Threads.cpp revision 69dce3343ffe33d2ba60ab4c6755953a7ec96899
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 "BufferProviders.h" 60#include "FastMixer.h" 61#include "FastCapture.h" 62#include "ServiceUtilities.h" 63#include "SchedulingPolicyService.h" 64 65#ifdef ADD_BATTERY_DATA 66#include <media/IMediaPlayerService.h> 67#include <media/IMediaDeathNotifier.h> 68#endif 69 70#ifdef DEBUG_CPU_USAGE 71#include <cpustats/CentralTendencyStatistics.h> 72#include <cpustats/ThreadCpuUsage.h> 73#endif 74 75// ---------------------------------------------------------------------------- 76 77// Note: the following macro is used for extremely verbose logging message. In 78// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 79// 0; but one side effect of this is to turn all LOGV's as well. Some messages 80// are so verbose that we want to suppress them even when we have ALOG_ASSERT 81// turned on. Do not uncomment the #def below unless you really know what you 82// are doing and want to see all of the extremely verbose messages. 83//#define VERY_VERY_VERBOSE_LOGGING 84#ifdef VERY_VERY_VERBOSE_LOGGING 85#define ALOGVV ALOGV 86#else 87#define ALOGVV(a...) do { } while(0) 88#endif 89 90// TODO: Move these macro/inlines to a header file. 91#define max(a, b) ((a) > (b) ? (a) : (b)) 92template <typename T> 93static inline T min(const T& a, const T& b) 94{ 95 return a < b ? a : b; 96} 97 98#ifndef ARRAY_SIZE 99#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 100#endif 101 102namespace android { 103 104// retry counts for buffer fill timeout 105// 50 * ~20msecs = 1 second 106static const int8_t kMaxTrackRetries = 50; 107static const int8_t kMaxTrackStartupRetries = 50; 108// allow less retry attempts on direct output thread. 109// direct outputs can be a scarce resource in audio hardware and should 110// be released as quickly as possible. 111static const int8_t kMaxTrackRetriesDirect = 2; 112 113// don't warn about blocked writes or record buffer overflows more often than this 114static const nsecs_t kWarningThrottleNs = seconds(5); 115 116// RecordThread loop sleep time upon application overrun or audio HAL read error 117static const int kRecordThreadSleepUs = 5000; 118 119// maximum time to wait in sendConfigEvent_l() for a status to be received 120static const nsecs_t kConfigEventTimeoutNs = seconds(2); 121 122// minimum sleep time for the mixer thread loop when tracks are active but in underrun 123static const uint32_t kMinThreadSleepTimeUs = 5000; 124// maximum divider applied to the active sleep time in the mixer thread loop 125static const uint32_t kMaxThreadSleepTimeShift = 2; 126 127// minimum normal sink buffer size, expressed in milliseconds rather than frames 128static const uint32_t kMinNormalSinkBufferSizeMs = 20; 129// maximum normal sink buffer size 130static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 131 132// Offloaded output thread standby delay: allows track transition without going to standby 133static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 134 135// Whether to use fast mixer 136static const enum { 137 FastMixer_Never, // never initialize or use: for debugging only 138 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 139 // normal mixer multiplier is 1 140 FastMixer_Static, // initialize if needed, then use all the time if initialized, 141 // multiplier is calculated based on min & max normal mixer buffer size 142 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 143 // multiplier is calculated based on min & max normal mixer buffer size 144 // FIXME for FastMixer_Dynamic: 145 // Supporting this option will require fixing HALs that can't handle large writes. 146 // For example, one HAL implementation returns an error from a large write, 147 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 148 // We could either fix the HAL implementations, or provide a wrapper that breaks 149 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 150} kUseFastMixer = FastMixer_Static; 151 152// Whether to use fast capture 153static const enum { 154 FastCapture_Never, // never initialize or use: for debugging only 155 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 156 FastCapture_Static, // initialize if needed, then use all the time if initialized 157} kUseFastCapture = FastCapture_Static; 158 159// Priorities for requestPriority 160static const int kPriorityAudioApp = 2; 161static const int kPriorityFastMixer = 3; 162static const int kPriorityFastCapture = 3; 163 164// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 165// for the track. The client then sub-divides this into smaller buffers for its use. 166// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 167// So for now we just assume that client is double-buffered for fast tracks. 168// FIXME It would be better for client to tell AudioFlinger the value of N, 169// so AudioFlinger could allocate the right amount of memory. 170// See the client's minBufCount and mNotificationFramesAct calculations for details. 171 172// This is the default value, if not specified by property. 173static const int kFastTrackMultiplier = 2; 174 175// The minimum and maximum allowed values 176static const int kFastTrackMultiplierMin = 1; 177static const int kFastTrackMultiplierMax = 2; 178 179// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 180static int sFastTrackMultiplier = kFastTrackMultiplier; 181 182// See Thread::readOnlyHeap(). 183// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 184// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 185// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 186static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 187 188// ---------------------------------------------------------------------------- 189 190static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 191 192static void sFastTrackMultiplierInit() 193{ 194 char value[PROPERTY_VALUE_MAX]; 195 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 196 char *endptr; 197 unsigned long ul = strtoul(value, &endptr, 0); 198 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 199 sFastTrackMultiplier = (int) ul; 200 } 201 } 202} 203 204// ---------------------------------------------------------------------------- 205 206#ifdef ADD_BATTERY_DATA 207// To collect the amplifier usage 208static void addBatteryData(uint32_t params) { 209 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 210 if (service == NULL) { 211 // it already logged 212 return; 213 } 214 215 service->addBatteryData(params); 216} 217#endif 218 219 220// ---------------------------------------------------------------------------- 221// CPU Stats 222// ---------------------------------------------------------------------------- 223 224class CpuStats { 225public: 226 CpuStats(); 227 void sample(const String8 &title); 228#ifdef DEBUG_CPU_USAGE 229private: 230 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 231 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 232 233 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 234 235 int mCpuNum; // thread's current CPU number 236 int mCpukHz; // frequency of thread's current CPU in kHz 237#endif 238}; 239 240CpuStats::CpuStats() 241#ifdef DEBUG_CPU_USAGE 242 : mCpuNum(-1), mCpukHz(-1) 243#endif 244{ 245} 246 247void CpuStats::sample(const String8 &title 248#ifndef DEBUG_CPU_USAGE 249 __unused 250#endif 251 ) { 252#ifdef DEBUG_CPU_USAGE 253 // get current thread's delta CPU time in wall clock ns 254 double wcNs; 255 bool valid = mCpuUsage.sampleAndEnable(wcNs); 256 257 // record sample for wall clock statistics 258 if (valid) { 259 mWcStats.sample(wcNs); 260 } 261 262 // get the current CPU number 263 int cpuNum = sched_getcpu(); 264 265 // get the current CPU frequency in kHz 266 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 267 268 // check if either CPU number or frequency changed 269 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 270 mCpuNum = cpuNum; 271 mCpukHz = cpukHz; 272 // ignore sample for purposes of cycles 273 valid = false; 274 } 275 276 // if no change in CPU number or frequency, then record sample for cycle statistics 277 if (valid && mCpukHz > 0) { 278 double cycles = wcNs * cpukHz * 0.000001; 279 mHzStats.sample(cycles); 280 } 281 282 unsigned n = mWcStats.n(); 283 // mCpuUsage.elapsed() is expensive, so don't call it every loop 284 if ((n & 127) == 1) { 285 long long elapsed = mCpuUsage.elapsed(); 286 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 287 double perLoop = elapsed / (double) n; 288 double perLoop100 = perLoop * 0.01; 289 double perLoop1k = perLoop * 0.001; 290 double mean = mWcStats.mean(); 291 double stddev = mWcStats.stddev(); 292 double minimum = mWcStats.minimum(); 293 double maximum = mWcStats.maximum(); 294 double meanCycles = mHzStats.mean(); 295 double stddevCycles = mHzStats.stddev(); 296 double minCycles = mHzStats.minimum(); 297 double maxCycles = mHzStats.maximum(); 298 mCpuUsage.resetElapsed(); 299 mWcStats.reset(); 300 mHzStats.reset(); 301 ALOGD("CPU usage for %s over past %.1f secs\n" 302 " (%u mixer loops at %.1f mean ms per loop):\n" 303 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 304 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 305 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 306 title.string(), 307 elapsed * .000000001, n, perLoop * .000001, 308 mean * .001, 309 stddev * .001, 310 minimum * .001, 311 maximum * .001, 312 mean / perLoop100, 313 stddev / perLoop100, 314 minimum / perLoop100, 315 maximum / perLoop100, 316 meanCycles / perLoop1k, 317 stddevCycles / perLoop1k, 318 minCycles / perLoop1k, 319 maxCycles / perLoop1k); 320 321 } 322 } 323#endif 324}; 325 326// ---------------------------------------------------------------------------- 327// ThreadBase 328// ---------------------------------------------------------------------------- 329 330// static 331const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 332{ 333 switch (type) { 334 case MIXER: 335 return "MIXER"; 336 case DIRECT: 337 return "DIRECT"; 338 case DUPLICATING: 339 return "DUPLICATING"; 340 case RECORD: 341 return "RECORD"; 342 case OFFLOAD: 343 return "OFFLOAD"; 344 default: 345 return "unknown"; 346 } 347} 348 349String8 devicesToString(audio_devices_t devices) 350{ 351 static const struct mapping { 352 audio_devices_t mDevices; 353 const char * mString; 354 } mappingsOut[] = { 355 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 356 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 357 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 358 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 359 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 360 AUDIO_DEVICE_NONE, "NONE", // must be last 361 }, mappingsIn[] = { 362 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 363 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 364 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 365 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 366 AUDIO_DEVICE_NONE, "NONE", // must be last 367 }; 368 String8 result; 369 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 370 const mapping *entry; 371 if (devices & AUDIO_DEVICE_BIT_IN) { 372 devices &= ~AUDIO_DEVICE_BIT_IN; 373 entry = mappingsIn; 374 } else { 375 entry = mappingsOut; 376 } 377 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 378 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 379 if (devices & entry->mDevices) { 380 if (!result.isEmpty()) { 381 result.append("|"); 382 } 383 result.append(entry->mString); 384 } 385 } 386 if (devices & ~allDevices) { 387 if (!result.isEmpty()) { 388 result.append("|"); 389 } 390 result.appendFormat("0x%X", devices & ~allDevices); 391 } 392 if (result.isEmpty()) { 393 result.append(entry->mString); 394 } 395 return result; 396} 397 398String8 inputFlagsToString(audio_input_flags_t flags) 399{ 400 static const struct mapping { 401 audio_input_flags_t mFlag; 402 const char * mString; 403 } mappings[] = { 404 AUDIO_INPUT_FLAG_FAST, "FAST", 405 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 406 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 407 }; 408 String8 result; 409 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 410 const mapping *entry; 411 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 412 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 413 if (flags & entry->mFlag) { 414 if (!result.isEmpty()) { 415 result.append("|"); 416 } 417 result.append(entry->mString); 418 } 419 } 420 if (flags & ~allFlags) { 421 if (!result.isEmpty()) { 422 result.append("|"); 423 } 424 result.appendFormat("0x%X", flags & ~allFlags); 425 } 426 if (result.isEmpty()) { 427 result.append(entry->mString); 428 } 429 return result; 430} 431 432String8 outputFlagsToString(audio_output_flags_t flags) 433{ 434 static const struct mapping { 435 audio_output_flags_t mFlag; 436 const char * mString; 437 } mappings[] = { 438 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 439 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 440 AUDIO_OUTPUT_FLAG_FAST, "FAST", 441 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 442 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 443 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 444 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 445 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 446 }; 447 String8 result; 448 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 449 const mapping *entry; 450 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 451 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 452 if (flags & entry->mFlag) { 453 if (!result.isEmpty()) { 454 result.append("|"); 455 } 456 result.append(entry->mString); 457 } 458 } 459 if (flags & ~allFlags) { 460 if (!result.isEmpty()) { 461 result.append("|"); 462 } 463 result.appendFormat("0x%X", flags & ~allFlags); 464 } 465 if (result.isEmpty()) { 466 result.append(entry->mString); 467 } 468 return result; 469} 470 471const char *sourceToString(audio_source_t source) 472{ 473 switch (source) { 474 case AUDIO_SOURCE_DEFAULT: return "default"; 475 case AUDIO_SOURCE_MIC: return "mic"; 476 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 477 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 478 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 479 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 480 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 481 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 482 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 483 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 484 case AUDIO_SOURCE_HOTWORD: return "hotword"; 485 default: return "unknown"; 486 } 487} 488 489AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 490 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 491 : Thread(false /*canCallJava*/), 492 mType(type), 493 mAudioFlinger(audioFlinger), 494 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 495 // are set by PlaybackThread::readOutputParameters_l() or 496 // RecordThread::readInputParameters_l() 497 //FIXME: mStandby should be true here. Is this some kind of hack? 498 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 499 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 500 // mName will be set by concrete (non-virtual) subclass 501 mDeathRecipient(new PMDeathRecipient(this)) 502{ 503 memset(&mPatch, 0, sizeof(struct audio_patch)); 504} 505 506AudioFlinger::ThreadBase::~ThreadBase() 507{ 508 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 509 mConfigEvents.clear(); 510 511 // do not lock the mutex in destructor 512 releaseWakeLock_l(); 513 if (mPowerManager != 0) { 514 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 515 binder->unlinkToDeath(mDeathRecipient); 516 } 517} 518 519status_t AudioFlinger::ThreadBase::readyToRun() 520{ 521 status_t status = initCheck(); 522 if (status == NO_ERROR) { 523 ALOGI("AudioFlinger's thread %p ready to run", this); 524 } else { 525 ALOGE("No working audio driver found."); 526 } 527 return status; 528} 529 530void AudioFlinger::ThreadBase::exit() 531{ 532 ALOGV("ThreadBase::exit"); 533 // do any cleanup required for exit to succeed 534 preExit(); 535 { 536 // This lock prevents the following race in thread (uniprocessor for illustration): 537 // if (!exitPending()) { 538 // // context switch from here to exit() 539 // // exit() calls requestExit(), what exitPending() observes 540 // // exit() calls signal(), which is dropped since no waiters 541 // // context switch back from exit() to here 542 // mWaitWorkCV.wait(...); 543 // // now thread is hung 544 // } 545 AutoMutex lock(mLock); 546 requestExit(); 547 mWaitWorkCV.broadcast(); 548 } 549 // When Thread::requestExitAndWait is made virtual and this method is renamed to 550 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 551 requestExitAndWait(); 552} 553 554status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 555{ 556 status_t status; 557 558 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 559 Mutex::Autolock _l(mLock); 560 561 return sendSetParameterConfigEvent_l(keyValuePairs); 562} 563 564// sendConfigEvent_l() must be called with ThreadBase::mLock held 565// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 566status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 567{ 568 status_t status = NO_ERROR; 569 570 mConfigEvents.add(event); 571 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 572 mWaitWorkCV.signal(); 573 mLock.unlock(); 574 { 575 Mutex::Autolock _l(event->mLock); 576 while (event->mWaitStatus) { 577 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 578 event->mStatus = TIMED_OUT; 579 event->mWaitStatus = false; 580 } 581 } 582 status = event->mStatus; 583 } 584 mLock.lock(); 585 return status; 586} 587 588void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event) 589{ 590 Mutex::Autolock _l(mLock); 591 sendIoConfigEvent_l(event); 592} 593 594// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 595void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event) 596{ 597 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event); 598 sendConfigEvent_l(configEvent); 599} 600 601// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 602void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 603{ 604 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 605 sendConfigEvent_l(configEvent); 606} 607 608// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 609status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 610{ 611 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 612 return sendConfigEvent_l(configEvent); 613} 614 615status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 616 const struct audio_patch *patch, 617 audio_patch_handle_t *handle) 618{ 619 Mutex::Autolock _l(mLock); 620 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 621 status_t status = sendConfigEvent_l(configEvent); 622 if (status == NO_ERROR) { 623 CreateAudioPatchConfigEventData *data = 624 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 625 *handle = data->mHandle; 626 } 627 return status; 628} 629 630status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 631 const audio_patch_handle_t handle) 632{ 633 Mutex::Autolock _l(mLock); 634 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 635 return sendConfigEvent_l(configEvent); 636} 637 638 639// post condition: mConfigEvents.isEmpty() 640void AudioFlinger::ThreadBase::processConfigEvents_l() 641{ 642 bool configChanged = false; 643 644 while (!mConfigEvents.isEmpty()) { 645 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 646 sp<ConfigEvent> event = mConfigEvents[0]; 647 mConfigEvents.removeAt(0); 648 switch (event->mType) { 649 case CFG_EVENT_PRIO: { 650 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 651 // FIXME Need to understand why this has to be done asynchronously 652 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 653 true /*asynchronous*/); 654 if (err != 0) { 655 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 656 data->mPrio, data->mPid, data->mTid, err); 657 } 658 } break; 659 case CFG_EVENT_IO: { 660 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 661 ioConfigChanged(data->mEvent); 662 } break; 663 case CFG_EVENT_SET_PARAMETER: { 664 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 665 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 666 configChanged = true; 667 } 668 } break; 669 case CFG_EVENT_CREATE_AUDIO_PATCH: { 670 CreateAudioPatchConfigEventData *data = 671 (CreateAudioPatchConfigEventData *)event->mData.get(); 672 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 673 } break; 674 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 675 ReleaseAudioPatchConfigEventData *data = 676 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 677 event->mStatus = releaseAudioPatch_l(data->mHandle); 678 } break; 679 default: 680 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 681 break; 682 } 683 { 684 Mutex::Autolock _l(event->mLock); 685 if (event->mWaitStatus) { 686 event->mWaitStatus = false; 687 event->mCond.signal(); 688 } 689 } 690 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 691 } 692 693 if (configChanged) { 694 cacheParameters_l(); 695 } 696} 697 698String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 699 String8 s; 700 if (output) { 701 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 702 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 703 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 704 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 705 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 706 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 707 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 708 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 709 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 710 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 711 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 712 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 713 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 714 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 715 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 716 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 717 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 718 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 719 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 720 } else { 721 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 722 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 723 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 724 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 725 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 726 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 727 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 728 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 729 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 730 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 731 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 732 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 733 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 734 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 735 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 736 } 737 int len = s.length(); 738 if (s.length() > 2) { 739 char *str = s.lockBuffer(len); 740 s.unlockBuffer(len - 2); 741 } 742 return s; 743} 744 745void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 746{ 747 const size_t SIZE = 256; 748 char buffer[SIZE]; 749 String8 result; 750 751 bool locked = AudioFlinger::dumpTryLock(mLock); 752 if (!locked) { 753 dprintf(fd, "thread %p may be deadlocked\n", this); 754 } 755 756 dprintf(fd, " Thread name: %s\n", mThreadName); 757 dprintf(fd, " I/O handle: %d\n", mId); 758 dprintf(fd, " TID: %d\n", getTid()); 759 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 760 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 761 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 762 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 763 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 764 dprintf(fd, " Channel count: %u\n", mChannelCount); 765 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 766 channelMaskToString(mChannelMask, mType != RECORD).string()); 767 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 768 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 769 dprintf(fd, " Pending config events:"); 770 size_t numConfig = mConfigEvents.size(); 771 if (numConfig) { 772 for (size_t i = 0; i < numConfig; i++) { 773 mConfigEvents[i]->dump(buffer, SIZE); 774 dprintf(fd, "\n %s", buffer); 775 } 776 dprintf(fd, "\n"); 777 } else { 778 dprintf(fd, " none\n"); 779 } 780 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 781 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 782 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 783 784 if (locked) { 785 mLock.unlock(); 786 } 787} 788 789void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 790{ 791 const size_t SIZE = 256; 792 char buffer[SIZE]; 793 String8 result; 794 795 size_t numEffectChains = mEffectChains.size(); 796 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 797 write(fd, buffer, strlen(buffer)); 798 799 for (size_t i = 0; i < numEffectChains; ++i) { 800 sp<EffectChain> chain = mEffectChains[i]; 801 if (chain != 0) { 802 chain->dump(fd, args); 803 } 804 } 805} 806 807void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 808{ 809 Mutex::Autolock _l(mLock); 810 acquireWakeLock_l(uid); 811} 812 813String16 AudioFlinger::ThreadBase::getWakeLockTag() 814{ 815 switch (mType) { 816 case MIXER: 817 return String16("AudioMix"); 818 case DIRECT: 819 return String16("AudioDirectOut"); 820 case DUPLICATING: 821 return String16("AudioDup"); 822 case RECORD: 823 return String16("AudioIn"); 824 case OFFLOAD: 825 return String16("AudioOffload"); 826 default: 827 ALOG_ASSERT(false); 828 return String16("AudioUnknown"); 829 } 830} 831 832void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 833{ 834 getPowerManager_l(); 835 if (mPowerManager != 0) { 836 sp<IBinder> binder = new BBinder(); 837 status_t status; 838 if (uid >= 0) { 839 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 840 binder, 841 getWakeLockTag(), 842 String16("media"), 843 uid, 844 true /* FIXME force oneway contrary to .aidl */); 845 } else { 846 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 847 binder, 848 getWakeLockTag(), 849 String16("media"), 850 true /* FIXME force oneway contrary to .aidl */); 851 } 852 if (status == NO_ERROR) { 853 mWakeLockToken = binder; 854 } 855 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 856 } 857} 858 859void AudioFlinger::ThreadBase::releaseWakeLock() 860{ 861 Mutex::Autolock _l(mLock); 862 releaseWakeLock_l(); 863} 864 865void AudioFlinger::ThreadBase::releaseWakeLock_l() 866{ 867 if (mWakeLockToken != 0) { 868 ALOGV("releaseWakeLock_l() %s", mThreadName); 869 if (mPowerManager != 0) { 870 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 871 true /* FIXME force oneway contrary to .aidl */); 872 } 873 mWakeLockToken.clear(); 874 } 875} 876 877void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 878 Mutex::Autolock _l(mLock); 879 updateWakeLockUids_l(uids); 880} 881 882void AudioFlinger::ThreadBase::getPowerManager_l() { 883 884 if (mPowerManager == 0) { 885 // use checkService() to avoid blocking if power service is not up yet 886 sp<IBinder> binder = 887 defaultServiceManager()->checkService(String16("power")); 888 if (binder == 0) { 889 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 890 } else { 891 mPowerManager = interface_cast<IPowerManager>(binder); 892 binder->linkToDeath(mDeathRecipient); 893 } 894 } 895} 896 897void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 898 899 getPowerManager_l(); 900 if (mWakeLockToken == NULL) { 901 ALOGE("no wake lock to update!"); 902 return; 903 } 904 if (mPowerManager != 0) { 905 sp<IBinder> binder = new BBinder(); 906 status_t status; 907 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 908 true /* FIXME force oneway contrary to .aidl */); 909 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 910 } 911} 912 913void AudioFlinger::ThreadBase::clearPowerManager() 914{ 915 Mutex::Autolock _l(mLock); 916 releaseWakeLock_l(); 917 mPowerManager.clear(); 918} 919 920void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 921{ 922 sp<ThreadBase> thread = mThread.promote(); 923 if (thread != 0) { 924 thread->clearPowerManager(); 925 } 926 ALOGW("power manager service died !!!"); 927} 928 929void AudioFlinger::ThreadBase::setEffectSuspended( 930 const effect_uuid_t *type, bool suspend, int sessionId) 931{ 932 Mutex::Autolock _l(mLock); 933 setEffectSuspended_l(type, suspend, sessionId); 934} 935 936void AudioFlinger::ThreadBase::setEffectSuspended_l( 937 const effect_uuid_t *type, bool suspend, int sessionId) 938{ 939 sp<EffectChain> chain = getEffectChain_l(sessionId); 940 if (chain != 0) { 941 if (type != NULL) { 942 chain->setEffectSuspended_l(type, suspend); 943 } else { 944 chain->setEffectSuspendedAll_l(suspend); 945 } 946 } 947 948 updateSuspendedSessions_l(type, suspend, sessionId); 949} 950 951void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 952{ 953 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 954 if (index < 0) { 955 return; 956 } 957 958 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 959 mSuspendedSessions.valueAt(index); 960 961 for (size_t i = 0; i < sessionEffects.size(); i++) { 962 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 963 for (int j = 0; j < desc->mRefCount; j++) { 964 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 965 chain->setEffectSuspendedAll_l(true); 966 } else { 967 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 968 desc->mType.timeLow); 969 chain->setEffectSuspended_l(&desc->mType, true); 970 } 971 } 972 } 973} 974 975void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 976 bool suspend, 977 int sessionId) 978{ 979 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 980 981 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 982 983 if (suspend) { 984 if (index >= 0) { 985 sessionEffects = mSuspendedSessions.valueAt(index); 986 } else { 987 mSuspendedSessions.add(sessionId, sessionEffects); 988 } 989 } else { 990 if (index < 0) { 991 return; 992 } 993 sessionEffects = mSuspendedSessions.valueAt(index); 994 } 995 996 997 int key = EffectChain::kKeyForSuspendAll; 998 if (type != NULL) { 999 key = type->timeLow; 1000 } 1001 index = sessionEffects.indexOfKey(key); 1002 1003 sp<SuspendedSessionDesc> desc; 1004 if (suspend) { 1005 if (index >= 0) { 1006 desc = sessionEffects.valueAt(index); 1007 } else { 1008 desc = new SuspendedSessionDesc(); 1009 if (type != NULL) { 1010 desc->mType = *type; 1011 } 1012 sessionEffects.add(key, desc); 1013 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1014 } 1015 desc->mRefCount++; 1016 } else { 1017 if (index < 0) { 1018 return; 1019 } 1020 desc = sessionEffects.valueAt(index); 1021 if (--desc->mRefCount == 0) { 1022 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1023 sessionEffects.removeItemsAt(index); 1024 if (sessionEffects.isEmpty()) { 1025 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1026 sessionId); 1027 mSuspendedSessions.removeItem(sessionId); 1028 } 1029 } 1030 } 1031 if (!sessionEffects.isEmpty()) { 1032 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1033 } 1034} 1035 1036void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1037 bool enabled, 1038 int sessionId) 1039{ 1040 Mutex::Autolock _l(mLock); 1041 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1042} 1043 1044void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1045 bool enabled, 1046 int sessionId) 1047{ 1048 if (mType != RECORD) { 1049 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1050 // another session. This gives the priority to well behaved effect control panels 1051 // and applications not using global effects. 