Threads.cpp revision 73e26b661af50be2c0a4ff6c9ac85f7347a8b235
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} 504 505AudioFlinger::ThreadBase::~ThreadBase() 506{ 507 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 508 mConfigEvents.clear(); 509 510 // do not lock the mutex in destructor 511 releaseWakeLock_l(); 512 if (mPowerManager != 0) { 513 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 514 binder->unlinkToDeath(mDeathRecipient); 515 } 516} 517 518status_t AudioFlinger::ThreadBase::readyToRun() 519{ 520 status_t status = initCheck(); 521 if (status == NO_ERROR) { 522 ALOGI("AudioFlinger's thread %p ready to run", this); 523 } else { 524 ALOGE("No working audio driver found."); 525 } 526 return status; 527} 528 529void AudioFlinger::ThreadBase::exit() 530{ 531 ALOGV("ThreadBase::exit"); 532 // do any cleanup required for exit to succeed 533 preExit(); 534 { 535 // This lock prevents the following race in thread (uniprocessor for illustration): 536 // if (!exitPending()) { 537 // // context switch from here to exit() 538 // // exit() calls requestExit(), what exitPending() observes 539 // // exit() calls signal(), which is dropped since no waiters 540 // // context switch back from exit() to here 541 // mWaitWorkCV.wait(...); 542 // // now thread is hung 543 // } 544 AutoMutex lock(mLock); 545 requestExit(); 546 mWaitWorkCV.broadcast(); 547 } 548 // When Thread::requestExitAndWait is made virtual and this method is renamed to 549 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 550 requestExitAndWait(); 551} 552 553status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 554{ 555 status_t status; 556 557 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 558 Mutex::Autolock _l(mLock); 559 560 return sendSetParameterConfigEvent_l(keyValuePairs); 561} 562 563// sendConfigEvent_l() must be called with ThreadBase::mLock held 564// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 565status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 566{ 567 status_t status = NO_ERROR; 568 569 mConfigEvents.add(event); 570 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 571 mWaitWorkCV.signal(); 572 mLock.unlock(); 573 { 574 Mutex::Autolock _l(event->mLock); 575 while (event->mWaitStatus) { 576 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 577 event->mStatus = TIMED_OUT; 578 event->mWaitStatus = false; 579 } 580 } 581 status = event->mStatus; 582 } 583 mLock.lock(); 584 return status; 585} 586 587void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event) 588{ 589 Mutex::Autolock _l(mLock); 590 sendIoConfigEvent_l(event); 591} 592 593// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 594void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event) 595{ 596 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event); 597 sendConfigEvent_l(configEvent); 598} 599 600// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 601void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 602{ 603 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 604 sendConfigEvent_l(configEvent); 605} 606 607// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 608status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 609{ 610 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 611 return sendConfigEvent_l(configEvent); 612} 613 614status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 615 const struct audio_patch *patch, 616 audio_patch_handle_t *handle) 617{ 618 Mutex::Autolock _l(mLock); 619 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 620 status_t status = sendConfigEvent_l(configEvent); 621 if (status == NO_ERROR) { 622 CreateAudioPatchConfigEventData *data = 623 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 624 *handle = data->mHandle; 625 } 626 return status; 627} 628 629status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 630 const audio_patch_handle_t handle) 631{ 632 Mutex::Autolock _l(mLock); 633 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 634 return sendConfigEvent_l(configEvent); 635} 636 637 638// post condition: mConfigEvents.isEmpty() 639void AudioFlinger::ThreadBase::processConfigEvents_l() 640{ 641 bool configChanged = false; 642 643 while (!mConfigEvents.isEmpty()) { 644 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 645 sp<ConfigEvent> event = mConfigEvents[0]; 646 mConfigEvents.removeAt(0); 647 switch (event->mType) { 648 case CFG_EVENT_PRIO: { 649 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 650 // FIXME Need to understand why this has to be done asynchronously 651 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 652 true /*asynchronous*/); 653 if (err != 0) { 654 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 655 data->mPrio, data->mPid, data->mTid, err); 656 } 657 } break; 658 case CFG_EVENT_IO: { 659 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 