Threads.cpp revision 0b89bc0d285b8fd4798df1ff0ba9f93851a3bd48
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/primitives.h> 40#include <audio_utils/format.h> 41#include <audio_utils/minifloat.h> 42 43// NBAIO implementations 44#include <media/nbaio/AudioStreamInSource.h> 45#include <media/nbaio/AudioStreamOutSink.h> 46#include <media/nbaio/MonoPipe.h> 47#include <media/nbaio/MonoPipeReader.h> 48#include <media/nbaio/Pipe.h> 49#include <media/nbaio/PipeReader.h> 50#include <media/nbaio/SourceAudioBufferProvider.h> 51 52#include <powermanager/PowerManager.h> 53 54#include <common_time/cc_helper.h> 55#include <common_time/local_clock.h> 56 57#include "AudioFlinger.h" 58#include "AudioMixer.h" 59#include "FastMixer.h" 60#include "FastCapture.h" 61#include "ServiceUtilities.h" 62#include "SchedulingPolicyService.h" 63 64#ifdef ADD_BATTERY_DATA 65#include <media/IMediaPlayerService.h> 66#include <media/IMediaDeathNotifier.h> 67#endif 68 69#ifdef DEBUG_CPU_USAGE 70#include <cpustats/CentralTendencyStatistics.h> 71#include <cpustats/ThreadCpuUsage.h> 72#endif 73 74// ---------------------------------------------------------------------------- 75 76// Note: the following macro is used for extremely verbose logging message. In 77// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80// turned on. Do not uncomment the #def below unless you really know what you 81// are doing and want to see all of the extremely verbose messages. 82//#define VERY_VERY_VERBOSE_LOGGING 83#ifdef VERY_VERY_VERBOSE_LOGGING 84#define ALOGVV ALOGV 85#else 86#define ALOGVV(a...) do { } while(0) 87#endif 88 89#define max(a, b) ((a) > (b) ? (a) : (b)) 90 91namespace android { 92 93// retry counts for buffer fill timeout 94// 50 * ~20msecs = 1 second 95static const int8_t kMaxTrackRetries = 50; 96static const int8_t kMaxTrackStartupRetries = 50; 97// allow less retry attempts on direct output thread. 98// direct outputs can be a scarce resource in audio hardware and should 99// be released as quickly as possible. 100static const int8_t kMaxTrackRetriesDirect = 2; 101 102// don't warn about blocked writes or record buffer overflows more often than this 103static const nsecs_t kWarningThrottleNs = seconds(5); 104 105// RecordThread loop sleep time upon application overrun or audio HAL read error 106static const int kRecordThreadSleepUs = 5000; 107 108// maximum time to wait in sendConfigEvent_l() for a status to be received 109static const nsecs_t kConfigEventTimeoutNs = seconds(2); 110 111// minimum sleep time for the mixer thread loop when tracks are active but in underrun 112static const uint32_t kMinThreadSleepTimeUs = 5000; 113// maximum divider applied to the active sleep time in the mixer thread loop 114static const uint32_t kMaxThreadSleepTimeShift = 2; 115 116// minimum normal sink buffer size, expressed in milliseconds rather than frames 117static const uint32_t kMinNormalSinkBufferSizeMs = 20; 118// maximum normal sink buffer size 119static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 120 121// Offloaded output thread standby delay: allows track transition without going to standby 122static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 123 124// Whether to use fast mixer 125static const enum { 126 FastMixer_Never, // never initialize or use: for debugging only 127 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 128 // normal mixer multiplier is 1 129 FastMixer_Static, // initialize if needed, then use all the time if initialized, 130 // multiplier is calculated based on min & max normal mixer buffer size 131 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 132 // multiplier is calculated based on min & max normal mixer buffer size 133 // FIXME for FastMixer_Dynamic: 134 // Supporting this option will require fixing HALs that can't handle large writes. 135 // For example, one HAL implementation returns an error from a large write, 136 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 137 // We could either fix the HAL implementations, or provide a wrapper that breaks 138 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 139} kUseFastMixer = FastMixer_Static; 140 141// Whether to use fast capture 142static const enum { 143 FastCapture_Never, // never initialize or use: for debugging only 144 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 145 FastCapture_Static, // initialize if needed, then use all the time if initialized 146} kUseFastCapture = FastCapture_Static; 147 148// Priorities for requestPriority 149static const int kPriorityAudioApp = 2; 150static const int kPriorityFastMixer = 3; 151static const int kPriorityFastCapture = 3; 152 153// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 154// for the track. The client then sub-divides this into smaller buffers for its use. 155// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 156// So for now we just assume that client is double-buffered for fast tracks. 157// FIXME It would be better for client to tell AudioFlinger the value of N, 158// so AudioFlinger could allocate the right amount of memory. 159// See the client's minBufCount and mNotificationFramesAct calculations for details. 160 161// This is the default value, if not specified by property. 162static const int kFastTrackMultiplier = 2; 163 164// The minimum and maximum allowed values 165static const int kFastTrackMultiplierMin = 1; 166static const int kFastTrackMultiplierMax = 2; 167 168// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 169static int sFastTrackMultiplier = kFastTrackMultiplier; 170 171// See Thread::readOnlyHeap(). 172// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 173// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 174// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 175static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 176 177// ---------------------------------------------------------------------------- 178 179static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 180 181static void sFastTrackMultiplierInit() 182{ 183 char value[PROPERTY_VALUE_MAX]; 184 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 185 char *endptr; 186 unsigned long ul = strtoul(value, &endptr, 0); 187 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 188 sFastTrackMultiplier = (int) ul; 189 } 190 } 191} 192 193// ---------------------------------------------------------------------------- 194 195#ifdef ADD_BATTERY_DATA 196// To collect the amplifier usage 197static void addBatteryData(uint32_t params) { 198 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 199 if (service == NULL) { 200 // it already logged 201 return; 202 } 203 204 service->addBatteryData(params); 205} 206#endif 207 208 209// ---------------------------------------------------------------------------- 210// CPU Stats 211// ---------------------------------------------------------------------------- 212 213class CpuStats { 214public: 215 CpuStats(); 216 void sample(const String8 &title); 217#ifdef DEBUG_CPU_USAGE 218private: 219 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 220 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 221 222 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 223 224 int mCpuNum; // thread's current CPU number 225 int mCpukHz; // frequency of thread's current CPU in kHz 226#endif 227}; 228 229CpuStats::CpuStats() 230#ifdef DEBUG_CPU_USAGE 231 : mCpuNum(-1), mCpukHz(-1) 232#endif 233{ 234} 235 236void CpuStats::sample(const String8 &title 237#ifndef DEBUG_CPU_USAGE 238 __unused 239#endif 240 ) { 241#ifdef DEBUG_CPU_USAGE 242 // get current thread's delta CPU time in wall clock ns 243 double wcNs; 244 bool valid = mCpuUsage.sampleAndEnable(wcNs); 245 246 // record sample for wall clock statistics 247 if (valid) { 248 mWcStats.sample(wcNs); 249 } 250 251 // get the current CPU number 252 int cpuNum = sched_getcpu(); 253 254 // get the current CPU frequency in kHz 255 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 256 257 // check if either CPU number or frequency changed 258 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 259 mCpuNum = cpuNum; 260 mCpukHz = cpukHz; 261 // ignore sample for purposes of cycles 262 valid = false; 263 } 264 265 // if no change in CPU number or frequency, then record sample for cycle statistics 266 if (valid && mCpukHz > 0) { 267 double cycles = wcNs * cpukHz * 0.000001; 268 mHzStats.sample(cycles); 269 } 270 271 unsigned n = mWcStats.n(); 272 // mCpuUsage.elapsed() is expensive, so don't call it every loop 273 if ((n & 127) == 1) { 274 long long elapsed = mCpuUsage.elapsed(); 275 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 276 double perLoop = elapsed / (double) n; 277 double perLoop100 = perLoop * 0.01; 278 double perLoop1k = perLoop * 0.001; 279 double mean = mWcStats.mean(); 280 double stddev = mWcStats.stddev(); 281 double minimum = mWcStats.minimum(); 282 double maximum = mWcStats.maximum(); 283 double meanCycles = mHzStats.mean(); 284 double stddevCycles = mHzStats.stddev(); 285 double minCycles = mHzStats.minimum(); 286 double maxCycles = mHzStats.maximum(); 287 mCpuUsage.resetElapsed(); 288 mWcStats.reset(); 289 mHzStats.reset(); 290 ALOGD("CPU usage for %s over past %.1f secs\n" 291 " (%u mixer loops at %.1f mean ms per loop):\n" 292 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 293 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 294 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 295 title.string(), 296 elapsed * .000000001, n, perLoop * .000001, 297 mean * .001, 298 stddev * .001, 299 minimum * .001, 300 maximum * .001, 301 mean / perLoop100, 302 stddev / perLoop100, 303 minimum / perLoop100, 304 maximum / perLoop100, 305 meanCycles / perLoop1k, 306 stddevCycles / perLoop1k, 307 minCycles / perLoop1k, 308 maxCycles / perLoop1k); 309 310 } 311 } 312#endif 313}; 314 315// ---------------------------------------------------------------------------- 316// ThreadBase 317// ---------------------------------------------------------------------------- 318 319// static 320const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 321{ 322 switch (type) { 323 case MIXER: 324 return "MIXER"; 325 case DIRECT: 326 return "DIRECT"; 327 case DUPLICATING: 328 return "DUPLICATING"; 329 case RECORD: 330 return "RECORD"; 331 case OFFLOAD: 332 return "OFFLOAD"; 333 default: 334 return "unknown"; 335 } 336} 337 338String8 devicesToString(audio_devices_t devices) 339{ 340 static const struct mapping { 341 audio_devices_t mDevices; 342 const char * mString; 343 } mappingsOut[] = { 344 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 345 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 346 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 347 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 348 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 349 AUDIO_DEVICE_NONE, "NONE", // must be last 350 }, mappingsIn[] = { 351 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 352 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 353 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 354 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 355 AUDIO_DEVICE_NONE, "NONE", // must be last 356 }; 357 String8 result; 358 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 359 const mapping *entry; 360 if (devices & AUDIO_DEVICE_BIT_IN) { 361 devices &= ~AUDIO_DEVICE_BIT_IN; 362 entry = mappingsIn; 363 } else { 364 entry = mappingsOut; 365 } 366 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 367 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 368 if (devices & entry->mDevices) { 369 if (!result.isEmpty()) { 370 result.append("|"); 371 } 372 result.append(entry->mString); 373 } 374 } 375 if (devices & ~allDevices) { 376 if (!result.isEmpty()) { 377 result.append("|"); 378 } 379 result.appendFormat("0x%X", devices & ~allDevices); 380 } 381 if (result.isEmpty()) { 382 result.append(entry->mString); 383 } 384 return result; 385} 386 387String8 inputFlagsToString(audio_input_flags_t flags) 388{ 389 static const struct mapping { 390 audio_input_flags_t mFlag; 391 const char * mString; 392 } mappings[] = { 393 AUDIO_INPUT_FLAG_FAST, "FAST", 394 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 395 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 396 }; 397 String8 result; 398 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 399 const mapping *entry; 400 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 401 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 402 if (flags & entry->mFlag) { 403 if (!result.isEmpty()) { 404 result.append("|"); 405 } 406 result.append(entry->mString); 407 } 408 } 409 if (flags & ~allFlags) { 410 if (!result.isEmpty()) { 411 result.append("|"); 412 } 413 result.appendFormat("0x%X", flags & ~allFlags); 414 } 415 if (result.isEmpty()) { 416 result.append(entry->mString); 417 } 418 return result; 419} 420 421String8 outputFlagsToString(audio_output_flags_t flags) 422{ 423 static const struct mapping { 424 audio_output_flags_t mFlag; 425 const char * mString; 426 } mappings[] = { 427 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 428 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 429 AUDIO_OUTPUT_FLAG_FAST, "FAST", 430 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 431 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 432 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 433 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 434 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 435 }; 436 String8 result; 437 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 438 const mapping *entry; 439 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 440 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 441 if (flags & entry->mFlag) { 442 if (!result.isEmpty()) { 443 result.append("|"); 444 } 445 result.append(entry->mString); 446 } 447 } 448 if (flags & ~allFlags) { 449 if (!result.isEmpty()) { 450 result.append("|"); 451 } 452 result.appendFormat("0x%X", flags & ~allFlags); 453 } 454 if (result.isEmpty()) { 455 result.append(entry->mString); 456 } 457 return result; 458} 459 460const char *sourceToString(audio_source_t source) 461{ 462 switch (source) { 463 case AUDIO_SOURCE_DEFAULT: return "default"; 464 case AUDIO_SOURCE_MIC: return "mic"; 465 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 466 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 467 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 468 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 469 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 470 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 471 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 472 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 473 case AUDIO_SOURCE_HOTWORD: return "hotword"; 474 default: return "unknown"; 475 } 476} 477 478AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 479 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 480 : Thread(false /*canCallJava*/), 481 mType(type), 482 mAudioFlinger(audioFlinger), 483 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 484 // are set by PlaybackThread::readOutputParameters_l() or 485 // RecordThread::readInputParameters_l() 486 //FIXME: mStandby should be true here. Is this some kind of hack? 487 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 488 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 489 // mName will be set by concrete (non-virtual) subclass 490 mDeathRecipient(new PMDeathRecipient(this)) 491{ 492} 493 494AudioFlinger::ThreadBase::~ThreadBase() 495{ 496 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 497 mConfigEvents.clear(); 498 499 // do not lock the mutex in destructor 500 releaseWakeLock_l(); 501 if (mPowerManager != 0) { 502 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 503 binder->unlinkToDeath(mDeathRecipient); 504 } 505} 506 507status_t AudioFlinger::ThreadBase::readyToRun() 508{ 509 status_t status = initCheck(); 510 if (status == NO_ERROR) { 511 ALOGI("AudioFlinger's thread %p ready to run", this); 512 } else { 513 ALOGE("No working audio driver found."); 514 } 515 return status; 516} 517 518void AudioFlinger::ThreadBase::exit() 519{ 520 ALOGV("ThreadBase::exit"); 521 // do any cleanup required for exit to succeed 522 preExit(); 523 { 524 // This lock prevents the following race in thread (uniprocessor for illustration): 525 // if (!exitPending()) { 526 // // context switch from here to exit() 527 // // exit() calls requestExit(), what exitPending() observes 528 // // exit() calls signal(), which is dropped since no waiters 529 // // context switch back from exit() to here 530 // mWaitWorkCV.wait(...); 531 // // now thread is hung 532 // } 533 AutoMutex lock(mLock); 534 requestExit(); 535 mWaitWorkCV.broadcast(); 536 } 537 // When Thread::requestExitAndWait is made virtual and this method is renamed to 538 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 539 requestExitAndWait(); 540} 541 542status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 543{ 544 status_t status; 545 546 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 547 Mutex::Autolock _l(mLock); 548 549 return sendSetParameterConfigEvent_l(keyValuePairs); 550} 551 552// sendConfigEvent_l() must be called with ThreadBase::mLock held 553// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 554status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 555{ 556 status_t status = NO_ERROR; 557 558 mConfigEvents.add(event); 559 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 560 mWaitWorkCV.signal(); 561 mLock.unlock(); 562 { 563 Mutex::Autolock _l(event->mLock); 564 while (event->mWaitStatus) { 565 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 566 event->mStatus = TIMED_OUT; 567 event->mWaitStatus = false; 568 } 569 } 570 status = event->mStatus; 571 } 572 mLock.lock(); 573 return status; 574} 575 576void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 577{ 578 Mutex::Autolock _l(mLock); 579 sendIoConfigEvent_l(event, param); 580} 581 582// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 583void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 584{ 585 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 586 sendConfigEvent_l(configEvent); 587} 588 589// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 590void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 591{ 592 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 593 sendConfigEvent_l(configEvent); 594} 595 596// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 597status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 598{ 599 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 600 return sendConfigEvent_l(configEvent); 601} 602 603status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 604 const struct audio_patch *patch, 605 audio_patch_handle_t *handle) 606{ 607 Mutex::Autolock _l(mLock); 608 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 609 status_t status = sendConfigEvent_l(configEvent); 610 if (status == NO_ERROR) { 611 CreateAudioPatchConfigEventData *data = 612 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 613 *handle = data->mHandle; 614 } 615 return status; 616} 617 618status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 619 const audio_patch_handle_t handle) 620{ 621 Mutex::Autolock _l(mLock); 622 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 623 return sendConfigEvent_l(configEvent); 624} 625 626 627// post condition: mConfigEvents.isEmpty() 628void AudioFlinger::ThreadBase::processConfigEvents_l() 629{ 630 bool configChanged = false; 631 632 while (!mConfigEvents.isEmpty()) { 633 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 634 sp<ConfigEvent> event = mConfigEvents[0]; 635 mConfigEvents.removeAt(0); 636 switch (event->mType) { 637 case CFG_EVENT_PRIO: { 638 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 639 // FIXME Need to understand why this has to be done asynchronously 640 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 641 true /*asynchronous*/); 642 if (err != 0) { 643 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 644 data->mPrio, data->mPid, data->mTid, err); 645 } 646 } break; 647 case CFG_EVENT_IO: { 648 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 649 audioConfigChanged(data->mEvent, data->mParam); 650 } break; 651 case CFG_EVENT_SET_PARAMETER: { 652 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 653 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 654 configChanged = true; 655 } 656 } break; 657 case CFG_EVENT_CREATE_AUDIO_PATCH: { 658 CreateAudioPatchConfigEventData *data = 659 (CreateAudioPatchConfigEventData *)event->mData.get(); 660 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 661 } break; 662 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 663 ReleaseAudioPatchConfigEventData *data = 664 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 665 event->mStatus = releaseAudioPatch_l(data->mHandle); 666 } break; 667 default: 668 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 669 break; 670 } 671 { 672 Mutex::Autolock _l(event->mLock); 673 if (event->mWaitStatus) { 674 event->mWaitStatus = false; 675 event->mCond.signal(); 676 } 677 } 678 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 679 } 680 681 if (configChanged) { 682 cacheParameters_l(); 683 } 684} 685 686String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 687 String8 s; 688 if (output) { 689 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 690 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 691 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 692 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 693 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 694 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 695 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 696 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 697 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 698 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 699 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 700 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 701 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 702 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 703 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 704 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 705 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 706 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 707 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 708 } else { 709 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 710 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 711 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 712 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 713 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 714 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 715 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 716 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 717 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 718 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 719 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 720 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 721 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 722 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 723 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 724 } 725 int len = s.length(); 726 if (s.length() > 2) { 727 char *str = s.lockBuffer(len); 728 s.unlockBuffer(len - 2); 729 } 730 return s; 731} 732 733void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 734{ 735 const size_t SIZE = 256; 736 char buffer[SIZE]; 737 String8 result; 738 739 bool locked = AudioFlinger::dumpTryLock(mLock); 740 if (!locked) { 741 dprintf(fd, "thread %p may be deadlocked\n", this); 742 } 743 744 dprintf(fd, " Thread name: %s\n", mThreadName); 745 dprintf(fd, " I/O handle: %d\n", mId); 746 dprintf(fd, " TID: %d\n", getTid()); 747 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 748 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 749 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 750 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 751 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 752 dprintf(fd, " Channel count: %u\n", mChannelCount); 753 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 754 channelMaskToString(mChannelMask, mType != RECORD).string()); 755 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 756 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 757 dprintf(fd, " Pending config events:"); 758 size_t numConfig = mConfigEvents.size(); 759 if (numConfig) { 760 for (size_t i = 0; i < numConfig; i++) { 761 mConfigEvents[i]->dump(buffer, SIZE); 762 dprintf(fd, "\n %s", buffer); 763 } 764 dprintf(fd, "\n"); 765 } else { 766 dprintf(fd, " none\n"); 767 } 768 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 769 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 770 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 771 772 if (locked) { 773 mLock.unlock(); 774 } 775} 776 777void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 778{ 779 const size_t SIZE = 256; 780 char buffer[SIZE]; 781 String8 result; 782 783 size_t numEffectChains = mEffectChains.size(); 784 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 785 write(fd, buffer, strlen(buffer)); 786 787 for (size_t i = 0; i < numEffectChains; ++i) { 788 sp<EffectChain> chain = mEffectChains[i]; 789 if (chain != 0) { 790 chain->dump(fd, args); 791 } 792 } 793} 794 795void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 796{ 797 Mutex::Autolock _l(mLock); 798 acquireWakeLock_l(uid); 799} 800 801String16 AudioFlinger::ThreadBase::getWakeLockTag() 802{ 803 switch (mType) { 804 case MIXER: 805 return String16("AudioMix"); 806 case DIRECT: 807 return String16("AudioDirectOut"); 808 case DUPLICATING: 809 return String16("AudioDup"); 810 case RECORD: 811 return String16("AudioIn"); 812 case OFFLOAD: 813 return String16("AudioOffload"); 814 default: 815 ALOG_ASSERT(false); 816 return String16("AudioUnknown"); 817 } 818} 819 820void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 821{ 822 getPowerManager_l(); 823 if (mPowerManager != 0) { 824 sp<IBinder> binder = new BBinder(); 825 status_t status; 826 if (uid >= 0) { 827 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 828 binder, 829 getWakeLockTag(), 830 String16("media"), 831 uid, 832 true /* FIXME force oneway contrary to .aidl */); 833 } else { 834 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 835 binder, 836 getWakeLockTag(), 837 String16("media"), 838 true /* FIXME force oneway contrary to .aidl */); 839 } 840 if (status == NO_ERROR) { 841 mWakeLockToken = binder; 