Threads.cpp revision d7dca050c630bddbd73a6623271b34b4290460ee
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: callback handler and frame count is default or at least as large as HAL 1549 ( 1550 (tid != -1) && 1551 ((frameCount == 0) || 1552 (frameCount >= mFrameCount)) 1553 ) 1554 ) && 1555 // PCM data 1556 audio_is_linear_pcm(format) && 1557 // identical channel mask to sink, or mono in and stereo sink 1558 (channelMask == mChannelMask || 1559 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1560 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1561 // hardware sample rate 1562 (sampleRate == mSampleRate) && 1563 // normal mixer has an associated fast mixer 1564 hasFastMixer() && 1565 // there are sufficient fast track slots available 1566 (mFastTrackAvailMask != 0) 1567 // FIXME test that MixerThread for this fast track has a capable output HAL 1568 // FIXME add a permission test also? 1569 ) { 1570 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1571 if (frameCount == 0) { 1572 // read the fast track multiplier property the first time it is needed 1573 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1574 if (ok != 0) { 1575 ALOGE("%s pthread_once failed: %d", __func__, ok); 1576 } 1577 frameCount = mFrameCount * sFastTrackMultiplier; 1578 } 1579 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1580 frameCount, mFrameCount); 1581 } else { 1582 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1583 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1584 "sampleRate=%u mSampleRate=%u " 1585 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1586 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1587 audio_is_linear_pcm(format), 1588 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1589 *flags &= ~IAudioFlinger::TRACK_FAST; 1590 } 1591 } 1592 // For normal PCM streaming tracks, update minimum frame count. 1593 // For compatibility with AudioTrack calculation, buffer depth is forced 1594 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1595 // This is probably too conservative, but legacy application code may depend on it. 1596 // If you change this calculation, also review the start threshold which is related. 1597 if (!(*flags & IAudioFlinger::TRACK_FAST) 1598 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1599 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1600 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1601 if (minBufCount < 2) { 1602 minBufCount = 2; 1603 } 1604 size_t minFrameCount = 1605 minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate); 1606 if (frameCount < minFrameCount) { // including frameCount == 0 1607 frameCount = minFrameCount; 1608 } 1609 } 1610 *pFrameCount = frameCount; 1611 1612 switch (mType) { 1613 1614 case DIRECT: 1615 if (audio_is_linear_pcm(format)) { 1616 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1617 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1618 "for output %p with format %#x", 1619 sampleRate, format, channelMask, mOutput, mFormat); 1620 lStatus = BAD_VALUE; 1621 goto Exit; 1622 } 1623 } 1624 break; 1625 1626 case OFFLOAD: 1627 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1628 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1629 "for output %p with format %#x", 1630 sampleRate, format, channelMask, mOutput, mFormat); 1631 lStatus = BAD_VALUE; 1632 goto Exit; 1633 } 1634 break; 1635 1636 default: 1637 if (!audio_is_linear_pcm(format)) { 1638 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1639 "for output %p with format %#x", 1640 format, mOutput, mFormat); 1641 lStatus = BAD_VALUE; 1642 goto Exit; 1643 } 1644 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1645 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1646 lStatus = BAD_VALUE; 1647 goto Exit; 1648 } 1649 break; 1650 1651 } 1652 1653 lStatus = initCheck(); 1654 if (lStatus != NO_ERROR) { 1655 ALOGE("createTrack_l() audio driver not initialized"); 1656 goto Exit; 1657 } 1658 1659 { // scope for mLock 1660 Mutex::Autolock _l(mLock); 1661 1662 // all tracks in same audio session must share the same routing strategy otherwise 1663 // conflicts will happen when tracks are moved from one output to another by audio policy 1664 // manager 1665 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1666 for (size_t i = 0; i < mTracks.size(); ++i) { 1667 sp<Track> t = mTracks[i]; 1668 if (t != 0 && t->isExternalTrack()) { 1669 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1670 if (sessionId == t->sessionId() && strategy != actual) { 1671 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1672 strategy, actual); 1673 lStatus = BAD_VALUE; 1674 goto Exit; 1675 } 1676 } 1677 } 1678 1679 if (!isTimed) { 1680 track = new Track(this, client, streamType, sampleRate, format, 1681 channelMask, frameCount, NULL, sharedBuffer, 1682 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1683 } else { 1684 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1685 channelMask, frameCount, sharedBuffer, sessionId, uid); 1686 } 1687 1688 // new Track always returns non-NULL, 1689 // but TimedTrack::create() is a factory that could fail by returning NULL 1690 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1691 if (lStatus != NO_ERROR) { 1692 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1693 // track must be cleared from the caller as the caller has the AF lock 1694 goto Exit; 1695 } 1696 mTracks.add(track); 1697 1698 sp<EffectChain> chain = getEffectChain_l(sessionId); 1699 if (chain != 0) { 1700 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1701 track->setMainBuffer(chain->inBuffer()); 1702 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1703 chain->incTrackCnt(); 1704 } 1705 1706 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1707 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1708 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1709 // so ask activity manager to do this on our behalf 1710 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1711 } 1712 } 1713 1714 lStatus = NO_ERROR; 1715 1716Exit: 1717 *status = lStatus; 1718 return track; 1719} 1720 1721uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1722{ 1723 return latency; 1724} 1725 1726uint32_t AudioFlinger::PlaybackThread::latency() const 1727{ 1728 Mutex::Autolock _l(mLock); 1729 return latency_l(); 1730} 1731uint32_t AudioFlinger::PlaybackThread::latency_l() const 1732{ 1733 if (initCheck() == NO_ERROR) { 1734 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1735 } else { 1736 return 0; 1737 } 1738} 1739 1740void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1741{ 1742 Mutex::Autolock _l(mLock); 1743 // Don't apply master volume in SW if our HAL can do it for us. 1744 if (mOutput && mOutput->audioHwDev && 1745 mOutput->audioHwDev->canSetMasterVolume()) { 1746 mMasterVolume = 1.0; 1747 } else { 1748 mMasterVolume = value; 1749 } 1750} 1751 1752void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1753{ 1754 Mutex::Autolock _l(mLock); 1755 // Don't apply master mute in SW if our HAL can do it for us. 1756 if (mOutput && mOutput->audioHwDev && 1757 mOutput->audioHwDev->canSetMasterMute()) { 1758 mMasterMute = false; 1759 } else { 1760 mMasterMute = muted; 1761 } 1762} 1763 1764void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1765{ 1766 Mutex::Autolock _l(mLock); 1767 mStreamTypes[stream].volume = value; 1768 broadcast_l(); 1769} 1770 1771void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1772{ 1773 Mutex::Autolock _l(mLock); 1774 mStreamTypes[stream].mute = muted; 1775 broadcast_l(); 1776} 1777 1778float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1779{ 1780 Mutex::Autolock _l(mLock); 1781 return mStreamTypes[stream].volume; 1782} 1783 1784// addTrack_l() must be called with ThreadBase::mLock held 1785status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1786{ 1787 status_t status = ALREADY_EXISTS; 1788 1789 // set retry count for buffer fill 1790 track->mRetryCount = kMaxTrackStartupRetries; 1791 if (mActiveTracks.indexOf(track) < 0) { 1792 // the track is newly added, make sure it fills up all its 1793 // buffers before playing. This is to ensure the client will 1794 // effectively get the latency it requested. 1795 if (track->isExternalTrack()) { 1796 TrackBase::track_state state = track->mState; 1797 mLock.unlock(); 1798 status = AudioSystem::startOutput(mId, track->streamType(), 1799 (audio_session_t)track->sessionId()); 1800 mLock.lock(); 1801 // abort track was stopped/paused while we released the lock 1802 if (state != track->mState) { 1803 if (status == NO_ERROR) { 1804 mLock.unlock(); 1805 AudioSystem::stopOutput(mId, track->streamType(), 1806 (audio_session_t)track->sessionId()); 1807 mLock.lock(); 1808 } 1809 return INVALID_OPERATION; 1810 } 1811 // abort if start is rejected by audio policy manager 1812 if (status != NO_ERROR) { 1813 return PERMISSION_DENIED; 1814 } 1815#ifdef ADD_BATTERY_DATA 1816 // to track the speaker usage 1817 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1818#endif 1819 } 1820 1821 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1822 track->mResetDone = false; 1823 track->mPresentationCompleteFrames = 0; 1824 mActiveTracks.add(track); 1825 mWakeLockUids.add(track->uid()); 1826 mActiveTracksGeneration++; 1827 mLatestActiveTrack = track; 1828 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1829 if (chain != 0) { 1830 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1831 track->sessionId()); 1832 chain->incActiveTrackCnt(); 1833 } 1834 1835 status = NO_ERROR; 1836 } 1837 1838 onAddNewTrack_l(); 1839 return status; 1840} 1841 1842bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1843{ 1844 track->terminate(); 1845 // active tracks are removed by threadLoop() 1846 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1847 track->mState = TrackBase::STOPPED; 1848 if (!trackActive) { 1849 removeTrack_l(track); 1850 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1851 track->mState = TrackBase::STOPPING_1; 1852 } 1853 1854 return trackActive; 1855} 1856 1857void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1858{ 1859 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1860 mTracks.remove(track); 1861 deleteTrackName_l(track->name()); 1862 // redundant as track is about to be destroyed, for dumpsys only 1863 track->mName = -1; 1864 if (track->isFastTrack()) { 1865 int index = track->mFastIndex; 1866 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1867 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1868 mFastTrackAvailMask |= 1 << index; 1869 // redundant as track is about to be destroyed, for dumpsys only 1870 track->mFastIndex = -1; 1871 } 1872 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1873 if (chain != 0) { 1874 chain->decTrackCnt(); 1875 } 1876} 1877 1878void AudioFlinger::PlaybackThread::broadcast_l() 1879{ 1880 // Thread could be blocked waiting for async 1881 // so signal it to handle state changes immediately 1882 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1883 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1884 mSignalPending = true; 1885 mWaitWorkCV.broadcast(); 1886} 1887 1888String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1889{ 1890 Mutex::Autolock _l(mLock); 1891 if (initCheck() != NO_ERROR) { 1892 return String8(); 1893 } 1894 1895 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1896 const String8 out_s8(s); 1897 free(s); 1898 return out_s8; 1899} 1900 1901void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1902 AudioSystem::OutputDescriptor desc; 1903 void *param2 = NULL; 1904 1905 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1906 param); 1907 1908 switch (event) { 1909 case AudioSystem::OUTPUT_OPENED: 1910 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1911 desc.channelMask = mChannelMask; 1912 desc.samplingRate = mSampleRate; 1913 desc.format = mFormat; 1914 desc.frameCount = mNormalFrameCount; // FIXME see 1915 // AudioFlinger::frameCount(audio_io_handle_t) 1916 desc.latency = latency_l(); 1917 param2 = &desc; 1918 break; 1919 1920 case AudioSystem::STREAM_CONFIG_CHANGED: 1921 param2 = ¶m; 1922 case AudioSystem::OUTPUT_CLOSED: 1923 default: 1924 break; 1925 } 1926 mAudioFlinger->audioConfigChanged(event, mId, param2); 1927} 1928 1929void AudioFlinger::PlaybackThread::writeCallback() 1930{ 1931 ALOG_ASSERT(mCallbackThread != 0); 1932 mCallbackThread->resetWriteBlocked(); 1933} 1934 1935void AudioFlinger::PlaybackThread::drainCallback() 1936{ 1937 ALOG_ASSERT(mCallbackThread != 0); 1938 mCallbackThread->resetDraining(); 1939} 1940 1941void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1942{ 1943 Mutex::Autolock _l(mLock); 1944 // reject out of sequence requests 1945 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1946 mWriteAckSequence &= ~1; 1947 mWaitWorkCV.signal(); 1948 } 1949} 1950 1951void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1952{ 1953 Mutex::Autolock _l(mLock); 1954 // reject out of sequence requests 1955 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1956 mDrainSequence &= ~1; 1957 mWaitWorkCV.signal(); 1958 } 1959} 1960 1961// static 1962int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1963 void *param __unused, 1964 void *cookie) 1965{ 1966 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1967 ALOGV("asyncCallback() event %d", event); 1968 switch (event) { 1969 case STREAM_CBK_EVENT_WRITE_READY: 1970 me->writeCallback(); 1971 break; 1972 case STREAM_CBK_EVENT_DRAIN_READY: 1973 me->drainCallback(); 1974 break; 1975 default: 1976 ALOGW("asyncCallback() unknown event %d", event); 1977 break; 1978 } 1979 return 0; 1980} 1981 1982void AudioFlinger::PlaybackThread::readOutputParameters_l() 1983{ 1984 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1985 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1986 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1987 if (!audio_is_output_channel(mChannelMask)) { 1988 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1989 } 1990 if ((mType == MIXER || mType == DUPLICATING) 1991 && !isValidPcmSinkChannelMask(mChannelMask)) { 1992 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1993 mChannelMask); 1994 } 1995 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1996 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1997 mFormat = mHALFormat; 1998 if (!audio_is_valid_format(mFormat)) { 1999 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2000 } 2001 if ((mType == MIXER || mType == DUPLICATING) 2002 && !isValidPcmSinkFormat(mFormat)) { 2003 LOG_FATAL("HAL format %#x not supported for mixed output", 2004 mFormat); 2005 } 2006 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 2007 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2008 mFrameCount = mBufferSize / mFrameSize; 2009 if (mFrameCount & 15) { 2010 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2011 mFrameCount); 2012 } 2013 2014 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2015 (mOutput->stream->set_callback != NULL)) { 2016 if (mOutput->stream->set_callback(mOutput->stream, 2017 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2018 mUseAsyncWrite = true; 2019 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2020 } 2021 } 2022 2023 mHwSupportsPause = false; 2024 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2025 if (mOutput->stream->pause != NULL) { 2026 if (mOutput->stream->resume != NULL) { 2027 mHwSupportsPause = true; 2028 } else { 2029 ALOGW("direct output implements pause but not resume"); 2030 } 2031 } else if (mOutput->stream->resume != NULL) { 2032 ALOGW("direct output implements resume but not pause"); 2033 } 2034 } 2035 2036 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2037 // For best precision, we use float instead of the associated output 2038 // device format (typically PCM 16 bit). 2039 2040 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2041 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2042 mBufferSize = mFrameSize * mFrameCount; 2043 2044 // TODO: We currently use the associated output device channel mask and sample rate. 2045 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2046 // (if a valid mask) to avoid premature downmix. 2047 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2048 // instead of the output device sample rate to avoid loss of high frequency information. 2049 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2050 } 2051 2052 // Calculate size of normal sink buffer relative to the HAL output buffer size 2053 double multiplier = 1.0; 2054 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2055 kUseFastMixer == FastMixer_Dynamic)) { 2056 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2057 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2058 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2059 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2060 maxNormalFrameCount = maxNormalFrameCount & ~15; 2061 if (maxNormalFrameCount < minNormalFrameCount) { 2062 maxNormalFrameCount = minNormalFrameCount; 2063 } 2064 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2065 if (multiplier <= 1.0) { 2066 multiplier = 1.0; 2067 } else if (multiplier <= 2.0) { 2068 if (2 * mFrameCount <= maxNormalFrameCount) { 2069 multiplier = 2.0; 2070 } else { 2071 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2072 } 2073 } else { 2074 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2075 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2076 // track, but we sometimes have to do this to satisfy the maximum frame count 2077 // constraint) 2078 // FIXME this rounding up should not be done if no HAL SRC 2079 uint32_t truncMult = (uint32_t) multiplier; 2080 if ((truncMult & 1)) { 2081 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2082 ++truncMult; 2083 } 2084 } 2085 multiplier = (double) truncMult; 2086 } 2087 } 2088 mNormalFrameCount = multiplier * mFrameCount; 2089 // round up to nearest 16 frames to satisfy AudioMixer 2090 if (mType == MIXER || mType == DUPLICATING) { 2091 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2092 } 2093 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2094 mNormalFrameCount); 2095 2096 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2097 // Originally this was int16_t[] array, need to remove legacy implications. 2098 free(mSinkBuffer); 2099 mSinkBuffer = NULL; 2100 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2101 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2102 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2103 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2104 2105 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2106 // drives the output. 2107 free(mMixerBuffer); 2108 mMixerBuffer = NULL; 2109 if (mMixerBufferEnabled) { 2110 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2111 mMixerBufferSize = mNormalFrameCount * mChannelCount 2112 * audio_bytes_per_sample(mMixerBufferFormat); 2113 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2114 } 2115 free(mEffectBuffer); 2116 mEffectBuffer = NULL; 2117 if (mEffectBufferEnabled) { 2118 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2119 mEffectBufferSize = mNormalFrameCount * mChannelCount 2120 * audio_bytes_per_sample(mEffectBufferFormat); 2121 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2122 } 2123 2124 // force reconfiguration of effect chains and engines to take new buffer size and audio 2125 // parameters into account 2126 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2127 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2128 // matter. 