Threads.cpp revision 1dfe2f9c2d03fc8d0ed0cdfe0b9fb894bc0bcc11
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, " I/O handle: %d\n", mId); 745 dprintf(fd, " TID: %d\n", getTid()); 746 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 747 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 748 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 749 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 750 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 751 dprintf(fd, " Channel count: %u\n", mChannelCount); 752 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 753 channelMaskToString(mChannelMask, mType != RECORD).string()); 754 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 755 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 756 dprintf(fd, " Pending config events:"); 757 size_t numConfig = mConfigEvents.size(); 758 if (numConfig) { 759 for (size_t i = 0; i < numConfig; i++) { 760 mConfigEvents[i]->dump(buffer, SIZE); 761 dprintf(fd, "\n %s", buffer); 762 } 763 dprintf(fd, "\n"); 764 } else { 765 dprintf(fd, " none\n"); 766 } 767 768 if (locked) { 769 mLock.unlock(); 770 } 771} 772 773void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 774{ 775 const size_t SIZE = 256; 776 char buffer[SIZE]; 777 String8 result; 778 779 size_t numEffectChains = mEffectChains.size(); 780 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 781 write(fd, buffer, strlen(buffer)); 782 783 for (size_t i = 0; i < numEffectChains; ++i) { 784 sp<EffectChain> chain = mEffectChains[i]; 785 if (chain != 0) { 786 chain->dump(fd, args); 787 } 788 } 789} 790 791void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 792{ 793 Mutex::Autolock _l(mLock); 794 acquireWakeLock_l(uid); 795} 796 797String16 AudioFlinger::ThreadBase::getWakeLockTag() 798{ 799 switch (mType) { 800 case MIXER: 801 return String16("AudioMix"); 802 case DIRECT: 803 return String16("AudioDirectOut"); 804 case DUPLICATING: 805 return String16("AudioDup"); 806 case RECORD: 807 return String16("AudioIn"); 808 case OFFLOAD: 809 return String16("AudioOffload"); 810 default: 811 ALOG_ASSERT(false); 812 return String16("AudioUnknown"); 813 } 814} 815 816void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 817{ 818 getPowerManager_l(); 819 if (mPowerManager != 0) { 820 sp<IBinder> binder = new BBinder(); 821 status_t status; 822 if (uid >= 0) { 823 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 824 binder, 825 getWakeLockTag(), 826 String16("media"), 827 uid, 828 true /* FIXME force oneway contrary to .aidl */); 829 } else { 830 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 831 binder, 832 getWakeLockTag(), 833 String16("media"), 834 true /* FIXME force oneway contrary to .aidl */); 835 } 836 if (status == NO_ERROR) { 837 mWakeLockToken = binder; 838 } 839 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 840 } 841} 842 843void AudioFlinger::ThreadBase::releaseWakeLock() 844{ 845 Mutex::Autolock _l(mLock); 846 releaseWakeLock_l(); 847} 848 849void AudioFlinger::ThreadBase::releaseWakeLock_l() 850{ 851 if (mWakeLockToken != 0) { 852 ALOGV("releaseWakeLock_l() %s", mThreadName); 853 if (mPowerManager != 0) { 854 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 855 true /* FIXME force oneway contrary to .aidl */); 856 } 857 mWakeLockToken.clear(); 858 } 859} 860 861void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 862 Mutex::Autolock _l(mLock); 863 updateWakeLockUids_l(uids); 864} 865 866void AudioFlinger::ThreadBase::getPowerManager_l() { 867 868 if (mPowerManager == 0) { 869 // use checkService() to avoid blocking if power service is not up yet 870 sp<IBinder> binder = 871 defaultServiceManager()->checkService(String16("power")); 872 if (binder == 0) { 873 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 874 } else { 875 mPowerManager = interface_cast<IPowerManager>(binder); 876 binder->linkToDeath(mDeathRecipient); 877 } 878 } 879} 880 881void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 882 883 getPowerManager_l(); 884 if (mWakeLockToken == NULL) { 885 ALOGE("no wake lock to update!"); 886 return; 887 } 888 if (mPowerManager != 0) { 889 sp<IBinder> binder = new BBinder(); 890 status_t status; 891 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 892 true /* FIXME force oneway contrary to .aidl */); 893 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 894 } 895} 896 897void AudioFlinger::ThreadBase::clearPowerManager() 898{ 899 Mutex::Autolock _l(mLock); 900 releaseWakeLock_l(); 901 mPowerManager.clear(); 902} 903 904void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 905{ 906 sp<ThreadBase> thread = mThread.promote(); 907 if (thread != 0) { 908 thread->clearPowerManager(); 909 } 910 ALOGW("power manager service died !!!"); 911} 912 913void AudioFlinger::ThreadBase::setEffectSuspended( 914 const effect_uuid_t *type, bool suspend, int sessionId) 915{ 916 Mutex::Autolock _l(mLock); 917 setEffectSuspended_l(type, suspend, sessionId); 918} 919 920void AudioFlinger::ThreadBase::setEffectSuspended_l( 921 const effect_uuid_t *type, bool suspend, int sessionId) 922{ 923 sp<EffectChain> chain = getEffectChain_l(sessionId); 924 if (chain != 0) { 925 if (type != NULL) { 926 chain->setEffectSuspended_l(type, suspend); 927 } else { 928 chain->setEffectSuspendedAll_l(suspend); 929 } 930 } 931 932 updateSuspendedSessions_l(type, suspend, sessionId); 933} 934 935void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 936{ 937 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 938 if (index < 0) { 939 return; 940 } 941 942 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 943 mSuspendedSessions.valueAt(index); 944 945 for (size_t i = 0; i < sessionEffects.size(); i++) { 946 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 947 for (int j = 0; j < desc->mRefCount; j++) { 948 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 949 chain->setEffectSuspendedAll_l(true); 950 } else { 951 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 952 desc->mType.timeLow); 953 chain->setEffectSuspended_l(&desc->mType, true); 954 } 955 } 956 } 957} 958 959void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 960 bool suspend, 961 int sessionId) 962{ 963 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 964 965 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 966 967 if (suspend) { 968 if (index >= 0) { 969 sessionEffects = mSuspendedSessions.valueAt(index); 970 } else { 971 mSuspendedSessions.add(sessionId, sessionEffects); 972 } 973 } else { 974 if (index < 0) { 975 return; 976 } 977 sessionEffects = mSuspendedSessions.valueAt(index); 978 } 979 980 981 int key = EffectChain::kKeyForSuspendAll; 982 if (type != NULL) { 983 key = type->timeLow; 984 } 985 index = sessionEffects.indexOfKey(key); 986 987 sp<SuspendedSessionDesc> desc; 988 if (suspend) { 989 if (index >= 0) { 990 desc = sessionEffects.valueAt(index); 991 } else { 992 desc = new SuspendedSessionDesc(); 993 if (type != NULL) { 994 desc->mType = *type; 995 } 996 sessionEffects.add(key, desc); 997 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 998 } 999 desc->mRefCount++; 1000 } else { 1001 if (index < 0) { 1002 return; 1003 } 1004 desc = sessionEffects.valueAt(index); 1005 if (--desc->mRefCount == 0) { 1006 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1007 sessionEffects.removeItemsAt(index); 1008 if (sessionEffects.isEmpty()) { 1009 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1010 sessionId); 1011 mSuspendedSessions.removeItem(sessionId); 1012 } 1013 } 1014 } 1015 if (!sessionEffects.isEmpty()) { 1016 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1017 } 1018} 1019 1020void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1021 bool enabled, 1022 int sessionId) 1023{ 1024 Mutex::Autolock _l(mLock); 1025 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1026} 1027 1028void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1029 bool enabled, 1030 int sessionId) 1031{ 1032 if (mType != RECORD) { 1033 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1034 // another session. This gives the priority to well behaved effect control panels 1035 // and applications not using global effects. 1036 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1037 // global effects 1038 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1039 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1040 } 1041 } 1042 1043 sp<EffectChain> chain = getEffectChain_l(sessionId); 1044 if (chain != 0) { 1045 chain->checkSuspendOnEffectEnabled(effect, enabled); 1046 } 1047} 1048 1049// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1050sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1051 const sp<AudioFlinger::Client>& client, 1052 const sp<IEffectClient>& effectClient, 1053 int32_t priority, 1054 int sessionId, 1055 effect_descriptor_t *desc, 1056 int *enabled, 1057 status_t *status) 1058{ 1059 sp<EffectModule> effect; 1060 sp<EffectHandle> handle; 1061 status_t lStatus; 1062 sp<EffectChain> chain; 1063 bool chainCreated = false; 1064 bool effectCreated = false; 1065 bool effectRegistered = false; 1066 1067 lStatus = initCheck(); 1068 if (lStatus != NO_ERROR) { 1069 ALOGW("createEffect_l() Audio driver not initialized."); 1070 goto Exit; 1071 } 1072 1073 // Reject any effect on Direct output threads for now, since the format of 1074 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1075 if (mType == DIRECT) { 1076 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1077 desc->name, mThreadName); 1078 lStatus = BAD_VALUE; 1079 goto Exit; 1080 } 1081 1082 // Reject any effect on mixer or duplicating multichannel sinks. 1083 // TODO: fix both format and multichannel issues with effects. 1084 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1085 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1086 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1087 lStatus = BAD_VALUE; 1088 goto Exit; 1089 } 1090 1091 // Allow global effects only on offloaded and mixer threads 1092 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1093 switch (mType) { 1094 case MIXER: 1095 case OFFLOAD: 1096 break; 1097 case DIRECT: 1098 case DUPLICATING: 1099 case RECORD: 1100 default: 1101 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1102 desc->name, mThreadName); 1103 lStatus = BAD_VALUE; 1104 goto Exit; 1105 } 1106 } 1107 1108 // Only Pre processor effects are allowed on input threads and only on input threads 1109 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1110 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1111 desc->name, desc->flags, mType); 1112 lStatus = BAD_VALUE; 1113 goto Exit; 1114 } 1115 1116 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1117 1118 { // scope for mLock 1119 Mutex::Autolock _l(mLock); 1120 1121 // check for existing effect chain with the requested audio session 1122 chain = getEffectChain_l(sessionId); 1123 if (chain == 0) { 1124 // create a new chain for this session 1125 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1126 chain = new EffectChain(this, sessionId); 1127 addEffectChain_l(chain); 1128 chain->setStrategy(getStrategyForSession_l(sessionId)); 1129 chainCreated = true; 1130 } else { 1131 effect = chain->getEffectFromDesc_l(desc); 1132 } 1133 1134 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1135 1136 if (effect == 0) { 1137 int id = mAudioFlinger->nextUniqueId(); 1138 // Check CPU and memory usage 1139 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1140 if (lStatus != NO_ERROR) { 1141 goto Exit; 1142 } 1143 effectRegistered = true; 1144 // create a new effect module if none present in the chain 1145 effect = new EffectModule(this, chain, desc, id, sessionId); 1146 lStatus = effect->status(); 1147 if (lStatus != NO_ERROR) { 1148 goto Exit; 1149 } 1150 effect->setOffloaded(mType == OFFLOAD, mId); 1151 1152 lStatus = chain->addEffect_l(effect); 1153 if (lStatus != NO_ERROR) { 1154 goto Exit; 1155 } 1156 effectCreated = true; 1157 1158 effect->setDevice(mOutDevice); 1159 effect->setDevice(mInDevice); 1160 effect->setMode(mAudioFlinger->getMode()); 1161 effect->setAudioSource(mAudioSource); 1162 } 1163 // create effect handle and connect it to effect module 1164 handle = new EffectHandle(effect, client, effectClient, priority); 1165 lStatus = handle->initCheck(); 1166 if (lStatus == OK) { 1167 lStatus = effect->addHandle(handle.get()); 1168 } 1169 if (enabled != NULL) { 1170 *enabled = (int)effect->isEnabled(); 1171 } 1172 } 1173 1174Exit: 1175 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1176 Mutex::Autolock _l(mLock); 1177 if (effectCreated) { 1178 chain->removeEffect_l(effect); 1179 } 1180 if (effectRegistered) { 1181 AudioSystem::unregisterEffect(effect->id()); 1182 } 1183 if (chainCreated) { 1184 removeEffectChain_l(chain); 1185 } 1186 handle.clear(); 1187 } 1188 1189 *status = lStatus; 1190 return handle; 1191} 1192 1193sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1194{ 1195 Mutex::Autolock _l(mLock); 1196 return getEffect_l(sessionId, effectId); 1197} 1198 1199sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1200{ 1201 sp<EffectChain> chain = getEffectChain_l(sessionId); 1202 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1203} 1204 1205// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1206// PlaybackThread::mLock held 1207status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1208{ 1209 // check for existing effect chain with the requested audio session 1210 int sessionId = effect->sessionId(); 1211 sp<EffectChain> chain = getEffectChain_l(sessionId); 1212 bool chainCreated = false; 1213 1214 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1215 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1216 this, effect->desc().name, effect->desc().flags); 1217 1218 if (chain == 0) { 1219 // create a new chain for this session 1220 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1221 chain = new EffectChain(this, sessionId); 1222 addEffectChain_l(chain); 1223 chain->setStrategy(getStrategyForSession_l(sessionId)); 1224 chainCreated = true; 1225 } 1226 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1227 1228 if (chain->getEffectFromId_l(effect->id()) != 0) { 1229 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1230 this, effect->desc().name, chain.get()); 1231 return BAD_VALUE; 1232 } 1233 1234 effect->setOffloaded(mType == OFFLOAD, mId); 1235 1236 status_t status = chain->addEffect_l(effect); 1237 if (status != NO_ERROR) { 1238 if (chainCreated) { 1239 removeEffectChain_l(chain); 1240 } 1241 return status; 1242 } 1243 1244 effect->setDevice(mOutDevice); 1245 effect->setDevice(mInDevice); 1246 effect->setMode(mAudioFlinger->getMode()); 1247 effect->setAudioSource(mAudioSource); 1248 return NO_ERROR; 1249} 1250 1251void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1252 1253 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1254 effect_descriptor_t desc = effect->desc(); 1255 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1256 detachAuxEffect_l(effect->id()); 1257 } 1258 1259 sp<EffectChain> chain = effect->chain().promote(); 1260 if (chain != 0) { 1261 // remove effect chain if removing last effect 1262 if (chain->removeEffect_l(effect) == 0) { 1263 removeEffectChain_l(chain); 1264 } 1265 } else { 1266 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1267 } 1268} 1269 1270void AudioFlinger::ThreadBase::lockEffectChains_l( 1271 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1272{ 1273 effectChains = mEffectChains; 1274 for (size_t i = 0; i < mEffectChains.size(); i++) { 1275 mEffectChains[i]->lock(); 1276 } 1277} 1278 1279void AudioFlinger::ThreadBase::unlockEffectChains( 1280 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1281{ 1282 for (size_t i = 0; i < effectChains.size(); i++) { 1283 effectChains[i]->unlock(); 1284 } 1285} 1286 1287sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1288{ 1289 Mutex::Autolock _l(mLock); 1290 return getEffectChain_l(sessionId); 1291} 1292 1293sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1294{ 1295 size_t size = mEffectChains.size(); 1296 for (size_t i = 0; i < size; i++) { 1297 if (mEffectChains[i]->sessionId() == sessionId) { 1298 return mEffectChains[i]; 1299 } 1300 } 1301 return 0; 1302} 1303 1304void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1305{ 1306 Mutex::Autolock _l(mLock); 1307 size_t size = mEffectChains.size(); 1308 for (size_t i = 0; i < size; i++) { 1309 mEffectChains[i]->setMode_l(mode); 1310 } 1311} 1312 1313void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1314{ 1315 config->type = AUDIO_PORT_TYPE_MIX; 1316 config->ext.mix.handle = mId; 1317 config->sample_rate = mSampleRate; 1318 config->format = mFormat; 1319 config->channel_mask = mChannelMask; 1320 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1321 AUDIO_PORT_CONFIG_FORMAT; 1322} 1323 1324 1325// ---------------------------------------------------------------------------- 1326// Playback 1327// ---------------------------------------------------------------------------- 1328 1329AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1330 AudioStreamOut* output, 1331 audio_io_handle_t id, 1332 audio_devices_t device, 1333 type_t type) 1334 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1335 mNormalFrameCount(0), mSinkBuffer(NULL), 1336 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1337 mMixerBuffer(NULL), 1338 mMixerBufferSize(0), 1339 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1340 mMixerBufferValid(false), 1341 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1342 mEffectBuffer(NULL), 1343 mEffectBufferSize(0), 1344 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1345 mEffectBufferValid(false), 1346 mSuspended(0), mBytesWritten(0), 1347 mActiveTracksGeneration(0), 1348 // mStreamTypes[] initialized in constructor body 1349 mOutput(output), 1350 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1351 mMixerStatus(MIXER_IDLE), 1352 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1353 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1354 mBytesRemaining(0), 1355 mCurrentWriteLength(0), 1356 mUseAsyncWrite(false), 1357 mWriteAckSequence(0), 1358 mDrainSequence(0), 1359 mSignalPending(false), 1360 mScreenState(AudioFlinger::mScreenState), 1361 // index 0 is reserved for normal mixer's submix 1362 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1363 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1364 // mLatchD, mLatchQ, 1365 mLatchDValid(false), mLatchQValid(false) 1366{ 1367 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1368 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1369 1370 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1371 // it would be safer to explicitly pass initial masterVolume/masterMute as 1372 // parameter. 1373 // 1374 // If the HAL we are using has support for master volume or master mute, 1375 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1376 // and the mute set to false). 1377 mMasterVolume = audioFlinger->masterVolume_l(); 1378 mMasterMute = audioFlinger->masterMute_l(); 1379 if (mOutput && mOutput->audioHwDev) { 1380 if (mOutput->audioHwDev->canSetMasterVolume()) { 1381 mMasterVolume = 1.0; 1382 } 1383 1384 if (mOutput->audioHwDev->canSetMasterMute()) { 1385 mMasterMute = false; 1386 } 1387 } 1388 1389 readOutputParameters_l(); 1390 1391 // ++ operator does not compile 1392 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1393 stream = (audio_stream_type_t) (stream + 1)) { 1394 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1395 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1396 } 1397} 1398 1399AudioFlinger::PlaybackThread::~PlaybackThread() 1400{ 1401 mAudioFlinger->unregisterWriter(mNBLogWriter); 1402 free(mSinkBuffer); 1403 free(mMixerBuffer); 1404 free(mEffectBuffer); 1405} 1406 1407void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1408{ 1409 dumpInternals(fd, args); 1410 dumpTracks(fd, args); 1411 dumpEffectChains(fd, args); 1412} 1413 1414void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1415{ 1416 const size_t SIZE = 256; 1417 char buffer[SIZE]; 1418 String8 result; 1419 1420 result.appendFormat(" Stream volumes in dB: "); 1421 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1422 const stream_type_t *st = &mStreamTypes[i]; 1423 if (i > 0) { 1424 result.appendFormat(", "); 1425 } 1426 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1427 if (st->mute) { 1428 result.append("M"); 1429 } 1430 } 1431 result.append("\n"); 1432 write(fd, result.string(), result.length()); 1433 result.clear(); 1434 1435 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1436 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1437 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1438 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1439 1440 size_t numtracks = mTracks.size(); 1441 size_t numactive = mActiveTracks.size(); 1442 dprintf(fd, " %d Tracks", numtracks); 1443 size_t numactiveseen = 0; 1444 if (numtracks) { 1445 dprintf(fd, " of which %d are active\n", numactive); 1446 Track::appendDumpHeader(result); 1447 for (size_t i = 0; i < numtracks; ++i) { 1448 sp<Track> track = mTracks[i]; 1449 if (track != 0) { 1450 bool active = mActiveTracks.indexOf(track) >= 0; 1451 if (active) { 1452 numactiveseen++; 1453 } 1454 track->dump(buffer, SIZE, active); 1455 result.append(buffer); 1456 } 1457 } 1458 } else { 1459 result.append("\n"); 1460 } 1461 if (numactiveseen != numactive) { 1462 // some tracks in the active list were not in the tracks list 1463 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1464 " not in the track list\n"); 1465 result.append(buffer); 1466 Track::appendDumpHeader(result); 1467 for (size_t i = 0; i < numactive; ++i) { 1468 sp<Track> track = mActiveTracks[i].promote(); 1469 if (track != 0 && mTracks.indexOf(track) < 0) { 1470 track->dump(buffer, SIZE, true); 1471 result.append(buffer); 1472 } 1473 } 1474 } 1475 1476 write(fd, result.string(), result.size()); 1477} 1478 1479void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1480{ 1481 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1482 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1483 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1484 dprintf(fd, " Total writes: %d\n", mNumWrites); 1485 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1486 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1487 dprintf(fd, " Suspend count: %d\n", mSuspended); 1488 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1489 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1490 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1491 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1492 AudioStreamOut *output = mOutput; 1493 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1494 String8 flagsAsString = outputFlagsToString(flags); 1495 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1496 1497 dumpBase(fd, args); 1498} 1499 1500// Thread virtuals 1501 1502void AudioFlinger::PlaybackThread::onFirstRef() 1503{ 1504 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1505} 1506 1507// ThreadBase virtuals 1508void AudioFlinger::PlaybackThread::preExit() 1509{ 1510 ALOGV(" preExit()"); 1511 // FIXME this is using hard-coded strings but in the future, this functionality will be 1512 // converted to use audio HAL extensions required to support tunneling 1513 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1514} 1515 1516// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1517sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1518 const sp<AudioFlinger::Client>& client, 1519 audio_stream_type_t streamType, 1520 uint32_t sampleRate, 1521 audio_format_t format, 1522 audio_channel_mask_t channelMask, 1523 size_t *pFrameCount, 1524 const sp<IMemory>& sharedBuffer, 1525 int sessionId, 1526 IAudioFlinger::track_flags_t *flags, 1527 pid_t tid, 1528 int uid, 1529 status_t *status) 1530{ 1531 size_t frameCount = *pFrameCount; 1532 sp<Track> track; 1533 status_t lStatus; 1534 1535 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1536 1537 // client expresses a preference for FAST, but we get the final say 1538 if (*flags & IAudioFlinger::TRACK_FAST) { 1539 if ( 1540 // not timed 1541 (!isTimed) && 1542 // either of these use cases: 1543 ( 1544 // use case 1: shared buffer with any frame count 1545 ( 1546 (sharedBuffer != 0) 1547 ) || 1548 // use case 2: frame count is default or at least as large as HAL 1549 ( 1550 // we formerly checked for a callback handler (non-0 tid), 1551 // but that is no longer required for TRANSFER_OBTAIN mode 1552 ((frameCount == 0) || 1553 (frameCount >= mFrameCount)) 1554 ) 1555 ) && 1556 // PCM data 1557 audio_is_linear_pcm(format) && 1558 // identical channel mask to sink, or mono in and stereo sink 1559 (channelMask == mChannelMask || 1560 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1561 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1562 // hardware sample rate 1563 (sampleRate == mSampleRate) && 1564 // normal mixer has an associated fast mixer 1565 hasFastMixer() && 1566 // there are sufficient fast track slots available 1567 (mFastTrackAvailMask != 0) 1568 // FIXME test that MixerThread for this fast track has a capable output HAL 1569 // FIXME add a permission test also? 1570 ) { 1571 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1572 if (frameCount == 0) { 1573 // read the fast track multiplier property the first time it is needed 1574 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1575 if (ok != 0) { 1576 ALOGE("%s pthread_once failed: %d", __func__, ok); 1577 } 1578 frameCount = mFrameCount * sFastTrackMultiplier; 1579 } 1580 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1581 frameCount, mFrameCount); 1582 } else { 1583 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1584 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1585 "sampleRate=%u mSampleRate=%u " 1586 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1587 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1588 audio_is_linear_pcm(format), 1589 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1590 *flags &= ~IAudioFlinger::TRACK_FAST; 1591 } 1592 } 1593 // For normal PCM streaming tracks, update minimum frame count. 