842 } 843 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 844 } 845} 846 847void AudioFlinger::ThreadBase::releaseWakeLock() 848{ 849 Mutex::Autolock _l(mLock); 850 releaseWakeLock_l(); 851} 852 853void AudioFlinger::ThreadBase::releaseWakeLock_l() 854{ 855 if (mWakeLockToken != 0) { 856 ALOGV("releaseWakeLock_l() %s", mThreadName); 857 if (mPowerManager != 0) { 858 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 859 true /* FIXME force oneway contrary to .aidl */); 860 } 861 mWakeLockToken.clear(); 862 } 863} 864 865void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 866 Mutex::Autolock _l(mLock); 867 updateWakeLockUids_l(uids); 868} 869 870void AudioFlinger::ThreadBase::getPowerManager_l() { 871 872 if (mPowerManager == 0) { 873 // use checkService() to avoid blocking if power service is not up yet 874 sp<IBinder> binder = 875 defaultServiceManager()->checkService(String16("power")); 876 if (binder == 0) { 877 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 878 } else { 879 mPowerManager = interface_cast<IPowerManager>(binder); 880 binder->linkToDeath(mDeathRecipient); 881 } 882 } 883} 884 885void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 886 887 getPowerManager_l(); 888 if (mWakeLockToken == NULL) { 889 ALOGE("no wake lock to update!"); 890 return; 891 } 892 if (mPowerManager != 0) { 893 sp<IBinder> binder = new BBinder(); 894 status_t status; 895 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 896 true /* FIXME force oneway contrary to .aidl */); 897 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 898 } 899} 900 901void AudioFlinger::ThreadBase::clearPowerManager() 902{ 903 Mutex::Autolock _l(mLock); 904 releaseWakeLock_l(); 905 mPowerManager.clear(); 906} 907 908void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 909{ 910 sp<ThreadBase> thread = mThread.promote(); 911 if (thread != 0) { 912 thread->clearPowerManager(); 913 } 914 ALOGW("power manager service died !!!"); 915} 916 917void AudioFlinger::ThreadBase::setEffectSuspended( 918 const effect_uuid_t *type, bool suspend, int sessionId) 919{ 920 Mutex::Autolock _l(mLock); 921 setEffectSuspended_l(type, suspend, sessionId); 922} 923 924void AudioFlinger::ThreadBase::setEffectSuspended_l( 925 const effect_uuid_t *type, bool suspend, int sessionId) 926{ 927 sp<EffectChain> chain = getEffectChain_l(sessionId); 928 if (chain != 0) { 929 if (type != NULL) { 930 chain->setEffectSuspended_l(type, suspend); 931 } else { 932 chain->setEffectSuspendedAll_l(suspend); 933 } 934 } 935 936 updateSuspendedSessions_l(type, suspend, sessionId); 937} 938 939void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 940{ 941 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 942 if (index < 0) { 943 return; 944 } 945 946 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 947 mSuspendedSessions.valueAt(index); 948 949 for (size_t i = 0; i < sessionEffects.size(); i++) { 950 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 951 for (int j = 0; j < desc->mRefCount; j++) { 952 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 953 chain->setEffectSuspendedAll_l(true); 954 } else { 955 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 956 desc->mType.timeLow); 957 chain->setEffectSuspended_l(&desc->mType, true); 958 } 959 } 960 } 961} 962 963void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 964 bool suspend, 965 int sessionId) 966{ 967 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 968 969 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 970 971 if (suspend) { 972 if (index >= 0) { 973 sessionEffects = mSuspendedSessions.valueAt(index); 974 } else { 975 mSuspendedSessions.add(sessionId, sessionEffects); 976 } 977 } else { 978 if (index < 0) { 979 return; 980 } 981 sessionEffects = mSuspendedSessions.valueAt(index); 982 } 983 984 985 int key = EffectChain::kKeyForSuspendAll; 986 if (type != NULL) { 987 key = type->timeLow; 988 } 989 index = sessionEffects.indexOfKey(key); 990 991 sp<SuspendedSessionDesc> desc; 992 if (suspend) { 993 if (index >= 0) { 994 desc = sessionEffects.valueAt(index); 995 } else { 996 desc = new SuspendedSessionDesc(); 997 if (type != NULL) { 998 desc->mType = *type; 999 } 1000 sessionEffects.add(key, desc); 1001 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1002 } 1003 desc->mRefCount++; 1004 } else { 1005 if (index < 0) { 1006 return; 1007 } 1008 desc = sessionEffects.valueAt(index); 1009 if (--desc->mRefCount == 0) { 1010 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1011 sessionEffects.removeItemsAt(index); 1012 if (sessionEffects.isEmpty()) { 1013 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1014 sessionId); 1015 mSuspendedSessions.removeItem(sessionId); 1016 } 1017 } 1018 } 1019 if (!sessionEffects.isEmpty()) { 1020 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1021 } 1022} 1023 1024void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1025 bool enabled, 1026 int sessionId) 1027{ 1028 Mutex::Autolock _l(mLock); 1029 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1030} 1031 1032void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1033 bool enabled, 1034 int sessionId) 1035{ 1036 if (mType != RECORD) { 1037 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1038 // another session. This gives the priority to well behaved effect control panels 1039 // and applications not using global effects. 1040 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1041 // global effects 1042 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1043 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1044 } 1045 } 1046 1047 sp<EffectChain> chain = getEffectChain_l(sessionId); 1048 if (chain != 0) { 1049 chain->checkSuspendOnEffectEnabled(effect, enabled); 1050 } 1051} 1052 1053// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1054sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1055 const sp<AudioFlinger::Client>& client, 1056 const sp<IEffectClient>& effectClient, 1057 int32_t priority, 1058 int sessionId, 1059 effect_descriptor_t *desc, 1060 int *enabled, 1061 status_t *status) 1062{ 1063 sp<EffectModule> effect; 1064 sp<EffectHandle> handle; 1065 status_t lStatus; 1066 sp<EffectChain> chain; 1067 bool chainCreated = false; 1068 bool effectCreated = false; 1069 bool effectRegistered = false; 1070 1071 lStatus = initCheck(); 1072 if (lStatus != NO_ERROR) { 1073 ALOGW("createEffect_l() Audio driver not initialized."); 1074 goto Exit; 1075 } 1076 1077 // Reject any effect on Direct output threads for now, since the format of 1078 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1079 if (mType == DIRECT) { 1080 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1081 desc->name, mThreadName); 1082 lStatus = BAD_VALUE; 1083 goto Exit; 1084 } 1085 1086 // Reject any effect on mixer or duplicating multichannel sinks. 1087 // TODO: fix both format and multichannel issues with effects. 1088 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1089 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1090 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1091 lStatus = BAD_VALUE; 1092 goto Exit; 1093 } 1094 1095 // Allow global effects only on offloaded and mixer threads 1096 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1097 switch (mType) { 1098 case MIXER: 1099 case OFFLOAD: 1100 break; 1101 case DIRECT: 1102 case DUPLICATING: 1103 case RECORD: 1104 default: 1105 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1106 desc->name, mThreadName); 1107 lStatus = BAD_VALUE; 1108 goto Exit; 1109 } 1110 } 1111 1112 // Only Pre processor effects are allowed on input threads and only on input threads 1113 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1114 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1115 desc->name, desc->flags, mType); 1116 lStatus = BAD_VALUE; 1117 goto Exit; 1118 } 1119 1120 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1121 1122 { // scope for mLock 1123 Mutex::Autolock _l(mLock); 1124 1125 // check for existing effect chain with the requested audio session 1126 chain = getEffectChain_l(sessionId); 1127 if (chain == 0) { 1128 // create a new chain for this session 1129 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1130 chain = new EffectChain(this, sessionId); 1131 addEffectChain_l(chain); 1132 chain->setStrategy(getStrategyForSession_l(sessionId)); 1133 chainCreated = true; 1134 } else { 1135 effect = chain->getEffectFromDesc_l(desc); 1136 } 1137 1138 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1139 1140 if (effect == 0) { 1141 int id = mAudioFlinger->nextUniqueId(); 1142 // Check CPU and memory usage 1143 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1144 if (lStatus != NO_ERROR) { 1145 goto Exit; 1146 } 1147 effectRegistered = true; 1148 // create a new effect module if none present in the chain 1149 effect = new EffectModule(this, chain, desc, id, sessionId); 1150 lStatus = effect->status(); 1151 if (lStatus != NO_ERROR) { 1152 goto Exit; 1153 } 1154 effect->setOffloaded(mType == OFFLOAD, mId); 1155 1156 lStatus = chain->addEffect_l(effect); 1157 if (lStatus != NO_ERROR) { 1158 goto Exit; 1159 } 1160 effectCreated = true; 1161 1162 effect->setDevice(mOutDevice); 1163 effect->setDevice(mInDevice); 1164 effect->setMode(mAudioFlinger->getMode()); 1165 effect->setAudioSource(mAudioSource); 1166 } 1167 // create effect handle and connect it to effect module 1168 handle = new EffectHandle(effect, client, effectClient, priority); 1169 lStatus = handle->initCheck(); 1170 if (lStatus == OK) { 1171 lStatus = effect->addHandle(handle.get()); 1172 } 1173 if (enabled != NULL) { 1174 *enabled = (int)effect->isEnabled(); 1175 } 1176 } 1177 1178Exit: 1179 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1180 Mutex::Autolock _l(mLock); 1181 if (effectCreated) { 1182 chain->removeEffect_l(effect); 1183 } 1184 if (effectRegistered) { 1185 AudioSystem::unregisterEffect(effect->id()); 1186 } 1187 if (chainCreated) { 1188 removeEffectChain_l(chain); 1189 } 1190 handle.clear(); 1191 } 1192 1193 *status = lStatus; 1194 return handle; 1195} 1196 1197sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1198{ 1199 Mutex::Autolock _l(mLock); 1200 return getEffect_l(sessionId, effectId); 1201} 1202 1203sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1204{ 1205 sp<EffectChain> chain = getEffectChain_l(sessionId); 1206 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1207} 1208 1209// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1210// PlaybackThread::mLock held 1211status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1212{ 1213 // check for existing effect chain with the requested audio session 1214 int sessionId = effect->sessionId(); 1215 sp<EffectChain> chain = getEffectChain_l(sessionId); 1216 bool chainCreated = false; 1217 1218 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1219 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1220 this, effect->desc().name, effect->desc().flags); 1221 1222 if (chain == 0) { 1223 // create a new chain for this session 1224 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1225 chain = new EffectChain(this, sessionId); 1226 addEffectChain_l(chain); 1227 chain->setStrategy(getStrategyForSession_l(sessionId)); 1228 chainCreated = true; 1229 } 1230 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1231 1232 if (chain->getEffectFromId_l(effect->id()) != 0) { 1233 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1234 this, effect->desc().name, chain.get()); 1235 return BAD_VALUE; 1236 } 1237 1238 effect->setOffloaded(mType == OFFLOAD, mId); 1239 1240 status_t status = chain->addEffect_l(effect); 1241 if (status != NO_ERROR) { 1242 if (chainCreated) { 1243 removeEffectChain_l(chain); 1244 } 1245 return status; 1246 } 1247 1248 effect->setDevice(mOutDevice); 1249 effect->setDevice(mInDevice); 1250 effect->setMode(mAudioFlinger->getMode()); 1251 effect->setAudioSource(mAudioSource); 1252 return NO_ERROR; 1253} 1254 1255void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1256 1257 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1258 effect_descriptor_t desc = effect->desc(); 1259 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1260 detachAuxEffect_l(effect->id()); 1261 } 1262 1263 sp<EffectChain> chain = effect->chain().promote(); 1264 if (chain != 0) { 1265 // remove effect chain if removing last effect 1266 if (chain->removeEffect_l(effect) == 0) { 1267 removeEffectChain_l(chain); 1268 } 1269 } else { 1270 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1271 } 1272} 1273 1274void AudioFlinger::ThreadBase::lockEffectChains_l( 1275 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1276{ 1277 effectChains = mEffectChains; 1278 for (size_t i = 0; i < mEffectChains.size(); i++) { 1279 mEffectChains[i]->lock(); 1280 } 1281} 1282 1283void AudioFlinger::ThreadBase::unlockEffectChains( 1284 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1285{ 1286 for (size_t i = 0; i < effectChains.size(); i++) { 1287 effectChains[i]->unlock(); 1288 } 1289} 1290 1291sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1292{ 1293 Mutex::Autolock _l(mLock); 1294 return getEffectChain_l(sessionId); 1295} 1296 1297sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1298{ 1299 size_t size = mEffectChains.size(); 1300 for (size_t i = 0; i < size; i++) { 1301 if (mEffectChains[i]->sessionId() == sessionId) { 1302 return mEffectChains[i]; 1303 } 1304 } 1305 return 0; 1306} 1307 1308void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1309{ 1310 Mutex::Autolock _l(mLock); 1311 size_t size = mEffectChains.size(); 1312 for (size_t i = 0; i < size; i++) { 1313 mEffectChains[i]->setMode_l(mode); 1314 } 1315} 1316 1317void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1318{ 1319 config->type = AUDIO_PORT_TYPE_MIX; 1320 config->ext.mix.handle = mId; 1321 config->sample_rate = mSampleRate; 1322 config->format = mFormat; 1323 config->channel_mask = mChannelMask; 1324 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1325 AUDIO_PORT_CONFIG_FORMAT; 1326} 1327 1328 1329// ---------------------------------------------------------------------------- 1330// Playback 1331// ---------------------------------------------------------------------------- 1332 1333AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1334 AudioStreamOut* output, 1335 audio_io_handle_t id, 1336 audio_devices_t device, 1337 type_t type) 1338 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1339 mNormalFrameCount(0), mSinkBuffer(NULL), 1340 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1341 mMixerBuffer(NULL), 1342 mMixerBufferSize(0), 1343 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1344 mMixerBufferValid(false), 1345 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1346 mEffectBuffer(NULL), 1347 mEffectBufferSize(0), 1348 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1349 mEffectBufferValid(false), 1350 mSuspended(0), mBytesWritten(0), 1351 mActiveTracksGeneration(0), 1352 // mStreamTypes[] initialized in constructor body 1353 mOutput(output), 1354 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1355 mMixerStatus(MIXER_IDLE), 1356 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1357 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1358 mBytesRemaining(0), 1359 mCurrentWriteLength(0), 1360 mUseAsyncWrite(false), 1361 mWriteAckSequence(0), 1362 mDrainSequence(0), 1363 mSignalPending(false), 1364 mScreenState(AudioFlinger::mScreenState), 1365 // index 0 is reserved for normal mixer's submix 1366 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1367 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1368 // mLatchD, mLatchQ, 1369 mLatchDValid(false), mLatchQValid(false) 1370{ 1371 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1372 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1373 1374 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1375 // it would be safer to explicitly pass initial masterVolume/masterMute as 1376 // parameter. 1377 // 1378 // If the HAL we are using has support for master volume or master mute, 1379 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1380 // and the mute set to false). 1381 mMasterVolume = audioFlinger->masterVolume_l(); 1382 mMasterMute = audioFlinger->masterMute_l(); 1383 if (mOutput && mOutput->audioHwDev) { 1384 if (mOutput->audioHwDev->canSetMasterVolume()) { 1385 mMasterVolume = 1.0; 1386 } 1387 1388 if (mOutput->audioHwDev->canSetMasterMute()) { 1389 mMasterMute = false; 1390 } 1391 } 1392 1393 readOutputParameters_l(); 1394 1395 // ++ operator does not compile 1396 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1397 stream = (audio_stream_type_t) (stream + 1)) { 1398 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1399 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1400 } 1401} 1402 1403AudioFlinger::PlaybackThread::~PlaybackThread() 1404{ 1405 mAudioFlinger->unregisterWriter(mNBLogWriter); 1406 free(mSinkBuffer); 1407 free(mMixerBuffer); 1408 free(mEffectBuffer); 1409} 1410 1411void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1412{ 1413 dumpInternals(fd, args); 1414 dumpTracks(fd, args); 1415 dumpEffectChains(fd, args); 1416} 1417 1418void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1419{ 1420 const size_t SIZE = 256; 1421 char buffer[SIZE]; 1422 String8 result; 1423 1424 result.appendFormat(" Stream volumes in dB: "); 1425 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1426 const stream_type_t *st = &mStreamTypes[i]; 1427 if (i > 0) { 1428 result.appendFormat(", "); 1429 } 1430 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1431 if (st->mute) { 1432 result.append("M"); 1433 } 1434 } 1435 result.append("\n"); 1436 write(fd, result.string(), result.length()); 1437 result.clear(); 1438 1439 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1440 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1441 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1442 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1443 1444 size_t numtracks = mTracks.size(); 1445 size_t numactive = mActiveTracks.size(); 1446 dprintf(fd, " %d Tracks", numtracks); 1447 size_t numactiveseen = 0; 1448 if (numtracks) { 1449 dprintf(fd, " of which %d are active\n", numactive); 1450 Track::appendDumpHeader(result); 1451 for (size_t i = 0; i < numtracks; ++i) { 1452 sp<Track> track = mTracks[i]; 1453 if (track != 0) { 1454 bool active = mActiveTracks.indexOf(track) >= 0; 1455 if (active) { 1456 numactiveseen++; 1457 } 1458 track->dump(buffer, SIZE, active); 1459 result.append(buffer); 1460 } 1461 } 1462 } else { 1463 result.append("\n"); 1464 } 1465 if (numactiveseen != numactive) { 1466 // some tracks in the active list were not in the tracks list 1467 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1468 " not in the track list\n"); 1469 result.append(buffer); 1470 Track::appendDumpHeader(result); 1471 for (size_t i = 0; i < numactive; ++i) { 1472 sp<Track> track = mActiveTracks[i].promote(); 1473 if (track != 0 && mTracks.indexOf(track) < 0) { 1474 track->dump(buffer, SIZE, true); 1475 result.append(buffer); 1476 } 1477 } 1478 } 1479 1480 write(fd, result.string(), result.size()); 1481} 1482 1483void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1484{ 1485 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1486 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1487 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1488 dprintf(fd, " Total writes: %d\n", mNumWrites); 1489 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1490 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1491 dprintf(fd, " Suspend count: %d\n", mSuspended); 1492 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1493 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1494 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1495 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1496 AudioStreamOut *output = mOutput; 1497 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1498 String8 flagsAsString = outputFlagsToString(flags); 1499 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1500 1501 dumpBase(fd, args); 1502} 1503 1504// Thread virtuals 1505 1506void AudioFlinger::PlaybackThread::onFirstRef() 1507{ 1508 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1509} 1510 1511// ThreadBase virtuals 1512void AudioFlinger::PlaybackThread::preExit() 1513{ 1514 ALOGV(" preExit()"); 1515 // FIXME this is using hard-coded strings but in the future, this functionality will be 1516 // converted to use audio HAL extensions required to support tunneling 1517 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1518} 1519 1520// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1521sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1522 const sp<AudioFlinger::Client>& client, 1523 audio_stream_type_t streamType, 1524 uint32_t sampleRate, 1525 audio_format_t format, 1526 audio_channel_mask_t channelMask, 1527 size_t *pFrameCount, 1528 const sp<IMemory>& sharedBuffer, 1529 int sessionId, 1530 IAudioFlinger::track_flags_t *flags, 1531 pid_t tid, 1532 int uid, 1533 status_t *status) 1534{ 1535 size_t frameCount = *pFrameCount; 1536 sp<Track> track; 1537 status_t lStatus; 1538 1539 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1540 1541 // client expresses a preference for FAST, but we get the final say 1542 if (*flags & IAudioFlinger::TRACK_FAST) { 1543 if ( 1544 // not timed 1545 (!isTimed) && 1546 // either of these use cases: 1547 ( 1548 // use case 1: shared buffer with any frame count 1549 ( 1550 (sharedBuffer != 0) 1551 ) || 1552 // use case 2: callback handler and frame count is default or at least as large as HAL 1553 ( 1554 (tid != -1) && 1555 ((frameCount == 0) || 1556 (frameCount >= mFrameCount)) 1557 ) 1558 ) && 1559 // PCM data 1560 audio_is_linear_pcm(format) && 1561 // identical channel mask to sink, or mono in and stereo sink 1562 (channelMask == mChannelMask || 1563 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1564 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1565 // hardware sample rate 1566 (sampleRate == mSampleRate) && 1567 // normal mixer has an associated fast mixer 1568 hasFastMixer() && 1569 // there are sufficient fast track slots available 1570 (mFastTrackAvailMask != 0) 1571 // FIXME test that MixerThread for this fast track has a capable output HAL 1572 // FIXME add a permission test also? 1573 ) { 1574 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1575 if (frameCount == 0) { 1576 // read the fast track multiplier property the first time it is needed 1577 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1578 if (ok != 0) { 1579 ALOGE("%s pthread_once failed: %d", __func__, ok); 1580 } 1581 frameCount = mFrameCount * sFastTrackMultiplier; 1582 } 1583 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1584 frameCount, mFrameCount); 1585 } else { 1586 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1587 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1588 "sampleRate=%u mSampleRate=%u " 1589 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1590 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1591 audio_is_linear_pcm(format), 1592 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1593 *flags &= ~IAudioFlinger::TRACK_FAST; 1594 } 1595 } 1596 // For normal PCM streaming tracks, update minimum frame count. 1597 // For compatibility with AudioTrack calculation, buffer depth is forced 1598 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1599 // This is probably too conservative, but legacy application code may depend on it. 1600 // If you change this calculation, also review the start threshold which is related. 1601 if (!(*flags & IAudioFlinger::TRACK_FAST) 1602 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1603 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1604 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1605 if (minBufCount < 2) { 1606 minBufCount = 2; 1607 } 1608 size_t minFrameCount = 1609 minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate); 1610 if (frameCount < minFrameCount) { // including frameCount == 0 1611 frameCount = minFrameCount; 1612 } 1613 } 1614 *pFrameCount = frameCount; 1615 1616 switch (mType) { 1617 1618 case DIRECT: 1619 if (audio_is_linear_pcm(format)) { 1620 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1621 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1622 "for output %p with format %#x", 1623 sampleRate, format, channelMask, mOutput, mFormat); 1624 lStatus = BAD_VALUE; 1625 goto Exit; 1626 } 1627 } 1628 break; 1629 1630 case OFFLOAD: 1631 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1632 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1633 "for output %p with format %#x", 1634 sampleRate, format, channelMask, mOutput, mFormat); 1635 lStatus = BAD_VALUE; 1636 goto Exit; 1637 } 1638 break; 1639 1640 default: 1641 if (!audio_is_linear_pcm(format)) { 1642 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1643 "for output %p with format %#x", 1644 format, mOutput, mFormat); 1645 lStatus = BAD_VALUE; 1646 goto Exit; 1647 } 1648 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1649 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1650 lStatus = BAD_VALUE; 1651 goto Exit; 1652 } 1653 break; 1654 1655 } 1656 1657 lStatus = initCheck(); 1658 if (lStatus != NO_ERROR) { 1659 ALOGE("createTrack_l() audio driver not initialized"); 1660 goto Exit; 1661 } 1662 1663 { // scope for mLock 1664 Mutex::Autolock _l(mLock); 1665 1666 // all tracks in same audio session must share the same routing strategy otherwise 1667 // conflicts will happen when tracks are moved from one output to another by audio policy 1668 // manager 1669 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1670 for (size_t i = 0; i < mTracks.size(); ++i) { 1671 sp<Track> t = mTracks[i]; 1672 if (t != 0 && t->isExternalTrack()) { 1673 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1674 if (sessionId == t->sessionId() && strategy != actual) { 1675 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1676 strategy, actual); 1677 lStatus = BAD_VALUE; 1678 goto Exit; 1679 } 1680 } 1681 } 1682 1683 if (!isTimed) { 1684 track = new Track(this, client, streamType, sampleRate, format, 1685 channelMask, frameCount, NULL, sharedBuffer, 1686 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1687 } else { 1688 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1689 channelMask, frameCount, sharedBuffer, sessionId, uid); 1690 } 1691 1692 // new Track always returns non-NULL, 1693 // but TimedTrack::create() is a factory that could fail by returning NULL 1694 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1695 if (lStatus != NO_ERROR) { 1696 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1697 // track must be cleared from the caller as the caller has the AF lock 1698 goto Exit; 1699 } 1700 mTracks.add(track); 1701 1702 sp<EffectChain> chain = getEffectChain_l(sessionId); 1703 if (chain != 0) { 1704 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1705 track->setMainBuffer(chain->inBuffer()); 1706 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1707 chain->incTrackCnt(); 1708 } 1709 1710 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1711 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1712 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1713 // so ask activity manager to do this on our behalf 1714 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1715 } 1716 } 1717 1718 lStatus = NO_ERROR; 1719 1720Exit: 1721 *status = lStatus; 1722 return track; 1723} 1724 1725uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1726{ 1727 return latency; 1728} 1729 1730uint32_t AudioFlinger::PlaybackThread::latency() const 1731{ 1732 Mutex::Autolock _l(mLock); 1733 return latency_l(); 1734} 1735uint32_t AudioFlinger::PlaybackThread::latency_l() const 1736{ 1737 if (initCheck() == NO_ERROR) { 1738 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1739 } else { 1740 return 0; 1741 } 1742} 1743 1744void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1745{ 1746 Mutex::Autolock _l(mLock); 1747 // Don't apply master volume in SW if our HAL can do it for us. 1748 if (mOutput && mOutput->audioHwDev && 1749 mOutput->audioHwDev->canSetMasterVolume()) { 1750 mMasterVolume = 1.0; 1751 } else { 1752 mMasterVolume = value; 1753 } 1754} 1755 1756void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1757{ 1758 Mutex::Autolock _l(mLock); 1759 // Don't apply master mute in SW if our HAL can do it for us. 1760 if (mOutput && mOutput->audioHwDev && 1761 mOutput->audioHwDev->canSetMasterMute()) { 1762 mMasterMute = false; 1763 } else { 1764 mMasterMute = muted; 1765 } 1766} 1767 1768void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1769{ 1770 Mutex::Autolock _l(mLock); 1771 mStreamTypes[stream].volume = value; 1772 broadcast_l(); 1773} 1774 1775void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1776{ 1777 Mutex::Autolock _l(mLock); 1778 mStreamTypes[stream].mute = muted; 1779 broadcast_l(); 1780} 1781 1782float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1783{ 1784 Mutex::Autolock _l(mLock); 1785 return mStreamTypes[stream].volume; 1786} 1787 1788// addTrack_l() must be called with ThreadBase::mLock held 1789status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1790{ 1791 status_t status = ALREADY_EXISTS; 1792 1793 // set retry count for buffer fill 1794 track->mRetryCount = kMaxTrackStartupRetries; 1795 if (mActiveTracks.indexOf(track) < 0) { 1796 // the track is newly added, make sure it fills up all its 1797 // buffers before playing. This is to ensure the client will 1798 // effectively get the latency it requested. 