2129 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2130 Vector< sp<EffectChain> > effectChains = mEffectChains; 2131 for (size_t i = 0; i < effectChains.size(); i ++) { 2132 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2133 } 2134} 2135 2136 2137status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2138{ 2139 if (halFrames == NULL || dspFrames == NULL) { 2140 return BAD_VALUE; 2141 } 2142 Mutex::Autolock _l(mLock); 2143 if (initCheck() != NO_ERROR) { 2144 return INVALID_OPERATION; 2145 } 2146 size_t framesWritten = mBytesWritten / mFrameSize; 2147 *halFrames = framesWritten; 2148 2149 if (isSuspended()) { 2150 // return an estimation of rendered frames when the output is suspended 2151 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2152 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2153 return NO_ERROR; 2154 } else { 2155 status_t status; 2156 uint32_t frames; 2157 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 2158 *dspFrames = (size_t)frames; 2159 return status; 2160 } 2161} 2162 2163uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2164{ 2165 Mutex::Autolock _l(mLock); 2166 uint32_t result = 0; 2167 if (getEffectChain_l(sessionId) != 0) { 2168 result = EFFECT_SESSION; 2169 } 2170 2171 for (size_t i = 0; i < mTracks.size(); ++i) { 2172 sp<Track> track = mTracks[i]; 2173 if (sessionId == track->sessionId() && !track->isInvalid()) { 2174 result |= TRACK_SESSION; 2175 break; 2176 } 2177 } 2178 2179 return result; 2180} 2181 2182uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2183{ 2184 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2185 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2186 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2187 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2188 } 2189 for (size_t i = 0; i < mTracks.size(); i++) { 2190 sp<Track> track = mTracks[i]; 2191 if (sessionId == track->sessionId() && !track->isInvalid()) { 2192 return AudioSystem::getStrategyForStream(track->streamType()); 2193 } 2194 } 2195 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2196} 2197 2198 2199AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2200{ 2201 Mutex::Autolock _l(mLock); 2202 return mOutput; 2203} 2204 2205AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2206{ 2207 Mutex::Autolock _l(mLock); 2208 AudioStreamOut *output = mOutput; 2209 mOutput = NULL; 2210 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2211 // must push a NULL and wait for ack 2212 mOutputSink.clear(); 2213 mPipeSink.clear(); 2214 mNormalSink.clear(); 2215 return output; 2216} 2217 2218// this method must always be called either with ThreadBase mLock held or inside the thread loop 2219audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2220{ 2221 if (mOutput == NULL) { 2222 return NULL; 2223 } 2224 return &mOutput->stream->common; 2225} 2226 2227uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2228{ 2229 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2230} 2231 2232status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2233{ 2234 if (!isValidSyncEvent(event)) { 2235 return BAD_VALUE; 2236 } 2237 2238 Mutex::Autolock _l(mLock); 2239 2240 for (size_t i = 0; i < mTracks.size(); ++i) { 2241 sp<Track> track = mTracks[i]; 2242 if (event->triggerSession() == track->sessionId()) { 2243 (void) track->setSyncEvent(event); 2244 return NO_ERROR; 2245 } 2246 } 2247 2248 return NAME_NOT_FOUND; 2249} 2250 2251bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2252{ 2253 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2254} 2255 2256void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2257 const Vector< sp<Track> >& tracksToRemove) 2258{ 2259 size_t count = tracksToRemove.size(); 2260 if (count > 0) { 2261 for (size_t i = 0 ; i < count ; i++) { 2262 const sp<Track>& track = tracksToRemove.itemAt(i); 2263 if (track->isExternalTrack()) { 2264 AudioSystem::stopOutput(mId, track->streamType(), 2265 (audio_session_t)track->sessionId()); 2266#ifdef ADD_BATTERY_DATA 2267 // to track the speaker usage 2268 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2269#endif 2270 if (track->isTerminated()) { 2271 AudioSystem::releaseOutput(mId, track->streamType(), 2272 (audio_session_t)track->sessionId()); 2273 } 2274 } 2275 } 2276 } 2277} 2278 2279void AudioFlinger::PlaybackThread::checkSilentMode_l() 2280{ 2281 if (!mMasterMute) { 2282 char value[PROPERTY_VALUE_MAX]; 2283 if (property_get("ro.audio.silent", value, "0") > 0) { 2284 char *endptr; 2285 unsigned long ul = strtoul(value, &endptr, 0); 2286 if (*endptr == '\0' && ul != 0) { 2287 ALOGD("Silence is golden"); 2288 // The setprop command will not allow a property to be changed after 2289 // the first time it is set, so we don't have to worry about un-muting. 2290 setMasterMute_l(true); 2291 } 2292 } 2293 } 2294} 2295 2296// shared by MIXER and DIRECT, overridden by DUPLICATING 2297ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2298{ 2299 // FIXME rewrite to reduce number of system calls 2300 mLastWriteTime = systemTime(); 2301 mInWrite = true; 2302 ssize_t bytesWritten; 2303 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2304 2305 // If an NBAIO sink is present, use it to write the normal mixer's submix 2306 if (mNormalSink != 0) { 2307 2308 const size_t count = mBytesRemaining / mFrameSize; 2309 2310 ATRACE_BEGIN("write"); 2311 // update the setpoint when AudioFlinger::mScreenState changes 2312 uint32_t screenState = AudioFlinger::mScreenState; 2313 if (screenState != mScreenState) { 2314 mScreenState = screenState; 2315 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2316 if (pipe != NULL) { 2317 pipe->setAvgFrames((mScreenState & 1) ? 2318 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2319 } 2320 } 2321 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2322 ATRACE_END(); 2323 if (framesWritten > 0) { 2324 bytesWritten = framesWritten * mFrameSize; 2325 } else { 2326 bytesWritten = framesWritten; 2327 } 2328 mLatchDValid = false; 2329 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2330 if (status == NO_ERROR) { 2331 size_t totalFramesWritten = mNormalSink->framesWritten(); 2332 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2333 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2334 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2335 mLatchDValid = true; 2336 } 2337 } 2338 // otherwise use the HAL / AudioStreamOut directly 2339 } else { 2340 // Direct output and offload threads 2341 2342 if (mUseAsyncWrite) { 2343 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2344 mWriteAckSequence += 2; 2345 mWriteAckSequence |= 1; 2346 ALOG_ASSERT(mCallbackThread != 0); 2347 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2348 } 2349 // FIXME We should have an implementation of timestamps for direct output threads. 2350 // They are used e.g for multichannel PCM playback over HDMI. 2351 bytesWritten = mOutput->stream->write(mOutput->stream, 2352 (char *)mSinkBuffer + offset, mBytesRemaining); 2353 if (mUseAsyncWrite && 2354 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2355 // do not wait for async callback in case of error of full write 2356 mWriteAckSequence &= ~1; 2357 ALOG_ASSERT(mCallbackThread != 0); 2358 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2359 } 2360 } 2361 2362 mNumWrites++; 2363 mInWrite = false; 2364 mStandby = false; 2365 return bytesWritten; 2366} 2367 2368void AudioFlinger::PlaybackThread::threadLoop_drain() 2369{ 2370 if (mOutput->stream->drain) { 2371 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2372 if (mUseAsyncWrite) { 2373 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2374 mDrainSequence |= 1; 2375 ALOG_ASSERT(mCallbackThread != 0); 2376 mCallbackThread->setDraining(mDrainSequence); 2377 } 2378 mOutput->stream->drain(mOutput->stream, 2379 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2380 : AUDIO_DRAIN_ALL); 2381 } 2382} 2383 2384void AudioFlinger::PlaybackThread::threadLoop_exit() 2385{ 2386 { 2387 Mutex::Autolock _l(mLock); 2388 for (size_t i = 0; i < mTracks.size(); i++) { 2389 sp<Track> track = mTracks[i]; 2390 track->invalidate(); 2391 } 2392 } 2393} 2394 2395/* 2396The derived values that are cached: 2397 - mSinkBufferSize from frame count * frame size 2398 - activeSleepTime from activeSleepTimeUs() 2399 - idleSleepTime from idleSleepTimeUs() 2400 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2401 - maxPeriod from frame count and sample rate (MIXER only) 2402 2403The parameters that affect these derived values are: 2404 - frame count 2405 - frame size 2406 - sample rate 2407 - device type: A2DP or not 2408 - device latency 2409 - format: PCM or not 2410 - active sleep time 2411 - idle sleep time 2412*/ 2413 2414void AudioFlinger::PlaybackThread::cacheParameters_l() 2415{ 2416 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2417 activeSleepTime = activeSleepTimeUs(); 2418 idleSleepTime = idleSleepTimeUs(); 2419} 2420 2421void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2422{ 2423 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2424 this, streamType, mTracks.size()); 2425 Mutex::Autolock _l(mLock); 2426 2427 size_t size = mTracks.size(); 2428 for (size_t i = 0; i < size; i++) { 2429 sp<Track> t = mTracks[i]; 2430 if (t->streamType() == streamType) { 2431 t->invalidate(); 2432 } 2433 } 2434} 2435 2436status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2437{ 2438 int session = chain->sessionId(); 2439 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2440 ? mEffectBuffer : mSinkBuffer); 2441 bool ownsBuffer = false; 2442 2443 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2444 if (session > 0) { 2445 // Only one effect chain can be present in direct output thread and it uses 2446 // the sink buffer as input 2447 if (mType != DIRECT) { 2448 size_t numSamples = mNormalFrameCount * mChannelCount; 2449 buffer = new int16_t[numSamples]; 2450 memset(buffer, 0, numSamples * sizeof(int16_t)); 2451 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2452 ownsBuffer = true; 2453 } 2454 2455 // Attach all tracks with same session ID to this chain. 2456 for (size_t i = 0; i < mTracks.size(); ++i) { 2457 sp<Track> track = mTracks[i]; 2458 if (session == track->sessionId()) { 2459 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2460 buffer); 2461 track->setMainBuffer(buffer); 2462 chain->incTrackCnt(); 2463 } 2464 } 2465 2466 // indicate all active tracks in the chain 2467 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2468 sp<Track> track = mActiveTracks[i].promote(); 2469 if (track == 0) { 2470 continue; 2471 } 2472 if (session == track->sessionId()) { 2473 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2474 chain->incActiveTrackCnt(); 2475 } 2476 } 2477 } 2478 chain->setThread(this); 2479 chain->setInBuffer(buffer, ownsBuffer); 2480 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2481 ? mEffectBuffer : mSinkBuffer)); 2482 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2483 // chains list in order to be processed last as it contains output stage effects 2484 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2485 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2486 // after track specific effects and before output stage 2487 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2488 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2489 // Effect chain for other sessions are inserted at beginning of effect 2490 // chains list to be processed before output mix effects. Relative order between other 2491 // sessions is not important 2492 size_t size = mEffectChains.size(); 2493 size_t i = 0; 2494 for (i = 0; i < size; i++) { 2495 if (mEffectChains[i]->sessionId() < session) { 2496 break; 2497 } 2498 } 2499 mEffectChains.insertAt(chain, i); 2500 checkSuspendOnAddEffectChain_l(chain); 2501 2502 return NO_ERROR; 2503} 2504 2505size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2506{ 2507 int session = chain->sessionId(); 2508 2509 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2510 2511 for (size_t i = 0; i < mEffectChains.size(); i++) { 2512 if (chain == mEffectChains[i]) { 2513 mEffectChains.removeAt(i); 2514 // detach all active tracks from the chain 2515 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2516 sp<Track> track = mActiveTracks[i].promote(); 2517 if (track == 0) { 2518 continue; 2519 } 2520 if (session == track->sessionId()) { 2521 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2522 chain.get(), session); 2523 chain->decActiveTrackCnt(); 2524 } 2525 } 2526 2527 // detach all tracks with same session ID from this chain 2528 for (size_t i = 0; i < mTracks.size(); ++i) { 2529 sp<Track> track = mTracks[i]; 2530 if (session == track->sessionId()) { 2531 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2532 chain->decTrackCnt(); 2533 } 2534 } 2535 break; 2536 } 2537 } 2538 return mEffectChains.size(); 2539} 2540 2541status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2542 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2543{ 2544 Mutex::Autolock _l(mLock); 2545 return attachAuxEffect_l(track, EffectId); 2546} 2547 2548status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2549 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2550{ 2551 status_t status = NO_ERROR; 2552 2553 if (EffectId == 0) { 2554 track->setAuxBuffer(0, NULL); 2555 } else { 2556 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2557 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2558 if (effect != 0) { 2559 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2560 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2561 } else { 2562 status = INVALID_OPERATION; 2563 } 2564 } else { 2565 status = BAD_VALUE; 2566 } 2567 } 2568 return status; 2569} 2570 2571void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2572{ 2573 for (size_t i = 0; i < mTracks.size(); ++i) { 2574 sp<Track> track = mTracks[i]; 2575 if (track->auxEffectId() == effectId) { 2576 attachAuxEffect_l(track, 0); 2577 } 2578 } 2579} 2580 2581bool AudioFlinger::PlaybackThread::threadLoop() 2582{ 2583 Vector< sp<Track> > tracksToRemove; 2584 2585 standbyTime = systemTime(); 2586 2587 // MIXER 2588 nsecs_t lastWarning = 0; 2589 2590 // DUPLICATING 2591 // FIXME could this be made local to while loop? 2592 writeFrames = 0; 2593 2594 int lastGeneration = 0; 2595 2596 cacheParameters_l(); 2597 sleepTime = idleSleepTime; 2598 2599 if (mType == MIXER) { 2600 sleepTimeShift = 0; 2601 } 2602 2603 CpuStats cpuStats; 2604 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2605 2606 acquireWakeLock(); 2607 2608 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2609 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2610 // and then that string will be logged at the next convenient opportunity. 2611 const char *logString = NULL; 2612 2613 checkSilentMode_l(); 2614 2615 while (!exitPending()) 2616 { 2617 cpuStats.sample(myName); 2618 2619 Vector< sp<EffectChain> > effectChains; 2620 2621 { // scope for mLock 2622 2623 Mutex::Autolock _l(mLock); 2624 2625 processConfigEvents_l(); 2626 2627 if (logString != NULL) { 2628 mNBLogWriter->logTimestamp(); 2629 mNBLogWriter->log(logString); 2630 logString = NULL; 2631 } 2632 2633 // Gather the framesReleased counters for all active tracks, 2634 // and latch them atomically with the timestamp. 2635 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2636 mLatchD.mFramesReleased.clear(); 2637 size_t size = mActiveTracks.size(); 2638 for (size_t i = 0; i < size; i++) { 2639 sp<Track> t = mActiveTracks[i].promote(); 2640 if (t != 0) { 2641 mLatchD.mFramesReleased.add(t.get(), 2642 t->mAudioTrackServerProxy->framesReleased()); 2643 } 2644 } 2645 if (mLatchDValid) { 2646 mLatchQ = mLatchD; 2647 mLatchDValid = false; 2648 mLatchQValid = true; 2649 } 2650 2651 saveOutputTracks(); 2652 if (mSignalPending) { 2653 // A signal was raised while we were unlocked 2654 mSignalPending = false; 2655 } else if (waitingAsyncCallback_l()) { 2656 if (exitPending()) { 2657 break; 2658 } 2659 releaseWakeLock_l(); 2660 mWakeLockUids.clear(); 2661 mActiveTracksGeneration++; 2662 ALOGV("wait async completion"); 2663 mWaitWorkCV.wait(mLock); 2664 ALOGV("async completion/wake"); 2665 acquireWakeLock_l(); 2666 standbyTime = systemTime() + standbyDelay; 2667 sleepTime = 0; 2668 2669 continue; 2670 } 2671 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2672 isSuspended()) { 2673 // put audio hardware into standby after short delay 2674 if (shouldStandby_l()) { 2675 2676 threadLoop_standby(); 2677 2678 mStandby = true; 2679 } 2680 2681 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2682 // we're about to wait, flush the binder command buffer 2683 IPCThreadState::self()->flushCommands(); 2684 2685 clearOutputTracks(); 2686 2687 if (exitPending()) { 2688 break; 2689 } 2690 2691 releaseWakeLock_l(); 2692 mWakeLockUids.clear(); 2693 mActiveTracksGeneration++; 2694 // wait until we have something to do... 2695 ALOGV("%s going to sleep", myName.string()); 2696 mWaitWorkCV.wait(mLock); 2697 ALOGV("%s waking up", myName.string()); 2698 acquireWakeLock_l(); 2699 2700 mMixerStatus = MIXER_IDLE; 2701 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2702 mBytesWritten = 0; 2703 mBytesRemaining = 0; 2704 checkSilentMode_l(); 2705 2706 standbyTime = systemTime() + standbyDelay; 2707 sleepTime = idleSleepTime; 2708 if (mType == MIXER) { 2709 sleepTimeShift = 0; 2710 } 2711 2712 continue; 2713 } 2714 } 2715 // mMixerStatusIgnoringFastTracks is also updated internally 2716 mMixerStatus = prepareTracks_l(&tracksToRemove); 2717 2718 // compare with previously applied list 2719 if (lastGeneration != mActiveTracksGeneration) { 2720 // update wakelock 2721 updateWakeLockUids_l(mWakeLockUids); 2722 lastGeneration = mActiveTracksGeneration; 2723 } 2724 2725 // prevent any changes in effect chain list and in each effect chain 2726 // during mixing and effect process as the audio buffers could be deleted 2727 // or modified if an effect is created or deleted 2728 lockEffectChains_l(effectChains); 2729 } // mLock scope ends 2730 2731 if (mBytesRemaining == 0) { 2732 mCurrentWriteLength = 0; 2733 if (mMixerStatus == MIXER_TRACKS_READY) { 2734 // threadLoop_mix() sets mCurrentWriteLength 2735 threadLoop_mix(); 2736 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2737 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2738 // threadLoop_sleepTime sets sleepTime to 0 if data 2739 // must be written to HAL 2740 threadLoop_sleepTime(); 2741 if (sleepTime == 0) { 2742 mCurrentWriteLength = mSinkBufferSize; 2743 } 2744 } 2745 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2746 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2747 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2748 // or mSinkBuffer (if there are no effects). 2749 // 2750 // This is done pre-effects computation; if effects change to 2751 // support higher precision, this needs to move. 2752 // 2753 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2754 // TODO use sleepTime == 0 as an additional condition. 2755 if (mMixerBufferValid) { 2756 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2757 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2758 2759 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2760 mNormalFrameCount * mChannelCount); 2761 } 2762 2763 mBytesRemaining = mCurrentWriteLength; 2764 if (isSuspended()) { 2765 sleepTime = suspendSleepTimeUs(); 2766 // simulate write to HAL when suspended 2767 mBytesWritten += mSinkBufferSize; 2768 mBytesRemaining = 0; 2769 } 2770 2771 // only process effects if we're going to write 2772 if (sleepTime == 0 && mType != OFFLOAD) { 2773 for (size_t i = 0; i < effectChains.size(); i ++) { 2774 effectChains[i]->process_l(); 2775 } 2776 } 2777 } 2778 // Process effect chains for offloaded thread even if no audio 2779 // was read from audio track: process only updates effect state 2780 // and thus does have to be synchronized with audio writes but may have 2781 // to be called while waiting for async write callback 2782 if (mType == OFFLOAD) { 2783 for (size_t i = 0; i < effectChains.size(); i ++) { 2784 effectChains[i]->process_l(); 2785 } 2786 } 2787 2788 // Only if the Effects buffer is enabled and there is data in the 2789 // Effects buffer (buffer valid), we need to 2790 // copy into the sink buffer. 2791 // TODO use sleepTime == 0 as an additional condition. 2792 if (mEffectBufferValid) { 2793 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2794 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2795 mNormalFrameCount * mChannelCount); 2796 } 2797 2798 // enable changes in effect chain 2799 unlockEffectChains(effectChains); 2800 2801 if (!waitingAsyncCallback()) { 2802 // sleepTime == 0 means we must write to audio hardware 2803 if (sleepTime == 0) { 2804 if (mBytesRemaining) { 2805 ssize_t ret = threadLoop_write(); 2806 if (ret < 0) { 2807 mBytesRemaining = 0; 2808 } else { 2809 mBytesWritten += ret; 2810 mBytesRemaining -= ret; 2811 } 2812 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2813 (mMixerStatus == MIXER_DRAIN_ALL)) { 2814 threadLoop_drain(); 2815 } 2816 if (mType == MIXER) { 2817 // write blocked detection 2818 nsecs_t now = systemTime(); 2819 nsecs_t delta = now - mLastWriteTime; 2820 if (!mStandby && delta > maxPeriod) { 2821 mNumDelayedWrites++; 2822 if ((now - lastWarning) > kWarningThrottleNs) { 2823 ATRACE_NAME("underrun"); 2824 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2825 ns2ms(delta), mNumDelayedWrites, this); 2826 lastWarning = now; 2827 } 2828 } 2829 } 2830 2831 } else { 2832 ATRACE_BEGIN("sleep"); 2833 usleep(sleepTime); 2834 ATRACE_END(); 2835 } 2836 } 2837 2838 // Finally let go of removed track(s), without the lock held 2839 // since we can't guarantee the destructors won't acquire that 2840 // same lock. This will also mutate and push a new fast mixer state. 