660 ioConfigChanged(data->mEvent); 661 } break; 662 case CFG_EVENT_SET_PARAMETER: { 663 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 664 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 665 configChanged = true; 666 } 667 } break; 668 case CFG_EVENT_CREATE_AUDIO_PATCH: { 669 CreateAudioPatchConfigEventData *data = 670 (CreateAudioPatchConfigEventData *)event->mData.get(); 671 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 672 } break; 673 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 674 ReleaseAudioPatchConfigEventData *data = 675 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 676 event->mStatus = releaseAudioPatch_l(data->mHandle); 677 } break; 678 default: 679 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 680 break; 681 } 682 { 683 Mutex::Autolock _l(event->mLock); 684 if (event->mWaitStatus) { 685 event->mWaitStatus = false; 686 event->mCond.signal(); 687 } 688 } 689 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 690 } 691 692 if (configChanged) { 693 cacheParameters_l(); 694 } 695} 696 697String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 698 String8 s; 699 if (output) { 700 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 701 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 702 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 703 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 704 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 705 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 706 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 707 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 708 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 709 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 710 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 711 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 712 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 713 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 714 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 715 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 716 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 717 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 718 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 719 } else { 720 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 721 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 722 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 723 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 724 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 725 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 726 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 727 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 728 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 729 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 730 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 731 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 732 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 733 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 734 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 735 } 736 int len = s.length(); 737 if (s.length() > 2) { 738 char *str = s.lockBuffer(len); 739 s.unlockBuffer(len - 2); 740 } 741 return s; 742} 743 744void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 745{ 746 const size_t SIZE = 256; 747 char buffer[SIZE]; 748 String8 result; 749 750 bool locked = AudioFlinger::dumpTryLock(mLock); 751 if (!locked) { 752 dprintf(fd, "thread %p may be deadlocked\n", this); 753 } 754 755 dprintf(fd, " Thread name: %s\n", mThreadName); 756 dprintf(fd, " I/O handle: %d\n", mId); 757 dprintf(fd, " TID: %d\n", getTid()); 758 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 759 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 760 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 761 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 762 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 763 dprintf(fd, " Channel count: %u\n", mChannelCount); 764 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 765 channelMaskToString(mChannelMask, mType != RECORD).string()); 766 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 767 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 768 dprintf(fd, " Pending config events:"); 769 size_t numConfig = mConfigEvents.size(); 770 if (numConfig) { 771 for (size_t i = 0; i < numConfig; i++) { 772 mConfigEvents[i]->dump(buffer, SIZE); 773 dprintf(fd, "\n %s", buffer); 774 } 775 dprintf(fd, "\n"); 776 } else { 777 dprintf(fd, " none\n"); 778 } 779 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 780 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 781 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 782 783 if (locked) { 784 mLock.unlock(); 785 } 786} 787 