1799 if (track->isExternalTrack()) { 1800 TrackBase::track_state state = track->mState; 1801 mLock.unlock(); 1802 status = AudioSystem::startOutput(mId, track->streamType(), 1803 (audio_session_t)track->sessionId()); 1804 mLock.lock(); 1805 // abort track was stopped/paused while we released the lock 1806 if (state != track->mState) { 1807 if (status == NO_ERROR) { 1808 mLock.unlock(); 1809 AudioSystem::stopOutput(mId, track->streamType(), 1810 (audio_session_t)track->sessionId()); 1811 mLock.lock(); 1812 } 1813 return INVALID_OPERATION; 1814 } 1815 // abort if start is rejected by audio policy manager 1816 if (status != NO_ERROR) { 1817 return PERMISSION_DENIED; 1818 } 1819#ifdef ADD_BATTERY_DATA 1820 // to track the speaker usage 1821 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1822#endif 1823 } 1824 1825 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1826 track->mResetDone = false; 1827 track->mPresentationCompleteFrames = 0; 1828 mActiveTracks.add(track); 1829 mWakeLockUids.add(track->uid()); 1830 mActiveTracksGeneration++; 1831 mLatestActiveTrack = track; 1832 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1833 if (chain != 0) { 1834 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1835 track->sessionId()); 1836 chain->incActiveTrackCnt(); 1837 } 1838 1839 status = NO_ERROR; 1840 } 1841 1842 onAddNewTrack_l(); 1843 return status; 1844} 1845 1846bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1847{ 1848 track->terminate(); 1849 // active tracks are removed by threadLoop() 1850 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1851 track->mState = TrackBase::STOPPED; 1852 if (!trackActive) { 1853 removeTrack_l(track); 1854 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1855 track->mState = TrackBase::STOPPING_1; 1856 } 1857 1858 return trackActive; 1859} 1860 1861void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1862{ 1863 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1864 mTracks.remove(track); 1865 deleteTrackName_l(track->name()); 1866 // redundant as track is about to be destroyed, for dumpsys only 1867 track->mName = -1; 1868 if (track->isFastTrack()) { 1869 int index = track->mFastIndex; 1870 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1871 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1872 mFastTrackAvailMask |= 1 << index; 1873 // redundant as track is about to be destroyed, for dumpsys only 1874 track->mFastIndex = -1; 1875 } 1876 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1877 if (chain != 0) { 1878 chain->decTrackCnt(); 1879 } 1880} 1881 1882void AudioFlinger::PlaybackThread::broadcast_l() 1883{ 1884 // Thread could be blocked waiting for async 1885 // so signal it to handle state changes immediately 1886 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1887 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1888 mSignalPending = true; 1889 mWaitWorkCV.broadcast(); 1890} 1891 1892String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1893{ 1894 Mutex::Autolock _l(mLock); 1895 if (initCheck() != NO_ERROR) { 1896 return String8(); 1897 } 1898 1899 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1900 const String8 out_s8(s); 1901 free(s); 1902 return out_s8; 1903} 1904 1905void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1906 AudioSystem::OutputDescriptor desc; 1907 void *param2 = NULL; 1908 1909 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1910 param); 1911 1912 switch (event) { 1913 case AudioSystem::OUTPUT_OPENED: 1914 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1915 desc.channelMask = mChannelMask; 1916 desc.samplingRate = mSampleRate; 1917 desc.format = mFormat; 1918 desc.frameCount = mNormalFrameCount; // FIXME see 1919 // AudioFlinger::frameCount(audio_io_handle_t) 1920 desc.latency = latency_l(); 1921 param2 = &desc; 1922 break; 1923 1924 case AudioSystem::STREAM_CONFIG_CHANGED: 1925 param2 = ¶m; 1926 case AudioSystem::OUTPUT_CLOSED: 1927 default: 1928 break; 1929 } 1930 mAudioFlinger->audioConfigChanged(event, mId, param2); 1931} 1932 1933void AudioFlinger::PlaybackThread::writeCallback() 1934{ 1935 ALOG_ASSERT(mCallbackThread != 0); 1936 mCallbackThread->resetWriteBlocked(); 1937} 1938 1939void AudioFlinger::PlaybackThread::drainCallback() 1940{ 1941 ALOG_ASSERT(mCallbackThread != 0); 1942 mCallbackThread->resetDraining(); 1943} 1944 1945void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1946{ 1947 Mutex::Autolock _l(mLock); 1948 // reject out of sequence requests 1949 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1950 mWriteAckSequence &= ~1; 1951 mWaitWorkCV.signal(); 1952 } 1953} 1954 1955void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1956{ 1957 Mutex::Autolock _l(mLock); 1958 // reject out of sequence requests 1959 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1960 mDrainSequence &= ~1; 1961 mWaitWorkCV.signal(); 1962 } 1963} 1964 1965// static 1966int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1967 void *param __unused, 1968 void *cookie) 1969{ 1970 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1971 ALOGV("asyncCallback() event %d", event); 1972 switch (event) { 1973 case STREAM_CBK_EVENT_WRITE_READY: 1974 me->writeCallback(); 1975 break; 1976 case STREAM_CBK_EVENT_DRAIN_READY: 1977 me->drainCallback(); 1978 break; 1979 default: 1980 ALOGW("asyncCallback() unknown event %d", event); 1981 break; 1982 } 1983 return 0; 1984} 1985 1986void AudioFlinger::PlaybackThread::readOutputParameters_l() 1987{ 1988 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1989 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1990 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1991 if (!audio_is_output_channel(mChannelMask)) { 1992 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1993 } 1994 if ((mType == MIXER || mType == DUPLICATING) 1995 && !isValidPcmSinkChannelMask(mChannelMask)) { 1996 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1997 mChannelMask); 1998 } 1999 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2000 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2001 mFormat = mHALFormat; 2002 if (!audio_is_valid_format(mFormat)) { 2003 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2004 } 2005 if ((mType == MIXER || mType == DUPLICATING) 2006 && !isValidPcmSinkFormat(mFormat)) { 2007 LOG_FATAL("HAL format %#x not supported for mixed output", 2008 mFormat); 2009 } 2010 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 2011 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2012 mFrameCount = mBufferSize / mFrameSize; 2013 if (mFrameCount & 15) { 2014 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2015 mFrameCount); 2016 } 2017 2018 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2019 (mOutput->stream->set_callback != NULL)) { 2020 if (mOutput->stream->set_callback(mOutput->stream, 2021 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2022 mUseAsyncWrite = true; 2023 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2024 } 2025 } 2026 2027 mHwSupportsPause = false; 2028 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2029 if (mOutput->stream->pause != NULL) { 2030 if (mOutput->stream->resume != NULL) { 2031 mHwSupportsPause = true; 2032 } else { 2033 ALOGW("direct output implements pause but not resume"); 2034 } 2035 } else if (mOutput->stream->resume != NULL) { 2036 ALOGW("direct output implements resume but not pause"); 2037 } 2038 } 2039 2040 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2041 // For best precision, we use float instead of the associated output 2042 // device format (typically PCM 16 bit). 2043 2044 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2045 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2046 mBufferSize = mFrameSize * mFrameCount; 2047 2048 // TODO: We currently use the associated output device channel mask and sample rate. 2049 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2050 // (if a valid mask) to avoid premature downmix. 2051 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2052 // instead of the output device sample rate to avoid loss of high frequency information. 2053 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2054 } 2055 2056 // Calculate size of normal sink buffer relative to the HAL output buffer size 2057 double multiplier = 1.0; 2058 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2059 kUseFastMixer == FastMixer_Dynamic)) { 2060 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2061 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2062 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2063 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2064 maxNormalFrameCount = maxNormalFrameCount & ~15; 2065 if (maxNormalFrameCount < minNormalFrameCount) { 2066 maxNormalFrameCount = minNormalFrameCount; 2067 } 2068 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2069 if (multiplier <= 1.0) { 2070 multiplier = 1.0; 2071 } else if (multiplier <= 2.0) { 2072 if (2 * mFrameCount <= maxNormalFrameCount) { 2073 multiplier = 2.0; 2074 } else { 2075 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2076 } 2077 } else { 2078 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2079 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2080 // track, but we sometimes have to do this to satisfy the maximum frame count 2081 // constraint) 2082 // FIXME this rounding up should not be done if no HAL SRC 2083 uint32_t truncMult = (uint32_t) multiplier; 2084 if ((truncMult & 1)) { 2085 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2086 ++truncMult; 2087 } 2088 } 2089 multiplier = (double) truncMult; 2090 } 2091 } 2092 mNormalFrameCount = multiplier * mFrameCount; 2093 // round up to nearest 16 frames to satisfy AudioMixer 2094 if (mType == MIXER || mType == DUPLICATING) { 2095 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2096 } 2097 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2098 mNormalFrameCount); 2099 2100 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2101 // Originally this was int16_t[] array, need to remove legacy implications. 2102 free(mSinkBuffer); 2103 mSinkBuffer = NULL; 2104 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2105 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2106 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2107 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2108 2109 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2110 // drives the output. 2111 free(mMixerBuffer); 2112 mMixerBuffer = NULL; 2113 if (mMixerBufferEnabled) { 2114 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2115 mMixerBufferSize = mNormalFrameCount * mChannelCount 2116 * audio_bytes_per_sample(mMixerBufferFormat); 2117 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2118 } 2119 free(mEffectBuffer); 2120 mEffectBuffer = NULL; 2121 if (mEffectBufferEnabled) { 2122 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2123 mEffectBufferSize = mNormalFrameCount * mChannelCount 2124 * audio_bytes_per_sample(mEffectBufferFormat); 2125 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2126 } 2127 2128 // force reconfiguration of effect chains and engines to take new buffer size and audio 2129 // parameters into account 2130 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2131 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2132 // matter. 2133 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2134 Vector< sp<EffectChain> > effectChains = mEffectChains; 2135 for (size_t i = 0; i < effectChains.size(); i ++) { 2136 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2137 } 2138} 2139 2140 2141status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2142{ 2143 if (halFrames == NULL || dspFrames == NULL) { 2144 return BAD_VALUE; 2145 } 2146 Mutex::Autolock _l(mLock); 2147 if (initCheck() != NO_ERROR) { 2148 return INVALID_OPERATION; 2149 } 2150 size_t framesWritten = mBytesWritten / mFrameSize; 2151 *halFrames = framesWritten; 2152 2153 if (isSuspended()) { 2154 // return an estimation of rendered frames when the output is suspended 2155 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2156 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2157 return NO_ERROR; 2158 } else { 2159 status_t status; 2160 uint32_t frames; 2161 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 2162 *dspFrames = (size_t)frames; 2163 return status; 2164 } 2165} 2166 2167uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2168{ 2169 Mutex::Autolock _l(mLock); 2170 uint32_t result = 0; 2171 if (getEffectChain_l(sessionId) != 0) { 2172 result = EFFECT_SESSION; 2173 } 2174 2175 for (size_t i = 0; i < mTracks.size(); ++i) { 2176 sp<Track> track = mTracks[i]; 2177 if (sessionId == track->sessionId() && !track->isInvalid()) { 2178 result |= TRACK_SESSION; 2179 break; 2180 } 2181 } 2182 2183 return result; 2184} 2185 2186uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2187{ 2188 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2189 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2190 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2191 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2192 } 2193 for (size_t i = 0; i < mTracks.size(); i++) { 2194 sp<Track> track = mTracks[i]; 2195 if (sessionId == track->sessionId() && !track->isInvalid()) { 2196 return AudioSystem::getStrategyForStream(track->streamType()); 2197 } 2198 } 2199 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2200} 2201 2202 2203AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2204{ 2205 Mutex::Autolock _l(mLock); 2206 return mOutput; 2207} 2208 2209AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2210{ 2211 Mutex::Autolock _l(mLock); 2212 AudioStreamOut *output = mOutput; 2213 mOutput = NULL; 2214 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2215 // must push a NULL and wait for ack 2216 mOutputSink.clear(); 2217 mPipeSink.clear(); 2218 mNormalSink.clear(); 2219 return output; 2220} 2221 2222// this method must always be called either with ThreadBase mLock held or inside the thread loop 2223audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2224{ 2225 if (mOutput == NULL) { 2226 return NULL; 2227 } 2228 return &mOutput->stream->common; 2229} 2230 2231uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2232{ 2233 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2234} 2235 2236status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2237{ 2238 if (!isValidSyncEvent(event)) { 2239 return BAD_VALUE; 2240 } 2241 2242 Mutex::Autolock _l(mLock); 2243 2244 for (size_t i = 0; i < mTracks.size(); ++i) { 2245 sp<Track> track = mTracks[i]; 2246 if (event->triggerSession() == track->sessionId()) { 2247 (void) track->setSyncEvent(event); 2248 return NO_ERROR; 2249 } 2250 } 2251 2252 return NAME_NOT_FOUND; 2253} 2254 2255bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2256{ 2257 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2258} 2259 2260void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2261 const Vector< sp<Track> >& tracksToRemove) 2262{ 2263 size_t count = tracksToRemove.size(); 2264 if (count > 0) { 2265 for (size_t i = 0 ; i < count ; i++) { 2266 const sp<Track>& track = tracksToRemove.itemAt(i); 2267 if (track->isExternalTrack()) { 2268 AudioSystem::stopOutput(mId, track->streamType(), 2269 (audio_session_t)track->sessionId()); 2270#ifdef ADD_BATTERY_DATA 2271 // to track the speaker usage 2272 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2273#endif 2274 if (track->isTerminated()) { 2275 AudioSystem::releaseOutput(mId, track->streamType(), 2276 (audio_session_t)track->sessionId()); 2277 } 2278 } 2279 } 2280 } 2281} 2282 2283void AudioFlinger::PlaybackThread::checkSilentMode_l() 2284{ 2285 if (!mMasterMute) { 2286 char value[PROPERTY_VALUE_MAX]; 2287 if (property_get("ro.audio.silent", value, "0") > 0) { 2288 char *endptr; 2289 unsigned long ul = strtoul(value, &endptr, 0); 2290 if (*endptr == '\0' && ul != 0) { 2291 ALOGD("Silence is golden"); 2292 // The setprop command will not allow a property to be changed after 2293 // the first time it is set, so we don't have to worry about un-muting. 2294 setMasterMute_l(true); 2295 } 2296 } 2297 } 2298} 2299 2300// shared by MIXER and DIRECT, overridden by DUPLICATING 2301ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2302{ 2303 // FIXME rewrite to reduce number of system calls 2304 mLastWriteTime = systemTime(); 2305 mInWrite = true; 2306 ssize_t bytesWritten; 2307 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2308 2309 // If an NBAIO sink is present, use it to write the normal mixer's submix 2310 if (mNormalSink != 0) { 2311 2312 const size_t count = mBytesRemaining / mFrameSize; 2313 2314 ATRACE_BEGIN("write"); 2315 // update the setpoint when AudioFlinger::mScreenState changes 2316 uint32_t screenState = AudioFlinger::mScreenState; 2317 if (screenState != mScreenState) { 2318 mScreenState = screenState; 2319 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2320 if (pipe != NULL) { 2321 pipe->setAvgFrames((mScreenState & 1) ? 2322 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2323 } 2324 } 2325 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2326 ATRACE_END(); 2327 if (framesWritten > 0) { 2328 bytesWritten = framesWritten * mFrameSize; 2329 } else { 2330 bytesWritten = framesWritten; 2331 } 2332 mLatchDValid = false; 2333 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2334 if (status == NO_ERROR) { 2335 size_t totalFramesWritten = mNormalSink->framesWritten(); 2336 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2337 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2338 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2339 mLatchDValid = true; 2340 } 2341 } 2342 // otherwise use the HAL / AudioStreamOut directly 2343 } else { 2344 // Direct output and offload threads 2345 2346 if (mUseAsyncWrite) { 2347 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2348 mWriteAckSequence += 2; 2349 mWriteAckSequence |= 1; 2350 ALOG_ASSERT(mCallbackThread != 0); 2351 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2352 } 2353 // FIXME We should have an implementation of timestamps for direct output threads. 2354 // They are used e.g for multichannel PCM playback over HDMI. 2355 bytesWritten = mOutput->stream->write(mOutput->stream, 2356 (char *)mSinkBuffer + offset, mBytesRemaining); 2357 if (mUseAsyncWrite && 2358 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2359 // do not wait for async callback in case of error of full write 2360 mWriteAckSequence &= ~1; 2361 ALOG_ASSERT(mCallbackThread != 0); 2362 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2363 } 2364 } 2365 2366 mNumWrites++; 2367 mInWrite = false; 2368 mStandby = false; 2369 return bytesWritten; 2370} 2371 2372void AudioFlinger::PlaybackThread::threadLoop_drain() 2373{ 2374 if (mOutput->stream->drain) { 2375 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2376 if (mUseAsyncWrite) { 2377 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2378 mDrainSequence |= 1; 2379 ALOG_ASSERT(mCallbackThread != 0); 2380 mCallbackThread->setDraining(mDrainSequence); 2381 } 2382 mOutput->stream->drain(mOutput->stream, 2383 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2384 : AUDIO_DRAIN_ALL); 2385 } 2386} 2387 2388void AudioFlinger::PlaybackThread::threadLoop_exit() 2389{ 2390 { 2391 Mutex::Autolock _l(mLock); 2392 for (size_t i = 0; i < mTracks.size(); i++) { 2393 sp<Track> track = mTracks[i]; 2394 track->invalidate(); 2395 } 2396 } 2397} 2398 2399/* 2400The derived values that are cached: 2401 - mSinkBufferSize from frame count * frame size 2402 - activeSleepTime from activeSleepTimeUs() 2403 - idleSleepTime from idleSleepTimeUs() 2404 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2405 - maxPeriod from frame count and sample rate (MIXER only) 2406 2407The parameters that affect these derived values are: 2408 - frame count 2409 - frame size 2410 - sample rate 2411 - device type: A2DP or not 2412 - device latency 2413 - format: PCM or not 2414 - active sleep time 2415 - idle sleep time 2416*/ 2417 2418void AudioFlinger::PlaybackThread::cacheParameters_l() 2419{ 2420 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2421 activeSleepTime = activeSleepTimeUs(); 2422 idleSleepTime = idleSleepTimeUs(); 2423} 2424 2425void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2426{ 2427 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2428 this, streamType, mTracks.size()); 2429 Mutex::Autolock _l(mLock); 2430 2431 size_t size = mTracks.size(); 2432 for (size_t i = 0; i < size; i++) { 2433 sp<Track> t = mTracks[i]; 2434 if (t->streamType() == streamType) { 2435 t->invalidate(); 2436 } 2437 } 2438} 2439 2440status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2441{ 2442 int session = chain->sessionId(); 2443 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2444 ? mEffectBuffer : mSinkBuffer); 2445 bool ownsBuffer = false; 2446 2447 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2448 if (session > 0) { 2449 // Only one effect chain can be present in direct output thread and it uses 2450 // the sink buffer as input 2451 if (mType != DIRECT) { 2452 size_t numSamples = mNormalFrameCount * mChannelCount; 2453 buffer = new int16_t[numSamples]; 2454 memset(buffer, 0, numSamples * sizeof(int16_t)); 2455 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2456 ownsBuffer = true; 2457 } 2458 2459 // Attach all tracks with same session ID to this chain. 2460 for (size_t i = 0; i < mTracks.size(); ++i) { 2461 sp<Track> track = mTracks[i]; 2462 if (session == track->sessionId()) { 2463 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2464 buffer); 2465 track->setMainBuffer(buffer); 2466 chain->incTrackCnt(); 2467 } 2468 } 2469 2470 // indicate all active tracks in the chain 2471 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2472 sp<Track> track = mActiveTracks[i].promote(); 2473 if (track == 0) { 2474 continue; 2475 } 2476 if (session == track->sessionId()) { 2477 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2478 chain->incActiveTrackCnt(); 2479 } 2480 } 2481 } 2482 chain->setThread(this); 2483 chain->setInBuffer(buffer, ownsBuffer); 2484 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2485 ? mEffectBuffer : mSinkBuffer)); 2486 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2487 // chains list in order to be processed last as it contains output stage effects 2488 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2489 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2490 // after track specific effects and before output stage 2491 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2492 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2493 // Effect chain for other sessions are inserted at beginning of effect 2494 // chains list to be processed before output mix effects. Relative order between other 2495 // sessions is not important 2496 size_t size = mEffectChains.size(); 2497 size_t i = 0; 2498 for (i = 0; i < size; i++) { 2499 if (mEffectChains[i]->sessionId() < session) { 2500 break; 2501 } 2502 } 2503 mEffectChains.insertAt(chain, i); 2504 checkSuspendOnAddEffectChain_l(chain); 2505 2506 return NO_ERROR; 2507} 2508 2509size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2510{ 2511 int session = chain->sessionId(); 2512 2513 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2514 2515 for (size_t i = 0; i < mEffectChains.size(); i++) { 2516 if (chain == mEffectChains[i]) { 2517 mEffectChains.removeAt(i); 2518 // detach all active tracks from the chain 2519 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2520 sp<Track> track = mActiveTracks[i].promote(); 2521 if (track == 0) { 2522 continue; 2523 } 2524 if (session == track->sessionId()) { 2525 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2526 chain.get(), session); 2527 chain->decActiveTrackCnt(); 2528 } 2529 } 2530 2531 // detach all tracks with same session ID from this chain 2532 for (size_t i = 0; i < mTracks.size(); ++i) { 2533 sp<Track> track = mTracks[i]; 2534 if (session == track->sessionId()) { 2535 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2536 chain->decTrackCnt(); 2537 } 2538 } 2539 break; 2540 } 2541 } 2542 return mEffectChains.size(); 2543} 2544 2545status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2546 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2547{ 2548 Mutex::Autolock _l(mLock); 2549 return attachAuxEffect_l(track, EffectId); 2550} 2551 2552status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2553 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2554{ 2555 status_t status = NO_ERROR; 2556 2557 if (EffectId == 0) { 2558 track->setAuxBuffer(0, NULL); 2559 } else { 2560 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2561 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2562 if (effect != 0) { 2563 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2564 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2565 } else { 2566 status = INVALID_OPERATION; 2567 } 2568 } else { 2569 status = BAD_VALUE; 2570 } 2571 } 2572 return status; 2573} 2574 2575void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2576{ 2577 for (size_t i = 0; i < mTracks.size(); ++i) { 2578 sp<Track> track = mTracks[i]; 2579 if (track->auxEffectId() == effectId) { 2580 attachAuxEffect_l(track, 0); 2581 } 2582 } 2583} 2584 2585bool AudioFlinger::PlaybackThread::threadLoop() 2586{ 2587 Vector< sp<Track> > tracksToRemove; 2588 2589 standbyTime = systemTime(); 2590 2591 // MIXER 2592 nsecs_t lastWarning = 0; 2593 2594 // DUPLICATING 2595 // FIXME could this be made local to while loop? 2596 writeFrames = 0; 2597 2598 int lastGeneration = 0; 2599 2600 cacheParameters_l(); 2601 sleepTime = idleSleepTime; 2602 2603 if (mType == MIXER) { 2604 sleepTimeShift = 0; 2605 } 2606 2607 CpuStats cpuStats; 2608 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2609 2610 acquireWakeLock(); 2611 2612 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2613 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2614 // and then that string will be logged at the next convenient opportunity. 