2841 threadLoop_removeTracks(tracksToRemove); 2842 tracksToRemove.clear(); 2843 2844 // FIXME I don't understand the need for this here; 2845 // it was in the original code but maybe the 2846 // assignment in saveOutputTracks() makes this unnecessary? 2847 clearOutputTracks(); 2848 2849 // Effect chains will be actually deleted here if they were removed from 2850 // mEffectChains list during mixing or effects processing 2851 effectChains.clear(); 2852 2853 // FIXME Note that the above .clear() is no longer necessary since effectChains 2854 // is now local to this block, but will keep it for now (at least until merge done). 2855 } 2856 2857 threadLoop_exit(); 2858 2859 if (!mStandby) { 2860 threadLoop_standby(); 2861 mStandby = true; 2862 } 2863 2864 releaseWakeLock(); 2865 mWakeLockUids.clear(); 2866 mActiveTracksGeneration++; 2867 2868 ALOGV("Thread %p type %d exiting", this, mType); 2869 return false; 2870} 2871 2872// removeTracks_l() must be called with ThreadBase::mLock held 2873void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2874{ 2875 size_t count = tracksToRemove.size(); 2876 if (count > 0) { 2877 for (size_t i=0 ; i<count ; i++) { 2878 const sp<Track>& track = tracksToRemove.itemAt(i); 2879 mActiveTracks.remove(track); 2880 mWakeLockUids.remove(track->uid()); 2881 mActiveTracksGeneration++; 2882 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2883 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2884 if (chain != 0) { 2885 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2886 track->sessionId()); 2887 chain->decActiveTrackCnt(); 2888 } 2889 if (track->isTerminated()) { 2890 removeTrack_l(track); 2891 } 2892 } 2893 } 2894 2895} 2896 2897status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2898{ 2899 if (mNormalSink != 0) { 2900 return mNormalSink->getTimestamp(timestamp); 2901 } 2902 if ((mType == OFFLOAD || mType == DIRECT) 2903 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2904 uint64_t position64; 2905 int ret = mOutput->stream->get_presentation_position( 2906 mOutput->stream, &position64, ×tamp.mTime); 2907 if (ret == 0) { 2908 timestamp.mPosition = (uint32_t)position64; 2909 return NO_ERROR; 2910 } 2911 } 2912 return INVALID_OPERATION; 2913} 2914 2915status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2916 audio_patch_handle_t *handle) 2917{ 2918 status_t status = NO_ERROR; 2919 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2920 // store new device and send to effects 2921 audio_devices_t type = AUDIO_DEVICE_NONE; 2922 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2923 type |= patch->sinks[i].ext.device.type; 2924 } 2925 mOutDevice = type; 2926 for (size_t i = 0; i < mEffectChains.size(); i++) { 2927 mEffectChains[i]->setDevice_l(mOutDevice); 2928 } 2929 2930 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2931 status = hwDevice->create_audio_patch(hwDevice, 2932 patch->num_sources, 2933 patch->sources, 2934 patch->num_sinks, 2935 patch->sinks, 2936 handle); 2937 } else { 2938 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2939 } 2940 return status; 2941} 2942 2943status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2944{ 2945 status_t status = NO_ERROR; 2946 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2947 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2948 status = hwDevice->release_audio_patch(hwDevice, handle); 2949 } else { 2950 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2951 } 2952 return status; 2953} 2954 2955void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2956{ 2957 Mutex::Autolock _l(mLock); 2958 mTracks.add(track); 2959} 2960 2961void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2962{ 2963 Mutex::Autolock _l(mLock); 2964 destroyTrack_l(track); 2965} 2966 2967void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2968{ 2969 ThreadBase::getAudioPortConfig(config); 2970 config->role = AUDIO_PORT_ROLE_SOURCE; 2971 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2972 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2973} 2974 2975// ---------------------------------------------------------------------------- 2976 2977AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2978 audio_io_handle_t id, audio_devices_t device, type_t type) 2979 : PlaybackThread(audioFlinger, output, id, device, type), 2980 // mAudioMixer below 2981 // mFastMixer below 2982 mFastMixerFutex(0) 2983 // mOutputSink below 2984 // mPipeSink below 2985 // mNormalSink below 2986{ 2987 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2988 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2989 "mFrameCount=%d, mNormalFrameCount=%d", 2990 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2991 mNormalFrameCount); 2992 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2993 2994 if (type == DUPLICATING) { 2995 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 2996 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 2997 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 2998 return; 2999 } 3000 // create an NBAIO sink for the HAL output stream, and negotiate 3001 mOutputSink = new AudioStreamOutSink(output->stream); 3002 size_t numCounterOffers = 0; 3003 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3004 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3005 ALOG_ASSERT(index == 0); 3006 3007 // initialize fast mixer depending on configuration 3008 bool initFastMixer; 3009 switch (kUseFastMixer) { 3010 case FastMixer_Never: 3011 initFastMixer = false; 3012 break; 3013 case FastMixer_Always: 3014 initFastMixer = true; 3015 break; 3016 case FastMixer_Static: 3017 case FastMixer_Dynamic: 3018 initFastMixer = mFrameCount < mNormalFrameCount; 3019 break; 3020 } 3021 if (initFastMixer) { 3022 audio_format_t fastMixerFormat; 3023 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3024 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3025 } else { 3026 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3027 } 3028 if (mFormat != fastMixerFormat) { 3029 // change our Sink format to accept our intermediate precision 3030 mFormat = fastMixerFormat; 3031 free(mSinkBuffer); 3032 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3033 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3034 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3035 } 3036 3037 // create a MonoPipe to connect our submix to FastMixer 3038 NBAIO_Format format = mOutputSink->format(); 3039 NBAIO_Format origformat = format; 3040 // adjust format to match that of the Fast Mixer 3041 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3042 format.mFormat = fastMixerFormat; 3043 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3044 3045 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3046 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3047 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3048 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3049 const NBAIO_Format offers[1] = {format}; 3050 size_t numCounterOffers = 0; 3051 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3052 ALOG_ASSERT(index == 0); 3053 monoPipe->setAvgFrames((mScreenState & 1) ? 3054 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3055 mPipeSink = monoPipe; 3056 3057#ifdef TEE_SINK 3058 if (mTeeSinkOutputEnabled) { 3059 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3060 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3061 const NBAIO_Format offers2[1] = {origformat}; 3062 numCounterOffers = 0; 3063 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3064 ALOG_ASSERT(index == 0); 3065 mTeeSink = teeSink; 3066 PipeReader *teeSource = new PipeReader(*teeSink); 3067 numCounterOffers = 0; 3068 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3069 ALOG_ASSERT(index == 0); 3070 mTeeSource = teeSource; 3071 } 3072#endif 3073 3074 // create fast mixer and configure it initially with just one fast track for our submix 3075 mFastMixer = new FastMixer(); 3076 FastMixerStateQueue *sq = mFastMixer->sq(); 3077#ifdef STATE_QUEUE_DUMP 3078 sq->setObserverDump(&mStateQueueObserverDump); 3079 sq->setMutatorDump(&mStateQueueMutatorDump); 3080#endif 3081 FastMixerState *state = sq->begin(); 3082 FastTrack *fastTrack = &state->mFastTracks[0]; 3083 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3084 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3085 fastTrack->mVolumeProvider = NULL; 3086 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3087 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3088 fastTrack->mGeneration++; 3089 state->mFastTracksGen++; 3090 state->mTrackMask = 1; 3091 // fast mixer will use the HAL output sink 3092 state->mOutputSink = mOutputSink.get(); 3093 state->mOutputSinkGen++; 3094 state->mFrameCount = mFrameCount; 3095 state->mCommand = FastMixerState::COLD_IDLE; 3096 // already done in constructor initialization list 3097 //mFastMixerFutex = 0; 3098 state->mColdFutexAddr = &mFastMixerFutex; 3099 state->mColdGen++; 3100 state->mDumpState = &mFastMixerDumpState; 3101#ifdef TEE_SINK 3102 state->mTeeSink = mTeeSink.get(); 3103#endif 3104 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3105 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3106 sq->end(); 3107 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3108 3109 // start the fast mixer 3110 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3111 pid_t tid = mFastMixer->getTid(); 3112 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3113 if (err != 0) { 3114 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3115 kPriorityFastMixer, getpid_cached, tid, err); 3116 } 3117 3118#ifdef AUDIO_WATCHDOG 3119 // create and start the watchdog 3120 mAudioWatchdog = new AudioWatchdog(); 3121 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3122 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3123 tid = mAudioWatchdog->getTid(); 3124 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3125 if (err != 0) { 3126 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3127 kPriorityFastMixer, getpid_cached, tid, err); 3128 } 3129#endif 3130 3131 } 3132 3133 switch (kUseFastMixer) { 3134 case FastMixer_Never: 3135 case FastMixer_Dynamic: 3136 mNormalSink = mOutputSink; 3137 break; 3138 case FastMixer_Always: 3139 mNormalSink = mPipeSink; 3140 break; 3141 case FastMixer_Static: 3142 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3143 break; 3144 } 3145} 3146 3147AudioFlinger::MixerThread::~MixerThread() 3148{ 3149 if (mFastMixer != 0) { 3150 FastMixerStateQueue *sq = mFastMixer->sq(); 3151 FastMixerState *state = sq->begin(); 3152 if (state->mCommand == FastMixerState::COLD_IDLE) { 3153 int32_t old = android_atomic_inc(&mFastMixerFutex); 3154 if (old == -1) { 3155 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3156 } 3157 } 3158 state->mCommand = FastMixerState::EXIT; 3159 sq->end(); 3160 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3161 mFastMixer->join(); 3162 // Though the fast mixer thread has exited, it's state queue is still valid. 3163 // We'll use that extract the final state which contains one remaining fast track 3164 // corresponding to our sub-mix. 3165 state = sq->begin(); 3166 ALOG_ASSERT(state->mTrackMask == 1); 3167 FastTrack *fastTrack = &state->mFastTracks[0]; 3168 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3169 delete fastTrack->mBufferProvider; 3170 sq->end(false /*didModify*/); 3171 mFastMixer.clear(); 3172#ifdef AUDIO_WATCHDOG 3173 if (mAudioWatchdog != 0) { 3174 mAudioWatchdog->requestExit(); 3175 mAudioWatchdog->requestExitAndWait(); 3176 mAudioWatchdog.clear(); 3177 } 3178#endif 3179 } 3180 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3181 delete mAudioMixer; 3182} 3183 3184 3185uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3186{ 3187 if (mFastMixer != 0) { 3188 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3189 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3190 } 3191 return latency; 3192} 3193 3194 3195void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3196{ 3197 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3198} 3199 3200ssize_t AudioFlinger::MixerThread::threadLoop_write() 3201{ 3202 // FIXME we should only do one push per cycle; confirm this is true 3203 // Start the fast mixer if it's not already running 3204 if (mFastMixer != 0) { 3205 FastMixerStateQueue *sq = mFastMixer->sq(); 3206 FastMixerState *state = sq->begin(); 3207 if (state->mCommand != FastMixerState::MIX_WRITE && 3208 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3209 if (state->mCommand == FastMixerState::COLD_IDLE) { 3210 int32_t old = android_atomic_inc(&mFastMixerFutex); 3211 if (old == -1) { 3212 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3213 } 3214#ifdef AUDIO_WATCHDOG 3215 if (mAudioWatchdog != 0) { 3216 mAudioWatchdog->resume(); 3217 } 3218#endif 3219 } 3220 state->mCommand = FastMixerState::MIX_WRITE; 3221#ifdef FAST_THREAD_STATISTICS 3222 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3223 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3224#endif 3225 sq->end(); 3226 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3227 if (kUseFastMixer == FastMixer_Dynamic) { 3228 mNormalSink = mPipeSink; 3229 } 3230 } else { 3231 sq->end(false /*didModify*/); 3232 } 3233 } 3234 return PlaybackThread::threadLoop_write(); 3235} 3236 3237void AudioFlinger::MixerThread::threadLoop_standby() 3238{ 3239 // Idle the fast mixer if it's currently running 3240 if (mFastMixer != 0) { 3241 FastMixerStateQueue *sq = mFastMixer->sq(); 3242 FastMixerState *state = sq->begin(); 3243 if (!(state->mCommand & FastMixerState::IDLE)) { 3244 state->mCommand = FastMixerState::COLD_IDLE; 3245 state->mColdFutexAddr = &mFastMixerFutex; 3246 state->mColdGen++; 3247 mFastMixerFutex = 0; 3248 sq->end(); 3249 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3250 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3251 if (kUseFastMixer == FastMixer_Dynamic) { 3252 mNormalSink = mOutputSink; 3253 } 3254#ifdef AUDIO_WATCHDOG 3255 if (mAudioWatchdog != 0) { 3256 mAudioWatchdog->pause(); 3257 } 3258#endif 3259 } else { 3260 sq->end(false /*didModify*/); 3261 } 3262 } 3263 PlaybackThread::threadLoop_standby(); 3264} 3265 3266bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3267{ 3268 return false; 3269} 3270 3271bool AudioFlinger::PlaybackThread::shouldStandby_l() 3272{ 3273 return !mStandby; 3274} 3275 3276bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3277{ 3278 Mutex::Autolock _l(mLock); 3279 return waitingAsyncCallback_l(); 3280} 3281 3282// shared by MIXER and DIRECT, overridden by DUPLICATING 3283void AudioFlinger::PlaybackThread::threadLoop_standby() 3284{ 3285 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3286 mOutput->stream->common.standby(&mOutput->stream->common); 3287 if (mUseAsyncWrite != 0) { 3288 // discard any pending drain or write ack by incrementing sequence 3289 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3290 mDrainSequence = (mDrainSequence + 2) & ~1; 3291 ALOG_ASSERT(mCallbackThread != 0); 3292 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3293 mCallbackThread->setDraining(mDrainSequence); 3294 } 3295 mHwPaused = false; 3296} 3297 3298void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3299{ 3300 ALOGV("signal playback thread"); 3301 broadcast_l(); 3302} 3303 3304void AudioFlinger::MixerThread::threadLoop_mix() 3305{ 3306 // obtain the presentation timestamp of the next output buffer 3307 int64_t pts; 3308 status_t status = INVALID_OPERATION; 3309 3310 if (mNormalSink != 0) { 3311 status = mNormalSink->getNextWriteTimestamp(&pts); 3312 } else { 3313 status = mOutputSink->getNextWriteTimestamp(&pts); 3314 } 3315 3316 if (status != NO_ERROR) { 3317 pts = AudioBufferProvider::kInvalidPTS; 3318 } 3319 3320 // mix buffers... 3321 mAudioMixer->process(pts); 3322 mCurrentWriteLength = mSinkBufferSize; 3323 // increase sleep time progressively when application underrun condition clears. 3324 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3325 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3326 // such that we would underrun the audio HAL. 3327 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3328 sleepTimeShift--; 3329 } 3330 sleepTime = 0; 3331 standbyTime = systemTime() + standbyDelay; 3332 //TODO: delay standby when effects have a tail 3333 3334} 3335 3336void AudioFlinger::MixerThread::threadLoop_sleepTime() 3337{ 3338 // If no tracks are ready, sleep once for the duration of an output 3339 // buffer size, then write 0s to the output 3340 if (sleepTime == 0) { 3341 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3342 sleepTime = activeSleepTime >> sleepTimeShift; 3343 if (sleepTime < kMinThreadSleepTimeUs) { 3344 sleepTime = kMinThreadSleepTimeUs; 3345 } 3346 // reduce sleep time in case of consecutive application underruns to avoid 3347 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3348 // duration we would end up writing less data than needed by the audio HAL if 3349 // the condition persists. 3350 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3351 sleepTimeShift++; 3352 } 3353 } else { 3354 sleepTime = idleSleepTime; 3355 } 3356 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3357 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3358 // before effects processing or output. 3359 if (mMixerBufferValid) { 3360 memset(mMixerBuffer, 0, mMixerBufferSize); 3361 } else { 3362 memset(mSinkBuffer, 0, mSinkBufferSize); 3363 } 3364 sleepTime = 0; 3365 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3366 "anticipated start"); 3367 } 3368 // TODO add standby time extension fct of effect tail 3369} 3370 3371// prepareTracks_l() must be called with ThreadBase::mLock held 3372AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3373 Vector< sp<Track> > *tracksToRemove) 3374{ 3375 3376 mixer_state mixerStatus = MIXER_IDLE; 3377 // find out which tracks need to be processed 3378 size_t count = mActiveTracks.size(); 3379 size_t mixedTracks = 0; 3380 size_t tracksWithEffect = 0; 3381 // counts only _active_ fast tracks 3382 size_t fastTracks = 0; 3383 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3384 3385 float masterVolume = mMasterVolume; 3386 bool masterMute = mMasterMute; 3387 3388 if (masterMute) { 3389 masterVolume = 0; 3390 } 3391 // Delegate master volume control to effect in output mix effect chain if needed 3392 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3393 if (chain != 0) { 3394 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3395 chain->setVolume_l(&v, &v); 3396 masterVolume = (float)((v + (1 << 23)) >> 24); 3397 chain.clear(); 3398 } 3399 3400 // prepare a new state to push 3401 FastMixerStateQueue *sq = NULL; 3402 FastMixerState *state = NULL; 3403 bool didModify = false; 3404 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3405 if (mFastMixer != 0) { 3406 sq = mFastMixer->sq(); 3407 state = sq->begin(); 3408 } 3409 3410 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3411 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3412 3413 for (size_t i=0 ; i<count ; i++) { 3414 const sp<Track> t = mActiveTracks[i].promote(); 3415 if (t == 0) { 3416 continue; 3417 } 3418 3419 // this const just means the local variable doesn't change 3420 Track* const track = t.get(); 3421 3422 // process fast tracks 3423 if (track->isFastTrack()) { 3424 3425 // It's theoretically possible (though unlikely) for a fast track to be created 3426 // and then removed within the same normal mix cycle. This is not a problem, as 3427 // the track never becomes active so it's fast mixer slot is never touched. 3428 // The converse, of removing an (active) track and then creating a new track 3429 // at the identical fast mixer slot within the same normal mix cycle, 3430 // is impossible because the slot isn't marked available until the end of each cycle. 3431 int j = track->mFastIndex; 3432 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3433 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3434 FastTrack *fastTrack = &state->mFastTracks[j]; 3435 3436 // Determine whether the track is currently in underrun condition, 3437 // and whether it had a recent underrun. 3438 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3439 FastTrackUnderruns underruns = ftDump->mUnderruns; 3440 uint32_t recentFull = (underruns.mBitFields.mFull - 3441 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3442 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3443 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3444 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3445 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3446 uint32_t recentUnderruns = recentPartial + recentEmpty; 3447 track->mObservedUnderruns = underruns; 3448 // don't count underruns that occur while stopping or pausing 3449 // or stopped which can occur when flush() is called while active 3450 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3451 recentUnderruns > 0) { 3452 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3453 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3454 } 3455 3456 // This is similar to the state machine for normal tracks, 3457 // with a few modifications for fast tracks. 