1594 // For compatibility with AudioTrack calculation, buffer depth is forced 1595 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1596 // This is probably too conservative, but legacy application code may depend on it. 1597 // If you change this calculation, also review the start threshold which is related. 1598 if (!(*flags & IAudioFlinger::TRACK_FAST) 1599 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1600 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1601 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1602 if (minBufCount < 2) { 1603 minBufCount = 2; 1604 } 1605 size_t minFrameCount = 1606 minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate); 1607 if (frameCount < minFrameCount) { // including frameCount == 0 1608 frameCount = minFrameCount; 1609 } 1610 } 1611 *pFrameCount = frameCount; 1612 1613 switch (mType) { 1614 1615 case DIRECT: 1616 if (audio_is_linear_pcm(format)) { 1617 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1618 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1619 "for output %p with format %#x", 1620 sampleRate, format, channelMask, mOutput, mFormat); 1621 lStatus = BAD_VALUE; 1622 goto Exit; 1623 } 1624 } 1625 break; 1626 1627 case OFFLOAD: 1628 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1629 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1630 "for output %p with format %#x", 1631 sampleRate, format, channelMask, mOutput, mFormat); 1632 lStatus = BAD_VALUE; 1633 goto Exit; 1634 } 1635 break; 1636 1637 default: 1638 if (!audio_is_linear_pcm(format)) { 1639 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1640 "for output %p with format %#x", 1641 format, mOutput, mFormat); 1642 lStatus = BAD_VALUE; 1643 goto Exit; 1644 } 1645 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1646 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1647 lStatus = BAD_VALUE; 1648 goto Exit; 1649 } 1650 break; 1651 1652 } 1653 1654 lStatus = initCheck(); 1655 if (lStatus != NO_ERROR) { 1656 ALOGE("createTrack_l() audio driver not initialized"); 1657 goto Exit; 1658 } 1659 1660 { // scope for mLock 1661 Mutex::Autolock _l(mLock); 1662 1663 // all tracks in same audio session must share the same routing strategy otherwise 1664 // conflicts will happen when tracks are moved from one output to another by audio policy 1665 // manager 1666 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1667 for (size_t i = 0; i < mTracks.size(); ++i) { 1668 sp<Track> t = mTracks[i]; 1669 if (t != 0 && t->isExternalTrack()) { 1670 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1671 if (sessionId == t->sessionId() && strategy != actual) { 1672 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1673 strategy, actual); 1674 lStatus = BAD_VALUE; 1675 goto Exit; 1676 } 1677 } 1678 } 1679 1680 if (!isTimed) { 1681 track = new Track(this, client, streamType, sampleRate, format, 1682 channelMask, frameCount, NULL, sharedBuffer, 1683 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1684 } else { 1685 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1686 channelMask, frameCount, sharedBuffer, sessionId, uid); 1687 } 1688 1689 // new Track always returns non-NULL, 1690 // but TimedTrack::create() is a factory that could fail by returning NULL 1691 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1692 if (lStatus != NO_ERROR) { 1693 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1694 // track must be cleared from the caller as the caller has the AF lock 1695 goto Exit; 1696 } 1697 mTracks.add(track); 1698 1699 sp<EffectChain> chain = getEffectChain_l(sessionId); 1700 if (chain != 0) { 1701 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1702 track->setMainBuffer(chain->inBuffer()); 1703 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1704 chain->incTrackCnt(); 1705 } 1706 1707 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1708 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1709 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1710 // so ask activity manager to do this on our behalf 1711 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1712 } 1713 } 1714 1715 lStatus = NO_ERROR; 1716 1717Exit: 1718 *status = lStatus; 1719 return track; 1720} 1721 1722uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1723{ 1724 return latency; 1725} 1726 1727uint32_t AudioFlinger::PlaybackThread::latency() const 1728{ 1729 Mutex::Autolock _l(mLock); 1730 return latency_l(); 1731} 1732uint32_t AudioFlinger::PlaybackThread::latency_l() const 1733{ 1734 if (initCheck() == NO_ERROR) { 1735 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1736 } else { 1737 return 0; 1738 } 1739} 1740 1741void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1742{ 1743 Mutex::Autolock _l(mLock); 1744 // Don't apply master volume in SW if our HAL can do it for us. 1745 if (mOutput && mOutput->audioHwDev && 1746 mOutput->audioHwDev->canSetMasterVolume()) { 1747 mMasterVolume = 1.0; 1748 } else { 1749 mMasterVolume = value; 1750 } 1751} 1752 1753void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1754{ 1755 Mutex::Autolock _l(mLock); 1756 // Don't apply master mute in SW if our HAL can do it for us. 1757 if (mOutput && mOutput->audioHwDev && 1758 mOutput->audioHwDev->canSetMasterMute()) { 1759 mMasterMute = false; 1760 } else { 1761 mMasterMute = muted; 1762 } 1763} 1764 1765void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1766{ 1767 Mutex::Autolock _l(mLock); 1768 mStreamTypes[stream].volume = value; 1769 broadcast_l(); 1770} 1771 1772void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1773{ 1774 Mutex::Autolock _l(mLock); 1775 mStreamTypes[stream].mute = muted; 1776 broadcast_l(); 1777} 1778 1779float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1780{ 1781 Mutex::Autolock _l(mLock); 1782 return mStreamTypes[stream].volume; 1783} 1784 1785// addTrack_l() must be called with ThreadBase::mLock held 1786status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1787{ 1788 status_t status = ALREADY_EXISTS; 1789 1790 // set retry count for buffer fill 1791 track->mRetryCount = kMaxTrackStartupRetries; 1792 if (mActiveTracks.indexOf(track) < 0) { 1793 // the track is newly added, make sure it fills up all its 1794 // buffers before playing. This is to ensure the client will 1795 // effectively get the latency it requested. 1796 if (track->isExternalTrack()) { 1797 TrackBase::track_state state = track->mState; 1798 mLock.unlock(); 1799 status = AudioSystem::startOutput(mId, track->streamType(), 1800 (audio_session_t)track->sessionId()); 1801 mLock.lock(); 1802 // abort track was stopped/paused while we released the lock 1803 if (state != track->mState) { 1804 if (status == NO_ERROR) { 1805 mLock.unlock(); 1806 AudioSystem::stopOutput(mId, track->streamType(), 1807 (audio_session_t)track->sessionId()); 1808 mLock.lock(); 1809 } 1810 return INVALID_OPERATION; 1811 } 1812 // abort if start is rejected by audio policy manager 1813 if (status != NO_ERROR) { 1814 return PERMISSION_DENIED; 1815 } 1816#ifdef ADD_BATTERY_DATA 1817 // to track the speaker usage 1818 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1819#endif 1820 } 1821 1822 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1823 track->mResetDone = false; 1824 track->mPresentationCompleteFrames = 0; 1825 mActiveTracks.add(track); 1826 mWakeLockUids.add(track->uid()); 1827 mActiveTracksGeneration++; 1828 mLatestActiveTrack = track; 1829 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1830 if (chain != 0) { 1831 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1832 track->sessionId()); 1833 chain->incActiveTrackCnt(); 1834 } 1835 1836 status = NO_ERROR; 1837 } 1838 1839 onAddNewTrack_l(); 1840 return status; 1841} 1842 1843bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1844{ 1845 track->terminate(); 1846 // active tracks are removed by threadLoop() 1847 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1848 track->mState = TrackBase::STOPPED; 1849 if (!trackActive) { 1850 removeTrack_l(track); 1851 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1852 track->mState = TrackBase::STOPPING_1; 1853 } 1854 1855 return trackActive; 1856} 1857 1858void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1859{ 1860 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1861 mTracks.remove(track); 1862 deleteTrackName_l(track->name()); 1863 // redundant as track is about to be destroyed, for dumpsys only 1864 track->mName = -1; 1865 if (track->isFastTrack()) { 1866 int index = track->mFastIndex; 1867 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1868 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1869 mFastTrackAvailMask |= 1 << index; 1870 // redundant as track is about to be destroyed, for dumpsys only 1871 track->mFastIndex = -1; 1872 } 1873 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1874 if (chain != 0) { 1875 chain->decTrackCnt(); 1876 } 1877} 1878 1879void AudioFlinger::PlaybackThread::broadcast_l() 1880{ 1881 // Thread could be blocked waiting for async 1882 // so signal it to handle state changes immediately 1883 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1884 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1885 mSignalPending = true; 1886 mWaitWorkCV.broadcast(); 1887} 1888 1889String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1890{ 1891 Mutex::Autolock _l(mLock); 1892 if (initCheck() != NO_ERROR) { 1893 return String8(); 1894 } 1895 1896 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1897 const String8 out_s8(s); 1898 free(s); 1899 return out_s8; 1900} 1901 1902void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1903 AudioSystem::OutputDescriptor desc; 1904 void *param2 = NULL; 1905 1906 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1907 param); 1908 1909 switch (event) { 1910 case AudioSystem::OUTPUT_OPENED: 1911 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1912 desc.channelMask = mChannelMask; 1913 desc.samplingRate = mSampleRate; 1914 desc.format = mFormat; 1915 desc.frameCount = mNormalFrameCount; // FIXME see 1916 // AudioFlinger::frameCount(audio_io_handle_t) 1917 desc.latency = latency_l(); 1918 param2 = &desc; 1919 break; 1920 1921 case AudioSystem::STREAM_CONFIG_CHANGED: 1922 param2 = ¶m; 1923 case AudioSystem::OUTPUT_CLOSED: 1924 default: 1925 break; 1926 } 1927 mAudioFlinger->audioConfigChanged(event, mId, param2); 1928} 1929 1930void AudioFlinger::PlaybackThread::writeCallback() 1931{ 1932 ALOG_ASSERT(mCallbackThread != 0); 1933 mCallbackThread->resetWriteBlocked(); 1934} 1935 1936void AudioFlinger::PlaybackThread::drainCallback() 1937{ 1938 ALOG_ASSERT(mCallbackThread != 0); 1939 mCallbackThread->resetDraining(); 1940} 1941 1942void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1943{ 1944 Mutex::Autolock _l(mLock); 1945 // reject out of sequence requests 1946 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1947 mWriteAckSequence &= ~1; 1948 mWaitWorkCV.signal(); 1949 } 1950} 1951 1952void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1953{ 1954 Mutex::Autolock _l(mLock); 1955 // reject out of sequence requests 1956 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1957 mDrainSequence &= ~1; 1958 mWaitWorkCV.signal(); 1959 } 1960} 1961 1962// static 1963int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1964 void *param __unused, 1965 void *cookie) 1966{ 1967 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1968 ALOGV("asyncCallback() event %d", event); 1969 switch (event) { 1970 case STREAM_CBK_EVENT_WRITE_READY: 1971 me->writeCallback(); 1972 break; 1973 case STREAM_CBK_EVENT_DRAIN_READY: 1974 me->drainCallback(); 1975 break; 1976 default: 1977 ALOGW("asyncCallback() unknown event %d", event); 1978 break; 1979 } 1980 return 0; 1981} 1982 1983void AudioFlinger::PlaybackThread::readOutputParameters_l() 1984{ 1985 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1986 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1987 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1988 if (!audio_is_output_channel(mChannelMask)) { 1989 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1990 } 1991 if ((mType == MIXER || mType == DUPLICATING) 1992 && !isValidPcmSinkChannelMask(mChannelMask)) { 1993 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1994 mChannelMask); 1995 } 1996 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1997 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1998 mFormat = mHALFormat; 1999 if (!audio_is_valid_format(mFormat)) { 2000 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2001 } 2002 if ((mType == MIXER || mType == DUPLICATING) 2003 && !isValidPcmSinkFormat(mFormat)) { 2004 LOG_FATAL("HAL format %#x not supported for mixed output", 2005 mFormat); 2006 } 2007 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 2008 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2009 mFrameCount = mBufferSize / mFrameSize; 2010 if (mFrameCount & 15) { 2011 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2012 mFrameCount); 2013 } 2014 2015 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2016 (mOutput->stream->set_callback != NULL)) { 2017 if (mOutput->stream->set_callback(mOutput->stream, 2018 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2019 mUseAsyncWrite = true; 2020 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2021 } 2022 } 2023 2024 mHwSupportsPause = false; 2025 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2026 if (mOutput->stream->pause != NULL) { 2027 if (mOutput->stream->resume != NULL) { 2028 mHwSupportsPause = true; 2029 } else { 2030 ALOGW("direct output implements pause but not resume"); 2031 } 2032 } else if (mOutput->stream->resume != NULL) { 2033 ALOGW("direct output implements resume but not pause"); 2034 } 2035 } 2036 2037 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2038 // For best precision, we use float instead of the associated output 2039 // device format (typically PCM 16 bit). 2040 2041 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2042 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2043 mBufferSize = mFrameSize * mFrameCount; 2044 2045 // TODO: We currently use the associated output device channel mask and sample rate. 2046 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2047 // (if a valid mask) to avoid premature downmix. 2048 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2049 // instead of the output device sample rate to avoid loss of high frequency information. 2050 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2051 } 2052 2053 // Calculate size of normal sink buffer relative to the HAL output buffer size 2054 double multiplier = 1.0; 2055 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2056 kUseFastMixer == FastMixer_Dynamic)) { 2057 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2058 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2059 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2060 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2061 maxNormalFrameCount = maxNormalFrameCount & ~15; 2062 if (maxNormalFrameCount < minNormalFrameCount) { 2063 maxNormalFrameCount = minNormalFrameCount; 2064 } 2065 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2066 if (multiplier <= 1.0) { 2067 multiplier = 1.0; 2068 } else if (multiplier <= 2.0) { 2069 if (2 * mFrameCount <= maxNormalFrameCount) { 2070 multiplier = 2.0; 2071 } else { 2072 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2073 } 2074 } else { 2075 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2076 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2077 // track, but we sometimes have to do this to satisfy the maximum frame count 2078 // constraint) 2079 // FIXME this rounding up should not be done if no HAL SRC 2080 uint32_t truncMult = (uint32_t) multiplier; 2081 if ((truncMult & 1)) { 2082 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2083 ++truncMult; 2084 } 2085 } 2086 multiplier = (double) truncMult; 2087 } 2088 } 2089 mNormalFrameCount = multiplier * mFrameCount; 2090 // round up to nearest 16 frames to satisfy AudioMixer 2091 if (mType == MIXER || mType == DUPLICATING) { 2092 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2093 } 2094 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2095 mNormalFrameCount); 2096 2097 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2098 // Originally this was int16_t[] array, need to remove legacy implications. 2099 free(mSinkBuffer); 2100 mSinkBuffer = NULL; 2101 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2102 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2103 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2104 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2105 2106 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2107 // drives the output. 2108 free(mMixerBuffer); 2109 mMixerBuffer = NULL; 2110 if (mMixerBufferEnabled) { 2111 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2112 mMixerBufferSize = mNormalFrameCount * mChannelCount 2113 * audio_bytes_per_sample(mMixerBufferFormat); 2114 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2115 } 2116 free(mEffectBuffer); 2117 mEffectBuffer = NULL; 2118 if (mEffectBufferEnabled) { 2119 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2120 mEffectBufferSize = mNormalFrameCount * mChannelCount 2121 * audio_bytes_per_sample(mEffectBufferFormat); 2122 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2123 } 2124 2125 // force reconfiguration of effect chains and engines to take new buffer size and audio 2126 // parameters into account 2127 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2128 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2129 // matter. 2130 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2131 Vector< sp<EffectChain> > effectChains = mEffectChains; 2132 for (size_t i = 0; i < effectChains.size(); i ++) { 2133 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2134 } 2135} 2136 2137 2138status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2139{ 2140 if (halFrames == NULL || dspFrames == NULL) { 2141 return BAD_VALUE; 2142 } 2143 Mutex::Autolock _l(mLock); 2144 if (initCheck() != NO_ERROR) { 2145 return INVALID_OPERATION; 2146 } 2147 size_t framesWritten = mBytesWritten / mFrameSize; 2148 *halFrames = framesWritten; 2149 2150 if (isSuspended()) { 2151 // return an estimation of rendered frames when the output is suspended 2152 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2153 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2154 return NO_ERROR; 2155 } else { 2156 status_t status; 2157 uint32_t frames; 2158 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 2159 *dspFrames = (size_t)frames; 2160 return status; 2161 } 2162} 2163 2164uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2165{ 2166 Mutex::Autolock _l(mLock); 2167 uint32_t result = 0; 2168 if (getEffectChain_l(sessionId) != 0) { 2169 result = EFFECT_SESSION; 2170 } 2171 2172 for (size_t i = 0; i < mTracks.size(); ++i) { 2173 sp<Track> track = mTracks[i]; 2174 if (sessionId == track->sessionId() && !track->isInvalid()) { 2175 result |= TRACK_SESSION; 2176 break; 2177 } 2178 } 2179 2180 return result; 2181} 2182 2183uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2184{ 2185 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2186 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2187 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2188 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2189 } 2190 for (size_t i = 0; i < mTracks.size(); i++) { 2191 sp<Track> track = mTracks[i]; 2192 if (sessionId == track->sessionId() && !track->isInvalid()) { 2193 return AudioSystem::getStrategyForStream(track->streamType()); 2194 } 2195 } 2196 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2197} 2198 2199 2200AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2201{ 2202 Mutex::Autolock _l(mLock); 2203 return mOutput; 2204} 2205 2206AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2207{ 2208 Mutex::Autolock _l(mLock); 2209 AudioStreamOut *output = mOutput; 2210 mOutput = NULL; 2211 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2212 // must push a NULL and wait for ack 2213 mOutputSink.clear(); 2214 mPipeSink.clear(); 2215 mNormalSink.clear(); 2216 return output; 2217} 2218 2219// this method must always be called either with ThreadBase mLock held or inside the thread loop 2220audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2221{ 2222 if (mOutput == NULL) { 2223 return NULL; 2224 } 2225 return &mOutput->stream->common; 2226} 2227 2228uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2229{ 2230 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2231} 2232 2233status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2234{ 2235 if (!isValidSyncEvent(event)) { 2236 return BAD_VALUE; 2237 } 2238 2239 Mutex::Autolock _l(mLock); 2240 2241 for (size_t i = 0; i < mTracks.size(); ++i) { 2242 sp<Track> track = mTracks[i]; 2243 if (event->triggerSession() == track->sessionId()) { 2244 (void) track->setSyncEvent(event); 2245 return NO_ERROR; 2246 } 2247 } 2248 2249 return NAME_NOT_FOUND; 2250} 2251 2252bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2253{ 2254 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2255} 2256 2257void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2258 const Vector< sp<Track> >& tracksToRemove) 2259{ 2260 size_t count = tracksToRemove.size(); 2261 if (count > 0) { 2262 for (size_t i = 0 ; i < count ; i++) { 2263 const sp<Track>& track = tracksToRemove.itemAt(i); 2264 if (track->isExternalTrack()) { 2265 AudioSystem::stopOutput(mId, track->streamType(), 2266 (audio_session_t)track->sessionId()); 2267#ifdef ADD_BATTERY_DATA 2268 // to track the speaker usage 2269 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2270#endif 2271 if (track->isTerminated()) { 2272 AudioSystem::releaseOutput(mId, track->streamType(), 2273 (audio_session_t)track->sessionId()); 2274 } 2275 } 2276 } 2277 } 2278} 2279 2280void AudioFlinger::PlaybackThread::checkSilentMode_l() 2281{ 2282 if (!mMasterMute) { 2283 char value[PROPERTY_VALUE_MAX]; 2284 if (property_get("ro.audio.silent", value, "0") > 0) { 2285 char *endptr; 2286 unsigned long ul = strtoul(value, &endptr, 0); 2287 if (*endptr == '\0' && ul != 0) { 2288 ALOGD("Silence is golden"); 2289 // The setprop command will not allow a property to be changed after 2290 // the first time it is set, so we don't have to worry about un-muting. 