788void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 789{ 790 const size_t SIZE = 256; 791 char buffer[SIZE]; 792 String8 result; 793 794 size_t numEffectChains = mEffectChains.size(); 795 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 796 write(fd, buffer, strlen(buffer)); 797 798 for (size_t i = 0; i < numEffectChains; ++i) { 799 sp<EffectChain> chain = mEffectChains[i]; 800 if (chain != 0) { 801 chain->dump(fd, args); 802 } 803 } 804} 805 806void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 807{ 808 Mutex::Autolock _l(mLock); 809 acquireWakeLock_l(uid); 810} 811 812String16 AudioFlinger::ThreadBase::getWakeLockTag() 813{ 814 switch (mType) { 815 case MIXER: 816 return String16("AudioMix"); 817 case DIRECT: 818 return String16("AudioDirectOut"); 819 case DUPLICATING: 820 return String16("AudioDup"); 821 case RECORD: 822 return String16("AudioIn"); 823 case OFFLOAD: 824 return String16("AudioOffload"); 825 default: 826 ALOG_ASSERT(false); 827 return String16("AudioUnknown"); 828 } 829} 830 831void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 832{ 833 getPowerManager_l(); 834 if (mPowerManager != 0) { 835 sp<IBinder> binder = new BBinder(); 836 status_t status; 837 if (uid >= 0) { 838 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 839 binder, 840 getWakeLockTag(), 841 String16("media"), 842 uid, 843 true /* FIXME force oneway contrary to .aidl */); 844 } else { 845 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 846 binder, 847 getWakeLockTag(), 848 String16("media"), 849 true /* FIXME force oneway contrary to .aidl */); 850 } 851 if (status == NO_ERROR) { 852 mWakeLockToken = binder; 853 } 854 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 855 } 856} 857 858void AudioFlinger::ThreadBase::releaseWakeLock() 859{ 860 Mutex::Autolock _l(mLock); 861 releaseWakeLock_l(); 862} 863 864void AudioFlinger::ThreadBase::releaseWakeLock_l() 865{ 866 if (mWakeLockToken != 0) { 867 ALOGV("releaseWakeLock_l() %s", mThreadName); 868 if (mPowerManager != 0) { 869 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 870 true /* FIXME force oneway contrary to .aidl */); 871 } 872 mWakeLockToken.clear(); 873 } 874} 875 876void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 877 Mutex::Autolock _l(mLock); 878 updateWakeLockUids_l(uids); 879} 880 881void AudioFlinger::ThreadBase::getPowerManager_l() { 882 883 if (mPowerManager == 0) { 884 // use checkService() to avoid blocking if power service is not up yet 885 sp<IBinder> binder = 886 defaultServiceManager()->checkService(String16("power")); 887 if (binder == 0) { 888 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 889 } else { 890 mPowerManager = interface_cast<IPowerManager>(binder); 891 binder->linkToDeath(mDeathRecipient); 892 } 893 } 894} 895 896void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 897 898 getPowerManager_l(); 899 if (mWakeLockToken == NULL) { 900 ALOGE("no wake lock to update!"); 901 return; 902 } 903 if (mPowerManager != 0) { 904 sp<IBinder> binder = new BBinder(); 905 status_t status; 906 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 907 true /* FIXME force oneway contrary to .aidl */); 908 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 909 } 910} 911 912void AudioFlinger::ThreadBase::clearPowerManager() 913{ 914 Mutex::Autolock _l(mLock); 915 releaseWakeLock_l(); 916 mPowerManager.clear(); 917} 918 919void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 920{ 921 sp<ThreadBase> thread = mThread.promote(); 922 if (thread != 0) { 923 thread->clearPowerManager(); 924 } 925 ALOGW("power manager service died !!!"); 926} 927 928void AudioFlinger::ThreadBase::setEffectSuspended( 929 const effect_uuid_t *type, bool suspend, int sessionId) 930{ 931 Mutex::Autolock _l(mLock); 932 setEffectSuspended_l(type, suspend, sessionId); 933} 934 935void AudioFlinger::ThreadBase::setEffectSuspended_l( 936 const effect_uuid_t *type, bool suspend, int sessionId) 937{ 938 sp<EffectChain> chain = getEffectChain_l(sessionId); 939 if (chain != 0) { 940 if (type != NULL) { 941 chain->setEffectSuspended_l(type, suspend); 942 } else { 943 chain->setEffectSuspendedAll_l(suspend); 944 } 945 } 946 947 updateSuspendedSessions_l(type, suspend, sessionId); 948} 949 950void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 951{ 952 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 953 if (index < 0) { 954 return; 955 } 956 957 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 958 mSuspendedSessions.valueAt(index); 959 960 for (size_t i = 0; i < sessionEffects.size(); i++) { 961 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 962 for (int j = 0; j < desc->mRefCount; j++) { 963 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 964 chain->setEffectSuspendedAll_l(true); 965 } else { 966 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 967 desc->mType.timeLow); 968 