3458 bool isActive = true; 3459 switch (track->mState) { 3460 case TrackBase::STOPPING_1: 3461 // track stays active in STOPPING_1 state until first underrun 3462 if (recentUnderruns > 0 || track->isTerminated()) { 3463 track->mState = TrackBase::STOPPING_2; 3464 } 3465 break; 3466 case TrackBase::PAUSING: 3467 // ramp down is not yet implemented 3468 track->setPaused(); 3469 break; 3470 case TrackBase::RESUMING: 3471 // ramp up is not yet implemented 3472 track->mState = TrackBase::ACTIVE; 3473 break; 3474 case TrackBase::ACTIVE: 3475 if (recentFull > 0 || recentPartial > 0) { 3476 // track has provided at least some frames recently: reset retry count 3477 track->mRetryCount = kMaxTrackRetries; 3478 } 3479 if (recentUnderruns == 0) { 3480 // no recent underruns: stay active 3481 break; 3482 } 3483 // there has recently been an underrun of some kind 3484 if (track->sharedBuffer() == 0) { 3485 // were any of the recent underruns "empty" (no frames available)? 3486 if (recentEmpty == 0) { 3487 // no, then ignore the partial underruns as they are allowed indefinitely 3488 break; 3489 } 3490 // there has recently been an "empty" underrun: decrement the retry counter 3491 if (--(track->mRetryCount) > 0) { 3492 break; 3493 } 3494 // indicate to client process that the track was disabled because of underrun; 3495 // it will then automatically call start() when data is available 3496 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3497 // remove from active list, but state remains ACTIVE [confusing but true] 3498 isActive = false; 3499 break; 3500 } 3501 // fall through 3502 case TrackBase::STOPPING_2: 3503 case TrackBase::PAUSED: 3504 case TrackBase::STOPPED: 3505 case TrackBase::FLUSHED: // flush() while active 3506 // Check for presentation complete if track is inactive 3507 // We have consumed all the buffers of this track. 3508 // This would be incomplete if we auto-paused on underrun 3509 { 3510 size_t audioHALFrames = 3511 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3512 size_t framesWritten = mBytesWritten / mFrameSize; 3513 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3514 // track stays in active list until presentation is complete 3515 break; 3516 } 3517 } 3518 if (track->isStopping_2()) { 3519 track->mState = TrackBase::STOPPED; 3520 } 3521 if (track->isStopped()) { 3522 // Can't reset directly, as fast mixer is still polling this track 3523 // track->reset(); 3524 // So instead mark this track as needing to be reset after push with ack 3525 resetMask |= 1 << i; 3526 } 3527 isActive = false; 3528 break; 3529 case TrackBase::IDLE: 3530 default: 3531 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3532 } 3533 3534 if (isActive) { 3535 // was it previously inactive? 3536 if (!(state->mTrackMask & (1 << j))) { 3537 ExtendedAudioBufferProvider *eabp = track; 3538 VolumeProvider *vp = track; 3539 fastTrack->mBufferProvider = eabp; 3540 fastTrack->mVolumeProvider = vp; 3541 fastTrack->mChannelMask = track->mChannelMask; 3542 fastTrack->mFormat = track->mFormat; 3543 fastTrack->mGeneration++; 3544 state->mTrackMask |= 1 << j; 3545 didModify = true; 3546 // no acknowledgement required for newly active tracks 3547 } 3548 // cache the combined master volume and stream type volume for fast mixer; this 3549 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3550 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3551 ++fastTracks; 3552 } else { 3553 // was it previously active? 3554 if (state->mTrackMask & (1 << j)) { 3555 fastTrack->mBufferProvider = NULL; 3556 fastTrack->mGeneration++; 3557 state->mTrackMask &= ~(1 << j); 3558 didModify = true; 3559 // If any fast tracks were removed, we must wait for acknowledgement 3560 // because we're about to decrement the last sp<> on those tracks. 3561 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3562 } else { 3563 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3564 } 3565 tracksToRemove->add(track); 3566 // Avoids a misleading display in dumpsys 3567 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3568 } 3569 continue; 3570 } 3571 3572 { // local variable scope to avoid goto warning 3573 3574 audio_track_cblk_t* cblk = track->cblk(); 3575 3576 // The first time a track is added we wait 3577 // for all its buffers to be filled before processing it 3578 int name = track->name(); 3579 // make sure that we have enough frames to mix one full buffer. 3580 // enforce this condition only once to enable draining the buffer in case the client 3581 // app does not call stop() and relies on underrun to stop: 3582 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3583 // during last round 3584 size_t desiredFrames; 3585 uint32_t sr = track->sampleRate(); 3586 if (sr == mSampleRate) { 3587 desiredFrames = mNormalFrameCount; 3588 } else { 3589 desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); 3590 // add frames already consumed but not yet released by the resampler 3591 // because mAudioTrackServerProxy->framesReady() will include these frames 3592 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3593#if 0 3594 // the minimum track buffer size is normally twice the number of frames necessary 3595 // to fill one buffer and the resampler should not leave more than one buffer worth 3596 // of unreleased frames after each pass, but just in case... 3597 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3598#endif 3599 } 3600 uint32_t minFrames = 1; 3601 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3602 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3603 minFrames = desiredFrames; 3604 } 3605 3606 size_t framesReady = track->framesReady(); 3607 if (ATRACE_ENABLED()) { 3608 // I wish we had formatted trace names 3609 char traceName[16]; 3610 strcpy(traceName, "nRdy"); 3611 int name = track->name(); 3612 if (AudioMixer::TRACK0 <= name && 3613 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3614 name -= AudioMixer::TRACK0; 3615 traceName[4] = (name / 10) + '0'; 3616 traceName[5] = (name % 10) + '0'; 3617 } else { 3618 traceName[4] = '?'; 3619 traceName[5] = '?'; 3620 } 3621 traceName[6] = '\0'; 3622 ATRACE_INT(traceName, framesReady); 3623 } 3624 if ((framesReady >= minFrames) && track->isReady() && 3625 !track->isPaused() && !track->isTerminated()) 3626 { 3627 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3628 3629 mixedTracks++; 3630 3631 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3632 // there is an effect chain connected to the track 3633 chain.clear(); 3634 if (track->mainBuffer() != mSinkBuffer && 3635 track->mainBuffer() != mMixerBuffer) { 3636 if (mEffectBufferEnabled) { 3637 mEffectBufferValid = true; // Later can set directly. 3638 } 3639 chain = getEffectChain_l(track->sessionId()); 3640 // Delegate volume control to effect in track effect chain if needed 3641 if (chain != 0) { 3642 tracksWithEffect++; 3643 } else { 3644 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3645 "session %d", 3646 name, track->sessionId()); 3647 } 3648 } 3649 3650 3651 int param = AudioMixer::VOLUME; 3652 if (track->mFillingUpStatus == Track::FS_FILLED) { 3653 // no ramp for the first volume setting 3654 track->mFillingUpStatus = Track::FS_ACTIVE; 3655 if (track->mState == TrackBase::RESUMING) { 3656 track->mState = TrackBase::ACTIVE; 3657 param = AudioMixer::RAMP_VOLUME; 3658 } 3659 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3660 // FIXME should not make a decision based on mServer 3661 } else if (cblk->mServer != 0) { 3662 // If the track is stopped before the first frame was mixed, 3663 // do not apply ramp 3664 param = AudioMixer::RAMP_VOLUME; 3665 } 3666 3667 // compute volume for this track 3668 uint32_t vl, vr; // in U8.24 integer format 3669 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3670 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3671 vl = vr = 0; 3672 vlf = vrf = vaf = 0.; 3673 if (track->isPausing()) { 3674 track->setPaused(); 3675 } 3676 } else { 3677 3678 // read original volumes with volume control 3679 float typeVolume = mStreamTypes[track->streamType()].volume; 3680 float v = masterVolume * typeVolume; 3681 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3682 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3683 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3684 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3685 // track volumes come from shared memory, so can't be trusted and must be clamped 3686 if (vlf > GAIN_FLOAT_UNITY) { 3687 ALOGV("Track left volume out of range: %.3g", vlf); 3688 vlf = GAIN_FLOAT_UNITY; 3689 } 3690 if (vrf > GAIN_FLOAT_UNITY) { 3691 ALOGV("Track right volume out of range: %.3g", vrf); 3692 vrf = GAIN_FLOAT_UNITY; 3693 } 3694 // now apply the master volume and stream type volume 3695 vlf *= v; 3696 vrf *= v; 3697 // assuming master volume and stream type volume each go up to 1.0, 3698 // then derive vl and vr as U8.24 versions for the effect chain 3699 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3700 vl = (uint32_t) (scaleto8_24 * vlf); 3701 vr = (uint32_t) (scaleto8_24 * vrf); 3702 // vl and vr are now in U8.24 format 3703 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3704 // send level comes from shared memory and so may be corrupt 3705 if (sendLevel > MAX_GAIN_INT) { 3706 ALOGV("Track send level out of range: %04X", sendLevel); 3707 sendLevel = MAX_GAIN_INT; 3708 } 3709 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3710 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3711 } 3712 3713 // Delegate volume control to effect in track effect chain if needed 3714 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3715 // Do not ramp volume if volume is controlled by effect 3716 param = AudioMixer::VOLUME; 3717 // Update remaining floating point volume levels 3718 vlf = (float)vl / (1 << 24); 3719 vrf = (float)vr / (1 << 24); 3720 track->mHasVolumeController = true; 3721 } else { 3722 // force no volume ramp when volume controller was just disabled or removed 3723 // from effect chain to avoid volume spike 3724 if (track->mHasVolumeController) { 3725 param = AudioMixer::VOLUME; 3726 } 3727 track->mHasVolumeController = false; 3728 } 3729 3730 // XXX: these things DON'T need to be done each time 3731 mAudioMixer->setBufferProvider(name, track); 3732 mAudioMixer->enable(name); 3733 3734 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3735 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3736 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3737 mAudioMixer->setParameter( 3738 name, 3739 AudioMixer::TRACK, 3740 AudioMixer::FORMAT, (void *)track->format()); 3741 mAudioMixer->setParameter( 3742 name, 3743 AudioMixer::TRACK, 3744 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3745 mAudioMixer->setParameter( 3746 name, 3747 AudioMixer::TRACK, 3748 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3749 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3750 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3751 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3752 if (reqSampleRate == 0) { 3753 reqSampleRate = mSampleRate; 3754 } else if (reqSampleRate > maxSampleRate) { 3755 reqSampleRate = maxSampleRate; 3756 } 3757 mAudioMixer->setParameter( 3758 name, 3759 AudioMixer::RESAMPLE, 3760 AudioMixer::SAMPLE_RATE, 3761 (void *)(uintptr_t)reqSampleRate); 3762 /* 3763 * Select the appropriate output buffer for the track. 3764 * 3765 * Tracks with effects go into their own effects chain buffer 3766 * and from there into either mEffectBuffer or mSinkBuffer. 3767 * 3768 * Other tracks can use mMixerBuffer for higher precision 3769 * channel accumulation. If this buffer is enabled 3770 * (mMixerBufferEnabled true), then selected tracks will accumulate 3771 * into it. 3772 * 3773 */ 3774 if (mMixerBufferEnabled 3775 && (track->mainBuffer() == mSinkBuffer 3776 || track->mainBuffer() == mMixerBuffer)) { 3777 mAudioMixer->setParameter( 3778 name, 3779 AudioMixer::TRACK, 3780 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3781 mAudioMixer->setParameter( 3782 name, 3783 AudioMixer::TRACK, 3784 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3785 // TODO: override track->mainBuffer()? 3786 mMixerBufferValid = true; 3787 } else { 3788 mAudioMixer->setParameter( 3789 name, 3790 AudioMixer::TRACK, 3791 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3792 mAudioMixer->setParameter( 3793 name, 3794 AudioMixer::TRACK, 3795 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3796 } 3797 mAudioMixer->setParameter( 3798 name, 3799 AudioMixer::TRACK, 3800 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3801 3802 // reset retry count 3803 track->mRetryCount = kMaxTrackRetries; 3804 3805 // If one track is ready, set the mixer ready if: 3806 // - the mixer was not ready during previous round OR 3807 // - no other track is not ready 3808 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3809 mixerStatus != MIXER_TRACKS_ENABLED) { 3810 mixerStatus = MIXER_TRACKS_READY; 3811 } 3812 } else { 3813 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3814 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3815 } 3816 // clear effect chain input buffer if an active track underruns to avoid sending 3817 // previous audio buffer again to effects 3818 chain = getEffectChain_l(track->sessionId()); 3819 if (chain != 0) { 3820 chain->clearInputBuffer(); 3821 } 3822 3823 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3824 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3825 track->isStopped() || track->isPaused()) { 3826 // We have consumed all the buffers of this track. 3827 // Remove it from the list of active tracks. 3828 // TODO: use actual buffer filling status instead of latency when available from 3829 // audio HAL 3830 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3831 size_t framesWritten = mBytesWritten / mFrameSize; 3832 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3833 if (track->isStopped()) { 3834 track->reset(); 3835 } 3836 tracksToRemove->add(track); 3837 } 3838 } else { 3839 // No buffers for this track. Give it a few chances to 3840 // fill a buffer, then remove it from active list. 3841 if (--(track->mRetryCount) <= 0) { 3842 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3843 tracksToRemove->add(track); 3844 // indicate to client process that the track was disabled because of underrun; 3845 // it will then automatically call start() when data is available 3846 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3847 // If one track is not ready, mark the mixer also not ready if: 3848 // - the mixer was ready during previous round OR 3849 // - no other track is ready 3850 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3851 mixerStatus != MIXER_TRACKS_READY) { 3852 mixerStatus = MIXER_TRACKS_ENABLED; 3853 } 3854 } 3855 mAudioMixer->disable(name); 3856 } 3857 3858 } // local variable scope to avoid goto warning 3859track_is_ready: ; 3860 3861 } 3862 3863 // Push the new FastMixer state if necessary 3864 bool pauseAudioWatchdog = false; 3865 if (didModify) { 3866 state->mFastTracksGen++; 3867 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3868 if (kUseFastMixer == FastMixer_Dynamic && 3869 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3870 state->mCommand = FastMixerState::COLD_IDLE; 3871 state->mColdFutexAddr = &mFastMixerFutex; 3872 state->mColdGen++; 3873 mFastMixerFutex = 0; 3874 if (kUseFastMixer == FastMixer_Dynamic) { 3875 mNormalSink = mOutputSink; 3876 } 3877 // If we go into cold idle, need to wait for acknowledgement 3878 // so that fast mixer stops doing I/O. 3879 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3880 pauseAudioWatchdog = true; 3881 } 3882 } 3883 if (sq != NULL) { 3884 sq->end(didModify); 3885 sq->push(block); 3886 } 3887#ifdef AUDIO_WATCHDOG 3888 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3889 mAudioWatchdog->pause(); 3890 } 3891#endif 3892 3893 // Now perform the deferred reset on fast tracks that have stopped 3894 while (resetMask != 0) { 3895 size_t i = __builtin_ctz(resetMask); 3896 ALOG_ASSERT(i < count); 3897 resetMask &= ~(1 << i); 3898 sp<Track> t = mActiveTracks[i].promote(); 3899 if (t == 0) { 3900 continue; 3901 } 3902 Track* track = t.get(); 3903 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3904 track->reset(); 3905 } 3906 3907 // remove all the tracks that need to be... 3908 removeTracks_l(*tracksToRemove); 3909 3910 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3911 mEffectBufferValid = true; 3912 } 3913 3914 if (mEffectBufferValid) { 3915 // as long as there are effects we should clear the effects buffer, to avoid 3916 // passing a non-clean buffer to the effect chain 3917 memset(mEffectBuffer, 0, mEffectBufferSize); 3918 } 3919 // sink or mix buffer must be cleared if all tracks are connected to an 3920 // effect chain as in this case the mixer will not write to the sink or mix buffer 3921 // and track effects will accumulate into it 3922 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3923 (mixedTracks == 0 && fastTracks > 0))) { 3924 // FIXME as a performance optimization, should remember previous zero status 3925 if (mMixerBufferValid) { 3926 memset(mMixerBuffer, 0, mMixerBufferSize); 3927 // TODO: In testing, mSinkBuffer below need not be cleared because 3928 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3929 // after mixing. 3930 // 3931 // To enforce this guarantee: 3932 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3933 // (mixedTracks == 0 && fastTracks > 0)) 3934 // must imply MIXER_TRACKS_READY. 3935 // Later, we may clear buffers regardless, and skip much of this logic. 3936 } 3937 // FIXME as a performance optimization, should remember previous zero status 3938 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3939 } 3940 3941 // if any fast tracks, then status is ready 3942 mMixerStatusIgnoringFastTracks = mixerStatus; 3943 if (fastTracks > 0) { 3944 mixerStatus = MIXER_TRACKS_READY; 3945 } 3946 return mixerStatus; 3947} 3948 3949// getTrackName_l() must be called with ThreadBase::mLock held 3950int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3951 audio_format_t format, int sessionId) 3952{ 3953 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3954} 3955 3956// deleteTrackName_l() must be called with ThreadBase::mLock held 3957void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3958{ 3959 ALOGV("remove track (%d) and delete from mixer", name); 3960 mAudioMixer->deleteTrackName(name); 3961} 3962 3963// checkForNewParameter_l() must be called with ThreadBase::mLock held 3964bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3965 status_t& status) 3966{ 3967 bool reconfig = false; 3968 3969 status = NO_ERROR; 3970 3971 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3972 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3973 if (mFastMixer != 0) { 3974 FastMixerStateQueue *sq = mFastMixer->sq(); 3975 FastMixerState *state = sq->begin(); 3976 if (!(state->mCommand & FastMixerState::IDLE)) { 3977 previousCommand = state->mCommand; 3978 state->mCommand = FastMixerState::HOT_IDLE; 3979 sq->end(); 3980 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3981 } else { 3982 sq->end(false /*didModify*/); 3983 } 3984 } 3985 3986 AudioParameter param = AudioParameter(keyValuePair); 3987 int value; 3988 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3989 reconfig = true; 3990 } 3991 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3992 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3993 status = BAD_VALUE; 3994 } else { 3995 // no need to save value, since it's constant 3996 reconfig = true; 3997 } 3998 } 3999 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4000 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4001 status = BAD_VALUE; 4002 } else { 4003 // no need to save value, since it's constant 4004 reconfig = true; 4005 } 4006 } 4007 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4008 // do not accept frame count changes if tracks are open as the track buffer 4009 // size depends on frame count and correct behavior would not be guaranteed 4010 // if frame count is changed after track creation 4011 if (!mTracks.isEmpty()) { 4012 status = INVALID_OPERATION; 4013 } else { 4014 reconfig = true; 4015 } 4016 } 4017 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4018#ifdef ADD_BATTERY_DATA 4019 // when changing the audio output device, call addBatteryData to notify 4020 // the change 4021 if (mOutDevice != value) { 4022 uint32_t params = 0; 4023 // check whether speaker is on 4024 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4025 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4026 } 4027 4028 audio_devices_t deviceWithoutSpeaker 4029 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4030 // check if any other device (except speaker) is on 4031 if (value & deviceWithoutSpeaker ) { 4032 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4033 } 4034 4035 if (params != 0) { 4036 addBatteryData(params); 4037 } 4038 } 4039#endif 4040 4041 // forward device change to effects that have requested to be 4042 // aware of attached audio device. 