1052 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1053 // global effects 1054 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1055 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1056 } 1057 } 1058 1059 sp<EffectChain> chain = getEffectChain_l(sessionId); 1060 if (chain != 0) { 1061 chain->checkSuspendOnEffectEnabled(effect, enabled); 1062 } 1063} 1064 1065// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1066sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1067 const sp<AudioFlinger::Client>& client, 1068 const sp<IEffectClient>& effectClient, 1069 int32_t priority, 1070 int sessionId, 1071 effect_descriptor_t *desc, 1072 int *enabled, 1073 status_t *status) 1074{ 1075 sp<EffectModule> effect; 1076 sp<EffectHandle> handle; 1077 status_t lStatus; 1078 sp<EffectChain> chain; 1079 bool chainCreated = false; 1080 bool effectCreated = false; 1081 bool effectRegistered = false; 1082 1083 lStatus = initCheck(); 1084 if (lStatus != NO_ERROR) { 1085 ALOGW("createEffect_l() Audio driver not initialized."); 1086 goto Exit; 1087 } 1088 1089 // Reject any effect on Direct output threads for now, since the format of 1090 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1091 if (mType == DIRECT) { 1092 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1093 desc->name, mThreadName); 1094 lStatus = BAD_VALUE; 1095 goto Exit; 1096 } 1097 1098 // Reject any effect on mixer or duplicating multichannel sinks. 1099 // TODO: fix both format and multichannel issues with effects. 1100 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1101 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1102 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1103 lStatus = BAD_VALUE; 1104 goto Exit; 1105 } 1106 1107 // Allow global effects only on offloaded and mixer threads 1108 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1109 switch (mType) { 1110 case MIXER: 1111 case OFFLOAD: 1112 break; 1113 case DIRECT: 1114 case DUPLICATING: 1115 case RECORD: 1116 default: 1117 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1118 desc->name, mThreadName); 1119 lStatus = BAD_VALUE; 1120 goto Exit; 1121 } 1122 } 1123 1124 // Only Pre processor effects are allowed on input threads and only on input threads 1125 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1126 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1127 desc->name, desc->flags, mType); 1128 lStatus = BAD_VALUE; 1129 goto Exit; 1130 } 1131 1132 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1133 1134 { // scope for mLock 1135 Mutex::Autolock _l(mLock); 1136 1137 // check for existing effect chain with the requested audio session 1138 chain = getEffectChain_l(sessionId); 1139 if (chain == 0) { 1140 // create a new chain for this session 1141 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1142 chain = new EffectChain(this, sessionId); 1143 addEffectChain_l(chain); 1144 chain->setStrategy(getStrategyForSession_l(sessionId)); 1145 chainCreated = true; 1146 } else { 1147 effect = chain->getEffectFromDesc_l(desc); 1148 } 1149 1150 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1151 1152 if (effect == 0) { 1153 int id = mAudioFlinger->nextUniqueId(); 1154 // Check CPU and memory usage 1155 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1156 if (lStatus != NO_ERROR) { 1157 goto Exit; 1158 } 1159 effectRegistered = true; 1160 // create a new effect module if none present in the chain 1161 effect = new EffectModule(this, chain, desc, id, sessionId); 1162 lStatus = effect->status(); 1163 if (lStatus != NO_ERROR) { 1164 goto Exit; 1165 } 1166 effect->setOffloaded(mType == OFFLOAD, mId); 1167 1168 lStatus = chain->addEffect_l(effect); 1169 if (lStatus != NO_ERROR) { 1170 goto Exit; 1171 } 1172 effectCreated = true; 1173 1174 effect->setDevice(mOutDevice); 1175 effect->setDevice(mInDevice); 1176 effect->setMode(mAudioFlinger->getMode()); 1177 effect->setAudioSource(mAudioSource); 1178 } 1179 // create effect handle and connect it to effect module 1180 handle = new EffectHandle(effect, client, effectClient, priority); 1181 lStatus = handle->initCheck(); 1182 if (lStatus == OK) { 1183 lStatus = effect->addHandle(handle.get()); 1184 } 1185 if (enabled != NULL) { 1186 *enabled = (int)effect->isEnabled(); 1187 } 1188 } 1189 1190Exit: 1191 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1192 Mutex::Autolock _l(mLock); 1193 if (effectCreated) { 1194 chain->removeEffect_l(effect); 1195 } 1196 if (effectRegistered) { 1197 AudioSystem::unregisterEffect(effect->id()); 1198 } 1199 if (chainCreated) { 1200 removeEffectChain_l(chain); 1201 } 1202 handle.clear(); 1203 } 1204 1205 *status = lStatus; 1206 return handle; 1207} 1208 1209sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1210{ 1211 Mutex::Autolock _l(mLock); 1212 return getEffect_l(sessionId, effectId); 1213} 1214 1215sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1216{ 1217 sp<EffectChain> chain = getEffectChain_l(sessionId); 1218 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1219} 1220 1221// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1222// PlaybackThread::mLock held 1223status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1224{ 1225 // check for existing effect chain with the requested audio session 1226 int sessionId = effect->sessionId(); 1227 sp<EffectChain> chain = getEffectChain_l(sessionId); 1228 bool chainCreated = false; 1229 1230 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1231 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1232 this, effect->desc().name, effect->desc().flags); 1233 1234 if (chain == 0) { 1235 // create a new chain for this session 1236 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1237 chain = new EffectChain(this, sessionId); 1238 addEffectChain_l(chain); 1239 chain->setStrategy(getStrategyForSession_l(sessionId)); 1240 chainCreated = true; 1241 } 1242 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1243 1244 if (chain->getEffectFromId_l(effect->id()) != 0) { 1245 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1246 this, effect->desc().name, chain.get()); 1247 return BAD_VALUE; 1248 } 1249 1250 effect->setOffloaded(mType == OFFLOAD, mId); 1251 1252 status_t status = chain->addEffect_l(effect); 1253 if (status != NO_ERROR) { 1254 if (chainCreated) { 1255 removeEffectChain_l(chain); 1256 } 1257 return status; 1258 } 1259 1260 effect->setDevice(mOutDevice); 1261 effect->setDevice(mInDevice); 1262 effect->setMode(mAudioFlinger->getMode()); 1263 effect->setAudioSource(mAudioSource); 1264 return NO_ERROR; 1265} 1266 1267void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1268 1269 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1270 effect_descriptor_t desc = effect->desc(); 1271 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1272 detachAuxEffect_l(effect->id()); 1273 } 1274 1275 sp<EffectChain> chain = effect->chain().promote(); 1276 if (chain != 0) { 1277 // remove effect chain if removing last effect 1278 if (chain->removeEffect_l(effect) == 0) { 1279 removeEffectChain_l(chain); 1280 } 1281 } else { 1282 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1283 } 1284} 1285 1286void AudioFlinger::ThreadBase::lockEffectChains_l( 1287 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1288{ 1289 effectChains = mEffectChains; 1290 for (size_t i = 0; i < mEffectChains.size(); i++) { 1291 mEffectChains[i]->lock(); 1292 } 1293} 1294 1295void AudioFlinger::ThreadBase::unlockEffectChains( 1296 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1297{ 1298 for (size_t i = 0; i < effectChains.size(); i++) { 1299 effectChains[i]->unlock(); 1300 } 1301} 1302 1303sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1304{ 1305 Mutex::Autolock _l(mLock); 1306 return getEffectChain_l(sessionId); 1307} 1308 1309sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1310{ 1311 size_t size = mEffectChains.size(); 1312 for (size_t i = 0; i < size; i++) { 1313 if (mEffectChains[i]->sessionId() == sessionId) { 1314 return mEffectChains[i]; 1315 } 1316 } 1317 return 0; 1318} 1319 1320void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1321{ 1322 Mutex::Autolock _l(mLock); 1323 size_t size = mEffectChains.size(); 1324 for (size_t i = 0; i < size; i++) { 1325 mEffectChains[i]->setMode_l(mode); 1326 } 1327} 1328 1329void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1330{ 1331 config->type = AUDIO_PORT_TYPE_MIX; 1332 config->ext.mix.handle = mId; 1333 config->sample_rate = mSampleRate; 1334 config->format = mFormat; 1335 config->channel_mask = mChannelMask; 1336 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1337 AUDIO_PORT_CONFIG_FORMAT; 1338} 1339 1340 1341// ---------------------------------------------------------------------------- 1342// Playback 1343// ---------------------------------------------------------------------------- 1344 1345AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1346 AudioStreamOut* output, 1347 audio_io_handle_t id, 1348 audio_devices_t device, 1349 type_t type) 1350 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1351 mNormalFrameCount(0), mSinkBuffer(NULL), 1352 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1353 mMixerBuffer(NULL), 1354 mMixerBufferSize(0), 1355 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1356 mMixerBufferValid(false), 1357 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1358 mEffectBuffer(NULL), 1359 mEffectBufferSize(0), 1360 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1361 mEffectBufferValid(false), 1362 mSuspended(0), mBytesWritten(0), 1363 mActiveTracksGeneration(0), 1364 // mStreamTypes[] initialized in constructor body 1365 mOutput(output), 1366 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1367 mMixerStatus(MIXER_IDLE), 1368 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1369 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1370 mBytesRemaining(0), 1371 mCurrentWriteLength(0), 1372 mUseAsyncWrite(false), 1373 mWriteAckSequence(0), 1374 mDrainSequence(0), 1375 mSignalPending(false), 1376 mScreenState(AudioFlinger::mScreenState), 1377 // index 0 is reserved for normal mixer's submix 1378 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1379 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1380 // mLatchD, mLatchQ, 1381 mLatchDValid(false), mLatchQValid(false) 1382{ 1383 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1384 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1385 1386 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1387 // it would be safer to explicitly pass initial masterVolume/masterMute as 1388 // parameter. 1389 // 1390 // If the HAL we are using has support for master volume or master mute, 1391 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1392 // and the mute set to false). 1393 mMasterVolume = audioFlinger->masterVolume_l(); 1394 mMasterMute = audioFlinger->masterMute_l(); 1395 if (mOutput && mOutput->audioHwDev) { 1396 if (mOutput->audioHwDev->canSetMasterVolume()) { 1397 mMasterVolume = 1.0; 1398 } 1399 1400 if (mOutput->audioHwDev->canSetMasterMute()) { 1401 mMasterMute = false; 1402 } 1403 } 1404 1405 readOutputParameters_l(); 1406 1407 // ++ operator does not compile 1408 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1409 stream = (audio_stream_type_t) (stream + 1)) { 1410 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1411 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1412 } 1413} 1414 1415AudioFlinger::PlaybackThread::~PlaybackThread() 1416{ 1417 mAudioFlinger->unregisterWriter(mNBLogWriter); 1418 free(mSinkBuffer); 1419 free(mMixerBuffer); 1420 free(mEffectBuffer); 1421} 1422 1423void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1424{ 1425 dumpInternals(fd, args); 1426 dumpTracks(fd, args); 1427 dumpEffectChains(fd, args); 1428} 1429 1430void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1431{ 1432 const size_t SIZE = 256; 1433 char buffer[SIZE]; 1434 String8 result; 1435 1436 result.appendFormat(" Stream volumes in dB: "); 1437 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1438 const stream_type_t *st = &mStreamTypes[i]; 1439 if (i > 0) { 1440 result.appendFormat(", "); 1441 } 1442 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1443 if (st->mute) { 1444 result.append("M"); 1445 } 1446 } 1447 result.append("\n"); 1448 write(fd, result.string(), result.length()); 1449 result.clear(); 1450 1451 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1452 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1453 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1454 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1455 1456 size_t numtracks = mTracks.size(); 1457 size_t numactive = mActiveTracks.size(); 1458 dprintf(fd, " %d Tracks", numtracks); 1459 size_t numactiveseen = 0; 1460 if (numtracks) { 1461 dprintf(fd, " of which %d are active\n", numactive); 1462 Track::appendDumpHeader(result); 1463 for (size_t i = 0; i < numtracks; ++i) { 1464 sp<Track> track = mTracks[i]; 1465 if (track != 0) { 1466 bool active = mActiveTracks.indexOf(track) >= 0; 1467 if (active) { 1468 numactiveseen++; 1469 } 1470 track->dump(buffer, SIZE, active); 1471 result.append(buffer); 1472 } 1473 } 1474 } else { 1475 result.append("\n"); 1476 } 1477 if (numactiveseen != numactive) { 1478 // some tracks in the active list were not in the tracks list 1479 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1480 " not in the track list\n"); 1481 result.append(buffer); 1482 Track::appendDumpHeader(result); 1483 for (size_t i = 0; i < numactive; ++i) { 1484 sp<Track> track = mActiveTracks[i].promote(); 1485 if (track != 0 && mTracks.indexOf(track) < 0) { 1486 track->dump(buffer, SIZE, true); 1487 result.append(buffer); 1488 } 1489 } 1490 } 1491 1492 write(fd, result.string(), result.size()); 1493} 1494 1495void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1496{ 1497 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1498 1499 dumpBase(fd, args); 1500 1501 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1502 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1503 dprintf(fd, " Total writes: %d\n", mNumWrites); 1504 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1505 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1506 dprintf(fd, " Suspend count: %d\n", mSuspended); 1507 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1508 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1509 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1510 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1511 AudioStreamOut *output = mOutput; 1512 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1513 String8 flagsAsString = outputFlagsToString(flags); 1514 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1515} 1516 1517// Thread virtuals 1518 1519void AudioFlinger::PlaybackThread::onFirstRef() 1520{ 1521 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1522} 1523 1524// ThreadBase virtuals 1525void AudioFlinger::PlaybackThread::preExit() 1526{ 1527 ALOGV(" preExit()"); 1528 // FIXME this is using hard-coded strings but in the future, this functionality will be 1529 // converted to use audio HAL extensions required to support tunneling 1530 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1531} 1532 1533// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1534sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1535 const sp<AudioFlinger::Client>& client, 1536 audio_stream_type_t streamType, 1537 uint32_t sampleRate, 1538 audio_format_t format, 1539 audio_channel_mask_t channelMask, 1540 size_t *pFrameCount, 1541 const sp<IMemory>& sharedBuffer, 1542 int sessionId, 1543 IAudioFlinger::track_flags_t *flags, 1544 pid_t tid, 1545 int uid, 1546 status_t *status) 1547{ 1548 size_t frameCount = *pFrameCount; 1549 sp<Track> track; 1550 status_t lStatus; 1551 1552 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1553 1554 // client expresses a preference for FAST, but we get the final say 1555 if (*flags & IAudioFlinger::TRACK_FAST) { 1556 if ( 1557 // not timed 1558 (!isTimed) && 1559 // either of these use cases: 1560 ( 1561 // use case 1: shared buffer with any frame count 1562 ( 1563 (sharedBuffer != 0) 1564 ) || 1565 // use case 2: frame count is default or at least as large as HAL 1566 ( 1567 // we formerly checked for a callback handler (non-0 tid), 1568 // but that is no longer required for TRANSFER_OBTAIN mode 1569 ((frameCount == 0) || 1570 (frameCount >= mFrameCount)) 1571 ) 1572 ) && 1573 // PCM data 1574 audio_is_linear_pcm(format) && 1575 // identical channel mask to sink, or mono in and stereo sink 1576 (channelMask == mChannelMask || 1577 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1578 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1579 // hardware sample rate 1580 (sampleRate == mSampleRate) && 1581 // normal mixer has an associated fast mixer 1582 hasFastMixer() && 1583 // there are sufficient fast track slots available 1584 (mFastTrackAvailMask != 0) 1585 // FIXME test that MixerThread for this fast track has a capable output HAL 1586 // FIXME add a permission test also? 1587 ) { 1588 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1589 if (frameCount == 0) { 1590 // read the fast track multiplier property the first time it is needed 1591 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1592 if (ok != 0) { 1593 ALOGE("%s pthread_once failed: %d", __func__, ok); 1594 } 1595 frameCount = mFrameCount * sFastTrackMultiplier; 1596 } 1597 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1598 frameCount, mFrameCount); 1599 } else { 1600 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1601 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1602 "sampleRate=%u mSampleRate=%u " 1603 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1604 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1605 audio_is_linear_pcm(format), 1606 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1607 *flags &= ~IAudioFlinger::TRACK_FAST; 1608 } 1609 } 1610 // For normal PCM streaming tracks, update minimum frame count. 1611 // For compatibility with AudioTrack calculation, buffer depth is forced 1612 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1613 // This is probably too conservative, but legacy application code may depend on it. 1614 // If you change this calculation, also review the start threshold which is related. 1615 if (!(*flags & IAudioFlinger::TRACK_FAST) 1616 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1617 // this must match AudioTrack.cpp calculateMinFrameCount(). 1618 // TODO: Move to a common library 1619 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1620 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1621 if (minBufCount < 2) { 1622 minBufCount = 2; 1623 } 1624 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1625 // or the client should compute and pass in a larger buffer request. 1626 size_t minFrameCount = 1627 minBufCount * sourceFramesNeededWithTimestretch( 1628 sampleRate, mNormalFrameCount, 1629 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1630 if (frameCount < minFrameCount) { // including frameCount == 0 1631 frameCount = minFrameCount; 1632 } 1633 } 1634 *pFrameCount = frameCount; 1635 1636 switch (mType) { 1637 1638 case DIRECT: 1639 if (audio_is_linear_pcm(format)) { 1640 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1641 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1642 "for output %p with format %#x", 1643 sampleRate, format, channelMask, mOutput, mFormat); 1644 lStatus = BAD_VALUE; 1645 goto Exit; 1646 } 1647 } 1648 break; 1649 1650 case OFFLOAD: 1651 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1652 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1653 "for output %p with format %#x", 1654 sampleRate, format, channelMask, mOutput, mFormat); 1655 lStatus = BAD_VALUE; 1656 goto Exit; 1657 } 1658 break; 1659 1660 default: 1661 if (!audio_is_linear_pcm(format)) { 1662 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1663 "for output %p with format %#x", 1664 format, mOutput, mFormat); 1665 lStatus = BAD_VALUE; 1666 goto Exit; 1667 } 1668 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1669 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1670 lStatus = BAD_VALUE; 1671 goto Exit; 1672 } 1673 break; 1674 1675 } 1676 1677 lStatus = initCheck(); 1678 if (lStatus != NO_ERROR) { 1679 ALOGE("createTrack_l() audio driver not initialized"); 1680 goto Exit; 1681 } 1682 1683 { // scope for mLock 1684 Mutex::Autolock _l(mLock); 1685 1686 // all tracks in same audio session must share the same routing strategy otherwise 1687 // conflicts will happen when tracks are moved from one output to another by audio policy 1688 // manager 1689 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1690 for (size_t i = 0; i < mTracks.size(); ++i) { 1691 sp<Track> t = mTracks[i]; 1692 if (t != 0 && t->isExternalTrack()) { 1693 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1694 if (sessionId == t->sessionId() && strategy != actual) { 1695 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1696 strategy, actual); 1697 lStatus = BAD_VALUE; 1698 goto Exit; 1699 } 1700 } 1701 } 1702 1703 if (!isTimed) { 1704 track = new Track(this, client, streamType, sampleRate, format, 1705 channelMask, frameCount, NULL, sharedBuffer, 1706 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1707 } else { 1708 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1709 channelMask, frameCount, sharedBuffer, sessionId, uid); 1710 } 1711 1712 // new Track always returns non-NULL, 1713 // but TimedTrack::create() is a factory that could fail by returning NULL 1714 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1715 if (lStatus != NO_ERROR) { 1716 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1717 // track must be cleared from the caller as the caller has the AF lock 1718 goto Exit; 1719 } 1720 mTracks.add(track); 1721 1722 sp<EffectChain> chain = getEffectChain_l(sessionId); 1723 if (chain != 0) { 1724 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1725 track->setMainBuffer(chain->inBuffer()); 1726 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1727 chain->incTrackCnt(); 1728 } 1729 1730 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1731 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1732 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1733 // so ask activity manager to do this on our behalf 1734 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1735 } 1736 } 1737 1738 lStatus = NO_ERROR; 1739 1740Exit: 1741 *status = lStatus; 1742 return track; 1743} 1744 1745uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1746{ 1747 return latency; 1748} 1749 1750uint32_t AudioFlinger::PlaybackThread::latency() const 1751{ 1752 Mutex::Autolock _l(mLock); 1753 return latency_l(); 1754} 1755uint32_t AudioFlinger::PlaybackThread::latency_l() const 1756{ 1757 if (initCheck() == NO_ERROR) { 1758 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1759 } else { 1760 return 0; 1761 } 1762} 1763 1764void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1765{ 1766 Mutex::Autolock _l(mLock); 1767 // Don't apply master volume in SW if our HAL can do it for us. 1768 if (mOutput && mOutput->audioHwDev && 1769 mOutput->audioHwDev->canSetMasterVolume()) { 1770 mMasterVolume = 1.0; 1771 } else { 1772 mMasterVolume = value; 1773 } 1774} 1775 1776void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1777{ 1778 Mutex::Autolock _l(mLock); 1779 // Don't apply master mute in SW if our HAL can do it for us. 1780 if (mOutput && mOutput->audioHwDev && 1781 mOutput->audioHwDev->canSetMasterMute()) { 1782 mMasterMute = false; 1783 } else { 1784 mMasterMute = muted; 1785 } 1786} 1787 1788void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1789{ 1790 Mutex::Autolock _l(mLock); 1791 mStreamTypes[stream].volume = value; 1792 broadcast_l(); 1793} 1794 1795void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1796{ 1797 Mutex::Autolock _l(mLock); 1798 mStreamTypes[stream].mute = muted; 1799 broadcast_l(); 1800} 1801 1802float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1803{ 1804 Mutex::Autolock _l(mLock); 1805 return mStreamTypes[stream].volume; 1806} 1807 1808// addTrack_l() must be called with ThreadBase::mLock held 1809status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1810{ 1811 status_t status = ALREADY_EXISTS; 1812 1813 // set retry count for buffer fill 1814 track->mRetryCount = kMaxTrackStartupRetries; 1815 if (mActiveTracks.indexOf(track) < 0) { 1816 // the track is newly added, make sure it fills up all its 1817 // buffers before playing. This is to ensure the client will 1818 // effectively get the latency it requested. 