ERROR; 2951 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2952 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2953 status = hwDevice->release_audio_patch(hwDevice, handle); 2954 } else { 2955 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2956 } 2957 return status; 2958} 2959 2960void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2961{ 2962 Mutex::Autolock _l(mLock); 2963 mTracks.add(track); 2964} 2965 2966void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2967{ 2968 Mutex::Autolock _l(mLock); 2969 destroyTrack_l(track); 2970} 2971 2972void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2973{ 2974 ThreadBase::getAudioPortConfig(config); 2975 config->role = AUDIO_PORT_ROLE_SOURCE; 2976 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2977 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2978} 2979 2980// ---------------------------------------------------------------------------- 2981 2982AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2983 audio_io_handle_t id, audio_devices_t device, type_t type) 2984 : PlaybackThread(audioFlinger, output, id, device, type), 2985 // mAudioMixer below 2986 // mFastMixer below 2987 mFastMixerFutex(0) 2988 // mOutputSink below 2989 // mPipeSink below 2990 // mNormalSink below 2991{ 2992 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2993 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2994 "mFrameCount=%d, mNormalFrameCount=%d", 2995 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2996 mNormalFrameCount); 2997 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2998 2999 if (type == DUPLICATING) { 3000 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3001 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3002 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3003 return; 3004 } 3005 // create an NBAIO sink for the HAL output stream, and negotiate 3006 mOutputSink = new AudioStreamOutSink(output->stream); 3007 size_t numCounterOffers = 0; 3008 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3009 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3010 ALOG_ASSERT(index == 0); 3011 3012 // initialize fast mixer depending on configuration 3013 bool initFastMixer; 3014 switch (kUseFastMixer) { 3015 case FastMixer_Never: 3016 initFastMixer = false; 3017 break; 3018 case FastMixer_Always: 3019 initFastMixer = true; 3020 break; 3021 case FastMixer_Static: 3022 case FastMixer_Dynamic: 3023 initFastMixer = mFrameCount < mNormalFrameCount; 3024 break; 3025 } 3026 if (initFastMixer) { 3027 audio_format_t fastMixerFormat; 3028 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3029 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3030 } else { 3031 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3032 } 3033 if (mFormat != fastMixerFormat) { 3034 // change our Sink format to accept our intermediate precision 3035 mFormat = fastMixerFormat; 3036 free(mSinkBuffer); 3037 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3038 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3039 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3040 } 3041 3042 // create a MonoPipe to connect our submix to FastMixer 3043 NBAIO_Format format = mOutputSink->format(); 3044 NBAIO_Format origformat = format; 3045 // adjust format to match that of the Fast Mixer 3046 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3047 format.mFormat = fastMixerFormat; 3048 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3049 3050 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3051 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3052 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3053 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3054 const NBAIO_Format offers[1] = {format}; 3055 size_t numCounterOffers = 0; 3056 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3057 ALOG_ASSERT(index == 0); 3058 monoPipe->setAvgFrames((mScreenState & 1) ? 3059 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3060 mPipeSink = monoPipe; 3061 3062#ifdef TEE_SINK 3063 if (mTeeSinkOutputEnabled) { 3064 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3065 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3066 const NBAIO_Format offers2[1] = {origformat}; 3067 numCounterOffers = 0; 3068 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3069 ALOG_ASSERT(index == 0); 3070 mTeeSink = teeSink; 3071 PipeReader *teeSource = new PipeReader(*teeSink); 3072 numCounterOffers = 0; 3073 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3074 ALOG_ASSERT(index == 0); 3075 mTeeSource = teeSource; 3076 } 3077#endif 3078 3079 // create fast mixer and configure it initially with just one fast track for our submix 3080 mFastMixer = new FastMixer(); 3081 FastMixerStateQueue *sq = mFastMixer->sq(); 3082#ifdef STATE_QUEUE_DUMP 3083 sq->setObserverDump(&mStateQueueObserverDump); 3084 sq->setMutatorDump(&mStateQueueMutatorDump); 3085#endif 3086 FastMixerState *state = sq->begin(); 3087 FastTrack *fastTrack = &state->mFastTracks[0]; 3088 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3089 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3090 fastTrack->mVolumeProvider = NULL; 3091 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3092 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3093 fastTrack->mGeneration++; 3094 state->mFastTracksGen++; 3095 state->mTrackMask = 1; 3096 // fast mixer will use the HAL output sink 3097 state->mOutputSink = mOutputSink.get(); 3098 state->mOutputSinkGen++; 3099 state->mFrameCount = mFrameCount; 3100 state->mCommand = FastMixerState::COLD_IDLE; 3101 // already done in constructor initialization list 3102 //mFastMixerFutex = 0; 3103 state->mColdFutexAddr = &mFastMixerFutex; 3104 state->mColdGen++; 3105 state->mDumpState = &mFastMixerDumpState; 3106#ifdef TEE_SINK 3107 state->mTeeSink = mTeeSink.get(); 3108#endif 3109 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3110 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3111 sq->end(); 3112 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3113 3114 // start the fast mixer 3115 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3116 pid_t tid = mFastMixer->getTid(); 3117 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3118 if (err != 0) { 3119 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3120 kPriorityFastMixer, getpid_cached, tid, err); 3121 } 3122 3123#ifdef AUDIO_WATCHDOG 3124 // create and start the watchdog 3125 mAudioWatchdog = new AudioWatchdog(); 3126 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3127 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3128 tid = mAudioWatchdog->getTid(); 3129 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3130 if (err != 0) { 3131 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3132 kPriorityFastMixer, getpid_cached, tid, err); 3133 } 3134#endif 3135 3136 } 3137 3138 switch (kUseFastMixer) { 3139 case FastMixer_Never: 3140 case FastMixer_Dynamic: 3141 mNormalSink = mOutputSink; 3142 break; 3143 case FastMixer_Always: 3144 mNormalSink = mPipeSink; 3145 break; 3146 case FastMixer_Static: 3147 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3148 break; 3149 } 3150} 3151 3152AudioFlinger::MixerThread::~MixerThread() 3153{ 3154 if (mFastMixer != 0) { 3155 FastMixerStateQueue *sq = mFastMixer->sq(); 3156 FastMixerState *state = sq->begin(); 3157 if (state->mCommand == FastMixerState::COLD_IDLE) { 3158 int32_t old = android_atomic_inc(&mFastMixerFutex); 3159 if (old == -1) { 3160 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3161 } 3162 } 3163 state->mCommand = FastMixerState::EXIT; 3164 sq->end(); 3165 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3166 mFastMixer->join(); 3167 // Though the fast mixer thread has exited, it's state queue is still valid. 3168 // We'll use that extract the final state which contains one remaining fast track 3169 // corresponding to our sub-mix. 3170 state = sq->begin(); 3171 ALOG_ASSERT(state->mTrackMask == 1); 3172 FastTrack *fastTrack = &state->mFastTracks[0]; 3173 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3174 delete fastTrack->mBufferProvider; 3175 sq->end(false /*didModify*/); 3176 mFastMixer.clear(); 3177#ifdef AUDIO_WATCHDOG 3178 if (mAudioWatchdog != 0) { 3179 mAudioWatchdog->requestExit(); 3180 mAudioWatchdog->requestExitAndWait(); 3181 mAudioWatchdog.clear(); 3182 } 3183#endif 3184 } 3185 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3186 delete mAudioMixer; 3187} 3188 3189 3190uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3191{ 3192 if (mFastMixer != 0) { 3193 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3194 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3195 } 3196 return latency; 3197} 3198 3199 3200void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3201{ 3202 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3203} 3204 3205ssize_t AudioFlinger::MixerThread::threadLoop_write() 3206{ 3207 // FIXME we should only do one push per cycle; confirm this is true 3208 // Start the fast mixer if it's not already running 3209 if (mFastMixer != 0) { 3210 FastMixerStateQueue *sq = mFastMixer->sq(); 3211 FastMixerState *state = sq->begin(); 3212 if (state->mCommand != FastMixerState::MIX_WRITE && 3213 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3214 if (state->mCommand == FastMixerState::COLD_IDLE) { 3215 int32_t old = android_atomic_inc(&mFastMixerFutex); 3216 if (old == -1) { 3217 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3218 } 3219#ifdef AUDIO_WATCHDOG 3220 if (mAudioWatchdog != 0) { 3221 mAudioWatchdog->resume(); 3222 } 3223#endif 3224 } 3225 state->mCommand = FastMixerState::MIX_WRITE; 3226#ifdef FAST_THREAD_STATISTICS 3227 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3228 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3229#endif 3230 sq->end(); 3231 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3232 if (kUseFastMixer == FastMixer_Dynamic) { 3233 mNormalSink = mPipeSink; 3234 } 3235 } else { 3236 sq->end(false /*didModify*/); 3237 } 3238 } 3239 return PlaybackThread::threadLoop_write(); 3240} 3241 3242void AudioFlinger::MixerThread::threadLoop_standby() 3243{ 3244 // Idle the fast mixer if it's currently running 3245 if (mFastMixer != 0) { 3246 FastMixerStateQueue *sq = mFastMixer->sq(); 3247 FastMixerState *state = sq->begin(); 3248 if (!(state->mCommand & FastMixerState::IDLE)) { 3249 state->mCommand = FastMixerState::COLD_IDLE; 3250 state->mColdFutexAddr = &mFastMixerFutex; 3251 state->mColdGen++; 3252 mFastMixerFutex = 0; 3253 sq->end(); 3254 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3255 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3256 if (kUseFastMixer == FastMixer_Dynamic) { 3257 mNormalSink = mOutputSink; 3258 } 3259#ifdef AUDIO_WATCHDOG 3260 if (mAudioWatchdog != 0) { 3261 mAudioWatchdog->pause(); 3262 } 3263#endif 3264 } else { 3265 sq->end(false /*didModify*/); 3266 } 3267 } 3268 PlaybackThread::threadLoop_standby(); 3269} 3270 3271bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3272{ 3273 return false; 3274} 3275 3276bool AudioFlinger::PlaybackThread::shouldStandby_l() 3277{ 3278 return !mStandby; 3279} 3280 3281bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3282{ 3283 Mutex::Autolock _l(mLock); 3284 return waitingAsyncCallback_l(); 3285} 3286 3287// shared by MIXER and DIRECT, overridden by DUPLICATING 3288void AudioFlinger::PlaybackThread::threadLoop_standby() 3289{ 3290 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3291 mOutput->stream->common.standby(&mOutput->stream->common); 3292 if (mUseAsyncWrite != 0) { 3293 // discard any pending drain or write ack by incrementing sequence 3294 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3295 mDrainSequence = (mDrainSequence + 2) & ~1; 3296 ALOG_ASSERT(mCallbackThread != 0); 3297 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3298 mCallbackThread->setDraining(mDrainSequence); 3299 } 3300 mHwPaused = false; 3301} 3302 3303void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3304{ 3305 ALOGV("signal playback thread"); 3306 broadcast_l(); 3307} 3308 3309void AudioFlinger::MixerThread::threadLoop_mix() 3310{ 3311 // obtain the presentation timestamp of the next output buffer 3312 int64_t pts; 3313 status_t status = INVALID_OPERATION; 3314 3315 if (mNormalSink != 0) { 3316 status = mNormalSink->getNextWriteTimestamp(&pts); 3317 } else { 3318 status = mOutputSink->getNextWriteTimestamp(&pts); 3319 } 3320 3321 if (status != NO_ERROR) { 3322 pts = AudioBufferProvider::kInvalidPTS; 3323 } 3324 3325 // mix buffers... 3326 mAudioMixer->process(pts); 3327 mCurrentWriteLength = mSinkBufferSize; 3328 // increase sleep time progressively when application underrun condition clears. 3329 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3330 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3331 // such that we would underrun the audio HAL. 3332 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3333 sleepTimeShift--; 3334 } 3335 sleepTime = 0; 3336 standbyTime = systemTime() + standbyDelay; 3337 //TODO: delay standby when effects have a tail 3338 3339} 3340 3341void AudioFlinger::MixerThread::threadLoop_sleepTime() 3342{ 3343 // If no tracks are ready, sleep once for the duration of an output 3344 // buffer size, then write 0s to the output 3345 if (sleepTime == 0) { 3346 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3347 sleepTime = activeSleepTime >> sleepTimeShift; 3348 if (sleepTime < kMinThreadSleepTimeUs) { 3349 sleepTime = kMinThreadSleepTimeUs; 3350 } 3351 // reduce sleep time in case of consecutive application underruns to avoid 3352 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3353 // duration we would end up writing less data than needed by the audio HAL if 3354 // the condition persists. 3355 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3356 sleepTimeShift++; 3357 } 3358 } else { 3359 sleepTime = idleSleepTime; 3360 } 3361 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3362 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3363 // before effects processing or output. 