4043 if (value != AUDIO_DEVICE_NONE) { 4044 mOutDevice = value; 4045 for (size_t i = 0; i < mEffectChains.size(); i++) { 4046 mEffectChains[i]->setDevice_l(mOutDevice); 4047 } 4048 } 4049 } 4050 4051 if (status == NO_ERROR) { 4052 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4053 keyValuePair.string()); 4054 if (!mStandby && status == INVALID_OPERATION) { 4055 mOutput->stream->common.standby(&mOutput->stream->common); 4056 mStandby = true; 4057 mBytesWritten = 0; 4058 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4059 keyValuePair.string()); 4060 } 4061 if (status == NO_ERROR && reconfig) { 4062 readOutputParameters_l(); 4063 delete mAudioMixer; 4064 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4065 for (size_t i = 0; i < mTracks.size() ; i++) { 4066 int name = getTrackName_l(mTracks[i]->mChannelMask, 4067 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4068 if (name < 0) { 4069 break; 4070 } 4071 mTracks[i]->mName = name; 4072 } 4073 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4074 } 4075 } 4076 4077 if (!(previousCommand & FastMixerState::IDLE)) { 4078 ALOG_ASSERT(mFastMixer != 0); 4079 FastMixerStateQueue *sq = mFastMixer->sq(); 4080 FastMixerState *state = sq->begin(); 4081 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4082 state->mCommand = previousCommand; 4083 sq->end(); 4084 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4085 } 4086 4087 return reconfig; 4088} 4089 4090 4091void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4092{ 4093 const size_t SIZE = 256; 4094 char buffer[SIZE]; 4095 String8 result; 4096 4097 PlaybackThread::dumpInternals(fd, args); 4098 4099 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4100 4101 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4102 const FastMixerDumpState copy(mFastMixerDumpState); 4103 copy.dump(fd); 4104 4105#ifdef STATE_QUEUE_DUMP 4106 // Similar for state queue 4107 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4108 observerCopy.dump(fd); 4109 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4110 mutatorCopy.dump(fd); 4111#endif 4112 4113#ifdef TEE_SINK 4114 // Write the tee output to a .wav file 4115 dumpTee(fd, mTeeSource, mId); 4116#endif 4117 4118#ifdef AUDIO_WATCHDOG 4119 if (mAudioWatchdog != 0) { 4120 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4121 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4122 wdCopy.dump(fd); 4123 } 4124#endif 4125} 4126 4127uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4128{ 4129 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4130} 4131 4132uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4133{ 4134 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4135} 4136 4137void AudioFlinger::MixerThread::cacheParameters_l() 4138{ 4139 PlaybackThread::cacheParameters_l(); 4140 4141 // FIXME: Relaxed timing because of a certain device that can't meet latency 4142 // Should be reduced to 2x after the vendor fixes the driver issue 4143 // increase threshold again due to low power audio mode. The way this warning 4144 // threshold is calculated and its usefulness should be reconsidered anyway. 4145 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4146} 4147 4148// ---------------------------------------------------------------------------- 4149 4150AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4151 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4152 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4153 // mLeftVolFloat, mRightVolFloat 4154{ 4155} 4156 4157AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4158 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4159 ThreadBase::type_t type) 4160 : PlaybackThread(audioFlinger, output, id, device, type) 4161 // mLeftVolFloat, mRightVolFloat 4162{ 4163} 4164 4165AudioFlinger::DirectOutputThread::~DirectOutputThread() 4166{ 4167} 4168 4169void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4170{ 4171 audio_track_cblk_t* cblk = track->cblk(); 4172 float left, right; 4173 4174 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4175 left = right = 0; 4176 } else { 4177 float typeVolume = mStreamTypes[track->streamType()].volume; 4178 float v = mMasterVolume * typeVolume; 4179 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4180 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4181 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4182 if (left > GAIN_FLOAT_UNITY) { 4183 left = GAIN_FLOAT_UNITY; 4184 } 4185 left *= v; 4186 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4187 if (right > GAIN_FLOAT_UNITY) { 4188 right = GAIN_FLOAT_UNITY; 4189 } 4190 right *= v; 4191 } 4192 4193 if (lastTrack) { 4194 if (left != mLeftVolFloat || right != mRightVolFloat) { 4195 mLeftVolFloat = left; 4196 mRightVolFloat = right; 4197 4198 // Convert volumes from float to 8.24 4199 uint32_t vl = (uint32_t)(left * (1 << 24)); 4200 uint32_t vr = (uint32_t)(right * (1 << 24)); 4201 4202 // Delegate volume control to effect in track effect chain if needed 4203 // only one effect chain can be present on DirectOutputThread, so if 4204 // there is one, the track is connected to it 4205 if (!mEffectChains.isEmpty()) { 4206 mEffectChains[0]->setVolume_l(&vl, &vr); 4207 left = (float)vl / (1 << 24); 4208 right = (float)vr / (1 << 24); 4209 } 4210 if (mOutput->stream->set_volume) { 4211 mOutput->stream->set_volume(mOutput->stream, left, right); 4212 } 4213 } 4214 } 4215} 4216 4217 4218AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4219 Vector< sp<Track> > *tracksToRemove 4220) 4221{ 4222 size_t count = mActiveTracks.size(); 4223 mixer_state mixerStatus = MIXER_IDLE; 4224 bool doHwPause = false; 4225 bool doHwResume = false; 4226 bool flushPending = false; 4227 4228 // find out which tracks need to be processed 4229 for (size_t i = 0; i < count; i++) { 4230 sp<Track> t = mActiveTracks[i].promote(); 4231 // The track died recently 4232 if (t == 0) { 4233 continue; 4234 } 4235 4236 Track* const track = t.get(); 4237 audio_track_cblk_t* cblk = track->cblk(); 4238 // Only consider last track started for volume and mixer state control. 4239 // In theory an older track could underrun and restart after the new one starts 4240 // but as we only care about the transition phase between two tracks on a 4241 // direct output, it is not a problem to ignore the underrun case. 4242 sp<Track> l = mLatestActiveTrack.promote(); 4243 bool last = l.get() == track; 4244 4245 if (mHwSupportsPause && track->isPausing()) { 4246 track->setPaused(); 4247 if (last && !mHwPaused) { 4248 doHwPause = true; 4249 mHwPaused = true; 4250 } 4251 tracksToRemove->add(track); 4252 } else if (track->isFlushPending()) { 4253 track->flushAck(); 4254 if (last) { 4255 flushPending = true; 4256 } 4257 } else if (mHwSupportsPause && track->isResumePending()){ 4258 track->resumeAck(); 4259 if (last) { 4260 if (mHwPaused) { 4261 doHwResume = true; 4262 mHwPaused = false; 4263 } 4264 } 4265 } 4266 4267 // The first time a track is added we wait 4268 // for all its buffers to be filled before processing it. 4269 // Allow draining the buffer in case the client 4270 // app does not call stop() and relies on underrun to stop: 4271 // hence the test on (track->mRetryCount > 1). 4272 // If retryCount<=1 then track is about to underrun and be removed. 4273 uint32_t minFrames; 4274 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4275 && (track->mRetryCount > 1)) { 4276 minFrames = mNormalFrameCount; 4277 } else { 4278 minFrames = 1; 4279 } 4280 4281 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4282 !track->isStopping_2() && !track->isStopped()) 4283 { 4284 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4285 4286 if (track->mFillingUpStatus == Track::FS_FILLED) { 4287 track->mFillingUpStatus = Track::FS_ACTIVE; 4288 // make sure processVolume_l() will apply new volume even if 0 4289 mLeftVolFloat = mRightVolFloat = -1.0; 4290 if (!mHwSupportsPause) { 4291 track->resumeAck(); 4292 } 4293 } 4294 4295 // compute volume for this track 4296 processVolume_l(track, last); 4297 if (last) { 4298 // reset retry count 4299 track->mRetryCount = kMaxTrackRetriesDirect; 4300 mActiveTrack = t; 4301 mixerStatus = MIXER_TRACKS_READY; 4302 if (usesHwAvSync() && mHwPaused) { 4303 doHwResume = true; 4304 mHwPaused = false; 4305 } 4306 } 4307 } else { 4308 // clear effect chain input buffer if the last active track started underruns 4309 // to avoid sending previous audio buffer again to effects 4310 if (!mEffectChains.isEmpty() && last) { 4311 mEffectChains[0]->clearInputBuffer(); 4312 } 4313 if (track->isStopping_1()) { 4314 track->mState = TrackBase::STOPPING_2; 4315 } 4316 if ((track->sharedBuffer() != 0) || track->isStopped() || 4317 track->isStopping_2() || track->isPaused()) { 4318 // We have consumed all the buffers of this track. 4319 // Remove it from the list of active tracks. 4320 size_t audioHALFrames; 4321 if (audio_is_linear_pcm(mFormat)) { 4322 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4323 } else { 4324 audioHALFrames = 0; 4325 } 4326 4327 size_t framesWritten = mBytesWritten / mFrameSize; 4328 if (mStandby || !last || 4329 track->presentationComplete(framesWritten, audioHALFrames)) { 4330 if (track->isStopping_2()) { 4331 track->mState = TrackBase::STOPPED; 4332 } 4333 if (track->isStopped()) { 4334 track->reset(); 4335 } 4336 tracksToRemove->add(track); 4337 } 4338 } else { 4339 // No buffers for this track. Give it a few chances to 4340 // fill a buffer, then remove it from active list. 4341 // Only consider last track started for mixer state control 4342 if (--(track->mRetryCount) <= 0) { 4343 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4344 tracksToRemove->add(track); 4345 // indicate to client process that the track was disabled because of underrun; 4346 // it will then automatically call start() when data is available 4347 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4348 } else if (last) { 4349 mixerStatus = MIXER_TRACKS_ENABLED; 4350 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4351 doHwPause = true; 4352 mHwPaused = true; 4353 } 4354 } 4355 } 4356 } 4357 } 4358 4359 // if an active track did not command a flush, check for pending flush on stopped tracks 4360 if (!flushPending) { 4361 for (size_t i = 0; i < mTracks.size(); i++) { 4362 if (mTracks[i]->isFlushPending()) { 4363 mTracks[i]->flushAck(); 4364 flushPending = true; 4365 } 4366 } 4367 } 4368 4369 // make sure the pause/flush/resume sequence is executed in the right order. 4370 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4371 // before flush and then resume HW. This can happen in case of pause/flush/resume 4372 // if resume is received before pause is executed. 4373 if (mHwSupportsPause && !mStandby && 4374 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4375 mOutput->stream->pause(mOutput->stream); 4376 } 4377 if (flushPending) { 4378 flushHw_l(); 4379 } 4380 if (mHwSupportsPause && !mStandby && doHwResume) { 4381 mOutput->stream->resume(mOutput->stream); 4382 } 4383 // remove all the tracks that need to be... 4384 removeTracks_l(*tracksToRemove); 4385 4386 return mixerStatus; 4387} 4388 4389void AudioFlinger::DirectOutputThread::threadLoop_mix() 4390{ 4391 size_t frameCount = mFrameCount; 4392 int8_t *curBuf = (int8_t *)mSinkBuffer; 4393 // output audio to hardware 4394 while (frameCount) { 4395 AudioBufferProvider::Buffer buffer; 4396 buffer.frameCount = frameCount; 4397 mActiveTrack->getNextBuffer(&buffer); 4398 if (buffer.raw == NULL) { 4399 memset(curBuf, 0, frameCount * mFrameSize); 4400 break; 4401 } 4402 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4403 frameCount -= buffer.frameCount; 4404 curBuf += buffer.frameCount * mFrameSize; 4405 mActiveTrack->releaseBuffer(&buffer); 4406 } 4407 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4408 sleepTime = 0; 4409 standbyTime = systemTime() + standbyDelay; 4410 mActiveTrack.clear(); 4411} 4412 4413void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4414{ 4415 // do not write to HAL when paused 4416 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4417 sleepTime = idleSleepTime; 4418 return; 4419 } 4420 if (sleepTime == 0) { 4421 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4422 sleepTime = activeSleepTime; 4423 } else { 4424 sleepTime = idleSleepTime; 4425 } 4426 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4427 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4428 sleepTime = 0; 4429 } 4430} 4431 4432void AudioFlinger::DirectOutputThread::threadLoop_exit() 4433{ 4434 { 4435 Mutex::Autolock _l(mLock); 4436 bool flushPending = false; 4437 for (size_t i = 0; i < mTracks.size(); i++) { 4438 if (mTracks[i]->isFlushPending()) { 4439 mTracks[i]->flushAck(); 4440 flushPending = true; 4441 } 4442 } 4443 if (flushPending) { 4444 flushHw_l(); 4445 } 4446 } 4447 PlaybackThread::threadLoop_exit(); 4448} 4449 4450// must be called with thread mutex locked 4451bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4452{ 4453 bool trackPaused = false; 4454 4455 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4456 // after a timeout and we will enter standby then. 4457 if (mTracks.size() > 0) { 4458 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4459 } 4460 4461 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused)); 4462} 4463 4464// getTrackName_l() must be called with ThreadBase::mLock held 4465int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4466 audio_format_t format __unused, int sessionId __unused) 4467{ 4468 return 0; 4469} 4470 4471// deleteTrackName_l() must be called with ThreadBase::mLock held 4472void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4473{ 4474} 4475 4476// checkForNewParameter_l() must be called with ThreadBase::mLock held 4477bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4478 status_t& status) 4479{ 4480 bool reconfig = false; 4481 4482 status = NO_ERROR; 4483 4484 AudioParameter param = AudioParameter(keyValuePair); 4485 int value; 4486 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4487 // forward device change to effects that have requested to be 4488 // aware of attached audio device. 4489 if (value != AUDIO_DEVICE_NONE) { 4490 mOutDevice = value; 4491 for (size_t i = 0; i < mEffectChains.size(); i++) { 4492 mEffectChains[i]->setDevice_l(mOutDevice); 4493 } 4494 } 4495 } 4496 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4497 // do not accept frame count changes if tracks are open as the track buffer 4498 // size depends on frame count and correct behavior would not be garantied 4499 // if frame count is changed after track creation 4500 if (!mTracks.isEmpty()) { 4501 status = INVALID_OPERATION; 4502 } else { 4503 reconfig = true; 4504 } 4505 } 4506 if (status == NO_ERROR) { 4507 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4508 keyValuePair.string()); 4509 if (!mStandby && status == INVALID_OPERATION) { 4510 mOutput->stream->common.standby(&mOutput->stream->common); 4511 mStandby = true; 4512 mBytesWritten = 0; 4513 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4514 keyValuePair.string()); 4515 } 4516 if (status == NO_ERROR && reconfig) { 4517 readOutputParameters_l(); 4518 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4519 } 4520 } 4521 4522 return reconfig; 4523} 4524 4525uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4526{ 4527 uint32_t time; 4528 if (audio_is_linear_pcm(mFormat)) { 4529 time = PlaybackThread::activeSleepTimeUs(); 4530 } else { 4531 time = 10000; 4532 } 4533 return time; 4534} 4535 4536uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4537{ 4538 uint32_t time; 4539 if (audio_is_linear_pcm(mFormat)) { 4540 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4541 } else { 4542 time = 10000; 4543 } 4544 return time; 4545} 4546 4547uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4548{ 4549 uint32_t time; 4550 if (audio_is_linear_pcm(mFormat)) { 4551 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4552 } else { 4553 time = 10000; 4554 } 4555 return time; 4556} 4557 4558void AudioFlinger::DirectOutputThread::cacheParameters_l() 4559{ 4560 PlaybackThread::cacheParameters_l(); 4561 4562 // use shorter standby delay as on normal output to release 4563 // hardware resources as soon as possible 4564 if (audio_is_linear_pcm(mFormat)) { 4565 standbyDelay = microseconds(activeSleepTime*2); 4566 } else { 4567 standbyDelay = kOffloadStandbyDelayNs; 4568 } 4569} 4570 4571void AudioFlinger::DirectOutputThread::flushHw_l() 4572{ 4573 if (mOutput->stream->flush != NULL) { 4574 mOutput->stream->flush(mOutput->stream); 4575 } 4576 mHwPaused = false; 4577} 4578 4579// ---------------------------------------------------------------------------- 4580 4581AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4582 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4583 : Thread(false /*canCallJava*/), 4584 mPlaybackThread(playbackThread), 4585 mWriteAckSequence(0), 4586 mDrainSequence(0) 4587{ 4588} 4589 4590AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4591{ 4592} 4593 4594void AudioFlinger::AsyncCallbackThread::onFirstRef() 4595{ 4596 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4597} 4598 4599bool AudioFlinger::AsyncCallbackThread::threadLoop() 4600{ 4601 while (!exitPending()) { 4602 uint32_t writeAckSequence; 4603 uint32_t drainSequence; 4604 4605 { 4606 Mutex::Autolock _l(mLock); 4607 while (!((mWriteAckSequence & 1) || 4608 (mDrainSequence & 1) || 4609 exitPending())) { 4610 mWaitWorkCV.wait(mLock); 4611 } 4612 4613 if (exitPending()) { 4614 break; 4615 } 4616 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4617 mWriteAckSequence, mDrainSequence); 4618 writeAckSequence = mWriteAckSequence; 4619 mWriteAckSequence &= ~1; 4620 drainSequence = mDrainSequence; 4621 mDrainSequence &= ~1; 4622 } 4623 { 4624 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4625 if (playbackThread != 0) { 4626 if (writeAckSequence & 1) { 4627 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4628 } 4629 if (drainSequence & 1) { 4630 playbackThread->resetDraining(drainSequence >> 1); 4631 } 4632 } 4633 } 4634 } 4635 return false; 4636} 4637 4638void AudioFlinger::AsyncCallbackThread::exit() 4639{ 4640 ALOGV("AsyncCallbackThread::exit"); 4641 Mutex::Autolock _l(mLock); 4642 requestExit(); 4643 mWaitWorkCV.broadcast(); 4644} 4645 4646void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4647{ 4648 Mutex::Autolock _l(mLock); 4649 // bit 0 is cleared 4650 mWriteAckSequence = sequence << 1; 4651} 4652 4653void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4654{ 4655 Mutex::Autolock _l(mLock); 4656 // ignore unexpected callbacks 4657 if (mWriteAckSequence & 2) { 4658 mWriteAckSequence |= 1; 4659 mWaitWorkCV.signal(); 4660 } 4661} 4662 4663void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4664{ 4665 Mutex::Autolock _l(mLock); 4666 // bit 0 is cleared 4667 mDrainSequence = sequence << 1; 4668} 4669 4670void AudioFlinger::AsyncCallbackThread::resetDraining() 4671{ 4672 Mutex::Autolock _l(mLock); 4673 // ignore unexpected callbacks 4674 if (mDrainSequence & 2) { 4675 mDrainSequence |= 1; 4676 mWaitWorkCV.signal(); 4677 } 4678} 4679 4680 4681// ---------------------------------------------------------------------------- 4682AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4683 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4684 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4685 mPausedBytesRemaining(0) 4686{ 4687 //FIXME: mStandby should be set to true by ThreadBase constructor 4688 mStandby = true; 4689} 4690 4691void AudioFlinger::OffloadThread::threadLoop_exit() 4692{ 4693 if (mFlushPending || mHwPaused) { 4694 // If a flush is pending or track was paused, just discard buffered data 4695 flushHw_l(); 4696 } else { 4697 mMixerStatus = MIXER_DRAIN_ALL; 4698 threadLoop_drain(); 4699 } 4700 if (mUseAsyncWrite) { 4701 ALOG_ASSERT(mCallbackThread != 0); 4702 mCallbackThread->exit(); 4703 } 4704 PlaybackThread::threadLoop_exit(); 4705} 4706 4707AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4708 Vector< sp<Track> > *tracksToRemove 4709) 4710{ 4711 size_t count = mActiveTracks.size(); 4712 4713 mixer_state mixerStatus = MIXER_IDLE; 4714 bool doHwPause = false; 4715 bool doHwResume = false; 4716 4717 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4718 4719 // find out which tracks need to be processed 4720 for (size_t i = 0; i < count; i++) { 4721 sp<Track> t = mActiveTracks[i].promote(); 4722 // The track died recently 4723 if (t == 0) { 4724 continue; 4725 } 4726 Track* const track = t.get(); 4727 audio_track_cblk_t* cblk = track->cblk(); 4728 // Only consider last track started for volume and mixer state control. 