1819 if (track->isExternalTrack()) { 1820 TrackBase::track_state state = track->mState; 1821 mLock.unlock(); 1822 status = AudioSystem::startOutput(mId, track->streamType(), 1823 (audio_session_t)track->sessionId()); 1824 mLock.lock(); 1825 // abort track was stopped/paused while we released the lock 1826 if (state != track->mState) { 1827 if (status == NO_ERROR) { 1828 mLock.unlock(); 1829 AudioSystem::stopOutput(mId, track->streamType(), 1830 (audio_session_t)track->sessionId()); 1831 mLock.lock(); 1832 } 1833 return INVALID_OPERATION; 1834 } 1835 // abort if start is rejected by audio policy manager 1836 if (status != NO_ERROR) { 1837 return PERMISSION_DENIED; 1838 } 1839#ifdef ADD_BATTERY_DATA 1840 // to track the speaker usage 1841 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1842#endif 1843 } 1844 1845 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1846 track->mResetDone = false; 1847 track->mPresentationCompleteFrames = 0; 1848 mActiveTracks.add(track); 1849 mWakeLockUids.add(track->uid()); 1850 mActiveTracksGeneration++; 1851 mLatestActiveTrack = track; 1852 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1853 if (chain != 0) { 1854 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1855 track->sessionId()); 1856 chain->incActiveTrackCnt(); 1857 } 1858 1859 status = NO_ERROR; 1860 } 1861 1862 onAddNewTrack_l(); 1863 return status; 1864} 1865 1866bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1867{ 1868 track->terminate(); 1869 // active tracks are removed by threadLoop() 1870 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1871 track->mState = TrackBase::STOPPED; 1872 if (!trackActive) { 1873 removeTrack_l(track); 1874 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1875 track->mState = TrackBase::STOPPING_1; 1876 } 1877 1878 return trackActive; 1879} 1880 1881void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1882{ 1883 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1884 mTracks.remove(track); 1885 deleteTrackName_l(track->name()); 1886 // redundant as track is about to be destroyed, for dumpsys only 1887 track->mName = -1; 1888 if (track->isFastTrack()) { 1889 int index = track->mFastIndex; 1890 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1891 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1892 mFastTrackAvailMask |= 1 << index; 1893 // redundant as track is about to be destroyed, for dumpsys only 1894 track->mFastIndex = -1; 1895 } 1896 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1897 if (chain != 0) { 1898 chain->decTrackCnt(); 1899 } 1900} 1901 1902void AudioFlinger::PlaybackThread::broadcast_l() 1903{ 1904 // Thread could be blocked waiting for async 1905 // so signal it to handle state changes immediately 1906 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1907 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1908 mSignalPending = true; 1909 mWaitWorkCV.broadcast(); 1910} 1911 1912String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1913{ 1914 Mutex::Autolock _l(mLock); 1915 if (initCheck() != NO_ERROR) { 1916 return String8(); 1917 } 1918 1919 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1920 const String8 out_s8(s); 1921 free(s); 1922 return out_s8; 1923} 1924 1925void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event) { 1926 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 1927 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 1928 1929 desc->mIoHandle = mId; 1930 1931 switch (event) { 1932 case AUDIO_OUTPUT_OPENED: 1933 case AUDIO_OUTPUT_CONFIG_CHANGED: 1934 desc->mPatch = mPatch; 1935 desc->mChannelMask = mChannelMask; 1936 desc->mSamplingRate = mSampleRate; 1937 desc->mFormat = mFormat; 1938 desc->mFrameCount = mNormalFrameCount; // FIXME see 1939 // AudioFlinger::frameCount(audio_io_handle_t) 1940 desc->mLatency = latency_l(); 1941 break; 1942 1943 case AUDIO_OUTPUT_CLOSED: 1944 default: 1945 break; 1946 } 1947 mAudioFlinger->ioConfigChanged(event, desc); 1948} 1949 1950void AudioFlinger::PlaybackThread::writeCallback() 1951{ 1952 ALOG_ASSERT(mCallbackThread != 0); 1953 mCallbackThread->resetWriteBlocked(); 1954} 1955 1956void AudioFlinger::PlaybackThread::drainCallback() 1957{ 1958 ALOG_ASSERT(mCallbackThread != 0); 1959 mCallbackThread->resetDraining(); 1960} 1961 1962void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1963{ 1964 Mutex::Autolock _l(mLock); 1965 // reject out of sequence requests 1966 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1967 mWriteAckSequence &= ~1; 1968 mWaitWorkCV.signal(); 1969 } 1970} 1971 1972void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1973{ 1974 Mutex::Autolock _l(mLock); 1975 // reject out of sequence requests 1976 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1977 mDrainSequence &= ~1; 1978 mWaitWorkCV.signal(); 1979 } 1980} 1981 1982// static 1983int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1984 void *param __unused, 1985 void *cookie) 1986{ 1987 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1988 ALOGV("asyncCallback() event %d", event); 1989 switch (event) { 1990 case STREAM_CBK_EVENT_WRITE_READY: 1991 me->writeCallback(); 1992 break; 1993 case STREAM_CBK_EVENT_DRAIN_READY: 1994 me->drainCallback(); 1995 break; 1996 default: 1997 ALOGW("asyncCallback() unknown event %d", event); 1998 break; 1999 } 2000 return 0; 2001} 2002 2003void AudioFlinger::PlaybackThread::readOutputParameters_l() 2004{ 2005 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2006 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2007 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2008 if (!audio_is_output_channel(mChannelMask)) { 2009 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2010 } 2011 if ((mType == MIXER || mType == DUPLICATING) 2012 && !isValidPcmSinkChannelMask(mChannelMask)) { 2013 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2014 mChannelMask); 2015 } 2016 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2017 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2018 mFormat = mHALFormat; 2019 if (!audio_is_valid_format(mFormat)) { 2020 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2021 } 2022 if ((mType == MIXER || mType == DUPLICATING) 2023 && !isValidPcmSinkFormat(mFormat)) { 2024 LOG_FATAL("HAL format %#x not supported for mixed output", 2025 mFormat); 2026 } 2027 mFrameSize = mOutput->getFrameSize(); 2028 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2029 mFrameCount = mBufferSize / mFrameSize; 2030 if (mFrameCount & 15) { 2031 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2032 mFrameCount); 2033 } 2034 2035 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2036 (mOutput->stream->set_callback != NULL)) { 2037 if (mOutput->stream->set_callback(mOutput->stream, 2038 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2039 mUseAsyncWrite = true; 2040 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2041 } 2042 } 2043 2044 mHwSupportsPause = false; 2045 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2046 if (mOutput->stream->pause != NULL) { 2047 if (mOutput->stream->resume != NULL) { 2048 mHwSupportsPause = true; 2049 } else { 2050 ALOGW("direct output implements pause but not resume"); 2051 } 2052 } else if (mOutput->stream->resume != NULL) { 2053 ALOGW("direct output implements resume but not pause"); 2054 } 2055 } 2056 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2057 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2058 } 2059 2060 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2061 // For best precision, we use float instead of the associated output 2062 // device format (typically PCM 16 bit). 2063 2064 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2065 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2066 mBufferSize = mFrameSize * mFrameCount; 2067 2068 // TODO: We currently use the associated output device channel mask and sample rate. 2069 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2070 // (if a valid mask) to avoid premature downmix. 2071 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2072 // instead of the output device sample rate to avoid loss of high frequency information. 2073 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2074 } 2075 2076 // Calculate size of normal sink buffer relative to the HAL output buffer size 2077 double multiplier = 1.0; 2078 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2079 kUseFastMixer == FastMixer_Dynamic)) { 2080 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2081 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2082 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2083 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2084 maxNormalFrameCount = maxNormalFrameCount & ~15; 2085 if (maxNormalFrameCount < minNormalFrameCount) { 2086 maxNormalFrameCount = minNormalFrameCount; 2087 } 2088 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2089 if (multiplier <= 1.0) { 2090 multiplier = 1.0; 2091 } else if (multiplier <= 2.0) { 2092 if (2 * mFrameCount <= maxNormalFrameCount) { 2093 multiplier = 2.0; 2094 } else { 2095 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2096 } 2097 } else { 2098 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2099 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2100 // track, but we sometimes have to do this to satisfy the maximum frame count 2101 // constraint) 2102 // FIXME this rounding up should not be done if no HAL SRC 2103 uint32_t truncMult = (uint32_t) multiplier; 2104 if ((truncMult & 1)) { 2105 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2106 ++truncMult; 2107 } 2108 } 2109 multiplier = (double) truncMult; 2110 } 2111 } 2112 mNormalFrameCount = multiplier * mFrameCount; 2113 // round up to nearest 16 frames to satisfy AudioMixer 2114 if (mType == MIXER || mType == DUPLICATING) { 2115 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2116 } 2117 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2118 mNormalFrameCount); 2119 2120 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2121 // Originally this was int16_t[] array, need to remove legacy implications. 2122 free(mSinkBuffer); 2123 mSinkBuffer = NULL; 2124 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2125 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2126 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2127 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2128 2129 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2130 // drives the output. 2131 free(mMixerBuffer); 2132 mMixerBuffer = NULL; 2133 if (mMixerBufferEnabled) { 2134 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2135 mMixerBufferSize = mNormalFrameCount * mChannelCount 2136 * audio_bytes_per_sample(mMixerBufferFormat); 2137 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2138 } 2139 free(mEffectBuffer); 2140 mEffectBuffer = NULL; 2141 if (mEffectBufferEnabled) { 2142 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2143 mEffectBufferSize = mNormalFrameCount * mChannelCount 2144 * audio_bytes_per_sample(mEffectBufferFormat); 2145 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2146 } 2147 2148 // force reconfiguration of effect chains and engines to take new buffer size and audio 2149 // parameters into account 2150 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2151 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2152 // matter. 2153 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2154 Vector< sp<EffectChain> > effectChains = mEffectChains; 2155 for (size_t i = 0; i < effectChains.size(); i ++) { 2156 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2157 } 2158} 2159 2160 2161status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2162{ 2163 if (halFrames == NULL || dspFrames == NULL) { 2164 return BAD_VALUE; 2165 } 2166 Mutex::Autolock _l(mLock); 2167 if (initCheck() != NO_ERROR) { 2168 return INVALID_OPERATION; 2169 } 2170 size_t framesWritten = mBytesWritten / mFrameSize; 2171 *halFrames = framesWritten; 2172 2173 if (isSuspended()) { 2174 // return an estimation of rendered frames when the output is suspended 2175 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2176 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2177 return NO_ERROR; 2178 } else { 2179 status_t status; 2180 uint32_t frames; 2181 status = mOutput->getRenderPosition(&frames); 2182 *dspFrames = (size_t)frames; 2183 return status; 2184 } 2185} 2186 2187uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2188{ 2189 Mutex::Autolock _l(mLock); 2190 uint32_t result = 0; 2191 if (getEffectChain_l(sessionId) != 0) { 2192 result = EFFECT_SESSION; 2193 } 2194 2195 for (size_t i = 0; i < mTracks.size(); ++i) { 2196 sp<Track> track = mTracks[i]; 2197 if (sessionId == track->sessionId() && !track->isInvalid()) { 2198 result |= TRACK_SESSION; 2199 break; 2200 } 2201 } 2202 2203 return result; 2204} 2205 2206uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2207{ 2208 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2209 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2210 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2211 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2212 } 2213 for (size_t i = 0; i < mTracks.size(); i++) { 2214 sp<Track> track = mTracks[i]; 2215 if (sessionId == track->sessionId() && !track->isInvalid()) { 2216 return AudioSystem::getStrategyForStream(track->streamType()); 2217 } 2218 } 2219 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2220} 2221 2222 2223AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2224{ 2225 Mutex::Autolock _l(mLock); 2226 return mOutput; 2227} 2228 2229AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2230{ 2231 Mutex::Autolock _l(mLock); 2232 AudioStreamOut *output = mOutput; 2233 mOutput = NULL; 2234 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2235 // must push a NULL and wait for ack 2236 mOutputSink.clear(); 2237 mPipeSink.clear(); 2238 mNormalSink.clear(); 2239 return output; 2240} 2241 2242// this method must always be called either with ThreadBase mLock held or inside the thread loop 2243audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2244{ 2245 if (mOutput == NULL) { 2246 return NULL; 2247 } 2248 return &mOutput->stream->common; 2249} 2250 2251uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2252{ 2253 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2254} 2255 2256status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2257{ 2258 if (!isValidSyncEvent(event)) { 2259 return BAD_VALUE; 2260 } 2261 2262 Mutex::Autolock _l(mLock); 2263 2264 for (size_t i = 0; i < mTracks.size(); ++i) { 2265 sp<Track> track = mTracks[i]; 2266 if (event->triggerSession() == track->sessionId()) { 2267 (void) track->setSyncEvent(event); 2268 return NO_ERROR; 2269 } 2270 } 2271 2272 return NAME_NOT_FOUND; 2273} 2274 2275bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2276{ 2277 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2278} 2279 2280void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2281 const Vector< sp<Track> >& tracksToRemove) 2282{ 2283 size_t count = tracksToRemove.size(); 2284 if (count > 0) { 2285 for (size_t i = 0 ; i < count ; i++) { 2286 const sp<Track>& track = tracksToRemove.itemAt(i); 2287 if (track->isExternalTrack()) { 2288 AudioSystem::stopOutput(mId, track->streamType(), 2289 (audio_session_t)track->sessionId()); 2290#ifdef ADD_BATTERY_DATA 2291 // to track the speaker usage 2292 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2293#endif 2294 if (track->isTerminated()) { 2295 AudioSystem::releaseOutput(mId, track->streamType(), 2296 (audio_session_t)track->sessionId()); 2297 } 2298 } 2299 } 2300 } 2301} 2302 2303void AudioFlinger::PlaybackThread::checkSilentMode_l() 2304{ 2305 if (!mMasterMute) { 2306 char value[PROPERTY_VALUE_MAX]; 2307 if (property_get("ro.audio.silent", value, "0") > 0) { 2308 char *endptr; 2309 unsigned long ul = strtoul(value, &endptr, 0); 2310 if (*endptr == '\0' && ul != 0) { 2311 ALOGD("Silence is golden"); 2312 // The setprop command will not allow a property to be changed after 2313 // the first time it is set, so we don't have to worry about un-muting. 2314 setMasterMute_l(true); 2315 } 2316 } 2317 } 2318} 2319 2320// shared by MIXER and DIRECT, overridden by DUPLICATING 2321ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2322{ 2323 // FIXME rewrite to reduce number of system calls 2324 mLastWriteTime = systemTime(); 2325 mInWrite = true; 2326 ssize_t bytesWritten; 2327 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2328 2329 // If an NBAIO sink is present, use it to write the normal mixer's submix 2330 if (mNormalSink != 0) { 2331 2332 const size_t count = mBytesRemaining / mFrameSize; 2333 2334 ATRACE_BEGIN("write"); 2335 // update the setpoint when AudioFlinger::mScreenState changes 2336 uint32_t screenState = AudioFlinger::mScreenState; 2337 if (screenState != mScreenState) { 2338 mScreenState = screenState; 2339 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2340 if (pipe != NULL) { 2341 pipe->setAvgFrames((mScreenState & 1) ? 2342 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2343 } 2344 } 2345 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2346 ATRACE_END(); 2347 if (framesWritten > 0) { 2348 bytesWritten = framesWritten * mFrameSize; 2349 } else { 2350 bytesWritten = framesWritten; 2351 } 2352 mLatchDValid = false; 2353 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2354 if (status == NO_ERROR) { 2355 size_t totalFramesWritten = mNormalSink->framesWritten(); 2356 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2357 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2358 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2359 mLatchDValid = true; 2360 } 2361 } 2362 // otherwise use the HAL / AudioStreamOut directly 2363 } else { 2364 // Direct output and offload threads 2365 2366 if (mUseAsyncWrite) { 2367 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2368 mWriteAckSequence += 2; 2369 mWriteAckSequence |= 1; 2370 ALOG_ASSERT(mCallbackThread != 0); 2371 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2372 } 2373 // FIXME We should have an implementation of timestamps for direct output threads. 2374 // They are used e.g for multichannel PCM playback over HDMI. 2375 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2376 if (mUseAsyncWrite && 2377 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2378 // do not wait for async callback in case of error of full write 2379 mWriteAckSequence &= ~1; 2380 ALOG_ASSERT(mCallbackThread != 0); 2381 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2382 } 2383 } 2384 2385 mNumWrites++; 2386 mInWrite = false; 2387 mStandby = false; 2388 return bytesWritten; 2389} 2390 2391void AudioFlinger::PlaybackThread::threadLoop_drain() 2392{ 2393 if (mOutput->stream->drain) { 2394 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2395 if (mUseAsyncWrite) { 2396 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2397 mDrainSequence |= 1; 2398 ALOG_ASSERT(mCallbackThread != 0); 2399 mCallbackThread->setDraining(mDrainSequence); 2400 } 2401 mOutput->stream->drain(mOutput->stream, 2402 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2403 : AUDIO_DRAIN_ALL); 2404 } 2405} 2406 2407void AudioFlinger::PlaybackThread::threadLoop_exit() 2408{ 2409 { 2410 Mutex::Autolock _l(mLock); 2411 for (size_t i = 0; i < mTracks.size(); i++) { 2412 sp<Track> track = mTracks[i]; 2413 track->invalidate(); 2414 } 2415 } 2416} 2417 2418/* 2419The derived values that are cached: 2420 - mSinkBufferSize from frame count * frame size 2421 - activeSleepTime from activeSleepTimeUs() 2422 - idleSleepTime from idleSleepTimeUs() 2423 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2424 - maxPeriod from frame count and sample rate (MIXER only) 2425 2426The parameters that affect these derived values are: 2427 - frame count 2428 - frame size 2429 - sample rate 2430 - device type: A2DP or not 2431 - device latency 2432 - format: PCM or not 2433 - active sleep time 2434 - idle sleep time 2435*/ 2436 2437void AudioFlinger::PlaybackThread::cacheParameters_l() 2438{ 2439 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2440 activeSleepTime = activeSleepTimeUs(); 2441 idleSleepTime = idleSleepTimeUs(); 2442} 2443 2444void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2445{ 2446 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2447 this, streamType, mTracks.size()); 2448 Mutex::Autolock _l(mLock); 2449 2450 size_t size = mTracks.size(); 2451 for (size_t i = 0; i < size; i++) { 2452 sp<Track> t = mTracks[i]; 2453 if (t->streamType() == streamType) { 2454 t->invalidate(); 2455 } 2456 } 2457} 2458 2459status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2460{ 2461 int session = chain->sessionId(); 2462 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2463 ? mEffectBuffer : mSinkBuffer); 2464 bool ownsBuffer = false; 2465 2466 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2467 if (session > 0) { 2468 // Only one effect chain can be present in direct output thread and it uses 2469 // the sink buffer as input 2470 if (mType != DIRECT) { 2471 size_t numSamples = mNormalFrameCount * mChannelCount; 2472 buffer = new int16_t[numSamples]; 2473 memset(buffer, 0, numSamples * sizeof(int16_t)); 2474 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2475 ownsBuffer = true; 2476 } 2477 2478 // Attach all tracks with same session ID to this chain. 2479 for (size_t i = 0; i < mTracks.size(); ++i) { 2480 sp<Track> track = mTracks[i]; 2481 if (session == track->sessionId()) { 2482 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2483 buffer); 2484 track->setMainBuffer(buffer); 2485 chain->incTrackCnt(); 2486 } 2487 } 2488 2489 // indicate all active tracks in the chain 2490 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2491 sp<Track> track = mActiveTracks[i].promote(); 2492 if (track == 0) { 2493 continue; 2494 } 2495 if (session == track->sessionId()) { 2496 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2497 chain->incActiveTrackCnt(); 2498 } 2499 } 2500 } 2501 chain->setThread(this); 2502 chain->setInBuffer(buffer, ownsBuffer); 2503 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2504 ? mEffectBuffer : mSinkBuffer)); 2505 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2506 // chains list in order to be processed last as it contains output stage effects 2507 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2508 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2509 // after track specific effects and before output stage 2510 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2511 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2512 // Effect chain for other sessions are inserted at beginning of effect 2513 // chains list to be processed before output mix effects. Relative order between other 2514 // sessions is not important 2515 size_t size = mEffectChains.size(); 2516 size_t i = 0; 2517 for (i = 0; i < size; i++) { 2518 if (mEffectChains[i]->sessionId() < session) { 2519 break; 2520 } 2521 } 2522 mEffectChains.insertAt(chain, i); 2523 checkSuspendOnAddEffectChain_l(chain); 2524 2525 return NO_ERROR; 2526} 2527 2528size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2529{ 2530 int session = chain->sessionId(); 2531 2532 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2533 2534 for (size_t i = 0; i < mEffectChains.size(); i++) { 2535 if (chain == mEffectChains[i]) { 2536 mEffectChains.removeAt(i); 2537 // detach all active tracks from the chain 2538 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2539 sp<Track> track = mActiveTracks[i].promote(); 2540 if (track == 0) { 2541 continue; 2542 } 2543 if (session == track->sessionId()) { 2544 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2545 chain.get(), session); 2546 chain->decActiveTrackCnt(); 2547 } 2548 } 2549 2550 // detach all tracks with same session ID from this chain 2551 for (size_t i = 0; i < mTracks.size(); ++i) { 2552 sp<Track> track = mTracks[i]; 2553 if (session == track->sessionId()) { 2554 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2555 chain->decTrackCnt(); 2556 } 2557 } 2558 break; 2559 } 2560 } 2561 return mEffectChains.size(); 2562} 2563 2564status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2565 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2566{ 2567 Mutex::Autolock _l(mLock); 2568 return attachAuxEffect_l(track, EffectId); 2569} 2570 2571status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2572 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2573{ 2574 status_t status = NO_ERROR; 2575 2576 if (EffectId == 0) { 2577 track->setAuxBuffer(0, NULL); 2578 } else { 2579 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2580 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2581 if (effect != 0) { 2582 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2583 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2584 } else { 2585 status = INVALID_OPERATION; 2586 } 2587 } else { 2588 status = BAD_VALUE; 2589 } 2590 } 2591 return status; 2592} 2593 2594void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2595{ 2596 for (size_t i = 0; i < mTracks.size(); ++i) { 2597 sp<Track> track = mTracks[i]; 2598 if (track->auxEffectId() == effectId) { 2599 attachAuxEffect_l(track, 0); 2600 } 2601 } 2602} 2603 2604bool AudioFlinger::PlaybackThread::threadLoop() 2605{ 2606 Vector< sp<Track> > tracksToRemove; 2607 2608 standbyTime = systemTime(); 2609 2610 // MIXER 2611 nsecs_t lastWarning = 0; 2612 2613 // DUPLICATING 2614 // FIXME could this be made local to while loop? 2615 writeFrames = 0; 2616 2617 int lastGeneration = 0; 2618 2619 cacheParameters_l(); 2620 sleepTime = idleSleepTime; 2621 2622 if (mType == MIXER) { 2623 sleepTimeShift = 0; 2624 } 2625 2626 CpuStats cpuStats; 2627 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2628 2629 acquireWakeLock(); 2630 2631 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2632 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2633 // and then that string will be logged at the next convenient opportunity. 2634 const char *logString = NULL; 2635 2636 checkSilentMode_l(); 2637 2638 while (!exitPending()) 2639 { 2640 cpuStats.sample(myName); 2641 2642 Vector< sp<EffectChain> > effectChains; 2643 2644 { // scope for mLock 2645 2646 Mutex::Autolock _l(mLock); 2647 2648 processConfigEvents_l(); 2649 2650 if (logString != NULL) { 2651 mNBLogWriter->logTimestamp(); 2652 mNBLogWriter->log(logString); 2653 logString = NULL; 2654 } 2655 2656 // Gather the framesReleased counters for all active tracks, 2657 // and latch them atomically with the timestamp. 2658 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2659 mLatchD.mFramesReleased.clear(); 2660 size_t size = mActiveTracks.size(); 2661 for (size_t i = 0; i < size; i++) { 2662 sp<Track> t = mActiveTracks[i].promote(); 2663 if (t != 0) { 2664 mLatchD.mFramesReleased.add(t.get(), 2665 t->mAudioTrackServerProxy->framesReleased()); 2666 } 2667 } 2668 if (mLatchDValid) { 2669 mLatchQ = mLatchD; 2670 mLatchDValid = false; 2671 mLatchQValid = true; 2672 } 2673 2674 saveOutputTracks(); 2675 if (mSignalPending) { 2676 // A signal was raised while we were unlocked 2677 mSignalPending = false; 2678 } else if (waitingAsyncCallback_l()) { 2679 if (exitPending()) { 2680 break; 2681 } 2682 bool released = false; 2683 // The following works around a bug in the offload driver. Ideally we would release 2684 // the wake lock every time, but that causes the last offload buffer(s) to be 2685 // dropped while the device is on battery, so we need to hold a wake lock during 2686 // the drain phase. 2687 if (mBytesRemaining && !