3364 if (mMixerBufferValid) { 3365 memset(mMixerBuffer, 0, mMixerBufferSize); 3366 } else { 3367 memset(mSinkBuffer, 0, mSinkBufferSize); 3368 } 3369 sleepTime = 0; 3370 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3371 "anticipated start"); 3372 } 3373 // TODO add standby time extension fct of effect tail 3374} 3375 3376// prepareTracks_l() must be called with ThreadBase::mLock held 3377AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3378 Vector< sp<Track> > *tracksToRemove) 3379{ 3380 3381 mixer_state mixerStatus = MIXER_IDLE; 3382 // find out which tracks need to be processed 3383 size_t count = mActiveTracks.size(); 3384 size_t mixedTracks = 0; 3385 size_t tracksWithEffect = 0; 3386 // counts only _active_ fast tracks 3387 size_t fastTracks = 0; 3388 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3389 3390 float masterVolume = mMasterVolume; 3391 bool masterMute = mMasterMute; 3392 3393 if (masterMute) { 3394 masterVolume = 0; 3395 } 3396 // Delegate master volume control to effect in output mix effect chain if needed 3397 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3398 if (chain != 0) { 3399 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3400 chain->setVolume_l(&v, &v); 3401 masterVolume = (float)((v + (1 << 23)) >> 24); 3402 chain.clear(); 3403 } 3404 3405 // prepare a new state to push 3406 FastMixerStateQueue *sq = NULL; 3407 FastMixerState *state = NULL; 3408 bool didModify = false; 3409 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3410 if (mFastMixer != 0) { 3411 sq = mFastMixer->sq(); 3412 state = sq->begin(); 3413 } 3414 3415 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3416 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3417 3418 for (size_t i=0 ; i<count ; i++) { 3419 const sp<Track> t = mActiveTracks[i].promote(); 3420 if (t == 0) { 3421 continue; 3422 } 3423 3424 // this const just means the local variable doesn't change 3425 Track* const track = t.get(); 3426 3427 // process fast tracks 3428 if (track->isFastTrack()) { 3429 3430 // It's theoretically possible (though unlikely) for a fast track to be created 3431 // and then removed within the same normal mix cycle. This is not a problem, as 3432 // the track never becomes active so it's fast mixer slot is never touched. 3433 // The converse, of removing an (active) track and then creating a new track 3434 // at the identical fast mixer slot within the same normal mix cycle, 3435 // is impossible because the slot isn't marked available until the end of each cycle. 3436 int j = track->mFastIndex; 3437 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3438 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3439 FastTrack *fastTrack = &state->mFastTracks[j]; 3440 3441 // Determine whether the track is currently in underrun condition, 3442 // and whether it had a recent underrun. 3443 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3444 FastTrackUnderruns underruns = ftDump->mUnderruns; 3445 uint32_t recentFull = (underruns.mBitFields.mFull - 3446 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3447 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3448 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3449 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3450 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3451 uint32_t recentUnderruns = recentPartial + recentEmpty; 3452 track->mObservedUnderruns = underruns; 3453 // don't count underruns that occur while stopping or pausing 3454 // or stopped which can occur when flush() is called while active 3455 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3456 recentUnderruns > 0) { 3457 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3458 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3459 } 3460 3461 // This is similar to the state machine for normal tracks, 3462 // with a few modifications for fast tracks. 3463 bool isActive = true; 3464 switch (track->mState) { 3465 case TrackBase::STOPPING_1: 3466 // track stays active in STOPPING_1 state until first underrun 3467 if (recentUnderruns > 0 || track->isTerminated()) { 3468 track->mState = TrackBase::STOPPING_2; 3469 } 3470 break; 3471 case TrackBase::PAUSING: 3472 // ramp down is not yet implemented 3473 track->setPaused(); 3474 break; 3475 case TrackBase::RESUMING: 3476 // ramp up is not yet implemented 3477 track->mState = TrackBase::ACTIVE; 3478 break; 3479 case TrackBase::ACTIVE: 3480 if (recentFull > 0 || recentPartial > 0) { 3481 // track has provided at least some frames recently: reset retry count 3482 track->mRetryCount = kMaxTrackRetries; 3483 } 3484 if (recentUnderruns == 0) { 3485 // no recent underruns: stay active 3486 break; 3487 } 3488 // there has recently been an underrun of some kind 3489 if (track->sharedBuffer() == 0) { 3490 // were any of the recent underruns "empty" (no frames available)? 3491 if (recentEmpty == 0) { 3492 // no, then ignore the partial underruns as they are allowed indefinitely 3493 break; 3494 } 3495 // there has recently been an "empty" underrun: decrement the retry counter 3496 if (--(track->mRetryCount) > 0) { 3497 break; 3498 } 3499 // indicate to client process that the track was disabled because of underrun; 3500 // it will then automatically call start() when data is available 3501 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3502 // remove from active list, but state remains ACTIVE [confusing but true] 3503 isActive = false; 3504 break; 3505 } 3506 // fall through 3507 case TrackBase::STOPPING_2: 3508 case TrackBase::PAUSED: 3509 case TrackBase::STOPPED: 3510 case TrackBase::FLUSHED: // flush() while active 3511 // Check for presentation complete if track is inactive 3512 // We have consumed all the buffers of this track. 3513 // This would be incomplete if we auto-paused on underrun 3514 { 3515 size_t audioHALFrames = 3516 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3517 size_t framesWritten = mBytesWritten / mFrameSize; 3518 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3519 // track stays in active list until presentation is complete 3520 break; 3521 } 3522 } 3523 if (track->isStopping_2()) { 3524 track->mState = TrackBase::STOPPED; 3525 } 3526 if (track->isStopped()) { 3527 // Can't reset directly, as fast mixer is still polling this track 3528 // track->reset(); 3529 // So instead mark this track as needing to be reset after push with ack 3530 resetMask |= 1 << i; 3531 } 3532 isActive = false; 3533 break; 3534 case TrackBase::IDLE: 3535 default: 3536 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3537 } 3538 3539 if (isActive) { 3540 // was it previously inactive? 3541 if (!(state->mTrackMask & (1 << j))) { 3542 ExtendedAudioBufferProvider *eabp = track; 3543 VolumeProvider *vp = track; 3544 fastTrack->mBufferProvider = eabp; 3545 fastTrack->mVolumeProvider = vp; 3546 fastTrack->mChannelMask = track->mChannelMask; 3547 fastTrack->mFormat = track->mFormat; 3548 fastTrack->mGeneration++; 3549 state->mTrackMask |= 1 << j; 3550 didModify = true; 3551 // no acknowledgement required for newly active tracks 3552 } 3553 // cache the combined master volume and stream type volume for fast mixer; this 3554 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3555 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3556 ++fastTracks; 3557 } else { 3558 // was it previously active? 3559 if (state->mTrackMask & (1 << j)) { 3560 fastTrack->mBufferProvider = NULL; 3561 fastTrack->mGeneration++; 3562 state->mTrackMask &= ~(1 << j); 3563 didModify = true; 3564 // If any fast tracks were removed, we must wait for acknowledgement 3565 // because we're about to decrement the last sp<> on those tracks. 3566 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3567 } else { 3568 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3569 } 3570 tracksToRemove->add(track); 3571 // Avoids a misleading display in dumpsys 3572 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3573 } 3574 continue; 3575 } 3576 3577 { // local variable scope to avoid goto warning 3578 3579 audio_track_cblk_t* cblk = track->cblk(); 3580 3581 // The first time a track is added we wait 3582 // for all its buffers to be filled before processing it 3583 int name = track->name(); 3584 // make sure that we have enough frames to mix one full buffer. 3585 // enforce this condition only once to enable draining the buffer in case the client 3586 // app does not call stop() and relies on underrun to stop: 3587 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3588 // during last round 3589 size_t desiredFrames; 3590 uint32_t sr = track->sampleRate(); 3591 if (sr == mSampleRate) { 3592 desiredFrames = mNormalFrameCount; 3593 } else { 3594 desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); 3595 // add frames already consumed but not yet released by the resampler 3596 // because mAudioTrackServerProxy->framesReady() will include these frames 3597 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3598#if 0 3599 // the minimum track buffer size is normally twice the number of frames necessary 3600 // to fill one buffer and the resampler should not leave more than one buffer worth 3601 // of unreleased frames after each pass, but just in case... 3602 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3603#endif 3604 } 3605 uint32_t minFrames = 1; 3606 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3607 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3608 minFrames = desiredFrames; 3609 } 3610 3611 size_t framesReady = track->framesReady(); 3612 if (ATRACE_ENABLED()) { 3613 // I wish we had formatted trace names 3614 char traceName[16]; 3615 strcpy(traceName, "nRdy"); 3616 int name = track->name(); 3617 if (AudioMixer::TRACK0 <= name && 3618 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3619 name -= AudioMixer::TRACK0; 3620 traceName[4] = (name / 10) + '0'; 3621 traceName[5] = (name % 10) + '0'; 3622 } else { 3623 traceName[4] = '?'; 3624 traceName[5] = '?'; 3625 } 3626 traceName[6] = '\0'; 3627 ATRACE_INT(traceName, framesReady); 3628 } 3629 if ((framesReady >= minFrames) && track->isReady() && 3630 !track->isPaused() && !track->isTerminated()) 3631 { 3632 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3633 3634 mixedTracks++; 3635 3636 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3637 // there is an effect chain connected to the track 3638 chain.clear(); 3639 if (track->mainBuffer() != mSinkBuffer && 3640 track->mainBuffer() != mMixerBuffer) { 3641 if (mEffectBufferEnabled) { 3642 mEffectBufferValid = true; // Later can set directly. 3643 } 3644 chain = getEffectChain_l(track->sessionId()); 3645 // Delegate volume control to effect in track effect chain if needed 3646 if (chain != 0) { 3647 tracksWithEffect++; 3648 } else { 3649 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3650 "session %d", 3651 name, track->sessionId()); 3652 } 3653 } 3654 3655 3656 int param = AudioMixer::VOLUME; 3657 if (track->mFillingUpStatus == Track::FS_FILLED) { 3658 // no ramp for the first volume setting 3659 track->mFillingUpStatus = Track::FS_ACTIVE; 3660 if (track->mState == TrackBase::RESUMING) { 3661 track->mState = TrackBase::ACTIVE; 3662 param = AudioMixer::RAMP_VOLUME; 3663 } 3664 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3665 // FIXME should not make a decision based on mServer 3666 } else if (cblk->mServer != 0) { 3667 // If the track is stopped before the first frame was mixed, 3668 // do not apply ramp 3669 param = AudioMixer::RAMP_VOLUME; 3670 } 3671 3672 // compute volume for this track 3673 uint32_t vl, vr; // in U8.24 integer format 3674 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3675 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3676 vl = vr = 0; 3677 vlf = vrf = vaf = 0.; 3678 if (track->isPausing()) { 3679 track->setPaused(); 3680 } 3681 } else { 3682 3683 // read original volumes with volume control 3684 float typeVolume = mStreamTypes[track->streamType()].volume; 3685 float v = masterVolume * typeVolume; 3686 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3687 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3688 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3689 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3690 // track volumes come from shared memory, so can't be trusted and must be clamped 3691 if (vlf > GAIN_FLOAT_UNITY) { 3692 ALOGV("Track left volume out of range: %.3g", vlf); 3693 vlf = GAIN_FLOAT_UNITY; 3694 } 3695 if (vrf > GAIN_FLOAT_UNITY) { 3696 ALOGV("Track right volume out of range: %.3g", vrf); 3697 vrf = GAIN_FLOAT_UNITY; 3698 } 3699 // now apply the master volume and stream type volume 3700 vlf *= v; 3701 vrf *= v; 3702 // assuming master volume and stream type volume each go up to 1.0, 3703 // then derive vl and vr as U8.24 versions for the effect chain 3704 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3705 vl = (uint32_t) (scaleto8_24 * vlf); 3706 vr = (uint32_t) (scaleto8_24 * vrf); 3707 // vl and vr are now in U8.24 format 3708 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3709 // send level comes from shared memory and so may be corrupt 3710 if (sendLevel > MAX_GAIN_INT) { 3711 ALOGV("Track send level out of range: %04X", sendLevel); 3712 sendLevel = MAX_GAIN_INT; 3713 } 3714 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3715 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3716 } 3717 3718 // Delegate volume control to effect in track effect chain if needed 3719 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3720 // Do not ramp volume if volume is controlled by effect 3721 param = AudioMixer::VOLUME; 3722 // Update remaining floating point volume levels 3723 vlf = (float)vl / (1 << 24); 3724 vrf = (float)vr / (1 << 24); 3725 track->mHasVolumeController = true; 3726 } else { 3727 // force no volume ramp when volume controller was just disabled or removed 3728 // from effect chain to avoid volume spike 3729 if (track->mHasVolumeController) { 3730 param = AudioMixer::VOLUME; 3731 } 3732 track->mHasVolumeController = false; 3733 } 3734 3735 // XXX: these things DON'T need to be done each time 3736 mAudioMixer->setBufferProvider(name, track); 3737 mAudioMixer->enable(name); 3738 3739 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3740 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3741 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3742 mAudioMixer->setParameter( 3743 name, 3744 AudioMixer::TRACK, 3745 AudioMixer::FORMAT, (void *)track->format()); 3746 mAudioMixer->setParameter( 3747 name, 3748 AudioMixer::TRACK, 3749 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3750 mAudioMixer->setParameter( 3751 name, 3752 AudioMixer::TRACK, 3753 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3754 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3755 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3756 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3757 if (reqSampleRate == 0) { 3758 reqSampleRate = mSampleRate; 3759 } else if (reqSampleRate > maxSampleRate) { 3760 reqSampleRate = maxSampleRate; 3761 } 3762 mAudioMixer->setParameter( 3763 name, 3764 AudioMixer::RESAMPLE, 3765 AudioMixer::SAMPLE_RATE, 3766 (void *)(uintptr_t)reqSampleRate); 3767 /* 3768 * Select the appropriate output buffer for the track. 3769 * 3770 * Tracks with effects go into their own effects chain buffer 3771 * and from there into either mEffectBuffer or mSinkBuffer. 3772 * 3773 * Other tracks can use mMixerBuffer for higher precision 3774 * channel accumulation. If this buffer is enabled 3775 * (mMixerBufferEnabled true), then selected tracks will accumulate 3776 * into it. 3777 * 3778 */ 3779 if (mMixerBufferEnabled 3780 && (track->mainBuffer() == mSinkBuffer 3781 || track->mainBuffer() == mMixerBuffer)) { 3782 mAudioMixer->setParameter( 3783 name, 3784 AudioMixer::TRACK, 3785 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3786 mAudioMixer->setParameter( 3787 name, 3788 AudioMixer::TRACK, 3789 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3790 // TODO: override track->mainBuffer()? 