4729 // In theory an older track could underrun and restart after the new one starts 4730 // but as we only care about the transition phase between two tracks on a 4731 // direct output, it is not a problem to ignore the underrun case. 4732 sp<Track> l = mLatestActiveTrack.promote(); 4733 bool last = l.get() == track; 4734 4735 if (track->isInvalid()) { 4736 ALOGW("An invalidated track shouldn't be in active list"); 4737 tracksToRemove->add(track); 4738 continue; 4739 } 4740 4741 if (track->mState == TrackBase::IDLE) { 4742 ALOGW("An idle track shouldn't be in active list"); 4743 continue; 4744 } 4745 4746 if (track->isPausing()) { 4747 track->setPaused(); 4748 if (last) { 4749 if (!mHwPaused) { 4750 doHwPause = true; 4751 mHwPaused = true; 4752 } 4753 // If we were part way through writing the mixbuffer to 4754 // the HAL we must save this until we resume 4755 // BUG - this will be wrong if a different track is made active, 4756 // in that case we want to discard the pending data in the 4757 // mixbuffer and tell the client to present it again when the 4758 // track is resumed 4759 mPausedWriteLength = mCurrentWriteLength; 4760 mPausedBytesRemaining = mBytesRemaining; 4761 mBytesRemaining = 0; // stop writing 4762 } 4763 tracksToRemove->add(track); 4764 } else if (track->isFlushPending()) { 4765 track->flushAck(); 4766 if (last) { 4767 mFlushPending = true; 4768 } 4769 } else if (track->isResumePending()){ 4770 track->resumeAck(); 4771 if (last) { 4772 if (mPausedBytesRemaining) { 4773 // Need to continue write that was interrupted 4774 mCurrentWriteLength = mPausedWriteLength; 4775 mBytesRemaining = mPausedBytesRemaining; 4776 mPausedBytesRemaining = 0; 4777 } 4778 if (mHwPaused) { 4779 doHwResume = true; 4780 mHwPaused = false; 4781 // threadLoop_mix() will handle the case that we need to 4782 // resume an interrupted write 4783 } 4784 // enable write to audio HAL 4785 sleepTime = 0; 4786 4787 // Do not handle new data in this iteration even if track->framesReady() 4788 mixerStatus = MIXER_TRACKS_ENABLED; 4789 } 4790 } else if (track->framesReady() && track->isReady() && 4791 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4792 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4793 if (track->mFillingUpStatus == Track::FS_FILLED) { 4794 track->mFillingUpStatus = Track::FS_ACTIVE; 4795 // make sure processVolume_l() will apply new volume even if 0 4796 mLeftVolFloat = mRightVolFloat = -1.0; 4797 } 4798 4799 if (last) { 4800 sp<Track> previousTrack = mPreviousTrack.promote(); 4801 if (previousTrack != 0) { 4802 if (track != previousTrack.get()) { 4803 // Flush any data still being written from last track 4804 mBytesRemaining = 0; 4805 if (mPausedBytesRemaining) { 4806 // Last track was paused so we also need to flush saved 4807 // mixbuffer state and invalidate track so that it will 4808 // re-submit that unwritten data when it is next resumed 4809 mPausedBytesRemaining = 0; 4810 // Invalidate is a bit drastic - would be more efficient 4811 // to have a flag to tell client that some of the 4812 // previously written data was lost 4813 previousTrack->invalidate(); 4814 } 4815 // flush data already sent to the DSP if changing audio session as audio 4816 // comes from a different source. Also invalidate previous track to force a 4817 // seek when resuming. 4818 if (previousTrack->sessionId() != track->sessionId()) { 4819 previousTrack->invalidate(); 4820 } 4821 } 4822 } 4823 mPreviousTrack = track; 4824 // reset retry count 4825 track->mRetryCount = kMaxTrackRetriesOffload; 4826 mActiveTrack = t; 4827 mixerStatus = MIXER_TRACKS_READY; 4828 } 4829 } else { 4830 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4831 if (track->isStopping_1()) { 4832 // Hardware buffer can hold a large amount of audio so we must 4833 // wait for all current track's data to drain before we say 4834 // that the track is stopped. 4835 if (mBytesRemaining == 0) { 4836 // Only start draining when all data in mixbuffer 4837 // has been written 4838 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4839 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4840 // do not drain if no data was ever sent to HAL (mStandby == true) 4841 if (last && !mStandby) { 4842 // do not modify drain sequence if we are already draining. This happens 4843 // when resuming from pause after drain. 4844 if ((mDrainSequence & 1) == 0) { 4845 sleepTime = 0; 4846 standbyTime = systemTime() + standbyDelay; 4847 mixerStatus = MIXER_DRAIN_TRACK; 4848 mDrainSequence += 2; 4849 } 4850 if (mHwPaused) { 4851 // It is possible to move from PAUSED to STOPPING_1 without 4852 // a resume so we must ensure hardware is running 4853 doHwResume = true; 4854 mHwPaused = false; 4855 } 4856 } 4857 } 4858 } else if (track->isStopping_2()) { 4859 // Drain has completed or we are in standby, signal presentation complete 4860 if (!(mDrainSequence & 1) || !last || mStandby) { 4861 track->mState = TrackBase::STOPPED; 4862 size_t audioHALFrames = 4863 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4864 size_t framesWritten = 4865 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4866 track->presentationComplete(framesWritten, audioHALFrames); 4867 track->reset(); 4868 tracksToRemove->add(track); 4869 } 4870 } else { 4871 // No buffers for this track. Give it a few chances to 4872 // fill a buffer, then remove it from active list. 4873 if (--(track->mRetryCount) <= 0) { 4874 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4875 track->name()); 4876 tracksToRemove->add(track); 4877 // indicate to client process that the track was disabled because of underrun; 4878 // it will then automatically call start() when data is available 4879 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4880 } else if (last){ 4881 mixerStatus = MIXER_TRACKS_ENABLED; 4882 } 4883 } 4884 } 4885 // compute volume for this track 4886 processVolume_l(track, last); 4887 } 4888 4889 // make sure the pause/flush/resume sequence is executed in the right order. 4890 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4891 // before flush and then resume HW. This can happen in case of pause/flush/resume 4892 // if resume is received before pause is executed. 4893 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4894 mOutput->stream->pause(mOutput->stream); 4895 } 4896 if (mFlushPending) { 4897 flushHw_l(); 4898 mFlushPending = false; 4899 } 4900 if (!mStandby && doHwResume) { 4901 mOutput->stream->resume(mOutput->stream); 4902 } 4903 4904 // remove all the tracks that need to be... 4905 removeTracks_l(*tracksToRemove); 4906 4907 return mixerStatus; 4908} 4909 4910// must be called with thread mutex locked 4911bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4912{ 4913 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4914 mWriteAckSequence, mDrainSequence); 4915 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4916 return true; 4917 } 4918 return false; 4919} 4920 4921bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4922{ 4923 Mutex::Autolock _l(mLock); 4924 return waitingAsyncCallback_l(); 4925} 4926 4927void AudioFlinger::OffloadThread::flushHw_l() 4928{ 4929 DirectOutputThread::flushHw_l(); 4930 // Flush anything still waiting in the mixbuffer 4931 mCurrentWriteLength = 0; 4932 mBytesRemaining = 0; 4933 mPausedWriteLength = 0; 4934 mPausedBytesRemaining = 0; 4935 4936 if (mUseAsyncWrite) { 4937 // discard any pending drain or write ack by incrementing sequence 4938 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4939 mDrainSequence = (mDrainSequence + 2) & ~1; 4940 ALOG_ASSERT(mCallbackThread != 0); 4941 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4942 mCallbackThread->setDraining(mDrainSequence); 4943 } 4944} 4945 4946void AudioFlinger::OffloadThread::onAddNewTrack_l() 4947{ 4948 sp<Track> previousTrack = mPreviousTrack.promote(); 4949 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4950 4951 if (previousTrack != 0 && latestTrack != 0 && 4952 (previousTrack->sessionId() != latestTrack->sessionId())) { 4953 mFlushPending = true; 4954 } 4955 PlaybackThread::onAddNewTrack_l(); 4956} 4957 4958// ---------------------------------------------------------------------------- 4959 4960AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4961 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4962 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4963 DUPLICATING), 4964 mWaitTimeMs(UINT_MAX) 4965{ 4966 addOutputTrack(mainThread); 4967} 4968 4969AudioFlinger::DuplicatingThread::~DuplicatingThread() 4970{ 4971 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4972 mOutputTracks[i]->destroy(); 4973 } 4974} 4975 4976void AudioFlinger::DuplicatingThread::threadLoop_mix() 4977{ 4978 // mix buffers... 4979 if (outputsReady(outputTracks)) { 4980 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4981 } else { 4982 if (mMixerBufferValid) { 4983 memset(mMixerBuffer, 0, mMixerBufferSize); 4984 } else { 4985 memset(mSinkBuffer, 0, mSinkBufferSize); 4986 } 4987 } 4988 sleepTime = 0; 4989 writeFrames = mNormalFrameCount; 4990 mCurrentWriteLength = mSinkBufferSize; 4991 standbyTime = systemTime() + standbyDelay; 4992} 4993 4994void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4995{ 4996 if (sleepTime == 0) { 4997 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4998 sleepTime = activeSleepTime; 4999 } else { 5000 sleepTime = idleSleepTime; 5001 } 5002 } else if (mBytesWritten != 0) { 5003 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5004 writeFrames = mNormalFrameCount; 5005 memset(mSinkBuffer, 0, mSinkBufferSize); 5006 } else { 5007 // flush remaining overflow buffers in output tracks 5008 writeFrames = 0; 5009 } 5010 sleepTime = 0; 5011 } 5012} 5013 5014ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5015{ 5016 for (size_t i = 0; i < outputTracks.size(); i++) { 5017 outputTracks[i]->write(mSinkBuffer, writeFrames); 5018 } 5019 mStandby = false; 5020 return (ssize_t)mSinkBufferSize; 5021} 5022 5023void AudioFlinger::DuplicatingThread::threadLoop_standby() 5024{ 5025 // DuplicatingThread implements standby by stopping all tracks 5026 for (size_t i = 0; i < outputTracks.size(); i++) { 5027 outputTracks[i]->stop(); 5028 } 5029} 5030 5031void AudioFlinger::DuplicatingThread::saveOutputTracks() 5032{ 5033 outputTracks = mOutputTracks; 5034} 5035 5036void AudioFlinger::DuplicatingThread::clearOutputTracks() 5037{ 5038 outputTracks.clear(); 5039} 5040 5041void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5042{ 5043 Mutex::Autolock _l(mLock); 5044 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5045 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5046 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5047 const size_t frameCount = 5048 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5049 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5050 // from different OutputTracks and their associated MixerThreads (e.g. one may 5051 // nearly empty and the other may be dropping data). 5052 5053 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5054 this, 5055 mSampleRate, 5056 mFormat, 5057 mChannelMask, 5058 frameCount, 5059 IPCThreadState::self()->getCallingUid()); 5060 if (outputTrack->cblk() != NULL) { 5061 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5062 mOutputTracks.add(outputTrack); 5063 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5064 updateWaitTime_l(); 5065 } 5066} 5067 5068void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5069{ 5070 Mutex::Autolock _l(mLock); 5071 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5072 if (mOutputTracks[i]->thread() == thread) { 5073 mOutputTracks[i]->destroy(); 5074 mOutputTracks.removeAt(i); 5075 updateWaitTime_l(); 5076 return; 5077 } 5078 } 5079 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5080} 5081 5082// caller must hold mLock 5083void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5084{ 5085 mWaitTimeMs = UINT_MAX; 5086 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5087 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5088 if (strong != 0) { 5089 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5090 if (waitTimeMs < mWaitTimeMs) { 5091 mWaitTimeMs = waitTimeMs; 5092 } 5093 } 5094 } 5095} 5096 5097 5098bool AudioFlinger::DuplicatingThread::outputsReady( 5099 const SortedVector< sp<OutputTrack> > &outputTracks) 5100{ 5101 for (size_t i = 0; i < outputTracks.size(); i++) { 5102 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5103 if (thread == 0) { 5104 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5105 outputTracks[i].get()); 5106 return false; 5107 } 5108 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5109 // see note at standby() declaration 5110 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5111 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5112 thread.get()); 5113 return false; 5114 } 5115 } 5116 return true; 5117} 5118 5119uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5120{ 5121 return (mWaitTimeMs * 1000) / 2; 5122} 5123 5124void AudioFlinger::DuplicatingThread::cacheParameters_l() 5125{ 5126 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5127 updateWaitTime_l(); 5128 5129 MixerThread::cacheParameters_l(); 5130} 5131 5132// ---------------------------------------------------------------------------- 5133// Record 5134// ---------------------------------------------------------------------------- 5135 5136AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5137 AudioStreamIn *input, 5138 audio_io_handle_t id, 5139 audio_devices_t outDevice, 5140 audio_devices_t inDevice 5141#ifdef TEE_SINK 5142 , const sp<NBAIO_Sink>& teeSink 5143#endif 5144 ) : 5145 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5146 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5147 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5148 mRsmpInRear(0) 5149#ifdef TEE_SINK 5150 , mTeeSink(teeSink) 5151#endif 5152 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5153 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5154 // mFastCapture below 5155 , mFastCaptureFutex(0) 5156 // mInputSource 5157 // mPipeSink 5158 // mPipeSource 5159 , mPipeFramesP2(0) 5160 // mPipeMemory 5161 // mFastCaptureNBLogWriter 5162 , mFastTrackAvail(false) 5163{ 5164 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5165 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5166 5167 readInputParameters_l(); 5168 5169 // create an NBAIO source for the HAL input stream, and negotiate 5170 mInputSource = new AudioStreamInSource(input->stream); 5171 size_t numCounterOffers = 0; 5172 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5173 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5174 ALOG_ASSERT(index == 0); 5175 5176 // initialize fast capture depending on configuration 5177 bool initFastCapture; 5178 switch (kUseFastCapture) { 5179 case FastCapture_Never: 5180 initFastCapture = false; 5181 break; 5182 case FastCapture_Always: 5183 initFastCapture = true; 5184 break; 5185 case FastCapture_Static: 5186 uint32_t primaryOutputSampleRate; 5187 { 5188 AutoMutex _l(audioFlinger->mHardwareLock); 5189 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5190 } 5191 initFastCapture = 5192 // either capture sample rate is same as (a reasonable) primary output sample rate 5193 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5194 (mSampleRate == primaryOutputSampleRate)) || 5195 // or primary output sample rate is unknown, and capture sample rate is reasonable 5196 ((primaryOutputSampleRate == 0) && 5197 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5198 // and the buffer size is < 12 ms 5199 (mFrameCount * 1000) / mSampleRate < 12; 5200 break; 5201 // case FastCapture_Dynamic: 5202 } 5203 5204 if (initFastCapture) { 5205 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 5206 NBAIO_Format format = mInputSource->format(); 5207 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5208 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5209 void *pipeBuffer; 5210 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5211 sp<IMemory> pipeMemory; 5212 if ((roHeap == 0) || 5213 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5214 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5215 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5216 goto failed; 5217 } 5218 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5219 memset(pipeBuffer, 0, pipeSize); 5220 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5221 const NBAIO_Format offers[1] = {format}; 5222 size_t numCounterOffers = 0; 5223 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5224 ALOG_ASSERT(index == 0); 5225 mPipeSink = pipe; 5226 PipeReader *pipeReader = new PipeReader(*pipe); 5227 numCounterOffers = 0; 5228 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5229 ALOG_ASSERT(index == 0); 5230 mPipeSource = pipeReader; 5231 mPipeFramesP2 = pipeFramesP2; 5232 mPipeMemory = pipeMemory; 5233 5234 // create fast capture 5235 mFastCapture = new FastCapture(); 5236 FastCaptureStateQueue *sq = mFastCapture->sq(); 5237#ifdef STATE_QUEUE_DUMP 5238 // FIXME 5239#endif 5240 FastCaptureState *state = sq->begin(); 5241 state->mCblk = NULL; 5242 state->mInputSource = mInputSource.get(); 5243 state->mInputSourceGen++; 5244 state->mPipeSink = pipe; 5245 state->mPipeSinkGen++; 5246 state->mFrameCount = mFrameCount; 5247 state->mCommand = FastCaptureState::COLD_IDLE; 5248 // already done in constructor initialization list 5249 //mFastCaptureFutex = 0; 5250 state->mColdFutexAddr = &mFastCaptureFutex; 5251 state->mColdGen++; 5252 state->mDumpState = &mFastCaptureDumpState; 5253#ifdef TEE_SINK 5254 // FIXME 5255#endif 5256 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5257 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5258 sq->end(); 5259 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5260 5261 // start the fast capture 5262 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5263 pid_t tid = mFastCapture->getTid(); 5264 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5265 if (err != 0) { 5266 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5267 kPriorityFastCapture, getpid_cached, tid, err); 5268 } 5269 5270#ifdef AUDIO_WATCHDOG 5271 // FIXME 5272#endif 5273 5274 mFastTrackAvail = true; 5275 } 5276failed: ; 5277 5278 // FIXME mNormalSource 5279} 5280 5281 5282AudioFlinger::RecordThread::~RecordThread() 5283{ 5284 if (mFastCapture != 0) { 5285 FastCaptureStateQueue *sq = mFastCapture->sq(); 5286 FastCaptureState *state = sq->begin(); 5287 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5288 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5289 if (old == -1) { 5290 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5291 } 5292 } 5293 state->mCommand = FastCaptureState::EXIT; 5294 sq->end(); 5295 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5296 mFastCapture->join(); 5297 mFastCapture.clear(); 5298 } 5299 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5300 mAudioFlinger->unregisterWriter(mNBLogWriter); 5301 delete[] mRsmpInBuffer; 5302} 5303 5304void AudioFlinger::RecordThread::onFirstRef() 5305{ 5306 run(mThreadName, PRIORITY_URGENT_AUDIO); 5307} 5308 5309bool AudioFlinger::RecordThread::threadLoop() 5310{ 5311 nsecs_t lastWarning = 0; 5312 5313 inputStandBy(); 5314 5315reacquire_wakelock: 5316 sp<RecordTrack> activeTrack; 5317 int activeTracksGen; 5318 { 5319 Mutex::Autolock _l(mLock); 5320 size_t size = mActiveTracks.size(); 5321 activeTracksGen = mActiveTracksGen; 5322 if (size > 0) { 5323 // FIXME an arbitrary choice 5324 activeTrack = mActiveTracks[0]; 5325 acquireWakeLock_l(activeTrack->uid()); 5326 if (size > 1) { 5327 SortedVector<int> tmp; 5328 for (size_t i = 0; i < size; i++) { 5329 tmp.add(mActiveTracks[i]->uid()); 5330 } 5331 updateWakeLockUids_l(tmp); 5332 } 5333 } else { 5334 acquireWakeLock_l(-1); 5335 } 5336 } 5337 5338 // used to request a