2615 const char *logString = NULL; 2616 2617 checkSilentMode_l(); 2618 2619 while (!exitPending()) 2620 { 2621 cpuStats.sample(myName); 2622 2623 Vector< sp<EffectChain> > effectChains; 2624 2625 { // scope for mLock 2626 2627 Mutex::Autolock _l(mLock); 2628 2629 processConfigEvents_l(); 2630 2631 if (logString != NULL) { 2632 mNBLogWriter->logTimestamp(); 2633 mNBLogWriter->log(logString); 2634 logString = NULL; 2635 } 2636 2637 // Gather the framesReleased counters for all active tracks, 2638 // and latch them atomically with the timestamp. 2639 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2640 mLatchD.mFramesReleased.clear(); 2641 size_t size = mActiveTracks.size(); 2642 for (size_t i = 0; i < size; i++) { 2643 sp<Track> t = mActiveTracks[i].promote(); 2644 if (t != 0) { 2645 mLatchD.mFramesReleased.add(t.get(), 2646 t->mAudioTrackServerProxy->framesReleased()); 2647 } 2648 } 2649 if (mLatchDValid) { 2650 mLatchQ = mLatchD; 2651 mLatchDValid = false; 2652 mLatchQValid = true; 2653 } 2654 2655 saveOutputTracks(); 2656 if (mSignalPending) { 2657 // A signal was raised while we were unlocked 2658 mSignalPending = false; 2659 } else if (waitingAsyncCallback_l()) { 2660 if (exitPending()) { 2661 break; 2662 } 2663 releaseWakeLock_l(); 2664 mWakeLockUids.clear(); 2665 mActiveTracksGeneration++; 2666 ALOGV("wait async completion"); 2667 mWaitWorkCV.wait(mLock); 2668 ALOGV("async completion/wake"); 2669 acquireWakeLock_l(); 2670 standbyTime = systemTime() + standbyDelay; 2671 sleepTime = 0; 2672 2673 continue; 2674 } 2675 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2676 isSuspended()) { 2677 // put audio hardware into standby after short delay 2678 if (shouldStandby_l()) { 2679 2680 threadLoop_standby(); 2681 2682 mStandby = true; 2683 } 2684 2685 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2686 // we're about to wait, flush the binder command buffer 2687 IPCThreadState::self()->flushCommands(); 2688 2689 clearOutputTracks(); 2690 2691 if (exitPending()) { 2692 break; 2693 } 2694 2695 releaseWakeLock_l(); 2696 mWakeLockUids.clear(); 2697 mActiveTracksGeneration++; 2698 // wait until we have something to do... 2699 ALOGV("%s going to sleep", myName.string()); 2700 mWaitWorkCV.wait(mLock); 2701 ALOGV("%s waking up", myName.string()); 2702 acquireWakeLock_l(); 2703 2704 mMixerStatus = MIXER_IDLE; 2705 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2706 mBytesWritten = 0; 2707 mBytesRemaining = 0; 2708 checkSilentMode_l(); 2709 2710 standbyTime = systemTime() + standbyDelay; 2711 sleepTime = idleSleepTime; 2712 if (mType == MIXER) { 2713 sleepTimeShift = 0; 2714 } 2715 2716 continue; 2717 } 2718 } 2719 // mMixerStatusIgnoringFastTracks is also updated internally 2720 mMixerStatus = prepareTracks_l(&tracksToRemove); 2721 2722 // compare with previously applied list 2723 if (lastGeneration != mActiveTracksGeneration) { 2724 // update wakelock 2725 updateWakeLockUids_l(mWakeLockUids); 2726 lastGeneration = mActiveTracksGeneration; 2727 } 2728 2729 // prevent any changes in effect chain list and in each effect chain 2730 // during mixing and effect process as the audio buffers could be deleted 2731 // or modified if an effect is created or deleted 2732 lockEffectChains_l(effectChains); 2733 } // mLock scope ends 2734 2735 if (mBytesRemaining == 0) { 2736 mCurrentWriteLength = 0; 2737 if (mMixerStatus == MIXER_TRACKS_READY) { 2738 // threadLoop_mix() sets mCurrentWriteLength 2739 threadLoop_mix(); 2740 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2741 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2742 // threadLoop_sleepTime sets sleepTime to 0 if data 2743 // must be written to HAL 2744 threadLoop_sleepTime(); 2745 if (sleepTime == 0) { 2746 mCurrentWriteLength = mSinkBufferSize; 2747 } 2748 } 2749 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2750 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2751 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2752 // or mSinkBuffer (if there are no effects). 2753 // 2754 // This is done pre-effects computation; if effects change to 2755 // support higher precision, this needs to move. 2756 // 2757 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2758 // TODO use sleepTime == 0 as an additional condition. 2759 if (mMixerBufferValid) { 2760 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2761 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2762 2763 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2764 mNormalFrameCount * mChannelCount); 2765 } 2766 2767 mBytesRemaining = mCurrentWriteLength; 2768 if (isSuspended()) { 2769 sleepTime = suspendSleepTimeUs(); 2770 // simulate write to HAL when suspended 2771 mBytesWritten += mSinkBufferSize; 2772 mBytesRemaining = 0; 2773 } 2774 2775 // only process effects if we're going to write 2776 if (sleepTime == 0 && mType != OFFLOAD) { 2777 for (size_t i = 0; i < effectChains.size(); i ++) { 2778 effectChains[i]->process_l(); 2779 } 2780 } 2781 } 2782 // Process effect chains for offloaded thread even if no audio 2783 // was read from audio track: process only updates effect state 2784 // and thus does have to be synchronized with audio writes but may have 2785 // to be called while waiting for async write callback 2786 if (mType == OFFLOAD) { 2787 for (size_t i = 0; i < effectChains.size(); i ++) { 2788 effectChains[i]->process_l(); 2789 } 2790 } 2791 2792 // Only if the Effects buffer is enabled and there is data in the 2793 // Effects buffer (buffer valid), we need to 2794 // copy into the sink buffer. 2795 // TODO use sleepTime == 0 as an additional condition. 2796 if (mEffectBufferValid) { 2797 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2798 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2799 mNormalFrameCount * mChannelCount); 2800 } 2801 2802 // enable changes in effect chain 2803 unlockEffectChains(effectChains); 2804 2805 if (!waitingAsyncCallback()) { 2806 // sleepTime == 0 means we must write to audio hardware 2807 if (sleepTime == 0) { 2808 if (mBytesRemaining) { 2809 ssize_t ret = threadLoop_write(); 2810 if (ret < 0) { 2811 mBytesRemaining = 0; 2812 } else { 2813 mBytesWritten += ret; 2814 mBytesRemaining -= ret; 2815 } 2816 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2817 (mMixerStatus == MIXER_DRAIN_ALL)) { 2818 threadLoop_drain(); 2819 } 2820 if (mType == MIXER) { 2821 // write blocked detection 2822 nsecs_t now = systemTime(); 2823 nsecs_t delta = now - mLastWriteTime; 2824 if (!mStandby && delta > maxPeriod) { 2825 mNumDelayedWrites++; 2826 if ((now - lastWarning) > kWarningThrottleNs) { 2827 ATRACE_NAME("underrun"); 2828 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2829 ns2ms(delta), mNumDelayedWrites, this); 2830 lastWarning = now; 2831 } 2832 } 2833 } 2834 2835 } else { 2836 ATRACE_BEGIN("sleep"); 2837 usleep(sleepTime); 2838 ATRACE_END(); 2839 } 2840 } 2841 2842 // Finally let go of removed track(s), without the lock held 2843 // since we can't guarantee the destructors won't acquire that 2844 // same lock. This will also mutate and push a new fast mixer state. 2845 threadLoop_removeTracks(tracksToRemove); 2846 tracksToRemove.clear(); 2847 2848 // FIXME I don't understand the need for this here; 2849 // it was in the original code but maybe the 2850 // assignment in saveOutputTracks() makes this unnecessary? 2851 clearOutputTracks(); 2852 2853 // Effect chains will be actually deleted here if they were removed from 2854 // mEffectChains list during mixing or effects processing 2855 effectChains.clear(); 2856 2857 // FIXME Note that the above .clear() is no longer necessary since effectChains 2858 // is now local to this block, but will keep it for now (at least until merge done). 2859 } 2860 2861 threadLoop_exit(); 2862 2863 if (!mStandby) { 2864 threadLoop_standby(); 2865 mStandby = true; 2866 } 2867 2868 releaseWakeLock(); 2869 mWakeLockUids.clear(); 2870 mActiveTracksGeneration++; 2871 2872 ALOGV("Thread %p type %d exiting", this, mType); 2873 return false; 2874} 2875 2876// removeTracks_l() must be called with ThreadBase::mLock held 2877void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2878{ 2879 size_t count = tracksToRemove.size(); 2880 if (count > 0) { 2881 for (size_t i=0 ; i<count ; i++) { 2882 const sp<Track>& track = tracksToRemove.itemAt(i); 2883 mActiveTracks.remove(track); 2884 mWakeLockUids.remove(track->uid()); 2885 mActiveTracksGeneration++; 2886 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2887 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2888 if (chain != 0) { 2889 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2890 track->sessionId()); 2891 chain->decActiveTrackCnt(); 2892 } 2893 if (track->isTerminated()) { 2894 removeTrack_l(track); 2895 } 2896 } 2897 } 2898 2899} 2900 2901status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2902{ 2903 if (mNormalSink != 0) { 2904 return mNormalSink->getTimestamp(timestamp); 2905 } 2906 if ((mType == OFFLOAD || mType == DIRECT) 2907 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2908 uint64_t position64; 2909 int ret = mOutput->stream->get_presentation_position( 2910 mOutput->stream, &position64, ×tamp.mTime); 2911 if (ret == 0) { 2912 timestamp.mPosition = (uint32_t)position64; 2913 return NO_ERROR; 2914 } 2915 } 2916 return INVALID_OPERATION; 2917} 2918 2919status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2920 audio_patch_handle_t *handle) 2921{ 2922 status_t status = NO_ERROR; 2923 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2924 // store new device and send to effects 2925 audio_devices_t type = AUDIO_DEVICE_NONE; 2926 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2927 type |= patch->sinks[i].ext.device.type; 2928 } 2929 mOutDevice = type; 2930 for (size_t i = 0; i < mEffectChains.size(); i++) { 2931 mEffectChains[i]->setDevice_l(mOutDevice); 2932 } 2933 2934 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2935 status = hwDevice->create_audio_patch(hwDevice, 2936 patch->num_sources, 2937 patch->sources, 2938 patch->num_sinks, 2939 patch->sinks, 2940 handle); 2941 } else { 2942 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2943 } 2944 return status; 2945} 2946 2947status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2948{ 2949 status_t status = NO_ERROR; 2950 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2951 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2952 status = hwDevice->release_audio_patch(hwDevice, handle); 2953 } else { 2954 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2955 } 2956 return status; 2957} 2958 2959void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2960{ 2961 Mutex::Autolock _l(mLock); 2962 mTracks.add(track); 2963} 2964 2965void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2966{ 2967 Mutex::Autolock _l(mLock); 2968 destroyTrack_l(track); 2969} 2970 2971void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2972{ 2973 ThreadBase::getAudioPortConfig(config); 2974 config->role = AUDIO_PORT_ROLE_SOURCE; 2975 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2976 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2977} 2978 2979// ---------------------------------------------------------------------------- 2980 2981AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2982 audio_io_handle_t id, audio_devices_t device, type_t type) 2983 : PlaybackThread(audioFlinger, output, id, device, type), 2984 // mAudioMixer below 2985 // mFastMixer below 2986 mFastMixerFutex(0) 2987 // mOutputSink below 2988 // mPipeSink below 2989 // mNormalSink below 2990{ 2991 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2992 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2993 "mFrameCount=%d, mNormalFrameCount=%d", 2994 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2995 mNormalFrameCount); 2996 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2997 2998 if (type == DUPLICATING) { 2999 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3000 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3001 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3002 return; 3003 } 3004 // create an NBAIO sink for the HAL output stream, and negotiate 3005 mOutputSink = new AudioStreamOutSink(output->stream); 3006 size_t numCounterOffers = 0; 3007 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3008 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3009 ALOG_ASSERT(index == 0); 3010 3011 // initialize fast mixer depending on configuration 3012 bool initFastMixer; 3013 switch (kUseFastMixer) { 3014 case FastMixer_Never: 3015 initFastMixer = false; 3016 break; 3017 case FastMixer_Always: 3018 initFastMixer = true; 3019 break; 3020 case FastMixer_Static: 3021 case FastMixer_Dynamic: 3022 initFastMixer = mFrameCount < mNormalFrameCount; 3023 break; 3024 } 3025 if (initFastMixer) { 3026 audio_format_t fastMixerFormat; 3027 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3028 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3029 } else { 3030 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3031 } 3032 if (mFormat != fastMixerFormat) { 3033 // change our Sink format to accept our intermediate precision 3034 mFormat = fastMixerFormat; 3035 free(mSinkBuffer); 3036 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3037 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3038 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3039 } 3040 3041 // create a MonoPipe to connect our submix to FastMixer 3042 NBAIO_Format format = mOutputSink->format(); 3043 NBAIO_Format origformat = format; 3044 // adjust format to match that of the Fast Mixer 3045 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3046 format.mFormat = fastMixerFormat; 3047 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3048 3049 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3050 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3051 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3052 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3053 const NBAIO_Format offers[1] = {format}; 3054 size_t numCounterOffers = 0; 3055 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3056 ALOG_ASSERT(index == 0); 3057 monoPipe->setAvgFrames((mScreenState & 1) ? 3058 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3059 mPipeSink = monoPipe; 3060 3061#ifdef TEE_SINK 3062 if (mTeeSinkOutputEnabled) { 3063 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3064 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3065 const NBAIO_Format offers2[1] = {origformat}; 3066 numCounterOffers = 0; 3067 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3068 ALOG_ASSERT(index == 0); 3069 mTeeSink = teeSink; 3070 PipeReader *teeSource = new PipeReader(*teeSink); 3071 numCounterOffers = 0; 3072 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3073 ALOG_ASSERT(index == 0); 3074 mTeeSource = teeSource; 3075 } 3076#endif 3077 3078 // create fast mixer and configure it initially with just one fast track for our submix 3079 mFastMixer = new FastMixer(); 3080 FastMixerStateQueue *sq = mFastMixer->sq(); 3081#ifdef STATE_QUEUE_DUMP 3082 sq->setObserverDump(&mStateQueueObserverDump); 3083 sq->setMutatorDump(&mStateQueueMutatorDump); 3084#endif 3085 FastMixerState *state = sq->begin(); 3086 FastTrack *fastTrack = &state->mFastTracks[0]; 3087 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3088 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3089 fastTrack->mVolumeProvider = NULL; 3090 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3091 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3092 fastTrack->mGeneration++; 3093 state->mFastTracksGen++; 3094 state->mTrackMask = 1; 3095 // fast mixer will use the HAL output sink 3096 state->mOutputSink = mOutputSink.get(); 3097 state->mOutputSinkGen++; 3098 state->mFrameCount = mFrameCount; 3099 state->mCommand = FastMixerState::COLD_IDLE; 3100 // already done in constructor initialization list 3101 //mFastMixerFutex = 0; 3102 state->mColdFutexAddr = &mFastMixerFutex; 3103 state->mColdGen++; 3104 state->mDumpState = &mFastMixerDumpState; 3105#ifdef TEE_SINK 3106 state->mTeeSink = mTeeSink.get(); 3107#endif 3108 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3109 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3110 sq->end(); 3111 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3112 3113 // start the fast mixer 3114 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3115 pid_t tid = mFastMixer->getTid(); 3116 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3117 if (err != 0) { 3118 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3119 kPriorityFastMixer, getpid_cached, tid, err); 3120 } 3121 3122#ifdef AUDIO_WATCHDOG 3123 // create and start the watchdog 3124 mAudioWatchdog = new AudioWatchdog(); 3125 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3126 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3127 tid = mAudioWatchdog->getTid(); 3128 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3129 if (err != 0) { 3130 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3131 kPriorityFastMixer, getpid_cached, tid, err); 3132 } 3133#endif 3134 3135 } 3136 3137 switch (kUseFastMixer) { 3138 case FastMixer_Never: 3139 case FastMixer_Dynamic: 3140 mNormalSink = mOutputSink; 3141 break; 3142 case FastMixer_Always: 3143 mNormalSink = mPipeSink; 3144 break; 3145 case FastMixer_Static: 3146 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3147 break; 3148 } 3149} 3150 3151AudioFlinger::MixerThread::~MixerThread() 3152{ 3153 if (mFastMixer != 0) { 3154 FastMixerStateQueue *sq = mFastMixer->sq(); 3155 FastMixerState *state = sq->begin(); 3156 if (state->mCommand == FastMixerState::COLD_IDLE) { 3157 int32_t old = android_atomic_inc(&mFastMixerFutex); 3158 if (old == -1) { 3159 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3160 } 3161 } 3162 state->mCommand = FastMixerState::EXIT; 3163 sq->end(); 3164 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3165 mFastMixer->join(); 3166 // Though the fast mixer thread has exited, it's state queue is still valid. 3167 // We'll use that extract the final state which contains one remaining fast track 3168 // corresponding to our sub-mix. 3169 state = sq->begin(); 3170 ALOG_ASSERT(state->mTrackMask == 1); 3171 FastTrack *fastTrack = &state->mFastTracks[0]; 3172 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3173 delete fastTrack->mBufferProvider; 3174 sq->end(false /*didModify*/); 3175 mFastMixer.clear(); 3176#ifdef AUDIO_WATCHDOG 3177 if (mAudioWatchdog != 0) { 3178 mAudioWatchdog->requestExit(); 3179 mAudioWatchdog->requestExitAndWait(); 3180 mAudioWatchdog.clear(); 3181 } 3182#endif 3183 } 3184 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3185 delete mAudioMixer; 3186} 3187 3188 3189uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3190{ 3191 if (mFastMixer != 0) { 3192 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3193 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3194 } 3195 return latency; 3196} 3197 3198 3199void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3200{ 3201 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3202} 3203 3204ssize_t AudioFlinger::MixerThread::threadLoop_write() 3205{ 3206 // FIXME we should only do one push per cycle; confirm this is true 3207 // Start the fast mixer if it's not already running 3208 if (mFastMixer != 0) { 3209 FastMixerStateQueue *sq = mFastMixer->sq(); 3210 FastMixerState *state = sq->begin(); 3211 if (state->mCommand != FastMixerState::MIX_WRITE && 3212 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3213 if (state->mCommand == FastMixerState::COLD_IDLE) { 3214 int32_t old = android_atomic_inc(&mFastMixerFutex); 3215 if (old == -1) { 3216 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3217 } 3218#ifdef AUDIO_WATCHDOG 3219 if (mAudioWatchdog != 0) { 3220 mAudioWatchdog->resume(); 3221 } 3222#endif 3223 } 3224 state->mCommand = FastMixerState::MIX_WRITE; 3225#ifdef FAST_THREAD_STATISTICS 3226 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3227 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3228#endif 3229 sq->end(); 3230 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3231 if (kUseFastMixer == FastMixer_Dynamic) { 3232 mNormalSink = mPipeSink; 3233 } 3234 } else { 3235 sq->end(false /*didModify*/); 3236 } 3237 } 3238 return PlaybackThread::threadLoop_write(); 3239} 3240 3241void AudioFlinger::MixerThread::threadLoop_standby() 3242{ 3243 // Idle the fast mixer if it's currently running 3244 if (mFastMixer != 0) { 3245 FastMixerStateQueue *sq = mFastMixer->sq(); 3246 FastMixerState *state = sq->begin(); 3247 if (!(state->mCommand & FastMixerState::IDLE)) { 3248 state->mCommand = FastMixerState::COLD_IDLE; 3249 state->mColdFutexAddr = &mFastMixerFutex; 3250 state->mColdGen++; 3251 mFastMixerFutex = 0; 3252 sq->end(); 3253 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3254 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3255 if (kUseFastMixer == FastMixer_Dynamic) { 3256 mNormalSink = mOutputSink; 3257 } 3258#ifdef AUDIO_WATCHDOG 3259 if (mAudioWatchdog != 0) { 3260 mAudioWatchdog->pause(); 3261 } 3262#endif 3263 } else { 3264 sq->end(false /*didModify*/); 3265 } 3266 } 3267 PlaybackThread::threadLoop_standby(); 3268} 3269 3270bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3271{ 3272 return false; 3273} 3274 3275bool AudioFlinger::PlaybackThread::shouldStandby_l() 3276{ 3277 return !mStandby; 3278} 3279 3280bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3281{ 3282 Mutex::Autolock _l(mLock); 3283 return waitingAsyncCallback_l(); 3284} 3285 3286// shared by MIXER and DIRECT, overridden by DUPLICATING 3287void AudioFlinger::PlaybackThread::threadLoop_standby() 3288{ 3289 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3290 mOutput->stream->common.standby(&mOutput->stream->common); 3291 if (mUseAsyncWrite != 0) { 3292 // discard any pending drain or write ack by incrementing sequence 3293 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3294 mDrainSequence = (mDrainSequence + 2) & ~1; 3295 ALOG_ASSERT(mCallbackThread != 0); 3296 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3297 mCallbackThread->setDraining(mDrainSequence); 3298 } 3299 mHwPaused = false; 3300} 3301 3302void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3303{ 3304 ALOGV("signal playback thread"); 3305 broadcast_l(); 3306} 3307 3308void AudioFlinger::MixerThread::threadLoop_mix() 3309{ 3310 // obtain the presentation timestamp of the next output buffer 3311 int64_t pts; 3312 status_t status = INVALID_OPERATION; 3313 3314 if (mNormalSink != 0) { 3315 status = mNormalSink->getNextWriteTimestamp(&pts); 3316 } else { 3317 status = mOutputSink->getNextWriteTimestamp(&pts); 3318 } 3319 3320 if (status != NO_ERROR) { 3321 pts = AudioBufferProvider::kInvalidPTS; 3322 } 3323 3324 // mix buffers... 3325 mAudioMixer->process(pts); 3326 mCurrentWriteLength = mSinkBufferSize; 3327 // increase sleep time progressively when application underrun condition clears. 3328 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3329 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3330 // such that we would underrun the audio HAL. 3331 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3332 sleepTimeShift--; 3333 } 3334 sleepTime = 0; 3335 standbyTime = systemTime() + standbyDelay; 3336 //TODO: delay standby when effects have a tail 3337 3338} 3339 3340void AudioFlinger::MixerThread::threadLoop_sleepTime() 3341{ 3342 // If no tracks are ready, sleep once for the duration of an output 3343 // buffer size, then write 0s to the output 3344 if (sleepTime == 0) { 3345 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3346 sleepTime = activeSleepTime >> sleepTimeShift; 3347 if (sleepTime < kMinThreadSleepTimeUs) { 3348 sleepTime = kMinThreadSleepTimeUs; 3349 } 3350 // reduce sleep time in case of consecutive application underruns to avoid 3351 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3352 // duration we would end up writing less data than needed by the audio HAL if 3353 // the condition persists. 3354 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3355 sleepTimeShift++; 3356 } 3357 } else { 3358 sleepTime = idleSleepTime; 3359 } 3360 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3361 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3362 // before effects processing or output. 