2291 setMasterMute_l(true); 2292 } 2293 } 2294 } 2295} 2296 2297// shared by MIXER and DIRECT, overridden by DUPLICATING 2298ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2299{ 2300 // FIXME rewrite to reduce number of system calls 2301 mLastWriteTime = systemTime(); 2302 mInWrite = true; 2303 ssize_t bytesWritten; 2304 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2305 2306 // If an NBAIO sink is present, use it to write the normal mixer's submix 2307 if (mNormalSink != 0) { 2308 2309 const size_t count = mBytesRemaining / mFrameSize; 2310 2311 ATRACE_BEGIN("write"); 2312 // update the setpoint when AudioFlinger::mScreenState changes 2313 uint32_t screenState = AudioFlinger::mScreenState; 2314 if (screenState != mScreenState) { 2315 mScreenState = screenState; 2316 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2317 if (pipe != NULL) { 2318 pipe->setAvgFrames((mScreenState & 1) ? 2319 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2320 } 2321 } 2322 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2323 ATRACE_END(); 2324 if (framesWritten > 0) { 2325 bytesWritten = framesWritten * mFrameSize; 2326 } else { 2327 bytesWritten = framesWritten; 2328 } 2329 mLatchDValid = false; 2330 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2331 if (status == NO_ERROR) { 2332 size_t totalFramesWritten = mNormalSink->framesWritten(); 2333 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2334 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2335 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2336 mLatchDValid = true; 2337 } 2338 } 2339 // otherwise use the HAL / AudioStreamOut directly 2340 } else { 2341 // Direct output and offload threads 2342 2343 if (mUseAsyncWrite) { 2344 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2345 mWriteAckSequence += 2; 2346 mWriteAckSequence |= 1; 2347 ALOG_ASSERT(mCallbackThread != 0); 2348 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2349 } 2350 // FIXME We should have an implementation of timestamps for direct output threads. 2351 // They are used e.g for multichannel PCM playback over HDMI. 2352 bytesWritten = mOutput->stream->write(mOutput->stream, 2353 (char *)mSinkBuffer + offset, mBytesRemaining); 2354 if (mUseAsyncWrite && 2355 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2356 // do not wait for async callback in case of error of full write 2357 mWriteAckSequence &= ~1; 2358 ALOG_ASSERT(mCallbackThread != 0); 2359 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2360 } 2361 } 2362 2363 mNumWrites++; 2364 mInWrite = false; 2365 mStandby = false; 2366 return bytesWritten; 2367} 2368 2369void AudioFlinger::PlaybackThread::threadLoop_drain() 2370{ 2371 if (mOutput->stream->drain) { 2372 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2373 if (mUseAsyncWrite) { 2374 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2375 mDrainSequence |= 1; 2376 ALOG_ASSERT(mCallbackThread != 0); 2377 mCallbackThread->setDraining(mDrainSequence); 2378 } 2379 mOutput->stream->drain(mOutput->stream, 2380 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2381 : AUDIO_DRAIN_ALL); 2382 } 2383} 2384 2385void AudioFlinger::PlaybackThread::threadLoop_exit() 2386{ 2387 { 2388 Mutex::Autolock _l(mLock); 2389 for (size_t i = 0; i < mTracks.size(); i++) { 2390 sp<Track> track = mTracks[i]; 2391 track->invalidate(); 2392 } 2393 } 2394} 2395 2396/* 2397The derived values that are cached: 2398 - mSinkBufferSize from frame count * frame size 2399 - activeSleepTime from activeSleepTimeUs() 2400 - idleSleepTime from idleSleepTimeUs() 2401 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2402 - maxPeriod from frame count and sample rate (MIXER only) 2403 2404The parameters that affect these derived values are: 2405 - frame count 2406 - frame size 2407 - sample rate 2408 - device type: A2DP or not 2409 - device latency 2410 - format: PCM or not 2411 - active sleep time 2412 - idle sleep time 2413*/ 2414 2415void AudioFlinger::PlaybackThread::cacheParameters_l() 2416{ 2417 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2418 activeSleepTime = activeSleepTimeUs(); 2419 idleSleepTime = idleSleepTimeUs(); 2420} 2421 2422void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2423{ 2424 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2425 this, streamType, mTracks.size()); 2426 Mutex::Autolock _l(mLock); 2427 2428 size_t size = mTracks.size(); 2429 for (size_t i = 0; i < size; i++) { 2430 sp<Track> t = mTracks[i]; 2431 if (t->streamType() == streamType) { 2432 t->invalidate(); 2433 } 2434 } 2435} 2436 2437status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2438{ 2439 int session = chain->sessionId(); 2440 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2441 ? mEffectBuffer : mSinkBuffer); 2442 bool ownsBuffer = false; 2443 2444 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2445 if (session > 0) { 2446 // Only one effect chain can be present in direct output thread and it uses 2447 // the sink buffer as input 2448 if (mType != DIRECT) { 2449 size_t numSamples = mNormalFrameCount * mChannelCount; 2450 buffer = new int16_t[numSamples]; 2451 memset(buffer, 0, numSamples * sizeof(int16_t)); 2452 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2453 ownsBuffer = true; 2454 } 2455 2456 // Attach all tracks with same session ID to this chain. 2457 for (size_t i = 0; i < mTracks.size(); ++i) { 2458 sp<Track> track = mTracks[i]; 2459 if (session == track->sessionId()) { 2460 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2461 buffer); 2462 track->setMainBuffer(buffer); 2463 chain->incTrackCnt(); 2464 } 2465 } 2466 2467 // indicate all active tracks in the chain 2468 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2469 sp<Track> track = mActiveTracks[i].promote(); 2470 if (track == 0) { 2471 continue; 2472 } 2473 if (session == track->sessionId()) { 2474 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2475 chain->incActiveTrackCnt(); 2476 } 2477 } 2478 } 2479 chain->setThread(this); 2480 chain->setInBuffer(buffer, ownsBuffer); 2481 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2482 ? mEffectBuffer : mSinkBuffer)); 2483 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2484 // chains list in order to be processed last as it contains output stage effects 2485 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2486 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2487 // after track specific effects and before output stage 2488 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2489 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2490 // Effect chain for other sessions are inserted at beginning of effect 2491 // chains list to be processed before output mix effects. Relative order between other 2492 // sessions is not important 2493 size_t size = mEffectChains.size(); 2494 size_t i = 0; 2495 for (i = 0; i < size; i++) { 2496 if (mEffectChains[i]->sessionId() < session) { 2497 break; 2498 } 2499 } 2500 mEffectChains.insertAt(chain, i); 2501 checkSuspendOnAddEffectChain_l(chain); 2502 2503 return NO_ERROR; 2504} 2505 2506size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2507{ 2508 int session = chain->sessionId(); 2509 2510 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2511 2512 for (size_t i = 0; i < mEffectChains.size(); i++) { 2513 if (chain == mEffectChains[i]) { 2514 mEffectChains.removeAt(i); 2515 // detach all active tracks from the chain 2516 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2517 sp<Track> track = mActiveTracks[i].promote(); 2518 if (track == 0) { 2519 continue; 2520 } 2521 if (session == track->sessionId()) { 2522 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2523 chain.get(), session); 2524 chain->decActiveTrackCnt(); 2525 } 2526 } 2527 2528 // detach all tracks with same session ID from this chain 2529 for (size_t i = 0; i < mTracks.size(); ++i) { 2530 sp<Track> track = mTracks[i]; 2531 if (session == track->sessionId()) { 2532 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2533 chain->decTrackCnt(); 2534 } 2535 } 2536 break; 2537 } 2538 } 2539 return mEffectChains.size(); 2540} 2541 2542status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2543 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2544{ 2545 Mutex::Autolock _l(mLock); 2546 return attachAuxEffect_l(track, EffectId); 2547} 2548 2549status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2550 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2551{ 2552 status_t status = NO_ERROR; 2553 2554 if (EffectId == 0) { 2555 track->setAuxBuffer(0, NULL); 2556 } else { 2557 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2558 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2559 if (effect != 0) { 2560 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2561 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2562 } else { 2563 status = INVALID_OPERATION; 2564 } 2565 } else { 2566 status = BAD_VALUE; 2567 } 2568 } 2569 return status; 2570} 2571 2572void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2573{ 2574 for (size_t i = 0; i < mTracks.size(); ++i) { 2575 sp<Track> track = mTracks[i]; 2576 if (track->auxEffectId() == effectId) { 2577 attachAuxEffect_l(track, 0); 2578 } 2579 } 2580} 2581 2582bool AudioFlinger::PlaybackThread::threadLoop() 2583{ 2584 Vector< sp<Track> > tracksToRemove; 2585 2586 standbyTime = systemTime(); 2587 2588 // MIXER 2589 nsecs_t lastWarning = 0; 2590 2591 // DUPLICATING 2592 // FIXME could this be made local to while loop? 2593 writeFrames = 0; 2594 2595 int lastGeneration = 0; 2596 2597 cacheParameters_l(); 2598 sleepTime = idleSleepTime; 2599 2600 if (mType == MIXER) { 2601 sleepTimeShift = 0; 2602 } 2603 2604 CpuStats cpuStats; 2605 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2606 2607 acquireWakeLock(); 2608 2609 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2610 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2611 // and then that string will be logged at the next convenient opportunity. 2612 const char *logString = NULL; 2613 2614 checkSilentMode_l(); 2615 2616 while (!exitPending()) 2617 { 2618 cpuStats.sample(myName); 2619 2620 Vector< sp<EffectChain> > effectChains; 2621 2622 { // scope for mLock 2623 2624 Mutex::Autolock _l(mLock); 2625 2626 processConfigEvents_l(); 2627 2628 if (logString != NULL) { 2629 mNBLogWriter->logTimestamp(); 2630 mNBLogWriter->log(logString); 2631 logString = NULL; 2632 } 2633 2634 // Gather the framesReleased counters for all active tracks, 2635 // and latch them atomically with the timestamp. 2636 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2637 mLatchD.mFramesReleased.clear(); 2638 size_t size = mActiveTracks.size(); 2639 for (size_t i = 0; i < size; i++) { 2640 sp<Track> t = mActiveTracks[i].promote(); 2641 if (t != 0) { 2642 mLatchD.mFramesReleased.add(t.get(), 2643 t->mAudioTrackServerProxy->framesReleased()); 2644 } 2645 } 2646 if (mLatchDValid) { 2647 mLatchQ = mLatchD; 2648 mLatchDValid = false; 2649 mLatchQValid = true; 2650 } 2651 2652 saveOutputTracks(); 2653 if (mSignalPending) { 2654 // A signal was raised while we were unlocked 2655 mSignalPending = false; 2656 } else if (waitingAsyncCallback_l()) { 2657 if (exitPending()) { 2658 break; 2659 } 2660 releaseWakeLock_l(); 2661 mWakeLockUids.clear(); 2662 mActiveTracksGeneration++; 2663 ALOGV("wait async completion"); 2664 mWaitWorkCV.wait(mLock); 2665 ALOGV("async completion/wake"); 2666 acquireWakeLock_l(); 2667 standbyTime = systemTime() + standbyDelay; 2668 sleepTime = 0; 2669 2670 continue; 2671 } 2672 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2673 isSuspended()) { 2674 // put audio hardware into standby after short delay 2675 if (shouldStandby_l()) { 2676 2677 threadLoop_standby(); 2678 2679 mStandby = true; 2680 } 2681 2682 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2683 // we're about to wait, flush the binder command buffer 2684 IPCThreadState::self()->flushCommands(); 2685 2686 clearOutputTracks(); 2687 2688 if (exitPending()) { 2689 break; 2690 } 2691 2692 releaseWakeLock_l(); 2693 mWakeLockUids.clear(); 2694 mActiveTracksGeneration++; 2695 // wait until we have something to do... 2696 ALOGV("%s going to sleep", myName.string()); 2697 mWaitWorkCV.wait(mLock); 2698 ALOGV("%s waking up", myName.string()); 2699 acquireWakeLock_l(); 2700 2701 mMixerStatus = MIXER_IDLE; 2702 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2703 mBytesWritten = 0; 2704 mBytesRemaining = 0; 2705 checkSilentMode_l(); 2706 2707 standbyTime = systemTime() + standbyDelay; 2708 sleepTime = idleSleepTime; 2709 if (mType == MIXER) { 2710 sleepTimeShift = 0; 2711 } 2712 2713 continue; 2714 } 2715 } 2716 // mMixerStatusIgnoringFastTracks is also updated internally 2717 mMixerStatus = prepareTracks_l(&tracksToRemove); 2718 2719 // compare with previously applied list 2720 if (lastGeneration != mActiveTracksGeneration) { 2721 // update wakelock 2722 updateWakeLockUids_l(mWakeLockUids); 2723 lastGeneration = mActiveTracksGeneration; 2724 } 2725 2726 // prevent any changes in effect chain list and in each effect chain 2727 // during mixing and effect process as the audio buffers could be deleted 2728 // or modified if an effect is created or deleted 2729 lockEffectChains_l(effectChains); 2730 } // mLock scope ends 2731 2732 if (mBytesRemaining == 0) { 2733 mCurrentWriteLength = 0; 2734 if (mMixerStatus == MIXER_TRACKS_READY) { 2735 // threadLoop_mix() sets mCurrentWriteLength 2736 threadLoop_mix(); 2737 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2738 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2739 // threadLoop_sleepTime sets sleepTime to 0 if data 2740 // must be written to HAL 2741 threadLoop_sleepTime(); 2742 if (sleepTime == 0) { 2743 mCurrentWriteLength = mSinkBufferSize; 2744 } 2745 } 2746 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2747 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2748 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2749 // or mSinkBuffer (if there are no effects). 2750 // 2751 // This is done pre-effects computation; if effects change to 2752 // support higher precision, this needs to move. 2753 // 2754 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2755 // TODO use sleepTime == 0 as an additional condition. 2756 if (mMixerBufferValid) { 2757 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2758 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2759 2760 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2761 mNormalFrameCount * mChannelCount); 2762 } 2763 2764 mBytesRemaining = mCurrentWriteLength; 2765 if (isSuspended()) { 2766 sleepTime = suspendSleepTimeUs(); 2767 // simulate write to HAL when suspended 2768 mBytesWritten += mSinkBufferSize; 2769 mBytesRemaining = 0; 2770 } 2771 2772 // only process effects if we're going to write 2773 if (sleepTime == 0 && mType != OFFLOAD) { 2774 for (size_t i = 0; i < effectChains.size(); i ++) { 2775 effectChains[i]->process_l(); 2776 } 2777 } 2778 } 2779 // Process effect chains for offloaded thread even if no audio 2780 // was read from audio track: process only updates effect state 2781 // and thus does have to be synchronized with audio writes but may have 2782 // to be called while waiting for async write callback 2783 if (mType == OFFLOAD) { 2784 for (size_t i = 0; i < effectChains.size(); i ++) { 2785 effectChains[i]->process_l(); 2786 } 2787 } 2788 2789 // Only if the Effects buffer is enabled and there is data in the 2790 // Effects buffer (buffer valid), we need to 2791 // copy into the sink buffer. 2792 // TODO use sleepTime == 0 as an additional condition. 2793 if (mEffectBufferValid) { 2794 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2795 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2796 mNormalFrameCount * mChannelCount); 2797 } 2798 2799 // enable changes in effect chain 2800 unlockEffectChains(effectChains); 2801 2802 if (!waitingAsyncCallback()) { 2803 // sleepTime == 0 means we must write to audio hardware 2804 if (sleepTime == 0) { 2805 if (mBytesRemaining) { 2806 ssize_t ret = threadLoop_write(); 2807 if (ret < 0) { 2808 mBytesRemaining = 0; 2809 } else { 2810 mBytesWritten += ret; 2811 mBytesRemaining -= ret; 2812 } 2813 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2814 (mMixerStatus == MIXER_DRAIN_ALL)) { 2815 threadLoop_drain(); 2816 } 2817 if (mType == MIXER) { 2818 // write blocked detection 2819 nsecs_t now = systemTime(); 2820 nsecs_t delta = now - mLastWriteTime; 2821 if (!mStandby && delta > maxPeriod) { 2822 mNumDelayedWrites++; 2823 if ((now - lastWarning) > kWarningThrottleNs) { 2824 ATRACE_NAME("underrun"); 2825 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2826 ns2ms(delta), mNumDelayedWrites, this); 2827 lastWarning = now; 2828 } 2829 } 2830 } 2831 2832 } else { 2833 ATRACE_BEGIN("sleep"); 2834 usleep(sleepTime); 2835 ATRACE_END(); 2836 } 2837 } 2838 2839 // Finally let go of removed track(s), without the lock held 2840 // since we can't guarantee the destructors won't acquire that 2841 // same lock. This will also mutate and push a new fast mixer state. 2842 threadLoop_removeTracks(tracksToRemove); 2843 tracksToRemove.clear(); 2844 2845 // FIXME I don't understand the need for this here; 2846 // it was in the original code but maybe the 2847 // assignment in saveOutputTracks() makes this unnecessary? 2848 clearOutputTracks(); 2849 2850 // Effect chains will be actually deleted here if they were removed from 2851 // mEffectChains list during mixing or effects processing 2852 effectChains.clear(); 2853 2854 // FIXME Note that the above .clear() is no longer necessary since effectChains 2855 // is now local to this block, but will keep it for now (at least until merge done). 2856 } 2857 2858 threadLoop_exit(); 2859 2860 if (!mStandby) { 2861 threadLoop_standby(); 2862 mStandby = true; 2863 } 2864 2865 releaseWakeLock(); 2866 mWakeLockUids.clear(); 2867 mActiveTracksGeneration++; 2868 2869 ALOGV("Thread %p type %d exiting", this, mType); 2870 return false; 2871} 2872 2873// removeTracks_l() must be called with ThreadBase::mLock held 2874void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2875{ 2876 size_t count = tracksToRemove.size(); 2877 if (count > 0) { 2878 for (size_t i=0 ; i<count ; i++) { 2879 const sp<Track>& track = tracksToRemove.itemAt(i); 2880 mActiveTracks.remove(track); 2881 mWakeLockUids.remove(track->uid()); 2882 mActiveTracksGeneration++; 2883 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2884 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2885 if (chain != 0) { 2886 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2887 track->sessionId()); 2888 chain->decActiveTrackCnt(); 2889 } 2890 if (track->isTerminated()) { 2891 removeTrack_l(track); 2892 } 2893 } 2894 } 2895 2896} 2897 2898status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2899{ 2900 if (mNormalSink != 0) { 2901 return mNormalSink->getTimestamp(timestamp); 2902 } 2903 if ((mType == OFFLOAD || mType == DIRECT) 2904 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2905 uint64_t position64; 2906 int ret = mOutput->stream->get_presentation_position( 2907 mOutput->stream, &position64, ×tamp.mTime); 2908 if (ret == 0) { 2909 timestamp.mPosition = (uint32_t)position64; 2910 return NO_ERROR; 2911 } 2912 } 2913 return INVALID_OPERATION; 2914} 2915 2916status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2917 audio_patch_handle_t *handle) 2918{ 2919 status_t status = NO_ERROR; 2920 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2921 // store new device and send to effects 2922 audio_devices_t type = AUDIO_DEVICE_NONE; 2923 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2924 type |= patch->sinks[i].ext.device.type; 2925 } 2926 mOutDevice = type; 2927 for (size_t i = 0; i < mEffectChains.size(); i++) { 2928 mEffectChains[i]->setDevice_l(mOutDevice); 2929 } 2930 2931 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2932 status = hwDevice->create_audio_patch(hwDevice, 2933 patch->num_sources, 2934 patch->sources, 2935 patch->num_sinks, 2936 patch->sinks, 2937 handle); 2938 } else { 2939 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2940 } 2941 return status; 2942} 2943 2944status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2945{ 2946 status_t status = NO_ERROR; 2947 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2948 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2949 status = hwDevice->release_audio_patch(hwDevice, handle); 2950 } else { 2951 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2952 } 2953 return status; 2954} 2955 2956void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2957{ 2958 Mutex::Autolock _l(mLock); 2959 mTracks.add(track); 2960} 2961 2962void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2963{ 2964 Mutex::Autolock _l(mLock); 2965 destroyTrack_l(track); 2966} 2967 2968void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2969{ 2970 ThreadBase::getAudioPortConfig(config); 2971 config->role = AUDIO_PORT_ROLE_SOURCE; 2972 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2973 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2974} 2975 2976// ---------------------------------------------------------------------------- 2977 2978AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2979 audio_io_handle_t id, audio_devices_t device, type_t type) 2980 : PlaybackThread(audioFlinger, output, id, device, type), 2981 // mAudioMixer below 2982 // mFastMixer below 2983 mFastMixerFutex(0) 2984 // mOutputSink below 2985 // mPipeSink below 2986 // mNormalSink below 2987{ 2988 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2989 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2990 "mFrameCount=%d, mNormalFrameCount=%d", 2991 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2992 mNormalFrameCount); 2993 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2994 2995 if (type == DUPLICATING) { 2996 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 2997 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 2998 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 2999 return; 3000 } 3001 // create an NBAIO sink for the HAL output stream, and negotiate 3002 mOutputSink = new AudioStreamOutSink(output->stream); 3003 size_t numCounterOffers = 0; 3004 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3005 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3006 ALOG_ASSERT(index == 0); 3007 3008 // initialize fast mixer depending on configuration 3009 bool initFastMixer; 3010 switch (kUseFastMixer) { 3011 case FastMixer_Never: 3012 initFastMixer = false; 3013 break; 3014 case FastMixer_Always: 3015 initFastMixer = true; 3016 break; 3017 case FastMixer_Static: 3018 case FastMixer_Dynamic: 3019 initFastMixer = mFrameCount < mNormalFrameCount; 3020 break; 3021 } 3022 if (initFastMixer) { 3023 audio_format_t fastMixerFormat; 3024 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3025 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3026 } else { 3027 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3028 } 3029 if (mFormat != fastMixerFormat) { 3030 // change our Sink format to accept our intermediate precision 3031 mFormat = fastMixerFormat; 3032 free(mSinkBuffer); 3033 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3034 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3035 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3036 } 3037 3038 // create a MonoPipe to connect our submix to FastMixer 3039 NBAIO_Format format = mOutputSink->format(); 3040 NBAIO_Format origformat = format; 3041 // adjust format to match that of the Fast Mixer 3042 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3043 format.mFormat = fastMixerFormat; 3044 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3045 3046 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3047 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3048 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3049 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3050 const NBAIO_Format offers[1] = {format}; 3051 size_t numCounterOffers = 0; 3052 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3053 ALOG_ASSERT(index == 0); 3054 monoPipe->setAvgFrames((mScreenState & 1) ? 