deferred sleep, to be executed later while mutex is unlocked 5339 uint32_t sleepUs = 0; 5340 5341 // loop while there is work to do 5342 for (;;) { 5343 Vector< sp<EffectChain> > effectChains; 5344 5345 // sleep with mutex unlocked 5346 if (sleepUs > 0) { 5347 ATRACE_BEGIN("sleep"); 5348 usleep(sleepUs); 5349 ATRACE_END(); 5350 sleepUs = 0; 5351 } 5352 5353 // activeTracks accumulates a copy of a subset of mActiveTracks 5354 Vector< sp<RecordTrack> > activeTracks; 5355 5356 // reference to the (first and only) active fast track 5357 sp<RecordTrack> fastTrack; 5358 5359 // reference to a fast track which is about to be removed 5360 sp<RecordTrack> fastTrackToRemove; 5361 5362 { // scope for mLock 5363 Mutex::Autolock _l(mLock); 5364 5365 processConfigEvents_l(); 5366 5367 // check exitPending here because checkForNewParameters_l() and 5368 // checkForNewParameters_l() can temporarily release mLock 5369 if (exitPending()) { 5370 break; 5371 } 5372 5373 // if no active track(s), then standby and release wakelock 5374 size_t size = mActiveTracks.size(); 5375 if (size == 0) { 5376 standbyIfNotAlreadyInStandby(); 5377 // exitPending() can't become true here 5378 releaseWakeLock_l(); 5379 ALOGV("RecordThread: loop stopping"); 5380 // go to sleep 5381 mWaitWorkCV.wait(mLock); 5382 ALOGV("RecordThread: loop starting"); 5383 goto reacquire_wakelock; 5384 } 5385 5386 if (mActiveTracksGen != activeTracksGen) { 5387 activeTracksGen = mActiveTracksGen; 5388 SortedVector<int> tmp; 5389 for (size_t i = 0; i < size; i++) { 5390 tmp.add(mActiveTracks[i]->uid()); 5391 } 5392 updateWakeLockUids_l(tmp); 5393 } 5394 5395 bool doBroadcast = false; 5396 for (size_t i = 0; i < size; ) { 5397 5398 activeTrack = mActiveTracks[i]; 5399 if (activeTrack->isTerminated()) { 5400 if (activeTrack->isFastTrack()) { 5401 ALOG_ASSERT(fastTrackToRemove == 0); 5402 fastTrackToRemove = activeTrack; 5403 } 5404 removeTrack_l(activeTrack); 5405 mActiveTracks.remove(activeTrack); 5406 mActiveTracksGen++; 5407 size--; 5408 continue; 5409 } 5410 5411 TrackBase::track_state activeTrackState = activeTrack->mState; 5412 switch (activeTrackState) { 5413 5414 case TrackBase::PAUSING: 5415 mActiveTracks.remove(activeTrack); 5416 mActiveTracksGen++; 5417 doBroadcast = true; 5418 size--; 5419 continue; 5420 5421 case TrackBase::STARTING_1: 5422 sleepUs = 10000; 5423 i++; 5424 continue; 5425 5426 case TrackBase::STARTING_2: 5427 doBroadcast = true; 5428 mStandby = false; 5429 activeTrack->mState = TrackBase::ACTIVE; 5430 break; 5431 5432 case TrackBase::ACTIVE: 5433 break; 5434 5435 case TrackBase::IDLE: 5436 i++; 5437 continue; 5438 5439 default: 5440 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5441 } 5442 5443 activeTracks.add(activeTrack); 5444 i++; 5445 5446 if (activeTrack->isFastTrack()) { 5447 ALOG_ASSERT(!mFastTrackAvail); 5448 ALOG_ASSERT(fastTrack == 0); 5449 fastTrack = activeTrack; 5450 } 5451 } 5452 if (doBroadcast) { 5453 mStartStopCond.broadcast(); 5454 } 5455 5456 // sleep if there are no active tracks to process 5457 if (activeTracks.size() == 0) { 5458 if (sleepUs == 0) { 5459 sleepUs = kRecordThreadSleepUs; 5460 } 5461 continue; 5462 } 5463 sleepUs = 0; 5464 5465 lockEffectChains_l(effectChains); 5466 } 5467 5468 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5469 5470 size_t size = effectChains.size(); 5471 for (size_t i = 0; i < size; i++) { 5472 // thread mutex is not locked, but effect chain is locked 5473 effectChains[i]->process_l(); 5474 } 5475 5476 // Push a new fast capture state if fast capture is not already running, or cblk change 5477 if (mFastCapture != 0) { 5478 FastCaptureStateQueue *sq = mFastCapture->sq(); 5479 FastCaptureState *state = sq->begin(); 5480 bool didModify = false; 5481 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5482 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5483 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5484 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5485 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5486 if (old == -1) { 5487 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5488 } 5489 } 5490 state->mCommand = FastCaptureState::READ_WRITE; 5491#if 0 // FIXME 5492 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5493 FastThreadDumpState::kSamplingNforLowRamDevice : 5494 FastThreadDumpState::kSamplingN); 5495#endif 5496 didModify = true; 5497 } 5498 audio_track_cblk_t *cblkOld = state->mCblk; 5499 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5500 if (cblkNew != cblkOld) { 5501 state->mCblk = cblkNew; 5502 // block until acked if removing a fast track 5503 if (cblkOld != NULL) { 5504 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5505 } 5506 didModify = true; 5507 } 5508 sq->end(didModify); 5509 if (didModify) { 5510 sq->push(block); 5511#if 0 5512 if (kUseFastCapture == FastCapture_Dynamic) { 5513 mNormalSource = mPipeSource; 5514 } 5515#endif 5516 } 5517 } 5518 5519 // now run the fast track destructor with thread mutex unlocked 5520 fastTrackToRemove.clear(); 5521 5522 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5523 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5524 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5525 // If destination is non-contiguous, first read past the nominal end of buffer, then 5526 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5527 5528 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5529 ssize_t framesRead; 5530 5531 // If an NBAIO source is present, use it to read the normal capture's data 5532 if (mPipeSource != 0) { 5533 size_t framesToRead = mBufferSize / mFrameSize; 5534 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5535 framesToRead, AudioBufferProvider::kInvalidPTS); 5536 if (framesRead == 0) { 5537 // since pipe is non-blocking, simulate blocking input 5538 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5539 } 5540 // otherwise use the HAL / AudioStreamIn directly 5541 } else { 5542 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5543 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5544 if (bytesRead < 0) { 5545 framesRead = bytesRead; 5546 } else { 5547 framesRead = bytesRead / mFrameSize; 5548 } 5549 } 5550 5551 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5552 ALOGE("read failed: framesRead=%d", framesRead); 5553 // Force input into standby so that it tries to recover at next read attempt 5554 inputStandBy(); 5555 sleepUs = kRecordThreadSleepUs; 5556 } 5557 if (framesRead <= 0) { 5558 goto unlock; 5559 } 5560 ALOG_ASSERT(framesRead > 0); 5561 5562 if (mTeeSink != 0) { 5563 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5564 } 5565 // If destination is non-contiguous, we now correct for reading past end of buffer. 5566 { 5567 size_t part1 = mRsmpInFramesP2 - rear; 5568 if ((size_t) framesRead > part1) { 5569 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5570 (framesRead - part1) * mFrameSize); 5571 } 5572 } 5573 rear = mRsmpInRear += framesRead; 5574 5575 size = activeTracks.size(); 5576 // loop over each active track 5577 for (size_t i = 0; i < size; i++) { 5578 activeTrack = activeTracks[i]; 5579 5580 // skip fast tracks, as those are handled directly by FastCapture 5581 if (activeTrack->isFastTrack()) { 5582 continue; 5583 } 5584 5585 enum { 5586 OVERRUN_UNKNOWN, 5587 OVERRUN_TRUE, 5588 OVERRUN_FALSE 5589 } overrun = OVERRUN_UNKNOWN; 5590 5591 // loop over getNextBuffer to handle circular sink 5592 for (;;) { 5593 5594 activeTrack->mSink.frameCount = ~0; 5595 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5596 size_t framesOut = activeTrack->mSink.frameCount; 5597 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5598 5599 int32_t front = activeTrack->mRsmpInFront; 5600 ssize_t filled = rear - front; 5601 size_t framesIn; 5602 5603 if (filled < 0) { 5604 // should not happen, but treat like a massive overrun and re-sync 5605 framesIn = 0; 5606 activeTrack->mRsmpInFront = rear; 5607 overrun = OVERRUN_TRUE; 5608 } else if ((size_t) filled <= mRsmpInFrames) { 5609 framesIn = (size_t) filled; 5610 } else { 5611 // client is not keeping up with server, but give it latest data 5612 framesIn = mRsmpInFrames; 5613 activeTrack->mRsmpInFront = front = rear - framesIn; 5614 overrun = OVERRUN_TRUE; 5615 } 5616 5617 if (framesOut == 0 || framesIn == 0) { 5618 break; 5619 } 5620 5621 if (activeTrack->mResampler == NULL) { 5622 // no resampling 5623 if (framesIn > framesOut) { 5624 framesIn = framesOut; 5625 } else { 5626 framesOut = framesIn; 5627 } 5628 int8_t *dst = activeTrack->mSink.i8; 5629 while (framesIn > 0) { 5630 front &= mRsmpInFramesP2 - 1; 5631 size_t part1 = mRsmpInFramesP2 - front; 5632 if (part1 > framesIn) { 5633 part1 = framesIn; 5634 } 5635 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5636 if (mChannelCount == activeTrack->mChannelCount) { 5637 memcpy(dst, src, part1 * mFrameSize); 5638 } else if (mChannelCount == 1) { 5639 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5640 part1); 5641 } else { 5642 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 5643 (const int16_t *)src, part1); 5644 } 5645 dst += part1 * activeTrack->mFrameSize; 5646 front += part1; 5647 framesIn -= part1; 5648 } 5649 activeTrack->mRsmpInFront += framesOut; 5650 5651 } else { 5652 // resampling 5653 // FIXME framesInNeeded should really be part of resampler API, and should 5654 // depend on the SRC ratio 5655 // to keep mRsmpInBuffer full so resampler always has sufficient input 5656 size_t framesInNeeded; 5657 // FIXME only re-calculate when it changes, and optimize for common ratios 5658 // Do not precompute in/out because floating point is not associative 5659 // e.g. a*b/c != a*(b/c). 5660 const double in(mSampleRate); 5661 const double out(activeTrack->mSampleRate); 5662 framesInNeeded = ceil(framesOut * in / out) + 1; 5663 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5664 framesInNeeded, framesOut, in / out); 5665 // Although we theoretically have framesIn in circular buffer, some of those are 5666 // unreleased frames, and thus must be discounted for purpose of budgeting. 5667 size_t unreleased = activeTrack->mRsmpInUnrel; 5668 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5669 if (framesIn < framesInNeeded) { 5670 ALOGV("not enough to resample: have %u frames in but need %u in to " 5671 "produce %u out given in/out ratio of %.4g", 5672 framesIn, framesInNeeded, framesOut, in / out); 5673 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5674 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5675 if (newFramesOut == 0) { 5676 break; 5677 } 5678 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5679 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5680 framesInNeeded, newFramesOut, out / in); 5681 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5682 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5683 "given in/out ratio of %.4g", 5684 framesIn, framesInNeeded, newFramesOut, in / out); 5685 framesOut = newFramesOut; 5686 } else { 5687 ALOGV("success 1: have %u in and need %u in to produce %u out " 5688 "given in/out ratio of %.4g", 5689 framesIn, framesInNeeded, framesOut, in / out); 5690 } 5691 5692 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5693 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5694 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5695 delete[] activeTrack->mRsmpOutBuffer; 5696 // resampler always outputs stereo 5697 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5698 activeTrack->mRsmpOutFrameCount = framesOut; 5699 } 5700 5701 // resampler accumulates, but we only have one source track 5702 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5703 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5704 // FIXME how about having activeTrack implement this interface itself? 5705 activeTrack->mResamplerBufferProvider 5706 /*this*/ /* AudioBufferProvider* */); 5707 // ditherAndClamp() works as long as all buffers returned by 5708 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5709 if (activeTrack->mChannelCount == 1) { 5710 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5711 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5712 framesOut); 5713 // the resampler always outputs stereo samples: 5714 // do post stereo to mono conversion 5715 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5716 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5717 } else { 5718 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5719 activeTrack->mRsmpOutBuffer, framesOut); 5720 } 5721 // now done with mRsmpOutBuffer 5722 5723 } 5724 5725 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5726 overrun = OVERRUN_FALSE; 5727 } 5728 5729 if (activeTrack->mFramesToDrop == 0) { 5730 if (framesOut > 0) { 5731 activeTrack->mSink.frameCount = framesOut; 5732 activeTrack->releaseBuffer(&activeTrack->mSink); 5733 } 5734 } else { 5735 // FIXME could do a partial drop of framesOut 5736 if (activeTrack->mFramesToDrop > 0) { 5737 activeTrack->mFramesToDrop -= framesOut; 5738 if (activeTrack->mFramesToDrop <= 0) { 5739 activeTrack->clearSyncStartEvent(); 5740 } 5741 } else { 5742 activeTrack->mFramesToDrop += framesOut; 5743 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5744 activeTrack->mSyncStartEvent->isCancelled()) { 5745 ALOGW("Synced record %s, session %d, trigger session %d", 5746 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5747 activeTrack->sessionId(), 5748 (activeTrack->mSyncStartEvent != 0) ? 5749 activeTrack->mSyncStartEvent->triggerSession() : 0); 5750 activeTrack->clearSyncStartEvent(); 5751 } 5752 } 5753 } 5754 5755 if (framesOut == 0) { 5756 break; 5757 } 5758 } 5759 5760 switch (overrun) { 5761 case OVERRUN_TRUE: 5762 // client isn't retrieving buffers fast enough 5763 if (!activeTrack->setOverflow()) { 5764 nsecs_t now = systemTime(); 5765 // FIXME should lastWarning per track? 5766 if ((now - lastWarning) > kWarningThrottleNs) { 5767 ALOGW("RecordThread: buffer overflow"); 5768 lastWarning = now; 5769 } 5770 } 5771 break; 5772 case OVERRUN_FALSE: 5773 activeTrack->clearOverflow(); 5774 break; 5775 case OVERRUN_UNKNOWN: 5776 break; 5777 } 5778 5779 } 5780 5781unlock: 5782 // enable changes in effect chain 5783 unlockEffectChains(effectChains); 5784 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5785 } 5786 5787 standbyIfNotAlreadyInStandby(); 5788 5789 { 5790 Mutex::Autolock _l(mLock); 5791 for (size_t i = 0; i < mTracks.size(); i++) { 5792 sp<RecordTrack> track = mTracks[i]; 5793 track->invalidate(); 5794 } 5795 mActiveTracks.clear(); 5796 mActiveTracksGen++; 5797 mStartStopCond.broadcast(); 5798 } 5799 5800 releaseWakeLock(); 5801 5802 ALOGV("RecordThread %p exiting", this); 5803 return false; 5804} 5805 5806void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5807{ 5808 if (!mStandby) { 5809 inputStandBy(); 5810 mStandby = true; 5811 } 5812} 5813 5814void AudioFlinger::RecordThread::inputStandBy() 5815{ 5816 // Idle the fast capture if it's currently running 5817 if (mFastCapture != 0) { 5818 FastCaptureStateQueue *sq = mFastCapture->sq(); 5819 FastCaptureState *state = sq->begin(); 5820 if (!(state->mCommand & FastCaptureState::IDLE)) { 5821 state->mCommand = FastCaptureState::COLD_IDLE; 5822 state->mColdFutexAddr = &mFastCaptureFutex; 5823 state->mColdGen++; 5824 mFastCaptureFutex = 0; 5825 sq->end(); 5826 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5827 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5828#if 0 5829 if (kUseFastCapture == FastCapture_Dynamic) { 5830 // FIXME 5831 } 5832#endif 5833#ifdef AUDIO_WATCHDOG 5834 // FIXME 5835#endif 5836 } else { 5837 sq->end(false /*didModify*/); 5838 } 5839 } 5840 mInput->stream->common.standby(&mInput->stream->common); 5841} 5842 5843// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5844sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5845 const sp<AudioFlinger::Client>& client, 5846 uint32_t sampleRate, 5847 audio_format_t format, 5848 audio_channel_mask_t channelMask, 5849 size_t *pFrameCount, 5850 int sessionId, 5851 size_t *notificationFrames, 5852 int uid, 5853 IAudioFlinger::track_flags_t *flags, 5854 pid_t tid, 5855 status_t *status) 5856{ 5857 size_t frameCount = *pFrameCount; 5858 sp<RecordTrack> track; 5859 status_t lStatus; 5860 5861 // client expresses a preference for FAST, but we get the final say 5862 if (*flags & IAudioFlinger::TRACK_FAST) { 5863 if ( 5864 // use case: callback handler 5865 (tid != -1) && 5866 // frame count is not specified, or is exactly the pipe depth 5867 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5868 // PCM data 5869 audio_is_linear_pcm(format) && 5870 // native format 5871 (format == mFormat) && 5872 // native channel mask 5873 (channelMask == mChannelMask) && 5874 // native hardware sample rate 5875 (sampleRate == mSampleRate) && 5876 // record thread has an associated fast capture 5877 hasFastCapture() && 5878 // there are sufficient fast track slots available 5879 mFastTrackAvail 5880 ) { 5881 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5882 frameCount, mFrameCount); 5883 } else { 5884 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5885 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5886 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5887 frameCount, mFrameCount, mPipeFramesP2, 5888 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5889 hasFastCapture(), tid, mFastTrackAvail); 5890 *flags &= ~IAudioFlinger::TRACK_FAST; 5891 } 5892 } 5893 5894 // compute track buffer size in frames, and suggest the notification frame count 5895 if (*flags & IAudioFlinger::TRACK_FAST) { 5896 // fast track: frame count is exactly the pipe depth 5897 frameCount = mPipeFramesP2; 5898 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5899 *notificationFrames = mFrameCount; 5900 } else { 5901 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5902 // or 20 ms if there is a fast capture 5903 // TODO This could be a roundupRatio inline, and const 5904 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5905 * sampleRate + mSampleRate - 1) / mSampleRate; 5906 // minimum number of notification periods is at least kMinNotifications, 5907 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5908 static const size_t kMinNotifications = 3; 5909 static const uint32_t kMinMs = 30; 5910 // TODO This could be a roundupRatio inline 5911 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5912 // TODO This could be a roundupRatio inline 5913 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5914 maxNotificationFrames; 5915 const size_t minFrameCount = maxNotificationFrames * 5916 max(kMinNotifications, minNotificationsByMs); 5917 frameCount = max(frameCount, minFrameCount); 5918 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5919 *notificationFrames = maxNotificationFrames; 5920 } 5921 } 5922 *pFrameCount = frameCount; 5923 5924 lStatus = initCheck(); 5925 if (lStatus != NO_ERROR) { 5926 ALOGE("createRecordTrack_l() audio driver not initialized"); 5927 goto Exit; 5928 } 5929 5930 { // scope for mLock 5931 Mutex::Autolock _l(mLock); 5932 5933 track = new RecordTrack(this, client, sampleRate, 5934 format, channelMask, frameCount, NULL, sessionId, uid, 5935 *flags, TrackBase::TYPE_DEFAULT); 5936 5937 lStatus = track->initCheck(); 5938 if (lStatus != NO_ERROR) { 5939 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5940 // track must be cleared from the caller as the caller has the AF lock 5941 goto Exit; 5942 } 5943 mTracks.add(track); 5944 5945 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5946 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5947 mAudioFlinger->btNrecIsOff(); 5948 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5949 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5950 5951 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5952 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5953 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5954 // so ask activity manager to do this on our behalf 5955 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5956 } 5957 } 5958 5959 lStatus = NO_ERROR; 5960 5961Exit: 5962 *status = lStatus; 5963 return track; 5964} 5965 5966status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5967 AudioSystem::sync_event_t event, 5968 int triggerSession) 5969{ 5970 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5971 sp<ThreadBase> strongMe = this; 5972 status_t status = NO_ERROR; 5973 5974 if (event == AudioSystem::SYNC_EVENT_NONE) { 5975 recordTrack->clearSyncStartEvent(); 5976 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5977 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5978 triggerSession, 5979 recordTrack->sessionId(), 5980 syncStartEventCallback, 5981 recordTrack); 5982 // Sync event can be cancelled by the trigger session if the track is not in a 5983 // compatible state in which case we start record immediately 5984 if (recordTrack->mSyncStartEvent->isCancelled()) { 5985 recordTrack->clearSyncStartEvent(); 5986 } else { 5987 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5988 recordTrack->mFramesToDrop = - 5989 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5990 } 5991 } 5992 5993 { 5994 // This section is a rendezvous between binder thread executing start() and RecordThread 5995 AutoMutex lock(mLock); 5996 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5997 if (recordTrack->mState == TrackBase::PAUSING) { 5998 ALOGV("active record track PAUSING -> ACTIVE"); 5999 recordTrack->mState = TrackBase::ACTIVE; 6000 } else { 6001 ALOGV("active record track state %d", recordTrack->mState); 6002 } 6003 return status; 6004 } 6005 6006 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6007 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6008 // or using a separate command thread 6009 recordTrack->mState = TrackBase::STARTING_1; 6010 mActiveTracks.add(recordTrack); 6011 mActiveTracksGen++; 6012 status_t status = NO_ERROR; 6013 if (recordTrack->isExternalTrack()) { 6014 mLock.unlock(); 6015 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6016 mLock.lock(); 6017 // FIXME should verify that recordTrack is still in mActiveTracks 6018 if (status != NO_ERROR) { 6019 mActiveTracks.remove(recordTrack); 6020 mActiveTracksGen++; 6021 recordTrack->clearSyncStartEvent(); 6022 ALOGV("RecordThread::start error %d", status); 6023 return status; 6024 } 6025 } 6026 // Catch up with current buffer indices if thread is already running. 