3363 if (mMixerBufferValid) { 3364 memset(mMixerBuffer, 0, mMixerBufferSize); 3365 } else { 3366 memset(mSinkBuffer, 0, mSinkBufferSize); 3367 } 3368 sleepTime = 0; 3369 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3370 "anticipated start"); 3371 } 3372 // TODO add standby time extension fct of effect tail 3373} 3374 3375// prepareTracks_l() must be called with ThreadBase::mLock held 3376AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3377 Vector< sp<Track> > *tracksToRemove) 3378{ 3379 3380 mixer_state mixerStatus = MIXER_IDLE; 3381 // find out which tracks need to be processed 3382 size_t count = mActiveTracks.size(); 3383 size_t mixedTracks = 0; 3384 size_t tracksWithEffect = 0; 3385 // counts only _active_ fast tracks 3386 size_t fastTracks = 0; 3387 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3388 3389 float masterVolume = mMasterVolume; 3390 bool masterMute = mMasterMute; 3391 3392 if (masterMute) { 3393 masterVolume = 0; 3394 } 3395 // Delegate master volume control to effect in output mix effect chain if needed 3396 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3397 if (chain != 0) { 3398 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3399 chain->setVolume_l(&v, &v); 3400 masterVolume = (float)((v + (1 << 23)) >> 24); 3401 chain.clear(); 3402 } 3403 3404 // prepare a new state to push 3405 FastMixerStateQueue *sq = NULL; 3406 FastMixerState *state = NULL; 3407 bool didModify = false; 3408 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3409 if (mFastMixer != 0) { 3410 sq = mFastMixer->sq(); 3411 state = sq->begin(); 3412 } 3413 3414 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3415 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3416 3417 for (size_t i=0 ; i<count ; i++) { 3418 const sp<Track> t = mActiveTracks[i].promote(); 3419 if (t == 0) { 3420 continue; 3421 } 3422 3423 // this const just means the local variable doesn't change 3424 Track* const track = t.get(); 3425 3426 // process fast tracks 3427 if (track->isFastTrack()) { 3428 3429 // It's theoretically possible (though unlikely) for a fast track to be created 3430 // and then removed within the same normal mix cycle. This is not a problem, as 3431 // the track never becomes active so it's fast mixer slot is never touched. 3432 // The converse, of removing an (active) track and then creating a new track 3433 // at the identical fast mixer slot within the same normal mix cycle, 3434 // is impossible because the slot isn't marked available until the end of each cycle. 3435 int j = track->mFastIndex; 3436 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3437 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3438 FastTrack *fastTrack = &state->mFastTracks[j]; 3439 3440 // Determine whether the track is currently in underrun condition, 3441 // and whether it had a recent underrun. 3442 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3443 FastTrackUnderruns underruns = ftDump->mUnderruns; 3444 uint32_t recentFull = (underruns.mBitFields.mFull - 3445 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3446 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3447 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3448 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3449 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3450 uint32_t recentUnderruns = recentPartial + recentEmpty; 3451 track->mObservedUnderruns = underruns; 3452 // don't count underruns that occur while stopping or pausing 3453 // or stopped which can occur when flush() is called while active 3454 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3455 recentUnderruns > 0) { 3456 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3457 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3458 } 3459 3460 // This is similar to the state machine for normal tracks, 3461 // with a few modifications for fast tracks. 3462 bool isActive = true; 3463 switch (track->mState) { 3464 case TrackBase::STOPPING_1: 3465 // track stays active in STOPPING_1 state until first underrun 3466 if (recentUnderruns > 0 || track->isTerminated()) { 3467 track->mState = TrackBase::STOPPING_2; 3468 } 3469 break; 3470 case TrackBase::PAUSING: 3471 // ramp down is not yet implemented 3472 track->setPaused(); 3473 break; 3474 case TrackBase::RESUMING: 3475 // ramp up is not yet implemented 3476 track->mState = TrackBase::ACTIVE; 3477 break; 3478 case TrackBase::ACTIVE: 3479 if (recentFull > 0 || recentPartial > 0) { 3480 // track has provided at least some frames recently: reset retry count 3481 track->mRetryCount = kMaxTrackRetries; 3482 } 3483 if (recentUnderruns == 0) { 3484 // no recent underruns: stay active 3485 break; 3486 } 3487 // there has recently been an underrun of some kind 3488 if (track->sharedBuffer() == 0) { 3489 // were any of the recent underruns "empty" (no frames available)? 3490 if (recentEmpty == 0) { 3491 // no, then ignore the partial underruns as they are allowed indefinitely 3492 break; 3493 } 3494 // there has recently been an "empty" underrun: decrement the retry counter 3495 if (--(track->mRetryCount) > 0) { 3496 break; 3497 } 3498 // indicate to client process that the track was disabled because of underrun; 3499 // it will then automatically call start() when data is available 3500 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3501 // remove from active list, but state remains ACTIVE [confusing but true] 3502 isActive = false; 3503 break; 3504 } 3505 // fall through 3506 case TrackBase::STOPPING_2: 3507 case TrackBase::PAUSED: 3508 case TrackBase::STOPPED: 3509 case TrackBase::FLUSHED: // flush() while active 3510 // Check for presentation complete if track is inactive 3511 // We have consumed all the buffers of this track. 3512 // This would be incomplete if we auto-paused on underrun 3513 { 3514 size_t audioHALFrames = 3515 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3516 size_t framesWritten = mBytesWritten / mFrameSize; 3517 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3518 // track stays in active list until presentation is complete 3519 break; 3520 } 3521 } 3522 if (track->isStopping_2()) { 3523 track->mState = TrackBase::STOPPED; 3524 } 3525 if (track->isStopped()) { 3526 // Can't reset directly, as fast mixer is still polling this track 3527 // track->reset(); 3528 // So instead mark this track as needing to be reset after push with ack 3529 resetMask |= 1 << i; 3530 } 3531 isActive = false; 3532 break; 3533 case TrackBase::IDLE: 3534 default: 3535 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3536 } 3537 3538 if (isActive) { 3539 // was it previously inactive? 3540 if (!(state->mTrackMask & (1 << j))) { 3541 ExtendedAudioBufferProvider *eabp = track; 3542 VolumeProvider *vp = track; 3543 fastTrack->mBufferProvider = eabp; 3544 fastTrack->mVolumeProvider = vp; 3545 fastTrack->mChannelMask = track->mChannelMask; 3546 fastTrack->mFormat = track->mFormat; 3547 fastTrack->mGeneration++; 3548 state->mTrackMask |= 1 << j; 3549 didModify = true; 3550 // no acknowledgement required for newly active tracks 3551 } 3552 // cache the combined master volume and stream type volume for fast mixer; this 3553 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3554 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3555 ++fastTracks; 3556 } else { 3557 // was it previously active? 3558 if (state->mTrackMask & (1 << j)) { 3559 fastTrack->mBufferProvider = NULL; 3560 fastTrack->mGeneration++; 3561 state->mTrackMask &= ~(1 << j); 3562 didModify = true; 3563 // If any fast tracks were removed, we must wait for acknowledgement 3564 // because we're about to decrement the last sp<> on those tracks. 3565 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3566 } else { 3567 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3568 } 3569 tracksToRemove->add(track); 3570 // Avoids a misleading display in dumpsys 3571 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3572 } 3573 continue; 3574 } 3575 3576 { // local variable scope to avoid goto warning 3577 3578 audio_track_cblk_t* cblk = track->cblk(); 3579 3580 // The first time a track is added we wait 3581 // for all its buffers to be filled before processing it 3582 int name = track->name(); 3583 // make sure that we have enough frames to mix one full buffer. 3584 // enforce this condition only once to enable draining the buffer in case the client 3585 // app does not call stop() and relies on underrun to stop: 3586 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3587 // during last round 3588 size_t desiredFrames; 3589 uint32_t sr = track->sampleRate(); 3590 if (sr == mSampleRate) { 3591 desiredFrames = mNormalFrameCount; 3592 } else { 3593 desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); 3594 // add frames already consumed but not yet released by the resampler 3595 // because mAudioTrackServerProxy->framesReady() will include these frames 3596 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3597#if 0 3598 // the minimum track buffer size is normally twice the number of frames necessary 3599 // to fill one buffer and the resampler should not leave more than one buffer worth 3600 // of unreleased frames after each pass, but just in case... 3601 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3602#endif 3603 } 3604 uint32_t minFrames = 1; 3605 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3606 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3607 minFrames = desiredFrames; 3608 } 3609 3610 size_t framesReady = track->framesReady(); 3611 if (ATRACE_ENABLED()) { 3612 // I wish we had formatted trace names 3613 char traceName[16]; 3614 strcpy(traceName, "nRdy"); 3615 int name = track->name(); 3616 if (AudioMixer::TRACK0 <= name && 3617 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3618 name -= AudioMixer::TRACK0; 3619 traceName[4] = (name / 10) + '0'; 3620 traceName[5] = (name % 10) + '0'; 3621 } else { 3622 traceName[4] = '?'; 3623 traceName[5] = '?'; 3624 } 3625 traceName[6] = '\0'; 3626 ATRACE_INT(traceName, framesReady); 3627 } 3628 if ((framesReady >= minFrames) && track->isReady() && 3629 !track->isPaused() && !track->isTerminated()) 3630 { 3631 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3632 3633 mixedTracks++; 3634 3635 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3636 // there is an effect chain connected to the track 3637 chain.clear(); 3638 if (track->mainBuffer() != mSinkBuffer && 3639 track->mainBuffer() != mMixerBuffer) { 3640 if (mEffectBufferEnabled) { 3641 mEffectBufferValid = true; // Later can set directly. 3642 } 3643 chain = getEffectChain_l(track->sessionId()); 3644 // Delegate volume control to effect in track effect chain if needed 3645 if (chain != 0) { 3646 tracksWithEffect++; 3647 } else { 3648 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3649 "session %d", 3650 name, track->sessionId()); 3651 } 3652 } 3653 3654 3655 int param = AudioMixer::VOLUME; 3656 if (track->mFillingUpStatus == Track::FS_FILLED) { 3657 // no ramp for the first volume setting 3658 track->mFillingUpStatus = Track::FS_ACTIVE; 3659 if (track->mState == TrackBase::RESUMING) { 3660 track->mState = TrackBase::ACTIVE; 3661 param = AudioMixer::RAMP_VOLUME; 3662 } 3663 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3664 // FIXME should not make a decision based on mServer 3665 } else if (cblk->mServer != 0) { 3666 // If the track is stopped before the first frame was mixed, 3667 // do not apply ramp 3668 param = AudioMixer::RAMP_VOLUME; 3669 } 3670 3671 // compute volume for this track 3672 uint32_t vl, vr; // in U8.24 integer format 3673 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3674 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3675 vl = vr = 0; 3676 vlf = vrf = vaf = 0.; 3677 if (track->isPausing()) { 3678 track->setPaused(); 3679 } 3680 } else { 3681 3682 // read original volumes with volume control 3683 float typeVolume = mStreamTypes[track->streamType()].volume; 3684 float v = masterVolume * typeVolume; 3685 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3686 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3687 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3688 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3689 // track volumes come from shared memory, so can't be trusted and must be clamped 3690 if (vlf > GAIN_FLOAT_UNITY) { 3691 ALOGV("Track left volume out of range: %.3g", vlf); 3692 vlf = GAIN_FLOAT_UNITY; 3693 } 3694 if (vrf > GAIN_FLOAT_UNITY) { 3695 ALOGV("Track right volume out of range: %.3g", vrf); 3696 vrf = GAIN_FLOAT_UNITY; 3697 } 3698 // now apply the master volume and stream type volume 3699 vlf *= v; 3700 vrf *= v; 3701 // assuming master volume and stream type volume each go up to 1.0, 3702 // then derive vl and vr as U8.24 versions for the effect chain 3703 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3704 vl = (uint32_t) (scaleto8_24 * vlf); 3705 vr = (uint32_t) (scaleto8_24 * vrf); 3706 // vl and vr are now in U8.24 format 3707 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3708 // send level comes from shared memory and so may be corrupt 3709 if (sendLevel > MAX_GAIN_INT) { 3710 ALOGV("Track send level out of range: %04X", sendLevel); 3711 sendLevel = MAX_GAIN_INT; 3712 } 3713 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3714 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3715 } 3716 3717 // Delegate volume control to effect in track effect chain if needed 3718 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3719 // Do not ramp volume if volume is controlled by effect 3720 param = AudioMixer::VOLUME; 3721 // Update remaining floating point volume levels 3722 vlf = (float)vl / (1 << 24); 3723 vrf = (float)vr / (1 << 24); 3724 track->mHasVolumeController = true; 3725 } else { 3726 // force no volume ramp when volume controller was just disabled or removed 3727 // from effect chain to avoid volume spike 3728 if (track->mHasVolumeController) { 3729 param = AudioMixer::VOLUME; 3730 } 3731 track->mHasVolumeController = false; 3732 } 3733 3734 // XXX: these things DON'T need to be done each time 3735 mAudioMixer->setBufferProvider(name, track); 3736 mAudioMixer->enable(name); 3737 3738 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3739 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3740 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3741 mAudioMixer->setParameter( 3742 name, 3743 AudioMixer::TRACK, 3744 AudioMixer::FORMAT, (void *)track->format()); 3745 mAudioMixer->setParameter( 3746 name, 3747 AudioMixer::TRACK, 3748 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3749 mAudioMixer->setParameter( 3750 name, 3751 AudioMixer::TRACK, 3752 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3753 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3754 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3755 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3756 if (reqSampleRate == 0) { 3757 reqSampleRate = mSampleRate; 3758 } else if (reqSampleRate > maxSampleRate) { 3759 reqSampleRate = maxSampleRate; 3760 } 3761 mAudioMixer->setParameter( 3762 name, 3763 AudioMixer::RESAMPLE, 3764 AudioMixer::SAMPLE_RATE, 3765 (void *)(uintptr_t)reqSampleRate); 3766 /* 3767 * Select the appropriate output buffer for the track. 3768 * 3769 * Tracks with effects go into their own effects chain buffer 3770 * and from there into either mEffectBuffer or mSinkBuffer. 3771 * 3772 * Other tracks can use mMixerBuffer for higher precision 3773 * channel accumulation. If this buffer is enabled 3774 * (mMixerBufferEnabled true), then selected tracks will accumulate 3775 * into it. 3776 * 3777 */ 3778 if (mMixerBufferEnabled 3779 && (track->mainBuffer() == mSinkBuffer 3780 || track->mainBuffer() == mMixerBuffer)) { 3781 mAudioMixer->setParameter( 3782 name, 3783 AudioMixer::TRACK, 3784 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3785 mAudioMixer->setParameter( 3786 name, 3787 AudioMixer::TRACK, 3788 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3789 // TODO: override track->mainBuffer()? 3790 mMixerBufferValid = true; 3791 } else { 3792 mAudioMixer->setParameter( 3793 name, 3794 AudioMixer::TRACK, 3795 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3796 mAudioMixer->setParameter( 3797 name, 3798 AudioMixer::TRACK, 3799 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3800 } 3801 mAudioMixer->setParameter( 3802 name, 3803 AudioMixer::TRACK, 3804 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3805 3806 // reset retry count 3807 track->mRetryCount = kMaxTrackRetries; 3808 3809 // If one track is ready, set the mixer ready if: 3810 // - the mixer was not ready during previous round OR 3811 // - no other track is not ready 3812 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3813 mixerStatus != MIXER_TRACKS_ENABLED) { 3814 mixerStatus = MIXER_TRACKS_READY; 3815 } 3816 } else { 3817 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3818 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3819 } 3820 // clear effect chain input buffer if an active track underruns to avoid sending 3821 // previous audio buffer again to effects 3822 chain = getEffectChain_l(track->sessionId()); 3823 if (chain != 0) { 3824 chain->clearInputBuffer(); 3825 } 3826 3827 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3828 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3829 track->isStopped() || track->isPaused()) { 3830 // We have consumed all the buffers of this track. 3831 // Remove it from the list of active tracks. 3832 // TODO: use actual buffer filling status instead of latency when available from 3833 // audio HAL 3834 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3835 size_t framesWritten = mBytesWritten / mFrameSize; 3836 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3837 if (track->isStopped()) { 3838 track->reset(); 3839 } 3840 tracksToRemove->add(track); 3841 } 3842 } else { 3843 // No buffers for this track. Give it a few chances to 3844 // fill a buffer, then remove it from active list. 3845 if (--(track->mRetryCount) <= 0) { 3846 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3847 tracksToRemove->add(track); 3848 // indicate to client process that the track was disabled because of underrun; 3849 // it will then automatically call start() when data is available 3850 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3851 // If one track is not ready, mark the mixer also not ready if: 3852 // - the mixer was ready during previous round OR 3853 // - no other track is ready 3854 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3855 mixerStatus != MIXER_TRACKS_READY) { 3856 mixerStatus = MIXER_TRACKS_ENABLED; 3857 } 3858 } 3859 mAudioMixer->disable(name); 3860 } 3861 3862 } // local variable scope to avoid goto warning 3863track_is_ready: ; 3864 3865 } 3866 3867 // Push the new FastMixer state if necessary 3868 bool pauseAudioWatchdog = false; 3869 if (didModify) { 3870 state->mFastTracksGen++; 3871 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3872 if (kUseFastMixer == FastMixer_Dynamic && 3873 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3874 state->mCommand = FastMixerState::COLD_IDLE; 3875 state->mColdFutexAddr = &mFastMixerFutex; 3876 state->mColdGen++; 3877 mFastMixerFutex = 0; 3878 if (kUseFastMixer == FastMixer_Dynamic) { 3879 mNormalSink = mOutputSink; 3880 } 3881 // If we go into cold idle, need to wait for acknowledgement 3882 // so that fast mixer stops doing I/O. 3883 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3884 pauseAudioWatchdog = true; 3885 } 3886 } 3887 if (sq != NULL) { 3888 sq->end(didModify); 3889 sq->push(block); 3890 } 3891#ifdef AUDIO_WATCHDOG 3892 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3893 mAudioWatchdog->pause(); 3894 } 3895#endif 3896 3897 // Now perform the deferred reset on fast tracks that have stopped 3898 while (resetMask != 0) { 3899 size_t i = __builtin_ctz(resetMask); 3900 ALOG_ASSERT(i < count); 3901 resetMask &= ~(1 << i); 3902 sp<Track> t = mActiveTracks[i].promote(); 3903 if (t == 0) { 3904 continue; 3905 } 3906 Track* track = t.get(); 3907 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3908 track->reset(); 3909 } 3910 3911 // remove all the tracks that need to be... 3912 removeTracks_l(*tracksToRemove); 3913 3914 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3915 mEffectBufferValid = true; 3916 } 3917 3918 if (mEffectBufferValid) { 3919 // as long as there are effects we should clear the effects buffer, to avoid 3920 // passing a non-clean buffer to the effect chain 3921 memset(mEffectBuffer, 0, mEffectBufferSize); 3922 } 3923 // sink or mix buffer must be cleared if all tracks are connected to an 3924 // effect chain as in this case the mixer will not write to the sink or mix buffer 3925 // and track effects will accumulate into it 3926 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3927 (mixedTracks == 0 && fastTracks > 0))) { 3928 // FIXME as a performance optimization, should remember previous zero status 3929 if (mMixerBufferValid) { 3930 memset(mMixerBuffer, 0, mMixerBufferSize); 3931 // TODO: In testing, mSinkBuffer below need not be cleared because 3932 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3933 // after mixing. 3934 // 3935 // To enforce this guarantee: 3936 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3937 // (mixedTracks == 0 && fastTracks > 0)) 3938 // must imply MIXER_TRACKS_READY. 3939 // Later, we may clear buffers regardless, and skip much of this logic. 3940 } 3941 // FIXME as a performance optimization, should remember previous zero status 3942 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3943 } 3944 3945 // if any fast tracks, then status is ready 3946 mMixerStatusIgnoringFastTracks = mixerStatus; 3947 if (fastTracks > 0) { 3948 mixerStatus = MIXER_TRACKS_READY; 3949 } 3950 return mixerStatus; 3951} 3952 3953// getTrackName_l() must be called with ThreadBase::mLock held 3954int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3955 audio_format_t format, int sessionId) 3956{ 3957 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3958} 3959 3960// deleteTrackName_l() must be called with ThreadBase::mLock held 3961void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3962{ 3963 ALOGV("remove track (%d) and delete from mixer", name); 3964 mAudioMixer->deleteTrackName(name); 3965} 3966 3967// checkForNewParameter_l() must be called with ThreadBase::mLock held 3968bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3969 status_t& status) 3970{ 3971 bool reconfig = false; 3972 3973 status = NO_ERROR; 3974 3975 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3976 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3977 if (mFastMixer != 0) { 3978 FastMixerStateQueue *sq = mFastMixer->sq(); 3979 FastMixerState *state = sq->begin(); 3980 if (!(state->mCommand & FastMixerState::IDLE)) { 3981 previousCommand = state->mCommand; 3982 state->mCommand = FastMixerState::HOT_IDLE; 3983 sq->end(); 3984 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3985 } else { 3986 sq->end(false /*didModify*/); 3987 } 3988 } 3989 3990 AudioParameter param = AudioParameter(keyValuePair); 3991 int value; 3992 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3993 reconfig = true; 3994 } 3995 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3996 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3997 status = BAD_VALUE; 3998 } else { 3999 // no need to save value, since it's constant 4000 reconfig = true; 4001 } 4002 } 4003 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4004 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4005 status = BAD_VALUE; 4006 } else { 4007 // no need to save value, since it's constant 4008 reconfig = true; 4009 } 4010 } 4011 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4012 // do not accept frame count changes if tracks are open as the track buffer 4013 // size depends on frame count and correct behavior would not be guaranteed 4014 // if frame count is changed after track creation 4015 if (!mTracks.isEmpty()) { 4016 status = INVALID_OPERATION; 4017 } else { 4018 reconfig = true; 4019 } 4020 } 4021 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4022#ifdef ADD_BATTERY_DATA 4023 // when changing the audio output device, call addBatteryData to notify 4024 // the change 4025 if (mOutDevice != value) { 4026 uint32_t params = 0; 4027 // check whether speaker is on 4028 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4029 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4030 } 4031 4032 audio_devices_t deviceWithoutSpeaker 4033 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4034 // check if any other device (except speaker) is on 4035 if (value & deviceWithoutSpeaker ) { 4036 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4037 } 4038 4039 if (params != 0) { 4040 addBatteryData(params); 4041 } 4042 } 4043#endif 4044 4045 // forward device change to effects that have requested to be 4046 // aware of attached audio device. 4047 if (value != AUDIO_DEVICE_NONE) { 4048 mOutDevice = value; 4049 for (size_t i = 0; i < mEffectChains.size(); i++) { 4050 mEffectChains[i]->setDevice_l(mOutDevice); 4051 } 4052 } 4053 } 4054 4055 if (status == NO_ERROR) { 4056 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4057 keyValuePair.string()); 4058 if (!mStandby && status == INVALID_OPERATION) { 4059 mOutput->stream->common.standby(&mOutput->stream->common); 4060 mStandby = true; 4061 mBytesWritten = 0; 4062 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4063 keyValuePair.string()); 4064 } 4065 if (status == NO_ERROR && reconfig) { 4066 readOutputParameters_l(); 4067 delete mAudioMixer; 4068 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4069 for (size_t i = 0; i < mTracks.size() ; i++) { 4070 int name = getTrackName_l(mTracks[i]->mChannelMask, 4071 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4072 if (name < 0) { 4073 break; 4074 } 4075 mTracks[i]->mName = name; 4076 } 4077 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4078 } 4079 } 4080 4081 if (!