3055 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3056 mPipeSink = monoPipe; 3057 3058#ifdef TEE_SINK 3059 if (mTeeSinkOutputEnabled) { 3060 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3061 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3062 const NBAIO_Format offers2[1] = {origformat}; 3063 numCounterOffers = 0; 3064 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3065 ALOG_ASSERT(index == 0); 3066 mTeeSink = teeSink; 3067 PipeReader *teeSource = new PipeReader(*teeSink); 3068 numCounterOffers = 0; 3069 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3070 ALOG_ASSERT(index == 0); 3071 mTeeSource = teeSource; 3072 } 3073#endif 3074 3075 // create fast mixer and configure it initially with just one fast track for our submix 3076 mFastMixer = new FastMixer(); 3077 FastMixerStateQueue *sq = mFastMixer->sq(); 3078#ifdef STATE_QUEUE_DUMP 3079 sq->setObserverDump(&mStateQueueObserverDump); 3080 sq->setMutatorDump(&mStateQueueMutatorDump); 3081#endif 3082 FastMixerState *state = sq->begin(); 3083 FastTrack *fastTrack = &state->mFastTracks[0]; 3084 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3085 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3086 fastTrack->mVolumeProvider = NULL; 3087 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3088 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3089 fastTrack->mGeneration++; 3090 state->mFastTracksGen++; 3091 state->mTrackMask = 1; 3092 // fast mixer will use the HAL output sink 3093 state->mOutputSink = mOutputSink.get(); 3094 state->mOutputSinkGen++; 3095 state->mFrameCount = mFrameCount; 3096 state->mCommand = FastMixerState::COLD_IDLE; 3097 // already done in constructor initialization list 3098 //mFastMixerFutex = 0; 3099 state->mColdFutexAddr = &mFastMixerFutex; 3100 state->mColdGen++; 3101 state->mDumpState = &mFastMixerDumpState; 3102#ifdef TEE_SINK 3103 state->mTeeSink = mTeeSink.get(); 3104#endif 3105 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3106 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3107 sq->end(); 3108 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3109 3110 // start the fast mixer 3111 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3112 pid_t tid = mFastMixer->getTid(); 3113 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3114 if (err != 0) { 3115 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3116 kPriorityFastMixer, getpid_cached, tid, err); 3117 } 3118 3119#ifdef AUDIO_WATCHDOG 3120 // create and start the watchdog 3121 mAudioWatchdog = new AudioWatchdog(); 3122 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3123 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3124 tid = mAudioWatchdog->getTid(); 3125 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3126 if (err != 0) { 3127 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3128 kPriorityFastMixer, getpid_cached, tid, err); 3129 } 3130#endif 3131 3132 } 3133 3134 switch (kUseFastMixer) { 3135 case FastMixer_Never: 3136 case FastMixer_Dynamic: 3137 mNormalSink = mOutputSink; 3138 break; 3139 case FastMixer_Always: 3140 mNormalSink = mPipeSink; 3141 break; 3142 case FastMixer_Static: 3143 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3144 break; 3145 } 3146} 3147 3148AudioFlinger::MixerThread::~MixerThread() 3149{ 3150 if (mFastMixer != 0) { 3151 FastMixerStateQueue *sq = mFastMixer->sq(); 3152 FastMixerState *state = sq->begin(); 3153 if (state->mCommand == FastMixerState::COLD_IDLE) { 3154 int32_t old = android_atomic_inc(&mFastMixerFutex); 3155 if (old == -1) { 3156 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3157 } 3158 } 3159 state->mCommand = FastMixerState::EXIT; 3160 sq->end(); 3161 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3162 mFastMixer->join(); 3163 // Though the fast mixer thread has exited, it's state queue is still valid. 3164 // We'll use that extract the final state which contains one remaining fast track 3165 // corresponding to our sub-mix. 3166 state = sq->begin(); 3167 ALOG_ASSERT(state->mTrackMask == 1); 3168 FastTrack *fastTrack = &state->mFastTracks[0]; 3169 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3170 delete fastTrack->mBufferProvider; 3171 sq->end(false /*didModify*/); 3172 mFastMixer.clear(); 3173#ifdef AUDIO_WATCHDOG 3174 if (mAudioWatchdog != 0) { 3175 mAudioWatchdog->requestExit(); 3176 mAudioWatchdog->requestExitAndWait(); 3177 mAudioWatchdog.clear(); 3178 } 3179#endif 3180 } 3181 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3182 delete mAudioMixer; 3183} 3184 3185 3186uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3187{ 3188 if (mFastMixer != 0) { 3189 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3190 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3191 } 3192 return latency; 3193} 3194 3195 3196void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3197{ 3198 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3199} 3200 3201ssize_t AudioFlinger::MixerThread::threadLoop_write() 3202{ 3203 // FIXME we should only do one push per cycle; confirm this is true 3204 // Start the fast mixer if it's not already running 3205 if (mFastMixer != 0) { 3206 FastMixerStateQueue *sq = mFastMixer->sq(); 3207 FastMixerState *state = sq->begin(); 3208 if (state->mCommand != FastMixerState::MIX_WRITE && 3209 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3210 if (state->mCommand == FastMixerState::COLD_IDLE) { 3211 int32_t old = android_atomic_inc(&mFastMixerFutex); 3212 if (old == -1) { 3213 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3214 } 3215#ifdef AUDIO_WATCHDOG 3216 if (mAudioWatchdog != 0) { 3217 mAudioWatchdog->resume(); 3218 } 3219#endif 3220 } 3221 state->mCommand = FastMixerState::MIX_WRITE; 3222#ifdef FAST_THREAD_STATISTICS 3223 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3224 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3225#endif 3226 sq->end(); 3227 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3228 if (kUseFastMixer == FastMixer_Dynamic) { 3229 mNormalSink = mPipeSink; 3230 } 3231 } else { 3232 sq->end(false /*didModify*/); 3233 } 3234 } 3235 return PlaybackThread::threadLoop_write(); 3236} 3237 3238void AudioFlinger::MixerThread::threadLoop_standby() 3239{ 3240 // Idle the fast mixer if it's currently running 3241 if (mFastMixer != 0) { 3242 FastMixerStateQueue *sq = mFastMixer->sq(); 3243 FastMixerState *state = sq->begin(); 3244 if (!(state->mCommand & FastMixerState::IDLE)) { 3245 state->mCommand = FastMixerState::COLD_IDLE; 3246 state->mColdFutexAddr = &mFastMixerFutex; 3247 state->mColdGen++; 3248 mFastMixerFutex = 0; 3249 sq->end(); 3250 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3251 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3252 if (kUseFastMixer == FastMixer_Dynamic) { 3253 mNormalSink = mOutputSink; 3254 } 3255#ifdef AUDIO_WATCHDOG 3256 if (mAudioWatchdog != 0) { 3257 mAudioWatchdog->pause(); 3258 } 3259#endif 3260 } else { 3261 sq->end(false /*didModify*/); 3262 } 3263 } 3264 PlaybackThread::threadLoop_standby(); 3265} 3266 3267bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3268{ 3269 return false; 3270} 3271 3272bool AudioFlinger::PlaybackThread::shouldStandby_l() 3273{ 3274 return !mStandby; 3275} 3276 3277bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3278{ 3279 Mutex::Autolock _l(mLock); 3280 return waitingAsyncCallback_l(); 3281} 3282 3283// shared by MIXER and DIRECT, overridden by DUPLICATING 3284void AudioFlinger::PlaybackThread::threadLoop_standby() 3285{ 3286 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3287 mOutput->stream->common.standby(&mOutput->stream->common); 3288 if (mUseAsyncWrite != 0) { 3289 // discard any pending drain or write ack by incrementing sequence 3290 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3291 mDrainSequence = (mDrainSequence + 2) & ~1; 3292 ALOG_ASSERT(mCallbackThread != 0); 3293 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3294 mCallbackThread->setDraining(mDrainSequence); 3295 } 3296 mHwPaused = false; 3297} 3298 3299void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3300{ 3301 ALOGV("signal playback thread"); 3302 broadcast_l(); 3303} 3304 3305void AudioFlinger::MixerThread::threadLoop_mix() 3306{ 3307 // obtain the presentation timestamp of the next output buffer 3308 int64_t pts; 3309 status_t status = INVALID_OPERATION; 3310 3311 if (mNormalSink != 0) { 3312 status = mNormalSink->getNextWriteTimestamp(&pts); 3313 } else { 3314 status = mOutputSink->getNextWriteTimestamp(&pts); 3315 } 3316 3317 if (status != NO_ERROR) { 3318 pts = AudioBufferProvider::kInvalidPTS; 3319 } 3320 3321 // mix buffers... 3322 mAudioMixer->process(pts); 3323 mCurrentWriteLength = mSinkBufferSize; 3324 // increase sleep time progressively when application underrun condition clears. 3325 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3326 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3327 // such that we would underrun the audio HAL. 3328 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3329 sleepTimeShift--; 3330 } 3331 sleepTime = 0; 3332 standbyTime = systemTime() + standbyDelay; 3333 //TODO: delay standby when effects have a tail 3334 3335} 3336 3337void AudioFlinger::MixerThread::threadLoop_sleepTime() 3338{ 3339 // If no tracks are ready, sleep once for the duration of an output 3340 // buffer size, then write 0s to the output 3341 if (sleepTime == 0) { 3342 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3343 sleepTime = activeSleepTime >> sleepTimeShift; 3344 if (sleepTime < kMinThreadSleepTimeUs) { 3345 sleepTime = kMinThreadSleepTimeUs; 3346 } 3347 // reduce sleep time in case of consecutive application underruns to avoid 3348 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3349 // duration we would end up writing less data than needed by the audio HAL if 3350 // the condition persists. 3351 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3352 sleepTimeShift++; 3353 } 3354 } else { 3355 sleepTime = idleSleepTime; 3356 } 3357 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3358 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3359 // before effects processing or output. 3360 if (mMixerBufferValid) { 3361 memset(mMixerBuffer, 0, mMixerBufferSize); 3362 } else { 3363 memset(mSinkBuffer, 0, mSinkBufferSize); 3364 } 3365 sleepTime = 0; 3366 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3367 "anticipated start"); 3368 } 3369 // TODO add standby time extension fct of effect tail 3370} 3371 3372// prepareTracks_l() must be called with ThreadBase::mLock held 3373AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3374 Vector< sp<Track> > *tracksToRemove) 3375{ 3376 3377 mixer_state mixerStatus = MIXER_IDLE; 3378 // find out which tracks need to be processed 3379 size_t count = mActiveTracks.size(); 3380 size_t mixedTracks = 0; 3381 size_t tracksWithEffect = 0; 3382 // counts only _active_ fast tracks 3383 size_t fastTracks = 0; 3384 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3385 3386 float masterVolume = mMasterVolume; 3387 bool masterMute = mMasterMute; 3388 3389 if (masterMute) { 3390 masterVolume = 0; 3391 } 3392 // Delegate master volume control to effect in output mix effect chain if needed 3393 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3394 if (chain != 0) { 3395 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3396 chain->setVolume_l(&v, &v); 3397 masterVolume = (float)((v + (1 << 23)) >> 24); 3398 chain.clear(); 3399 } 3400 3401 // prepare a new state to push 3402 FastMixerStateQueue *sq = NULL; 3403 FastMixerState *state = NULL; 3404 bool didModify = false; 3405 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3406 if (mFastMixer != 0) { 3407 sq = mFastMixer->sq(); 3408 state = sq->begin(); 3409 } 3410 3411 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3412 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3413 3414 for (size_t i=0 ; i<count ; i++) { 3415 const sp<Track> t = mActiveTracks[i].promote(); 3416 if (t == 0) { 3417 continue; 3418 } 3419 3420 // this const just means the local variable doesn't change 3421 Track* const track = t.get(); 3422 3423 // process fast tracks 3424 if (track->isFastTrack()) { 3425 3426 // It's theoretically possible (though unlikely) for a fast track to be created 3427 // and then removed within the same normal mix cycle. This is not a problem, as 3428 // the track never becomes active so it's fast mixer slot is never touched. 3429 // The converse, of removing an (active) track and then creating a new track 3430 // at the identical fast mixer slot within the same normal mix cycle, 3431 // is impossible because the slot isn't marked available until the end of each cycle. 3432 int j = track->mFastIndex; 3433 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3434 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3435 FastTrack *fastTrack = &state->mFastTracks[j]; 3436 3437 // Determine whether the track is currently in underrun condition, 3438 // and whether it had a recent underrun. 3439 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3440 FastTrackUnderruns underruns = ftDump->mUnderruns; 3441 uint32_t recentFull = (underruns.mBitFields.mFull - 3442 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3443 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3444 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3445 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3446 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3447 uint32_t recentUnderruns = recentPartial + recentEmpty; 3448 track->mObservedUnderruns = underruns; 3449 // don't count underruns that occur while stopping or pausing 3450 // or stopped which can occur when flush() is called while active 3451 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3452 recentUnderruns > 0) { 3453 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3454 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3455 } 3456 3457 // This is similar to the state machine for normal tracks, 3458 // with a few modifications for fast tracks. 3459 bool isActive = true; 3460 switch (track->mState) { 3461 case TrackBase::STOPPING_1: 3462 // track stays active in STOPPING_1 state until first underrun 3463 if (recentUnderruns > 0 || track->isTerminated()) { 3464 track->mState = TrackBase::STOPPING_2; 3465 } 3466 break; 3467 case TrackBase::PAUSING: 3468 // ramp down is not yet implemented 3469 track->setPaused(); 3470 break; 3471 case TrackBase::RESUMING: 3472 // ramp up is not yet implemented 3473 track->mState = TrackBase::ACTIVE; 3474 break; 3475 case TrackBase::ACTIVE: 3476 if (recentFull > 0 || recentPartial > 0) { 3477 // track has provided at least some frames recently: reset retry count 3478 track->mRetryCount = kMaxTrackRetries; 3479 } 3480 if (recentUnderruns == 0) { 3481 // no recent underruns: stay active 3482 break; 3483 } 3484 // there has recently been an underrun of some kind 3485 if (track->sharedBuffer() == 0) { 3486 // were any of the recent underruns "empty" (no frames available)? 3487 if (recentEmpty == 0) { 3488 // no, then ignore the partial underruns as they are allowed indefinitely 3489 break; 3490 } 3491 // there has recently been an "empty" underrun: decrement the retry counter 3492 if (--(track->mRetryCount) > 0) { 3493 break; 3494 } 3495 // indicate to client process that the track was disabled because of underrun; 3496 // it will then automatically call start() when data is available 3497 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3498 // remove from active list, but state remains ACTIVE [confusing but true] 3499 isActive = false; 3500 break; 3501 } 3502 // fall through 3503 case TrackBase::STOPPING_2: 3504 case TrackBase::PAUSED: 3505 case TrackBase::STOPPED: 3506 case TrackBase::FLUSHED: // flush() while active 3507 // Check for presentation complete if track is inactive 3508 // We have consumed all the buffers of this track. 3509 // This would be incomplete if we auto-paused on underrun 3510 { 3511 size_t audioHALFrames = 3512 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3513 size_t framesWritten = mBytesWritten / mFrameSize; 3514 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3515 // track stays in active list until presentation is complete 3516 break; 3517 } 3518 } 3519 if (track->isStopping_2()) { 3520 track->mState = TrackBase::STOPPED; 3521 } 3522 if (track->isStopped()) { 3523 // Can't reset directly, as fast mixer is still polling this track 3524 // track->reset(); 3525 // So instead mark this track as needing to be reset after push with ack 3526 resetMask |= 1 << i; 3527 } 3528 isActive = false; 3529 break; 3530 case TrackBase::IDLE: 3531 default: 3532 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3533 } 3534 3535 if (isActive) { 3536 // was it previously inactive? 3537 if (!(state->mTrackMask & (1 << j))) { 3538 ExtendedAudioBufferProvider *eabp = track; 3539 VolumeProvider *vp = track; 3540 fastTrack->mBufferProvider = eabp; 3541 fastTrack->mVolumeProvider = vp; 3542 fastTrack->mChannelMask = track->mChannelMask; 3543 fastTrack->mFormat = track->mFormat; 3544 fastTrack->mGeneration++; 3545 state->mTrackMask |= 1 << j; 3546 didModify = true; 3547 // no acknowledgement required for newly active tracks 3548 } 3549 // cache the combined master volume and stream type volume for fast mixer; this 3550 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3551 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3552 ++fastTracks; 3553 } else { 3554 // was it previously active? 3555 if (state->mTrackMask & (1 << j)) { 3556 fastTrack->mBufferProvider = NULL; 3557 fastTrack->mGeneration++; 3558 state->mTrackMask &= ~(1 << j); 3559 didModify = true; 3560 // If any fast tracks were removed, we must wait for acknowledgement 3561 // because we're about to decrement the last sp<> on those tracks. 3562 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3563 } else { 3564 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3565 } 3566 tracksToRemove->add(track); 3567 // Avoids a misleading display in dumpsys 3568 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3569 } 3570 continue; 3571 } 3572 3573 { // local variable scope to avoid goto warning 3574 3575 audio_track_cblk_t* cblk = track->cblk(); 3576 3577 // The first time a track is added we wait 3578 // for all its buffers to be filled before processing it 3579 int name = track->name(); 3580 // make sure that we have enough frames to mix one full buffer. 3581 // enforce this condition only once to enable draining the buffer in case the client 3582 // app does not call stop() and relies on underrun to stop: 3583 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3584 // during last round 3585 size_t desiredFrames; 3586 uint32_t sr = track->sampleRate(); 3587 if (sr == mSampleRate) { 3588 desiredFrames = mNormalFrameCount; 3589 } else { 3590 desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); 3591 // add frames already consumed but not yet released by the resampler 3592 // because mAudioTrackServerProxy->framesReady() will include these frames 3593 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3594#if 0 3595 // the minimum track buffer size is normally twice the number of frames necessary 3596 // to fill one buffer and the resampler should not leave more than one buffer worth 3597 // of unreleased frames after each pass, but just in case... 3598 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3599#endif 3600 } 3601 uint32_t minFrames = 1; 3602 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3603 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3604 minFrames = desiredFrames; 3605 } 3606 3607 size_t framesReady = track->framesReady(); 3608 if (ATRACE_ENABLED()) { 3609 // I wish we had formatted trace names 3610 char traceName[16]; 3611 strcpy(traceName, "nRdy"); 3612 int name = track->name(); 3613 if (AudioMixer::TRACK0 <= name && 3614 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3615 name -= AudioMixer::TRACK0; 3616 traceName[4] = (name / 10) + '0'; 3617 traceName[5] = (name % 10) + '0'; 3618 } else { 3619 traceName[4] = '?'; 3620 traceName[5] = '?'; 3621 } 3622 traceName[6] = '\0'; 3623 ATRACE_INT(traceName, framesReady); 3624 } 3625 if ((framesReady >= minFrames) && track->isReady() && 3626 !track->isPaused() && !track->isTerminated()) 3627 { 3628 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3629 3630 mixedTracks++; 3631 3632 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3633 // there is an effect chain connected to the track 3634 chain.clear(); 3635 if (track->mainBuffer() != mSinkBuffer && 3636 track->mainBuffer() != mMixerBuffer) { 3637 if (mEffectBufferEnabled) { 3638 mEffectBufferValid = true; // Later can set directly. 