3791 mMixerBufferValid = true; 3792 } else { 3793 mAudioMixer->setParameter( 3794 name, 3795 AudioMixer::TRACK, 3796 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3797 mAudioMixer->setParameter( 3798 name, 3799 AudioMixer::TRACK, 3800 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3801 } 3802 mAudioMixer->setParameter( 3803 name, 3804 AudioMixer::TRACK, 3805 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3806 3807 // reset retry count 3808 track->mRetryCount = kMaxTrackRetries; 3809 3810 // If one track is ready, set the mixer ready if: 3811 // - the mixer was not ready during previous round OR 3812 // - no other track is not ready 3813 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3814 mixerStatus != MIXER_TRACKS_ENABLED) { 3815 mixerStatus = MIXER_TRACKS_READY; 3816 } 3817 } else { 3818 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3819 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3820 } 3821 // clear effect chain input buffer if an active track underruns to avoid sending 3822 // previous audio buffer again to effects 3823 chain = getEffectChain_l(track->sessionId()); 3824 if (chain != 0) { 3825 chain->clearInputBuffer(); 3826 } 3827 3828 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3829 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3830 track->isStopped() || track->isPaused()) { 3831 // We have consumed all the buffers of this track. 3832 // Remove it from the list of active tracks. 3833 // TODO: use actual buffer filling status instead of latency when available from 3834 // audio HAL 3835 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3836 size_t framesWritten = mBytesWritten / mFrameSize; 3837 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3838 if (track->isStopped()) { 3839 track->reset(); 3840 } 3841 tracksToRemove->add(track); 3842 } 3843 } else { 3844 // No buffers for this track. Give it a few chances to 3845 // fill a buffer, then remove it from active list. 3846 if (--(track->mRetryCount) <= 0) { 3847 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3848 tracksToRemove->add(track); 3849 // indicate to client process that the track was disabled because of underrun; 3850 // it will then automatically call start() when data is available 3851 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3852 // If one track is not ready, mark the mixer also not ready if: 3853 // - the mixer was ready during previous round OR 3854 // - no other track is ready 3855 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3856 mixerStatus != MIXER_TRACKS_READY) { 3857 mixerStatus = MIXER_TRACKS_ENABLED; 3858 } 3859 } 3860 mAudioMixer->disable(name); 3861 } 3862 3863 } // local variable scope to avoid goto warning 3864track_is_ready: ; 3865 3866 } 3867 3868 // Push the new FastMixer state if necessary 3869 bool pauseAudioWatchdog = false; 3870 if (didModify) { 3871 state->mFastTracksGen++; 3872 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3873 if (kUseFastMixer == FastMixer_Dynamic && 3874 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3875 state->mCommand = FastMixerState::COLD_IDLE; 3876 state->mColdFutexAddr = &mFastMixerFutex; 3877 state->mColdGen++; 3878 mFastMixerFutex = 0; 3879 if (kUseFastMixer == FastMixer_Dynamic) { 3880 mNormalSink = mOutputSink; 3881 } 3882 // If we go into cold idle, need to wait for acknowledgement 3883 // so that fast mixer stops doing I/O. 3884 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3885 pauseAudioWatchdog = true; 3886 } 3887 } 3888 if (sq != NULL) { 3889 sq->end(didModify); 3890 sq->push(block); 3891 } 3892#ifdef AUDIO_WATCHDOG 3893 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3894 mAudioWatchdog->pause(); 3895 } 3896#endif 3897 3898 // Now perform the deferred reset on fast tracks that have stopped 3899 while (resetMask != 0) { 3900 size_t i = __builtin_ctz(resetMask); 3901 ALOG_ASSERT(i < count); 3902 resetMask &= ~(1 << i); 3903 sp<Track> t = mActiveTracks[i].promote(); 3904 if (t == 0) { 3905 continue; 3906 } 3907 Track* track = t.get(); 3908 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3909 track->reset(); 3910 } 3911 3912 // remove all the tracks that need to be... 3913 removeTracks_l(*tracksToRemove); 3914 3915 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3916 mEffectBufferValid = true; 3917 } 3918 3919 if (mEffectBufferValid) { 3920 // as long as there are effects we should clear the effects buffer, to avoid 3921 // passing a non-clean buffer to the effect chain 3922 memset(mEffectBuffer, 0, mEffectBufferSize); 3923 } 3924 // sink or mix buffer must be cleared if all tracks are connected to an 3925 // effect chain as in this case the mixer will not write to the sink or mix buffer 3926 // and track effects will accumulate into it 3927 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3928 (mixedTracks == 0 && fastTracks > 0))) { 3929 // FIXME as a performance optimization, should remember previous zero status 3930 if (mMixerBufferValid) { 3931 memset(mMixerBuffer, 0, mMixerBufferSize); 3932 // TODO: In testing, mSinkBuffer below need not be cleared because 3933 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3934 // after mixing. 3935 // 3936 // To enforce this guarantee: 3937 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3938 // (mixedTracks == 0 && fastTracks > 0)) 3939 // must imply MIXER_TRACKS_READY. 3940 // Later, we may clear buffers regardless, and skip much of this logic. 3941 } 3942 // FIXME as a performance optimization, should remember previous zero status 3943 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3944 } 3945 3946 // if any fast tracks, then status is ready 3947 mMixerStatusIgnoringFastTracks = mixerStatus; 3948 if (fastTracks > 0) { 3949 mixerStatus = MIXER_TRACKS_READY; 3950 } 3951 return mixerStatus; 3952} 3953 3954// getTrackName_l() must be called with ThreadBase::mLock held 3955int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3956 audio_format_t format, int sessionId) 3957{ 3958 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3959} 3960 3961// deleteTrackName_l() must be called with ThreadBase::mLock held 3962void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3963{ 3964 ALOGV("remove track (%d) and delete from mixer", name); 3965 mAudioMixer->deleteTrackName(name); 3966} 3967 3968// checkForNewParameter_l() must be called with ThreadBase::mLock held 3969bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3970 status_t& status) 3971{ 3972 bool reconfig = false; 3973 3974 status = NO_ERROR; 3975 3976 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3977 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3978 if (mFastMixer != 0) { 3979 FastMixerStateQueue *sq = mFastMixer->sq(); 3980 FastMixerState *state = sq->begin(); 3981 if (!(state->mCommand & FastMixerState::IDLE)) { 3982 previousCommand = state->mCommand; 3983 state->mCommand = FastMixerState::HOT_IDLE; 3984 sq->end(); 3985 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3986 } else { 3987 sq->end(false /*didModify*/); 3988 } 3989 } 3990 3991 AudioParameter param = AudioParameter(keyValuePair); 3992 int value; 3993 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3994 reconfig = true; 3995 } 3996 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3997 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3998 status = BAD_VALUE; 3999 } else { 4000 // no need to save value, since it's constant 4001 reconfig = true; 4002 } 4003 } 4004 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4005 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4006 status = BAD_VALUE; 4007 } else { 4008 // no need to save value, since it's constant 4009 reconfig = true; 4010 } 4011 } 4012 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4013 // do not accept frame count changes if tracks are open as the track buffer 4014 // size depends on frame count and correct behavior would not be guaranteed 4015 // if frame count is changed after track creation 4016 if (!mTracks.isEmpty()) { 4017 status = INVALID_OPERATION; 4018 } else { 4019 reconfig = true; 4020 } 4021 } 4022 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4023#ifdef ADD_BATTERY_DATA 4024 // when changing the audio output device, call addBatteryData to notify 4025 // the change 4026 if (mOutDevice != value) { 4027 uint32_t params = 0; 4028 // check whether speaker is on 4029 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4030 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4031 } 4032 4033 audio_devices_t deviceWithoutSpeaker 4034 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4035 // check if any other device (except speaker) is on 4036 if (value & deviceWithoutSpeaker ) { 4037 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4038 } 4039 4040 if (params != 0) { 4041 addBatteryData(params); 4042 } 4043 } 4044#endif 4045 4046 // forward device change to effects that have requested to be 4047 // aware of attached audio device. 4048 if (value != AUDIO_DEVICE_NONE) { 4049 mOutDevice = value; 4050 for (size_t i = 0; i < mEffectChains.size(); i++) { 4051 mEffectChains[i]->setDevice_l(mOutDevice); 4052 } 4053 } 4054 } 4055 4056 if (status == NO_ERROR) { 4057 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4058 keyValuePair.string()); 4059 if (!mStandby && status == INVALID_OPERATION) { 4060 mOutput->stream->common.standby(&mOutput->stream->common); 4061 mStandby = true; 4062 mBytesWritten = 0; 4063 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4064 keyValuePair.string()); 4065 } 4066 if (status == NO_ERROR && reconfig) { 4067 readOutputParameters_l(); 4068 delete mAudioMixer; 4069 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4070 for (size_t i = 0; i < mTracks.size() ; i++) { 4071 int name = getTrackName_l(mTracks[i]->mChannelMask, 4072 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4073 if (name < 0) { 4074 break; 4075 } 4076 mTracks[i]->mName = name; 4077 } 4078 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4079 } 4080 } 4081 4082 if (!(previousCommand & FastMixerState::IDLE)) { 4083 ALOG_ASSERT(mFastMixer != 0); 4084 FastMixerStateQueue *sq = mFastMixer->sq(); 4085 FastMixerState *state = sq->begin(); 4086 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4087 state->mCommand = previousCommand; 4088 sq->end(); 4089 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4090 } 4091 4092 return reconfig; 4093} 4094 4095 4096void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4097{ 4098 const size_t SIZE = 256; 4099 char buffer[SIZE]; 4100 String8 result; 4101 4102 PlaybackThread::dumpInternals(fd, args); 4103 4104 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4105 4106 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4107 const FastMixerDumpState copy(mFastMixerDumpState); 4108 copy.dump(fd); 4109 4110#ifdef STATE_QUEUE_DUMP 4111 // Similar for state queue 4112 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4113 observerCopy.dump(fd); 4114 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4115 mutatorCopy.dump(fd); 4116#endif 4117 4118#ifdef TEE_SINK 4119 // Write the tee output to a .wav file 4120 dumpTee(fd, mTeeSource, mId); 4121#endif 4122 4123#ifdef AUDIO_WATCHDOG 4124 if (mAudioWatchdog != 0) { 4125 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4126 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4127 wdCopy.dump(fd); 4128 } 4129#endif 4130} 4131 4132uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4133{ 4134 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4135} 4136 4137uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4138{ 4139 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4140} 4141 4142void AudioFlinger::MixerThread::cacheParameters_l() 4143{ 4144 PlaybackThread::cacheParameters_l(); 4145 4146 // FIXME: Relaxed timing because of a certain device that can't meet latency 4147 // Should be reduced to 2x after the vendor fixes the driver issue 4148 // increase threshold again due to low power audio mode. The way this warning 4149 // threshold is calculated and its usefulness should be reconsidered anyway. 4150 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4151} 4152 4153// ---------------------------------------------------------------------------- 4154 4155AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4156 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4157 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4158 // mLeftVolFloat, mRightVolFloat 4159{ 4160} 4161 4162AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4163 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4164 ThreadBase::type_t type) 4165 : PlaybackThread(audioFlinger, output, id, device, type) 4166 // mLeftVolFloat, mRightVolFloat 4167{ 4168} 4169 4170AudioFlinger::DirectOutputThread::~DirectOutputThread() 4171{ 4172} 4173 4174void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4175{ 4176 audio_track_cblk_t* cblk = track->cblk(); 4177 float left, right; 4178 4179 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4180 left = right = 0; 4181 } else { 4182 float typeVolume = mStreamTypes[track->streamType()].volume; 4183 float v = mMasterVolume * typeVolume; 4184 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4185 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4186 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4187 if (left > GAIN_FLOAT_UNITY) { 4188 left = GAIN_FLOAT_UNITY; 4189 } 4190 left *= v; 4191 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4192 if (right > GAIN_FLOAT_UNITY) { 4193 right = GAIN_FLOAT_UNITY; 4194 } 4195 right *= v; 4196 } 4197 4198 if (lastTrack) { 4199 if (left != mLeftVolFloat || right != mRightVolFloat) { 4200 mLeftVolFloat = left; 4201 mRightVolFloat = right; 4202 4203 // Convert volumes from float to 8.24 4204 uint32_t vl = (uint32_t)(left * (1 << 24)); 4205 uint32_t vr = (uint32_t)(right * (1 << 24)); 4206 4207 // Delegate volume control to effect in track effect chain if needed 4208 // only one effect chain can be present on DirectOutputThread, so if 4209 // there is one, the track is connected to it 4210 if (!mEffectChains.isEmpty()) { 4211 mEffectChains[0]->setVolume_l(&vl, &vr); 4212 left = (float)vl / (1 << 24); 4213 right = (float)vr / (1 << 24); 4214 } 4215 if (mOutput->stream->set_volume) { 4216 mOutput->stream->set_volume(mOutput->stream, left, right); 4217 } 4218 } 4219 } 4220} 4221 4222 4223AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4224 Vector< sp<Track> > *tracksToRemove 4225) 4226{ 4227 size_t count = mActiveTracks.size(); 4228 mixer_state mixerStatus = MIXER_IDLE; 4229 bool doHwPause = false; 4230 bool doHwResume = false; 4231 bool flushPending = false; 4232 4233 // find out which tracks need to be processed 4234 for (size_t i = 0; i < count; i++) { 4235 sp<Track> t = mActiveTracks[i].promote(); 4236 // The track died recently 4237 if (t == 0) { 4238 continue; 4239 } 4240 4241 Track* const track = t.get(); 4242 audio_track_cblk_t* cblk = track->cblk(); 4243 // Only consider last track started for volume and mixer state control. 