(previousCommand & FastMixerState::IDLE)) { 4082 ALOG_ASSERT(mFastMixer != 0); 4083 FastMixerStateQueue *sq = mFastMixer->sq(); 4084 FastMixerState *state = sq->begin(); 4085 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4086 state->mCommand = previousCommand; 4087 sq->end(); 4088 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4089 } 4090 4091 return reconfig; 4092} 4093 4094 4095void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4096{ 4097 const size_t SIZE = 256; 4098 char buffer[SIZE]; 4099 String8 result; 4100 4101 PlaybackThread::dumpInternals(fd, args); 4102 4103 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4104 4105 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4106 const FastMixerDumpState copy(mFastMixerDumpState); 4107 copy.dump(fd); 4108 4109#ifdef STATE_QUEUE_DUMP 4110 // Similar for state queue 4111 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4112 observerCopy.dump(fd); 4113 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4114 mutatorCopy.dump(fd); 4115#endif 4116 4117#ifdef TEE_SINK 4118 // Write the tee output to a .wav file 4119 dumpTee(fd, mTeeSource, mId); 4120#endif 4121 4122#ifdef AUDIO_WATCHDOG 4123 if (mAudioWatchdog != 0) { 4124 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4125 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4126 wdCopy.dump(fd); 4127 } 4128#endif 4129} 4130 4131uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4132{ 4133 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4134} 4135 4136uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4137{ 4138 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4139} 4140 4141void AudioFlinger::MixerThread::cacheParameters_l() 4142{ 4143 PlaybackThread::cacheParameters_l(); 4144 4145 // FIXME: Relaxed timing because of a certain device that can't meet latency 4146 // Should be reduced to 2x after the vendor fixes the driver issue 4147 // increase threshold again due to low power audio mode. The way this warning 4148 // threshold is calculated and its usefulness should be reconsidered anyway. 4149 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4150} 4151 4152// ---------------------------------------------------------------------------- 4153 4154AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4155 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4156 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4157 // mLeftVolFloat, mRightVolFloat 4158{ 4159} 4160 4161AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4162 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4163 ThreadBase::type_t type) 4164 : PlaybackThread(audioFlinger, output, id, device, type) 4165 // mLeftVolFloat, mRightVolFloat 4166{ 4167} 4168 4169AudioFlinger::DirectOutputThread::~DirectOutputThread() 4170{ 4171} 4172 4173void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4174{ 4175 audio_track_cblk_t* cblk = track->cblk(); 4176 float left, right; 4177 4178 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4179 left = right = 0; 4180 } else { 4181 float typeVolume = mStreamTypes[track->streamType()].volume; 4182 float v = mMasterVolume * typeVolume; 4183 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4184 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4185 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4186 if (left > GAIN_FLOAT_UNITY) { 4187 left = GAIN_FLOAT_UNITY; 4188 } 4189 left *= v; 4190 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4191 if (right > GAIN_FLOAT_UNITY) { 4192 right = GAIN_FLOAT_UNITY; 4193 } 4194 right *= v; 4195 } 4196 4197 if (lastTrack) { 4198 if (left != mLeftVolFloat || right != mRightVolFloat) { 4199 mLeftVolFloat = left; 4200 mRightVolFloat = right; 4201 4202 // Convert volumes from float to 8.24 4203 uint32_t vl = (uint32_t)(left * (1 << 24)); 4204 uint32_t vr = (uint32_t)(right * (1 << 24)); 4205 4206 // Delegate volume control to effect in track effect chain if needed 4207 // only one effect chain can be present on DirectOutputThread, so if 4208 // there is one, the track is connected to it 4209 if (!mEffectChains.isEmpty()) { 4210 mEffectChains[0]->setVolume_l(&vl, &vr); 4211 left = (float)vl / (1 << 24); 4212 right = (float)vr / (1 << 24); 4213 } 4214 if (mOutput->stream->set_volume) { 4215 mOutput->stream->set_volume(mOutput->stream, left, right); 4216 } 4217 } 4218 } 4219} 4220 4221 4222AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4223 Vector< sp<Track> > *tracksToRemove 4224) 4225{ 4226 size_t count = mActiveTracks.size(); 4227 mixer_state mixerStatus = MIXER_IDLE; 4228 bool doHwPause = false; 4229 bool doHwResume = false; 4230 bool flushPending = false; 4231 4232 // find out which tracks need to be processed 4233 for (size_t i = 0; i < count; i++) { 4234 sp<Track> t = mActiveTracks[i].promote(); 4235 // The track died recently 4236 if (t == 0) { 4237 continue; 4238 } 4239 4240 Track* const track = t.get(); 4241 audio_track_cblk_t* cblk = track->cblk(); 4242 // Only consider last track started for volume and mixer state control. 4243 // In theory an older track could underrun and restart after the new one starts 4244 // but as we only care about the transition phase between two tracks on a 4245 // direct output, it is not a problem to ignore the underrun case. 4246 sp<Track> l = mLatestActiveTrack.promote(); 4247 bool last = l.get() == track; 4248 4249 if (mHwSupportsPause && track->isPausing()) { 4250 track->setPaused(); 4251 if (last && !mHwPaused) { 4252 doHwPause = true; 4253 mHwPaused = true; 4254 } 4255 tracksToRemove->add(track); 4256 } else if (track->isFlushPending()) { 4257 track->flushAck(); 4258 if (last) { 4259 flushPending = true; 4260 } 4261 } else if (mHwSupportsPause && track->isResumePending()){ 4262 track->resumeAck(); 4263 if (last) { 4264 if (mHwPaused) { 4265 doHwResume = true; 4266 mHwPaused = false; 4267 } 4268 } 4269 } 4270 4271 // The first time a track is added we wait 4272 // for all its buffers to be filled before processing it. 4273 // Allow draining the buffer in case the client 4274 // app does not call stop() and relies on underrun to stop: 4275 // hence the test on (track->mRetryCount > 1). 4276 // If retryCount<=1 then track is about to underrun and be removed. 4277 uint32_t minFrames; 4278 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4279 && (track->mRetryCount > 1)) { 4280 minFrames = mNormalFrameCount; 4281 } else { 4282 minFrames = 1; 4283 } 4284 4285 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4286 !track->isStopping_2() && !track->isStopped()) 4287 { 4288 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4289 4290 if (track->mFillingUpStatus == Track::FS_FILLED) { 4291 track->mFillingUpStatus = Track::FS_ACTIVE; 4292 // make sure processVolume_l() will apply new volume even if 0 4293 mLeftVolFloat = mRightVolFloat = -1.0; 4294 if (!mHwSupportsPause) { 4295 track->resumeAck(); 4296 } 4297 } 4298 4299 // compute volume for this track 4300 processVolume_l(track, last); 4301 if (last) { 4302 // reset retry count 4303 track->mRetryCount = kMaxTrackRetriesDirect; 4304 mActiveTrack = t; 4305 mixerStatus = MIXER_TRACKS_READY; 4306 if (usesHwAvSync() && mHwPaused) { 4307 doHwResume = true; 4308 mHwPaused = false; 4309 } 4310 } 4311 } else { 4312 // clear effect chain input buffer if the last active track started underruns 4313 // to avoid sending previous audio buffer again to effects 4314 if (!mEffectChains.isEmpty() && last) { 4315 mEffectChains[0]->clearInputBuffer(); 4316 } 4317 if (track->isStopping_1()) { 4318 track->mState = TrackBase::STOPPING_2; 4319 } 4320 if ((track->sharedBuffer() != 0) || track->isStopped() || 4321 track->isStopping_2() || track->isPaused()) { 4322 // We have consumed all the buffers of this track. 4323 // Remove it from the list of active tracks. 4324 size_t audioHALFrames; 4325 if (audio_is_linear_pcm(mFormat)) { 4326 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4327 } else { 4328 audioHALFrames = 0; 4329 } 4330 4331 size_t framesWritten = mBytesWritten / mFrameSize; 4332 if (mStandby || !last || 4333 track->presentationComplete(framesWritten, audioHALFrames)) { 4334 if (track->isStopping_2()) { 4335 track->mState = TrackBase::STOPPED; 4336 } 4337 if (track->isStopped()) { 4338 track->reset(); 4339 } 4340 tracksToRemove->add(track); 4341 } 4342 } else { 4343 // No buffers for this track. Give it a few chances to 4344 // fill a buffer, then remove it from active list. 4345 // Only consider last track started for mixer state control 4346 if (--(track->mRetryCount) <= 0) { 4347 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4348 tracksToRemove->add(track); 4349 // indicate to client process that the track was disabled because of underrun; 4350 // it will then automatically call start() when data is available 4351 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4352 } else if (last) { 4353 mixerStatus = MIXER_TRACKS_ENABLED; 4354 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4355 doHwPause = true; 4356 mHwPaused = true; 4357 } 4358 } 4359 } 4360 } 4361 } 4362 4363 // if an active track did not command a flush, check for pending flush on stopped tracks 4364 if (!flushPending) { 4365 for (size_t i = 0; i < mTracks.size(); i++) { 4366 if (mTracks[i]->isFlushPending()) { 4367 mTracks[i]->flushAck(); 4368 flushPending = true; 4369 } 4370 } 4371 } 4372 4373 // make sure the pause/flush/resume sequence is executed in the right order. 4374 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4375 // before flush and then resume HW. This can happen in case of pause/flush/resume 4376 // if resume is received before pause is executed. 4377 if (mHwSupportsPause && !mStandby && 4378 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4379 mOutput->stream->pause(mOutput->stream); 4380 } 4381 if (flushPending) { 4382 flushHw_l(); 4383 } 4384 if (mHwSupportsPause && !mStandby && doHwResume) { 4385 mOutput->stream->resume(mOutput->stream); 4386 } 4387 // remove all the tracks that need to be... 4388 removeTracks_l(*tracksToRemove); 4389 4390 return mixerStatus; 4391} 4392 4393void AudioFlinger::DirectOutputThread::threadLoop_mix() 4394{ 4395 size_t frameCount = mFrameCount; 4396 int8_t *curBuf = (int8_t *)mSinkBuffer; 4397 // output audio to hardware 4398 while (frameCount) { 4399 AudioBufferProvider::Buffer buffer; 4400 buffer.frameCount = frameCount; 4401 mActiveTrack->getNextBuffer(&buffer); 4402 if (buffer.raw == NULL) { 4403 memset(curBuf, 0, frameCount * mFrameSize); 4404 break; 4405 } 4406 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4407 frameCount -= buffer.frameCount; 4408 curBuf += buffer.frameCount * mFrameSize; 4409 mActiveTrack->releaseBuffer(&buffer); 4410 } 4411 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4412 sleepTime = 0; 4413 standbyTime = systemTime() + standbyDelay; 4414 mActiveTrack.clear(); 4415} 4416 4417void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4418{ 4419 // do not write to HAL when paused 4420 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4421 sleepTime = idleSleepTime; 4422 return; 4423 } 4424 if (sleepTime == 0) { 4425 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4426 sleepTime = activeSleepTime; 4427 } else { 4428 sleepTime = idleSleepTime; 4429 } 4430 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4431 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4432 sleepTime = 0; 4433 } 4434} 4435 4436void AudioFlinger::DirectOutputThread::threadLoop_exit() 4437{ 4438 { 4439 Mutex::Autolock _l(mLock); 4440 bool flushPending = false; 4441 for (size_t i = 0; i < mTracks.size(); i++) { 4442 if (mTracks[i]->isFlushPending()) { 4443 mTracks[i]->flushAck(); 4444 flushPending = true; 4445 } 4446 } 4447 if (flushPending) { 4448 flushHw_l(); 4449 } 4450 } 4451 PlaybackThread::threadLoop_exit(); 4452} 4453 4454// must be called with thread mutex locked 4455bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4456{ 4457 bool trackPaused = false; 4458 4459 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4460 // after a timeout and we will enter standby then. 4461 if (mTracks.size() > 0) { 4462 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4463 } 4464 4465 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused)); 4466} 4467 4468// getTrackName_l() must be called with ThreadBase::mLock held 4469int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4470 audio_format_t format __unused, int sessionId __unused) 4471{ 4472 return 0; 4473} 4474 4475// deleteTrackName_l() must be called with ThreadBase::mLock held 4476void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4477{ 4478} 4479 4480// checkForNewParameter_l() must be called with ThreadBase::mLock held 4481bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4482 status_t& status) 4483{ 4484 bool reconfig = false; 4485 4486 status = NO_ERROR; 4487 4488 AudioParameter param = AudioParameter(keyValuePair); 4489 int value; 4490 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4491 // forward device change to effects that have requested to be 4492 // aware of attached audio device. 4493 if (value != AUDIO_DEVICE_NONE) { 4494 mOutDevice = value; 4495 for (size_t i = 0; i < mEffectChains.size(); i++) { 4496 mEffectChains[i]->setDevice_l(mOutDevice); 4497 } 4498 } 4499 } 4500 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4501 // do not accept frame count changes if tracks are open as the track buffer 4502 // size depends on frame count and correct behavior would not be garantied 4503 // if frame count is changed after track creation 4504 if (!mTracks.isEmpty()) { 4505 status = INVALID_OPERATION; 4506 } else { 4507 reconfig = true; 4508 } 4509 } 4510 if (status == NO_ERROR) { 4511 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4512 keyValuePair.string()); 4513 if (!mStandby && status == INVALID_OPERATION) { 4514 mOutput->stream->common.standby(&mOutput->stream->common); 4515 mStandby = true; 4516 mBytesWritten = 0; 4517 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4518 keyValuePair.string()); 4519 } 4520 if (status == NO_ERROR && reconfig) { 4521 readOutputParameters_l(); 4522 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4523 } 4524 } 4525 4526 return reconfig; 4527} 4528 4529uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4530{ 4531 uint32_t time; 4532 if (audio_is_linear_pcm(mFormat)) { 4533 time = PlaybackThread::activeSleepTimeUs(); 4534 } else { 4535 time = 10000; 4536 } 4537 return time; 4538} 4539 4540uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4541{ 4542 uint32_t time; 4543 if (audio_is_linear_pcm(mFormat)) { 4544 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4545 } else { 4546 time = 10000; 4547 } 4548 return time; 4549} 4550 4551uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4552{ 4553 uint32_t time; 4554 if (audio_is_linear_pcm(mFormat)) { 4555 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4556 } else { 4557 time = 10000; 4558 } 4559 return time; 4560} 4561 4562void AudioFlinger::DirectOutputThread::cacheParameters_l() 4563{ 4564 PlaybackThread::cacheParameters_l(); 4565 4566 // use shorter standby delay as on normal output to release 4567 // hardware resources as soon as possible 4568 if (audio_is_linear_pcm(mFormat)) { 4569 standbyDelay = microseconds(activeSleepTime*2); 4570 } else { 4571 standbyDelay = kOffloadStandbyDelayNs; 4572 } 4573} 4574 4575void AudioFlinger::DirectOutputThread::flushHw_l() 4576{ 4577 if (mOutput->stream->flush != NULL) { 4578 mOutput->stream->flush(mOutput->stream); 4579 } 4580 mHwPaused = false; 4581} 4582 4583// ---------------------------------------------------------------------------- 4584 4585AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4586 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4587 : Thread(false /*canCallJava*/), 4588 mPlaybackThread(playbackThread), 4589 mWriteAckSequence(0), 4590 mDrainSequence(0) 4591{ 4592} 4593 4594AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4595{ 4596} 4597 4598void AudioFlinger::AsyncCallbackThread::onFirstRef() 4599{ 4600 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4601} 4602 4603bool AudioFlinger::AsyncCallbackThread::threadLoop() 4604{ 4605 while (!exitPending()) { 4606 uint32_t writeAckSequence; 4607 uint32_t drainSequence; 4608 4609 { 4610 Mutex::Autolock _l(mLock); 4611 while (!((mWriteAckSequence & 1) || 4612 (mDrainSequence & 1) || 4613 exitPending())) { 4614 mWaitWorkCV.wait(mLock); 4615 } 4616 4617 if (exitPending()) { 4618 break; 4619 } 4620 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4621 mWriteAckSequence, mDrainSequence); 4622 writeAckSequence = mWriteAckSequence; 4623 mWriteAckSequence &= ~1; 4624 drainSequence = mDrainSequence; 4625 mDrainSequence &= ~1; 4626 } 4627 { 4628 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4629 if (playbackThread != 0) { 4630 if (writeAckSequence & 1) { 4631 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4632 } 4633 if (drainSequence & 1) { 4634 playbackThread->resetDraining(drainSequence >> 1); 4635 } 4636 } 4637 } 4638 } 4639 return false; 4640} 4641 4642void AudioFlinger::AsyncCallbackThread::exit() 4643{ 4644 ALOGV("AsyncCallbackThread::exit"); 4645 Mutex::Autolock _l(mLock); 4646 requestExit(); 4647 mWaitWorkCV.broadcast(); 4648} 4649 4650void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4651{ 4652 Mutex::Autolock _l(mLock); 4653 // bit 0 is cleared 4654 mWriteAckSequence = sequence << 1; 4655} 4656 4657void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4658{ 4659 Mutex::Autolock _l(mLock); 4660 // ignore unexpected callbacks 4661 if (mWriteAckSequence & 2) { 4662 mWriteAckSequence |= 1; 4663 mWaitWorkCV.signal(); 4664 } 4665} 4666 4667void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4668{ 4669 Mutex::Autolock _l(mLock); 4670 // bit 0 is cleared 4671 mDrainSequence = sequence << 1; 4672} 4673 4674void AudioFlinger::AsyncCallbackThread::resetDraining() 4675{ 4676 Mutex::Autolock _l(mLock); 4677 // ignore unexpected callbacks 4678 if (mDrainSequence & 2) { 4679 mDrainSequence |= 1; 4680 mWaitWorkCV.signal(); 4681 } 4682} 4683 4684 4685// ---------------------------------------------------------------------------- 4686AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4687 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4688 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4689 mPausedBytesRemaining(0) 4690{ 4691 //FIXME: mStandby should be set to true by ThreadBase constructor 4692 mStandby = true; 4693} 4694 4695void AudioFlinger::OffloadThread::threadLoop_exit() 4696{ 4697 if (mFlushPending || mHwPaused) { 4698 // If a flush is pending or track was paused, just discard buffered data 4699 flushHw_l(); 4700 } else { 4701 mMixerStatus = MIXER_DRAIN_ALL; 4702 threadLoop_drain(); 4703 } 4704 if (mUseAsyncWrite) { 4705 ALOG_ASSERT(mCallbackThread != 0); 4706 mCallbackThread->exit(); 4707 } 4708 PlaybackThread::threadLoop_exit(); 4709} 4710 4711AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4712 Vector< sp<Track> > *tracksToRemove 4713) 4714{ 4715 size_t count = mActiveTracks.size(); 4716 4717 mixer_state mixerStatus = MIXER_IDLE; 4718 bool doHwPause = false; 4719 bool doHwResume = false; 4720 4721 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4722 4723 // find out which tracks need to be processed 4724 for (size_t i = 0; i < count; i++) { 4725 sp<Track> t = mActiveTracks[i].promote(); 4726 // The track died recently 4727 if (t == 0) { 4728 continue; 4729 } 4730 Track* const track = t.get(); 4731 audio_track_cblk_t* cblk = track->cblk(); 4732 // Only consider last track started for volume and mixer state control. 4733 // In theory an older track could underrun and restart after the new one starts 4734 // but as we only care about the transition phase between two tracks on a 4735 // direct output, it is not a problem to ignore the underrun case. 4736 sp<Track> l = mLatestActiveTrack.promote(); 4737 bool last = l.get() == track; 4738 4739 if (track->isInvalid()) { 4740 ALOGW("An invalidated track shouldn't be in active list"); 4741 tracksToRemove->add(track); 4742 continue; 4743 } 4744 4745 if (track->mState == TrackBase::IDLE) { 4746 ALOGW("An idle track shouldn't be in active list"); 4747 continue; 4748 } 4749 4750 if (track->isPausing()) { 4751 track->setPaused(); 4752 if (last) { 4753 if (!mHwPaused) { 4754 doHwPause = true; 4755 mHwPaused = true; 4756 } 4757 // If we were part way through writing the mixbuffer to 4758 // the HAL we must save this until we resume 4759 // BUG - this will be wrong if a different track is made active, 4760 // in that case we want to discard the pending data in the 4761 // mixbuffer and tell the client to present it again when the 4762 // track is resumed 4763 mPausedWriteLength = mCurrentWriteLength; 4764 mPausedBytesRemaining = mBytesRemaining; 4765 mBytesRemaining = 0; // stop writing 4766 } 4767 tracksToRemove->add(track); 4768 } else if (track->isFlushPending()) { 4769 track->flushAck(); 4770 if (last) { 4771 mFlushPending = true; 4772 } 4773 } else if (track->isResumePending()){ 4774 track->resumeAck(); 4775 if (last) { 4776 if (mPausedBytesRemaining) { 4777 // Need to continue write that was interrupted 4778 mCurrentWriteLength = mPausedWriteLength; 4779 mBytesRemaining = mPausedBytesRemaining; 4780 mPausedBytesRemaining = 0; 4781 } 4782 if (mHwPaused) { 4783 doHwResume = true; 4784 mHwPaused = false; 4785 // threadLoop_mix() will handle the case that we need to 4786 // resume an interrupted write 4787 } 4788 // enable write to audio HAL 4789 sleepTime = 0; 4790 4791 // Do not handle new data in this iteration even if track->framesReady() 4792 mixerStatus = MIXER_TRACKS_ENABLED; 4793 } 4794 } else if (track->framesReady() && track->isReady() && 4795 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4796 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4797 if (track->mFillingUpStatus == Track::FS_FILLED) { 4798 track->mFillingUpStatus = Track::FS_ACTIVE; 4799 // make sure processVolume_l() will apply new volume even if 0 4800 mLeftVolFloat = mRightVolFloat = -1.0; 4801 } 4802 4803 if (last) { 4804 sp<Track> previousTrack = mPreviousTrack.promote(); 4805 if (previousTrack != 0) { 4806 if (track != previousTrack.get()) { 4807 // Flush any data still being written from last track 4808 mBytesRemaining = 0; 4809 if (mPausedBytesRemaining) { 4810 // Last track was paused so we also need to flush saved 4811 // mixbuffer state and invalidate track so that it will 4812 // re-submit that unwritten data when it is next resumed 4813 mPausedBytesRemaining = 0; 4814 // Invalidate is a bit drastic - would be more efficient 4815 // to have a flag to tell client that some of the 4816 // previously written data was lost 4817 previousTrack->invalidate(); 4818 } 4819 // flush data already sent to the DSP if changing audio session as audio 4820 // comes from a different source. Also invalidate previous track to force a 4821 // seek when resuming. 4822 if (previousTrack->sessionId() != track->sessionId()) { 4823 previousTrack->invalidate(); 4824 } 4825 } 4826 } 4827 mPreviousTrack = track; 4828 // reset retry count 4829 track->mRetryCount = kMaxTrackRetriesOffload; 4830 mActiveTrack = t; 4831 mixerStatus = MIXER_TRACKS_READY; 4832 } 4833 } else { 4834 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4835 if (track->isStopping_1()) { 4836 // Hardware buffer can hold a large amount of audio so we must 4837 // wait for all current track's data to drain before we say 4838 // that the track is stopped. 4839 if (mBytesRemaining == 0) { 4840 // Only start draining when all data in mixbuffer 4841 // has been written 4842 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4843 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4844 // do not drain if no data was ever sent to HAL (mStandby == true) 4845 if (last && !mStandby) { 4846 // do not modify drain sequence if we are already draining. This happens 4847 // when resuming from pause after drain. 4848 if ((mDrainSequence & 1) == 0) { 4849 sleepTime = 0; 4850 standbyTime = systemTime() + standbyDelay; 4851 mixerStatus = MIXER_DRAIN_TRACK; 4852 mDrainSequence += 2; 4853 } 4854 if (mHwPaused) { 4855 // It is possible to move from PAUSED to STOPPING_1 without 4856 // a resume so we must ensure hardware is running 4857 doHwResume = true; 4858 mHwPaused = false; 4859 } 4860 } 4861 } 4862 } else if (track->isStopping_2()) { 4863 // Drain has completed or we are in standby, signal presentation complete 4864 if (!(mDrainSequence & 1) || !last || mStandby) { 4865 track->mState = TrackBase::STOPPED; 4866 size_t audioHALFrames = 4867 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4868 size_t framesWritten = 4869 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4870 track->presentationComplete(framesWritten, audioHALFrames); 4871 track->reset(); 4872 tracksToRemove->add(track); 4873 } 4874 } else { 4875 // No buffers for this track. Give it a few chances to 4876 // fill a buffer, then remove it from active list. 4877 if (--(track->mRetryCount) <= 0) { 4878 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4879 track->name()); 4880 tracksToRemove->add(track); 4881 // indicate to client process that the track was disabled because of underrun; 4882 // it will then automatically call start() when data is available 4883 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4884 } else if (last){ 4885 mixerStatus = MIXER_TRACKS_ENABLED; 4886 } 4887 } 4888 } 4889 // compute volume for this track 4890 processVolume_l(track, last); 4891 } 4892 4893 // make sure the pause/flush/resume sequence is executed in the right order. 4894 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4895 // before flush and then resume HW. This can happen in case of pause/flush/resume 4896 // if resume is received before pause is executed. 4897 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4898 mOutput->stream->pause(mOutput->stream); 4899 } 4900 if (mFlushPending) { 4901 flushHw_l(); 4902 mFlushPending = false; 4903 } 4904 if (!mStandby && doHwResume) { 4905 mOutput->stream->resume(mOutput->stream); 4906 } 4907 4908 // remove all the tracks that need to be... 4909 removeTracks_l(*tracksToRemove); 4910 4911 return mixerStatus; 4912} 4913 4914// must be called with thread mutex locked 4915bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4916{ 4917 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4918 mWriteAckSequence, mDrainSequence); 4919 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4920 return true; 4921 } 4922 return false; 4923} 4924 4925bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4926{ 4927 Mutex::Autolock _l(mLock); 4928 return waitingAsyncCallback_l(); 4929} 4930 4931void AudioFlinger::OffloadThread::flushHw_l() 4932{ 4933 DirectOutputThread::flushHw_l(); 4934 // Flush anything still waiting in the mixbuffer 4935 mCurrentWriteLength = 0; 4936 mBytesRemaining = 0; 4937 mPausedWriteLength = 0; 4938 mPausedBytesRemaining = 0; 4939 4940 if (mUseAsyncWrite) { 4941 // discard any pending drain or write ack by incrementing sequence 4942 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4943 mDrainSequence = (mDrainSequence + 2) & ~1; 4944 ALOG_ASSERT(mCallbackThread != 0); 4945 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4946 mCallbackThread->setDraining(mDrainSequence); 4947 } 4948} 4949 4950void AudioFlinger::OffloadThread::onAddNewTrack_l() 4951{ 4952 sp<Track> previousTrack = mPreviousTrack.promote(); 4953 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4954 4955 if (previousTrack != 0 && latestTrack != 0 && 4956 (previousTrack->sessionId() != latestTrack->sessionId())) { 4957 mFlushPending = true; 4958 } 4959 PlaybackThread::onAddNewTrack_l(); 4960} 4961 4962// ---------------------------------------------------------------------------- 4963 4964AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4965 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4966 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4967 DUPLICATING), 4968 mWaitTimeMs(UINT_MAX) 4969{ 4970 addOutputTrack(mainThread); 4971} 4972 4973AudioFlinger::DuplicatingThread::~DuplicatingThread() 4974{ 4975 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4976 mOutputTracks[i]->destroy(); 4977 } 4978} 4979 4980void AudioFlinger::DuplicatingThread::threadLoop_mix() 4981{ 4982 // mix buffers... 