3639 } 3640 chain = getEffectChain_l(track->sessionId()); 3641 // Delegate volume control to effect in track effect chain if needed 3642 if (chain != 0) { 3643 tracksWithEffect++; 3644 } else { 3645 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3646 "session %d", 3647 name, track->sessionId()); 3648 } 3649 } 3650 3651 3652 int param = AudioMixer::VOLUME; 3653 if (track->mFillingUpStatus == Track::FS_FILLED) { 3654 // no ramp for the first volume setting 3655 track->mFillingUpStatus = Track::FS_ACTIVE; 3656 if (track->mState == TrackBase::RESUMING) { 3657 track->mState = TrackBase::ACTIVE; 3658 param = AudioMixer::RAMP_VOLUME; 3659 } 3660 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3661 // FIXME should not make a decision based on mServer 3662 } else if (cblk->mServer != 0) { 3663 // If the track is stopped before the first frame was mixed, 3664 // do not apply ramp 3665 param = AudioMixer::RAMP_VOLUME; 3666 } 3667 3668 // compute volume for this track 3669 uint32_t vl, vr; // in U8.24 integer format 3670 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3671 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3672 vl = vr = 0; 3673 vlf = vrf = vaf = 0.; 3674 if (track->isPausing()) { 3675 track->setPaused(); 3676 } 3677 } else { 3678 3679 // read original volumes with volume control 3680 float typeVolume = mStreamTypes[track->streamType()].volume; 3681 float v = masterVolume * typeVolume; 3682 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3683 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3684 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3685 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3686 // track volumes come from shared memory, so can't be trusted and must be clamped 3687 if (vlf > GAIN_FLOAT_UNITY) { 3688 ALOGV("Track left volume out of range: %.3g", vlf); 3689 vlf = GAIN_FLOAT_UNITY; 3690 } 3691 if (vrf > GAIN_FLOAT_UNITY) { 3692 ALOGV("Track right volume out of range: %.3g", vrf); 3693 vrf = GAIN_FLOAT_UNITY; 3694 } 3695 // now apply the master volume and stream type volume 3696 vlf *= v; 3697 vrf *= v; 3698 // assuming master volume and stream type volume each go up to 1.0, 3699 // then derive vl and vr as U8.24 versions for the effect chain 3700 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3701 vl = (uint32_t) (scaleto8_24 * vlf); 3702 vr = (uint32_t) (scaleto8_24 * vrf); 3703 // vl and vr are now in U8.24 format 3704 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3705 // send level comes from shared memory and so may be corrupt 3706 if (sendLevel > MAX_GAIN_INT) { 3707 ALOGV("Track send level out of range: %04X", sendLevel); 3708 sendLevel = MAX_GAIN_INT; 3709 } 3710 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3711 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3712 } 3713 3714 // Delegate volume control to effect in track effect chain if needed 3715 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3716 // Do not ramp volume if volume is controlled by effect 3717 param = AudioMixer::VOLUME; 3718 // Update remaining floating point volume levels 3719 vlf = (float)vl / (1 << 24); 3720 vrf = (float)vr / (1 << 24); 3721 track->mHasVolumeController = true; 3722 } else { 3723 // force no volume ramp when volume controller was just disabled or removed 3724 // from effect chain to avoid volume spike 3725 if (track->mHasVolumeController) { 3726 param = AudioMixer::VOLUME; 3727 } 3728 track->mHasVolumeController = false; 3729 } 3730 3731 // XXX: these things DON'T need to be done each time 3732 mAudioMixer->setBufferProvider(name, track); 3733 mAudioMixer->enable(name); 3734 3735 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3736 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3737 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3738 mAudioMixer->setParameter( 3739 name, 3740 AudioMixer::TRACK, 3741 AudioMixer::FORMAT, (void *)track->format()); 3742 mAudioMixer->setParameter( 3743 name, 3744 AudioMixer::TRACK, 3745 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3746 mAudioMixer->setParameter( 3747 name, 3748 AudioMixer::TRACK, 3749 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3750 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3751 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3752 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3753 if (reqSampleRate == 0) { 3754 reqSampleRate = mSampleRate; 3755 } else if (reqSampleRate > maxSampleRate) { 3756 reqSampleRate = maxSampleRate; 3757 } 3758 mAudioMixer->setParameter( 3759 name, 3760 AudioMixer::RESAMPLE, 3761 AudioMixer::SAMPLE_RATE, 3762 (void *)(uintptr_t)reqSampleRate); 3763 /* 3764 * Select the appropriate output buffer for the track. 3765 * 3766 * Tracks with effects go into their own effects chain buffer 3767 * and from there into either mEffectBuffer or mSinkBuffer. 3768 * 3769 * Other tracks can use mMixerBuffer for higher precision 3770 * channel accumulation. If this buffer is enabled 3771 * (mMixerBufferEnabled true), then selected tracks will accumulate 3772 * into it. 3773 * 3774 */ 3775 if (mMixerBufferEnabled 3776 && (track->mainBuffer() == mSinkBuffer 3777 || track->mainBuffer() == mMixerBuffer)) { 3778 mAudioMixer->setParameter( 3779 name, 3780 AudioMixer::TRACK, 3781 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3782 mAudioMixer->setParameter( 3783 name, 3784 AudioMixer::TRACK, 3785 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3786 // TODO: override track->mainBuffer()? 3787 mMixerBufferValid = true; 3788 } else { 3789 mAudioMixer->setParameter( 3790 name, 3791 AudioMixer::TRACK, 3792 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3793 mAudioMixer->setParameter( 3794 name, 3795 AudioMixer::TRACK, 3796 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3797 } 3798 mAudioMixer->setParameter( 3799 name, 3800 AudioMixer::TRACK, 3801 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3802 3803 // reset retry count 3804 track->mRetryCount = kMaxTrackRetries; 3805 3806 // If one track is ready, set the mixer ready if: 3807 // - the mixer was not ready during previous round OR 3808 // - no other track is not ready 3809 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3810 mixerStatus != MIXER_TRACKS_ENABLED) { 3811 mixerStatus = MIXER_TRACKS_READY; 3812 } 3813 } else { 3814 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3815 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3816 } 3817 // clear effect chain input buffer if an active track underruns to avoid sending 3818 // previous audio buffer again to effects 3819 chain = getEffectChain_l(track->sessionId()); 3820 if (chain != 0) { 3821 chain->clearInputBuffer(); 3822 } 3823 3824 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3825 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3826 track->isStopped() || track->isPaused()) { 3827 // We have consumed all the buffers of this track. 3828 // Remove it from the list of active tracks. 3829 // TODO: use actual buffer filling status instead of latency when available from 3830 // audio HAL 3831 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3832 size_t framesWritten = mBytesWritten / mFrameSize; 3833 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3834 if (track->isStopped()) { 3835 track->reset(); 3836 } 3837 tracksToRemove->add(track); 3838 } 3839 } else { 3840 // No buffers for this track. Give it a few chances to 3841 // fill a buffer, then remove it from active list. 3842 if (--(track->mRetryCount) <= 0) { 3843 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3844 tracksToRemove->add(track); 3845 // indicate to client process that the track was disabled because of underrun; 3846 // it will then automatically call start() when data is available 3847 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3848 // If one track is not ready, mark the mixer also not ready if: 3849 // - the mixer was ready during previous round OR 3850 // - no other track is ready 3851 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3852 mixerStatus != MIXER_TRACKS_READY) { 3853 mixerStatus = MIXER_TRACKS_ENABLED; 3854 } 3855 } 3856 mAudioMixer->disable(name); 3857 } 3858 3859 } // local variable scope to avoid goto warning 3860track_is_ready: ; 3861 3862 } 3863 3864 // Push the new FastMixer state if necessary 3865 bool pauseAudioWatchdog = false; 3866 if (didModify) { 3867 state->mFastTracksGen++; 3868 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3869 if (kUseFastMixer == FastMixer_Dynamic && 3870 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3871 state->mCommand = FastMixerState::COLD_IDLE; 3872 state->mColdFutexAddr = &mFastMixerFutex; 3873 state->mColdGen++; 3874 mFastMixerFutex = 0; 3875 if (kUseFastMixer == FastMixer_Dynamic) { 3876 mNormalSink = mOutputSink; 3877 } 3878 // If we go into cold idle, need to wait for acknowledgement 3879 // so that fast mixer stops doing I/O. 3880 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3881 pauseAudioWatchdog = true; 3882 } 3883 } 3884 if (sq != NULL) { 3885 sq->end(didModify); 3886 sq->push(block); 3887 } 3888#ifdef AUDIO_WATCHDOG 3889 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3890 mAudioWatchdog->pause(); 3891 } 3892#endif 3893 3894 // Now perform the deferred reset on fast tracks that have stopped 3895 while (resetMask != 0) { 3896 size_t i = __builtin_ctz(resetMask); 3897 ALOG_ASSERT(i < count); 3898 resetMask &= ~(1 << i); 3899 sp<Track> t = mActiveTracks[i].promote(); 3900 if (t == 0) { 3901 continue; 3902 } 3903 Track* track = t.get(); 3904 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3905 track->reset(); 3906 } 3907 3908 // remove all the tracks that need to be... 3909 removeTracks_l(*tracksToRemove); 3910 3911 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3912 mEffectBufferValid = true; 3913 } 3914 3915 if (mEffectBufferValid) { 3916 // as long as there are effects we should clear the effects buffer, to avoid 3917 // passing a non-clean buffer to the effect chain 3918 memset(mEffectBuffer, 0, mEffectBufferSize); 3919 } 3920 // sink or mix buffer must be cleared if all tracks are connected to an 3921 // effect chain as in this case the mixer will not write to the sink or mix buffer 3922 // and track effects will accumulate into it 3923 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3924 (mixedTracks == 0 && fastTracks > 0))) { 3925 // FIXME as a performance optimization, should remember previous zero status 3926 if (mMixerBufferValid) { 3927 memset(mMixerBuffer, 0, mMixerBufferSize); 3928 // TODO: In testing, mSinkBuffer below need not be cleared because 3929 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3930 // after mixing. 3931 // 3932 // To enforce this guarantee: 3933 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3934 // (mixedTracks == 0 && fastTracks > 0)) 3935 // must imply MIXER_TRACKS_READY. 3936 // Later, we may clear buffers regardless, and skip much of this logic. 3937 } 3938 // FIXME as a performance optimization, should remember previous zero status 3939 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3940 } 3941 3942 // if any fast tracks, then status is ready 3943 mMixerStatusIgnoringFastTracks = mixerStatus; 3944 if (fastTracks > 0) { 3945 mixerStatus = MIXER_TRACKS_READY; 3946 } 3947 return mixerStatus; 3948} 3949 3950// getTrackName_l() must be called with ThreadBase::mLock held 3951int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3952 audio_format_t format, int sessionId) 3953{ 3954 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3955} 3956 3957// deleteTrackName_l() must be called with ThreadBase::mLock held 3958void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3959{ 3960 ALOGV("remove track (%d) and delete from mixer", name); 3961 mAudioMixer->deleteTrackName(name); 3962} 3963 3964// checkForNewParameter_l() must be called with ThreadBase::mLock held 3965bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3966 status_t& status) 3967{ 3968 bool reconfig = false; 3969 3970 status = NO_ERROR; 3971 3972 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3973 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3974 if (mFastMixer != 0) { 3975 FastMixerStateQueue *sq = mFastMixer->sq(); 3976 FastMixerState *state = sq->begin(); 3977 if (!(state->mCommand & FastMixerState::IDLE)) { 3978 previousCommand = state->mCommand; 3979 state->mCommand = FastMixerState::HOT_IDLE; 3980 sq->end(); 3981 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3982 } else { 3983 sq->end(false /*didModify*/); 3984 } 3985 } 3986 3987 AudioParameter param = AudioParameter(keyValuePair); 3988 int value; 3989 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3990 reconfig = true; 3991 } 3992 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3993 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3994 status = BAD_VALUE; 3995 } else { 3996 // no need to save value, since it's constant 3997 reconfig = true; 3998 } 3999 } 4000 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4001 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4002 status = BAD_VALUE; 4003 } else { 4004 // no need to save value, since it's constant 4005 reconfig = true; 4006 } 4007 } 4008 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4009 // do not accept frame count changes if tracks are open as the track buffer 4010 // size depends on frame count and correct behavior would not be guaranteed 4011 // if frame count is changed after track creation 4012 if (!mTracks.isEmpty()) { 4013 status = INVALID_OPERATION; 4014 } else { 4015 reconfig = true; 4016 } 4017 } 4018 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4019#ifdef ADD_BATTERY_DATA 4020 // when changing the audio output device, call addBatteryData to notify 4021 // the change 4022 if (mOutDevice != value) { 4023 uint32_t params = 0; 4024 // check whether speaker is on 4025 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4026 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4027 } 4028 4029 audio_devices_t deviceWithoutSpeaker 4030 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4031 // check if any other device (except speaker) is on 4032 if (value & deviceWithoutSpeaker ) { 4033 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4034 } 4035 4036 if (params != 0) { 4037 addBatteryData(params); 4038 } 4039 } 4040#endif 4041 4042 // forward device change to effects that have requested to be 4043 // aware of attached audio device. 4044 if (value != AUDIO_DEVICE_NONE) { 4045 mOutDevice = value; 4046 for (size_t i = 0; i < mEffectChains.size(); i++) { 4047 mEffectChains[i]->setDevice_l(mOutDevice); 4048 } 4049 } 4050 } 4051 4052 if (status == NO_ERROR) { 4053 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4054 keyValuePair.string()); 4055 if (!mStandby && status == INVALID_OPERATION) { 4056 mOutput->stream->common.standby(&mOutput->stream->common); 4057 mStandby = true; 4058 mBytesWritten = 0; 4059 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4060 keyValuePair.string()); 4061 } 4062 if (status == NO_ERROR && reconfig) { 4063 readOutputParameters_l(); 4064 delete mAudioMixer; 4065 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4066 for (size_t i = 0; i < mTracks.size() ; i++) { 4067 int name = getTrackName_l(mTracks[i]->mChannelMask, 4068 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4069 if (name < 0) { 4070 break; 4071 } 4072 mTracks[i]->mName = name; 4073 } 4074 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4075 } 4076 } 4077 4078 if (!(previousCommand & FastMixerState::IDLE)) { 4079 ALOG_ASSERT(mFastMixer != 0); 4080 FastMixerStateQueue *sq = mFastMixer->sq(); 4081 FastMixerState *state = sq->begin(); 4082 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4083 state->mCommand = previousCommand; 4084 sq->end(); 4085 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4086 } 4087 4088 return reconfig; 4089} 4090 4091 4092void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4093{ 4094 const size_t SIZE = 256; 4095 char buffer[SIZE]; 4096 String8 result; 4097 4098 PlaybackThread::dumpInternals(fd, args); 4099 4100 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4101 4102 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4103 const FastMixerDumpState copy(mFastMixerDumpState); 4104 copy.dump(fd); 4105 4106#ifdef STATE_QUEUE_DUMP 4107 // Similar for state queue 4108 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4109 observerCopy.dump(fd); 4110 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4111 mutatorCopy.dump(fd); 4112#endif 4113 4114#ifdef TEE_SINK 4115 // Write the tee output to a .wav file 4116 dumpTee(fd, mTeeSource, mId); 4117#endif 4118 4119#ifdef AUDIO_WATCHDOG 4120 if (mAudioWatchdog != 0) { 4121 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4122 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4123 wdCopy.dump(fd); 4124 } 4125#endif 4126} 4127 4128uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4129{ 4130 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4131} 4132 4133uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4134{ 4135 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4136} 4137 4138void AudioFlinger::MixerThread::cacheParameters_l() 4139{ 4140 PlaybackThread::cacheParameters_l(); 4141 4142 // FIXME: Relaxed timing because of a certain device that can't meet latency 4143 // Should be reduced to 2x after the vendor fixes the driver issue 4144 // increase threshold again due to low power audio mode. The way this warning 4145 // threshold is calculated and its usefulness should be reconsidered anyway. 4146 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4147} 4148 4149// ---------------------------------------------------------------------------- 4150 4151AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4152 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4153 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4154 // mLeftVolFloat, mRightVolFloat 4155{ 4156} 4157 4158AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4159 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4160 ThreadBase::type_t type) 4161 : PlaybackThread(audioFlinger, output, id, device, type) 4162 // mLeftVolFloat, mRightVolFloat 4163{ 4164} 4165 4166AudioFlinger::DirectOutputThread::~DirectOutputThread() 4167{ 4168} 4169 4170void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4171{ 4172 audio_track_cblk_t* cblk = track->cblk(); 4173 float left, right; 4174 4175 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4176 left = right = 0; 4177 } else { 4178 float typeVolume = mStreamTypes[track->streamType()].volume; 4179 float v = mMasterVolume * typeVolume; 4180 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4181 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4182 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4183 if (left > GAIN_FLOAT_UNITY) { 4184 left = GAIN_FLOAT_UNITY; 4185 } 4186 left *= v; 4187 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4188 if (right > GAIN_FLOAT_UNITY) { 4189 right = GAIN_FLOAT_UNITY; 4190 } 4191 right *= v; 4192 } 4193 4194 if (lastTrack) { 4195 if (left != mLeftVolFloat || right != mRightVolFloat) { 4196 mLeftVolFloat = left; 4197 mRightVolFloat = right; 4198 4199 // Convert volumes from float to 8.24 4200 uint32_t vl = (uint32_t)(left * (1 << 24)); 4201 uint32_t vr = (uint32_t)(right * (1 << 24)); 4202 4203 // Delegate volume control to effect in track effect chain if needed 4204 // only one effect chain can be present on DirectOutputThread, so if 4205 // there is one, the track is connected to it 4206 if (!mEffectChains.isEmpty()) { 4207 mEffectChains[0]->setVolume_l(&vl, &vr); 4208 left = (float)vl / (1 << 24); 4209 right = (float)vr / (1 << 24); 4210 } 4211 if (mOutput->stream->set_volume) { 4212 mOutput->stream->set_volume(mOutput->stream, left, right); 4213 } 4214 } 4215 } 4216} 4217 4218 4219AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4220 Vector< sp<Track> > *tracksToRemove 4221) 4222{ 4223 size_t count = mActiveTracks.size(); 4224 mixer_state mixerStatus = MIXER_IDLE; 4225 bool doHwPause = false; 4226 bool doHwResume = false; 4227 bool flushPending = false; 4228 4229 // find out which tracks need to be processed 4230 for (size_t i = 0; i < count; i++) { 4231 sp<Track> t = mActiveTracks[i].promote(); 4232 // The track died recently 4233 if (t == 0) { 4234 continue; 4235 } 4236 4237 Track* const track = t.get(); 4238 audio_track_cblk_t* cblk = track->cblk(); 4239 // Only consider last track started for volume and mixer state control. 4240 // In theory an older track could underrun and restart after the new one starts 4241 // but as we only care about the transition phase between two tracks on a 4242 // direct output, it is not a problem to ignore the underrun case. 4243 sp<Track> l = mLatestActiveTrack.promote(); 4244 bool last = l.get() == track; 4245 4246 if (mHwSupportsPause && track->isPausing()) { 4247 track->setPaused(); 4248 if (last && !mHwPaused) { 4249 doHwPause = true; 4250 mHwPaused = true; 4251 } 4252 tracksToRemove->add(track); 4253 } else if (track->isFlushPending()) { 4254 track->flushAck(); 4255 if (last) { 4256 flushPending = true; 4257 } 4258 } else if (mHwSupportsPause && track->isResumePending()){ 4259 track->resumeAck(); 4260 if (last) { 4261 if (mHwPaused) { 4262 doHwResume = true; 4263 mHwPaused = false; 4264 } 4265 } 4266 } 4267 4268 // The first time a track is added we wait 4269 // for all its buffers to be filled before processing it. 4270 // Allow draining the buffer in case the client 4271 // app does not call stop() and relies on underrun to stop: 4272 // hence the test on (track->mRetryCount > 1). 4273 // If retryCount<=1 then track is about to underrun and be removed. 4274 uint32_t minFrames; 4275 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4276 && (track->mRetryCount > 1)) { 4277 minFrames = mNormalFrameCount; 4278 } else { 4279 minFrames = 1; 4280 } 4281 4282 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4283 !track->isStopping_2() && !track->isStopped()) 4284 { 4285 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4286 4287 if (track->mFillingUpStatus == Track::FS_FILLED) { 4288 track->mFillingUpStatus = Track::FS_ACTIVE; 4289 // make sure processVolume_l() will apply new volume even if 0 4290 mLeftVolFloat = mRightVolFloat = -1.0; 4291 if (!mHwSupportsPause) { 4292 track->resumeAck(); 4293 } 4294 } 4295 4296 // compute volume for this track 4297 processVolume_l(track, last); 4298 if (last) { 4299 // reset retry count 4300 track->mRetryCount = kMaxTrackRetriesDirect; 4301 mActiveTrack = t; 4302 mixerStatus = MIXER_TRACKS_READY; 4303 if (usesHwAvSync() && mHwPaused) { 4304 doHwResume = true; 4305 mHwPaused = false; 4306 } 4307 } 4308 } else { 4309 // clear effect chain input buffer if the last active track started underruns 4310 // to avoid sending previous audio buffer again to effects 4311 if (!mEffectChains.isEmpty() && last) { 4312 mEffectChains[0]->clearInputBuffer(); 4313 } 4314 if (track->isStopping_1()) { 4315 track->mState = TrackBase::STOPPING_2; 4316 } 4317 if ((track->sharedBuffer() != 0) || track->isStopped() || 4318 track->isStopping_2() || track->isPaused()) { 4319 // We have consumed all the buffers of this track. 4320 // Remove it from the list of active tracks. 4321 size_t audioHALFrames; 4322 if (audio_is_linear_pcm(mFormat)) { 4323 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4324 } else { 4325 audioHALFrames = 0; 4326 } 4327 4328 size_t framesWritten = mBytesWritten / mFrameSize; 4329 if (mStandby || !last || 4330 track->presentationComplete(framesWritten, audioHALFrames)) { 4331 if (track->isStopping_2()) { 4332 track->mState = TrackBase::STOPPED; 4333 } 4334 if (track->isStopped()) { 4335 track->reset(); 4336 } 4337 tracksToRemove->add(track); 4338 } 4339 } else { 4340 // No buffers for this track. Give it a few chances to 4341 // fill a buffer, then remove it from active list. 4342 // Only consider last track started for mixer state control 4343 if (--(track->mRetryCount) <= 0) { 4344 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4345 tracksToRemove->add(track); 4346 // indicate to client process that the track was disabled because of underrun; 4347 // it will then automatically call start() when data is available 4348 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4349 } else if (last) { 4350 mixerStatus = MIXER_TRACKS_ENABLED; 4351 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4352 doHwPause = true; 4353 mHwPaused = true; 4354 } 4355 } 4356 } 4357 } 4358 } 4359 4360 // if an active track did not command a flush, check for pending flush on stopped tracks 4361 if (!flushPending) { 4362 for (size_t i = 0; i < mTracks.size(); i++) { 4363 if (mTracks[i]->isFlushPending()) { 4364 mTracks[i]->flushAck(); 4365 flushPending = true; 4366 } 4367 } 4368 } 4369 4370 // make sure the pause/flush/resume sequence is executed in the right order. 4371 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4372 // before flush and then resume HW. This can happen in case of pause/flush/resume 4373 // if resume is received before pause is executed. 