4244 // In theory an older track could underrun and restart after the new one starts 4245 // but as we only care about the transition phase between two tracks on a 4246 // direct output, it is not a problem to ignore the underrun case. 4247 sp<Track> l = mLatestActiveTrack.promote(); 4248 bool last = l.get() == track; 4249 4250 if (mHwSupportsPause && track->isPausing()) { 4251 track->setPaused(); 4252 if (last && !mHwPaused) { 4253 doHwPause = true; 4254 mHwPaused = true; 4255 } 4256 tracksToRemove->add(track); 4257 } else if (track->isFlushPending()) { 4258 track->flushAck(); 4259 if (last) { 4260 flushPending = true; 4261 } 4262 } else if (mHwSupportsPause && track->isResumePending()){ 4263 track->resumeAck(); 4264 if (last) { 4265 if (mHwPaused) { 4266 doHwResume = true; 4267 mHwPaused = false; 4268 } 4269 } 4270 } 4271 4272 // The first time a track is added we wait 4273 // for all its buffers to be filled before processing it. 4274 // Allow draining the buffer in case the client 4275 // app does not call stop() and relies on underrun to stop: 4276 // hence the test on (track->mRetryCount > 1). 4277 // If retryCount<=1 then track is about to underrun and be removed. 4278 uint32_t minFrames; 4279 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4280 && (track->mRetryCount > 1)) { 4281 minFrames = mNormalFrameCount; 4282 } else { 4283 minFrames = 1; 4284 } 4285 4286 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4287 !track->isStopping_2() && !track->isStopped()) 4288 { 4289 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4290 4291 if (track->mFillingUpStatus == Track::FS_FILLED) { 4292 track->mFillingUpStatus = Track::FS_ACTIVE; 4293 // make sure processVolume_l() will apply new volume even if 0 4294 mLeftVolFloat = mRightVolFloat = -1.0; 4295 if (!mHwSupportsPause) { 4296 track->resumeAck(); 4297 } 4298 } 4299 4300 // compute volume for this track 4301 processVolume_l(track, last); 4302 if (last) { 4303 // reset retry count 4304 track->mRetryCount = kMaxTrackRetriesDirect; 4305 mActiveTrack = t; 4306 mixerStatus = MIXER_TRACKS_READY; 4307 if (usesHwAvSync() && mHwPaused) { 4308 doHwResume = true; 4309 mHwPaused = false; 4310 } 4311 } 4312 } else { 4313 // clear effect chain input buffer if the last active track started underruns 4314 // to avoid sending previous audio buffer again to effects 4315 if (!mEffectChains.isEmpty() && last) { 4316 mEffectChains[0]->clearInputBuffer(); 4317 } 4318 if (track->isStopping_1()) { 4319 track->mState = TrackBase::STOPPING_2; 4320 } 4321 if ((track->sharedBuffer() != 0) || track->isStopped() || 4322 track->isStopping_2() || track->isPaused()) { 4323 // We have consumed all the buffers of this track. 4324 // Remove it from the list of active tracks. 4325 size_t audioHALFrames; 4326 if (audio_is_linear_pcm(mFormat)) { 4327 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4328 } else { 4329 audioHALFrames = 0; 4330 } 4331 4332 size_t framesWritten = mBytesWritten / mFrameSize; 4333 if (mStandby || !last || 4334 track->presentationComplete(framesWritten, audioHALFrames)) { 4335 if (track->isStopping_2()) { 4336 track->mState = TrackBase::STOPPED; 4337 } 4338 if (track->isStopped()) { 4339 track->reset(); 4340 } 4341 tracksToRemove->add(track); 4342 } 4343 } else { 4344 // No buffers for this track. Give it a few chances to 4345 // fill a buffer, then remove it from active list. 4346 // Only consider last track started for mixer state control 4347 if (--(track->mRetryCount) <= 0) { 4348 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4349 tracksToRemove->add(track); 4350 // indicate to client process that the track was disabled because of underrun; 4351 // it will then automatically call start() when data is available 4352 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4353 } else if (last) { 4354 mixerStatus = MIXER_TRACKS_ENABLED; 4355 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4356 doHwPause = true; 4357 mHwPaused = true; 4358 } 4359 } 4360 } 4361 } 4362 } 4363 4364 // if an active track did not command a flush, check for pending flush on stopped tracks 4365 if (!flushPending) { 4366 for (size_t i = 0; i < mTracks.size(); i++) { 4367 if (mTracks[i]->isFlushPending()) { 4368 mTracks[i]->flushAck(); 4369 flushPending = true; 4370 } 4371 } 4372 } 4373 4374 // make sure the pause/flush/resume sequence is executed in the right order. 4375 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4376 // before flush and then resume HW. This can happen in case of pause/flush/resume 4377 // if resume is received before pause is executed. 4378 if (mHwSupportsPause && !mStandby && 4379 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4380 mOutput->stream->pause(mOutput->stream); 4381 } 4382 if (flushPending) { 4383 flushHw_l(); 4384 } 4385 if (mHwSupportsPause && !mStandby && doHwResume) { 4386 mOutput->stream->resume(mOutput->stream); 4387 } 4388 // remove all the tracks that need to be... 4389 removeTracks_l(*tracksToRemove); 4390 4391 return mixerStatus; 4392} 4393 4394void AudioFlinger::DirectOutputThread::threadLoop_mix() 4395{ 4396 size_t frameCount = mFrameCount; 4397 int8_t *curBuf = (int8_t *)mSinkBuffer; 4398 // output audio to hardware 4399 while (frameCount) { 4400 AudioBufferProvider::Buffer buffer; 4401 buffer.frameCount = frameCount; 4402 mActiveTrack->getNextBuffer(&buffer); 4403 if (buffer.raw == NULL) { 4404 memset(curBuf, 0, frameCount * mFrameSize); 4405 break; 4406 } 4407 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4408 frameCount -= buffer.frameCount; 4409 curBuf += buffer.frameCount * mFrameSize; 4410 mActiveTrack->releaseBuffer(&buffer); 4411 } 4412 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4413 sleepTime = 0; 4414 standbyTime = systemTime() + standbyDelay; 4415 mActiveTrack.clear(); 4416} 4417 4418void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4419{ 4420 // do not write to HAL when paused 4421 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4422 sleepTime = idleSleepTime; 4423 return; 4424 } 4425 if (sleepTime == 0) { 4426 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4427 sleepTime = activeSleepTime; 4428 } else { 4429 sleepTime = idleSleepTime; 4430 } 4431 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4432 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4433 sleepTime = 0; 4434 } 4435} 4436 4437void AudioFlinger::DirectOutputThread::threadLoop_exit() 4438{ 4439 { 4440 Mutex::Autolock _l(mLock); 4441 bool flushPending = false; 4442 for (size_t i = 0; i < mTracks.size(); i++) { 4443 if (mTracks[i]->isFlushPending()) { 4444 mTracks[i]->flushAck(); 4445 flushPending = true; 4446 } 4447 } 4448 if (flushPending) { 4449 flushHw_l(); 4450 } 4451 } 4452 PlaybackThread::threadLoop_exit(); 4453} 4454 4455// must be called with thread mutex locked 4456bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4457{ 4458 bool trackPaused = false; 4459 4460 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4461 // after a timeout and we will enter standby then. 4462 if (mTracks.size() > 0) { 4463 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4464 } 4465 4466 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused)); 4467} 4468 4469// getTrackName_l() must be called with ThreadBase::mLock held 4470int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4471 audio_format_t format __unused, int sessionId __unused) 4472{ 4473 return 0; 4474} 4475 4476// deleteTrackName_l() must be called with ThreadBase::mLock held 4477void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4478{ 4479} 4480 4481// checkForNewParameter_l() must be called with ThreadBase::mLock held 4482bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4483 status_t& status) 4484{ 4485 bool reconfig = false; 4486 4487 status = NO_ERROR; 4488 4489 AudioParameter param = AudioParameter(keyValuePair); 4490 int value; 4491 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4492 // forward device change to effects that have requested to be 4493 // aware of attached audio device. 4494 if (value != AUDIO_DEVICE_NONE) { 4495 mOutDevice = value; 4496 for (size_t i = 0; i < mEffectChains.size(); i++) { 4497 mEffectChains[i]->setDevice_l(mOutDevice); 4498 } 4499 } 4500 } 4501 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4502 // do not accept frame count changes if tracks are open as the track buffer 4503 // size depends on frame count and correct behavior would not be garantied 4504 // if frame count is changed after track creation 4505 if (!mTracks.isEmpty()) { 4506 status = INVALID_OPERATION; 4507 } else { 4508 reconfig = true; 4509 } 4510 } 4511 if (status == NO_ERROR) { 4512 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4513 keyValuePair.string()); 4514 if (!mStandby && status == INVALID_OPERATION) { 4515 mOutput->stream->common.standby(&mOutput->stream->common); 4516 mStandby = true; 4517 mBytesWritten = 0; 4518 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4519 keyValuePair.string()); 4520 } 4521 if (status == NO_ERROR && reconfig) { 4522 readOutputParameters_l(); 4523 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4524 } 4525 } 4526 4527 return reconfig; 4528} 4529 4530uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4531{ 4532 uint32_t time; 4533 if (audio_is_linear_pcm(mFormat)) { 4534 time = PlaybackThread::activeSleepTimeUs(); 4535 } else { 4536 time = 10000; 4537 } 4538 return time; 4539} 4540 4541uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4542{ 4543 uint32_t time; 4544 if (audio_is_linear_pcm(mFormat)) { 4545 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4546 } else { 4547 time = 10000; 4548 } 4549 return time; 4550} 4551 4552uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4553{ 4554 uint32_t time; 4555 if (audio_is_linear_pcm(mFormat)) { 4556 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4557 } else { 4558 time = 10000; 4559 } 4560 return time; 4561} 4562 4563void AudioFlinger::DirectOutputThread::cacheParameters_l() 4564{ 4565 PlaybackThread::cacheParameters_l(); 4566 4567 // use shorter standby delay as on normal output to release 4568 // hardware resources as soon as possible 4569 if (audio_is_linear_pcm(mFormat)) { 4570 standbyDelay = microseconds(activeSleepTime*2); 4571 } else { 4572 standbyDelay = kOffloadStandbyDelayNs; 4573 } 4574} 4575 4576void AudioFlinger::DirectOutputThread::flushHw_l() 4577{ 4578 if (mOutput->stream->flush != NULL) { 4579 mOutput->stream->flush(mOutput->stream); 4580 } 4581 mHwPaused = false; 4582} 4583 4584// ---------------------------------------------------------------------------- 4585 4586AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4587 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4588 : Thread(false /*canCallJava*/), 4589 mPlaybackThread(playbackThread), 4590 mWriteAckSequence(0), 4591 mDrainSequence(0) 4592{ 4593} 4594 4595AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4596{ 4597} 4598 4599void AudioFlinger::AsyncCallbackThread::onFirstRef() 4600{ 4601 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4602} 4603 4604bool AudioFlinger::AsyncCallbackThread::threadLoop() 4605{ 4606 while (!exitPending()) { 4607 uint32_t writeAckSequence; 4608 uint32_t drainSequence; 4609 4610 { 4611 Mutex::Autolock _l(mLock); 4612 while (!((mWriteAckSequence & 1) || 4613 (mDrainSequence & 1) || 4614 exitPending())) { 4615 mWaitWorkCV.wait(mLock); 4616 } 4617 4618 if (exitPending()) { 4619 break; 4620 } 4621 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4622 mWriteAckSequence, mDrainSequence); 4623 writeAckSequence = mWriteAckSequence; 4624 mWriteAckSequence &= ~1; 4625 drainSequence = mDrainSequence; 4626 mDrainSequence &= ~1; 4627 } 4628 { 4629 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4630 if (playbackThread != 0) { 4631 if (writeAckSequence & 1) { 4632 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4633 } 4634 if (drainSequence & 1) { 4635 playbackThread->resetDraining(drainSequence >> 1); 4636 } 4637 } 4638 } 4639 } 4640 return false; 4641} 4642 4643void AudioFlinger::AsyncCallbackThread::exit() 4644{ 4645 ALOGV("AsyncCallbackThread::exit"); 4646 Mutex::Autolock _l(mLock); 4647 requestExit(); 4648 mWaitWorkCV.broadcast(); 4649} 4650 4651void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4652{ 4653 Mutex::Autolock _l(mLock); 4654 // bit 0 is cleared 4655 mWriteAckSequence = sequence << 1; 4656} 4657 4658void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4659{ 4660 Mutex::Autolock _l(mLock); 4661 // ignore unexpected callbacks 4662 if (mWriteAckSequence & 2) { 4663 mWriteAckSequence |= 1; 4664 mWaitWorkCV.signal(); 4665 } 4666} 4667 4668void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4669{ 4670 Mutex::Autolock _l(mLock); 4671 // bit 0 is cleared 4672 mDrainSequence = sequence << 1; 4673} 4674 4675void AudioFlinger::AsyncCallbackThread::resetDraining() 4676{ 4677 Mutex::Autolock _l(mLock); 4678 // ignore unexpected callbacks 4679 if (mDrainSequence & 2) { 4680 mDrainSequence |= 1; 4681 mWaitWorkCV.signal(); 4682 } 4683} 4684 4685 4686// ---------------------------------------------------------------------------- 4687AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4688 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4689 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4690 mPausedBytesRemaining(0) 4691{ 4692 //FIXME: mStandby should be set to true by ThreadBase constructor 4693 mStandby = true; 4694} 4695 4696void AudioFlinger::OffloadThread::threadLoop_exit() 4697{ 4698 if (mFlushPending || mHwPaused) { 4699 // If a flush is pending or track was paused, just discard buffered data 4700 flushHw_l(); 4701 } else { 4702 mMixerStatus = MIXER_DRAIN_ALL; 4703 threadLoop_drain(); 4704 } 4705 if (mUseAsyncWrite) { 4706 ALOG_ASSERT(mCallbackThread != 0); 4707 mCallbackThread->exit(); 4708 } 4709 PlaybackThread::threadLoop_exit(); 4710} 4711 4712AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4713 Vector< sp<Track> > *tracksToRemove 4714) 4715{ 4716 size_t count = mActiveTracks.size(); 4717 4718 mixer_state mixerStatus = MIXER_IDLE; 4719 bool doHwPause = false; 4720 bool doHwResume = false; 4721 4722 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4723 4724 // find out which tracks need to be processed 4725 for (size_t i = 0; i < count; i++) { 4726 sp<Track> t = mActiveTracks[i].promote(); 4727 // The track died recently 4728 if (t == 0) { 4729 continue; 4730 } 4731 Track* const track = t.get(); 4732 audio_track_cblk_t* cblk = track->cblk(); 4733 // Only consider last track started for volume and mixer state control. 