4983 if (outputsReady(outputTracks)) { 4984 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4985 } else { 4986 if (mMixerBufferValid) { 4987 memset(mMixerBuffer, 0, mMixerBufferSize); 4988 } else { 4989 memset(mSinkBuffer, 0, mSinkBufferSize); 4990 } 4991 } 4992 sleepTime = 0; 4993 writeFrames = mNormalFrameCount; 4994 mCurrentWriteLength = mSinkBufferSize; 4995 standbyTime = systemTime() + standbyDelay; 4996} 4997 4998void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4999{ 5000 if (sleepTime == 0) { 5001 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5002 sleepTime = activeSleepTime; 5003 } else { 5004 sleepTime = idleSleepTime; 5005 } 5006 } else if (mBytesWritten != 0) { 5007 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5008 writeFrames = mNormalFrameCount; 5009 memset(mSinkBuffer, 0, mSinkBufferSize); 5010 } else { 5011 // flush remaining overflow buffers in output tracks 5012 writeFrames = 0; 5013 } 5014 sleepTime = 0; 5015 } 5016} 5017 5018ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5019{ 5020 for (size_t i = 0; i < outputTracks.size(); i++) { 5021 outputTracks[i]->write(mSinkBuffer, writeFrames); 5022 } 5023 mStandby = false; 5024 return (ssize_t)mSinkBufferSize; 5025} 5026 5027void AudioFlinger::DuplicatingThread::threadLoop_standby() 5028{ 5029 // DuplicatingThread implements standby by stopping all tracks 5030 for (size_t i = 0; i < outputTracks.size(); i++) { 5031 outputTracks[i]->stop(); 5032 } 5033} 5034 5035void AudioFlinger::DuplicatingThread::saveOutputTracks() 5036{ 5037 outputTracks = mOutputTracks; 5038} 5039 5040void AudioFlinger::DuplicatingThread::clearOutputTracks() 5041{ 5042 outputTracks.clear(); 5043} 5044 5045void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5046{ 5047 Mutex::Autolock _l(mLock); 5048 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5049 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5050 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5051 const size_t frameCount = 5052 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5053 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5054 // from different OutputTracks and their associated MixerThreads (e.g. one may 5055 // nearly empty and the other may be dropping data). 5056 5057 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5058 this, 5059 mSampleRate, 5060 mFormat, 5061 mChannelMask, 5062 frameCount, 5063 IPCThreadState::self()->getCallingUid()); 5064 if (outputTrack->cblk() != NULL) { 5065 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5066 mOutputTracks.add(outputTrack); 5067 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5068 updateWaitTime_l(); 5069 } 5070} 5071 5072void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5073{ 5074 Mutex::Autolock _l(mLock); 5075 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5076 if (mOutputTracks[i]->thread() == thread) { 5077 mOutputTracks[i]->destroy(); 5078 mOutputTracks.removeAt(i); 5079 updateWaitTime_l(); 5080 return; 5081 } 5082 } 5083 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5084} 5085 5086// caller must hold mLock 5087void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5088{ 5089 mWaitTimeMs = UINT_MAX; 5090 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5091 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5092 if (strong != 0) { 5093 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5094 if (waitTimeMs < mWaitTimeMs) { 5095 mWaitTimeMs = waitTimeMs; 5096 } 5097 } 5098 } 5099} 5100 5101 5102bool AudioFlinger::DuplicatingThread::outputsReady( 5103 const SortedVector< sp<OutputTrack> > &outputTracks) 5104{ 5105 for (size_t i = 0; i < outputTracks.size(); i++) { 5106 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5107 if (thread == 0) { 5108 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5109 outputTracks[i].get()); 5110 return false; 5111 } 5112 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5113 // see note at standby() declaration 5114 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5115 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5116 thread.get()); 5117 return false; 5118 } 5119 } 5120 return true; 5121} 5122 5123uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5124{ 5125 return (mWaitTimeMs * 1000) / 2; 5126} 5127 5128void AudioFlinger::DuplicatingThread::cacheParameters_l() 5129{ 5130 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5131 updateWaitTime_l(); 5132 5133 MixerThread::cacheParameters_l(); 5134} 5135 5136// ---------------------------------------------------------------------------- 5137// Record 5138// ---------------------------------------------------------------------------- 5139 5140AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5141 AudioStreamIn *input, 5142 audio_io_handle_t id, 5143 audio_devices_t outDevice, 5144 audio_devices_t inDevice 5145#ifdef TEE_SINK 5146 , const sp<NBAIO_Sink>& teeSink 5147#endif 5148 ) : 5149 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5150 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5151 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5152 mRsmpInRear(0) 5153#ifdef TEE_SINK 5154 , mTeeSink(teeSink) 5155#endif 5156 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5157 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5158 // mFastCapture below 5159 , mFastCaptureFutex(0) 5160 // mInputSource 5161 // mPipeSink 5162 // mPipeSource 5163 , mPipeFramesP2(0) 5164 // mPipeMemory 5165 // mFastCaptureNBLogWriter 5166 , mFastTrackAvail(false) 5167{ 5168 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5169 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5170 5171 readInputParameters_l(); 5172 5173 // create an NBAIO source for the HAL input stream, and negotiate 5174 mInputSource = new AudioStreamInSource(input->stream); 5175 size_t numCounterOffers = 0; 5176 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5177 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5178 ALOG_ASSERT(index == 0); 5179 5180 // initialize fast capture depending on configuration 5181 bool initFastCapture; 5182 switch (kUseFastCapture) { 5183 case FastCapture_Never: 5184 initFastCapture = false; 5185 break; 5186 case FastCapture_Always: 5187 initFastCapture = true; 5188 break; 5189 case FastCapture_Static: 5190 uint32_t primaryOutputSampleRate; 5191 { 5192 AutoMutex _l(audioFlinger->mHardwareLock); 5193 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5194 } 5195 initFastCapture = 5196 // either capture sample rate is same as (a reasonable) primary output sample rate 5197 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5198 (mSampleRate == primaryOutputSampleRate)) || 5199 // or primary output sample rate is unknown, and capture sample rate is reasonable 5200 ((primaryOutputSampleRate == 0) && 5201 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5202 // and the buffer size is < 12 ms 5203 (mFrameCount * 1000) / mSampleRate < 12; 5204 break; 5205 // case FastCapture_Dynamic: 5206 } 5207 5208 if (initFastCapture) { 5209 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 5210 NBAIO_Format format = mInputSource->format(); 5211 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5212 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5213 void *pipeBuffer; 5214 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5215 sp<IMemory> pipeMemory; 5216 if ((roHeap == 0) || 5217 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5218 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5219 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5220 goto failed; 5221 } 5222 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5223 memset(pipeBuffer, 0, pipeSize); 5224 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5225 const NBAIO_Format offers[1] = {format}; 5226 size_t numCounterOffers = 0; 5227 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5228 ALOG_ASSERT(index == 0); 5229 mPipeSink = pipe; 5230 PipeReader *pipeReader = new PipeReader(*pipe); 5231 numCounterOffers = 0; 5232 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5233 ALOG_ASSERT(index == 0); 5234 mPipeSource = pipeReader; 5235 mPipeFramesP2 = pipeFramesP2; 5236 mPipeMemory = pipeMemory; 5237 5238 // create fast capture 5239 mFastCapture = new FastCapture(); 5240 FastCaptureStateQueue *sq = mFastCapture->sq(); 5241#ifdef STATE_QUEUE_DUMP 5242 // FIXME 5243#endif 5244 FastCaptureState *state = sq->begin(); 5245 state->mCblk = NULL; 5246 state->mInputSource = mInputSource.get(); 5247 state->mInputSourceGen++; 5248 state->mPipeSink = pipe; 5249 state->mPipeSinkGen++; 5250 state->mFrameCount = mFrameCount; 5251 state->mCommand = FastCaptureState::COLD_IDLE; 5252 // already done in constructor initialization list 5253 //mFastCaptureFutex = 0; 5254 state->mColdFutexAddr = &mFastCaptureFutex; 5255 state->mColdGen++; 5256 state->mDumpState = &mFastCaptureDumpState; 5257#ifdef TEE_SINK 5258 // FIXME 5259#endif 5260 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5261 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5262 sq->end(); 5263 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5264 5265 // start the fast capture 5266 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5267 pid_t tid = mFastCapture->getTid(); 5268 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5269 if (err != 0) { 5270 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5271 kPriorityFastCapture, getpid_cached, tid, err); 5272 } 5273 5274#ifdef AUDIO_WATCHDOG 5275 // FIXME 5276#endif 5277 5278 mFastTrackAvail = true; 5279 } 5280failed: ; 5281 5282 // FIXME mNormalSource 5283} 5284 5285 5286AudioFlinger::RecordThread::~RecordThread() 5287{ 5288 if (mFastCapture != 0) { 5289 FastCaptureStateQueue *sq = mFastCapture->sq(); 5290 FastCaptureState *state = sq->begin(); 5291 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5292 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5293 if (old == -1) { 5294 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5295 } 5296 } 5297 state->mCommand = FastCaptureState::EXIT; 5298 sq->end(); 5299 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5300 mFastCapture->join(); 5301 mFastCapture.clear(); 5302 } 5303 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5304 mAudioFlinger->unregisterWriter(mNBLogWriter); 5305 delete[] mRsmpInBuffer; 5306} 5307 5308void AudioFlinger::RecordThread::onFirstRef() 5309{ 5310 run(mThreadName, PRIORITY_URGENT_AUDIO); 5311} 5312 5313bool AudioFlinger::RecordThread::threadLoop() 5314{ 5315 nsecs_t lastWarning = 0; 5316 5317 inputStandBy(); 5318 5319reacquire_wakelock: 5320 sp<RecordTrack> activeTrack; 5321 int activeTracksGen; 5322 { 5323 Mutex::Autolock _l(mLock); 5324 size_t size = mActiveTracks.size(); 5325 activeTracksGen = mActiveTracksGen; 5326 if (size > 0) { 5327 // FIXME an arbitrary choice 5328 activeTrack = mActiveTracks[0]; 5329 acquireWakeLock_l(activeTrack->uid()); 5330 if (size > 1) { 5331 SortedVector<int> tmp; 5332 for (size_t i = 0; i < size; i++) { 5333 tmp.add(mActiveTracks[i]->uid()); 5334 } 5335 updateWakeLockUids_l(tmp); 5336 } 5337 } else { 5338 acquireWakeLock_l(-1); 5339 } 5340 } 5341 5342 // used to request a deferred sleep, to be executed later while mutex is unlocked 5343 uint32_t sleepUs = 0; 5344 5345 // loop while there is work to do 5346 for (;;) { 5347 Vector< sp<EffectChain> > effectChains; 5348 5349 // sleep with mutex unlocked 5350 if (sleepUs > 0) { 5351 ATRACE_BEGIN("sleep"); 5352 usleep(sleepUs); 5353 ATRACE_END(); 5354 sleepUs = 0; 5355 } 5356 5357 // activeTracks accumulates a copy of a subset of mActiveTracks 5358 Vector< sp<RecordTrack> > activeTracks; 5359 5360 // reference to the (first and only) active fast track 5361 sp<RecordTrack> fastTrack; 5362 5363 // reference to a fast track which is about to be removed 5364 sp<RecordTrack> fastTrackToRemove; 5365 5366 { // scope for mLock 5367 Mutex::Autolock _l(mLock); 5368 5369 processConfigEvents_l(); 5370 5371 // check exitPending here because checkForNewParameters_l() and 5372 // checkForNewParameters_l() can temporarily release mLock 5373 if (exitPending()) { 5374 break; 5375 } 5376 5377 // if no active track(s), then standby and release wakelock 5378 size_t size = mActiveTracks.size(); 5379 if (size == 0) { 5380 standbyIfNotAlreadyInStandby(); 5381 // exitPending() can't become true here 5382 releaseWakeLock_l(); 5383 ALOGV("RecordThread: loop stopping"); 5384 // go to sleep 5385 mWaitWorkCV.wait(mLock); 5386 ALOGV("RecordThread: loop starting"); 5387 goto reacquire_wakelock; 5388 } 5389 5390 if (mActiveTracksGen != activeTracksGen) { 5391 activeTracksGen = mActiveTracksGen; 5392 SortedVector<int> tmp; 5393 for (size_t i = 0; i < size; i++) { 5394 tmp.add(mActiveTracks[i]->uid()); 5395 } 5396 updateWakeLockUids_l(tmp); 5397 } 5398 5399 bool doBroadcast = false; 5400 for (size_t i = 0; i < size; ) { 5401 5402 activeTrack = mActiveTracks[i]; 5403 if (activeTrack->isTerminated()) { 5404 if (activeTrack->isFastTrack()) { 5405 ALOG_ASSERT(fastTrackToRemove == 0); 5406 fastTrackToRemove = activeTrack; 5407 } 5408 removeTrack_l(activeTrack); 5409 mActiveTracks.remove(activeTrack); 5410 mActiveTracksGen++; 5411 size--; 5412 continue; 5413 } 5414 5415 TrackBase::track_state activeTrackState = activeTrack->mState; 5416 switch (activeTrackState) { 5417 5418 case TrackBase::PAUSING: 5419 mActiveTracks.remove(activeTrack); 5420 mActiveTracksGen++; 5421 doBroadcast = true; 5422 size--; 5423 continue; 5424 5425 case TrackBase::STARTING_1: 5426 sleepUs = 10000; 5427 i++; 5428 continue; 5429 5430 case TrackBase::STARTING_2: 5431 doBroadcast = true; 5432 mStandby = false; 5433 activeTrack->mState = TrackBase::ACTIVE; 5434 break; 5435 5436 case TrackBase::ACTIVE: 5437 break; 5438 5439 case TrackBase::IDLE: 5440 i++; 5441 continue; 5442 5443 default: 5444 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5445 } 5446 5447 activeTracks.add(activeTrack); 5448 i++; 5449 5450 if (activeTrack->isFastTrack()) { 5451 ALOG_ASSERT(!mFastTrackAvail); 5452 ALOG_ASSERT(fastTrack == 0); 5453 fastTrack = activeTrack; 5454 } 5455 } 5456 if (doBroadcast) { 5457 mStartStopCond.broadcast(); 5458 } 5459 5460 // sleep if there are no active tracks to process 5461 if (activeTracks.size() == 0) { 5462 if (sleepUs == 0) { 5463 sleepUs = kRecordThreadSleepUs; 5464 } 5465 continue; 5466 } 5467 sleepUs = 0; 5468 5469 lockEffectChains_l(effectChains); 5470 } 5471 5472 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5473 5474 size_t size = effectChains.size(); 5475 for (size_t i = 0; i < size; i++) { 5476 // thread mutex is not locked, but effect chain is locked 5477 effectChains[i]->process_l(); 5478 } 5479 5480 // Push a new fast capture state if fast capture is not already running, or cblk change 5481 if (mFastCapture != 0) { 5482 FastCaptureStateQueue *sq = mFastCapture->sq(); 5483 FastCaptureState *state = sq->begin(); 5484 bool didModify = false; 5485 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5486 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5487 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5488 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5489 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5490 if (old == -1) { 5491 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5492 } 5493 } 5494 state->mCommand = FastCaptureState::READ_WRITE; 5495#if 0 // FIXME 5496 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5497 FastThreadDumpState::kSamplingNforLowRamDevice : 5498 FastThreadDumpState::kSamplingN); 5499#endif 5500 didModify = true; 5501 } 5502 audio_track_cblk_t *cblkOld = state->mCblk; 5503 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5504 if (cblkNew != cblkOld) { 5505 state->mCblk = cblkNew; 5506 // block until acked if removing a fast track 5507 if (cblkOld != NULL) { 5508 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5509 } 5510 didModify = true; 5511 } 5512 sq->end(didModify); 5513 if (didModify) { 5514 sq->push(block); 5515#if 0 5516 if (kUseFastCapture == FastCapture_Dynamic) { 5517 mNormalSource = mPipeSource; 5518 } 5519#endif 5520 } 5521 } 5522 5523 // now run the fast track destructor with thread mutex unlocked 5524 fastTrackToRemove.clear(); 5525 5526 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5527 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5528 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5529 // If destination is non-contiguous, first read past the nominal end of buffer, then 5530 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5531 5532 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5533 ssize_t framesRead; 5534 5535 // If an NBAIO source is present, use it to read the normal capture's data 5536 if (mPipeSource != 0) { 5537 size_t framesToRead = mBufferSize / mFrameSize; 5538 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5539 framesToRead, AudioBufferProvider::kInvalidPTS); 5540 if (framesRead == 0) { 5541 // since pipe is non-blocking, simulate blocking input 5542 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5543 } 5544 // otherwise use the HAL / AudioStreamIn directly 5545 } else { 5546 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5547 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5548 if (bytesRead < 0) { 5549 framesRead = bytesRead; 5550 } else { 5551 framesRead = bytesRead / mFrameSize; 5552 } 5553 } 5554 5555 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5556 ALOGE("read failed: framesRead=%d", framesRead); 5557 // Force input into standby so that it tries to recover at next read attempt 5558 inputStandBy(); 5559 sleepUs = kRecordThreadSleepUs; 5560 } 5561 if (framesRead <= 0) { 5562 goto unlock; 5563 } 5564 ALOG_ASSERT(framesRead > 0); 5565 5566 if (mTeeSink != 0) { 5567 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5568 } 5569 // If destination is non-contiguous, we now correct for reading past end of buffer. 5570 { 5571 size_t part1 = mRsmpInFramesP2 - rear; 5572 if ((size_t) framesRead > part1) { 5573 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5574 (framesRead - part1) * mFrameSize); 5575 } 5576 } 5577 rear = mRsmpInRear += framesRead; 5578 5579 size = activeTracks.size(); 5580 // loop over each active track 5581 for (size_t i = 0; i < size; i++) { 5582 activeTrack = activeTracks[i]; 5583 5584 // skip fast tracks, as those are handled directly by FastCapture 5585 if (activeTrack->isFastTrack()) { 5586 continue; 5587 } 5588 5589 enum { 5590 OVERRUN_UNKNOWN, 5591 OVERRUN_TRUE, 5592 OVERRUN_FALSE 5593 } overrun = OVERRUN_UNKNOWN; 5594 5595 // loop over getNextBuffer to handle circular sink 5596 for (;;) { 5597 5598 activeTrack->mSink.frameCount = ~0; 5599 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5600 size_t framesOut = activeTrack->mSink.frameCount; 5601 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5602 5603 int32_t front = activeTrack->mRsmpInFront; 5604 ssize_t filled = rear - front; 5605 size_t framesIn; 5606 5607 if (filled < 0) { 5608 // should not happen, but treat like a massive overrun and re-sync 5609 framesIn = 0; 5610 activeTrack->mRsmpInFront = rear; 5611 overrun = OVERRUN_TRUE; 5612 } else if ((size_t) filled <= mRsmpInFrames) { 5613 framesIn = (size_t) filled; 5614 } else { 5615 // client is not keeping up with server, but give it latest data 5616 framesIn = mRsmpInFrames; 5617 activeTrack->mRsmpInFront = front = rear - framesIn; 5618 overrun = OVERRUN_TRUE; 5619 } 5620 5621 if (framesOut == 0 || framesIn == 0) { 5622 break; 5623 } 5624 5625 if (activeTrack->mResampler == NULL) { 5626 // no resampling 5627 if (framesIn > framesOut) { 5628 framesIn = framesOut; 5629 } else { 5630 framesOut = framesIn; 5631 } 5632 int8_t *dst = activeTrack->mSink.i8; 5633 while (framesIn > 0) { 5634 front &= mRsmpInFramesP2 - 1; 5635 size_t part1 = mRsmpInFramesP2 - front; 5636 if (part1 > framesIn) { 5637 part1 = framesIn; 5638 } 5639 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5640 if (mChannelCount == activeTrack->mChannelCount) { 5641 memcpy(dst, src, part1 * mFrameSize); 5642 } else if (mChannelCount == 1) { 5643 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5644 part1); 5645 } else { 5646 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 5647 (const int16_t *)src, part1); 5648 } 5649 dst += part1 * activeTrack->mFrameSize; 5650 front += part1; 5651 framesIn -= part1; 5652 } 5653 activeTrack->mRsmpInFront += framesOut; 5654 5655 } else { 5656 // resampling 5657 // FIXME framesInNeeded should really be part of resampler API, and should 5658 // depend on the SRC ratio 5659 // to keep mRsmpInBuffer full so resampler always has sufficient input 5660 size_t framesInNeeded; 5661 // FIXME only re-calculate when it changes, and optimize for common ratios 5662 // Do not precompute in/out because floating point is not associative 5663 // e.g. a*b/c != a*(b/c). 5664 const double in(mSampleRate); 5665 const double out(activeTrack->mSampleRate); 5666 framesInNeeded = ceil(framesOut * in / out) + 1; 5667 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5668 framesInNeeded, framesOut, in / out); 5669 // Although we theoretically have framesIn in circular buffer, some of those are 5670 // unreleased frames, and thus must be discounted for purpose of budgeting. 5671 size_t unreleased = activeTrack->mRsmpInUnrel; 5672 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5673 if (framesIn < framesInNeeded) { 5674 ALOGV("not enough to resample: have %u frames in but need %u in to " 5675 "produce %u out given in/out ratio of %.4g", 5676 framesIn, framesInNeeded, framesOut, in / out); 5677 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5678 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5679 if (newFramesOut == 0) { 5680 break; 5681 } 5682 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5683 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5684 framesInNeeded, newFramesOut, out / in); 5685 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5686 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5687 "given in/out ratio of %.4g", 5688 framesIn, framesInNeeded, newFramesOut, in / out); 5689 framesOut = newFramesOut; 5690 } else { 5691 ALOGV("success 1: have %u in and need %u in to produce %u out " 5692 "given in/out ratio of %.4g", 5693 framesIn, framesInNeeded, framesOut, in / out); 5694 } 5695 5696 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5697 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5698 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5699 delete[] activeTrack->mRsmpOutBuffer; 5700 // resampler always outputs stereo 5701 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5702 activeTrack->mRsmpOutFrameCount = framesOut; 5703 } 5704 5705 // resampler accumulates, but we only have one source track 5706 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5707 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5708 // FIXME how about having activeTrack implement this interface itself? 5709 activeTrack->mResamplerBufferProvider 5710 /*this*/ /* AudioBufferProvider* */); 5711 // ditherAndClamp() works as long as all buffers returned by 5712 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5713 if (activeTrack->mChannelCount == 1) { 5714 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5715 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5716 framesOut); 5717 // the resampler always outputs stereo samples: 5718 // do post stereo to mono conversion 5719 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5720 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5721 } else { 5722 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5723 activeTrack->mRsmpOutBuffer, framesOut); 5724 } 5725 // now done with mRsmpOutBuffer 5726 5727 } 5728 5729 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5730 overrun = OVERRUN_FALSE; 5731 } 5732 5733 if (activeTrack->mFramesToDrop == 0) { 5734 if (framesOut > 0) { 5735 activeTrack->mSink.frameCount = framesOut; 5736 activeTrack->releaseBuffer(&activeTrack->mSink); 5737 } 5738 } else { 5739 // FIXME could do a partial drop of framesOut 5740 if (activeTrack->mFramesToDrop > 0) { 5741 activeTrack->mFramesToDrop -= framesOut; 5742 if (activeTrack->mFramesToDrop <= 0) { 5743 activeTrack->clearSyncStartEvent(); 5744 } 5745 } else { 5746 activeTrack->mFramesToDrop += framesOut; 5747 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5748 activeTrack->mSyncStartEvent->isCancelled()) { 5749 ALOGW("Synced record %s, session %d, trigger session %d", 5750 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5751 activeTrack->sessionId(), 5752 (activeTrack->mSyncStartEvent != 0) ? 5753 activeTrack->mSyncStartEvent->triggerSession() : 0); 5754 activeTrack->clearSyncStartEvent(); 5755 } 5756 } 5757 } 5758 5759 if (framesOut == 0) { 5760 break; 5761 } 5762 } 5763 5764 switch (overrun) { 5765 case OVERRUN_TRUE: 5766 // client isn't retrieving buffers fast enough 5767 if (!activeTrack->setOverflow()) { 5768 nsecs_t now = systemTime(); 5769 // FIXME should lastWarning per track? 5770 if ((now - lastWarning) > kWarningThrottleNs) { 5771 ALOGW("RecordThread: buffer overflow"); 5772 lastWarning = now; 5773 } 5774 } 5775 break; 5776 case OVERRUN_FALSE: 5777 activeTrack->clearOverflow(); 5778 break; 5779 case OVERRUN_UNKNOWN: 5780 break; 5781 } 5782 5783 } 5784 5785unlock: 5786 // enable changes in effect chain 5787 unlockEffectChains(effectChains); 5788 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5789 } 5790 5791 standbyIfNotAlreadyInStandby(); 5792 5793 { 5794 Mutex::Autolock _l(mLock); 5795 for (size_t i = 0; i < mTracks.size(); i++) { 5796 sp<RecordTrack> track = mTracks[i]; 5797 track->invalidate(); 5798 } 5799 mActiveTracks.clear(); 5800 mActiveTracksGen++; 5801 mStartStopCond.broadcast(); 5802 } 5803 5804 releaseWakeLock(); 5805 5806 ALOGV("RecordThread %p exiting", this); 5807 return false; 5808} 5809 5810void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5811{ 5812 if (!mStandby) { 5813 inputStandBy(); 5814 mStandby = true; 5815 } 5816} 5817 5818void AudioFlinger::RecordThread::inputStandBy() 5819{ 5820 // Idle the fast capture if it's currently running 5821 if (mFastCapture != 0) { 5822 FastCaptureStateQueue *sq = mFastCapture->sq(); 5823 FastCaptureState *state = sq->begin(); 5824 if (!