4374 if (mHwSupportsPause && !mStandby && 4375 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4376 mOutput->stream->pause(mOutput->stream); 4377 } 4378 if (flushPending) { 4379 flushHw_l(); 4380 } 4381 if (mHwSupportsPause && !mStandby && doHwResume) { 4382 mOutput->stream->resume(mOutput->stream); 4383 } 4384 // remove all the tracks that need to be... 4385 removeTracks_l(*tracksToRemove); 4386 4387 return mixerStatus; 4388} 4389 4390void AudioFlinger::DirectOutputThread::threadLoop_mix() 4391{ 4392 size_t frameCount = mFrameCount; 4393 int8_t *curBuf = (int8_t *)mSinkBuffer; 4394 // output audio to hardware 4395 while (frameCount) { 4396 AudioBufferProvider::Buffer buffer; 4397 buffer.frameCount = frameCount; 4398 mActiveTrack->getNextBuffer(&buffer); 4399 if (buffer.raw == NULL) { 4400 memset(curBuf, 0, frameCount * mFrameSize); 4401 break; 4402 } 4403 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4404 frameCount -= buffer.frameCount; 4405 curBuf += buffer.frameCount * mFrameSize; 4406 mActiveTrack->releaseBuffer(&buffer); 4407 } 4408 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4409 sleepTime = 0; 4410 standbyTime = systemTime() + standbyDelay; 4411 mActiveTrack.clear(); 4412} 4413 4414void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4415{ 4416 // do not write to HAL when paused 4417 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4418 sleepTime = idleSleepTime; 4419 return; 4420 } 4421 if (sleepTime == 0) { 4422 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4423 sleepTime = activeSleepTime; 4424 } else { 4425 sleepTime = idleSleepTime; 4426 } 4427 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4428 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4429 sleepTime = 0; 4430 } 4431} 4432 4433void AudioFlinger::DirectOutputThread::threadLoop_exit() 4434{ 4435 { 4436 Mutex::Autolock _l(mLock); 4437 bool flushPending = false; 4438 for (size_t i = 0; i < mTracks.size(); i++) { 4439 if (mTracks[i]->isFlushPending()) { 4440 mTracks[i]->flushAck(); 4441 flushPending = true; 4442 } 4443 } 4444 if (flushPending) { 4445 flushHw_l(); 4446 } 4447 } 4448 PlaybackThread::threadLoop_exit(); 4449} 4450 4451// must be called with thread mutex locked 4452bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4453{ 4454 bool trackPaused = false; 4455 4456 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4457 // after a timeout and we will enter standby then. 4458 if (mTracks.size() > 0) { 4459 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4460 } 4461 4462 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused)); 4463} 4464 4465// getTrackName_l() must be called with ThreadBase::mLock held 4466int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4467 audio_format_t format __unused, int sessionId __unused) 4468{ 4469 return 0; 4470} 4471 4472// deleteTrackName_l() must be called with ThreadBase::mLock held 4473void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4474{ 4475} 4476 4477// checkForNewParameter_l() must be called with ThreadBase::mLock held 4478bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4479 status_t& status) 4480{ 4481 bool reconfig = false; 4482 4483 status = NO_ERROR; 4484 4485 AudioParameter param = AudioParameter(keyValuePair); 4486 int value; 4487 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4488 // forward device change to effects that have requested to be 4489 // aware of attached audio device. 4490 if (value != AUDIO_DEVICE_NONE) { 4491 mOutDevice = value; 4492 for (size_t i = 0; i < mEffectChains.size(); i++) { 4493 mEffectChains[i]->setDevice_l(mOutDevice); 4494 } 4495 } 4496 } 4497 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4498 // do not accept frame count changes if tracks are open as the track buffer 4499 // size depends on frame count and correct behavior would not be garantied 4500 // if frame count is changed after track creation 4501 if (!mTracks.isEmpty()) { 4502 status = INVALID_OPERATION; 4503 } else { 4504 reconfig = true; 4505 } 4506 } 4507 if (status == NO_ERROR) { 4508 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4509 keyValuePair.string()); 4510 if (!mStandby && status == INVALID_OPERATION) { 4511 mOutput->stream->common.standby(&mOutput->stream->common); 4512 mStandby = true; 4513 mBytesWritten = 0; 4514 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4515 keyValuePair.string()); 4516 } 4517 if (status == NO_ERROR && reconfig) { 4518 readOutputParameters_l(); 4519 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4520 } 4521 } 4522 4523 return reconfig; 4524} 4525 4526uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4527{ 4528 uint32_t time; 4529 if (audio_is_linear_pcm(mFormat)) { 4530 time = PlaybackThread::activeSleepTimeUs(); 4531 } else { 4532 time = 10000; 4533 } 4534 return time; 4535} 4536 4537uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4538{ 4539 uint32_t time; 4540 if (audio_is_linear_pcm(mFormat)) { 4541 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4542 } else { 4543 time = 10000; 4544 } 4545 return time; 4546} 4547 4548uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4549{ 4550 uint32_t time; 4551 if (audio_is_linear_pcm(mFormat)) { 4552 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4553 } else { 4554 time = 10000; 4555 } 4556 return time; 4557} 4558 4559void AudioFlinger::DirectOutputThread::cacheParameters_l() 4560{ 4561 PlaybackThread::cacheParameters_l(); 4562 4563 // use shorter standby delay as on normal output to release 4564 // hardware resources as soon as possible 4565 if (audio_is_linear_pcm(mFormat)) { 4566 standbyDelay = microseconds(activeSleepTime*2); 4567 } else { 4568 standbyDelay = kOffloadStandbyDelayNs; 4569 } 4570} 4571 4572void AudioFlinger::DirectOutputThread::flushHw_l() 4573{ 4574 if (mOutput->stream->flush != NULL) { 4575 mOutput->stream->flush(mOutput->stream); 4576 } 4577 mHwPaused = false; 4578} 4579 4580// ---------------------------------------------------------------------------- 4581 4582AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4583 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4584 : Thread(false /*canCallJava*/), 4585 mPlaybackThread(playbackThread), 4586 mWriteAckSequence(0), 4587 mDrainSequence(0) 4588{ 4589} 4590 4591AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4592{ 4593} 4594 4595void AudioFlinger::AsyncCallbackThread::onFirstRef() 4596{ 4597 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4598} 4599 4600bool AudioFlinger::AsyncCallbackThread::threadLoop() 4601{ 4602 while (!exitPending()) { 4603 uint32_t writeAckSequence; 4604 uint32_t drainSequence; 4605 4606 { 4607 Mutex::Autolock _l(mLock); 4608 while (!((mWriteAckSequence & 1) || 4609 (mDrainSequence & 1) || 4610 exitPending())) { 4611 mWaitWorkCV.wait(mLock); 4612 } 4613 4614 if (exitPending()) { 4615 break; 4616 } 4617 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4618 mWriteAckSequence, mDrainSequence); 4619 writeAckSequence = mWriteAckSequence; 4620 mWriteAckSequence &= ~1; 4621 drainSequence = mDrainSequence; 4622 mDrainSequence &= ~1; 4623 } 4624 { 4625 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4626 if (playbackThread != 0) { 4627 if (writeAckSequence & 1) { 4628 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4629 } 4630 if (drainSequence & 1) { 4631 playbackThread->resetDraining(drainSequence >> 1); 4632 } 4633 } 4634 } 4635 } 4636 return false; 4637} 4638 4639void AudioFlinger::AsyncCallbackThread::exit() 4640{ 4641 ALOGV("AsyncCallbackThread::exit"); 4642 Mutex::Autolock _l(mLock); 4643 requestExit(); 4644 mWaitWorkCV.broadcast(); 4645} 4646 4647void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4648{ 4649 Mutex::Autolock _l(mLock); 4650 // bit 0 is cleared 4651 mWriteAckSequence = sequence << 1; 4652} 4653 4654void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4655{ 4656 Mutex::Autolock _l(mLock); 4657 // ignore unexpected callbacks 4658 if (mWriteAckSequence & 2) { 4659 mWriteAckSequence |= 1; 4660 mWaitWorkCV.signal(); 4661 } 4662} 4663 4664void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4665{ 4666 Mutex::Autolock _l(mLock); 4667 // bit 0 is cleared 4668 mDrainSequence = sequence << 1; 4669} 4670 4671void AudioFlinger::AsyncCallbackThread::resetDraining() 4672{ 4673 Mutex::Autolock _l(mLock); 4674 // ignore unexpected callbacks 4675 if (mDrainSequence & 2) { 4676 mDrainSequence |= 1; 4677 mWaitWorkCV.signal(); 4678 } 4679} 4680 4681 4682// ---------------------------------------------------------------------------- 4683AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4684 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4685 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4686 mPausedBytesRemaining(0) 4687{ 4688 //FIXME: mStandby should be set to true by ThreadBase constructor 4689 mStandby = true; 4690} 4691 4692void AudioFlinger::OffloadThread::threadLoop_exit() 4693{ 4694 if (mFlushPending || mHwPaused) { 4695 // If a flush is pending or track was paused, just discard buffered data 4696 flushHw_l(); 4697 } else { 4698 mMixerStatus = MIXER_DRAIN_ALL; 4699 threadLoop_drain(); 4700 } 4701 if (mUseAsyncWrite) { 4702 ALOG_ASSERT(mCallbackThread != 0); 4703 mCallbackThread->exit(); 4704 } 4705 PlaybackThread::threadLoop_exit(); 4706} 4707 4708AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4709 Vector< sp<Track> > *tracksToRemove 4710) 4711{ 4712 size_t count = mActiveTracks.size(); 4713 4714 mixer_state mixerStatus = MIXER_IDLE; 4715 bool doHwPause = false; 4716 bool doHwResume = false; 4717 4718 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4719 4720 // find out which tracks need to be processed 4721 for (size_t i = 0; i < count; i++) { 4722 sp<Track> t = mActiveTracks[i].promote(); 4723 // The track died recently 4724 if (t == 0) { 4725 continue; 4726 } 4727 Track* const track = t.get(); 4728 audio_track_cblk_t* cblk = track->cblk(); 4729 // Only consider last track started for volume and mixer state control. 4730 // In theory an older track could underrun and restart after the new one starts 4731 // but as we only care about the transition phase between two tracks on a 4732 // direct output, it is not a problem to ignore the underrun case. 4733 sp<Track> l = mLatestActiveTrack.promote(); 4734 bool last = l.get() == track; 4735 4736 if (track->isInvalid()) { 4737 ALOGW("An invalidated track shouldn't be in active list"); 4738 tracksToRemove->add(track); 4739 continue; 4740 } 4741 4742 if (track->mState == TrackBase::IDLE) { 4743 ALOGW("An idle track shouldn't be in active list"); 4744 continue; 4745 } 4746 4747 if (track->isPausing()) { 4748 track->setPaused(); 4749 if (last) { 4750 if (!mHwPaused) { 4751 doHwPause = true; 4752 mHwPaused = true; 4753 } 4754 // If we were part way through writing the mixbuffer to 4755 // the HAL we must save this until we resume 4756 // BUG - this will be wrong if a different track is made active, 4757 // in that case we want to discard the pending data in the 4758 // mixbuffer and tell the client to present it again when the 4759 // track is resumed 4760 mPausedWriteLength = mCurrentWriteLength; 4761 mPausedBytesRemaining = mBytesRemaining; 4762 mBytesRemaining = 0; // stop writing 4763 } 4764 tracksToRemove->add(track); 4765 } else if (track->isFlushPending()) { 4766 track->flushAck(); 4767 if (last) { 4768 mFlushPending = true; 4769 } 4770 } else if (track->isResumePending()){ 4771 track->resumeAck(); 4772 if (last) { 4773 if (mPausedBytesRemaining) { 4774 // Need to continue write that was interrupted 4775 mCurrentWriteLength = mPausedWriteLength; 4776 mBytesRemaining = mPausedBytesRemaining; 4777 mPausedBytesRemaining = 0; 4778 } 4779 if (mHwPaused) { 4780 doHwResume = true; 4781 mHwPaused = false; 4782 // threadLoop_mix() will handle the case that we need to 4783 // resume an interrupted write 4784 } 4785 // enable write to audio HAL 4786 sleepTime = 0; 4787 4788 // Do not handle new data in this iteration even if track->framesReady() 4789 mixerStatus = MIXER_TRACKS_ENABLED; 4790 } 4791 } else if (track->framesReady() && track->isReady() && 4792 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4793 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4794 if (track->mFillingUpStatus == Track::FS_FILLED) { 4795 track->mFillingUpStatus = Track::FS_ACTIVE; 4796 // make sure processVolume_l() will apply new volume even if 0 4797 mLeftVolFloat = mRightVolFloat = -1.0; 4798 } 4799 4800 if (last) { 4801 sp<Track> previousTrack = mPreviousTrack.promote(); 4802 if (previousTrack != 0) { 4803 if (track != previousTrack.get()) { 4804 // Flush any data still being written from last track 4805 mBytesRemaining = 0; 4806 if (mPausedBytesRemaining) { 4807 // Last track was paused so we also need to flush saved 4808 // mixbuffer state and invalidate track so that it will 4809 // re-submit that unwritten data when it is next resumed 4810 mPausedBytesRemaining = 0; 4811 // Invalidate is a bit drastic - would be more efficient 4812 // to have a flag to tell client that some of the 4813 // previously written data was lost 4814 previousTrack->invalidate(); 4815 } 4816 // flush data already sent to the DSP if changing audio session as audio 4817 // comes from a different source. Also invalidate previous track to force a 4818 // seek when resuming. 4819 if (previousTrack->sessionId() != track->sessionId()) { 4820 previousTrack->invalidate(); 4821 } 4822 } 4823 } 4824 mPreviousTrack = track; 4825 // reset retry count 4826 track->mRetryCount = kMaxTrackRetriesOffload; 4827 mActiveTrack = t; 4828 mixerStatus = MIXER_TRACKS_READY; 4829 } 4830 } else { 4831 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4832 if (track->isStopping_1()) { 4833 // Hardware buffer can hold a large amount of audio so we must 4834 // wait for all current track's data to drain before we say 4835 // that the track is stopped. 4836 if (mBytesRemaining == 0) { 4837 // Only start draining when all data in mixbuffer 4838 // has been written 4839 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4840 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4841 // do not drain if no data was ever sent to HAL (mStandby == true) 4842 if (last && !mStandby) { 4843 // do not modify drain sequence if we are already draining. This happens 4844 // when resuming from pause after drain. 4845 if ((mDrainSequence & 1) == 0) { 4846 sleepTime = 0; 4847 standbyTime = systemTime() + standbyDelay; 4848 mixerStatus = MIXER_DRAIN_TRACK; 4849 mDrainSequence += 2; 4850 } 4851 if (mHwPaused) { 4852 // It is possible to move from PAUSED to STOPPING_1 without 4853 // a resume so we must ensure hardware is running 4854 doHwResume = true; 4855 mHwPaused = false; 4856 } 4857 } 4858 } 4859 } else if (track->isStopping_2()) { 4860 // Drain has completed or we are in standby, signal presentation complete 4861 if (!(mDrainSequence & 1) || !last || mStandby) { 4862 track->mState = TrackBase::STOPPED; 4863 size_t audioHALFrames = 4864 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4865 size_t framesWritten = 4866 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4867 track->presentationComplete(framesWritten, audioHALFrames); 4868 track->reset(); 4869 tracksToRemove->add(track); 4870 } 4871 } else { 4872 // No buffers for this track. Give it a few chances to 4873 // fill a buffer, then remove it from active list. 4874 if (--(track->mRetryCount) <= 0) { 4875 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4876 track->name()); 4877 tracksToRemove->add(track); 4878 // indicate to client process that the track was disabled because of underrun; 4879 // it will then automatically call start() when data is available 4880 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4881 } else if (last){ 4882 mixerStatus = MIXER_TRACKS_ENABLED; 4883 } 4884 } 4885 } 4886 // compute volume for this track 4887 processVolume_l(track, last); 4888 } 4889 4890 // make sure the pause/flush/resume sequence is executed in the right order. 4891 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4892 // before flush and then resume HW. This can happen in case of pause/flush/resume 4893 // if resume is received before pause is executed. 4894 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4895 mOutput->stream->pause(mOutput->stream); 4896 } 4897 if (mFlushPending) { 4898 flushHw_l(); 4899 mFlushPending = false; 4900 } 4901 if (!mStandby && doHwResume) { 4902 mOutput->stream->resume(mOutput->stream); 4903 } 4904 4905 // remove all the tracks that need to be... 4906 removeTracks_l(*tracksToRemove); 4907 4908 return mixerStatus; 4909} 4910 4911// must be called with thread mutex locked 4912bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4913{ 4914 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4915 mWriteAckSequence, mDrainSequence); 4916 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4917 return true; 4918 } 4919 return false; 4920} 4921 4922bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4923{ 4924 Mutex::Autolock _l(mLock); 4925 return waitingAsyncCallback_l(); 4926} 4927 4928void AudioFlinger::OffloadThread::flushHw_l() 4929{ 4930 DirectOutputThread::flushHw_l(); 4931 // Flush anything still waiting in the mixbuffer 4932 mCurrentWriteLength = 0; 4933 mBytesRemaining = 0; 4934 mPausedWriteLength = 0; 4935 mPausedBytesRemaining = 0; 4936 4937 if (mUseAsyncWrite) { 4938 // discard any pending drain or write ack by incrementing sequence 4939 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4940 mDrainSequence = (mDrainSequence + 2) & ~1; 4941 ALOG_ASSERT(mCallbackThread != 0); 4942 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4943 mCallbackThread->setDraining(mDrainSequence); 4944 } 4945} 4946 4947void AudioFlinger::OffloadThread::onAddNewTrack_l() 4948{ 4949 sp<Track> previousTrack = mPreviousTrack.promote(); 4950 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4951 4952 if (previousTrack != 0 && latestTrack != 0 && 4953 (previousTrack->sessionId() != latestTrack->sessionId())) { 4954 mFlushPending = true; 4955 } 4956 PlaybackThread::onAddNewTrack_l(); 4957} 4958 4959// ---------------------------------------------------------------------------- 4960 4961AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4962 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4963 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4964 DUPLICATING), 4965 mWaitTimeMs(UINT_MAX) 4966{ 4967 addOutputTrack(mainThread); 4968} 4969 4970AudioFlinger::DuplicatingThread::~DuplicatingThread() 4971{ 4972 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4973 mOutputTracks[i]->destroy(); 4974 } 4975} 4976 4977void AudioFlinger::DuplicatingThread::threadLoop_mix() 4978{ 4979 // mix buffers... 4980 if (outputsReady(outputTracks)) { 4981 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4982 } else { 4983 if (mMixerBufferValid) { 4984 memset(mMixerBuffer, 0, mMixerBufferSize); 4985 } else { 4986 memset(mSinkBuffer, 0, mSinkBufferSize); 4987 } 4988 } 4989 sleepTime = 0; 4990 writeFrames = mNormalFrameCount; 4991 mCurrentWriteLength = mSinkBufferSize; 4992 standbyTime = systemTime() + standbyDelay; 4993} 4994 4995void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4996{ 4997 if (sleepTime == 0) { 4998 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4999 sleepTime = activeSleepTime; 5000 } else { 5001 sleepTime = idleSleepTime; 5002 } 5003 } else if (mBytesWritten != 0) { 5004 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5005 writeFrames = mNormalFrameCount; 5006 memset(mSinkBuffer, 0, mSinkBufferSize); 5007 } else { 5008 // flush remaining overflow buffers in output tracks 5009 writeFrames = 0; 5010 } 5011 sleepTime = 0; 5012 } 5013} 5014 5015ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5016{ 5017 for (size_t i = 0; i < outputTracks.size(); i++) { 5018 outputTracks[i]->write(mSinkBuffer, writeFrames); 5019 } 5020 mStandby = false; 5021 return (ssize_t)mSinkBufferSize; 5022} 5023 5024void AudioFlinger::DuplicatingThread::threadLoop_standby() 5025{ 5026 // DuplicatingThread implements standby by stopping all tracks 5027 for (size_t i = 0; i < outputTracks.size(); i++) { 5028 outputTracks[i]->stop(); 5029 } 5030} 5031 5032void AudioFlinger::DuplicatingThread::saveOutputTracks() 5033{ 5034 outputTracks = mOutputTracks; 5035} 5036 5037void AudioFlinger::DuplicatingThread::clearOutputTracks() 5038{ 5039 outputTracks.clear(); 5040} 5041 5042void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5043{ 5044 Mutex::Autolock _l(mLock); 5045 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5046 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5047 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5048 const size_t frameCount = 5049 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5050 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5051 // from different OutputTracks and their associated MixerThreads (e.g. one may 5052 // nearly empty and the other may be dropping data). 