4734 // In theory an older track could underrun and restart after the new one starts 4735 // but as we only care about the transition phase between two tracks on a 4736 // direct output, it is not a problem to ignore the underrun case. 4737 sp<Track> l = mLatestActiveTrack.promote(); 4738 bool last = l.get() == track; 4739 4740 if (track->isInvalid()) { 4741 ALOGW("An invalidated track shouldn't be in active list"); 4742 tracksToRemove->add(track); 4743 continue; 4744 } 4745 4746 if (track->mState == TrackBase::IDLE) { 4747 ALOGW("An idle track shouldn't be in active list"); 4748 continue; 4749 } 4750 4751 if (track->isPausing()) { 4752 track->setPaused(); 4753 if (last) { 4754 if (!mHwPaused) { 4755 doHwPause = true; 4756 mHwPaused = true; 4757 } 4758 // If we were part way through writing the mixbuffer to 4759 // the HAL we must save this until we resume 4760 // BUG - this will be wrong if a different track is made active, 4761 // in that case we want to discard the pending data in the 4762 // mixbuffer and tell the client to present it again when the 4763 // track is resumed 4764 mPausedWriteLength = mCurrentWriteLength; 4765 mPausedBytesRemaining = mBytesRemaining; 4766 mBytesRemaining = 0; // stop writing 4767 } 4768 tracksToRemove->add(track); 4769 } else if (track->isFlushPending()) { 4770 track->flushAck(); 4771 if (last) { 4772 mFlushPending = true; 4773 } 4774 } else if (track->isResumePending()){ 4775 track->resumeAck(); 4776 if (last) { 4777 if (mPausedBytesRemaining) { 4778 // Need to continue write that was interrupted 4779 mCurrentWriteLength = mPausedWriteLength; 4780 mBytesRemaining = mPausedBytesRemaining; 4781 mPausedBytesRemaining = 0; 4782 } 4783 if (mHwPaused) { 4784 doHwResume = true; 4785 mHwPaused = false; 4786 // threadLoop_mix() will handle the case that we need to 4787 // resume an interrupted write 4788 } 4789 // enable write to audio HAL 4790 sleepTime = 0; 4791 4792 // Do not handle new data in this iteration even if track->framesReady() 4793 mixerStatus = MIXER_TRACKS_ENABLED; 4794 } 4795 } else if (track->framesReady() && track->isReady() && 4796 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4797 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4798 if (track->mFillingUpStatus == Track::FS_FILLED) { 4799 track->mFillingUpStatus = Track::FS_ACTIVE; 4800 // make sure processVolume_l() will apply new volume even if 0 4801 mLeftVolFloat = mRightVolFloat = -1.0; 4802 } 4803 4804 if (last) { 4805 sp<Track> previousTrack = mPreviousTrack.promote(); 4806 if (previousTrack != 0) { 4807 if (track != previousTrack.get()) { 4808 // Flush any data still being written from last track 4809 mBytesRemaining = 0; 4810 if (mPausedBytesRemaining) { 4811 // Last track was paused so we also need to flush saved 4812 // mixbuffer state and invalidate track so that it will 4813 // re-submit that unwritten data when it is next resumed 4814 mPausedBytesRemaining = 0; 4815 // Invalidate is a bit drastic - would be more efficient 4816 // to have a flag to tell client that some of the 4817 // previously written data was lost 4818 previousTrack->invalidate(); 4819 } 4820 // flush data already sent to the DSP if changing audio session as audio 4821 // comes from a different source. Also invalidate previous track to force a 4822 // seek when resuming. 4823 if (previousTrack->sessionId() != track->sessionId()) { 4824 previousTrack->invalidate(); 4825 } 4826 } 4827 } 4828 mPreviousTrack = track; 4829 // reset retry count 4830 track->mRetryCount = kMaxTrackRetriesOffload; 4831 mActiveTrack = t; 4832 mixerStatus = MIXER_TRACKS_READY; 4833 } 4834 } else { 4835 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4836 if (track->isStopping_1()) { 4837 // Hardware buffer can hold a large amount of audio so we must 4838 // wait for all current track's data to drain before we say 4839 // that the track is stopped. 4840 if (mBytesRemaining == 0) { 4841 // Only start draining when all data in mixbuffer 4842 // has been written 4843 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4844 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4845 // do not drain if no data was ever sent to HAL (mStandby == true) 4846 if (last && !mStandby) { 4847 // do not modify drain sequence if we are already draining. This happens 4848 // when resuming from pause after drain. 4849 if ((mDrainSequence & 1) == 0) { 4850 sleepTime = 0; 4851 standbyTime = systemTime() + standbyDelay; 4852 mixerStatus = MIXER_DRAIN_TRACK; 4853 mDrainSequence += 2; 4854 } 4855 if (mHwPaused) { 4856 // It is possible to move from PAUSED to STOPPING_1 without 4857 // a resume so we must ensure hardware is running 4858 doHwResume = true; 4859 mHwPaused = false; 4860 } 4861 } 4862 } 4863 } else if (track->isStopping_2()) { 4864 // Drain has completed or we are in standby, signal presentation complete 4865 if (!(mDrainSequence & 1) || !last || mStandby) { 4866 track->mState = TrackBase::STOPPED; 4867 size_t audioHALFrames = 4868 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4869 size_t framesWritten = 4870 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4871 track->presentationComplete(framesWritten, audioHALFrames); 4872 track->reset(); 4873 tracksToRemove->add(track); 4874 } 4875 } else { 4876 // No buffers for this track. Give it a few chances to 4877 // fill a buffer, then remove it from active list. 4878 if (--(track->mRetryCount) <= 0) { 4879 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4880 track->name()); 4881 tracksToRemove->add(track); 4882 // indicate to client process that the track was disabled because of underrun; 4883 // it will then automatically call start() when data is available 4884 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4885 } else if (last){ 4886 mixerStatus = MIXER_TRACKS_ENABLED; 4887 } 4888 } 4889 } 4890 // compute volume for this track 4891 processVolume_l(track, last); 4892 } 4893 4894 // make sure the pause/flush/resume sequence is executed in the right order. 4895 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4896 // before flush and then resume HW. This can happen in case of pause/flush/resume 4897 // if resume is received before pause is executed. 4898 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4899 mOutput->stream->pause(mOutput->stream); 4900 } 4901 if (mFlushPending) { 4902 flushHw_l(); 4903 mFlushPending = false; 4904 } 4905 if (!mStandby && doHwResume) { 4906 mOutput->stream->resume(mOutput->stream); 4907 } 4908 4909 // remove all the tracks that need to be... 4910 removeTracks_l(*tracksToRemove); 4911 4912 return mixerStatus; 4913} 4914 4915// must be called with thread mutex locked 4916bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4917{ 4918 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4919 mWriteAckSequence, mDrainSequence); 4920 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4921 return true; 4922 } 4923 return false; 4924} 4925 4926bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4927{ 4928 Mutex::Autolock _l(mLock); 4929 return waitingAsyncCallback_l(); 4930} 4931 4932void AudioFlinger::OffloadThread::flushHw_l() 4933{ 4934 DirectOutputThread::flushHw_l(); 4935 // Flush anything still waiting in the mixbuffer 4936 mCurrentWriteLength = 0; 4937 mBytesRemaining = 0; 4938 mPausedWriteLength = 0; 4939 mPausedBytesRemaining = 0; 4940 4941 if (mUseAsyncWrite) { 4942 // discard any pending drain or write ack by incrementing sequence 4943 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4944 mDrainSequence = (mDrainSequence + 2) & ~1; 4945 ALOG_ASSERT(mCallbackThread != 0); 4946 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4947 mCallbackThread->setDraining(mDrainSequence); 4948 } 4949} 4950 4951void AudioFlinger::OffloadThread::onAddNewTrack_l() 4952{ 4953 sp<Track> previousTrack = mPreviousTrack.promote(); 4954 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4955 4956 if (previousTrack != 0 && latestTrack != 0 && 4957 (previousTrack->sessionId() != latestTrack->sessionId())) { 4958 mFlushPending = true; 4959 } 4960 PlaybackThread::onAddNewTrack_l(); 4961} 4962 4963// ---------------------------------------------------------------------------- 4964 4965AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4966 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4967 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4968 DUPLICATING), 4969 mWaitTimeMs(UINT_MAX) 4970{ 4971 addOutputTrack(mainThread); 4972} 4973 4974AudioFlinger::DuplicatingThread::~DuplicatingThread() 4975{ 4976 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4977 mOutputTracks[i]->destroy(); 4978 } 4979} 4980 4981void AudioFlinger::DuplicatingThread::threadLoop_mix() 4982{ 4983 // mix buffers... 4984 if (outputsReady(outputTracks)) { 4985 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4986 } else { 4987 if (mMixerBufferValid) { 4988 memset(mMixerBuffer, 0, mMixerBufferSize); 4989 } else { 4990 memset(mSinkBuffer, 0, mSinkBufferSize); 4991 } 4992 } 4993 sleepTime = 0; 4994 writeFrames = mNormalFrameCount; 4995 mCurrentWriteLength = mSinkBufferSize; 4996 standbyTime = systemTime() + standbyDelay; 4997} 4998 4999void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5000{ 5001 if (sleepTime == 0) { 5002 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5003 sleepTime = activeSleepTime; 5004 } else { 5005 sleepTime = idleSleepTime; 5006 } 5007 } else if (mBytesWritten != 0) { 5008 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5009 writeFrames = mNormalFrameCount; 5010 memset(mSinkBuffer, 0, mSinkBufferSize); 5011 } else { 5012 // flush remaining overflow buffers in output tracks 5013 writeFrames = 0; 5014 } 5015 sleepTime = 0; 5016 } 5017} 5018 5019ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5020{ 5021 for (size_t i = 0; i < outputTracks.size(); i++) { 5022 outputTracks[i]->write(mSinkBuffer, writeFrames); 5023 } 5024 mStandby = false; 5025 return (ssize_t)mSinkBufferSize; 5026} 5027 5028void AudioFlinger::DuplicatingThread::threadLoop_standby() 5029{ 5030 // DuplicatingThread implements standby by stopping all tracks 5031 for (size_t i = 0; i < outputTracks.size(); i++) { 5032 outputTracks[i]->stop(); 5033 } 5034} 5035 5036void AudioFlinger::DuplicatingThread::saveOutputTracks() 5037{ 5038 outputTracks = mOutputTracks; 5039} 5040 5041void AudioFlinger::DuplicatingThread::clearOutputTracks() 5042{ 5043 outputTracks.clear(); 5044} 5045 5046void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5047{ 5048 Mutex::Autolock _l(mLock); 5049 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5050 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5051 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5052 const size_t frameCount = 5053 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5054 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5055 // from different OutputTracks and their associated MixerThreads (e.g. one may 5056 // nearly empty and the other may be dropping data). 5057 5058 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5059 this, 5060 mSampleRate, 5061 mFormat, 5062 mChannelMask, 5063 frameCount, 5064 IPCThreadState::self()->getCallingUid()); 5065 if (outputTrack->cblk() != NULL) { 5066 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5067 mOutputTracks.add(outputTrack); 5068 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5069 updateWaitTime_l(); 5070 } 5071} 5072 5073void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5074{ 5075 Mutex::Autolock _l(mLock); 5076 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5077 if (mOutputTracks[i]->thread() == thread) { 5078 mOutputTracks[i]->destroy(); 5079 mOutputTracks.removeAt(i); 5080 updateWaitTime_l(); 5081 return; 5082 } 5083 } 5084 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5085} 5086 5087// caller must hold mLock 5088void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5089{ 5090 mWaitTimeMs = UINT_MAX; 5091 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5092 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5093 if (strong != 0) { 5094 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5095 if (waitTimeMs < mWaitTimeMs) { 5096 mWaitTimeMs = waitTimeMs; 5097 } 5098 } 5099 } 5100} 5101 5102 5103bool AudioFlinger::DuplicatingThread::outputsReady( 5104 const SortedVector< sp<OutputTrack> > &outputTracks) 5105{ 5106 for (size_t i = 0; i < outputTracks.size(); i++) { 5107 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5108 if (thread == 0) { 5109 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5110 outputTracks[i].get()); 5111 return false; 5112 } 5113 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5114 // see note at standby() declaration 5115 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5116 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5117 thread.get()); 5118 return false; 5119 } 5120 } 5121 return true; 5122} 5123 5124uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5125{ 5126 return (mWaitTimeMs * 1000) / 2; 5127} 5128 5129void AudioFlinger::DuplicatingThread::cacheParameters_l() 5130{ 5131 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5132 updateWaitTime_l(); 5133 5134 MixerThread::cacheParameters_l(); 5135} 5136 5137// ---------------------------------------------------------------------------- 5138// Record 5139// ---------------------------------------------------------------------------- 5140 5141AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5142 AudioStreamIn *input, 5143 audio_io_handle_t id, 5144 audio_devices_t outDevice, 5145 audio_devices_t inDevice 5146#ifdef TEE_SINK 5147 , const sp<NBAIO_Sink>& teeSink 5148#endif 5149 ) : 5150 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5151 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5152 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5153 mRsmpInRear(0) 5154#ifdef TEE_SINK 5155 , mTeeSink(teeSink) 5156#endif 5157 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5158 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5159 // mFastCapture below 5160 , mFastCaptureFutex(0) 5161 // mInputSource 5162 // mPipeSink 5163 // mPipeSource 5164 , mPipeFramesP2(0) 5165 // mPipeMemory 5166 // mFastCaptureNBLogWriter 5167 , mFastTrackAvail(false) 5168{ 5169 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5170 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5171 5172 readInputParameters_l(); 5173 5174 // create an NBAIO source for the HAL input stream, and negotiate 5175 mInputSource = new AudioStreamInSource(input->stream); 5176 size_t numCounterOffers = 0; 5177 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5178 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5179 ALOG_ASSERT(index == 0); 5180 5181 // initialize fast capture depending on configuration 5182 bool initFastCapture; 5183 switch (kUseFastCapture) { 5184 case FastCapture_Never: 5185 initFastCapture = false; 5186 break; 5187 case FastCapture_Always: 5188 initFastCapture = true; 5189 break; 5190 case FastCapture_Static: 5191 uint32_t primaryOutputSampleRate; 5192 { 5193 AutoMutex _l(audioFlinger->mHardwareLock); 5194 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5195 } 5196 initFastCapture = 5197 // either capture sample rate is same as (a reasonable) primary output sample rate 5198 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5199 (mSampleRate == primaryOutputSampleRate)) || 5200 // or primary output sample rate is unknown, and capture sample rate is reasonable 5201 ((primaryOutputSampleRate == 0) && 5202 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5203 // and the buffer size is < 12 ms 5204 (mFrameCount * 1000) / mSampleRate < 12; 5205 break; 5206 // case FastCapture_Dynamic: 5207 } 5208 5209 if (initFastCapture) { 5210 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 5211 NBAIO_Format format = mInputSource->format(); 