(mDrainSequence & 1)) { 2688 releaseWakeLock_l(); 2689 released = true; 2690 } 2691 mWakeLockUids.clear(); 2692 mActiveTracksGeneration++; 2693 ALOGV("wait async completion"); 2694 mWaitWorkCV.wait(mLock); 2695 ALOGV("async completion/wake"); 2696 if (released) { 2697 acquireWakeLock_l(); 2698 } 2699 standbyTime = systemTime() + standbyDelay; 2700 sleepTime = 0; 2701 2702 continue; 2703 } 2704 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2705 isSuspended()) { 2706 // put audio hardware into standby after short delay 2707 if (shouldStandby_l()) { 2708 2709 threadLoop_standby(); 2710 2711 mStandby = true; 2712 } 2713 2714 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2715 // we're about to wait, flush the binder command buffer 2716 IPCThreadState::self()->flushCommands(); 2717 2718 clearOutputTracks(); 2719 2720 if (exitPending()) { 2721 break; 2722 } 2723 2724 releaseWakeLock_l(); 2725 mWakeLockUids.clear(); 2726 mActiveTracksGeneration++; 2727 // wait until we have something to do... 2728 ALOGV("%s going to sleep", myName.string()); 2729 mWaitWorkCV.wait(mLock); 2730 ALOGV("%s waking up", myName.string()); 2731 acquireWakeLock_l(); 2732 2733 mMixerStatus = MIXER_IDLE; 2734 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2735 mBytesWritten = 0; 2736 mBytesRemaining = 0; 2737 checkSilentMode_l(); 2738 2739 standbyTime = systemTime() + standbyDelay; 2740 sleepTime = idleSleepTime; 2741 if (mType == MIXER) { 2742 sleepTimeShift = 0; 2743 } 2744 2745 continue; 2746 } 2747 } 2748 // mMixerStatusIgnoringFastTracks is also updated internally 2749 mMixerStatus = prepareTracks_l(&tracksToRemove); 2750 2751 // compare with previously applied list 2752 if (lastGeneration != mActiveTracksGeneration) { 2753 // update wakelock 2754 updateWakeLockUids_l(mWakeLockUids); 2755 lastGeneration = mActiveTracksGeneration; 2756 } 2757 2758 // prevent any changes in effect chain list and in each effect chain 2759 // during mixing and effect process as the audio buffers could be deleted 2760 // or modified if an effect is created or deleted 2761 lockEffectChains_l(effectChains); 2762 } // mLock scope ends 2763 2764 if (mBytesRemaining == 0) { 2765 mCurrentWriteLength = 0; 2766 if (mMixerStatus == MIXER_TRACKS_READY) { 2767 // threadLoop_mix() sets mCurrentWriteLength 2768 threadLoop_mix(); 2769 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2770 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2771 // threadLoop_sleepTime sets sleepTime to 0 if data 2772 // must be written to HAL 2773 threadLoop_sleepTime(); 2774 if (sleepTime == 0) { 2775 mCurrentWriteLength = mSinkBufferSize; 2776 } 2777 } 2778 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2779 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2780 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2781 // or mSinkBuffer (if there are no effects). 2782 // 2783 // This is done pre-effects computation; if effects change to 2784 // support higher precision, this needs to move. 2785 // 2786 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2787 // TODO use sleepTime == 0 as an additional condition. 2788 if (mMixerBufferValid) { 2789 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2790 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2791 2792 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2793 mNormalFrameCount * mChannelCount); 2794 } 2795 2796 mBytesRemaining = mCurrentWriteLength; 2797 if (isSuspended()) { 2798 sleepTime = suspendSleepTimeUs(); 2799 // simulate write to HAL when suspended 2800 mBytesWritten += mSinkBufferSize; 2801 mBytesRemaining = 0; 2802 } 2803 2804 // only process effects if we're going to write 2805 if (sleepTime == 0 && mType != OFFLOAD) { 2806 for (size_t i = 0; i < effectChains.size(); i ++) { 2807 effectChains[i]->process_l(); 2808 } 2809 } 2810 } 2811 // Process effect chains for offloaded thread even if no audio 2812 // was read from audio track: process only updates effect state 2813 // and thus does have to be synchronized with audio writes but may have 2814 // to be called while waiting for async write callback 2815 if (mType == OFFLOAD) { 2816 for (size_t i = 0; i < effectChains.size(); i ++) { 2817 effectChains[i]->process_l(); 2818 } 2819 } 2820 2821 // Only if the Effects buffer is enabled and there is data in the 2822 // Effects buffer (buffer valid), we need to 2823 // copy into the sink buffer. 2824 // TODO use sleepTime == 0 as an additional condition. 2825 if (mEffectBufferValid) { 2826 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2827 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2828 mNormalFrameCount * mChannelCount); 2829 } 2830 2831 // enable changes in effect chain 2832 unlockEffectChains(effectChains); 2833 2834 if (!waitingAsyncCallback()) { 2835 // sleepTime == 0 means we must write to audio hardware 2836 if (sleepTime == 0) { 2837 if (mBytesRemaining) { 2838 ssize_t ret = threadLoop_write(); 2839 if (ret < 0) { 2840 mBytesRemaining = 0; 2841 } else { 2842 mBytesWritten += ret; 2843 mBytesRemaining -= ret; 2844 } 2845 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2846 (mMixerStatus == MIXER_DRAIN_ALL)) { 2847 threadLoop_drain(); 2848 } 2849 if (mType == MIXER) { 2850 // write blocked detection 2851 nsecs_t now = systemTime(); 2852 nsecs_t delta = now - mLastWriteTime; 2853 if (!mStandby && delta > maxPeriod) { 2854 mNumDelayedWrites++; 2855 if ((now - lastWarning) > kWarningThrottleNs) { 2856 ATRACE_NAME("underrun"); 2857 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2858 ns2ms(delta), mNumDelayedWrites, this); 2859 lastWarning = now; 2860 } 2861 } 2862 } 2863 2864 } else { 2865 ATRACE_BEGIN("sleep"); 2866 usleep(sleepTime); 2867 ATRACE_END(); 2868 } 2869 } 2870 2871 // Finally let go of removed track(s), without the lock held 2872 // since we can't guarantee the destructors won't acquire that 2873 // same lock. This will also mutate and push a new fast mixer state. 2874 threadLoop_removeTracks(tracksToRemove); 2875 tracksToRemove.clear(); 2876 2877 // FIXME I don't understand the need for this here; 2878 // it was in the original code but maybe the 2879 // assignment in saveOutputTracks() makes this unnecessary? 2880 clearOutputTracks(); 2881 2882 // Effect chains will be actually deleted here if they were removed from 2883 // mEffectChains list during mixing or effects processing 2884 effectChains.clear(); 2885 2886 // FIXME Note that the above .clear() is no longer necessary since effectChains 2887 // is now local to this block, but will keep it for now (at least until merge done). 2888 } 2889 2890 threadLoop_exit(); 2891 2892 if (!mStandby) { 2893 threadLoop_standby(); 2894 mStandby = true; 2895 } 2896 2897 releaseWakeLock(); 2898 mWakeLockUids.clear(); 2899 mActiveTracksGeneration++; 2900 2901 ALOGV("Thread %p type %d exiting", this, mType); 2902 return false; 2903} 2904 2905// removeTracks_l() must be called with ThreadBase::mLock held 2906void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2907{ 2908 size_t count = tracksToRemove.size(); 2909 if (count > 0) { 2910 for (size_t i=0 ; i<count ; i++) { 2911 const sp<Track>& track = tracksToRemove.itemAt(i); 2912 mActiveTracks.remove(track); 2913 mWakeLockUids.remove(track->uid()); 2914 mActiveTracksGeneration++; 2915 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2916 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2917 if (chain != 0) { 2918 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2919 track->sessionId()); 2920 chain->decActiveTrackCnt(); 2921 } 2922 if (track->isTerminated()) { 2923 removeTrack_l(track); 2924 } 2925 } 2926 } 2927 2928} 2929 2930status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2931{ 2932 if (mNormalSink != 0) { 2933 return mNormalSink->getTimestamp(timestamp); 2934 } 2935 if ((mType == OFFLOAD || mType == DIRECT) 2936 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2937 uint64_t position64; 2938 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 2939 if (ret == 0) { 2940 timestamp.mPosition = (uint32_t)position64; 2941 return NO_ERROR; 2942 } 2943 } 2944 return INVALID_OPERATION; 2945} 2946 2947status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 2948 audio_patch_handle_t *handle) 2949{ 2950 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2951 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2952 if (mFastMixer != 0) { 2953 FastMixerStateQueue *sq = mFastMixer->sq(); 2954 FastMixerState *state = sq->begin(); 2955 if (!(state->mCommand & FastMixerState::IDLE)) { 2956 previousCommand = state->mCommand; 2957 state->mCommand = FastMixerState::HOT_IDLE; 2958 sq->end(); 2959 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2960 } else { 2961 sq->end(false /*didModify*/); 2962 } 2963 } 2964 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 2965 2966 if (!(previousCommand & FastMixerState::IDLE)) { 2967 ALOG_ASSERT(mFastMixer != 0); 2968 FastMixerStateQueue *sq = mFastMixer->sq(); 2969 FastMixerState *state = sq->begin(); 2970 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 2971 state->mCommand = previousCommand; 2972 sq->end(); 2973 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2974 } 2975 2976 return status; 2977} 2978 2979status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2980 audio_patch_handle_t *handle) 2981{ 2982 status_t status = NO_ERROR; 2983 2984 // store new device and send to effects 2985 audio_devices_t type = AUDIO_DEVICE_NONE; 2986 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2987 type |= patch->sinks[i].ext.device.type; 2988 } 2989 2990#ifdef ADD_BATTERY_DATA 2991 // when changing the audio output device, call addBatteryData to notify 2992 // the change 2993 if (mOutDevice != type) { 2994 uint32_t params = 0; 2995 // check whether speaker is on 2996 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 2997 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2998 } 2999 3000 audio_devices_t deviceWithoutSpeaker 3001 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3002 // check if any other device (except speaker) is on 3003 if (type & deviceWithoutSpeaker) { 3004 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3005 } 3006 3007 if (params != 0) { 3008 addBatteryData(params); 3009 } 3010 } 3011#endif 3012 3013 for (size_t i = 0; i < mEffectChains.size(); i++) { 3014 mEffectChains[i]->setDevice_l(type); 3015 } 3016 mOutDevice = type; 3017 mPatch = *patch; 3018 3019 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3020 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3021 status = hwDevice->create_audio_patch(hwDevice, 3022 patch->num_sources, 3023 patch->sources, 3024 patch->num_sinks, 3025 patch->sinks, 3026 handle); 3027 } else { 3028 char *address; 3029 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3030 //FIXME: we only support address on first sink with HAL version < 3.0 3031 address = audio_device_address_to_parameter( 3032 patch->sinks[0].ext.device.type, 3033 patch->sinks[0].ext.device.address); 3034 } else { 3035 address = (char *)calloc(1, 1); 3036 } 3037 AudioParameter param = AudioParameter(String8(address)); 3038 free(address); 3039 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3040 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3041 param.toString().string()); 3042 *handle = AUDIO_PATCH_HANDLE_NONE; 3043 } 3044 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3045 return status; 3046} 3047 3048status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3049{ 3050 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3051 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3052 if (mFastMixer != 0) { 3053 FastMixerStateQueue *sq = mFastMixer->sq(); 3054 FastMixerState *state = sq->begin(); 3055 if (!(state->mCommand & FastMixerState::IDLE)) { 3056 previousCommand = state->mCommand; 3057 state->mCommand = FastMixerState::HOT_IDLE; 3058 sq->end(); 3059 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3060 } else { 3061 sq->end(false /*didModify*/); 3062 } 3063 } 3064 3065 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3066 3067 if (!(previousCommand & FastMixerState::IDLE)) { 3068 ALOG_ASSERT(mFastMixer != 0); 3069 FastMixerStateQueue *sq = mFastMixer->sq(); 3070 FastMixerState *state = sq->begin(); 3071 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3072 state->mCommand = previousCommand; 3073 sq->end(); 3074 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3075 } 3076 3077 return status; 3078} 3079 3080status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3081{ 3082 status_t status = NO_ERROR; 3083 3084 mOutDevice = AUDIO_DEVICE_NONE; 3085 3086 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3087 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3088 status = hwDevice->release_audio_patch(hwDevice, handle); 3089 } else { 3090 AudioParameter param; 3091 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3092 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3093 param.toString().string()); 3094 } 3095 return status; 3096} 3097 3098void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3099{ 3100 Mutex::Autolock _l(mLock); 3101 mTracks.add(track); 3102} 3103 3104void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3105{ 3106 Mutex::Autolock _l(mLock); 3107 destroyTrack_l(track); 3108} 3109 3110void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3111{ 3112 ThreadBase::getAudioPortConfig(config); 3113 config->role = AUDIO_PORT_ROLE_SOURCE; 3114 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3115 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3116} 3117 3118// ---------------------------------------------------------------------------- 3119 3120AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3121 audio_io_handle_t id, audio_devices_t device, type_t type) 3122 : PlaybackThread(audioFlinger, output, id, device, type), 3123 // mAudioMixer below 3124 // mFastMixer below 3125 mFastMixerFutex(0) 3126 // mOutputSink below 3127 // mPipeSink below 3128 // mNormalSink below 3129{ 3130 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3131 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3132 "mFrameCount=%d, mNormalFrameCount=%d", 3133 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3134 mNormalFrameCount); 3135 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3136 3137 if (type == DUPLICATING) { 3138 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3139 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3140 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3141 return; 3142 } 3143 // create an NBAIO sink for the HAL output stream, and negotiate 3144 mOutputSink = new AudioStreamOutSink(output->stream); 3145 size_t numCounterOffers = 0; 3146 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3147 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3148 ALOG_ASSERT(index == 0); 3149 3150 // initialize fast mixer depending on configuration 3151 bool initFastMixer; 3152 switch (kUseFastMixer) { 3153 case FastMixer_Never: 3154 initFastMixer = false; 3155 break; 3156 case FastMixer_Always: 3157 initFastMixer = true; 3158 break; 3159 case FastMixer_Static: 3160 case FastMixer_Dynamic: 3161 initFastMixer = mFrameCount < mNormalFrameCount; 3162 break; 3163 } 3164 if (initFastMixer) { 3165 audio_format_t fastMixerFormat; 3166 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3167 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3168 } else { 3169 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3170 } 3171 if (mFormat != fastMixerFormat) { 3172 // change our Sink format to accept our intermediate precision 3173 mFormat = fastMixerFormat; 3174 free(mSinkBuffer); 3175 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3176 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3177 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3178 } 3179 3180 // create a MonoPipe to connect our submix to FastMixer 3181 NBAIO_Format format = mOutputSink->format(); 3182 NBAIO_Format origformat = format; 3183 // adjust format to match that of the Fast Mixer 3184 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3185 format.mFormat = fastMixerFormat; 3186 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3187 3188 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3189 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3190 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3191 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3192 const NBAIO_Format offers[1] = {format}; 3193 size_t numCounterOffers = 0; 3194 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3195 ALOG_ASSERT(index == 0); 3196 monoPipe->setAvgFrames((mScreenState & 1) ? 3197 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3198 mPipeSink = monoPipe; 3199 3200#ifdef TEE_SINK 3201 if (mTeeSinkOutputEnabled) { 3202 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3203 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3204 const NBAIO_Format offers2[1] = {origformat}; 3205 numCounterOffers = 0; 3206 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3207 ALOG_ASSERT(index == 0); 3208 mTeeSink = teeSink; 3209 PipeReader *teeSource = new PipeReader(*teeSink); 3210 numCounterOffers = 0; 3211 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3212 ALOG_ASSERT(index == 0); 3213 mTeeSource = teeSource; 3214 } 3215#endif 3216 3217 // create fast mixer and configure it initially with just one fast track for our submix 3218 mFastMixer = new FastMixer(); 3219 FastMixerStateQueue *sq = mFastMixer->sq(); 3220#ifdef STATE_QUEUE_DUMP 3221 sq->setObserverDump(&mStateQueueObserverDump); 3222 sq->setMutatorDump(&mStateQueueMutatorDump); 3223#endif 3224 FastMixerState *state = sq->begin(); 3225 FastTrack *fastTrack = &state->mFastTracks[0]; 3226 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3227 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3228 fastTrack->mVolumeProvider = NULL; 3229 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3230 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3231 fastTrack->mGeneration++; 3232 state->mFastTracksGen++; 3233 state->mTrackMask = 1; 3234 // fast mixer will use the HAL output sink 3235 state->mOutputSink = mOutputSink.get(); 3236 state->mOutputSinkGen++; 3237 state->mFrameCount = mFrameCount; 3238 state->mCommand = FastMixerState::COLD_IDLE; 3239 // already done in constructor initialization list 3240 //mFastMixerFutex = 0; 3241 state->mColdFutexAddr = &mFastMixerFutex; 3242 state->mColdGen++; 3243 state->mDumpState = &mFastMixerDumpState; 3244#ifdef TEE_SINK 3245 state->mTeeSink = mTeeSink.get(); 3246#endif 3247 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3248 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3249 sq->end(); 3250 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3251 3252 // start the fast mixer 3253 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3254 pid_t tid = mFastMixer->getTid(); 3255 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3256 if (err != 0) { 3257 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3258 kPriorityFastMixer, getpid_cached, tid, err); 3259 } 3260 3261#ifdef AUDIO_WATCHDOG 3262 // create and start the watchdog 3263 mAudioWatchdog = new AudioWatchdog(); 3264 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3265 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3266 tid = mAudioWatchdog->getTid(); 3267 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3268 if (err != 0) { 3269 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3270 kPriorityFastMixer, getpid_cached, tid, err); 3271 } 3272#endif 3273 3274 } 3275 3276 switch (kUseFastMixer) { 3277 case FastMixer_Never: 3278 case FastMixer_Dynamic: 3279 mNormalSink = mOutputSink; 3280 break; 3281 case FastMixer_Always: 3282 mNormalSink = mPipeSink; 3283 break; 3284 case FastMixer_Static: 3285 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3286 break; 3287 } 3288} 3289 3290AudioFlinger::MixerThread::~MixerThread() 3291{ 3292 if (mFastMixer != 0) { 3293 FastMixerStateQueue *sq = mFastMixer->sq(); 3294 FastMixerState *state = sq->begin(); 3295 if (state->mCommand == FastMixerState::COLD_IDLE) { 3296 int32_t old = android_atomic_inc(&mFastMixerFutex); 3297 if (old == -1) { 3298 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3299 } 3300 } 3301 state->mCommand = FastMixerState::EXIT; 3302 sq->end(); 3303 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3304 mFastMixer->join(); 3305 // Though the fast mixer thread has exited, it's state queue is still valid. 3306 // We'll use that extract the final state which contains one remaining fast track 3307 // corresponding to our sub-mix. 3308 state = sq->begin(); 3309 ALOG_ASSERT(state->mTrackMask == 1); 3310 FastTrack *fastTrack = &state->mFastTracks[0]; 3311 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3312 delete fastTrack->mBufferProvider; 3313 sq->end(false /*didModify*/); 3314 mFastMixer.clear(); 3315#ifdef AUDIO_WATCHDOG 3316 if (mAudioWatchdog != 0) { 3317 mAudioWatchdog->requestExit(); 3318 mAudioWatchdog->requestExitAndWait(); 3319 mAudioWatchdog.clear(); 3320 } 3321#endif 3322 } 3323 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3324 delete mAudioMixer; 3325} 3326 3327 3328uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3329{ 3330 if (mFastMixer != 0) { 3331 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3332 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3333 } 3334 return latency; 3335} 3336 3337 3338void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3339{ 3340 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3341} 3342 3343ssize_t AudioFlinger::MixerThread::threadLoop_write() 3344{ 3345 // FIXME we should only do one push per cycle; confirm this is true 3346 // Start the fast mixer if it's not already running 3347 if (mFastMixer != 0) { 3348 FastMixerStateQueue *sq = mFastMixer->sq(); 3349 FastMixerState *state = sq->begin(); 3350 if (state->mCommand != FastMixerState::MIX_WRITE && 3351 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3352 if (state->mCommand == FastMixerState::COLD_IDLE) { 3353 int32_t old = android_atomic_inc(&mFastMixerFutex); 3354 if (old == -1) { 3355 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3356 } 3357#ifdef AUDIO_WATCHDOG 3358 if (mAudioWatchdog != 0) { 3359 mAudioWatchdog->resume(); 3360 } 3361#endif 3362 } 3363 state->mCommand = FastMixerState::MIX_WRITE; 3364#ifdef FAST_THREAD_STATISTICS 3365 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3366 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3367#endif 3368 sq->end(); 3369 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3370 if (kUseFastMixer == FastMixer_Dynamic) { 3371 mNormalSink = mPipeSink; 3372 } 3373 } else { 3374 sq->end(false /*didModify*/); 3375 } 3376 } 3377 return PlaybackThread::threadLoop_write(); 3378} 3379 3380void AudioFlinger::MixerThread::threadLoop_standby() 3381{ 3382 // Idle the fast mixer if it's currently running 3383 if (mFastMixer != 0) { 3384 FastMixerStateQueue *sq = mFastMixer->sq(); 3385 FastMixerState *state = sq->begin(); 3386 if (!(state->mCommand & FastMixerState::IDLE)) { 3387 state->mCommand = FastMixerState::COLD_IDLE; 3388 state->mColdFutexAddr = &mFastMixerFutex; 3389 state->mColdGen++; 3390 mFastMixerFutex = 0; 3391 sq->end(); 3392 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3393 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3394 if (kUseFastMixer == FastMixer_Dynamic) { 3395 mNormalSink = mOutputSink; 3396 } 3397#ifdef AUDIO_WATCHDOG 3398 if (mAudioWatchdog != 0) { 3399 mAudioWatchdog->pause(); 3400 } 3401#endif 3402 } else { 3403 sq->end(false /*didModify*/); 3404 } 3405 } 3406 PlaybackThread::threadLoop_standby(); 3407} 3408 3409bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3410{ 3411 return false; 3412} 3413 3414bool AudioFlinger::PlaybackThread::shouldStandby_l() 3415{ 3416 return !mStandby; 3417} 3418 3419bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3420{ 3421 Mutex::Autolock _l(mLock); 3422 return waitingAsyncCallback_l(); 3423} 3424 3425// shared by MIXER and DIRECT, overridden by DUPLICATING 3426void AudioFlinger::PlaybackThread::threadLoop_standby() 3427{ 3428 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3429 mOutput->standby(); 3430 if (mUseAsyncWrite != 0) { 3431 // discard any pending drain or write ack by incrementing sequence 3432 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3433 mDrainSequence = (mDrainSequence + 2) & ~1; 3434 ALOG_ASSERT(mCallbackThread != 0); 3435 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3436 mCallbackThread->setDraining(mDrainSequence); 3437 } 3438 mHwPaused = false; 3439} 3440 3441void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3442{ 3443 ALOGV("signal playback thread"); 3444 broadcast_l(); 3445} 3446 3447void AudioFlinger::MixerThread::threadLoop_mix() 3448{ 3449 // obtain the presentation timestamp of the next output buffer 3450 int64_t pts; 3451 status_t status = INVALID_OPERATION; 3452 3453 if (mNormalSink != 0) { 3454 status = mNormalSink->getNextWriteTimestamp(&pts); 3455 } else { 3456 status = mOutputSink->getNextWriteTimestamp(&pts); 3457 } 3458 3459 if (status != NO_ERROR) { 3460 pts = AudioBufferProvider::kInvalidPTS; 3461 } 3462 3463 // mix buffers... 3464 mAudioMixer->process(pts); 3465 mCurrentWriteLength = mSinkBufferSize; 3466 // increase sleep time progressively when application underrun condition clears. 3467 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3468 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3469 // such that we would underrun the audio HAL. 3470 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3471 sleepTimeShift--; 3472 } 3473 sleepTime = 0; 3474 standbyTime = systemTime() + standbyDelay; 3475 //TODO: delay standby when effects have a tail 3476 3477} 3478 3479void AudioFlinger::MixerThread::threadLoop_sleepTime() 3480{ 3481 // If no tracks are ready, sleep once for the duration of an output 3482 // buffer size, then write 0s to the output 3483 if (sleepTime == 0) { 3484 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3485 sleepTime = activeSleepTime >> sleepTimeShift; 3486 if (sleepTime < kMinThreadSleepTimeUs) { 3487 sleepTime = kMinThreadSleepTimeUs; 3488 } 3489 // reduce sleep time in case of consecutive application underruns to avoid 3490 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3491 // duration we would end up writing less data than needed by the audio HAL if 3492 // the condition persists. 3493 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3494 sleepTimeShift++; 3495 } 3496 } else { 3497 sleepTime = idleSleepTime; 3498 } 3499 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3500 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3501 // before effects processing or output. 3502 if (mMixerBufferValid) { 3503 memset(mMixerBuffer, 0, mMixerBufferSize); 3504 } else { 3505 memset(mSinkBuffer, 0, mSinkBufferSize); 3506 } 3507 sleepTime = 0; 3508 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3509 "anticipated start"); 3510 } 3511 // TODO add standby time extension fct of effect tail 3512} 3513 3514// prepareTracks_l() must be called with ThreadBase::mLock held 3515AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3516 Vector< sp<Track> > *tracksToRemove) 3517{ 3518 3519 mixer_state mixerStatus = MIXER_IDLE; 3520 // find out which tracks need to be processed 3521 size_t count = mActiveTracks.size(); 3522 size_t mixedTracks = 0; 3523 size_t tracksWithEffect = 0; 3524 // counts only _active_ fast tracks 3525 size_t fastTracks = 0; 3526 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3527 3528 float masterVolume = mMasterVolume; 3529 bool masterMute = mMasterMute; 3530 3531 if (masterMute) { 3532 masterVolume = 0; 3533 } 3534 // Delegate master volume control to effect in output mix effect chain if needed 3535 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3536 if (chain != 0) { 3537 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3538 chain->setVolume_l(&v, &v); 3539 masterVolume = (float)((v + (1 << 23)) >> 24); 3540 chain.clear(); 3541 } 3542 3543 // prepare a new state to push 3544 FastMixerStateQueue *sq = NULL; 3545 FastMixerState *state = NULL; 3546 bool didModify = false; 3547 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3548 if (mFastMixer != 0) { 3549 sq = mFastMixer->sq(); 3550 state = sq->begin(); 3551 } 3552 3553 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3554 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3555 3556 for (size_t i=0 ; i<count ; i++) { 3557 const sp<Track> t = mActiveTracks[i].promote(); 3558 if (t == 0) { 3559 continue; 3560 } 3561 3562 // this const just means the local variable doesn't change 3563 Track* const track = t.get(); 3564 3565 // process fast tracks 3566 if (track->isFastTrack()) { 3567 3568 // It's theoretically possible (though unlikely) for a fast track to be created 3569 // and then removed within the same normal mix cycle. This is not a problem, as 3570 // the track never becomes active so it's fast mixer slot is never touched. 