6027 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6028 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6029 // see previously buffered data before it called start(), but with greater risk of overrun. 6030 6031 recordTrack->mRsmpInFront = mRsmpInRear; 6032 recordTrack->mRsmpInUnrel = 0; 6033 // FIXME why reset? 6034 if (recordTrack->mResampler != NULL) { 6035 recordTrack->mResampler->reset(); 6036 } 6037 recordTrack->mState = TrackBase::STARTING_2; 6038 // signal thread to start 6039 mWaitWorkCV.broadcast(); 6040 if (mActiveTracks.indexOf(recordTrack) < 0) { 6041 ALOGV("Record failed to start"); 6042 status = BAD_VALUE; 6043 goto startError; 6044 } 6045 return status; 6046 } 6047 6048startError: 6049 if (recordTrack->isExternalTrack()) { 6050 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6051 } 6052 recordTrack->clearSyncStartEvent(); 6053 // FIXME I wonder why we do not reset the state here? 6054 return status; 6055} 6056 6057void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6058{ 6059 sp<SyncEvent> strongEvent = event.promote(); 6060 6061 if (strongEvent != 0) { 6062 sp<RefBase> ptr = strongEvent->cookie().promote(); 6063 if (ptr != 0) { 6064 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6065 recordTrack->handleSyncStartEvent(strongEvent); 6066 } 6067 } 6068} 6069 6070bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6071 ALOGV("RecordThread::stop"); 6072 AutoMutex _l(mLock); 6073 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6074 return false; 6075 } 6076 // note that threadLoop may still be processing the track at this point [without lock] 6077 recordTrack->mState = TrackBase::PAUSING; 6078 // do not wait for mStartStopCond if exiting 6079 if (exitPending()) { 6080 return true; 6081 } 6082 // FIXME incorrect usage of wait: no explicit predicate or loop 6083 mStartStopCond.wait(mLock); 6084 // if we have been restarted, recordTrack is in mActiveTracks here 6085 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6086 ALOGV("Record stopped OK"); 6087 return true; 6088 } 6089 return false; 6090} 6091 6092bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6093{ 6094 return false; 6095} 6096 6097status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6098{ 6099#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6100 if (!isValidSyncEvent(event)) { 6101 return BAD_VALUE; 6102 } 6103 6104 int eventSession = event->triggerSession(); 6105 status_t ret = NAME_NOT_FOUND; 6106 6107 Mutex::Autolock _l(mLock); 6108 6109 for (size_t i = 0; i < mTracks.size(); i++) { 6110 sp<RecordTrack> track = mTracks[i]; 6111 if (eventSession == track->sessionId()) { 6112 (void) track->setSyncEvent(event); 6113 ret = NO_ERROR; 6114 } 6115 } 6116 return ret; 6117#else 6118 return BAD_VALUE; 6119#endif 6120} 6121 6122// destroyTrack_l() must be called with ThreadBase::mLock held 6123void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6124{ 6125 track->terminate(); 6126 track->mState = TrackBase::STOPPED; 6127 // active tracks are removed by threadLoop() 6128 if (mActiveTracks.indexOf(track) < 0) { 6129 removeTrack_l(track); 6130 } 6131} 6132 6133void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6134{ 6135 mTracks.remove(track); 6136 // need anything related to effects here? 6137 if (track->isFastTrack()) { 6138 ALOG_ASSERT(!mFastTrackAvail); 6139 mFastTrackAvail = true; 6140 } 6141} 6142 6143void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6144{ 6145 dumpInternals(fd, args); 6146 dumpTracks(fd, args); 6147 dumpEffectChains(fd, args); 6148} 6149 6150void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6151{ 6152 dprintf(fd, "\nInput thread %p:\n", this); 6153 6154 if (mActiveTracks.size() > 0) { 6155 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 6156 } else { 6157 dprintf(fd, " No active record clients\n"); 6158 } 6159 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6160 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6161 6162 dumpBase(fd, args); 6163} 6164 6165void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6166{ 6167 const size_t SIZE = 256; 6168 char buffer[SIZE]; 6169 String8 result; 6170 6171 size_t numtracks = mTracks.size(); 6172 size_t numactive = mActiveTracks.size(); 6173 size_t numactiveseen = 0; 6174 dprintf(fd, " %d Tracks", numtracks); 6175 if (numtracks) { 6176 dprintf(fd, " of which %d are active\n", numactive); 6177 RecordTrack::appendDumpHeader(result); 6178 for (size_t i = 0; i < numtracks ; ++i) { 6179 sp<RecordTrack> track = mTracks[i]; 6180 if (track != 0) { 6181 bool active = mActiveTracks.indexOf(track) >= 0; 6182 if (active) { 6183 numactiveseen++; 6184 } 6185 track->dump(buffer, SIZE, active); 6186 result.append(buffer); 6187 } 6188 } 6189 } else { 6190 dprintf(fd, "\n"); 6191 } 6192 6193 if (numactiveseen != numactive) { 6194 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6195 " not in the track list\n"); 6196 result.append(buffer); 6197 RecordTrack::appendDumpHeader(result); 6198 for (size_t i = 0; i < numactive; ++i) { 6199 sp<RecordTrack> track = mActiveTracks[i]; 6200 if (mTracks.indexOf(track) < 0) { 6201 track->dump(buffer, SIZE, true); 6202 result.append(buffer); 6203 } 6204 } 6205 6206 } 6207 write(fd, result.string(), result.size()); 6208} 6209 6210// AudioBufferProvider interface 6211status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6212 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6213{ 6214 RecordTrack *activeTrack = mRecordTrack; 6215 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 6216 if (threadBase == 0) { 6217 buffer->frameCount = 0; 6218 buffer->raw = NULL; 6219 return NOT_ENOUGH_DATA; 6220 } 6221 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6222 int32_t rear = recordThread->mRsmpInRear; 6223 int32_t front = activeTrack->mRsmpInFront; 6224 ssize_t filled = rear - front; 6225 // FIXME should not be P2 (don't want to increase latency) 6226 // FIXME if client not keeping up, discard 6227 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6228 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6229 front &= recordThread->mRsmpInFramesP2 - 1; 6230 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6231 if (part1 > (size_t) filled) { 6232 part1 = filled; 6233 } 6234 size_t ask = buffer->frameCount; 6235 ALOG_ASSERT(ask > 0); 6236 if (part1 > ask) { 6237 part1 = ask; 6238 } 6239 if (part1 == 0) { 6240 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 6241 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 6242 buffer->raw = NULL; 6243 buffer->frameCount = 0; 6244 activeTrack->mRsmpInUnrel = 0; 6245 return NOT_ENOUGH_DATA; 6246 } 6247 6248 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6249 buffer->frameCount = part1; 6250 activeTrack->mRsmpInUnrel = part1; 6251 return NO_ERROR; 6252} 6253 6254// AudioBufferProvider interface 6255void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6256 AudioBufferProvider::Buffer* buffer) 6257{ 6258 RecordTrack *activeTrack = mRecordTrack; 6259 size_t stepCount = buffer->frameCount; 6260 if (stepCount == 0) { 6261 return; 6262 } 6263 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 6264 activeTrack->mRsmpInUnrel -= stepCount; 6265 activeTrack->mRsmpInFront += stepCount; 6266 buffer->raw = NULL; 6267 buffer->frameCount = 0; 6268} 6269 6270bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6271 status_t& status) 6272{ 6273 bool reconfig = false; 6274 6275 status = NO_ERROR; 6276 6277 audio_format_t reqFormat = mFormat; 6278 uint32_t samplingRate = mSampleRate; 6279 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6280 6281 AudioParameter param = AudioParameter(keyValuePair); 6282 int value; 6283 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6284 // channel count change can be requested. Do we mandate the first client defines the 6285 // HAL sampling rate and channel count or do we allow changes on the fly? 6286 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6287 samplingRate = value; 6288 reconfig = true; 6289 } 6290 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6291 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 6292 status = BAD_VALUE; 6293 } else { 6294 reqFormat = (audio_format_t) value; 6295 reconfig = true; 6296 } 6297 } 6298 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6299 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6300 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6301 status = BAD_VALUE; 6302 } else { 6303 channelMask = mask; 6304 reconfig = true; 6305 } 6306 } 6307 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6308 // do not accept frame count changes if tracks are open as the track buffer 6309 // size depends on frame count and correct behavior would not be guaranteed 6310 // if frame count is changed after track creation 6311 if (mActiveTracks.size() > 0) { 6312 status = INVALID_OPERATION; 6313 } else { 6314 reconfig = true; 6315 } 6316 } 6317 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6318 // forward device change to effects that have requested to be 6319 // aware of attached audio device. 6320 for (size_t i = 0; i < mEffectChains.size(); i++) { 6321 mEffectChains[i]->setDevice_l(value); 6322 } 6323 6324 // store input device and output device but do not forward output device to audio HAL. 6325 // Note that status is ignored by the caller for output device 6326 // (see AudioFlinger::setParameters() 6327 if (audio_is_output_devices(value)) { 6328 mOutDevice = value; 6329 status = BAD_VALUE; 6330 } else { 6331 mInDevice = value; 6332 // disable AEC and NS if the device is a BT SCO headset supporting those 6333 // pre processings 6334 if (mTracks.size() > 0) { 6335 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6336 mAudioFlinger->btNrecIsOff(); 6337 for (size_t i = 0; i < mTracks.size(); i++) { 6338 sp<RecordTrack> track = mTracks[i]; 6339 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6340 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6341 } 6342 } 6343 } 6344 } 6345 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6346 mAudioSource != (audio_source_t)value) { 6347 // forward device change to effects that have requested to be 6348 // aware of attached audio device. 6349 for (size_t i = 0; i < mEffectChains.size(); i++) { 6350 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6351 } 6352 mAudioSource = (audio_source_t)value; 6353 } 6354 6355 if (status == NO_ERROR) { 6356 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6357 keyValuePair.string()); 6358 if (status == INVALID_OPERATION) { 6359 inputStandBy(); 6360 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6361 keyValuePair.string()); 6362 } 6363 if (reconfig) { 6364 if (status == BAD_VALUE && 6365 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6366 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6367 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6368 <= (2 * samplingRate)) && 6369 audio_channel_count_from_in_mask( 6370 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6371 (channelMask == AUDIO_CHANNEL_IN_MONO || 6372 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6373 status = NO_ERROR; 6374 } 6375 if (status == NO_ERROR) { 6376 readInputParameters_l(); 6377 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6378 } 6379 } 6380 } 6381 6382 return reconfig; 6383} 6384 6385String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6386{ 6387 Mutex::Autolock _l(mLock); 6388 if (initCheck() != NO_ERROR) { 6389 return String8(); 6390 } 6391 6392 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6393 const String8 out_s8(s); 6394 free(s); 6395 return out_s8; 6396} 6397 6398void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6399 AudioSystem::OutputDescriptor desc; 6400 const void *param2 = NULL; 6401 6402 switch (event) { 6403 case AudioSystem::INPUT_OPENED: 6404 case AudioSystem::INPUT_CONFIG_CHANGED: 6405 desc.channelMask = mChannelMask; 6406 desc.samplingRate = mSampleRate; 6407 desc.format = mFormat; 6408 desc.frameCount = mFrameCount; 6409 desc.latency = 0; 6410 param2 = &desc; 6411 break; 6412 6413 case AudioSystem::INPUT_CLOSED: 6414 default: 6415 break; 6416 } 6417 mAudioFlinger->audioConfigChanged(event, mId, param2); 6418} 6419 6420void AudioFlinger::RecordThread::readInputParameters_l() 6421{ 6422 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6423 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6424 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6425 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6426 mFormat = mHALFormat; 6427 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6428 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6429 } 6430 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6431 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6432 mFrameCount = mBufferSize / mFrameSize; 6433 // This is the formula for calculating the temporary buffer size. 6434 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6435 // 1 full output buffer, regardless of the alignment of the available input. 6436 // The value is somewhat arbitrary, and could probably be even larger. 6437 // A larger value should allow more old data to be read after a track calls start(), 6438 // without increasing latency. 6439 mRsmpInFrames = mFrameCount * 7; 6440 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6441 delete[] mRsmpInBuffer; 6442 6443 // TODO optimize audio capture buffer sizes ... 6444 // Here we calculate the size of the sliding buffer used as a source 6445 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6446 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6447 // be better to have it derived from the pipe depth in the long term. 6448 // The current value is higher than necessary. However it should not add to latency. 6449 6450 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6451 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6452 6453 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6454 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6455} 6456 6457uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6458{ 6459 Mutex::Autolock _l(mLock); 6460 if (initCheck() != NO_ERROR) { 6461 return 0; 6462 } 6463 6464 return mInput->stream->get_input_frames_lost(mInput->stream); 6465} 6466 6467uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6468{ 6469 Mutex::Autolock _l(mLock); 6470 uint32_t result = 0; 6471 if (getEffectChain_l(sessionId) != 0) { 6472 result = EFFECT_SESSION; 6473 } 6474 6475 for (size_t i = 0; i < mTracks.size(); ++i) { 6476 if (sessionId == mTracks[i]->sessionId()) { 6477 result |= TRACK_SESSION; 6478 break; 6479 } 6480 } 6481 6482 return result; 6483} 6484 6485KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6486{ 6487 KeyedVector<int, bool> ids; 6488 Mutex::Autolock _l(mLock); 6489 for (size_t j = 0; j < mTracks.size(); ++j) { 6490 sp<RecordThread::RecordTrack> track = mTracks[j]; 6491 int sessionId = track->sessionId(); 6492 if (ids.indexOfKey(sessionId) < 0) { 6493 ids.add(sessionId, true); 6494 } 6495 } 6496 return ids; 6497} 6498 6499AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6500{ 6501 Mutex::Autolock _l(mLock); 6502 AudioStreamIn *input = mInput; 6503 mInput = NULL; 6504 return input; 6505} 6506 6507// this method must always be called either with ThreadBase mLock held or inside the thread loop 6508audio_stream_t* AudioFlinger::RecordThread::stream() const 6509{ 6510 if (mInput == NULL) { 6511 return NULL; 6512 } 6513 return &mInput->stream->common; 6514} 6515 6516status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6517{ 6518 // only one chain per input thread 6519 if (mEffectChains.size() != 0) { 6520 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6521 return INVALID_OPERATION; 6522 } 6523 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6524 chain->setThread(this); 6525 chain->setInBuffer(NULL); 6526 chain->setOutBuffer(NULL); 6527 6528 checkSuspendOnAddEffectChain_l(chain); 6529 6530 // make sure enabled pre processing effects state is communicated to the HAL as we 6531 // just moved them to a new input stream. 6532 chain->syncHalEffectsState(); 6533 6534 mEffectChains.add(chain); 6535 6536 return NO_ERROR; 6537} 6538 6539size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6540{ 6541 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6542 ALOGW_IF(mEffectChains.size() != 1, 6543 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6544 chain.get(), mEffectChains.size(), this); 6545 if (mEffectChains.size() == 1) { 6546 mEffectChains.removeAt(0); 6547 } 6548 return 0; 6549} 6550 6551status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6552 audio_patch_handle_t *handle) 6553{ 6554 status_t status = NO_ERROR; 6555 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6556 // store new device and send to effects 6557 mInDevice = patch->sources[0].ext.device.type; 6558 for (size_t i = 0; i < mEffectChains.size(); i++) { 6559 mEffectChains[i]->setDevice_l(mInDevice); 6560 } 6561 6562 // disable AEC and NS if the device is a BT SCO headset supporting those 6563 // pre processings 6564 if (mTracks.size() > 0) { 6565 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6566 mAudioFlinger->btNrecIsOff(); 6567 for (size_t i = 0; i < mTracks.size(); i++) { 6568 sp<RecordTrack> track = mTracks[i]; 6569 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6570 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6571 } 6572 } 6573 6574 // store new source and send to effects 6575 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6576 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6577 for (size_t i = 0; i < mEffectChains.size(); i++) { 6578 mEffectChains[i]->setAudioSource_l(mAudioSource); 6579 } 6580 } 6581 6582 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6583 status = hwDevice->create_audio_patch(hwDevice, 6584 patch->num_sources, 6585 patch->sources, 6586 patch->num_sinks, 6587 patch->sinks, 6588 handle); 6589 } else { 6590 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6591 } 6592 return status; 6593} 6594 6595status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6596{ 6597 status_t status = NO_ERROR; 6598 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6599 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6600 status = hwDevice->release_audio_patch(hwDevice, handle); 6601 } else { 6602 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6603 } 6604 return status; 6605} 6606 6607void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6608{ 6609 Mutex::Autolock _l(mLock); 6610 mTracks.add(record); 6611} 6612 6613void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6614{ 6615 Mutex::Autolock _l(mLock); 6616 destroyTrack_l(record); 6617} 6618 6619void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6620{ 6621 ThreadBase::getAudioPortConfig(config); 6622 config->role = AUDIO_PORT_ROLE_SINK; 6623 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6624 config->ext.mix.usecase.source = mAudioSource; 6625} 6626 6627} // namespace android 6628