5212 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5213 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5214 void *pipeBuffer; 5215 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5216 sp<IMemory> pipeMemory; 5217 if ((roHeap == 0) || 5218 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5219 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5220 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5221 goto failed; 5222 } 5223 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5224 memset(pipeBuffer, 0, pipeSize); 5225 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5226 const NBAIO_Format offers[1] = {format}; 5227 size_t numCounterOffers = 0; 5228 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5229 ALOG_ASSERT(index == 0); 5230 mPipeSink = pipe; 5231 PipeReader *pipeReader = new PipeReader(*pipe); 5232 numCounterOffers = 0; 5233 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5234 ALOG_ASSERT(index == 0); 5235 mPipeSource = pipeReader; 5236 mPipeFramesP2 = pipeFramesP2; 5237 mPipeMemory = pipeMemory; 5238 5239 // create fast capture 5240 mFastCapture = new FastCapture(); 5241 FastCaptureStateQueue *sq = mFastCapture->sq(); 5242#ifdef STATE_QUEUE_DUMP 5243 // FIXME 5244#endif 5245 FastCaptureState *state = sq->begin(); 5246 state->mCblk = NULL; 5247 state->mInputSource = mInputSource.get(); 5248 state->mInputSourceGen++; 5249 state->mPipeSink = pipe; 5250 state->mPipeSinkGen++; 5251 state->mFrameCount = mFrameCount; 5252 state->mCommand = FastCaptureState::COLD_IDLE; 5253 // already done in constructor initialization list 5254 //mFastCaptureFutex = 0; 5255 state->mColdFutexAddr = &mFastCaptureFutex; 5256 state->mColdGen++; 5257 state->mDumpState = &mFastCaptureDumpState; 5258#ifdef TEE_SINK 5259 // FIXME 5260#endif 5261 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5262 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5263 sq->end(); 5264 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5265 5266 // start the fast capture 5267 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5268 pid_t tid = mFastCapture->getTid(); 5269 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5270 if (err != 0) { 5271 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5272 kPriorityFastCapture, getpid_cached, tid, err); 5273 } 5274 5275#ifdef AUDIO_WATCHDOG 5276 // FIXME 5277#endif 5278 5279 mFastTrackAvail = true; 5280 } 5281failed: ; 5282 5283 // FIXME mNormalSource 5284} 5285 5286 5287AudioFlinger::RecordThread::~RecordThread() 5288{ 5289 if (mFastCapture != 0) { 5290 FastCaptureStateQueue *sq = mFastCapture->sq(); 5291 FastCaptureState *state = sq->begin(); 5292 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5293 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5294 if (old == -1) { 5295 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5296 } 5297 } 5298 state->mCommand = FastCaptureState::EXIT; 5299 sq->end(); 5300 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5301 mFastCapture->join(); 5302 mFastCapture.clear(); 5303 } 5304 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5305 mAudioFlinger->unregisterWriter(mNBLogWriter); 5306 delete[] mRsmpInBuffer; 5307} 5308 5309void AudioFlinger::RecordThread::onFirstRef() 5310{ 5311 run(mThreadName, PRIORITY_URGENT_AUDIO); 5312} 5313 5314bool AudioFlinger::RecordThread::threadLoop() 5315{ 5316 nsecs_t lastWarning = 0; 5317 5318 inputStandBy(); 5319 5320reacquire_wakelock: 5321 sp<RecordTrack> activeTrack; 5322 int activeTracksGen; 5323 { 5324 Mutex::Autolock _l(mLock); 5325 size_t size = mActiveTracks.size(); 5326 activeTracksGen = mActiveTracksGen; 5327 if (size > 0) { 5328 // FIXME an arbitrary choice 5329 activeTrack = mActiveTracks[0]; 5330 acquireWakeLock_l(activeTrack->uid()); 5331 if (size > 1) { 5332 SortedVector<int> tmp; 5333 for (size_t i = 0; i < size; i++) { 5334 tmp.add(mActiveTracks[i]->uid()); 5335 } 5336 updateWakeLockUids_l(tmp); 5337 } 5338 } else { 5339 acquireWakeLock_l(-1); 5340 } 5341 } 5342 5343 // used to request a deferred sleep, to be executed later while mutex is unlocked 5344 uint32_t sleepUs = 0; 5345 5346 // loop while there is work to do 5347 for (;;) { 5348 Vector< sp<EffectChain> > effectChains; 5349 5350 // sleep with mutex unlocked 5351 if (sleepUs > 0) { 5352 ATRACE_BEGIN("sleep"); 5353 usleep(sleepUs); 5354 ATRACE_END(); 5355 sleepUs = 0; 5356 } 5357 5358 // activeTracks accumulates a copy of a subset of mActiveTracks 5359 Vector< sp<RecordTrack> > activeTracks; 5360 5361 // reference to the (first and only) active fast track 5362 sp<RecordTrack> fastTrack; 5363 5364 // reference to a fast track which is about to be removed 5365 sp<RecordTrack> fastTrackToRemove; 5366 5367 { // scope for mLock 5368 Mutex::Autolock _l(mLock); 5369 5370 processConfigEvents_l(); 5371 5372 // check exitPending here because checkForNewParameters_l() and 5373 // checkForNewParameters_l() can temporarily release mLock 5374 if (exitPending()) { 5375 break; 5376 } 5377 5378 // if no active track(s), then standby and release wakelock 5379 size_t size = mActiveTracks.size(); 5380 if (size == 0) { 5381 standbyIfNotAlreadyInStandby(); 5382 // exitPending() can't become true here 5383 releaseWakeLock_l(); 5384 ALOGV("RecordThread: loop stopping"); 5385 // go to sleep 5386 mWaitWorkCV.wait(mLock); 5387 ALOGV("RecordThread: loop starting"); 5388 goto reacquire_wakelock; 5389 } 5390 5391 if (mActiveTracksGen != activeTracksGen) { 5392 activeTracksGen = mActiveTracksGen; 5393 SortedVector<int> tmp; 5394 for (size_t i = 0; i < size; i++) { 5395 tmp.add(mActiveTracks[i]->uid()); 5396 } 5397 updateWakeLockUids_l(tmp); 5398 } 5399 5400 bool doBroadcast = false; 5401 for (size_t i = 0; i < size; ) { 5402 5403 activeTrack = mActiveTracks[i]; 5404 if (activeTrack->isTerminated()) { 5405 if (activeTrack->isFastTrack()) { 5406 ALOG_ASSERT(fastTrackToRemove == 0); 5407 fastTrackToRemove = activeTrack; 5408 } 5409 removeTrack_l(activeTrack); 5410 mActiveTracks.remove(activeTrack); 5411 mActiveTracksGen++; 5412 size--; 5413 continue; 5414 } 5415 5416 TrackBase::track_state activeTrackState = activeTrack->mState; 5417 switch (activeTrackState) { 5418 5419 case TrackBase::PAUSING: 5420 mActiveTracks.remove(activeTrack); 5421 mActiveTracksGen++; 5422 doBroadcast = true; 5423 size--; 5424 continue; 5425 5426 case TrackBase::STARTING_1: 5427 sleepUs = 10000; 5428 i++; 5429 continue; 5430 5431 case TrackBase::STARTING_2: 5432 doBroadcast = true; 5433 mStandby = false; 5434 activeTrack->mState = TrackBase::ACTIVE; 5435 break; 5436 5437 case TrackBase::ACTIVE: 5438 break; 5439 5440 case TrackBase::IDLE: 5441 i++; 5442 continue; 5443 5444 default: 5445 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5446 } 5447 5448 activeTracks.add(activeTrack); 5449 i++; 5450 5451 if (activeTrack->isFastTrack()) { 5452 ALOG_ASSERT(!mFastTrackAvail); 5453 ALOG_ASSERT(fastTrack == 0); 5454 fastTrack = activeTrack; 5455 } 5456 } 5457 if (doBroadcast) { 5458 mStartStopCond.broadcast(); 5459 } 5460 5461 // sleep if there are no active tracks to process 5462 if (activeTracks.size() == 0) { 5463 if (sleepUs == 0) { 5464 sleepUs = kRecordThreadSleepUs; 5465 } 5466 continue; 5467 } 5468 sleepUs = 0; 5469 5470 lockEffectChains_l(effectChains); 5471 } 5472 5473 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5474 5475 size_t size = effectChains.size(); 5476 for (size_t i = 0; i < size; i++) { 5477 // thread mutex is not locked, but effect chain is locked 5478 effectChains[i]->process_l(); 5479 } 5480 5481 // Push a new fast capture state if fast capture is not already running, or cblk change 5482 if (mFastCapture != 0) { 5483 FastCaptureStateQueue *sq = mFastCapture->sq(); 5484 FastCaptureState *state = sq->begin(); 5485 bool didModify = false; 5486 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5487 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5488 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5489 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5490 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5491 if (old == -1) { 5492 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5493 } 5494 } 5495 state->mCommand = FastCaptureState::READ_WRITE; 5496#if 0 // FIXME 5497 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5498 FastThreadDumpState::kSamplingNforLowRamDevice : 5499 FastThreadDumpState::kSamplingN); 5500#endif 5501 didModify = true; 5502 } 5503 audio_track_cblk_t *cblkOld = state->mCblk; 5504 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5505 if (cblkNew != cblkOld) { 5506 state->mCblk = cblkNew; 5507 // block until acked if removing a fast track 5508 if (cblkOld != NULL) { 5509 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5510 } 5511 didModify = true; 5512 } 5513 sq->end(didModify); 5514 if (didModify) { 5515 sq->push(block); 5516#if 0 5517 if (kUseFastCapture == FastCapture_Dynamic) { 5518 mNormalSource = mPipeSource; 5519 } 5520#endif 5521 } 5522 } 5523 5524 // now run the fast track destructor with thread mutex unlocked 5525 fastTrackToRemove.clear(); 5526 5527 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5528 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5529 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5530 // If destination is non-contiguous, first read past the nominal end of buffer, then 5531 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5532 5533 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5534 ssize_t framesRead; 5535 5536 // If an NBAIO source is present, use it to read the normal capture's data 5537 if (mPipeSource != 0) { 5538 size_t framesToRead = mBufferSize / mFrameSize; 5539 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5540 framesToRead, AudioBufferProvider::kInvalidPTS); 5541 if (framesRead == 0) { 5542 // since pipe is non-blocking, simulate blocking input 5543 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5544 } 5545 // otherwise use the HAL / AudioStreamIn directly 5546 } else { 5547 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5548 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5549 if (bytesRead < 0) { 5550 framesRead = bytesRead; 5551 } else { 5552 framesRead = bytesRead / mFrameSize; 5553 } 5554 } 5555 5556 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5557 ALOGE("read failed: framesRead=%d", framesRead); 5558 // Force input into standby so that it tries to recover at next read attempt 5559 inputStandBy(); 5560 sleepUs = kRecordThreadSleepUs; 5561 } 5562 if (framesRead <= 0) { 5563 goto unlock; 5564 } 5565 ALOG_ASSERT(framesRead > 0); 5566 5567 if (mTeeSink != 0) { 5568 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5569 } 5570 // If destination is non-contiguous, we now correct for reading past end of buffer. 5571 { 5572 size_t part1 = mRsmpInFramesP2 - rear; 5573 if ((size_t) framesRead > part1) { 5574 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5575 (framesRead - part1) * mFrameSize); 5576 } 5577 } 5578 rear = mRsmpInRear += framesRead; 5579 5580 size = activeTracks.size(); 5581 // loop over each active track 5582 for (size_t i = 0; i < size; i++) { 5583 activeTrack = activeTracks[i]; 5584 5585 // skip fast tracks, as those are handled directly by FastCapture 5586 if (activeTrack->isFastTrack()) { 5587 continue; 5588 } 5589 5590 enum { 5591 OVERRUN_UNKNOWN, 5592 OVERRUN_TRUE, 5593 OVERRUN_FALSE 5594 } overrun = OVERRUN_UNKNOWN; 5595 5596 // loop over getNextBuffer to handle circular sink 5597 for (;;) { 5598 5599 activeTrack->mSink.frameCount = ~0; 5600 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5601 size_t framesOut = activeTrack->mSink.frameCount; 5602 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5603 5604 int32_t front = activeTrack->mRsmpInFront; 5605 ssize_t filled = rear - front; 5606 size_t framesIn; 5607 5608 if (filled < 0) { 5609 // should not happen, but treat like a massive overrun and re-sync 5610 framesIn = 0; 5611 activeTrack->mRsmpInFront = rear; 5612 overrun = OVERRUN_TRUE; 5613 } else if ((size_t) filled <= mRsmpInFrames) { 5614 framesIn = (size_t) filled; 5615 } else { 5616 // client is not keeping up with server, but give it latest data 5617 framesIn = mRsmpInFrames; 5618 activeTrack->mRsmpInFront = front = rear - framesIn; 5619 overrun = OVERRUN_TRUE; 5620 } 5621 5622 if (framesOut == 0 || framesIn == 0) { 5623 break; 5624 } 5625 5626 if (activeTrack->mResampler == NULL) { 5627 // no resampling 5628 if (framesIn > framesOut) { 5629 framesIn = framesOut; 5630 } else { 5631 framesOut = framesIn; 5632 } 5633 int8_t *dst = activeTrack->mSink.i8; 5634 while (framesIn > 0) { 5635 front &= mRsmpInFramesP2 - 1; 5636 size_t part1 = mRsmpInFramesP2 - front; 5637 if (part1 > framesIn) { 5638 part1 = framesIn; 5639 } 5640 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5641 if (mChannelCount == activeTrack->mChannelCount) { 5642 memcpy(dst, src, part1 * mFrameSize); 5643 } else if (mChannelCount == 1) { 5644 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5645 part1); 5646 } else { 5647 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 5648 (const int16_t *)src, part1); 5649 } 5650 dst += part1 * activeTrack->mFrameSize; 5651 front += part1; 5652 framesIn -= part1; 5653 } 5654 activeTrack->mRsmpInFront += framesOut; 5655 5656 } else { 5657 // resampling 5658 // FIXME framesInNeeded should really be part of resampler API, and should 5659 // depend on the SRC ratio 5660 // to keep mRsmpInBuffer full so resampler always has sufficient input 5661 size_t framesInNeeded; 5662 // FIXME only re-calculate when it changes, and optimize for common ratios 5663 // Do not precompute in/out because floating point is not associative 5664 // e.g. a*b/c != a*(b/c). 5665 const double in(mSampleRate); 5666 const double out(activeTrack->mSampleRate); 5667 framesInNeeded = ceil(framesOut * in / out) + 1; 5668 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5669 framesInNeeded, framesOut, in / out); 5670 // Although we theoretically have framesIn in circular buffer, some of those are 5671 // unreleased frames, and thus must be discounted for purpose of budgeting. 5672 size_t unreleased = activeTrack->mRsmpInUnrel; 5673 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5674 if (framesIn < framesInNeeded) { 5675 ALOGV("not enough to resample: have %u frames in but need %u in to " 5676 "produce %u out given in/out ratio of %.4g", 5677 framesIn, framesInNeeded, framesOut, in / out); 5678 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5679 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5680 if (newFramesOut == 0) { 5681 break; 5682 } 5683 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5684 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5685 framesInNeeded, newFramesOut, out / in); 5686 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5687 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5688 "given in/out ratio of %.4g", 5689 framesIn, framesInNeeded, newFramesOut, in / out); 5690 framesOut = newFramesOut; 5691 } else { 5692 ALOGV("success 1: have %u in and need %u in to produce %u out " 5693 "given in/out ratio of %.4g", 5694 framesIn, framesInNeeded, framesOut, in / out); 5695 } 5696 5697 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5698 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5699 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5700 delete[] activeTrack->mRsmpOutBuffer; 5701 // resampler always outputs stereo 5702 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5703 activeTrack->mRsmpOutFrameCount = framesOut; 5704 } 5705 5706 // resampler accumulates, but we only have one source track 5707 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5708 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5709 // FIXME how about having activeTrack implement this interface itself? 