3571 // The converse, of removing an (active) track and then creating a new track 3572 // at the identical fast mixer slot within the same normal mix cycle, 3573 // is impossible because the slot isn't marked available until the end of each cycle. 3574 int j = track->mFastIndex; 3575 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3576 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3577 FastTrack *fastTrack = &state->mFastTracks[j]; 3578 3579 // Determine whether the track is currently in underrun condition, 3580 // and whether it had a recent underrun. 3581 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3582 FastTrackUnderruns underruns = ftDump->mUnderruns; 3583 uint32_t recentFull = (underruns.mBitFields.mFull - 3584 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3585 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3586 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3587 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3588 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3589 uint32_t recentUnderruns = recentPartial + recentEmpty; 3590 track->mObservedUnderruns = underruns; 3591 // don't count underruns that occur while stopping or pausing 3592 // or stopped which can occur when flush() is called while active 3593 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3594 recentUnderruns > 0) { 3595 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3596 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3597 } 3598 3599 // This is similar to the state machine for normal tracks, 3600 // with a few modifications for fast tracks. 3601 bool isActive = true; 3602 switch (track->mState) { 3603 case TrackBase::STOPPING_1: 3604 // track stays active in STOPPING_1 state until first underrun 3605 if (recentUnderruns > 0 || track->isTerminated()) { 3606 track->mState = TrackBase::STOPPING_2; 3607 } 3608 break; 3609 case TrackBase::PAUSING: 3610 // ramp down is not yet implemented 3611 track->setPaused(); 3612 break; 3613 case TrackBase::RESUMING: 3614 // ramp up is not yet implemented 3615 track->mState = TrackBase::ACTIVE; 3616 break; 3617 case TrackBase::ACTIVE: 3618 if (recentFull > 0 || recentPartial > 0) { 3619 // track has provided at least some frames recently: reset retry count 3620 track->mRetryCount = kMaxTrackRetries; 3621 } 3622 if (recentUnderruns == 0) { 3623 // no recent underruns: stay active 3624 break; 3625 } 3626 // there has recently been an underrun of some kind 3627 if (track->sharedBuffer() == 0) { 3628 // were any of the recent underruns "empty" (no frames available)? 3629 if (recentEmpty == 0) { 3630 // no, then ignore the partial underruns as they are allowed indefinitely 3631 break; 3632 } 3633 // there has recently been an "empty" underrun: decrement the retry counter 3634 if (--(track->mRetryCount) > 0) { 3635 break; 3636 } 3637 // indicate to client process that the track was disabled because of underrun; 3638 // it will then automatically call start() when data is available 3639 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3640 // remove from active list, but state remains ACTIVE [confusing but true] 3641 isActive = false; 3642 break; 3643 } 3644 // fall through 3645 case TrackBase::STOPPING_2: 3646 case TrackBase::PAUSED: 3647 case TrackBase::STOPPED: 3648 case TrackBase::FLUSHED: // flush() while active 3649 // Check for presentation complete if track is inactive 3650 // We have consumed all the buffers of this track. 3651 // This would be incomplete if we auto-paused on underrun 3652 { 3653 size_t audioHALFrames = 3654 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3655 size_t framesWritten = mBytesWritten / mFrameSize; 3656 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3657 // track stays in active list until presentation is complete 3658 break; 3659 } 3660 } 3661 if (track->isStopping_2()) { 3662 track->mState = TrackBase::STOPPED; 3663 } 3664 if (track->isStopped()) { 3665 // Can't reset directly, as fast mixer is still polling this track 3666 // track->reset(); 3667 // So instead mark this track as needing to be reset after push with ack 3668 resetMask |= 1 << i; 3669 } 3670 isActive = false; 3671 break; 3672 case TrackBase::IDLE: 3673 default: 3674 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3675 } 3676 3677 if (isActive) { 3678 // was it previously inactive? 3679 if (!(state->mTrackMask & (1 << j))) { 3680 ExtendedAudioBufferProvider *eabp = track; 3681 VolumeProvider *vp = track; 3682 fastTrack->mBufferProvider = eabp; 3683 fastTrack->mVolumeProvider = vp; 3684 fastTrack->mChannelMask = track->mChannelMask; 3685 fastTrack->mFormat = track->mFormat; 3686 fastTrack->mGeneration++; 3687 state->mTrackMask |= 1 << j; 3688 didModify = true; 3689 // no acknowledgement required for newly active tracks 3690 } 3691 // cache the combined master volume and stream type volume for fast mixer; this 3692 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3693 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3694 ++fastTracks; 3695 } else { 3696 // was it previously active? 3697 if (state->mTrackMask & (1 << j)) { 3698 fastTrack->mBufferProvider = NULL; 3699 fastTrack->mGeneration++; 3700 state->mTrackMask &= ~(1 << j); 3701 didModify = true; 3702 // If any fast tracks were removed, we must wait for acknowledgement 3703 // because we're about to decrement the last sp<> on those tracks. 3704 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3705 } else { 3706 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3707 } 3708 tracksToRemove->add(track); 3709 // Avoids a misleading display in dumpsys 3710 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3711 } 3712 continue; 3713 } 3714 3715 { // local variable scope to avoid goto warning 3716 3717 audio_track_cblk_t* cblk = track->cblk(); 3718 3719 // The first time a track is added we wait 3720 // for all its buffers to be filled before processing it 3721 int name = track->name(); 3722 // make sure that we have enough frames to mix one full buffer. 3723 // enforce this condition only once to enable draining the buffer in case the client 3724 // app does not call stop() and relies on underrun to stop: 3725 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3726 // during last round 3727 size_t desiredFrames; 3728 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3729 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3730 3731 desiredFrames = sourceFramesNeededWithTimestretch( 3732 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3733 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3734 // add frames already consumed but not yet released by the resampler 3735 // because mAudioTrackServerProxy->framesReady() will include these frames 3736 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3737 3738 uint32_t minFrames = 1; 3739 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3740 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3741 minFrames = desiredFrames; 3742 } 3743 3744 size_t framesReady = track->framesReady(); 3745 if (ATRACE_ENABLED()) { 3746 // I wish we had formatted trace names 3747 char traceName[16]; 3748 strcpy(traceName, "nRdy"); 3749 int name = track->name(); 3750 if (AudioMixer::TRACK0 <= name && 3751 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3752 name -= AudioMixer::TRACK0; 3753 traceName[4] = (name / 10) + '0'; 3754 traceName[5] = (name % 10) + '0'; 3755 } else { 3756 traceName[4] = '?'; 3757 traceName[5] = '?'; 3758 } 3759 traceName[6] = '\0'; 3760 ATRACE_INT(traceName, framesReady); 3761 } 3762 if ((framesReady >= minFrames) && track->isReady() && 3763 !track->isPaused() && !track->isTerminated()) 3764 { 3765 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3766 3767 mixedTracks++; 3768 3769 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3770 // there is an effect chain connected to the track 3771 chain.clear(); 3772 if (track->mainBuffer() != mSinkBuffer && 3773 track->mainBuffer() != mMixerBuffer) { 3774 if (mEffectBufferEnabled) { 3775 mEffectBufferValid = true; // Later can set directly. 3776 } 3777 chain = getEffectChain_l(track->sessionId()); 3778 // Delegate volume control to effect in track effect chain if needed 3779 if (chain != 0) { 3780 tracksWithEffect++; 3781 } else { 3782 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3783 "session %d", 3784 name, track->sessionId()); 3785 } 3786 } 3787 3788 3789 int param = AudioMixer::VOLUME; 3790 if (track->mFillingUpStatus == Track::FS_FILLED) { 3791 // no ramp for the first volume setting 3792 track->mFillingUpStatus = Track::FS_ACTIVE; 3793 if (track->mState == TrackBase::RESUMING) { 3794 track->mState = TrackBase::ACTIVE; 3795 param = AudioMixer::RAMP_VOLUME; 3796 } 3797 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3798 // FIXME should not make a decision based on mServer 3799 } else if (cblk->mServer != 0) { 3800 // If the track is stopped before the first frame was mixed, 3801 // do not apply ramp 3802 param = AudioMixer::RAMP_VOLUME; 3803 } 3804 3805 // compute volume for this track 3806 uint32_t vl, vr; // in U8.24 integer format 3807 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3808 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3809 vl = vr = 0; 3810 vlf = vrf = vaf = 0.; 3811 if (track->isPausing()) { 3812 track->setPaused(); 3813 } 3814 } else { 3815 3816 // read original volumes with volume control 3817 float typeVolume = mStreamTypes[track->streamType()].volume; 3818 float v = masterVolume * typeVolume; 3819 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3820 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3821 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3822 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3823 // track volumes come from shared memory, so can't be trusted and must be clamped 3824 if (vlf > GAIN_FLOAT_UNITY) { 3825 ALOGV("Track left volume out of range: %.3g", vlf); 3826 vlf = GAIN_FLOAT_UNITY; 3827 } 3828 if (vrf > GAIN_FLOAT_UNITY) { 3829 ALOGV("Track right volume out of range: %.3g", vrf); 3830 vrf = GAIN_FLOAT_UNITY; 3831 } 3832 // now apply the master volume and stream type volume 3833 vlf *= v; 3834 vrf *= v; 3835 // assuming master volume and stream type volume each go up to 1.0, 3836 // then derive vl and vr as U8.24 versions for the effect chain 3837 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3838 vl = (uint32_t) (scaleto8_24 * vlf); 3839 vr = (uint32_t) (scaleto8_24 * vrf); 3840 // vl and vr are now in U8.24 format 3841 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3842 // send level comes from shared memory and so may be corrupt 3843 if (sendLevel > MAX_GAIN_INT) { 3844 ALOGV("Track send level out of range: %04X", sendLevel); 3845 sendLevel = MAX_GAIN_INT; 3846 } 3847 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3848 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3849 } 3850 3851 // Delegate volume control to effect in track effect chain if needed 3852 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3853 // Do not ramp volume if volume is controlled by effect 3854 param = AudioMixer::VOLUME; 3855 // Update remaining floating point volume levels 3856 vlf = (float)vl / (1 << 24); 3857 vrf = (float)vr / (1 << 24); 3858 track->mHasVolumeController = true; 3859 } else { 3860 // force no volume ramp when volume controller was just disabled or removed 3861 // from effect chain to avoid volume spike 3862 if (track->mHasVolumeController) { 3863 param = AudioMixer::VOLUME; 3864 } 3865 track->mHasVolumeController = false; 3866 } 3867 3868 // XXX: these things DON'T need to be done each time 3869 mAudioMixer->setBufferProvider(name, track); 3870 mAudioMixer->enable(name); 3871 3872 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3873 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3874 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3875 mAudioMixer->setParameter( 3876 name, 3877 AudioMixer::TRACK, 3878 AudioMixer::FORMAT, (void *)track->format()); 3879 mAudioMixer->setParameter( 3880 name, 3881 AudioMixer::TRACK, 3882 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3883 mAudioMixer->setParameter( 3884 name, 3885 AudioMixer::TRACK, 3886 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3887 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3888 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3889 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3890 if (reqSampleRate == 0) { 3891 reqSampleRate = mSampleRate; 3892 } else if (reqSampleRate > maxSampleRate) { 3893 reqSampleRate = maxSampleRate; 3894 } 3895 mAudioMixer->setParameter( 3896 name, 3897 AudioMixer::RESAMPLE, 3898 AudioMixer::SAMPLE_RATE, 3899 (void *)(uintptr_t)reqSampleRate); 3900 3901 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3902 mAudioMixer->setParameter( 3903 name, 3904 AudioMixer::TIMESTRETCH, 3905 AudioMixer::PLAYBACK_RATE, 3906 &playbackRate); 3907 3908 /* 3909 * Select the appropriate output buffer for the track. 3910 * 3911 * Tracks with effects go into their own effects chain buffer 3912 * and from there into either mEffectBuffer or mSinkBuffer. 3913 * 3914 * Other tracks can use mMixerBuffer for higher precision 3915 * channel accumulation. If this buffer is enabled 3916 * (mMixerBufferEnabled true), then selected tracks will accumulate 3917 * into it. 3918 * 3919 */ 3920 if (mMixerBufferEnabled 3921 && (track->mainBuffer() == mSinkBuffer 3922 || track->mainBuffer() == mMixerBuffer)) { 3923 mAudioMixer->setParameter( 3924 name, 3925 AudioMixer::TRACK, 3926 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3927 mAudioMixer->setParameter( 3928 name, 3929 AudioMixer::TRACK, 3930 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3931 // TODO: override track->mainBuffer()? 3932 mMixerBufferValid = true; 3933 } else { 3934 mAudioMixer->setParameter( 3935 name, 3936 AudioMixer::TRACK, 3937 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3938 mAudioMixer->setParameter( 3939 name, 3940 AudioMixer::TRACK, 3941 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3942 } 3943 mAudioMixer->setParameter( 3944 name, 3945 AudioMixer::TRACK, 3946 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3947 3948 // reset retry count 3949 track->mRetryCount = kMaxTrackRetries; 3950 3951 // If one track is ready, set the mixer ready if: 3952 // - the mixer was not ready during previous round OR 3953 // - no other track is not ready 3954 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3955 mixerStatus != MIXER_TRACKS_ENABLED) { 3956 mixerStatus = MIXER_TRACKS_READY; 3957 } 3958 } else { 3959 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3960 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3961 } 3962 // clear effect chain input buffer if an active track underruns to avoid sending 3963 // previous audio buffer again to effects 3964 chain = getEffectChain_l(track->sessionId()); 3965 if (chain != 0) { 3966 chain->clearInputBuffer(); 3967 } 3968 3969 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3970 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3971 track->isStopped() || track->isPaused()) { 3972 // We have consumed all the buffers of this track. 3973 // Remove it from the list of active tracks. 3974 // TODO: use actual buffer filling status instead of latency when available from 3975 // audio HAL 3976 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3977 size_t framesWritten = mBytesWritten / mFrameSize; 3978 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3979 if (track->isStopped()) { 3980 track->reset(); 3981 } 3982 tracksToRemove->add(track); 3983 } 3984 } else { 3985 // No buffers for this track. Give it a few chances to 3986 // fill a buffer, then remove it from active list. 3987 if (--(track->mRetryCount) <= 0) { 3988 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3989 tracksToRemove->add(track); 3990 // indicate to client process that the track was disabled because of underrun; 3991 // it will then automatically call start() when data is available 3992 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3993 // If one track is not ready, mark the mixer also not ready if: 3994 // - the mixer was ready during previous round OR 3995 // - no other track is ready 3996 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3997 mixerStatus != MIXER_TRACKS_READY) { 3998 mixerStatus = MIXER_TRACKS_ENABLED; 3999 } 4000 } 4001 mAudioMixer->disable(name); 4002 } 4003 4004 } // local variable scope to avoid goto warning 4005track_is_ready: ; 4006 4007 } 4008 4009 // Push the new FastMixer state if necessary 4010 bool pauseAudioWatchdog = false; 4011 if (didModify) { 4012 state->mFastTracksGen++; 4013 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4014 if (kUseFastMixer == FastMixer_Dynamic && 4015 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4016 state->mCommand = FastMixerState::COLD_IDLE; 4017 state->mColdFutexAddr = &mFastMixerFutex; 4018 state->mColdGen++; 4019 mFastMixerFutex = 0; 4020 if (kUseFastMixer == FastMixer_Dynamic) { 4021 mNormalSink = mOutputSink; 4022 } 4023 // If we go into cold idle, need to wait for acknowledgement 4024 // so that fast mixer stops doing I/O. 4025 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4026 pauseAudioWatchdog = true; 4027 } 4028 } 4029 if (sq != NULL) { 4030 sq->end(didModify); 4031 sq->push(block); 4032 } 4033#ifdef AUDIO_WATCHDOG 4034 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4035 mAudioWatchdog->pause(); 4036 } 4037#endif 4038 4039 // Now perform the deferred reset on fast tracks that have stopped 4040 while (resetMask != 0) { 4041 size_t i = __builtin_ctz(resetMask); 4042 ALOG_ASSERT(i < count); 4043 resetMask &= ~(1 << i); 4044 sp<Track> t = mActiveTracks[i].promote(); 4045 if (t == 0) { 4046 continue; 4047 } 4048 Track* track = t.get(); 4049 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4050 track->reset(); 4051 } 4052 4053 // remove all the tracks that need to be... 4054 removeTracks_l(*tracksToRemove); 4055 4056 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4057 mEffectBufferValid = true; 4058 } 4059 4060 if (mEffectBufferValid) { 4061 // as long as there are effects we should clear the effects buffer, to avoid 4062 // passing a non-clean buffer to the effect chain 4063 memset(mEffectBuffer, 0, mEffectBufferSize); 4064 } 4065 // sink or mix buffer must be cleared if all tracks are connected to an 4066 // effect chain as in this case the mixer will not write to the sink or mix buffer 4067 // and track effects will accumulate into it 4068 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4069 (mixedTracks == 0 && fastTracks > 0))) { 4070 // FIXME as a performance optimization, should remember previous zero status 4071 if (mMixerBufferValid) { 4072 memset(mMixerBuffer, 0, mMixerBufferSize); 4073 // TODO: In testing, mSinkBuffer below need not be cleared because 4074 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4075 // after mixing. 4076 // 4077 // To enforce this guarantee: 4078 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4079 // (mixedTracks == 0 && fastTracks > 0)) 4080 // must imply MIXER_TRACKS_READY. 4081 // Later, we may clear buffers regardless, and skip much of this logic. 4082 } 4083 // FIXME as a performance optimization, should remember previous zero status 4084 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4085 } 4086 4087 // if any fast tracks, then status is ready 4088 mMixerStatusIgnoringFastTracks = mixerStatus; 4089 if (fastTracks > 0) { 4090 mixerStatus = MIXER_TRACKS_READY; 4091 } 4092 return mixerStatus; 4093} 4094 4095// getTrackName_l() must be called with ThreadBase::mLock held 4096int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4097 audio_format_t format, int sessionId) 4098{ 4099 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4100} 4101 4102// deleteTrackName_l() must be called with ThreadBase::mLock held 4103void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4104{ 4105 ALOGV("remove track (%d) and delete from mixer", name); 4106 mAudioMixer->deleteTrackName(name); 4107} 4108 4109// checkForNewParameter_l() must be called with ThreadBase::mLock held 4110bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4111 status_t& status) 4112{ 4113 bool reconfig = false; 4114 4115 status = NO_ERROR; 4116 4117 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4118 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4119 if (mFastMixer != 0) { 4120 FastMixerStateQueue *sq = mFastMixer->sq(); 4121 FastMixerState *state = sq->begin(); 4122 if (!(state->mCommand & FastMixerState::IDLE)) { 4123 previousCommand = state->mCommand; 4124 state->mCommand = FastMixerState::HOT_IDLE; 4125 sq->end(); 4126 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4127 } else { 4128 sq->end(false /*didModify*/); 4129 } 4130 } 4131 4132 AudioParameter param = AudioParameter(keyValuePair); 4133 int value; 4134 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4135 reconfig = true; 4136 } 4137 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4138 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4139 status = BAD_VALUE; 4140 } else { 4141 // no need to save value, since it's constant 4142 reconfig = true; 4143 } 4144 } 4145 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4146 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4147 status = BAD_VALUE; 4148 } else { 4149 // no need to save value, since it's constant 4150 reconfig = true; 4151 } 4152 } 4153 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4154 // do not accept frame count changes if tracks are open as the track buffer 4155 // size depends on frame count and correct behavior would not be guaranteed 4156 // if frame count is changed after track creation 4157 if (!mTracks.isEmpty()) { 4158 status = INVALID_OPERATION; 4159 } else { 4160 reconfig = true; 4161 } 4162 } 4163 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4164#ifdef ADD_BATTERY_DATA 4165 // when changing the audio output device, call addBatteryData to notify 4166 // the change 4167 if (mOutDevice != value) { 4168 uint32_t params = 0; 4169 // check whether speaker is on 4170 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4171 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4172 } 4173 4174 audio_devices_t deviceWithoutSpeaker 4175 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4176 // check if any other device (except speaker) is on 4177 if (value & deviceWithoutSpeaker) { 4178 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4179 } 4180 4181 if (params != 0) { 4182 addBatteryData(params); 4183 } 4184 } 4185#endif 4186 4187 // forward device change to effects that have requested to be 4188 // aware of attached audio device. 4189 if (value != AUDIO_DEVICE_NONE) { 4190 mOutDevice = value; 4191 for (size_t i = 0; i < mEffectChains.size(); i++) { 4192 mEffectChains[i]->setDevice_l(mOutDevice); 4193 } 4194 } 4195 } 4196 4197 if (status == NO_ERROR) { 4198 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4199 keyValuePair.string()); 4200 if (!mStandby && status == INVALID_OPERATION) { 4201 mOutput->standby(); 4202 mStandby = true; 4203 mBytesWritten = 0; 4204 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4205 keyValuePair.string()); 4206 } 4207 if (status == NO_ERROR && reconfig) { 4208 readOutputParameters_l(); 4209 delete mAudioMixer; 4210 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4211 for (size_t i = 0; i < mTracks.size() ; i++) { 4212 int name = getTrackName_l(mTracks[i]->mChannelMask, 4213 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4214 if (name < 0) { 4215 break; 4216 } 4217 mTracks[i]->mName = name; 4218 } 4219 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4220 } 4221 } 4222 4223 if (!(previousCommand & FastMixerState::IDLE)) { 4224 ALOG_ASSERT(mFastMixer != 0); 4225 FastMixerStateQueue *sq = mFastMixer->sq(); 4226 FastMixerState *state = sq->begin(); 4227 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4228 state->mCommand = previousCommand; 4229 sq->end(); 4230 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4231 } 4232 4233 return reconfig; 4234} 4235 4236 4237void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4238{ 4239 const size_t SIZE = 256; 4240 char buffer[SIZE]; 4241 String8 result; 4242 4243 PlaybackThread::dumpInternals(fd, args); 4244 4245 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4246 4247 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4248 const FastMixerDumpState copy(mFastMixerDumpState); 4249 copy.dump(fd); 4250 4251#ifdef STATE_QUEUE_DUMP 4252 // Similar for state queue 4253 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4254 observerCopy.dump(fd); 4255 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4256 mutatorCopy.dump(fd); 4257#endif 4258 4259#ifdef TEE_SINK 4260 // Write the tee output to a .wav file 4261 dumpTee(fd, mTeeSource, mId); 4262#endif 4263 4264#ifdef AUDIO_WATCHDOG 4265 if (mAudioWatchdog != 0) { 4266 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4267 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4268 wdCopy.dump(fd); 4269 } 4270#endif 4271} 4272 4273uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4274{ 4275 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4276} 4277 4278uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4279{ 4280 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4281} 4282 4283void AudioFlinger::MixerThread::cacheParameters_l() 4284{ 4285 PlaybackThread::cacheParameters_l(); 4286 4287 // FIXME: Relaxed timing because of a certain device that can't meet latency 4288 // Should be reduced to 2x after the vendor fixes the driver issue 4289 // increase threshold again due to low power audio mode. The way this warning 4290 // threshold is calculated and its usefulness should be reconsidered anyway. 4291 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4292} 4293 4294// ---------------------------------------------------------------------------- 4295 4296AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4297 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4298 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4299 // mLeftVolFloat, mRightVolFloat 4300{ 4301} 4302 4303AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4304 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4305 ThreadBase::type_t type) 4306 : PlaybackThread(audioFlinger, output, id, device, type) 4307 // mLeftVolFloat, mRightVolFloat 4308{ 4309} 4310 4311AudioFlinger::DirectOutputThread::~DirectOutputThread() 4312{ 4313} 4314 4315void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4316{ 4317 audio_track_cblk_t* cblk = track->cblk(); 4318 float left, right; 4319 4320 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4321 left = right = 0; 4322 } else { 4323 float typeVolume = mStreamTypes[track->streamType()].volume; 4324 float v = mMasterVolume * typeVolume; 4325 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4326 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4327 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4328 if (left > GAIN_FLOAT_UNITY) { 4329 left = GAIN_FLOAT_UNITY; 4330 } 4331 left *= v; 4332 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4333 if (right > GAIN_FLOAT_UNITY) { 4334 right = GAIN_FLOAT_UNITY; 4335 } 4336 right *= v; 4337 } 4338 4339 if (lastTrack) { 4340 if (left != mLeftVolFloat || right != mRightVolFloat) { 4341 mLeftVolFloat = left; 4342 mRightVolFloat = right; 4343 4344 // Convert volumes from float to 8.24 4345 uint32_t vl = (uint32_t)(left * (1 << 24)); 4346 uint32_t vr = (uint32_t)(right * (1 << 24)); 4347 4348 // Delegate volume control to effect in track effect chain if needed 4349 // only one effect chain can be present on DirectOutputThread, so if 4350 // there is one, the track is connected to it 4351 if (!mEffectChains.isEmpty()) { 4352 mEffectChains[0]->setVolume_l(&vl, &vr); 4353 left = (float)vl / (1 << 24); 4354 right = (float)vr / (1 << 24); 4355 } 4356 if (mOutput->stream->set_volume) { 4357 mOutput->stream->set_volume(mOutput->stream, left, right); 4358 } 4359 } 4360 } 4361} 4362 4363 4364AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4365 Vector< sp<Track> > *tracksToRemove 4366) 4367{ 4368 size_t count = mActiveTracks.size(); 4369 mixer_state mixerStatus = MIXER_IDLE; 4370 bool doHwPause = false; 4371 bool doHwResume = false; 4372 bool flushPending = false; 4373 4374 // find out which tracks need to be processed 4375 for (size_t i = 0; i < count; i++) { 4376 sp<Track> t = mActiveTracks[i].promote(); 4377 // The track died recently 4378 if (t == 0) { 4379 continue; 4380 } 4381 4382 Track* const track = t.get(); 4383 audio_track_cblk_t* cblk = track->cblk(); 4384 // Only consider last track started for volume and mixer state control. 