5053 5054 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5055 this, 5056 mSampleRate, 5057 mFormat, 5058 mChannelMask, 5059 frameCount, 5060 IPCThreadState::self()->getCallingUid()); 5061 if (outputTrack->cblk() != NULL) { 5062 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5063 mOutputTracks.add(outputTrack); 5064 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5065 updateWaitTime_l(); 5066 } 5067} 5068 5069void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5070{ 5071 Mutex::Autolock _l(mLock); 5072 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5073 if (mOutputTracks[i]->thread() == thread) { 5074 mOutputTracks[i]->destroy(); 5075 mOutputTracks.removeAt(i); 5076 updateWaitTime_l(); 5077 return; 5078 } 5079 } 5080 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5081} 5082 5083// caller must hold mLock 5084void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5085{ 5086 mWaitTimeMs = UINT_MAX; 5087 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5088 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5089 if (strong != 0) { 5090 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5091 if (waitTimeMs < mWaitTimeMs) { 5092 mWaitTimeMs = waitTimeMs; 5093 } 5094 } 5095 } 5096} 5097 5098 5099bool AudioFlinger::DuplicatingThread::outputsReady( 5100 const SortedVector< sp<OutputTrack> > &outputTracks) 5101{ 5102 for (size_t i = 0; i < outputTracks.size(); i++) { 5103 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5104 if (thread == 0) { 5105 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5106 outputTracks[i].get()); 5107 return false; 5108 } 5109 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5110 // see note at standby() declaration 5111 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5112 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5113 thread.get()); 5114 return false; 5115 } 5116 } 5117 return true; 5118} 5119 5120uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5121{ 5122 return (mWaitTimeMs * 1000) / 2; 5123} 5124 5125void AudioFlinger::DuplicatingThread::cacheParameters_l() 5126{ 5127 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5128 updateWaitTime_l(); 5129 5130 MixerThread::cacheParameters_l(); 5131} 5132 5133// ---------------------------------------------------------------------------- 5134// Record 5135// ---------------------------------------------------------------------------- 5136 5137AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5138 AudioStreamIn *input, 5139 audio_io_handle_t id, 5140 audio_devices_t outDevice, 5141 audio_devices_t inDevice 5142#ifdef TEE_SINK 5143 , const sp<NBAIO_Sink>& teeSink 5144#endif 5145 ) : 5146 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5147 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5148 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5149 mRsmpInRear(0) 5150#ifdef TEE_SINK 5151 , mTeeSink(teeSink) 5152#endif 5153 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5154 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5155 // mFastCapture below 5156 , mFastCaptureFutex(0) 5157 // mInputSource 5158 // mPipeSink 5159 // mPipeSource 5160 , mPipeFramesP2(0) 5161 // mPipeMemory 5162 // mFastCaptureNBLogWriter 5163 , mFastTrackAvail(false) 5164{ 5165 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5166 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5167 5168 readInputParameters_l(); 5169 5170 // create an NBAIO source for the HAL input stream, and negotiate 5171 mInputSource = new AudioStreamInSource(input->stream); 5172 size_t numCounterOffers = 0; 5173 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5174 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5175 ALOG_ASSERT(index == 0); 5176 5177 // initialize fast capture depending on configuration 5178 bool initFastCapture; 5179 switch (kUseFastCapture) { 5180 case FastCapture_Never: 5181 initFastCapture = false; 5182 break; 5183 case FastCapture_Always: 5184 initFastCapture = true; 5185 break; 5186 case FastCapture_Static: 5187 uint32_t primaryOutputSampleRate; 5188 { 5189 AutoMutex _l(audioFlinger->mHardwareLock); 5190 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5191 } 5192 initFastCapture = 5193 // either capture sample rate is same as (a reasonable) primary output sample rate 5194 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5195 (mSampleRate == primaryOutputSampleRate)) || 5196 // or primary output sample rate is unknown, and capture sample rate is reasonable 5197 ((primaryOutputSampleRate == 0) && 5198 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5199 // and the buffer size is < 12 ms 5200 (mFrameCount * 1000) / mSampleRate < 12; 5201 break; 5202 // case FastCapture_Dynamic: 5203 } 5204 5205 if (initFastCapture) { 5206 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 5207 NBAIO_Format format = mInputSource->format(); 5208 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5209 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5210 void *pipeBuffer; 5211 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5212 sp<IMemory> pipeMemory; 5213 if ((roHeap == 0) || 5214 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5215 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5216 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5217 goto failed; 5218 } 5219 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5220 memset(pipeBuffer, 0, pipeSize); 5221 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5222 const NBAIO_Format offers[1] = {format}; 5223 size_t numCounterOffers = 0; 5224 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5225 ALOG_ASSERT(index == 0); 5226 mPipeSink = pipe; 5227 PipeReader *pipeReader = new PipeReader(*pipe); 5228 numCounterOffers = 0; 5229 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5230 ALOG_ASSERT(index == 0); 5231 mPipeSource = pipeReader; 5232 mPipeFramesP2 = pipeFramesP2; 5233 mPipeMemory = pipeMemory; 5234 5235 // create fast capture 5236 mFastCapture = new FastCapture(); 5237 FastCaptureStateQueue *sq = mFastCapture->sq(); 5238#ifdef STATE_QUEUE_DUMP 5239 // FIXME 5240#endif 5241 FastCaptureState *state = sq->begin(); 5242 state->mCblk = NULL; 5243 state->mInputSource = mInputSource.get(); 5244 state->mInputSourceGen++; 5245 state->mPipeSink = pipe; 5246 state->mPipeSinkGen++; 5247 state->mFrameCount = mFrameCount; 5248 state->mCommand = FastCaptureState::COLD_IDLE; 5249 // already done in constructor initialization list 5250 //mFastCaptureFutex = 0; 5251 state->mColdFutexAddr = &mFastCaptureFutex; 5252 state->mColdGen++; 5253 state->mDumpState = &mFastCaptureDumpState; 5254#ifdef TEE_SINK 5255 // FIXME 5256#endif 5257 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5258 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5259 sq->end(); 5260 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5261 5262 // start the fast capture 5263 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5264 pid_t tid = mFastCapture->getTid(); 5265 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5266 if (err != 0) { 5267 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5268 kPriorityFastCapture, getpid_cached, tid, err); 5269 } 5270 5271#ifdef AUDIO_WATCHDOG 5272 // FIXME 5273#endif 5274 5275 mFastTrackAvail = true; 5276 } 5277failed: ; 5278 5279 // FIXME mNormalSource 5280} 5281 5282 5283AudioFlinger::RecordThread::~RecordThread() 5284{ 5285 if (mFastCapture != 0) { 5286 FastCaptureStateQueue *sq = mFastCapture->sq(); 5287 FastCaptureState *state = sq->begin(); 5288 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5289 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5290 if (old == -1) { 5291 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5292 } 5293 } 5294 state->mCommand = FastCaptureState::EXIT; 5295 sq->end(); 5296 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5297 mFastCapture->join(); 5298 mFastCapture.clear(); 5299 } 5300 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5301 mAudioFlinger->unregisterWriter(mNBLogWriter); 5302 delete[] mRsmpInBuffer; 5303} 5304 5305void AudioFlinger::RecordThread::onFirstRef() 5306{ 5307 run(mThreadName, PRIORITY_URGENT_AUDIO); 5308} 5309 5310bool AudioFlinger::RecordThread::threadLoop() 5311{ 5312 nsecs_t lastWarning = 0; 5313 5314 inputStandBy(); 5315 5316reacquire_wakelock: 5317 sp<RecordTrack> activeTrack; 5318 int activeTracksGen; 5319 { 5320 Mutex::Autolock _l(mLock); 5321 size_t size = mActiveTracks.size(); 5322 activeTracksGen = mActiveTracksGen; 5323 if (size > 0) { 5324 // FIXME an arbitrary choice 5325 activeTrack = mActiveTracks[0]; 5326 acquireWakeLock_l(activeTrack->uid()); 5327 if (size > 1) { 5328 SortedVector<int> tmp; 5329 for (size_t i = 0; i < size; i++) { 5330 tmp.add(mActiveTracks[i]->uid()); 5331 } 5332 updateWakeLockUids_l(tmp); 5333 } 5334 } else { 5335 acquireWakeLock_l(-1); 5336 } 5337 } 5338 5339 // used to request a deferred sleep, to be executed later while mutex is unlocked 5340 uint32_t sleepUs = 0; 5341 5342 // loop while there is work to do 5343 for (;;) { 5344 Vector< sp<EffectChain> > effectChains; 5345 5346 // sleep with mutex unlocked 5347 if (sleepUs > 0) { 5348 ATRACE_BEGIN("sleep"); 5349 usleep(sleepUs); 5350 ATRACE_END(); 5351 sleepUs = 0; 5352 } 5353 5354 // activeTracks accumulates a copy of a subset of mActiveTracks 5355 Vector< sp<RecordTrack> > activeTracks; 5356 5357 // reference to the (first and only) active fast track 5358 sp<RecordTrack> fastTrack; 5359 5360 // reference to a fast track which is about to be removed 5361 sp<RecordTrack> fastTrackToRemove; 5362 5363 { // scope for mLock 5364 Mutex::Autolock _l(mLock); 5365 5366 processConfigEvents_l(); 5367 5368 // check exitPending here because checkForNewParameters_l() and 5369 // checkForNewParameters_l() can temporarily release mLock 5370 if (exitPending()) { 5371 break; 5372 } 5373 5374 // if no active track(s), then standby and release wakelock 5375 size_t size = mActiveTracks.size(); 5376 if (size == 0) { 5377 standbyIfNotAlreadyInStandby(); 5378 // exitPending() can't become true here 5379 releaseWakeLock_l(); 5380 ALOGV("RecordThread: loop stopping"); 5381 // go to sleep 5382 mWaitWorkCV.wait(mLock); 5383 ALOGV("RecordThread: loop starting"); 5384 goto reacquire_wakelock; 5385 } 5386 5387 if (mActiveTracksGen != activeTracksGen) { 5388 activeTracksGen = mActiveTracksGen; 5389 SortedVector<int> tmp; 5390 for (size_t i = 0; i < size; i++) { 5391 tmp.add(mActiveTracks[i]->uid()); 5392 } 5393 updateWakeLockUids_l(tmp); 5394 } 5395 5396 bool doBroadcast = false; 5397 for (size_t i = 0; i < size; ) { 5398 5399 activeTrack = mActiveTracks[i]; 5400 if (activeTrack->isTerminated()) { 5401 if (activeTrack->isFastTrack()) { 5402 ALOG_ASSERT(fastTrackToRemove == 0); 5403 fastTrackToRemove = activeTrack; 5404 } 5405 removeTrack_l(activeTrack); 5406 mActiveTracks.remove(activeTrack); 5407 mActiveTracksGen++; 5408 size--; 5409 continue; 5410 } 5411 5412 TrackBase::track_state activeTrackState = activeTrack->mState; 5413 switch (activeTrackState) { 5414 5415 case TrackBase::PAUSING: 5416 mActiveTracks.remove(activeTrack); 5417 mActiveTracksGen++; 5418 doBroadcast = true; 5419 size--; 5420 continue; 5421 5422 case TrackBase::STARTING_1: 5423 sleepUs = 10000; 5424 i++; 5425 continue; 5426 5427 case TrackBase::STARTING_2: 5428 doBroadcast = true; 5429 mStandby = false; 5430 activeTrack->mState = TrackBase::ACTIVE; 5431 break; 5432 5433 case TrackBase::ACTIVE: 5434 break; 5435 5436 case TrackBase::IDLE: 5437 i++; 5438 continue; 5439 5440 default: 5441 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5442 } 5443 5444 activeTracks.add(activeTrack); 5445 i++; 5446 5447 if (activeTrack->isFastTrack()) { 5448 ALOG_ASSERT(!mFastTrackAvail); 5449 ALOG_ASSERT(fastTrack == 0); 5450 fastTrack = activeTrack; 5451 } 5452 } 5453 if (doBroadcast) { 5454 mStartStopCond.broadcast(); 5455 } 5456 5457 // sleep if there are no active tracks to process 5458 if (activeTracks.size() == 0) { 5459 if (sleepUs == 0) { 5460 sleepUs = kRecordThreadSleepUs; 5461 } 5462 continue; 5463 } 5464 sleepUs = 0; 5465 5466 lockEffectChains_l(effectChains); 5467 } 5468 5469 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5470 5471 size_t size = effectChains.size(); 5472 for (size_t i = 0; i < size; i++) { 5473 // thread mutex is not locked, but effect chain is locked 5474 effectChains[i]->process_l(); 5475 } 5476 5477 // Push a new fast capture state if fast capture is not already running, or cblk change 5478 if (mFastCapture != 0) { 5479 FastCaptureStateQueue *sq = mFastCapture->sq(); 5480 FastCaptureState *state = sq->begin(); 5481 bool didModify = false; 5482 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5483 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5484 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5485 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5486 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5487 if (old == -1) { 5488 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5489 } 5490 } 5491 state->mCommand = FastCaptureState::READ_WRITE; 5492#if 0 // FIXME 5493 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5494 FastThreadDumpState::kSamplingNforLowRamDevice : 5495 FastThreadDumpState::kSamplingN); 5496#endif 5497 didModify = true; 5498 } 5499 audio_track_cblk_t *cblkOld = state->mCblk; 5500 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5501 if (cblkNew != cblkOld) { 5502 state->mCblk = cblkNew; 5503 // block until acked if removing a fast track 5504 if (cblkOld != NULL) { 5505 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5506 } 5507 didModify = true; 5508 } 5509 sq->end(didModify); 5510 if (didModify) { 5511 sq->push(block); 5512#if 0 5513 if (kUseFastCapture == FastCapture_Dynamic) { 5514 mNormalSource = mPipeSource; 5515 } 5516#endif 5517 } 5518 } 5519 5520 // now run the fast track destructor with thread mutex unlocked 5521 fastTrackToRemove.clear(); 5522 5523 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5524 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5525 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5526 // If destination is non-contiguous, first read past the nominal end of buffer, then 5527 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5528 5529 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5530 ssize_t framesRead; 5531 5532 // If an NBAIO source is present, use it to read the normal capture's data 5533 if (mPipeSource != 0) { 5534 size_t framesToRead = mBufferSize / mFrameSize; 5535 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5536 framesToRead, AudioBufferProvider::kInvalidPTS); 5537 if (framesRead == 0) { 5538 // since pipe is non-blocking, simulate blocking input 5539 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5540 } 5541 // otherwise use the HAL / AudioStreamIn directly 5542 } else { 5543 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5544 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5545 if (bytesRead < 0) { 5546 framesRead = bytesRead; 5547 } else { 5548 framesRead = bytesRead / mFrameSize; 5549 } 5550 } 5551 5552 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5553 ALOGE("read failed: framesRead=%d", framesRead); 5554 // Force input into standby so that it tries to recover at next read attempt 5555 inputStandBy(); 5556 sleepUs = kRecordThreadSleepUs; 5557 } 5558 if (framesRead <= 0) { 5559 goto unlock; 5560 } 5561 ALOG_ASSERT(framesRead > 0); 5562 5563 if (mTeeSink != 0) { 5564 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5565 } 5566 // If destination is non-contiguous, we now correct for reading past end of buffer. 5567 { 5568 size_t part1 = mRsmpInFramesP2 - rear; 5569 if ((size_t) framesRead > part1) { 5570 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5571 (framesRead - part1) * mFrameSize); 5572 } 5573 } 5574 rear = mRsmpInRear += framesRead; 5575 5576 size = activeTracks.size(); 5577 // loop over each active track 5578 for (size_t i = 0; i < size; i++) { 5579 activeTrack = activeTracks[i]; 5580 5581 // skip fast tracks, as those are handled directly by FastCapture 5582 if (activeTrack->isFastTrack()) { 5583 continue; 5584 } 5585 5586 enum { 5587 OVERRUN_UNKNOWN, 5588 OVERRUN_TRUE, 5589 OVERRUN_FALSE 5590 } overrun = OVERRUN_UNKNOWN; 5591 5592 // loop over getNextBuffer to handle circular sink 5593 for (;;) { 5594 5595 activeTrack->mSink.frameCount = ~0; 5596 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5597 size_t framesOut = activeTrack->mSink.frameCount; 5598 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5599 5600 int32_t front = activeTrack->mRsmpInFront; 5601 ssize_t filled = rear - front; 5602 size_t framesIn; 5603 5604 if (filled < 0) { 5605 // should not happen, but treat like a massive overrun and re-sync 5606 framesIn = 0; 5607 activeTrack->mRsmpInFront = rear; 5608 overrun = OVERRUN_TRUE; 5609 } else if ((size_t) filled <= mRsmpInFrames) { 5610 framesIn = (size_t) filled; 5611 } else { 5612 // client is not keeping up with server, but give it latest data 5613 framesIn = mRsmpInFrames; 5614 activeTrack->mRsmpInFront = front = rear - framesIn; 5615 overrun = OVERRUN_TRUE; 5616 } 5617 5618 if (framesOut == 0 || framesIn == 0) { 5619 break; 5620 } 5621 5622 if (activeTrack->mResampler == NULL) { 5623 // no resampling 5624 if (framesIn > framesOut) { 5625 framesIn = framesOut; 5626 } else { 5627 framesOut = framesIn; 5628 } 5629 int8_t *dst = activeTrack->mSink.i8; 5630 while (framesIn > 0) { 5631 front &= mRsmpInFramesP2 - 1; 5632 size_t part1 = mRsmpInFramesP2 - front; 5633 if (part1 > framesIn) { 5634 part1 = framesIn; 5635 } 5636 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5637 if (mChannelCount == activeTrack->mChannelCount) { 5638 memcpy(dst, src, part1 * mFrameSize); 5639 } else if (mChannelCount == 1) { 5640 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5641 part1); 5642 } else { 5643 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 5644 (const int16_t *)src, part1); 5645 } 5646 dst += part1 * activeTrack->mFrameSize; 5647 front += part1; 5648 framesIn -= part1; 5649 } 5650 activeTrack->mRsmpInFront += framesOut; 5651 5652 } else { 5653 // resampling 5654 // FIXME framesInNeeded should really be part of resampler API, and should 5655 // depend on the SRC ratio 5656 // to keep mRsmpInBuffer full so resampler always has sufficient input 5657 size_t framesInNeeded; 5658 // FIXME only re-calculate when it changes, and optimize for common ratios 5659 // Do not precompute in/out because floating point is not associative 5660 // e.g. a*b/c != a*(b/c). 5661 const double in(mSampleRate); 5662 const double out(activeTrack->mSampleRate); 5663 framesInNeeded = ceil(framesOut * in / out) + 1; 5664 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5665 framesInNeeded, framesOut, in / out); 5666 // Although we theoretically have framesIn in circular buffer, some of those are 5667 // unreleased frames, and thus must be discounted for purpose of budgeting. 5668 size_t unreleased = activeTrack->mRsmpInUnrel; 5669 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5670 if (framesIn < framesInNeeded) { 5671 ALOGV("not enough to resample: have %u frames in but need %u in to " 5672 "produce %u out given in/out ratio of %.4g", 5673 framesIn, framesInNeeded, framesOut, in / out); 5674 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5675 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5676 if (newFramesOut == 0) { 5677 break; 5678 } 5679 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5680 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5681 framesInNeeded, newFramesOut, out / in); 5682 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5683 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5684 "given in/out ratio of %.4g", 5685 framesIn, framesInNeeded, newFramesOut, in / out); 5686 framesOut = newFramesOut; 5687 } else { 5688 ALOGV("success 1: have %u in and need %u in to produce %u out " 5689 "given in/out ratio of %.4g", 5690 framesIn, framesInNeeded, framesOut, in / out); 5691 } 5692 5693 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5694 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5695 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5696 delete[] activeTrack->mRsmpOutBuffer; 5697 // resampler always outputs stereo 5698 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5699 activeTrack->mRsmpOutFrameCount = framesOut; 5700 } 5701 5702 // resampler accumulates, but we only have one source track 5703 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5704 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5705 // FIXME how about having activeTrack implement this interface itself? 5706 activeTrack->mResamplerBufferProvider 5707 /*this*/ /* AudioBufferProvider* */); 5708 // ditherAndClamp() works as long as all buffers returned by 5709 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5710 if (activeTrack->mChannelCount == 1) { 5711 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5712 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5713 framesOut); 5714 // the resampler always outputs stereo samples: 5715 // do post stereo to mono conversion 5716 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5717 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5718 } else { 5719 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5720 activeTrack->mRsmpOutBuffer, framesOut); 5721 } 5722 // now done with mRsmpOutBuffer 5723 5724 } 5725 5726 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5727 overrun = OVERRUN_FALSE; 5728 } 5729 5730 if (activeTrack->mFramesToDrop == 0) { 5731 if (framesOut > 0) { 5732 activeTrack->mSink.frameCount = framesOut; 5733 activeTrack->releaseBuffer(&activeTrack->mSink); 5734 } 5735 } else { 5736 // FIXME could do a partial drop of framesOut 5737 if (activeTrack->mFramesToDrop > 0) { 5738 activeTrack->mFramesToDrop -= framesOut; 5739 if (activeTrack->mFramesToDrop <= 0) { 5740 activeTrack->clearSyncStartEvent(); 5741 } 5742 } else { 5743 activeTrack->mFramesToDrop += framesOut; 5744 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5745 activeTrack->mSyncStartEvent->isCancelled()) { 5746 ALOGW("Synced record %s, session %d, trigger session %d", 5747 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5748 activeTrack->sessionId(), 5749 (activeTrack->mSyncStartEvent != 0) ? 5750 activeTrack->mSyncStartEvent->triggerSession() : 0); 5751 activeTrack->clearSyncStartEvent(); 5752 } 5753 } 5754 } 5755 5756 if (framesOut == 0) { 5757 break; 5758 } 5759 } 5760 5761 switch (overrun) { 5762 case OVERRUN_TRUE: 5763 // client isn't retrieving buffers fast enough 5764 if (!activeTrack->setOverflow()) { 5765 nsecs_t now = systemTime(); 5766 // FIXME should lastWarning per track? 5767 if ((now - lastWarning) > kWarningThrottleNs) { 5768 ALOGW("RecordThread: buffer overflow"); 5769 lastWarning = now; 5770 } 5771 } 5772 break; 5773 case OVERRUN_FALSE: 5774 activeTrack->clearOverflow(); 5775 break; 5776 case OVERRUN_UNKNOWN: 5777 break; 5778 } 5779 5780 } 5781 5782unlock: 5783 // enable changes in effect chain 5784 unlockEffectChains(effectChains); 5785 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5786 } 5787 5788 standbyIfNotAlreadyInStandby(); 5789 5790 { 5791 Mutex::Autolock _l(mLock); 5792 for (size_t i = 0; i < mTracks.size(); i++) { 5793 sp<RecordTrack> track = mTracks[i]; 5794 track->invalidate(); 5795 } 5796 mActiveTracks.clear(); 5797 mActiveTracksGen++; 5798 mStartStopCond.broadcast(); 5799 } 5800 5801 releaseWakeLock(); 5802 5803 ALOGV("RecordThread %p exiting", this); 5804 return false; 5805} 5806 5807void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5808{ 5809 if (!mStandby) { 5810 inputStandBy(); 5811 mStandby = true; 5812 } 5813} 5814 5815void AudioFlinger::RecordThread::inputStandBy() 5816{ 5817 // Idle the fast capture if it's currently running 5818 if (mFastCapture != 0) { 5819 FastCaptureStateQueue *sq = mFastCapture->sq(); 5820 FastCaptureState *state = sq->begin(); 5821 if (!