5710 activeTrack->mResamplerBufferProvider 5711 /*this*/ /* AudioBufferProvider* */); 5712 // ditherAndClamp() works as long as all buffers returned by 5713 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5714 if (activeTrack->mChannelCount == 1) { 5715 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5716 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5717 framesOut); 5718 // the resampler always outputs stereo samples: 5719 // do post stereo to mono conversion 5720 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5721 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5722 } else { 5723 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5724 activeTrack->mRsmpOutBuffer, framesOut); 5725 } 5726 // now done with mRsmpOutBuffer 5727 5728 } 5729 5730 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5731 overrun = OVERRUN_FALSE; 5732 } 5733 5734 if (activeTrack->mFramesToDrop == 0) { 5735 if (framesOut > 0) { 5736 activeTrack->mSink.frameCount = framesOut; 5737 activeTrack->releaseBuffer(&activeTrack->mSink); 5738 } 5739 } else { 5740 // FIXME could do a partial drop of framesOut 5741 if (activeTrack->mFramesToDrop > 0) { 5742 activeTrack->mFramesToDrop -= framesOut; 5743 if (activeTrack->mFramesToDrop <= 0) { 5744 activeTrack->clearSyncStartEvent(); 5745 } 5746 } else { 5747 activeTrack->mFramesToDrop += framesOut; 5748 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5749 activeTrack->mSyncStartEvent->isCancelled()) { 5750 ALOGW("Synced record %s, session %d, trigger session %d", 5751 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5752 activeTrack->sessionId(), 5753 (activeTrack->mSyncStartEvent != 0) ? 5754 activeTrack->mSyncStartEvent->triggerSession() : 0); 5755 activeTrack->clearSyncStartEvent(); 5756 } 5757 } 5758 } 5759 5760 if (framesOut == 0) { 5761 break; 5762 } 5763 } 5764 5765 switch (overrun) { 5766 case OVERRUN_TRUE: 5767 // client isn't retrieving buffers fast enough 5768 if (!activeTrack->setOverflow()) { 5769 nsecs_t now = systemTime(); 5770 // FIXME should lastWarning per track? 5771 if ((now - lastWarning) > kWarningThrottleNs) { 5772 ALOGW("RecordThread: buffer overflow"); 5773 lastWarning = now; 5774 } 5775 } 5776 break; 5777 case OVERRUN_FALSE: 5778 activeTrack->clearOverflow(); 5779 break; 5780 case OVERRUN_UNKNOWN: 5781 break; 5782 } 5783 5784 } 5785 5786unlock: 5787 // enable changes in effect chain 5788 unlockEffectChains(effectChains); 5789 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5790 } 5791 5792 standbyIfNotAlreadyInStandby(); 5793 5794 { 5795 Mutex::Autolock _l(mLock); 5796 for (size_t i = 0; i < mTracks.size(); i++) { 5797 sp<RecordTrack> track = mTracks[i]; 5798 track->invalidate(); 5799 } 5800 mActiveTracks.clear(); 5801 mActiveTracksGen++; 5802 mStartStopCond.broadcast(); 5803 } 5804 5805 releaseWakeLock(); 5806 5807 ALOGV("RecordThread %p exiting", this); 5808 return false; 5809} 5810 5811void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5812{ 5813 if (!mStandby) { 5814 inputStandBy(); 5815 mStandby = true; 5816 } 5817} 5818 5819void AudioFlinger::RecordThread::inputStandBy() 5820{ 5821 // Idle the fast capture if it's currently running 5822 if (mFastCapture != 0) { 5823 FastCaptureStateQueue *sq = mFastCapture->sq(); 5824 FastCaptureState *state = sq->begin(); 5825 if (!(state->mCommand & FastCaptureState::IDLE)) { 5826 state->mCommand = FastCaptureState::COLD_IDLE; 5827 state->mColdFutexAddr = &mFastCaptureFutex; 5828 state->mColdGen++; 5829 mFastCaptureFutex = 0; 5830 sq->end(); 5831 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5832 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5833#if 0 5834 if (kUseFastCapture == FastCapture_Dynamic) { 5835 // FIXME 5836 } 5837#endif 5838#ifdef AUDIO_WATCHDOG 5839 // FIXME 5840#endif 5841 } else { 5842 sq->end(false /*didModify*/); 5843 } 5844 } 5845 mInput->stream->common.standby(&mInput->stream->common); 5846} 5847 5848// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5849sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5850 const sp<AudioFlinger::Client>& client, 5851 uint32_t sampleRate, 5852 audio_format_t format, 5853 audio_channel_mask_t channelMask, 5854 size_t *pFrameCount, 5855 int sessionId, 5856 size_t *notificationFrames, 5857 int uid, 5858 IAudioFlinger::track_flags_t *flags, 5859 pid_t tid, 5860 status_t *status) 5861{ 5862 size_t frameCount = *pFrameCount; 5863 sp<RecordTrack> track; 5864 status_t lStatus; 5865 5866 // client expresses a preference for FAST, but we get the final say 5867 if (*flags & IAudioFlinger::TRACK_FAST) { 5868 if ( 5869 // use case: callback handler 5870 (tid != -1) && 5871 // frame count is not specified, or is exactly the pipe depth 5872 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5873 // PCM data 5874 audio_is_linear_pcm(format) && 5875 // native format 5876 (format == mFormat) && 5877 // native channel mask 5878 (channelMask == mChannelMask) && 5879 // native hardware sample rate 5880 (sampleRate == mSampleRate) && 5881 // record thread has an associated fast capture 5882 hasFastCapture() && 5883 // there are sufficient fast track slots available 5884 mFastTrackAvail 5885 ) { 5886 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5887 frameCount, mFrameCount); 5888 } else { 5889 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5890 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5891 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5892 frameCount, mFrameCount, mPipeFramesP2, 5893 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5894 hasFastCapture(), tid, mFastTrackAvail); 5895 *flags &= ~IAudioFlinger::TRACK_FAST; 5896 } 5897 } 5898 5899 // compute track buffer size in frames, and suggest the notification frame count 5900 if (*flags & IAudioFlinger::TRACK_FAST) { 5901 // fast track: frame count is exactly the pipe depth 5902 frameCount = mPipeFramesP2; 5903 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5904 *notificationFrames = mFrameCount; 5905 } else { 5906 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5907 // or 20 ms if there is a fast capture 5908 // TODO This could be a roundupRatio inline, and const 5909 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5910 * sampleRate + mSampleRate - 1) / mSampleRate; 5911 // minimum number of notification periods is at least kMinNotifications, 5912 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5913 static const size_t kMinNotifications = 3; 5914 static const uint32_t kMinMs = 30; 5915 // TODO This could be a roundupRatio inline 5916 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5917 // TODO This could be a roundupRatio inline 5918 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5919 maxNotificationFrames; 5920 const size_t minFrameCount = maxNotificationFrames * 5921 max(kMinNotifications, minNotificationsByMs); 5922 frameCount = max(frameCount, minFrameCount); 5923 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5924 *notificationFrames = maxNotificationFrames; 5925 } 5926 } 5927 *pFrameCount = frameCount; 5928 5929 lStatus = initCheck(); 5930 if (lStatus != NO_ERROR) { 5931 ALOGE("createRecordTrack_l() audio driver not initialized"); 5932 goto Exit; 5933 } 5934 5935 { // scope for mLock 5936 Mutex::Autolock _l(mLock); 5937 5938 track = new RecordTrack(this, client, sampleRate, 5939 format, channelMask, frameCount, NULL, sessionId, uid, 5940 *flags, TrackBase::TYPE_DEFAULT); 5941 5942 lStatus = track->initCheck(); 5943 if (lStatus != NO_ERROR) { 5944 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5945 // track must be cleared from the caller as the caller has the AF lock 5946 goto Exit; 5947 } 5948 mTracks.add(track); 5949 5950 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5951 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5952 mAudioFlinger->btNrecIsOff(); 5953 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5954 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5955 5956 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5957 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5958 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5959 // so ask activity manager to do this on our behalf 5960 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5961 } 5962 } 5963 5964 lStatus = NO_ERROR; 5965 5966Exit: 5967 *status = lStatus; 5968 return track; 5969} 5970 5971status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5972 AudioSystem::sync_event_t event, 5973 int triggerSession) 5974{ 5975 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5976 sp<ThreadBase> strongMe = this; 5977 status_t status = NO_ERROR; 5978 5979 if (event == AudioSystem::SYNC_EVENT_NONE) { 5980 recordTrack->clearSyncStartEvent(); 5981 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5982 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5983 triggerSession, 5984 recordTrack->sessionId(), 5985 syncStartEventCallback, 5986 recordTrack); 5987 // Sync event can be cancelled by the trigger session if the track is not in a 5988 // compatible state in which case we start record immediately 5989 if (recordTrack->mSyncStartEvent->isCancelled()) { 5990 recordTrack->clearSyncStartEvent(); 5991 } else { 5992 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5993 recordTrack->mFramesToDrop = - 5994 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5995 } 5996 } 5997 5998 { 5999 // This section is a rendezvous between binder thread executing start() and RecordThread 6000 AutoMutex lock(mLock); 6001 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6002 if (recordTrack->mState == TrackBase::PAUSING) { 6003 ALOGV("active record track PAUSING -> ACTIVE"); 6004 recordTrack->mState = TrackBase::ACTIVE; 6005 } else { 6006 ALOGV("active record track state %d", recordTrack->mState); 6007 } 6008 return status; 6009 } 6010 6011 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6012 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6013 // or using a separate command thread 6014 recordTrack->mState = TrackBase::STARTING_1; 6015 mActiveTracks.add(recordTrack); 6016 mActiveTracksGen++; 6017 status_t status = NO_ERROR; 6018 if (recordTrack->isExternalTrack()) { 6019 mLock.unlock(); 6020 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6021 mLock.lock(); 6022 // FIXME should verify that recordTrack is still in mActiveTracks 6023 if (status != NO_ERROR) { 6024 mActiveTracks.remove(recordTrack); 6025 mActiveTracksGen++; 6026 recordTrack->clearSyncStartEvent(); 6027 ALOGV("RecordThread::start error %d", status); 6028 return status; 6029 } 6030 } 6031 // Catch up with current buffer indices if thread is already running. 6032 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6033 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6034 // see previously buffered data before it called start(), but with greater risk of overrun. 6035 6036 recordTrack->mRsmpInFront = mRsmpInRear; 6037 recordTrack->mRsmpInUnrel = 0; 6038 // FIXME why reset? 6039 if (recordTrack->mResampler != NULL) { 6040 recordTrack->mResampler->reset(); 6041 } 6042 recordTrack->mState = TrackBase::STARTING_2; 6043 // signal thread to start 6044 mWaitWorkCV.broadcast(); 6045 if (mActiveTracks.indexOf(recordTrack) < 0) { 6046 ALOGV("Record failed to start"); 6047 status = BAD_VALUE; 6048 goto startError; 6049 } 6050 return status; 6051 } 6052 6053startError: 6054 if (recordTrack->isExternalTrack()) { 6055 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6056 } 6057 recordTrack->clearSyncStartEvent(); 6058 // FIXME I wonder why we do not reset the state here? 6059 return status; 6060} 6061 6062void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6063{ 6064 sp<SyncEvent> strongEvent = event.promote(); 6065 6066 if (strongEvent != 0) { 6067 sp<RefBase> ptr = strongEvent->cookie().promote(); 6068 if (ptr != 0) { 6069 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6070 recordTrack->handleSyncStartEvent(strongEvent); 6071 } 6072 } 6073} 6074 6075bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6076 ALOGV("RecordThread::stop"); 6077 AutoMutex _l(mLock); 6078 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6079 return false; 6080 } 6081 // note that threadLoop may still be processing the track at this point [without lock] 6082 recordTrack->mState = TrackBase::PAUSING; 6083 // do not wait for mStartStopCond if exiting 6084 if (exitPending()) { 6085 return true; 6086 } 6087 // FIXME incorrect usage of wait: no explicit predicate or loop 6088 mStartStopCond.wait(mLock); 6089 // if we have been restarted, recordTrack is in mActiveTracks here 6090 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6091 ALOGV("Record stopped OK"); 6092 return true; 6093 } 6094 return false; 6095} 6096 6097bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6098{ 6099 return false; 6100} 6101 6102status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6103{ 6104#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6105 if (!isValidSyncEvent(event)) { 6106 return BAD_VALUE; 6107 } 6108 6109 int eventSession = event->triggerSession(); 6110 status_t ret = NAME_NOT_FOUND; 6111 6112 Mutex::Autolock _l(mLock); 6113 6114 for (size_t i = 0; i < mTracks.size(); i++) { 6115 sp<RecordTrack> track = mTracks[i]; 6116 if (eventSession == track->sessionId()) { 6117 (void) track->setSyncEvent(event); 6118 ret = NO_ERROR; 6119 } 6120 } 6121 return ret; 6122#else 6123 return BAD_VALUE; 6124#endif 6125} 6126 6127// destroyTrack_l() must be called with ThreadBase::mLock held 6128void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6129{ 6130 track->terminate(); 6131 track->mState = TrackBase::STOPPED; 6132 // active tracks are removed by threadLoop() 6133 if (mActiveTracks.indexOf(track) < 0) { 6134 removeTrack_l(track); 6135 } 6136} 6137 6138void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6139{ 6140 mTracks.remove(track); 6141 // need anything related to effects here? 6142 if (track->isFastTrack()) { 6143 ALOG_ASSERT(!mFastTrackAvail); 6144 mFastTrackAvail = true; 6145 } 6146} 6147 6148void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6149{ 6150 dumpInternals(fd, args); 6151 dumpTracks(fd, args); 6152 dumpEffectChains(fd, args); 6153} 6154 6155void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6156{ 6157 dprintf(fd, "\nInput thread %p:\n", this); 6158 6159 dumpBase(fd, args); 6160 6161 if (mActiveTracks.size() == 0) { 6162 dprintf(fd, " No active record clients\n"); 6163 } 