4385 // In theory an older track could underrun and restart after the new one starts 4386 // but as we only care about the transition phase between two tracks on a 4387 // direct output, it is not a problem to ignore the underrun case. 4388 sp<Track> l = mLatestActiveTrack.promote(); 4389 bool last = l.get() == track; 4390 4391 if (track->isPausing()) { 4392 track->setPaused(); 4393 if (mHwSupportsPause && last && !mHwPaused) { 4394 doHwPause = true; 4395 mHwPaused = true; 4396 } 4397 tracksToRemove->add(track); 4398 } else if (track->isFlushPending()) { 4399 track->flushAck(); 4400 if (last) { 4401 flushPending = true; 4402 } 4403 } else if (track->isResumePending()) { 4404 track->resumeAck(); 4405 if (last && mHwPaused) { 4406 doHwResume = true; 4407 mHwPaused = false; 4408 } 4409 } 4410 4411 // The first time a track is added we wait 4412 // for all its buffers to be filled before processing it. 4413 // Allow draining the buffer in case the client 4414 // app does not call stop() and relies on underrun to stop: 4415 // hence the test on (track->mRetryCount > 1). 4416 // If retryCount<=1 then track is about to underrun and be removed. 4417 uint32_t minFrames; 4418 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4419 && (track->mRetryCount > 1)) { 4420 minFrames = mNormalFrameCount; 4421 } else { 4422 minFrames = 1; 4423 } 4424 4425 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4426 !track->isStopping_2() && !track->isStopped()) 4427 { 4428 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4429 4430 if (track->mFillingUpStatus == Track::FS_FILLED) { 4431 track->mFillingUpStatus = Track::FS_ACTIVE; 4432 // make sure processVolume_l() will apply new volume even if 0 4433 mLeftVolFloat = mRightVolFloat = -1.0; 4434 if (!mHwSupportsPause) { 4435 track->resumeAck(); 4436 } 4437 } 4438 4439 // compute volume for this track 4440 processVolume_l(track, last); 4441 if (last) { 4442 // reset retry count 4443 track->mRetryCount = kMaxTrackRetriesDirect; 4444 mActiveTrack = t; 4445 mixerStatus = MIXER_TRACKS_READY; 4446 if (usesHwAvSync() && mHwPaused) { 4447 doHwResume = true; 4448 mHwPaused = false; 4449 } 4450 } 4451 } else { 4452 // clear effect chain input buffer if the last active track started underruns 4453 // to avoid sending previous audio buffer again to effects 4454 if (!mEffectChains.isEmpty() && last) { 4455 mEffectChains[0]->clearInputBuffer(); 4456 } 4457 if (track->isStopping_1()) { 4458 track->mState = TrackBase::STOPPING_2; 4459 if (last && mHwPaused) { 4460 doHwResume = true; 4461 mHwPaused = false; 4462 } 4463 } 4464 if ((track->sharedBuffer() != 0) || track->isStopped() || 4465 track->isStopping_2() || track->isPaused()) { 4466 // We have consumed all the buffers of this track. 4467 // Remove it from the list of active tracks. 4468 size_t audioHALFrames; 4469 if (audio_is_linear_pcm(mFormat)) { 4470 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4471 } else { 4472 audioHALFrames = 0; 4473 } 4474 4475 size_t framesWritten = mBytesWritten / mFrameSize; 4476 if (mStandby || !last || 4477 track->presentationComplete(framesWritten, audioHALFrames)) { 4478 if (track->isStopping_2()) { 4479 track->mState = TrackBase::STOPPED; 4480 } 4481 if (track->isStopped()) { 4482 track->reset(); 4483 } 4484 tracksToRemove->add(track); 4485 } 4486 } else { 4487 // No buffers for this track. Give it a few chances to 4488 // fill a buffer, then remove it from active list. 4489 // Only consider last track started for mixer state control 4490 if (--(track->mRetryCount) <= 0) { 4491 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4492 tracksToRemove->add(track); 4493 // indicate to client process that the track was disabled because of underrun; 4494 // it will then automatically call start() when data is available 4495 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4496 } else if (last) { 4497 mixerStatus = MIXER_TRACKS_ENABLED; 4498 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4499 doHwPause = true; 4500 mHwPaused = true; 4501 } 4502 } 4503 } 4504 } 4505 } 4506 4507 // if an active track did not command a flush, check for pending flush on stopped tracks 4508 if (!flushPending) { 4509 for (size_t i = 0; i < mTracks.size(); i++) { 4510 if (mTracks[i]->isFlushPending()) { 4511 mTracks[i]->flushAck(); 4512 flushPending = true; 4513 } 4514 } 4515 } 4516 4517 // make sure the pause/flush/resume sequence is executed in the right order. 4518 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4519 // before flush and then resume HW. This can happen in case of pause/flush/resume 4520 // if resume is received before pause is executed. 4521 if (mHwSupportsPause && !mStandby && 4522 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4523 mOutput->stream->pause(mOutput->stream); 4524 } 4525 if (flushPending) { 4526 flushHw_l(); 4527 } 4528 if (mHwSupportsPause && !mStandby && doHwResume) { 4529 mOutput->stream->resume(mOutput->stream); 4530 } 4531 // remove all the tracks that need to be... 4532 removeTracks_l(*tracksToRemove); 4533 4534 return mixerStatus; 4535} 4536 4537void AudioFlinger::DirectOutputThread::threadLoop_mix() 4538{ 4539 size_t frameCount = mFrameCount; 4540 int8_t *curBuf = (int8_t *)mSinkBuffer; 4541 // output audio to hardware 4542 while (frameCount) { 4543 AudioBufferProvider::Buffer buffer; 4544 buffer.frameCount = frameCount; 4545 status_t status = mActiveTrack->getNextBuffer(&buffer); 4546 if (status != NO_ERROR || buffer.raw == NULL) { 4547 memset(curBuf, 0, frameCount * mFrameSize); 4548 break; 4549 } 4550 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4551 frameCount -= buffer.frameCount; 4552 curBuf += buffer.frameCount * mFrameSize; 4553 mActiveTrack->releaseBuffer(&buffer); 4554 } 4555 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4556 sleepTime = 0; 4557 standbyTime = systemTime() + standbyDelay; 4558 mActiveTrack.clear(); 4559} 4560 4561void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4562{ 4563 // do not write to HAL when paused 4564 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4565 sleepTime = idleSleepTime; 4566 return; 4567 } 4568 if (sleepTime == 0) { 4569 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4570 sleepTime = activeSleepTime; 4571 } else { 4572 sleepTime = idleSleepTime; 4573 } 4574 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4575 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4576 sleepTime = 0; 4577 } 4578} 4579 4580void AudioFlinger::DirectOutputThread::threadLoop_exit() 4581{ 4582 { 4583 Mutex::Autolock _l(mLock); 4584 bool flushPending = false; 4585 for (size_t i = 0; i < mTracks.size(); i++) { 4586 if (mTracks[i]->isFlushPending()) { 4587 mTracks[i]->flushAck(); 4588 flushPending = true; 4589 } 4590 } 4591 if (flushPending) { 4592 flushHw_l(); 4593 } 4594 } 4595 PlaybackThread::threadLoop_exit(); 4596} 4597 4598// must be called with thread mutex locked 4599bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4600{ 4601 bool trackPaused = false; 4602 bool trackStopped = false; 4603 4604 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4605 // after a timeout and we will enter standby then. 4606 if (mTracks.size() > 0) { 4607 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4608 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4609 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4610 } 4611 4612 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped)); 4613} 4614 4615// getTrackName_l() must be called with ThreadBase::mLock held 4616int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4617 audio_format_t format __unused, int sessionId __unused) 4618{ 4619 return 0; 4620} 4621 4622// deleteTrackName_l() must be called with ThreadBase::mLock held 4623void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4624{ 4625} 4626 4627// checkForNewParameter_l() must be called with ThreadBase::mLock held 4628bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4629 status_t& status) 4630{ 4631 bool reconfig = false; 4632 4633 status = NO_ERROR; 4634 4635 AudioParameter param = AudioParameter(keyValuePair); 4636 int value; 4637 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4638 // forward device change to effects that have requested to be 4639 // aware of attached audio device. 4640 if (value != AUDIO_DEVICE_NONE) { 4641 mOutDevice = value; 4642 for (size_t i = 0; i < mEffectChains.size(); i++) { 4643 mEffectChains[i]->setDevice_l(mOutDevice); 4644 } 4645 } 4646 } 4647 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4648 // do not accept frame count changes if tracks are open as the track buffer 4649 // size depends on frame count and correct behavior would not be garantied 4650 // if frame count is changed after track creation 4651 if (!mTracks.isEmpty()) { 4652 status = INVALID_OPERATION; 4653 } else { 4654 reconfig = true; 4655 } 4656 } 4657 if (status == NO_ERROR) { 4658 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4659 keyValuePair.string()); 4660 if (!mStandby && status == INVALID_OPERATION) { 4661 mOutput->standby(); 4662 mStandby = true; 4663 mBytesWritten = 0; 4664 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4665 keyValuePair.string()); 4666 } 4667 if (status == NO_ERROR && reconfig) { 4668 readOutputParameters_l(); 4669 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4670 } 4671 } 4672 4673 return reconfig; 4674} 4675 4676uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4677{ 4678 uint32_t time; 4679 if (audio_is_linear_pcm(mFormat)) { 4680 time = PlaybackThread::activeSleepTimeUs(); 4681 } else { 4682 time = 10000; 4683 } 4684 return time; 4685} 4686 4687uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4688{ 4689 uint32_t time; 4690 if (audio_is_linear_pcm(mFormat)) { 4691 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4692 } else { 4693 time = 10000; 4694 } 4695 return time; 4696} 4697 4698uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4699{ 4700 uint32_t time; 4701 if (audio_is_linear_pcm(mFormat)) { 4702 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4703 } else { 4704 time = 10000; 4705 } 4706 return time; 4707} 4708 4709void AudioFlinger::DirectOutputThread::cacheParameters_l() 4710{ 4711 PlaybackThread::cacheParameters_l(); 4712 4713 // use shorter standby delay as on normal output to release 4714 // hardware resources as soon as possible 4715 // no delay on outputs with HW A/V sync 4716 if (usesHwAvSync()) { 4717 standbyDelay = 0; 4718 } else if (audio_is_linear_pcm(mFormat)) { 4719 standbyDelay = microseconds(activeSleepTime*2); 4720 } else { 4721 standbyDelay = kOffloadStandbyDelayNs; 4722 } 4723} 4724 4725void AudioFlinger::DirectOutputThread::flushHw_l() 4726{ 4727 mOutput->flush(); 4728 mHwPaused = false; 4729} 4730 4731// ---------------------------------------------------------------------------- 4732 4733AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4734 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4735 : Thread(false /*canCallJava*/), 4736 mPlaybackThread(playbackThread), 4737 mWriteAckSequence(0), 4738 mDrainSequence(0) 4739{ 4740} 4741 4742AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4743{ 4744} 4745 4746void AudioFlinger::AsyncCallbackThread::onFirstRef() 4747{ 4748 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4749} 4750 4751bool AudioFlinger::AsyncCallbackThread::threadLoop() 4752{ 4753 while (!exitPending()) { 4754 uint32_t writeAckSequence; 4755 uint32_t drainSequence; 4756 4757 { 4758 Mutex::Autolock _l(mLock); 4759 while (!((mWriteAckSequence & 1) || 4760 (mDrainSequence & 1) || 4761 exitPending())) { 4762 mWaitWorkCV.wait(mLock); 4763 } 4764 4765 if (exitPending()) { 4766 break; 4767 } 4768 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4769 mWriteAckSequence, mDrainSequence); 4770 writeAckSequence = mWriteAckSequence; 4771 mWriteAckSequence &= ~1; 4772 drainSequence = mDrainSequence; 4773 mDrainSequence &= ~1; 4774 } 4775 { 4776 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4777 if (playbackThread != 0) { 4778 if (writeAckSequence & 1) { 4779 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4780 } 4781 if (drainSequence & 1) { 4782 playbackThread->resetDraining(drainSequence >> 1); 4783 } 4784 } 4785 } 4786 } 4787 return false; 4788} 4789 4790void AudioFlinger::AsyncCallbackThread::exit() 4791{ 4792 ALOGV("AsyncCallbackThread::exit"); 4793 Mutex::Autolock _l(mLock); 4794 requestExit(); 4795 mWaitWorkCV.broadcast(); 4796} 4797 4798void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4799{ 4800 Mutex::Autolock _l(mLock); 4801 // bit 0 is cleared 4802 mWriteAckSequence = sequence << 1; 4803} 4804 4805void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4806{ 4807 Mutex::Autolock _l(mLock); 4808 // ignore unexpected callbacks 4809 if (mWriteAckSequence & 2) { 4810 mWriteAckSequence |= 1; 4811 mWaitWorkCV.signal(); 4812 } 4813} 4814 4815void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4816{ 4817 Mutex::Autolock _l(mLock); 4818 // bit 0 is cleared 4819 mDrainSequence = sequence << 1; 4820} 4821 4822void AudioFlinger::AsyncCallbackThread::resetDraining() 4823{ 4824 Mutex::Autolock _l(mLock); 4825 // ignore unexpected callbacks 4826 if (mDrainSequence & 2) { 4827 mDrainSequence |= 1; 4828 mWaitWorkCV.signal(); 4829 } 4830} 4831 4832 4833// ---------------------------------------------------------------------------- 4834AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4835 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4836 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4837 mPausedBytesRemaining(0) 4838{ 4839 //FIXME: mStandby should be set to true by ThreadBase constructor 4840 mStandby = true; 4841} 4842 4843void AudioFlinger::OffloadThread::threadLoop_exit() 4844{ 4845 if (mFlushPending || mHwPaused) { 4846 // If a flush is pending or track was paused, just discard buffered data 4847 flushHw_l(); 4848 } else { 4849 mMixerStatus = MIXER_DRAIN_ALL; 4850 threadLoop_drain(); 4851 } 4852 if (mUseAsyncWrite) { 4853 ALOG_ASSERT(mCallbackThread != 0); 4854 mCallbackThread->exit(); 4855 } 4856 PlaybackThread::threadLoop_exit(); 4857} 4858 4859AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4860 Vector< sp<Track> > *tracksToRemove 4861) 4862{ 4863 size_t count = mActiveTracks.size(); 4864 4865 mixer_state mixerStatus = MIXER_IDLE; 4866 bool doHwPause = false; 4867 bool doHwResume = false; 4868 4869 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4870 4871 // find out which tracks need to be processed 4872 for (size_t i = 0; i < count; i++) { 4873 sp<Track> t = mActiveTracks[i].promote(); 4874 // The track died recently 4875 if (t == 0) { 4876 continue; 4877 } 4878 Track* const track = t.get(); 4879 audio_track_cblk_t* cblk = track->cblk(); 4880 // Only consider last track started for volume and mixer state control. 4881 // In theory an older track could underrun and restart after the new one starts 4882 // but as we only care about the transition phase between two tracks on a 4883 // direct output, it is not a problem to ignore the underrun case. 4884 sp<Track> l = mLatestActiveTrack.promote(); 4885 bool last = l.get() == track; 4886 4887 if (track->isInvalid()) { 4888 ALOGW("An invalidated track shouldn't be in active list"); 4889 tracksToRemove->add(track); 4890 continue; 4891 } 4892 4893 if (track->mState == TrackBase::IDLE) { 4894 ALOGW("An idle track shouldn't be in active list"); 4895 continue; 4896 } 4897 4898 if (track->isPausing()) { 4899 track->setPaused(); 4900 if (last) { 4901 if (!mHwPaused) { 4902 doHwPause = true; 4903 mHwPaused = true; 4904 } 4905 // If we were part way through writing the mixbuffer to 4906 // the HAL we must save this until we resume 4907 // BUG - this will be wrong if a different track is made active, 4908 // in that case we want to discard the pending data in the 4909 // mixbuffer and tell the client to present it again when the 4910 // track is resumed 4911 mPausedWriteLength = mCurrentWriteLength; 4912 mPausedBytesRemaining = mBytesRemaining; 4913 mBytesRemaining = 0; // stop writing 4914 } 4915 tracksToRemove->add(track); 4916 } else if (track->isFlushPending()) { 4917 track->flushAck(); 4918 if (last) { 4919 mFlushPending = true; 4920 } 4921 } else if (track->isResumePending()){ 4922 track->resumeAck(); 4923 if (last) { 4924 if (mPausedBytesRemaining) { 4925 // Need to continue write that was interrupted 4926 mCurrentWriteLength = mPausedWriteLength; 4927 mBytesRemaining = mPausedBytesRemaining; 4928 mPausedBytesRemaining = 0; 4929 } 4930 if (mHwPaused) { 4931 doHwResume = true; 4932 mHwPaused = false; 4933 // threadLoop_mix() will handle the case that we need to 4934 // resume an interrupted write 4935 } 4936 // enable write to audio HAL 4937 sleepTime = 0; 4938 4939 // Do not handle new data in this iteration even if track->framesReady() 4940 mixerStatus = MIXER_TRACKS_ENABLED; 4941 } 4942 } else if (track->framesReady() && track->isReady() && 4943 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4944 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4945 if (track->mFillingUpStatus == Track::FS_FILLED) { 4946 track->mFillingUpStatus = Track::FS_ACTIVE; 4947 // make sure processVolume_l() will apply new volume even if 0 4948 mLeftVolFloat = mRightVolFloat = -1.0; 4949 } 4950 4951 if (last) { 4952 sp<Track> previousTrack = mPreviousTrack.promote(); 4953 if (previousTrack != 0) { 4954 if (track != previousTrack.get()) { 4955 // Flush any data still being written from last track 4956 mBytesRemaining = 0; 4957 if (mPausedBytesRemaining) { 4958 // Last track was paused so we also need to flush saved 4959 // mixbuffer state and invalidate track so that it will 4960 // re-submit that unwritten data when it is next resumed 4961 mPausedBytesRemaining = 0; 4962 // Invalidate is a bit drastic - would be more efficient 4963 // to have a flag to tell client that some of the 4964 // previously written data was lost 4965 previousTrack->invalidate(); 4966 } 4967 // flush data already sent to the DSP if changing audio session as audio 4968 // comes from a different source. Also invalidate previous track to force a 4969 // seek when resuming. 4970 if (previousTrack->sessionId() != track->sessionId()) { 4971 previousTrack->invalidate(); 4972 } 4973 } 4974 } 4975 mPreviousTrack = track; 4976 // reset retry count 4977 track->mRetryCount = kMaxTrackRetriesOffload; 4978 mActiveTrack = t; 4979 mixerStatus = MIXER_TRACKS_READY; 4980 } 4981 } else { 4982 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4983 if (track->isStopping_1()) { 4984 // Hardware buffer can hold a large amount of audio so we must 4985 // wait for all current track's data to drain before we say 4986 // that the track is stopped. 4987 if (mBytesRemaining == 0) { 4988 // Only start draining when all data in mixbuffer 4989 // has been written 4990 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4991 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4992 // do not drain if no data was ever sent to HAL (mStandby == true) 4993 if (last && !mStandby) { 4994 // do not modify drain sequence if we are already draining. This happens 4995 // when resuming from pause after drain. 4996 if ((mDrainSequence & 1) == 0) { 4997 sleepTime = 0; 4998 standbyTime = systemTime() + standbyDelay; 4999 mixerStatus = MIXER_DRAIN_TRACK; 5000 mDrainSequence += 2; 5001 } 5002 if (mHwPaused) { 5003 // It is possible to move from PAUSED to STOPPING_1 without 5004 // a resume so we must ensure hardware is running 5005 doHwResume = true; 5006 mHwPaused = false; 5007 } 5008 } 5009 } 5010 } else if (track->isStopping_2()) { 5011 // Drain has completed or we are in standby, signal presentation complete 5012 if (!(mDrainSequence & 1) || !last || mStandby) { 5013 track->mState = TrackBase::STOPPED; 5014 size_t audioHALFrames = 5015 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5016 size_t framesWritten = 5017 mBytesWritten / mOutput->getFrameSize(); 5018 track->presentationComplete(framesWritten, audioHALFrames); 5019 track->reset(); 5020 tracksToRemove->add(track); 5021 } 5022 } else { 5023 // No buffers for this track. Give it a few chances to 5024 // fill a buffer, then remove it from active list. 5025 if (--(track->mRetryCount) <= 0) { 5026 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5027 track->name()); 5028 tracksToRemove->add(track); 5029 // indicate to client process that the track was disabled because of underrun; 5030 // it will then automatically call start() when data is available 5031 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5032 } else if (last){ 5033 mixerStatus = MIXER_TRACKS_ENABLED; 5034 } 5035 } 5036 } 5037 // compute volume for this track 5038 processVolume_l(track, last); 5039 } 5040 5041 // make sure the pause/flush/resume sequence is executed in the right order. 5042 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5043 // before flush and then resume HW. This can happen in case of pause/flush/resume 5044 // if resume is received before pause is executed. 5045 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5046 mOutput->stream->pause(mOutput->stream); 5047 } 5048 if (mFlushPending) { 5049 flushHw_l(); 5050 mFlushPending = false; 5051 } 5052 if (!mStandby && doHwResume) { 5053 mOutput->stream->resume(mOutput->stream); 5054 } 5055 5056 // remove all the tracks that need to be... 5057 removeTracks_l(*tracksToRemove); 5058 5059 return mixerStatus; 5060} 5061 5062// must be called with thread mutex locked 5063bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5064{ 5065 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5066 mWriteAckSequence, mDrainSequence); 5067 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5068 return true; 5069 } 5070 return false; 5071} 5072 5073bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5074{ 5075 Mutex::Autolock _l(mLock); 5076 return waitingAsyncCallback_l(); 5077} 5078 5079void AudioFlinger::OffloadThread::flushHw_l() 5080{ 5081 DirectOutputThread::flushHw_l(); 5082 // Flush anything still waiting in the mixbuffer 5083 mCurrentWriteLength = 0; 5084 mBytesRemaining = 0; 5085 mPausedWriteLength = 0; 5086 mPausedBytesRemaining = 0; 5087 5088 if (mUseAsyncWrite) { 5089 // discard any pending drain or write ack by incrementing sequence 5090 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5091 mDrainSequence = (mDrainSequence + 2) & ~1; 5092 ALOG_ASSERT(mCallbackThread != 0); 5093 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5094 mCallbackThread->setDraining(mDrainSequence); 5095 } 5096} 5097 5098void AudioFlinger::OffloadThread::onAddNewTrack_l() 5099{ 5100 sp<Track> previousTrack = mPreviousTrack.promote(); 5101 sp<Track> latestTrack = mLatestActiveTrack.promote(); 5102 5103 if (previousTrack != 0 && latestTrack != 0 && 5104 (previousTrack->sessionId() != latestTrack->sessionId())) { 5105 mFlushPending = true; 5106 } 5107 PlaybackThread::onAddNewTrack_l(); 5108} 5109 5110// ---------------------------------------------------------------------------- 5111 5112AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5113 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 5114 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5115 DUPLICATING), 5116 mWaitTimeMs(UINT_MAX) 5117{ 5118 addOutputTrack(mainThread); 5119} 5120 5121AudioFlinger::DuplicatingThread::~DuplicatingThread() 5122{ 5123 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5124 mOutputTracks[i]->destroy(); 5125 } 5126} 5127 5128void AudioFlinger::DuplicatingThread::threadLoop_mix() 5129{ 5130 // mix buffers... 5131 if (outputsReady(outputTracks)) { 5132 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5133 } else { 5134 if (mMixerBufferValid) { 5135 memset(mMixerBuffer, 0, mMixerBufferSize); 5136 } else { 5137 memset(mSinkBuffer, 0, mSinkBufferSize); 5138 } 5139 } 5140 sleepTime = 0; 5141 writeFrames = mNormalFrameCount; 5142 mCurrentWriteLength = mSinkBufferSize; 5143 standbyTime = systemTime() + standbyDelay; 5144} 5145 5146void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5147{ 5148 if (sleepTime == 0) { 5149 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5150 sleepTime = activeSleepTime; 5151 } else { 5152 sleepTime = idleSleepTime; 5153 } 5154 } else if (mBytesWritten != 0) { 5155 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5156 writeFrames = mNormalFrameCount; 5157 memset(mSinkBuffer, 0, mSinkBufferSize); 5158 } else { 5159 // flush remaining overflow buffers in output tracks 5160 writeFrames = 0; 5161 } 5162 sleepTime = 0; 5163 } 5164} 5165 5166ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5167{ 5168 for (size_t i = 0; i < outputTracks.size(); i++) { 5169 outputTracks[i]->write(mSinkBuffer, writeFrames); 5170 } 5171 mStandby = false; 5172 return (ssize_t)mSinkBufferSize; 5173} 5174 5175void AudioFlinger::DuplicatingThread::threadLoop_standby() 5176{ 5177 // DuplicatingThread implements standby by stopping all tracks 5178 for (size_t i = 0; i < outputTracks.size(); i++) { 5179 outputTracks[i]->stop(); 5180 } 5181} 5182 5183void AudioFlinger::DuplicatingThread::saveOutputTracks() 5184{ 5185 outputTracks = mOutputTracks; 5186} 5187 5188void AudioFlinger::DuplicatingThread::clearOutputTracks() 5189{ 5190 outputTracks.clear(); 5191} 5192 5193void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5194{ 5195 Mutex::Autolock _l(mLock); 5196 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5197 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5198 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5199 const size_t frameCount = 5200 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5201 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5202 // from different OutputTracks and their associated MixerThreads (e.g. one may 5203 // nearly empty and the other may be dropping data). 