(state->mCommand & FastCaptureState::IDLE)) { 5825 state->mCommand = FastCaptureState::COLD_IDLE; 5826 state->mColdFutexAddr = &mFastCaptureFutex; 5827 state->mColdGen++; 5828 mFastCaptureFutex = 0; 5829 sq->end(); 5830 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5831 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5832#if 0 5833 if (kUseFastCapture == FastCapture_Dynamic) { 5834 // FIXME 5835 } 5836#endif 5837#ifdef AUDIO_WATCHDOG 5838 // FIXME 5839#endif 5840 } else { 5841 sq->end(false /*didModify*/); 5842 } 5843 } 5844 mInput->stream->common.standby(&mInput->stream->common); 5845} 5846 5847// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5848sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5849 const sp<AudioFlinger::Client>& client, 5850 uint32_t sampleRate, 5851 audio_format_t format, 5852 audio_channel_mask_t channelMask, 5853 size_t *pFrameCount, 5854 int sessionId, 5855 size_t *notificationFrames, 5856 int uid, 5857 IAudioFlinger::track_flags_t *flags, 5858 pid_t tid, 5859 status_t *status) 5860{ 5861 size_t frameCount = *pFrameCount; 5862 sp<RecordTrack> track; 5863 status_t lStatus; 5864 5865 // client expresses a preference for FAST, but we get the final say 5866 if (*flags & IAudioFlinger::TRACK_FAST) { 5867 if ( 5868 // use case: callback handler 5869 (tid != -1) && 5870 // frame count is not specified, or is exactly the pipe depth 5871 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5872 // PCM data 5873 audio_is_linear_pcm(format) && 5874 // native format 5875 (format == mFormat) && 5876 // native channel mask 5877 (channelMask == mChannelMask) && 5878 // native hardware sample rate 5879 (sampleRate == mSampleRate) && 5880 // record thread has an associated fast capture 5881 hasFastCapture() && 5882 // there are sufficient fast track slots available 5883 mFastTrackAvail 5884 ) { 5885 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5886 frameCount, mFrameCount); 5887 } else { 5888 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5889 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5890 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5891 frameCount, mFrameCount, mPipeFramesP2, 5892 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5893 hasFastCapture(), tid, mFastTrackAvail); 5894 *flags &= ~IAudioFlinger::TRACK_FAST; 5895 } 5896 } 5897 5898 // compute track buffer size in frames, and suggest the notification frame count 5899 if (*flags & IAudioFlinger::TRACK_FAST) { 5900 // fast track: frame count is exactly the pipe depth 5901 frameCount = mPipeFramesP2; 5902 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5903 *notificationFrames = mFrameCount; 5904 } else { 5905 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5906 // or 20 ms if there is a fast capture 5907 // TODO This could be a roundupRatio inline, and const 5908 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5909 * sampleRate + mSampleRate - 1) / mSampleRate; 5910 // minimum number of notification periods is at least kMinNotifications, 5911 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5912 static const size_t kMinNotifications = 3; 5913 static const uint32_t kMinMs = 30; 5914 // TODO This could be a roundupRatio inline 5915 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5916 // TODO This could be a roundupRatio inline 5917 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5918 maxNotificationFrames; 5919 const size_t minFrameCount = maxNotificationFrames * 5920 max(kMinNotifications, minNotificationsByMs); 5921 frameCount = max(frameCount, minFrameCount); 5922 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5923 *notificationFrames = maxNotificationFrames; 5924 } 5925 } 5926 *pFrameCount = frameCount; 5927 5928 lStatus = initCheck(); 5929 if (lStatus != NO_ERROR) { 5930 ALOGE("createRecordTrack_l() audio driver not initialized"); 5931 goto Exit; 5932 } 5933 5934 { // scope for mLock 5935 Mutex::Autolock _l(mLock); 5936 5937 track = new RecordTrack(this, client, sampleRate, 5938 format, channelMask, frameCount, NULL, sessionId, uid, 5939 *flags, TrackBase::TYPE_DEFAULT); 5940 5941 lStatus = track->initCheck(); 5942 if (lStatus != NO_ERROR) { 5943 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5944 // track must be cleared from the caller as the caller has the AF lock 5945 goto Exit; 5946 } 5947 mTracks.add(track); 5948 5949 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5950 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5951 mAudioFlinger->btNrecIsOff(); 5952 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5953 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5954 5955 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5956 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5957 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5958 // so ask activity manager to do this on our behalf 5959 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5960 } 5961 } 5962 5963 lStatus = NO_ERROR; 5964 5965Exit: 5966 *status = lStatus; 5967 return track; 5968} 5969 5970status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5971 AudioSystem::sync_event_t event, 5972 int triggerSession) 5973{ 5974 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5975 sp<ThreadBase> strongMe = this; 5976 status_t status = NO_ERROR; 5977 5978 if (event == AudioSystem::SYNC_EVENT_NONE) { 5979 recordTrack->clearSyncStartEvent(); 5980 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5981 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5982 triggerSession, 5983 recordTrack->sessionId(), 5984 syncStartEventCallback, 5985 recordTrack); 5986 // Sync event can be cancelled by the trigger session if the track is not in a 5987 // compatible state in which case we start record immediately 5988 if (recordTrack->mSyncStartEvent->isCancelled()) { 5989 recordTrack->clearSyncStartEvent(); 5990 } else { 5991 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5992 recordTrack->mFramesToDrop = - 5993 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5994 } 5995 } 5996 5997 { 5998 // This section is a rendezvous between binder thread executing start() and RecordThread 5999 AutoMutex lock(mLock); 6000 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6001 if (recordTrack->mState == TrackBase::PAUSING) { 6002 ALOGV("active record track PAUSING -> ACTIVE"); 6003 recordTrack->mState = TrackBase::ACTIVE; 6004 } else { 6005 ALOGV("active record track state %d", recordTrack->mState); 6006 } 6007 return status; 6008 } 6009 6010 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6011 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6012 // or using a separate command thread 6013 recordTrack->mState = TrackBase::STARTING_1; 6014 mActiveTracks.add(recordTrack); 6015 mActiveTracksGen++; 6016 status_t status = NO_ERROR; 6017 if (recordTrack->isExternalTrack()) { 6018 mLock.unlock(); 6019 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6020 mLock.lock(); 6021 // FIXME should verify that recordTrack is still in mActiveTracks 6022 if (status != NO_ERROR) { 6023 mActiveTracks.remove(recordTrack); 6024 mActiveTracksGen++; 6025 recordTrack->clearSyncStartEvent(); 6026 ALOGV("RecordThread::start error %d", status); 6027 return status; 6028 } 6029 } 6030 // Catch up with current buffer indices if thread is already running. 6031 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6032 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6033 // see previously buffered data before it called start(), but with greater risk of overrun. 6034 6035 recordTrack->mRsmpInFront = mRsmpInRear; 6036 recordTrack->mRsmpInUnrel = 0; 6037 // FIXME why reset? 6038 if (recordTrack->mResampler != NULL) { 6039 recordTrack->mResampler->reset(); 6040 } 6041 recordTrack->mState = TrackBase::STARTING_2; 6042 // signal thread to start 6043 mWaitWorkCV.broadcast(); 6044 if (mActiveTracks.indexOf(recordTrack) < 0) { 6045 ALOGV("Record failed to start"); 6046 status = BAD_VALUE; 6047 goto startError; 6048 } 6049 return status; 6050 } 6051 6052startError: 6053 if (recordTrack->isExternalTrack()) { 6054 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6055 } 6056 recordTrack->clearSyncStartEvent(); 6057 // FIXME I wonder why we do not reset the state here? 6058 return status; 6059} 6060 6061void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6062{ 6063 sp<SyncEvent> strongEvent = event.promote(); 6064 6065 if (strongEvent != 0) { 6066 sp<RefBase> ptr = strongEvent->cookie().promote(); 6067 if (ptr != 0) { 6068 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6069 recordTrack->handleSyncStartEvent(strongEvent); 6070 } 6071 } 6072} 6073 6074bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6075 ALOGV("RecordThread::stop"); 6076 AutoMutex _l(mLock); 6077 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6078 return false; 6079 } 6080 // note that threadLoop may still be processing the track at this point [without lock] 6081 recordTrack->mState = TrackBase::PAUSING; 6082 // do not wait for mStartStopCond if exiting 6083 if (exitPending()) { 6084 return true; 6085 } 6086 // FIXME incorrect usage of wait: no explicit predicate or loop 6087 mStartStopCond.wait(mLock); 6088 // if we have been restarted, recordTrack is in mActiveTracks here 6089 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6090 ALOGV("Record stopped OK"); 6091 return true; 6092 } 6093 return false; 6094} 6095 6096bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6097{ 6098 return false; 6099} 6100 6101status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6102{ 6103#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6104 if (!isValidSyncEvent(event)) { 6105 return BAD_VALUE; 6106 } 6107 6108 int eventSession = event->triggerSession(); 6109 status_t ret = NAME_NOT_FOUND; 6110 6111 Mutex::Autolock _l(mLock); 6112 6113 for (size_t i = 0; i < mTracks.size(); i++) { 6114 sp<RecordTrack> track = mTracks[i]; 6115 if (eventSession == track->sessionId()) { 6116 (void) track->setSyncEvent(event); 6117 ret = NO_ERROR; 6118 } 6119 } 6120 return ret; 6121#else 6122 return BAD_VALUE; 6123#endif 6124} 6125 6126// destroyTrack_l() must be called with ThreadBase::mLock held 6127void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6128{ 6129 track->terminate(); 6130 track->mState = TrackBase::STOPPED; 6131 // active tracks are removed by threadLoop() 6132 if (mActiveTracks.indexOf(track) < 0) { 6133 removeTrack_l(track); 6134 } 6135} 6136 6137void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6138{ 6139 mTracks.remove(track); 6140 // need anything related to effects here? 6141 if (track->isFastTrack()) { 6142 ALOG_ASSERT(!mFastTrackAvail); 6143 mFastTrackAvail = true; 6144 } 6145} 6146 6147void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6148{ 6149 dumpInternals(fd, args); 6150 dumpTracks(fd, args); 6151 dumpEffectChains(fd, args); 6152} 6153 6154void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6155{ 6156 dprintf(fd, "\nInput thread %p:\n", this); 6157 6158 if (mActiveTracks.size() > 0) { 6159 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 6160 } else { 6161 dprintf(fd, " No active record clients\n"); 6162 } 6163 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6164 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6165 6166 dumpBase(fd, args); 6167} 6168 6169void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6170{ 6171 const size_t SIZE = 256; 6172 char buffer[SIZE]; 6173 String8 result; 6174 6175 size_t numtracks = mTracks.size(); 6176 size_t numactive = mActiveTracks.size(); 6177 size_t numactiveseen = 0; 6178 dprintf(fd, " %d Tracks", numtracks); 6179 if (numtracks) { 6180 dprintf(fd, " of which %d are active\n", numactive); 6181 RecordTrack::appendDumpHeader(result); 6182 for (size_t i = 0; i < numtracks ; ++i) { 6183 sp<RecordTrack> track = mTracks[i]; 6184 if (track != 0) { 6185 bool active = mActiveTracks.indexOf(track) >= 0; 6186 if (active) { 6187 numactiveseen++; 6188 } 6189 track->dump(buffer, SIZE, active); 6190 result.append(buffer); 6191 } 6192 } 6193 } else { 6194 dprintf(fd, "\n"); 6195 } 6196 6197 if (numactiveseen != numactive) { 6198 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6199 " not in the track list\n"); 6200 result.append(buffer); 6201 RecordTrack::appendDumpHeader(result); 6202 for (size_t i = 0; i < numactive; ++i) { 6203 sp<RecordTrack> track = mActiveTracks[i]; 6204 if (mTracks.indexOf(track) < 0) { 6205 track->dump(buffer, SIZE, true); 6206 result.append(buffer); 6207 } 6208 } 6209 6210 } 6211 write(fd, result.string(), result.size()); 6212} 6213 6214// AudioBufferProvider interface 6215status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6216 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6217{ 6218 RecordTrack *activeTrack = mRecordTrack; 6219 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 6220 if (threadBase == 0) { 6221 buffer->frameCount = 0; 6222 buffer->raw = NULL; 6223 return NOT_ENOUGH_DATA; 6224 } 6225 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6226 int32_t rear = recordThread->mRsmpInRear; 6227 int32_t front = activeTrack->mRsmpInFront; 6228 ssize_t filled = rear - front; 6229 // FIXME should not be P2 (don't want to increase latency) 6230 // FIXME if client not keeping up, discard 6231 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6232 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6233 front &= recordThread->mRsmpInFramesP2 - 1; 6234 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6235 if (part1 > (size_t) filled) { 6236 part1 = filled; 6237 } 6238 size_t ask = buffer->frameCount; 6239 ALOG_ASSERT(ask > 0); 6240 if (part1 > ask) { 6241 part1 = ask; 6242 } 6243 if (part1 == 0) { 6244 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 6245 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 6246 buffer->raw = NULL; 6247 buffer->frameCount = 0; 6248 activeTrack->mRsmpInUnrel = 0; 6249 return NOT_ENOUGH_DATA; 6250 } 6251 6252 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6253 buffer->frameCount = part1; 6254 activeTrack->mRsmpInUnrel = part1; 6255 return NO_ERROR; 6256} 6257 6258// AudioBufferProvider interface 6259void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6260 AudioBufferProvider::Buffer* buffer) 6261{ 6262 RecordTrack *activeTrack = mRecordTrack; 6263 size_t stepCount = buffer->frameCount; 6264 if (stepCount == 0) { 6265 return; 6266 } 6267 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 6268 activeTrack->mRsmpInUnrel -= stepCount; 6269 activeTrack->mRsmpInFront += stepCount; 6270 buffer->raw = NULL; 6271 buffer->frameCount = 0; 6272} 6273 6274bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6275 status_t& status) 6276{ 6277 bool reconfig = false; 6278 6279 status = NO_ERROR; 6280 6281 audio_format_t reqFormat = mFormat; 6282 uint32_t samplingRate = mSampleRate; 6283 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6284 6285 AudioParameter param = AudioParameter(keyValuePair); 6286 int value; 6287 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6288 // channel count change can be requested. Do we mandate the first client defines the 6289 // HAL sampling rate and channel count or do we allow changes on the fly? 6290 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6291 samplingRate = value; 6292 reconfig = true; 6293 } 6294 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6295 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 6296 status = BAD_VALUE; 6297 } else { 6298 reqFormat = (audio_format_t) value; 6299 reconfig = true; 6300 } 6301 } 6302 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6303 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6304 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6305 status = BAD_VALUE; 6306 } else { 6307 channelMask = mask; 6308 reconfig = true; 6309 } 6310 } 6311 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6312 // do not accept frame count changes if tracks are open as the track buffer 6313 // size depends on frame count and correct behavior would not be guaranteed 6314 // if frame count is changed after track creation 6315 if (mActiveTracks.size() > 0) { 6316 status = INVALID_OPERATION; 6317 } else { 6318 reconfig = true; 6319 } 6320 } 6321 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6322 // forward device change to effects that have requested to be 6323 // aware of attached audio device. 6324 for (size_t i = 0; i < mEffectChains.size(); i++) { 6325 mEffectChains[i]->setDevice_l(value); 6326 } 6327 6328 // store input device and output device but do not forward output device to audio HAL. 6329 // Note that status is ignored by the caller for output device 6330 // (see AudioFlinger::setParameters() 6331 if (audio_is_output_devices(value)) { 6332 mOutDevice = value; 6333 status = BAD_VALUE; 6334 } else { 6335 mInDevice = value; 6336 // disable AEC and NS if the device is a BT SCO headset supporting those 6337 // pre processings 6338 if (mTracks.size() > 0) { 6339 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6340 mAudioFlinger->btNrecIsOff(); 6341 for (size_t i = 0; i < mTracks.size(); i++) { 6342 sp<RecordTrack> track = mTracks[i]; 6343 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6344 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6345 } 6346 } 6347 } 6348 } 6349 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6350 mAudioSource != (audio_source_t)value) { 6351 // forward device change to effects that have requested to be 6352 // aware of attached audio device. 6353 for (size_t i = 0; i < mEffectChains.size(); i++) { 6354 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6355 } 6356 mAudioSource = (audio_source_t)value; 6357 } 6358 6359 if (status == NO_ERROR) { 6360 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6361 keyValuePair.string()); 6362 if (status == INVALID_OPERATION) { 6363 inputStandBy(); 6364 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6365 keyValuePair.string()); 6366 } 6367 if (reconfig) { 6368 if (status == BAD_VALUE && 6369 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6370 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6371 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6372 <= (2 * samplingRate)) && 6373 audio_channel_count_from_in_mask( 6374 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6375 (channelMask == AUDIO_CHANNEL_IN_MONO || 6376 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6377 status = NO_ERROR; 6378 } 6379 if (status == NO_ERROR) { 6380 readInputParameters_l(); 6381 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6382 } 6383 } 6384 } 6385 6386 return reconfig; 6387} 6388 6389String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6390{ 6391 Mutex::Autolock _l(mLock); 6392 if (initCheck() != NO_ERROR) { 6393 return String8(); 6394 } 6395 6396 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6397 const String8 out_s8(s); 6398 free(s); 6399 return out_s8; 6400} 6401 6402void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6403 AudioSystem::OutputDescriptor desc; 6404 const void *param2 = NULL; 6405 6406 switch (event) { 6407 case AudioSystem::INPUT_OPENED: 6408 case AudioSystem::INPUT_CONFIG_CHANGED: 6409 desc.channelMask = mChannelMask; 6410 desc.samplingRate = mSampleRate; 6411 desc.format = mFormat; 6412 desc.frameCount = mFrameCount; 6413 desc.latency = 0; 6414 param2 = &desc; 6415 break; 6416 6417 case AudioSystem::INPUT_CLOSED: 6418 default: 6419 break; 6420 } 6421 mAudioFlinger->audioConfigChanged(event, mId, param2); 6422} 6423 6424void AudioFlinger::RecordThread::readInputParameters_l() 6425{ 6426 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6427 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6428 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6429 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6430 mFormat = mHALFormat; 6431 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6432 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6433 } 6434 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6435 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6436 mFrameCount = mBufferSize / mFrameSize; 6437 // This is the formula for calculating the temporary buffer size. 6438 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6439 // 1 full output buffer, regardless of the alignment of the available input. 6440 // The value is somewhat arbitrary, and could probably be even larger. 6441 // A larger value should allow more old data to be read after a track calls start(), 6442 // without increasing latency. 6443 mRsmpInFrames = mFrameCount * 7; 6444 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6445 delete[] mRsmpInBuffer; 6446 6447 // TODO optimize audio capture buffer sizes ... 6448 // Here we calculate the size of the sliding buffer used as a source 6449 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6450 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6451 // be better to have it derived from the pipe depth in the long term. 6452 // The current value is higher than necessary. However it should not add to latency. 6453 6454 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6455 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6456 6457 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6458 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6459} 6460 6461uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6462{ 6463 Mutex::Autolock _l(mLock); 6464 if (initCheck() != NO_ERROR) { 6465 return 0; 6466 } 6467 6468 return mInput->stream->get_input_frames_lost(mInput->stream); 6469} 6470 6471uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6472{ 6473 Mutex::Autolock _l(mLock); 6474 uint32_t result = 0; 6475 if (getEffectChain_l(sessionId) != 0) { 6476 result = EFFECT_SESSION; 6477 } 6478 6479 for (size_t i = 0; i < mTracks.size(); ++i) { 6480 if (sessionId == mTracks[i]->sessionId()) { 6481 result |= TRACK_SESSION; 6482 break; 6483 } 6484 } 6485 6486 return result; 6487} 6488 6489KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6490{ 6491 KeyedVector<int, bool> ids; 6492 Mutex::Autolock _l(mLock); 6493 for (size_t j = 0; j < mTracks.size(); ++j) { 6494 sp<RecordThread::RecordTrack> track = mTracks[j]; 6495 int sessionId = track->sessionId(); 6496 if (ids.indexOfKey(sessionId) < 0) { 6497 ids.add(sessionId, true); 6498 } 6499 } 6500 return ids; 6501} 6502 6503AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6504{ 6505 Mutex::Autolock _l(mLock); 6506 AudioStreamIn *input = mInput; 6507 mInput = NULL; 6508 return input; 6509} 6510 6511// this method must always be called either with ThreadBase mLock held or inside the thread loop 6512audio_stream_t* AudioFlinger::RecordThread::stream() const 6513{ 6514 if (mInput == NULL) { 6515 return NULL; 6516 } 6517 return &mInput->stream->common; 6518} 6519 6520status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6521{ 6522 // only one chain per input thread 6523 if (mEffectChains.size() != 0) { 6524 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6525 return INVALID_OPERATION; 6526 } 6527 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6528 chain->setThread(this); 6529 chain->setInBuffer(NULL); 6530 chain->setOutBuffer(NULL); 6531 6532 checkSuspendOnAddEffectChain_l(chain); 6533 6534 // make sure enabled pre processing effects state is communicated to the HAL as we 6535 // just moved them to a new input stream. 6536 chain->syncHalEffectsState(); 6537 6538 mEffectChains.add(chain); 6539 6540 return NO_ERROR; 6541} 6542 6543size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6544{ 6545 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6546 ALOGW_IF(mEffectChains.size() != 1, 6547 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6548 chain.get(), mEffectChains.size(), this); 6549 if (mEffectChains.size() == 1) { 6550 mEffectChains.removeAt(0); 6551 } 6552 return 0; 6553} 6554 6555status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6556 audio_patch_handle_t *handle) 6557{ 6558 status_t status = NO_ERROR; 6559 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6560 // store new device and send to effects 6561 mInDevice = patch->sources[0].ext.device.type; 6562 for (size_t i = 0; i < mEffectChains.size(); i++) { 6563 mEffectChains[i]->setDevice_l(mInDevice); 6564 } 6565 6566 // disable AEC and NS if the device is a BT SCO headset supporting those 6567 // pre processings 6568 if (mTracks.size() > 0) { 6569 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6570 mAudioFlinger->btNrecIsOff(); 6571 for (size_t i = 0; i < mTracks.size(); i++) { 6572 sp<RecordTrack> track = mTracks[i]; 6573 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6574 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6575 } 6576 } 6577 6578 // store new source and send to effects 6579 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6580 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6581 for (size_t i = 0; i < mEffectChains.size(); i++) { 6582 mEffectChains[i]->setAudioSource_l(mAudioSource); 6583 } 6584 } 6585 6586 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6587 status = hwDevice->create_audio_patch(hwDevice, 6588 patch->num_sources, 6589 patch->sources, 6590 patch->num_sinks, 6591 patch->sinks, 6592 handle); 6593 } else { 6594 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6595 } 6596 return status; 6597} 6598 6599status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6600{ 6601 status_t status = NO_ERROR; 6602 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6603 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6604 status = hwDevice->release_audio_patch(hwDevice, handle); 6605 } else { 6606 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6607 } 6608 return status; 6609} 6610 6611void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6612{ 6613 Mutex::Autolock _l(mLock); 6614 mTracks.add(record); 6615} 6616 6617void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6618{ 6619 Mutex::Autolock _l(mLock); 6620 destroyTrack_l(record); 6621} 6622 6623void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6624{ 6625 ThreadBase::getAudioPortConfig(config); 6626 config->role = AUDIO_PORT_ROLE_SINK; 6627 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6628 config->ext.mix.usecase.source = mAudioSource; 6629} 6630 6631} // namespace android 6632