6164 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6165 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6166} 6167 6168void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6169{ 6170 const size_t SIZE = 256; 6171 char buffer[SIZE]; 6172 String8 result; 6173 6174 size_t numtracks = mTracks.size(); 6175 size_t numactive = mActiveTracks.size(); 6176 size_t numactiveseen = 0; 6177 dprintf(fd, " %d Tracks", numtracks); 6178 if (numtracks) { 6179 dprintf(fd, " of which %d are active\n", numactive); 6180 RecordTrack::appendDumpHeader(result); 6181 for (size_t i = 0; i < numtracks ; ++i) { 6182 sp<RecordTrack> track = mTracks[i]; 6183 if (track != 0) { 6184 bool active = mActiveTracks.indexOf(track) >= 0; 6185 if (active) { 6186 numactiveseen++; 6187 } 6188 track->dump(buffer, SIZE, active); 6189 result.append(buffer); 6190 } 6191 } 6192 } else { 6193 dprintf(fd, "\n"); 6194 } 6195 6196 if (numactiveseen != numactive) { 6197 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6198 " not in the track list\n"); 6199 result.append(buffer); 6200 RecordTrack::appendDumpHeader(result); 6201 for (size_t i = 0; i < numactive; ++i) { 6202 sp<RecordTrack> track = mActiveTracks[i]; 6203 if (mTracks.indexOf(track) < 0) { 6204 track->dump(buffer, SIZE, true); 6205 result.append(buffer); 6206 } 6207 } 6208 6209 } 6210 write(fd, result.string(), result.size()); 6211} 6212 6213// AudioBufferProvider interface 6214status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6215 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6216{ 6217 RecordTrack *activeTrack = mRecordTrack; 6218 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 6219 if (threadBase == 0) { 6220 buffer->frameCount = 0; 6221 buffer->raw = NULL; 6222 return NOT_ENOUGH_DATA; 6223 } 6224 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6225 int32_t rear = recordThread->mRsmpInRear; 6226 int32_t front = activeTrack->mRsmpInFront; 6227 ssize_t filled = rear - front; 6228 // FIXME should not be P2 (don't want to increase latency) 6229 // FIXME if client not keeping up, discard 6230 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6231 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6232 front &= recordThread->mRsmpInFramesP2 - 1; 6233 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6234 if (part1 > (size_t) filled) { 6235 part1 = filled; 6236 } 6237 size_t ask = buffer->frameCount; 6238 ALOG_ASSERT(ask > 0); 6239 if (part1 > ask) { 6240 part1 = ask; 6241 } 6242 if (part1 == 0) { 6243 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 6244 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 6245 buffer->raw = NULL; 6246 buffer->frameCount = 0; 6247 activeTrack->mRsmpInUnrel = 0; 6248 return NOT_ENOUGH_DATA; 6249 } 6250 6251 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6252 buffer->frameCount = part1; 6253 activeTrack->mRsmpInUnrel = part1; 6254 return NO_ERROR; 6255} 6256 6257// AudioBufferProvider interface 6258void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6259 AudioBufferProvider::Buffer* buffer) 6260{ 6261 RecordTrack *activeTrack = mRecordTrack; 6262 size_t stepCount = buffer->frameCount; 6263 if (stepCount == 0) { 6264 return; 6265 } 6266 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 6267 activeTrack->mRsmpInUnrel -= stepCount; 6268 activeTrack->mRsmpInFront += stepCount; 6269 buffer->raw = NULL; 6270 buffer->frameCount = 0; 6271} 6272 6273bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6274 status_t& status) 6275{ 6276 bool reconfig = false; 6277 6278 status = NO_ERROR; 6279 6280 audio_format_t reqFormat = mFormat; 6281 uint32_t samplingRate = mSampleRate; 6282 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6283 6284 AudioParameter param = AudioParameter(keyValuePair); 6285 int value; 6286 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6287 // channel count change can be requested. Do we mandate the first client defines the 6288 // HAL sampling rate and channel count or do we allow changes on the fly? 6289 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6290 samplingRate = value; 6291 reconfig = true; 6292 } 6293 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6294 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 6295 status = BAD_VALUE; 6296 } else { 6297 reqFormat = (audio_format_t) value; 6298 reconfig = true; 6299 } 6300 } 6301 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6302 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6303 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6304 status = BAD_VALUE; 6305 } else { 6306 channelMask = mask; 6307 reconfig = true; 6308 } 6309 } 6310 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6311 // do not accept frame count changes if tracks are open as the track buffer 6312 // size depends on frame count and correct behavior would not be guaranteed 6313 // if frame count is changed after track creation 6314 if (mActiveTracks.size() > 0) { 6315 status = INVALID_OPERATION; 6316 } else { 6317 reconfig = true; 6318 } 6319 } 6320 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6321 // forward device change to effects that have requested to be 6322 // aware of attached audio device. 6323 for (size_t i = 0; i < mEffectChains.size(); i++) { 6324 mEffectChains[i]->setDevice_l(value); 6325 } 6326 6327 // store input device and output device but do not forward output device to audio HAL. 6328 // Note that status is ignored by the caller for output device 6329 // (see AudioFlinger::setParameters() 6330 if (audio_is_output_devices(value)) { 6331 mOutDevice = value; 6332 status = BAD_VALUE; 6333 } else { 6334 mInDevice = value; 6335 // disable AEC and NS if the device is a BT SCO headset supporting those 6336 // pre processings 6337 if (mTracks.size() > 0) { 6338 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6339 mAudioFlinger->btNrecIsOff(); 6340 for (size_t i = 0; i < mTracks.size(); i++) { 6341 sp<RecordTrack> track = mTracks[i]; 6342 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6343 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6344 } 6345 } 6346 } 6347 } 6348 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6349 mAudioSource != (audio_source_t)value) { 6350 // forward device change to effects that have requested to be 6351 // aware of attached audio device. 6352 for (size_t i = 0; i < mEffectChains.size(); i++) { 6353 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6354 } 6355 mAudioSource = (audio_source_t)value; 6356 } 6357 6358 if (status == NO_ERROR) { 6359 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6360 keyValuePair.string()); 6361 if (status == INVALID_OPERATION) { 6362 inputStandBy(); 6363 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6364 keyValuePair.string()); 6365 } 6366 if (reconfig) { 6367 if (status == BAD_VALUE && 6368 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6369 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6370 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6371 <= (2 * samplingRate)) && 6372 audio_channel_count_from_in_mask( 6373 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6374 (channelMask == AUDIO_CHANNEL_IN_MONO || 6375 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6376 status = NO_ERROR; 6377 } 6378 if (status == NO_ERROR) { 6379 readInputParameters_l(); 6380 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6381 } 6382 } 6383 } 6384 6385 return reconfig; 6386} 6387 6388String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6389{ 6390 Mutex::Autolock _l(mLock); 6391 if (initCheck() != NO_ERROR) { 6392 return String8(); 6393 } 6394 6395 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6396 const String8 out_s8(s); 6397 free(s); 6398 return out_s8; 6399} 6400 6401void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6402 AudioSystem::OutputDescriptor desc; 6403 const void *param2 = NULL; 6404 6405 switch (event) { 6406 case AudioSystem::INPUT_OPENED: 6407 case AudioSystem::INPUT_CONFIG_CHANGED: 6408 desc.channelMask = mChannelMask; 6409 desc.samplingRate = mSampleRate; 6410 desc.format = mFormat; 6411 desc.frameCount = mFrameCount; 6412 desc.latency = 0; 6413 param2 = &desc; 6414 break; 6415 6416 case AudioSystem::INPUT_CLOSED: 6417 default: 6418 break; 6419 } 6420 mAudioFlinger->audioConfigChanged(event, mId, param2); 6421} 6422 6423void AudioFlinger::RecordThread::readInputParameters_l() 6424{ 6425 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6426 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6427 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6428 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6429 mFormat = mHALFormat; 6430 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6431 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6432 } 6433 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6434 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6435 mFrameCount = mBufferSize / mFrameSize; 6436 // This is the formula for calculating the temporary buffer size. 6437 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6438 // 1 full output buffer, regardless of the alignment of the available input. 6439 // The value is somewhat arbitrary, and could probably be even larger. 6440 // A larger value should allow more old data to be read after a track calls start(), 6441 // without increasing latency. 6442 mRsmpInFrames = mFrameCount * 7; 6443 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6444 delete[] mRsmpInBuffer; 6445 6446 // TODO optimize audio capture buffer sizes ... 6447 // Here we calculate the size of the sliding buffer used as a source 6448 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6449 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6450 // be better to have it derived from the pipe depth in the long term. 6451 // The current value is higher than necessary. However it should not add to latency. 6452 6453 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6454 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6455 6456 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6457 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6458} 6459 6460uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6461{ 6462 Mutex::Autolock _l(mLock); 6463 if (initCheck() != NO_ERROR) { 6464 return 0; 6465 } 6466 6467 return mInput->stream->get_input_frames_lost(mInput->stream); 6468} 6469 6470uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6471{ 6472 Mutex::Autolock _l(mLock); 6473 uint32_t result = 0; 6474 if (getEffectChain_l(sessionId) != 0) { 6475 result = EFFECT_SESSION; 6476 } 6477 6478 for (size_t i = 0; i < mTracks.size(); ++i) { 6479 if (sessionId == mTracks[i]->sessionId()) { 6480 result |= TRACK_SESSION; 6481 break; 6482 } 6483 } 6484 6485 return result; 6486} 6487 6488KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6489{ 6490 KeyedVector<int, bool> ids; 6491 Mutex::Autolock _l(mLock); 6492 for (size_t j = 0; j < mTracks.size(); ++j) { 6493 sp<RecordThread::RecordTrack> track = mTracks[j]; 6494 int sessionId = track->sessionId(); 6495 if (ids.indexOfKey(sessionId) < 0) { 6496 ids.add(sessionId, true); 6497 } 6498 } 6499 return ids; 6500} 6501 6502AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6503{ 6504 Mutex::Autolock _l(mLock); 6505 AudioStreamIn *input = mInput; 6506 mInput = NULL; 6507 return input; 6508} 6509 6510// this method must always be called either with ThreadBase mLock held or inside the thread loop 6511audio_stream_t* AudioFlinger::RecordThread::stream() const 6512{ 6513 if (mInput == NULL) { 6514 return NULL; 6515 } 6516 return &mInput->stream->common; 6517} 6518 6519status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6520{ 6521 // only one chain per input thread 6522 if (mEffectChains.size() != 0) { 6523 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6524 return INVALID_OPERATION; 6525 } 6526 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6527 chain->setThread(this); 6528 chain->setInBuffer(NULL); 6529 chain->setOutBuffer(NULL); 6530 6531 checkSuspendOnAddEffectChain_l(chain); 6532 6533 // make sure enabled pre processing effects state is communicated to the HAL as we 6534 // just moved them to a new input stream. 6535 chain->syncHalEffectsState(); 6536 6537 mEffectChains.add(chain); 6538 6539 return NO_ERROR; 6540} 6541 6542size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6543{ 6544 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6545 ALOGW_IF(mEffectChains.size() != 1, 6546 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6547 chain.get(), mEffectChains.size(), this); 6548 if (mEffectChains.size() == 1) { 6549 mEffectChains.removeAt(0); 6550 } 6551 return 0; 6552} 6553 6554status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6555 audio_patch_handle_t *handle) 6556{ 6557 status_t status = NO_ERROR; 6558 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6559 // store new device and send to effects 6560 mInDevice = patch->sources[0].ext.device.type; 6561 for (size_t i = 0; i < mEffectChains.size(); i++) { 6562 mEffectChains[i]->setDevice_l(mInDevice); 6563 } 6564 6565 // disable AEC and NS if the device is a BT SCO headset supporting those 6566 // pre processings 6567 if (mTracks.size() > 0) { 6568 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6569 mAudioFlinger->btNrecIsOff(); 6570 for (size_t i = 0; i < mTracks.size(); i++) { 6571 sp<RecordTrack> track = mTracks[i]; 6572 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6573 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6574 } 6575 } 6576 6577 // store new source and send to effects 6578 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6579 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6580 for (size_t i = 0; i < mEffectChains.size(); i++) { 6581 mEffectChains[i]->setAudioSource_l(mAudioSource); 6582 } 6583 } 6584 6585 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6586 status = hwDevice->create_audio_patch(hwDevice, 6587 patch->num_sources, 6588 patch->sources, 6589 patch->num_sinks, 6590 patch->sinks, 6591 handle); 6592 } else { 6593 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6594 } 6595 return status; 6596} 6597 6598status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6599{ 6600 status_t status = NO_ERROR; 6601 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6602 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6603 status = hwDevice->release_audio_patch(hwDevice, handle); 6604 } else { 6605 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6606 } 6607 return status; 6608} 6609 6610void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6611{ 6612 Mutex::Autolock _l(mLock); 6613 mTracks.add(record); 6614} 6615 6616void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6617{ 6618 Mutex::Autolock _l(mLock); 6619 destroyTrack_l(record); 6620} 6621 6622void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6623{ 6624 ThreadBase::getAudioPortConfig(config); 6625 config->role = AUDIO_PORT_ROLE_SINK; 6626 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6627 config->ext.mix.usecase.source = mAudioSource; 6628} 6629 6630} // namespace android 6631