5204 5205 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5206 this, 5207 mSampleRate, 5208 mFormat, 5209 mChannelMask, 5210 frameCount, 5211 IPCThreadState::self()->getCallingUid()); 5212 if (outputTrack->cblk() != NULL) { 5213 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5214 mOutputTracks.add(outputTrack); 5215 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5216 updateWaitTime_l(); 5217 } 5218} 5219 5220void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5221{ 5222 Mutex::Autolock _l(mLock); 5223 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5224 if (mOutputTracks[i]->thread() == thread) { 5225 mOutputTracks[i]->destroy(); 5226 mOutputTracks.removeAt(i); 5227 updateWaitTime_l(); 5228 if (thread->getOutput() == mOutput) { 5229 mOutput = NULL; 5230 } 5231 return; 5232 } 5233 } 5234 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5235} 5236 5237// caller must hold mLock 5238void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5239{ 5240 mWaitTimeMs = UINT_MAX; 5241 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5242 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5243 if (strong != 0) { 5244 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5245 if (waitTimeMs < mWaitTimeMs) { 5246 mWaitTimeMs = waitTimeMs; 5247 } 5248 } 5249 } 5250} 5251 5252 5253bool AudioFlinger::DuplicatingThread::outputsReady( 5254 const SortedVector< sp<OutputTrack> > &outputTracks) 5255{ 5256 for (size_t i = 0; i < outputTracks.size(); i++) { 5257 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5258 if (thread == 0) { 5259 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5260 outputTracks[i].get()); 5261 return false; 5262 } 5263 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5264 // see note at standby() declaration 5265 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5266 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5267 thread.get()); 5268 return false; 5269 } 5270 } 5271 return true; 5272} 5273 5274uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5275{ 5276 return (mWaitTimeMs * 1000) / 2; 5277} 5278 5279void AudioFlinger::DuplicatingThread::cacheParameters_l() 5280{ 5281 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5282 updateWaitTime_l(); 5283 5284 MixerThread::cacheParameters_l(); 5285} 5286 5287// ---------------------------------------------------------------------------- 5288// Record 5289// ---------------------------------------------------------------------------- 5290 5291AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5292 AudioStreamIn *input, 5293 audio_io_handle_t id, 5294 audio_devices_t outDevice, 5295 audio_devices_t inDevice 5296#ifdef TEE_SINK 5297 , const sp<NBAIO_Sink>& teeSink 5298#endif 5299 ) : 5300 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5301 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5302 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5303 mRsmpInRear(0) 5304#ifdef TEE_SINK 5305 , mTeeSink(teeSink) 5306#endif 5307 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5308 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5309 // mFastCapture below 5310 , mFastCaptureFutex(0) 5311 // mInputSource 5312 // mPipeSink 5313 // mPipeSource 5314 , mPipeFramesP2(0) 5315 // mPipeMemory 5316 // mFastCaptureNBLogWriter 5317 , mFastTrackAvail(false) 5318{ 5319 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5320 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5321 5322 readInputParameters_l(); 5323 5324 // create an NBAIO source for the HAL input stream, and negotiate 5325 mInputSource = new AudioStreamInSource(input->stream); 5326 size_t numCounterOffers = 0; 5327 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5328 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5329 ALOG_ASSERT(index == 0); 5330 5331 // initialize fast capture depending on configuration 5332 bool initFastCapture; 5333 switch (kUseFastCapture) { 5334 case FastCapture_Never: 5335 initFastCapture = false; 5336 break; 5337 case FastCapture_Always: 5338 initFastCapture = true; 5339 break; 5340 case FastCapture_Static: 5341 uint32_t primaryOutputSampleRate; 5342 { 5343 AutoMutex _l(audioFlinger->mHardwareLock); 5344 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5345 } 5346 initFastCapture = 5347 // either capture sample rate is same as (a reasonable) primary output sample rate 5348 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5349 (mSampleRate == primaryOutputSampleRate)) || 5350 // or primary output sample rate is unknown, and capture sample rate is reasonable 5351 ((primaryOutputSampleRate == 0) && 5352 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5353 // and the buffer size is < 12 ms 5354 (mFrameCount * 1000) / mSampleRate < 12; 5355 break; 5356 // case FastCapture_Dynamic: 5357 } 5358 5359 if (initFastCapture) { 5360 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5361 NBAIO_Format format = mInputSource->format(); 5362 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5363 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5364 void *pipeBuffer; 5365 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5366 sp<IMemory> pipeMemory; 5367 if ((roHeap == 0) || 5368 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5369 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5370 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5371 goto failed; 5372 } 5373 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5374 memset(pipeBuffer, 0, pipeSize); 5375 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5376 const NBAIO_Format offers[1] = {format}; 5377 size_t numCounterOffers = 0; 5378 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5379 ALOG_ASSERT(index == 0); 5380 mPipeSink = pipe; 5381 PipeReader *pipeReader = new PipeReader(*pipe); 5382 numCounterOffers = 0; 5383 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5384 ALOG_ASSERT(index == 0); 5385 mPipeSource = pipeReader; 5386 mPipeFramesP2 = pipeFramesP2; 5387 mPipeMemory = pipeMemory; 5388 5389 // create fast capture 5390 mFastCapture = new FastCapture(); 5391 FastCaptureStateQueue *sq = mFastCapture->sq(); 5392#ifdef STATE_QUEUE_DUMP 5393 // FIXME 5394#endif 5395 FastCaptureState *state = sq->begin(); 5396 state->mCblk = NULL; 5397 state->mInputSource = mInputSource.get(); 5398 state->mInputSourceGen++; 5399 state->mPipeSink = pipe; 5400 state->mPipeSinkGen++; 5401 state->mFrameCount = mFrameCount; 5402 state->mCommand = FastCaptureState::COLD_IDLE; 5403 // already done in constructor initialization list 5404 //mFastCaptureFutex = 0; 5405 state->mColdFutexAddr = &mFastCaptureFutex; 5406 state->mColdGen++; 5407 state->mDumpState = &mFastCaptureDumpState; 5408#ifdef TEE_SINK 5409 // FIXME 5410#endif 5411 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5412 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5413 sq->end(); 5414 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5415 5416 // start the fast capture 5417 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5418 pid_t tid = mFastCapture->getTid(); 5419 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5420 if (err != 0) { 5421 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5422 kPriorityFastCapture, getpid_cached, tid, err); 5423 } 5424 5425#ifdef AUDIO_WATCHDOG 5426 // FIXME 5427#endif 5428 5429 mFastTrackAvail = true; 5430 } 5431failed: ; 5432 5433 // FIXME mNormalSource 5434} 5435 5436AudioFlinger::RecordThread::~RecordThread() 5437{ 5438 if (mFastCapture != 0) { 5439 FastCaptureStateQueue *sq = mFastCapture->sq(); 5440 FastCaptureState *state = sq->begin(); 5441 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5442 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5443 if (old == -1) { 5444 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5445 } 5446 } 5447 state->mCommand = FastCaptureState::EXIT; 5448 sq->end(); 5449 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5450 mFastCapture->join(); 5451 mFastCapture.clear(); 5452 } 5453 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5454 mAudioFlinger->unregisterWriter(mNBLogWriter); 5455 free(mRsmpInBuffer); 5456} 5457 5458void AudioFlinger::RecordThread::onFirstRef() 5459{ 5460 run(mThreadName, PRIORITY_URGENT_AUDIO); 5461} 5462 5463bool AudioFlinger::RecordThread::threadLoop() 5464{ 5465 nsecs_t lastWarning = 0; 5466 5467 inputStandBy(); 5468 5469reacquire_wakelock: 5470 sp<RecordTrack> activeTrack; 5471 int activeTracksGen; 5472 { 5473 Mutex::Autolock _l(mLock); 5474 size_t size = mActiveTracks.size(); 5475 activeTracksGen = mActiveTracksGen; 5476 if (size > 0) { 5477 // FIXME an arbitrary choice 5478 activeTrack = mActiveTracks[0]; 5479 acquireWakeLock_l(activeTrack->uid()); 5480 if (size > 1) { 5481 SortedVector<int> tmp; 5482 for (size_t i = 0; i < size; i++) { 5483 tmp.add(mActiveTracks[i]->uid()); 5484 } 5485 updateWakeLockUids_l(tmp); 5486 } 5487 } else { 5488 acquireWakeLock_l(-1); 5489 } 5490 } 5491 5492 // used to request a deferred sleep, to be executed later while mutex is unlocked 5493 uint32_t sleepUs = 0; 5494 5495 // loop while there is work to do 5496 for (;;) { 5497 Vector< sp<EffectChain> > effectChains; 5498 5499 // sleep with mutex unlocked 5500 if (sleepUs > 0) { 5501 ATRACE_BEGIN("sleep"); 5502 usleep(sleepUs); 5503 ATRACE_END(); 5504 sleepUs = 0; 5505 } 5506 5507 // activeTracks accumulates a copy of a subset of mActiveTracks 5508 Vector< sp<RecordTrack> > activeTracks; 5509 5510 // reference to the (first and only) active fast track 5511 sp<RecordTrack> fastTrack; 5512 5513 // reference to a fast track which is about to be removed 5514 sp<RecordTrack> fastTrackToRemove; 5515 5516 { // scope for mLock 5517 Mutex::Autolock _l(mLock); 5518 5519 processConfigEvents_l(); 5520 5521 // check exitPending here because checkForNewParameters_l() and 5522 // checkForNewParameters_l() can temporarily release mLock 5523 if (exitPending()) { 5524 break; 5525 } 5526 5527 // if no active track(s), then standby and release wakelock 5528 size_t size = mActiveTracks.size(); 5529 if (size == 0) { 5530 standbyIfNotAlreadyInStandby(); 5531 // exitPending() can't become true here 5532 releaseWakeLock_l(); 5533 ALOGV("RecordThread: loop stopping"); 5534 // go to sleep 5535 mWaitWorkCV.wait(mLock); 5536 ALOGV("RecordThread: loop starting"); 5537 goto reacquire_wakelock; 5538 } 5539 5540 if (mActiveTracksGen != activeTracksGen) { 5541 activeTracksGen = mActiveTracksGen; 5542 SortedVector<int> tmp; 5543 for (size_t i = 0; i < size; i++) { 5544 tmp.add(mActiveTracks[i]->uid()); 5545 } 5546 updateWakeLockUids_l(tmp); 5547 } 5548 5549 bool doBroadcast = false; 5550 for (size_t i = 0; i < size; ) { 5551 5552 activeTrack = mActiveTracks[i]; 5553 if (activeTrack->isTerminated()) { 5554 if (activeTrack->isFastTrack()) { 5555 ALOG_ASSERT(fastTrackToRemove == 0); 5556 fastTrackToRemove = activeTrack; 5557 } 5558 removeTrack_l(activeTrack); 5559 mActiveTracks.remove(activeTrack); 5560 mActiveTracksGen++; 5561 size--; 5562 continue; 5563 } 5564 5565 TrackBase::track_state activeTrackState = activeTrack->mState; 5566 switch (activeTrackState) { 5567 5568 case TrackBase::PAUSING: 5569 mActiveTracks.remove(activeTrack); 5570 mActiveTracksGen++; 5571 doBroadcast = true; 5572 size--; 5573 continue; 5574 5575 case TrackBase::STARTING_1: 5576 sleepUs = 10000; 5577 i++; 5578 continue; 5579 5580 case TrackBase::STARTING_2: 5581 doBroadcast = true; 5582 mStandby = false; 5583 activeTrack->mState = TrackBase::ACTIVE; 5584 break; 5585 5586 case TrackBase::ACTIVE: 5587 break; 5588 5589 case TrackBase::IDLE: 5590 i++; 5591 continue; 5592 5593 default: 5594 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5595 } 5596 5597 activeTracks.add(activeTrack); 5598 i++; 5599 5600 if (activeTrack->isFastTrack()) { 5601 ALOG_ASSERT(!mFastTrackAvail); 5602 ALOG_ASSERT(fastTrack == 0); 5603 fastTrack = activeTrack; 5604 } 5605 } 5606 if (doBroadcast) { 5607 mStartStopCond.broadcast(); 5608 } 5609 5610 // sleep if there are no active tracks to process 5611 if (activeTracks.size() == 0) { 5612 if (sleepUs == 0) { 5613 sleepUs = kRecordThreadSleepUs; 5614 } 5615 continue; 5616 } 5617 sleepUs = 0; 5618 5619 lockEffectChains_l(effectChains); 5620 } 5621 5622 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5623 5624 size_t size = effectChains.size(); 5625 for (size_t i = 0; i < size; i++) { 5626 // thread mutex is not locked, but effect chain is locked 5627 effectChains[i]->process_l(); 5628 } 5629 5630 // Push a new fast capture state if fast capture is not already running, or cblk change 5631 if (mFastCapture != 0) { 5632 FastCaptureStateQueue *sq = mFastCapture->sq(); 5633 FastCaptureState *state = sq->begin(); 5634 bool didModify = false; 5635 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5636 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5637 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5638 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5639 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5640 if (old == -1) { 5641 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5642 } 5643 } 5644 state->mCommand = FastCaptureState::READ_WRITE; 5645#if 0 // FIXME 5646 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5647 FastThreadDumpState::kSamplingNforLowRamDevice : 5648 FastThreadDumpState::kSamplingN); 5649#endif 5650 didModify = true; 5651 } 5652 audio_track_cblk_t *cblkOld = state->mCblk; 5653 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5654 if (cblkNew != cblkOld) { 5655 state->mCblk = cblkNew; 5656 // block until acked if removing a fast track 5657 if (cblkOld != NULL) { 5658 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5659 } 5660 didModify = true; 5661 } 5662 sq->end(didModify); 5663 if (didModify) { 5664 sq->push(block); 5665#if 0 5666 if (kUseFastCapture == FastCapture_Dynamic) { 5667 mNormalSource = mPipeSource; 5668 } 5669#endif 5670 } 5671 } 5672 5673 // now run the fast track destructor with thread mutex unlocked 5674 fastTrackToRemove.clear(); 5675 5676 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5677 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5678 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5679 // If destination is non-contiguous, first read past the nominal end of buffer, then 5680 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5681 5682 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5683 ssize_t framesRead; 5684 5685 // If an NBAIO source is present, use it to read the normal capture's data 5686 if (mPipeSource != 0) { 5687 size_t framesToRead = mBufferSize / mFrameSize; 5688 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5689 framesToRead, AudioBufferProvider::kInvalidPTS); 5690 if (framesRead == 0) { 5691 // since pipe is non-blocking, simulate blocking input 5692 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5693 } 5694 // otherwise use the HAL / AudioStreamIn directly 5695 } else { 5696 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5697 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5698 if (bytesRead < 0) { 5699 framesRead = bytesRead; 5700 } else { 5701 framesRead = bytesRead / mFrameSize; 5702 } 5703 } 5704 5705 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5706 ALOGE("read failed: framesRead=%d", framesRead); 5707 // Force input into standby so that it tries to recover at next read attempt 5708 inputStandBy(); 5709 sleepUs = kRecordThreadSleepUs; 5710 } 5711 if (framesRead <= 0) { 5712 goto unlock; 5713 } 5714 ALOG_ASSERT(framesRead > 0); 5715 5716 if (mTeeSink != 0) { 5717 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5718 } 5719 // If destination is non-contiguous, we now correct for reading past end of buffer. 5720 { 5721 size_t part1 = mRsmpInFramesP2 - rear; 5722 if ((size_t) framesRead > part1) { 5723 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5724 (framesRead - part1) * mFrameSize); 5725 } 5726 } 5727 rear = mRsmpInRear += framesRead; 5728 5729 size = activeTracks.size(); 5730 // loop over each active track 5731 for (size_t i = 0; i < size; i++) { 5732 activeTrack = activeTracks[i]; 5733 5734 // skip fast tracks, as those are handled directly by FastCapture 5735 if (activeTrack->isFastTrack()) { 5736 continue; 5737 } 5738 5739 // TODO: This code probably should be moved to RecordTrack. 5740 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5741 5742 enum { 5743 OVERRUN_UNKNOWN, 5744 OVERRUN_TRUE, 5745 OVERRUN_FALSE 5746 } overrun = OVERRUN_UNKNOWN; 5747 5748 // loop over getNextBuffer to handle circular sink 5749 for (;;) { 5750 5751 activeTrack->mSink.frameCount = ~0; 5752 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5753 size_t framesOut = activeTrack->mSink.frameCount; 5754 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5755 5756 // check available frames and handle overrun conditions 5757 // if the record track isn't draining fast enough. 5758 bool hasOverrun; 5759 size_t framesIn; 5760 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5761 if (hasOverrun) { 5762 overrun = OVERRUN_TRUE; 5763 } 5764 if (framesOut == 0 || framesIn == 0) { 5765 break; 5766 } 5767 5768 // Don't allow framesOut to be larger than what is possible with resampling 5769 // from framesIn. 5770 // This isn't strictly necessary but helps limit buffer resizing in 5771 // RecordBufferConverter. TODO: remove when no longer needed. 5772 framesOut = min(framesOut, 5773 destinationFramesPossible( 5774 framesIn, mSampleRate, activeTrack->mSampleRate)); 5775 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5776 framesOut = activeTrack->mRecordBufferConverter->convert( 5777 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5778 5779 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5780 overrun = OVERRUN_FALSE; 5781 } 5782 5783 if (activeTrack->mFramesToDrop == 0) { 5784 if (framesOut > 0) { 5785 activeTrack->mSink.frameCount = framesOut; 5786 activeTrack->releaseBuffer(&activeTrack->mSink); 5787 } 5788 } else { 5789 // FIXME could do a partial drop of framesOut 5790 if (activeTrack->mFramesToDrop > 0) { 5791 activeTrack->mFramesToDrop -= framesOut; 5792 if (activeTrack->mFramesToDrop <= 0) { 5793 activeTrack->clearSyncStartEvent(); 5794 } 5795 } else { 5796 activeTrack->mFramesToDrop += framesOut; 5797 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5798 activeTrack->mSyncStartEvent->isCancelled()) { 5799 ALOGW("Synced record %s, session %d, trigger session %d", 5800 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5801 activeTrack->sessionId(), 5802 (activeTrack->mSyncStartEvent != 0) ? 5803 activeTrack->mSyncStartEvent->triggerSession() : 0); 5804 activeTrack->clearSyncStartEvent(); 5805 } 5806 } 5807 } 5808 5809 if (framesOut == 0) { 5810 break; 5811 } 5812 } 5813 5814 switch (overrun) { 5815 case OVERRUN_TRUE: 5816 // client isn't retrieving buffers fast enough 5817 if (!activeTrack->setOverflow()) { 5818 nsecs_t now = systemTime(); 5819 // FIXME should lastWarning per track? 5820 if ((now - lastWarning) > kWarningThrottleNs) { 5821 ALOGW("RecordThread: buffer overflow"); 5822 lastWarning = now; 5823 } 5824 } 5825 break; 5826 case OVERRUN_FALSE: 5827 activeTrack->clearOverflow(); 5828 break; 5829 case OVERRUN_UNKNOWN: 5830 break; 5831 } 5832 5833 } 5834 5835unlock: 5836 // enable changes in effect chain 5837 unlockEffectChains(effectChains); 5838 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5839 } 5840 5841 standbyIfNotAlreadyInStandby(); 5842 5843 { 5844 Mutex::Autolock _l(mLock); 5845 for (size_t i = 0; i < mTracks.size(); i++) { 5846 sp<RecordTrack> track = mTracks[i]; 5847 track->invalidate(); 5848 } 5849 mActiveTracks.clear(); 5850 mActiveTracksGen++; 5851 mStartStopCond.broadcast(); 5852 } 5853 5854 releaseWakeLock(); 5855 5856 ALOGV("RecordThread %p exiting", this); 5857 return false; 5858} 5859 5860void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5861{ 5862 if (!mStandby) { 5863 inputStandBy(); 5864 mStandby = true; 5865 } 5866} 5867 5868void AudioFlinger::RecordThread::inputStandBy() 5869{ 5870 // Idle the fast capture if it's currently running 5871 if (mFastCapture != 0) { 5872 FastCaptureStateQueue *sq = mFastCapture->sq(); 5873 FastCaptureState *state = sq->begin(); 5874 if (!(state->mCommand & FastCaptureState::IDLE)) { 5875 state->mCommand = FastCaptureState::COLD_IDLE; 5876 state->mColdFutexAddr = &mFastCaptureFutex; 5877 state->mColdGen++; 5878 mFastCaptureFutex = 0; 5879 sq->end(); 5880 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5881 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5882#if 0 5883 if (kUseFastCapture == FastCapture_Dynamic) { 5884 // FIXME 5885 } 5886#endif 5887#ifdef AUDIO_WATCHDOG 5888 // FIXME 5889#endif 5890 } else { 5891 sq->end(false /*didModify*/); 5892 } 5893 } 5894 mInput->stream->common.standby(&mInput->stream->common); 5895} 5896 5897// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5898sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5899 const sp<AudioFlinger::Client>& client, 5900 uint32_t sampleRate, 5901 audio_format_t format, 5902 audio_channel_mask_t channelMask, 5903 size_t *pFrameCount, 5904 int sessionId, 5905 size_t *notificationFrames, 5906 int uid, 5907 IAudioFlinger::track_flags_t *flags, 5908 pid_t tid, 5909 status_t *status) 5910{ 5911 size_t frameCount = *pFrameCount; 5912 sp<RecordTrack> track; 5913 status_t lStatus; 5914 5915 // client expresses a preference for FAST, but we get the final say 5916 if (*flags & IAudioFlinger::TRACK_FAST) { 5917 if ( 5918 // we formerly checked for a callback handler (non-0 tid), 5919 // but that is no longer required for TRANSFER_OBTAIN mode 5920 // 5921 // frame count is not specified, or is exactly the pipe depth 5922 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5923 // PCM data 5924 audio_is_linear_pcm(format) && 5925 // native format 5926 (format == mFormat) && 5927 // native channel mask 5928 (channelMask == mChannelMask) && 5929 // native hardware sample rate 5930 (sampleRate == mSampleRate) && 5931 // record thread has an associated fast capture 5932 hasFastCapture() && 5933 // there are sufficient fast track slots available 5934 mFastTrackAvail 5935 ) { 5936 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5937 frameCount, mFrameCount); 5938 } else { 5939 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5940 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5941 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5942 frameCount, mFrameCount, mPipeFramesP2, 5943 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5944 hasFastCapture(), tid, mFastTrackAvail); 5945 *flags &= ~IAudioFlinger::TRACK_FAST; 5946 } 5947 } 5948 5949 // compute track buffer size in frames, and suggest the notification frame count 5950 if (*flags & IAudioFlinger::TRACK_FAST) { 5951 // fast track: frame count is exactly the pipe depth 5952 frameCount = mPipeFramesP2; 5953 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5954 *notificationFrames = mFrameCount; 5955 } else { 5956 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5957 // or 20 ms if there is a fast capture 5958 // TODO This could be a roundupRatio inline, and const 5959 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5960 * sampleRate + mSampleRate - 1) / mSampleRate; 5961 // minimum number of notification periods is at least kMinNotifications, 5962 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5963 static const size_t kMinNotifications = 3; 5964 static const uint32_t kMinMs = 30; 5965 // TODO This could be a roundupRatio inline 5966 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5967 // TODO This could be a roundupRatio inline 5968 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5969 maxNotificationFrames; 5970 const size_t minFrameCount = maxNotificationFrames * 5971 max(kMinNotifications, minNotificationsByMs); 5972 frameCount = max(frameCount, minFrameCount); 5973 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5974 *notificationFrames = maxNotificationFrames; 5975 } 5976 } 5977 *pFrameCount = frameCount; 5978 5979 lStatus = initCheck(); 5980 if (lStatus != NO_ERROR) { 5981 ALOGE("createRecordTrack_l() audio driver not initialized"); 5982 goto Exit; 5983 } 5984 5985 { // scope for mLock 5986 Mutex::Autolock _l(mLock); 5987 5988 track = new RecordTrack(this, client, sampleRate, 5989 format, channelMask, frameCount, NULL, sessionId, uid, 5990 *flags, TrackBase::TYPE_DEFAULT); 5991 5992 lStatus = track->initCheck(); 5993 if (lStatus != NO_ERROR) { 5994 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5995 // track must be cleared from the caller as the caller has the AF lock 5996 goto Exit; 5997 } 5998 mTracks.add(track); 5999 6000 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6001 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6002 mAudioFlinger->btNrecIsOff(); 6003 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6004 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6005 6006 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6007 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6008 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6009 // so ask activity manager to do this on our behalf 6010 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6011 } 6012 } 6013 6014 lStatus = NO_ERROR; 6015 6016Exit: 6017 *status = lStatus; 6018 return track; 6019} 6020 6021status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6022 AudioSystem::sync_event_t event, 6023 int triggerSession) 6024{ 6025 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6026 sp<ThreadBase> strongMe = this; 6027 status_t status = NO_ERROR; 6028 6029 if (event == AudioSystem::SYNC_EVENT_NONE) { 6030 recordTrack->clearSyncStartEvent(); 6031 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6032 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6033 triggerSession, 6034 recordTrack->sessionId(), 6035 syncStartEventCallback, 6036 recordTrack); 6037 // Sync event can be cancelled by the trigger session if the track is not in a 6038 // compatible state in which case we start record immediately 6039 if (recordTrack->mSyncStartEvent->isCancelled()) { 6040 recordTrack->clearSyncStartEvent(); 6041 } else { 6042 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6043 recordTrack->mFramesToDrop = - 6044 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6045 } 6046 } 6047 6048 { 6049 // This section is a rendezvous between binder thread executing start() and RecordThread 6050 AutoMutex lock(mLock); 6051 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6052 if (recordTrack->mState == TrackBase::PAUSING) { 6053 ALOGV("active record track PAUSING -> ACTIVE"); 6054 recordTrack->mState = TrackBase::ACTIVE; 6055 } else { 6056 ALOGV("active record track state %d", recordTrack->mState); 6057 } 6058 return status; 6059 } 6060 6061 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6062 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6063 // or using a separate command thread 6064 recordTrack->mState = TrackBase::STARTING_1; 6065 mActiveTracks.add(recordTrack); 6066 mActiveTracksGen++; 6067 status_t status = NO_ERROR; 6068 if (recordTrack->isExternalTrack()) { 6069 mLock.unlock(); 6070 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6071 mLock.lock(); 6072 // FIXME should verify that recordTrack is still in mActiveTracks 6073 if (status != NO_ERROR) { 6074 mActiveTracks.remove(recordTrack); 6075 mActiveTracksGen++; 6076 recordTrack->clearSyncStartEvent(); 6077 ALOGV("RecordThread::start error %d", status); 6078 return status; 6079 } 6080 } 6081 // Catch up with current buffer indices if thread is already running. 6082 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6083 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6084 // see previously buffered data before it called start(), but with greater risk of overrun. 