(state->mCommand & FastCaptureState::IDLE)) { 5822 state->mCommand = FastCaptureState::COLD_IDLE; 5823 state->mColdFutexAddr = &mFastCaptureFutex; 5824 state->mColdGen++; 5825 mFastCaptureFutex = 0; 5826 sq->end(); 5827 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5828 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5829#if 0 5830 if (kUseFastCapture == FastCapture_Dynamic) { 5831 // FIXME 5832 } 5833#endif 5834#ifdef AUDIO_WATCHDOG 5835 // FIXME 5836#endif 5837 } else { 5838 sq->end(false /*didModify*/); 5839 } 5840 } 5841 mInput->stream->common.standby(&mInput->stream->common); 5842} 5843 5844// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5845sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5846 const sp<AudioFlinger::Client>& client, 5847 uint32_t sampleRate, 5848 audio_format_t format, 5849 audio_channel_mask_t channelMask, 5850 size_t *pFrameCount, 5851 int sessionId, 5852 size_t *notificationFrames, 5853 int uid, 5854 IAudioFlinger::track_flags_t *flags, 5855 pid_t tid, 5856 status_t *status) 5857{ 5858 size_t frameCount = *pFrameCount; 5859 sp<RecordTrack> track; 5860 status_t lStatus; 5861 5862 // client expresses a preference for FAST, but we get the final say 5863 if (*flags & IAudioFlinger::TRACK_FAST) { 5864 if ( 5865 // use case: callback handler 5866 (tid != -1) && 5867 // frame count is not specified, or is exactly the pipe depth 5868 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5869 // PCM data 5870 audio_is_linear_pcm(format) && 5871 // native format 5872 (format == mFormat) && 5873 // native channel mask 5874 (channelMask == mChannelMask) && 5875 // native hardware sample rate 5876 (sampleRate == mSampleRate) && 5877 // record thread has an associated fast capture 5878 hasFastCapture() && 5879 // there are sufficient fast track slots available 5880 mFastTrackAvail 5881 ) { 5882 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5883 frameCount, mFrameCount); 5884 } else { 5885 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5886 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5887 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5888 frameCount, mFrameCount, mPipeFramesP2, 5889 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5890 hasFastCapture(), tid, mFastTrackAvail); 5891 *flags &= ~IAudioFlinger::TRACK_FAST; 5892 } 5893 } 5894 5895 // compute track buffer size in frames, and suggest the notification frame count 5896 if (*flags & IAudioFlinger::TRACK_FAST) { 5897 // fast track: frame count is exactly the pipe depth 5898 frameCount = mPipeFramesP2; 5899 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5900 *notificationFrames = mFrameCount; 5901 } else { 5902 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5903 // or 20 ms if there is a fast capture 5904 // TODO This could be a roundupRatio inline, and const 5905 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5906 * sampleRate + mSampleRate - 1) / mSampleRate; 5907 // minimum number of notification periods is at least kMinNotifications, 5908 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5909 static const size_t kMinNotifications = 3; 5910 static const uint32_t kMinMs = 30; 5911 // TODO This could be a roundupRatio inline 5912 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5913 // TODO This could be a roundupRatio inline 5914 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5915 maxNotificationFrames; 5916 const size_t minFrameCount = maxNotificationFrames * 5917 max(kMinNotifications, minNotificationsByMs); 5918 frameCount = max(frameCount, minFrameCount); 5919 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5920 *notificationFrames = maxNotificationFrames; 5921 } 5922 } 5923 *pFrameCount = frameCount; 5924 5925 lStatus = initCheck(); 5926 if (lStatus != NO_ERROR) { 5927 ALOGE("createRecordTrack_l() audio driver not initialized"); 5928 goto Exit; 5929 } 5930 5931 { // scope for mLock 5932 Mutex::Autolock _l(mLock); 5933 5934 track = new RecordTrack(this, client, sampleRate, 5935 format, channelMask, frameCount, NULL, sessionId, uid, 5936 *flags, TrackBase::TYPE_DEFAULT); 5937 5938 lStatus = track->initCheck(); 5939 if (lStatus != NO_ERROR) { 5940 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5941 // track must be cleared from the caller as the caller has the AF lock 5942 goto Exit; 5943 } 5944 mTracks.add(track); 5945 5946 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5947 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5948 mAudioFlinger->btNrecIsOff(); 5949 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5950 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5951 5952 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5953 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5954 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5955 // so ask activity manager to do this on our behalf 5956 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5957 } 5958 } 5959 5960 lStatus = NO_ERROR; 5961 5962Exit: 5963 *status = lStatus; 5964 return track; 5965} 5966 5967status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5968 AudioSystem::sync_event_t event, 5969 int triggerSession) 5970{ 5971 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5972 sp<ThreadBase> strongMe = this; 5973 status_t status = NO_ERROR; 5974 5975 if (event == AudioSystem::SYNC_EVENT_NONE) { 5976 recordTrack->clearSyncStartEvent(); 5977 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5978 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5979 triggerSession, 5980 recordTrack->sessionId(), 5981 syncStartEventCallback, 5982 recordTrack); 5983 // Sync event can be cancelled by the trigger session if the track is not in a 5984 // compatible state in which case we start record immediately 5985 if (recordTrack->mSyncStartEvent->isCancelled()) { 5986 recordTrack->clearSyncStartEvent(); 5987 } else { 5988 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5989 recordTrack->mFramesToDrop = - 5990 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5991 } 5992 } 5993 5994 { 5995 // This section is a rendezvous between binder thread executing start() and RecordThread 5996 AutoMutex lock(mLock); 5997 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5998 if (recordTrack->mState == TrackBase::PAUSING) { 5999 ALOGV("active record track PAUSING -> ACTIVE"); 6000 recordTrack->mState = TrackBase::ACTIVE; 6001 } else { 6002 ALOGV("active record track state %d", recordTrack->mState); 6003 } 6004 return status; 6005 } 6006 6007 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6008 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6009 // or using a separate command thread 6010 recordTrack->mState = TrackBase::STARTING_1; 6011 mActiveTracks.add(recordTrack); 6012 mActiveTracksGen++; 6013 status_t status = NO_ERROR; 6014 if (recordTrack->isExternalTrack()) { 6015 mLock.unlock(); 6016 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6017 mLock.lock(); 6018 // FIXME should verify that recordTrack is still in mActiveTracks 6019 if (status != NO_ERROR) { 6020 mActiveTracks.remove(recordTrack); 6021 mActiveTracksGen++; 6022 recordTrack->clearSyncStartEvent(); 6023 ALOGV("RecordThread::start error %d", status); 6024 return status; 6025 } 6026 } 6027 // Catch up with current buffer indices if thread is already running. 6028 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6029 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6030 // see previously buffered data before it called start(), but with greater risk of overrun. 6031 6032 recordTrack->mRsmpInFront = mRsmpInRear; 6033 recordTrack->mRsmpInUnrel = 0; 6034 // FIXME why reset? 6035 if (recordTrack->mResampler != NULL) { 6036 recordTrack->mResampler->reset(); 6037 } 6038 recordTrack->mState = TrackBase::STARTING_2; 6039 // signal thread to start 6040 mWaitWorkCV.broadcast(); 6041 if (mActiveTracks.indexOf(recordTrack) < 0) { 6042 ALOGV("Record failed to start"); 6043 status = BAD_VALUE; 6044 goto startError; 6045 } 6046 return status; 6047 } 6048 6049startError: 6050 if (recordTrack->isExternalTrack()) { 6051 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6052 } 6053 recordTrack->clearSyncStartEvent(); 6054 // FIXME I wonder why we do not reset the state here? 6055 return status; 6056} 6057 6058void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6059{ 6060 sp<SyncEvent> strongEvent = event.promote(); 6061 6062 if (strongEvent != 0) { 6063 sp<RefBase> ptr = strongEvent->cookie().promote(); 6064 if (ptr != 0) { 6065 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6066 recordTrack->handleSyncStartEvent(strongEvent); 6067 } 6068 } 6069} 6070 6071bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6072 ALOGV("RecordThread::stop"); 6073 AutoMutex _l(mLock); 6074 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6075 return false; 6076 } 6077 // note that threadLoop may still be processing the track at this point [without lock] 6078 recordTrack->mState = TrackBase::PAUSING; 6079 // do not wait for mStartStopCond if exiting 6080 if (exitPending()) { 6081 return true; 6082 } 6083 // FIXME incorrect usage of wait: no explicit predicate or loop 6084 mStartStopCond.wait(mLock); 6085 // if we have been restarted, recordTrack is in mActiveTracks here 6086 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6087 ALOGV("Record stopped OK"); 6088 return true; 6089 } 6090 return false; 6091} 6092 6093bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6094{ 6095 return false; 6096} 6097 6098status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6099{ 6100#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6101 if (!isValidSyncEvent(event)) { 6102 return BAD_VALUE; 6103 } 6104 6105 int eventSession = event->triggerSession(); 6106 status_t ret = NAME_NOT_FOUND; 6107 6108 Mutex::Autolock _l(mLock); 6109 6110 for (size_t i = 0; i < mTracks.size(); i++) { 6111 sp<RecordTrack> track = mTracks[i]; 6112 if (eventSession == track->sessionId()) { 6113 (void) track->setSyncEvent(event); 6114 ret = NO_ERROR; 6115 } 6116 } 6117 return ret; 6118#else 6119 return BAD_VALUE; 6120#endif 6121} 6122 6123// destroyTrack_l() must be called with ThreadBase::mLock held 6124void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6125{ 6126 track->terminate(); 6127 track->mState = TrackBase::STOPPED; 6128 // active tracks are removed by threadLoop() 6129 if (mActiveTracks.indexOf(track) < 0) { 6130 removeTrack_l(track); 6131 } 6132} 6133 6134void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6135{ 6136 mTracks.remove(track); 6137 // need anything related to effects here? 6138 if (track->isFastTrack()) { 6139 ALOG_ASSERT(!mFastTrackAvail); 6140 mFastTrackAvail = true; 6141 } 6142} 6143 6144void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6145{ 6146 dumpInternals(fd, args); 6147 dumpTracks(fd, args); 6148 dumpEffectChains(fd, args); 6149} 6150 6151void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6152{ 6153 dprintf(fd, "\nInput thread %p:\n", this); 6154 6155 if (mActiveTracks.size() > 0) { 6156 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 6157 } else { 6158 dprintf(fd, " No active record clients\n"); 6159 } 6160 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6161 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6162 6163 dumpBase(fd, args); 6164} 6165 6166void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6167{ 6168 const size_t SIZE = 256; 6169 char buffer[SIZE]; 6170 String8 result; 6171 6172 size_t numtracks = mTracks.size(); 6173 size_t numactive = mActiveTracks.size(); 6174 size_t numactiveseen = 0; 6175 dprintf(fd, " %d Tracks", numtracks); 6176 if (numtracks) { 6177 dprintf(fd, " of which %d are active\n", numactive); 6178 RecordTrack::appendDumpHeader(result); 6179 for (size_t i = 0; i < numtracks ; ++i) { 6180 sp<RecordTrack> track = mTracks[i]; 6181 if (track != 0) { 6182 bool active = mActiveTracks.indexOf(track) >= 0; 6183 if (active) { 6184 numactiveseen++; 6185 } 6186 track->dump(buffer, SIZE, active); 6187 result.append(buffer); 6188 } 6189 } 6190 } else { 6191 dprintf(fd, "\n"); 6192 } 6193 6194 if (numactiveseen != numactive) { 6195 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6196 " not in the track list\n"); 6197 result.append(buffer); 6198 RecordTrack::appendDumpHeader(result); 6199 for (size_t i = 0; i < numactive; ++i) { 6200 sp<RecordTrack> track = mActiveTracks[i]; 6201 if (mTracks.indexOf(track) < 0) { 6202 track->dump(buffer, SIZE, true); 6203 result.append(buffer); 6204 } 6205 } 6206 6207 } 6208 write(fd, result.string(), result.size()); 6209} 6210 6211// AudioBufferProvider interface 6212status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6213 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6214{ 6215 RecordTrack *activeTrack = mRecordTrack; 6216 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 6217 if (threadBase == 0) { 6218 buffer->frameCount = 0; 6219 buffer->raw = NULL; 6220 return NOT_ENOUGH_DATA; 6221 } 6222 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6223 int32_t rear = recordThread->mRsmpInRear; 6224 int32_t front = activeTrack->mRsmpInFront; 6225 ssize_t filled = rear - front; 6226 // FIXME should not be P2 (don't want to increase latency) 6227 // FIXME if client not keeping up, discard 6228 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6229 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6230 front &= recordThread->mRsmpInFramesP2 - 1; 6231 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6232 if (part1 > (size_t) filled) { 6233 part1 = filled; 6234 } 6235 size_t ask = buffer->frameCount; 6236 ALOG_ASSERT(ask > 0); 6237 if (part1 > ask) { 6238 part1 = ask; 6239 } 6240 if (part1 == 0) { 6241 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 6242 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 6243 buffer->raw = NULL; 6244 buffer->frameCount = 0; 6245 activeTrack->mRsmpInUnrel = 0; 6246 return NOT_ENOUGH_DATA; 6247 } 6248 6249 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6250 buffer->frameCount = part1; 6251 activeTrack->mRsmpInUnrel = part1; 6252 return NO_ERROR; 6253} 6254 6255// AudioBufferProvider interface 6256void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6257 AudioBufferProvider::Buffer* buffer) 6258{ 6259 RecordTrack *activeTrack = mRecordTrack; 6260 size_t stepCount = buffer->frameCount; 6261 if (stepCount == 0) { 6262 return; 6263 } 6264 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 6265 activeTrack->mRsmpInUnrel -= stepCount; 6266 activeTrack->mRsmpInFront += stepCount; 6267 buffer->raw = NULL; 6268 buffer->frameCount = 0; 6269} 6270 6271bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6272 status_t& status) 6273{ 6274 bool reconfig = false; 6275 6276 status = NO_ERROR; 6277 6278 audio_format_t reqFormat = mFormat; 6279 uint32_t samplingRate = mSampleRate; 6280 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6281 6282 AudioParameter param = AudioParameter(keyValuePair); 6283 int value; 6284 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6285 // channel count change can be requested. Do we mandate the first client defines the 6286 // HAL sampling rate and channel count or do we allow changes on the fly? 6287 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6288 samplingRate = value; 6289 reconfig = true; 6290 } 6291 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6292 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 6293 status = BAD_VALUE; 6294 } else { 6295 reqFormat = (audio_format_t) value; 6296 reconfig = true; 6297 } 6298 } 6299 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6300 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6301 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6302 status = BAD_VALUE; 6303 } else { 6304 channelMask = mask; 6305 reconfig = true; 6306 } 6307 } 6308 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6309 // do not accept frame count changes if tracks are open as the track buffer 6310 // size depends on frame count and correct behavior would not be guaranteed 6311 // if frame count is changed after track creation 6312 if (mActiveTracks.size() > 0) { 6313 status = INVALID_OPERATION; 6314 } else { 6315 reconfig = true; 6316 } 6317 } 6318 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6319 // forward device change to effects that have requested to be 6320 // aware of attached audio device. 6321 for (size_t i = 0; i < mEffectChains.size(); i++) { 6322 mEffectChains[i]->setDevice_l(value); 6323 } 6324 6325 // store input device and output device but do not forward output device to audio HAL. 6326 // Note that status is ignored by the caller for output device 6327 // (see AudioFlinger::setParameters() 6328 if (audio_is_output_devices(value)) { 6329 mOutDevice = value; 6330 status = BAD_VALUE; 6331 } else { 6332 mInDevice = value; 6333 // disable AEC and NS if the device is a BT SCO headset supporting those 6334 // pre processings 6335 if (mTracks.size() > 0) { 6336 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6337 mAudioFlinger->btNrecIsOff(); 6338 for (size_t i = 0; i < mTracks.size(); i++) { 6339 sp<RecordTrack> track = mTracks[i]; 6340 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6341 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6342 } 6343 } 6344 } 6345 } 6346 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6347 mAudioSource != (audio_source_t)value) { 6348 // forward device change to effects that have requested to be 6349 // aware of attached audio device. 6350 for (size_t i = 0; i < mEffectChains.size(); i++) { 6351 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6352 } 6353 mAudioSource = (audio_source_t)value; 6354 } 6355 6356 if (status == NO_ERROR) { 6357 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6358 keyValuePair.string()); 6359 if (status == INVALID_OPERATION) { 6360 inputStandBy(); 6361 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6362 keyValuePair.string()); 6363 } 6364 if (reconfig) { 6365 if (status == BAD_VALUE && 6366 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6367 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6368 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6369 <= (2 * samplingRate)) && 6370 audio_channel_count_from_in_mask( 6371 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6372 (channelMask == AUDIO_CHANNEL_IN_MONO || 6373 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6374 status = NO_ERROR; 6375 } 6376 if (status == NO_ERROR) { 6377 readInputParameters_l(); 6378 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6379 } 6380 } 6381 } 6382 6383 return reconfig; 6384} 6385 6386String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6387{ 6388 Mutex::Autolock _l(mLock); 6389 if (initCheck() != NO_ERROR) { 6390 return String8(); 6391 } 6392 6393 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6394 const String8 out_s8(s); 6395 free(s); 6396 return out_s8; 6397} 6398 6399void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6400 AudioSystem::OutputDescriptor desc; 6401 const void *param2 = NULL; 6402 6403 switch (event) { 6404 case AudioSystem::INPUT_OPENED: 6405 case AudioSystem::INPUT_CONFIG_CHANGED: 6406 desc.channelMask = mChannelMask; 6407 desc.samplingRate = mSampleRate; 6408 desc.format = mFormat; 6409 desc.frameCount = mFrameCount; 6410 desc.latency = 0; 6411 param2 = &desc; 6412 break; 6413 6414 case AudioSystem::INPUT_CLOSED: 6415 default: 6416 break; 6417 } 6418 mAudioFlinger->audioConfigChanged(event, mId, param2); 6419} 6420 6421void AudioFlinger::RecordThread::readInputParameters_l() 6422{ 6423 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6424 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6425 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6426 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6427 mFormat = mHALFormat; 6428 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6429 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6430 } 6431 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6432 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6433 mFrameCount = mBufferSize / mFrameSize; 6434 // This is the formula for calculating the temporary buffer size. 6435 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6436 // 1 full output buffer, regardless of the alignment of the available input. 6437 // The value is somewhat arbitrary, and could probably be even larger. 6438 // A larger value should allow more old data to be read after a track calls start(), 6439 // without increasing latency. 6440 mRsmpInFrames = mFrameCount * 7; 6441 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6442 delete[] mRsmpInBuffer; 6443 6444 // TODO optimize audio capture buffer sizes ... 6445 // Here we calculate the size of the sliding buffer used as a source 6446 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6447 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6448 // be better to have it derived from the pipe depth in the long term. 6449 // The current value is higher than necessary. However it should not add to latency. 6450 6451 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6452 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6453 6454 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6455 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6456} 6457 6458uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6459{ 6460 Mutex::Autolock _l(mLock); 6461 if (initCheck() != NO_ERROR) { 6462 return 0; 6463 } 6464 6465 return mInput->stream->get_input_frames_lost(mInput->stream); 6466} 6467 6468uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6469{ 6470 Mutex::Autolock _l(mLock); 6471 uint32_t result = 0; 6472 if (getEffectChain_l(sessionId) != 0) { 6473 result = EFFECT_SESSION; 6474 } 6475 6476 for (size_t i = 0; i < mTracks.size(); ++i) { 6477 if (sessionId == mTracks[i]->sessionId()) { 6478 result |= TRACK_SESSION; 6479 break; 6480 } 6481 } 6482 6483 return result; 6484} 6485 6486KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6487{ 6488 KeyedVector<int, bool> ids; 6489 Mutex::Autolock _l(mLock); 6490 for (size_t j = 0; j < mTracks.size(); ++j) { 6491 sp<RecordThread::RecordTrack> track = mTracks[j]; 6492 int sessionId = track->sessionId(); 6493 if (ids.indexOfKey(sessionId) < 0) { 6494 ids.add(sessionId, true); 6495 } 6496 } 6497 return ids; 6498} 6499 6500AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6501{ 6502 Mutex::Autolock _l(mLock); 6503 AudioStreamIn *input = mInput; 6504 mInput = NULL; 6505 return input; 6506} 6507 6508// this method must always be called either with ThreadBase mLock held or inside the thread loop 6509audio_stream_t* AudioFlinger::RecordThread::stream() const 6510{ 6511 if (mInput == NULL) { 6512 return NULL; 6513 } 6514 return &mInput->stream->common; 6515} 6516 6517status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6518{ 6519 // only one chain per input thread 6520 if (mEffectChains.size() != 0) { 6521 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6522 return INVALID_OPERATION; 6523 } 6524 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6525 chain->setThread(this); 6526 chain->setInBuffer(NULL); 6527 chain->setOutBuffer(NULL); 6528 6529 checkSuspendOnAddEffectChain_l(chain); 6530 6531 // make sure enabled pre processing effects state is communicated to the HAL as we 6532 // just moved them to a new input stream. 6533 chain->syncHalEffectsState(); 6534 6535 mEffectChains.add(chain); 6536 6537 return NO_ERROR; 6538} 6539 6540size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6541{ 6542 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6543 ALOGW_IF(mEffectChains.size() != 1, 6544 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6545 chain.get(), mEffectChains.size(), this); 6546 if (mEffectChains.size() == 1) { 6547 mEffectChains.removeAt(0); 6548 } 6549 return 0; 6550} 6551 6552status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6553 audio_patch_handle_t *handle) 6554{ 6555 status_t status = NO_ERROR; 6556 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6557 // store new device and send to effects 6558 mInDevice = patch->sources[0].ext.device.type; 6559 for (size_t i = 0; i < mEffectChains.size(); i++) { 6560 mEffectChains[i]->setDevice_l(mInDevice); 6561 } 6562 6563 // disable AEC and NS if the device is a BT SCO headset supporting those 6564 // pre processings 6565 if (mTracks.size() > 0) { 6566 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6567 mAudioFlinger->btNrecIsOff(); 6568 for (size_t i = 0; i < mTracks.size(); i++) { 6569 sp<RecordTrack> track = mTracks[i]; 6570 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6571 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6572 } 6573 } 6574 6575 // store new source and send to effects 6576 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6577 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6578 for (size_t i = 0; i < mEffectChains.size(); i++) { 6579 mEffectChains[i]->setAudioSource_l(mAudioSource); 6580 } 6581 } 6582 6583 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6584 status = hwDevice->create_audio_patch(hwDevice, 6585 patch->num_sources, 6586 patch->sources, 6587 patch->num_sinks, 6588 patch->sinks, 6589 handle); 6590 } else { 6591 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6592 } 6593 return status; 6594} 6595 6596status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6597{ 6598 status_t status = NO_ERROR; 6599 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6600 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6601 status = hwDevice->release_audio_patch(hwDevice, handle); 6602 } else { 6603 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6604 } 6605 return status; 6606} 6607 6608void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6609{ 6610 Mutex::Autolock _l(mLock); 6611 mTracks.add(record); 6612} 6613 6614void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6615{ 6616 Mutex::Autolock _l(mLock); 6617 destroyTrack_l(record); 6618} 6619 6620void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6621{ 6622 ThreadBase::getAudioPortConfig(config); 6623 config->role = AUDIO_PORT_ROLE_SINK; 6624 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6625 config->ext.mix.usecase.source = mAudioSource; 6626} 6627 6628} // namespace android 6629