6085 6086 recordTrack->mResamplerBufferProvider->reset(); 6087 // clear any converter state as new data will be discontinuous 6088 recordTrack->mRecordBufferConverter->reset(); 6089 recordTrack->mState = TrackBase::STARTING_2; 6090 // signal thread to start 6091 mWaitWorkCV.broadcast(); 6092 if (mActiveTracks.indexOf(recordTrack) < 0) { 6093 ALOGV("Record failed to start"); 6094 status = BAD_VALUE; 6095 goto startError; 6096 } 6097 return status; 6098 } 6099 6100startError: 6101 if (recordTrack->isExternalTrack()) { 6102 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6103 } 6104 recordTrack->clearSyncStartEvent(); 6105 // FIXME I wonder why we do not reset the state here? 6106 return status; 6107} 6108 6109void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6110{ 6111 sp<SyncEvent> strongEvent = event.promote(); 6112 6113 if (strongEvent != 0) { 6114 sp<RefBase> ptr = strongEvent->cookie().promote(); 6115 if (ptr != 0) { 6116 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6117 recordTrack->handleSyncStartEvent(strongEvent); 6118 } 6119 } 6120} 6121 6122bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6123 ALOGV("RecordThread::stop"); 6124 AutoMutex _l(mLock); 6125 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6126 return false; 6127 } 6128 // note that threadLoop may still be processing the track at this point [without lock] 6129 recordTrack->mState = TrackBase::PAUSING; 6130 // do not wait for mStartStopCond if exiting 6131 if (exitPending()) { 6132 return true; 6133 } 6134 // FIXME incorrect usage of wait: no explicit predicate or loop 6135 mStartStopCond.wait(mLock); 6136 // if we have been restarted, recordTrack is in mActiveTracks here 6137 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6138 ALOGV("Record stopped OK"); 6139 return true; 6140 } 6141 return false; 6142} 6143 6144bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6145{ 6146 return false; 6147} 6148 6149status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6150{ 6151#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6152 if (!isValidSyncEvent(event)) { 6153 return BAD_VALUE; 6154 } 6155 6156 int eventSession = event->triggerSession(); 6157 status_t ret = NAME_NOT_FOUND; 6158 6159 Mutex::Autolock _l(mLock); 6160 6161 for (size_t i = 0; i < mTracks.size(); i++) { 6162 sp<RecordTrack> track = mTracks[i]; 6163 if (eventSession == track->sessionId()) { 6164 (void) track->setSyncEvent(event); 6165 ret = NO_ERROR; 6166 } 6167 } 6168 return ret; 6169#else 6170 return BAD_VALUE; 6171#endif 6172} 6173 6174// destroyTrack_l() must be called with ThreadBase::mLock held 6175void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6176{ 6177 track->terminate(); 6178 track->mState = TrackBase::STOPPED; 6179 // active tracks are removed by threadLoop() 6180 if (mActiveTracks.indexOf(track) < 0) { 6181 removeTrack_l(track); 6182 } 6183} 6184 6185void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6186{ 6187 mTracks.remove(track); 6188 // need anything related to effects here? 6189 if (track->isFastTrack()) { 6190 ALOG_ASSERT(!mFastTrackAvail); 6191 mFastTrackAvail = true; 6192 } 6193} 6194 6195void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6196{ 6197 dumpInternals(fd, args); 6198 dumpTracks(fd, args); 6199 dumpEffectChains(fd, args); 6200} 6201 6202void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6203{ 6204 dprintf(fd, "\nInput thread %p:\n", this); 6205 6206 dumpBase(fd, args); 6207 6208 if (mActiveTracks.size() == 0) { 6209 dprintf(fd, " No active record clients\n"); 6210 } 6211 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6212 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6213 6214 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6215 const FastCaptureDumpState copy(mFastCaptureDumpState); 6216 copy.dump(fd); 6217} 6218 6219void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6220{ 6221 const size_t SIZE = 256; 6222 char buffer[SIZE]; 6223 String8 result; 6224 6225 size_t numtracks = mTracks.size(); 6226 size_t numactive = mActiveTracks.size(); 6227 size_t numactiveseen = 0; 6228 dprintf(fd, " %d Tracks", numtracks); 6229 if (numtracks) { 6230 dprintf(fd, " of which %d are active\n", numactive); 6231 RecordTrack::appendDumpHeader(result); 6232 for (size_t i = 0; i < numtracks ; ++i) { 6233 sp<RecordTrack> track = mTracks[i]; 6234 if (track != 0) { 6235 bool active = mActiveTracks.indexOf(track) >= 0; 6236 if (active) { 6237 numactiveseen++; 6238 } 6239 track->dump(buffer, SIZE, active); 6240 result.append(buffer); 6241 } 6242 } 6243 } else { 6244 dprintf(fd, "\n"); 6245 } 6246 6247 if (numactiveseen != numactive) { 6248 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6249 " not in the track list\n"); 6250 result.append(buffer); 6251 RecordTrack::appendDumpHeader(result); 6252 for (size_t i = 0; i < numactive; ++i) { 6253 sp<RecordTrack> track = mActiveTracks[i]; 6254 if (mTracks.indexOf(track) < 0) { 6255 track->dump(buffer, SIZE, true); 6256 result.append(buffer); 6257 } 6258 } 6259 6260 } 6261 write(fd, result.string(), result.size()); 6262} 6263 6264 6265void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6266{ 6267 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6268 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6269 mRsmpInFront = recordThread->mRsmpInRear; 6270 mRsmpInUnrel = 0; 6271} 6272 6273void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6274 size_t *framesAvailable, bool *hasOverrun) 6275{ 6276 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6277 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6278 const int32_t rear = recordThread->mRsmpInRear; 6279 const int32_t front = mRsmpInFront; 6280 const ssize_t filled = rear - front; 6281 6282 size_t framesIn; 6283 bool overrun = false; 6284 if (filled < 0) { 6285 // should not happen, but treat like a massive overrun and re-sync 6286 framesIn = 0; 6287 mRsmpInFront = rear; 6288 overrun = true; 6289 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6290 framesIn = (size_t) filled; 6291 } else { 6292 // client is not keeping up with server, but give it latest data 6293 framesIn = recordThread->mRsmpInFrames; 6294 mRsmpInFront = /* front = */ rear - framesIn; 6295 overrun = true; 6296 } 6297 if (framesAvailable != NULL) { 6298 *framesAvailable = framesIn; 6299 } 6300 if (hasOverrun != NULL) { 6301 *hasOverrun = overrun; 6302 } 6303} 6304 6305// AudioBufferProvider interface 6306status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6307 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6308{ 6309 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6310 if (threadBase == 0) { 6311 buffer->frameCount = 0; 6312 buffer->raw = NULL; 6313 return NOT_ENOUGH_DATA; 6314 } 6315 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6316 int32_t rear = recordThread->mRsmpInRear; 6317 int32_t front = mRsmpInFront; 6318 ssize_t filled = rear - front; 6319 // FIXME should not be P2 (don't want to increase latency) 6320 // FIXME if client not keeping up, discard 6321 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6322 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6323 front &= recordThread->mRsmpInFramesP2 - 1; 6324 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6325 if (part1 > (size_t) filled) { 6326 part1 = filled; 6327 } 6328 size_t ask = buffer->frameCount; 6329 ALOG_ASSERT(ask > 0); 6330 if (part1 > ask) { 6331 part1 = ask; 6332 } 6333 if (part1 == 0) { 6334 // out of data is fine since the resampler will return a short-count. 6335 buffer->raw = NULL; 6336 buffer->frameCount = 0; 6337 mRsmpInUnrel = 0; 6338 return NOT_ENOUGH_DATA; 6339 } 6340 6341 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6342 buffer->frameCount = part1; 6343 mRsmpInUnrel = part1; 6344 return NO_ERROR; 6345} 6346 6347// AudioBufferProvider interface 6348void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6349 AudioBufferProvider::Buffer* buffer) 6350{ 6351 size_t stepCount = buffer->frameCount; 6352 if (stepCount == 0) { 6353 return; 6354 } 6355 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6356 mRsmpInUnrel -= stepCount; 6357 mRsmpInFront += stepCount; 6358 buffer->raw = NULL; 6359 buffer->frameCount = 0; 6360} 6361 6362AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6363 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6364 uint32_t srcSampleRate, 6365 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6366 uint32_t dstSampleRate) : 6367 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6368 // mSrcFormat 6369 // mSrcSampleRate 6370 // mDstChannelMask 6371 // mDstFormat 6372 // mDstSampleRate 6373 // mSrcChannelCount 6374 // mDstChannelCount 6375 // mDstFrameSize 6376 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6377 mResampler(NULL), 6378 mIsLegacyDownmix(false), 6379 mIsLegacyUpmix(false), 6380 mRequiresFloat(false), 6381 mInputConverterProvider(NULL) 6382{ 6383 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6384 dstChannelMask, dstFormat, dstSampleRate); 6385} 6386 6387AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6388 free(mBuf); 6389 delete mResampler; 6390 delete mInputConverterProvider; 6391} 6392 6393size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6394 AudioBufferProvider *provider, size_t frames) 6395{ 6396 if (mInputConverterProvider != NULL) { 6397 mInputConverterProvider->setBufferProvider(provider); 6398 provider = mInputConverterProvider; 6399 } 6400 6401 if (mResampler == NULL) { 6402 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6403 mSrcSampleRate, mSrcFormat, mDstFormat); 6404 6405 AudioBufferProvider::Buffer buffer; 6406 for (size_t i = frames; i > 0; ) { 6407 buffer.frameCount = i; 6408 status_t status = provider->getNextBuffer(&buffer, 0); 6409 if (status != OK || buffer.frameCount == 0) { 6410 frames -= i; // cannot fill request. 6411 break; 6412 } 6413 // format convert to destination buffer 6414 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6415 6416 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6417 i -= buffer.frameCount; 6418 provider->releaseBuffer(&buffer); 6419 } 6420 } else { 6421 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6422 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6423 6424 // reallocate buffer if needed 6425 if (mBufFrameSize != 0 && mBufFrames < frames) { 6426 free(mBuf); 6427 mBufFrames = frames; 6428 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6429 } 6430 // resampler accumulates, but we only have one source track 6431 memset(mBuf, 0, frames * mBufFrameSize); 6432 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6433 // format convert to destination buffer 6434 convertResampler(dst, mBuf, frames); 6435 } 6436 return frames; 6437} 6438 6439status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6440 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6441 uint32_t srcSampleRate, 6442 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6443 uint32_t dstSampleRate) 6444{ 6445 // quick evaluation if there is any change. 6446 if (mSrcFormat == srcFormat 6447 && mSrcChannelMask == srcChannelMask 6448 && mSrcSampleRate == srcSampleRate 6449 && mDstFormat == dstFormat 6450 && mDstChannelMask == dstChannelMask 6451 && mDstSampleRate == dstSampleRate) { 6452 return NO_ERROR; 6453 } 6454 6455 const bool valid = 6456 audio_is_input_channel(srcChannelMask) 6457 && audio_is_input_channel(dstChannelMask) 6458 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6459 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6460 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6461 ; // no upsampling checks for now 6462 if (!valid) { 6463 return BAD_VALUE; 6464 } 6465 6466 mSrcFormat = srcFormat; 6467 mSrcChannelMask = srcChannelMask; 6468 mSrcSampleRate = srcSampleRate; 6469 mDstFormat = dstFormat; 6470 mDstChannelMask = dstChannelMask; 6471 mDstSampleRate = dstSampleRate; 6472 6473 // compute derived parameters 6474 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6475 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6476 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6477 6478 // do we need to resample? 6479 delete mResampler; 6480 mResampler = NULL; 6481 if (mSrcSampleRate != mDstSampleRate) { 6482 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6483 mSrcChannelCount, mDstSampleRate); 6484 mResampler->setSampleRate(mSrcSampleRate); 6485 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6486 } 6487 6488 // are we running legacy channel conversion modes? 6489 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6490 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6491 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6492 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6493 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6494 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6495 6496 // do we need to process in float? 6497 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6498 6499 // do we need a staging buffer to convert for destination (we can still optimize this)? 6500 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6501 if (mResampler != NULL) { 6502 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6503 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6504 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6505 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6506 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6507 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6508 } else { 6509 mBufFrameSize = 0; 6510 } 6511 mBufFrames = 0; // force the buffer to be resized. 6512 6513 // do we need an input converter buffer provider to give us float? 6514 delete mInputConverterProvider; 6515 mInputConverterProvider = NULL; 6516 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6517 mInputConverterProvider = new ReformatBufferProvider( 6518 audio_channel_count_from_in_mask(mSrcChannelMask), 6519 mSrcFormat, 6520 AUDIO_FORMAT_PCM_FLOAT, 6521 256 /* provider buffer frame count */); 6522 } 6523 6524 // do we need a remixer to do channel mask conversion 6525 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6526 (void) memcpy_by_index_array_initialization_from_channel_mask( 6527 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6528 } 6529 return NO_ERROR; 6530} 6531 6532void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6533 void *dst, const void *src, size_t frames) 6534{ 6535 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6536 if (mBufFrameSize != 0 && mBufFrames < frames) { 6537 free(mBuf); 6538 mBufFrames = frames; 6539 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6540 } 6541 // do we need to do legacy upmix and downmix? 6542 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6543 void *dstBuf = mBuf != NULL ? mBuf : dst; 6544 if (mIsLegacyUpmix) { 6545 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6546 (const float *)src, frames); 6547 } else /*mIsLegacyDownmix */ { 6548 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6549 (const float *)src, frames); 6550 } 6551 if (mBuf != NULL) { 6552 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6553 frames * mDstChannelCount); 6554 } 6555 return; 6556 } 6557 // do we need to do channel mask conversion? 6558 if (mSrcChannelMask != mDstChannelMask) { 6559 void *dstBuf = mBuf != NULL ? mBuf : dst; 6560 memcpy_by_index_array(dstBuf, mDstChannelCount, 6561 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6562 if (dstBuf == dst) { 6563 return; // format is the same 6564 } 6565 } 6566 // convert to destination buffer 6567 const void *convertBuf = mBuf != NULL ? mBuf : src; 6568 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6569 frames * mDstChannelCount); 6570} 6571 6572void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6573 void *dst, /*not-a-const*/ void *src, size_t frames) 6574{ 6575 // src buffer format is ALWAYS float when entering this routine 6576 if (mIsLegacyUpmix) { 6577 ; // mono to stereo already handled by resampler 6578 } else if (mIsLegacyDownmix 6579 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6580 // the resampler outputs stereo for mono input channel (a feature?) 6581 // must convert to mono 6582 downmix_to_mono_float_from_stereo_float((float *)src, 6583 (const float *)src, frames); 6584 } else if (mSrcChannelMask != mDstChannelMask) { 6585 // convert to mono channel again for channel mask conversion (could be skipped 6586 // with further optimization). 6587 if (mSrcChannelCount == 1) { 6588 downmix_to_mono_float_from_stereo_float((float *)src, 6589 (const float *)src, frames); 6590 } 6591 // convert to destination format (in place, OK as float is larger than other types) 6592 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6593 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6594 frames * mSrcChannelCount); 6595 } 6596 // channel convert and save to dst 6597 memcpy_by_index_array(dst, mDstChannelCount, 6598 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6599 return; 6600 } 6601 // convert to destination format and save to dst 6602 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6603 frames * mDstChannelCount); 6604} 6605 6606bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6607 status_t& status) 6608{ 6609 bool reconfig = false; 6610 6611 status = NO_ERROR; 6612 6613 audio_format_t reqFormat = mFormat; 6614 uint32_t samplingRate = mSampleRate; 6615 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6616 // possible that we are > 2 channels, use channel index mask 6617 if (channelMask == AUDIO_CHANNEL_INVALID && mChannelCount <= FCC_8) { 6618 audio_channel_mask_for_index_assignment_from_count(mChannelCount); 6619 } 6620 6621 AudioParameter param = AudioParameter(keyValuePair); 6622 int value; 6623 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6624 // channel count change can be requested. Do we mandate the first client defines the 6625 // HAL sampling rate and channel count or do we allow changes on the fly? 6626 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6627 samplingRate = value; 6628 reconfig = true; 6629 } 6630 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6631 if (!audio_is_linear_pcm((audio_format_t) value)) { 6632 status = BAD_VALUE; 6633 } else { 6634 reqFormat = (audio_format_t) value; 6635 reconfig = true; 6636 } 6637 } 6638 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6639 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6640 if (!audio_is_input_channel(mask) || 6641 audio_channel_count_from_in_mask(mask) > FCC_8) { 6642 status = BAD_VALUE; 6643 } else { 6644 channelMask = mask; 6645 reconfig = true; 6646 } 6647 } 6648 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6649 // do not accept frame count changes if tracks are open as the track buffer 6650 // size depends on frame count and correct behavior would not be guaranteed 6651 // if frame count is changed after track creation 6652 if (mActiveTracks.size() > 0) { 6653 status = INVALID_OPERATION; 6654 } else { 6655 reconfig = true; 6656 } 6657 } 6658 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6659 // forward device change to effects that have requested to be 6660 // aware of attached audio device. 6661 for (size_t i = 0; i < mEffectChains.size(); i++) { 6662 mEffectChains[i]->setDevice_l(value); 6663 } 6664 6665 // store input device and output device but do not forward output device to audio HAL. 6666 // Note that status is ignored by the caller for output device 6667 // (see AudioFlinger::setParameters() 6668 if (audio_is_output_devices(value)) { 6669 mOutDevice = value; 6670 status = BAD_VALUE; 6671 } else { 6672 mInDevice = value; 6673 // disable AEC and NS if the device is a BT SCO headset supporting those 6674 // pre processings 6675 if (mTracks.size() > 0) { 6676 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6677 mAudioFlinger->btNrecIsOff(); 6678 for (size_t i = 0; i < mTracks.size(); i++) { 6679 sp<RecordTrack> track = mTracks[i]; 6680 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6681 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6682 } 6683 } 6684 } 6685 } 6686 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6687 mAudioSource != (audio_source_t)value) { 6688 // forward device change to effects that have requested to be 6689 // aware of attached audio device. 6690 for (size_t i = 0; i < mEffectChains.size(); i++) { 6691 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6692 } 6693 mAudioSource = (audio_source_t)value; 6694 } 6695 6696 if (status == NO_ERROR) { 6697 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6698 keyValuePair.string()); 6699 if (status == INVALID_OPERATION) { 6700 inputStandBy(); 6701 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6702 keyValuePair.string()); 6703 } 6704 if (reconfig) { 6705 if (status == BAD_VALUE && 6706 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6707 audio_is_linear_pcm(reqFormat) && 6708 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6709 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6710 audio_channel_count_from_in_mask( 6711 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6712 status = NO_ERROR; 6713 } 6714 if (status == NO_ERROR) { 6715 readInputParameters_l(); 6716 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6717 } 6718 } 6719 } 6720 6721 return reconfig; 6722} 6723 6724String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6725{ 6726 Mutex::Autolock _l(mLock); 6727 if (initCheck() != NO_ERROR) { 6728 return String8(); 6729 } 6730 6731 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6732 const String8 out_s8(s); 6733 free(s); 6734 return out_s8; 6735} 6736 6737void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event) { 6738 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6739 6740 desc->mIoHandle = mId; 6741 6742 switch (event) { 6743 case AUDIO_INPUT_OPENED: 6744 case AUDIO_INPUT_CONFIG_CHANGED: 6745 desc->mPatch = mPatch; 6746 desc->mChannelMask = mChannelMask; 6747 desc->mSamplingRate = mSampleRate; 6748 desc->mFormat = mFormat; 6749 desc->mFrameCount = mFrameCount; 6750 desc->mLatency = 0; 6751 break; 6752 6753 case AUDIO_INPUT_CLOSED: 6754 default: 6755 break; 6756 } 6757 mAudioFlinger->ioConfigChanged(event, desc); 6758} 6759 6760void AudioFlinger::RecordThread::readInputParameters_l() 6761{ 6762 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6763 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6764 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6765 if (mChannelCount > FCC_8) { 6766 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6767 } 6768 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6769 mFormat = mHALFormat; 6770 if (!audio_is_linear_pcm(mFormat)) { 6771 ALOGE("HAL format %#x is not linear pcm", mFormat); 6772 } 6773 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6774 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6775 mFrameCount = mBufferSize / mFrameSize; 6776 // This is the formula for calculating the temporary buffer size. 6777 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6778 // 1 full output buffer, regardless of the alignment of the available input. 6779 // The value is somewhat arbitrary, and could probably be even larger. 6780 // A larger value should allow more old data to be read after a track calls start(), 6781 // without increasing latency. 6782 // 6783 // Note this is independent of the maximum downsampling ratio permitted for capture. 6784 mRsmpInFrames = mFrameCount * 7; 6785 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6786 free(mRsmpInBuffer); 6787 6788 // TODO optimize audio capture buffer sizes ... 6789 // Here we calculate the size of the sliding buffer used as a source 6790 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6791 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6792 // be better to have it derived from the pipe depth in the long term. 6793 // The current value is higher than necessary. However it should not add to latency. 6794 6795 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6796 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6797 6798 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6799 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6800} 6801 6802uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6803{ 6804 Mutex::Autolock _l(mLock); 6805 if (initCheck() != NO_ERROR) { 6806 return 0; 6807 } 6808 6809 return mInput->stream->get_input_frames_lost(mInput->stream); 6810} 6811 6812uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6813{ 6814 Mutex::Autolock _l(mLock); 6815 uint32_t result = 0; 6816 if (getEffectChain_l(sessionId) != 0) { 6817 result = EFFECT_SESSION; 6818 } 6819 6820 for (size_t i = 0; i < mTracks.size(); ++i) { 6821 if (sessionId == mTracks[i]->sessionId()) { 6822 result |= TRACK_SESSION; 6823 break; 6824 } 6825 } 6826 6827 return result; 6828} 6829 6830KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6831{ 6832 KeyedVector<int, bool> ids; 6833 Mutex::Autolock _l(mLock); 6834 for (size_t j = 0; j < mTracks.size(); ++j) { 6835 sp<RecordThread::RecordTrack> track = mTracks[j]; 6836 int sessionId = track->sessionId(); 6837 if (ids.indexOfKey(sessionId) < 0) { 6838 ids.add(sessionId, true); 6839 } 6840 } 6841 return ids; 6842} 6843 6844AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6845{ 6846 Mutex::Autolock _l(mLock); 6847 AudioStreamIn *input = mInput; 6848 mInput = NULL; 6849 return input; 6850} 6851 6852// this method must always be called either with ThreadBase mLock held or inside the thread loop 6853audio_stream_t* AudioFlinger::RecordThread::stream() const 6854{ 6855 if (mInput == NULL) { 6856 return NULL; 6857 } 6858 return &mInput->stream->common; 6859} 6860 6861status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6862{ 6863 // only one chain per input thread 6864 if (mEffectChains.size() != 0) { 6865 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6866 return INVALID_OPERATION; 6867 } 6868 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6869 chain->setThread(this); 6870 chain->setInBuffer(NULL); 6871 chain->setOutBuffer(NULL); 6872 6873 checkSuspendOnAddEffectChain_l(chain); 6874 6875 // make sure enabled pre processing effects state is communicated to the HAL as we 6876 // just moved them to a new input stream. 6877 chain->syncHalEffectsState(); 6878 6879 mEffectChains.add(chain); 6880 6881 return NO_ERROR; 6882} 6883 6884size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6885{ 6886 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6887 ALOGW_IF(mEffectChains.size() != 1, 6888 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6889 chain.get(), mEffectChains.size(), this); 6890 if (mEffectChains.size() == 1) { 6891 mEffectChains.removeAt(0); 6892 } 6893 return 0; 6894} 6895 6896status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6897 audio_patch_handle_t *handle) 6898{ 6899 status_t status = NO_ERROR; 6900 6901 // store new device and send to effects 6902 mInDevice = patch->sources[0].ext.device.type; 6903 mPatch = *patch; 6904 for (size_t i = 0; i < mEffectChains.size(); i++) { 6905 mEffectChains[i]->setDevice_l(mInDevice); 6906 } 6907 6908 // disable AEC and NS if the device is a BT SCO headset supporting those 6909 // pre processings 6910 if (mTracks.size() > 0) { 6911 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6912 mAudioFlinger->btNrecIsOff(); 6913 for (size_t i = 0; i < mTracks.size(); i++) { 6914 sp<RecordTrack> track = mTracks[i]; 6915 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6916 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6917 } 6918 } 6919 6920 // store new source and send to effects 6921 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6922 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6923 for (size_t i = 0; i < mEffectChains.size(); i++) { 6924 mEffectChains[i]->setAudioSource_l(mAudioSource); 6925 } 6926 } 6927 6928 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6929 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6930 status = hwDevice->create_audio_patch(hwDevice, 6931 patch->num_sources, 6932 patch->sources, 6933 patch->num_sinks, 6934 patch->sinks, 6935 handle); 6936 } else { 6937 char *address; 6938 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 6939 address = audio_device_address_to_parameter( 6940 patch->sources[0].ext.device.type, 6941 patch->sources[0].ext.device.address); 6942 } else { 6943 address = (char *)calloc(1, 1); 6944 } 6945 AudioParameter param = AudioParameter(String8(address)); 6946 free(address); 6947 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 6948 (int)patch->sources[0].ext.device.type); 6949 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 6950 (int)patch->sinks[0].ext.mix.usecase.source); 6951 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6952 param.toString().string()); 6953 *handle = AUDIO_PATCH_HANDLE_NONE; 6954 } 6955 6956 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6957 6958 return status; 6959} 6960 6961status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6962{ 6963 status_t status = NO_ERROR; 6964 6965 mInDevice = AUDIO_DEVICE_NONE; 6966 6967 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6968 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6969 status = hwDevice->release_audio_patch(hwDevice, handle); 6970 } else { 6971 AudioParameter param; 6972 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 6973 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6974 param.toString().string()); 6975 } 6976 return status; 6977} 6978 6979void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6980{ 6981 Mutex::Autolock _l(mLock); 6982 mTracks.add(record); 6983} 6984 6985void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6986{ 6987 Mutex::Autolock _l(mLock); 6988 destroyTrack_l(record); 6989} 6990 6991void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6992{ 6993 ThreadBase::getAudioPortConfig(config); 6994 config->role = AUDIO_PORT_ROLE_SINK; 6995 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6996 config->ext.mix.usecase.source = mAudioSource; 6997} 6998 6999} // namespace android 7000