chain->setEffectSuspended_l(&desc->mType, true); 969 } 970 } 971 } 972} 973 974void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 975 bool suspend, 976 int sessionId) 977{ 978 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 979 980 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 981 982 if (suspend) { 983 if (index >= 0) { 984 sessionEffects = mSuspendedSessions.valueAt(index); 985 } else { 986 mSuspendedSessions.add(sessionId, sessionEffects); 987 } 988 } else { 989 if (index < 0) { 990 return; 991 } 992 sessionEffects = mSuspendedSessions.valueAt(index); 993 } 994 995 996 int key = EffectChain::kKeyForSuspendAll; 997 if (type != NULL) { 998 key = type->timeLow; 999 } 1000 index = sessionEffects.indexOfKey(key); 1001 1002 sp<SuspendedSessionDesc> desc; 1003 if (suspend) { 1004 if (index >= 0) { 1005 desc = sessionEffects.valueAt(index); 1006 } else { 1007 desc = new SuspendedSessionDesc(); 1008 if (type != NULL) { 1009 desc->mType = *type; 1010 } 1011 sessionEffects.add(key, desc); 1012 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1013 } 1014 desc->mRefCount++; 1015 } else { 1016 if (index < 0) { 1017 return; 1018 } 1019 desc = sessionEffects.valueAt(index); 1020 if (--desc->mRefCount == 0) { 1021 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1022 sessionEffects.removeItemsAt(index); 1023 if (sessionEffects.isEmpty()) { 1024 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1025 sessionId); 1026 mSuspendedSessions.removeItem(sessionId); 1027 } 1028 } 1029 } 1030 if (!sessionEffects.isEmpty()) { 1031 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1032 } 1033} 1034 1035void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1036 bool enabled, 1037 int sessionId) 1038{ 1039 Mutex::Autolock _l(mLock); 1040 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1041} 1042 1043void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1044 bool enabled, 1045 int sessionId) 1046{ 1047 if (mType != RECORD) { 1048 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1049 // another session. This gives the priority to well behaved effect control panels 1050 // and applications not using global effects. 1051 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1052 // global effects 1053 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1054 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1055 } 1056 } 1057 1058 sp<EffectChain> chain = getEffectChain_l(sessionId); 1059 if (chain != 0) { 1060 chain->checkSuspendOnEffectEnabled(effect, enabled); 1061 } 1062} 1063 1064// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1065sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1066 const sp<AudioFlinger::Client>& client, 1067 const sp<IEffectClient>& effectClient, 1068 int32_t priority, 1069 int sessionId, 1070 effect_descriptor_t *desc, 1071 int *enabled, 1072 status_t *status) 1073{ 1074 sp<EffectModule> effect; 1075 sp<EffectHandle> handle; 1076 status_t lStatus; 1077 sp<EffectChain> chain; 1078 bool chainCreated = false; 1079 bool effectCreated = false; 1080 bool effectRegistered = false; 1081 1082 lStatus = initCheck(); 1083 if (lStatus != NO_ERROR) { 1084 ALOGW("createEffect_l() Audio driver not initialized."); 1085 goto Exit; 1086 } 1087 1088 // Reject any effect on Direct output threads for now, since the format of 1089 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1090 if (mType == DIRECT) { 1091 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1092 desc->name, mThreadName); 1093 lStatus = BAD_VALUE; 1094 goto Exit; 1095 } 1096 1097 // Reject any effect on mixer or duplicating multichannel sinks. 1098 // TODO: fix both format and multichannel issues with effects. 1099 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1100 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1101 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1102 lStatus = BAD_VALUE; 1103 goto Exit; 1104 } 1105 1106 // Allow global effects only on offloaded and mixer threads 1107 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1108 switch (mType) { 1109 case MIXER: 1110 case OFFLOAD: 1111 break; 1112 case DIRECT: 1113 case DUPLICATING: 1114 case RECORD: 1115 default: 1116 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1117 desc->name, mThreadName); 1118 lStatus = BAD_VALUE; 1119 goto Exit; 1120 } 1121 } 1122 1123 // Only Pre processor effects are allowed on input threads and only on input threads 1124 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1125 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1126 desc->name, desc->flags, mType); 1127 lStatus = BAD_VALUE; 1128 goto Exit; 1129 } 1130 1131 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1132 1133 { // scope for mLock 1134 Mutex::Autolock _l(mLock); 1135 1136 // check for existing effect chain with the requested audio session 1137 chain = getEffectChain_l(sessionId); 1138 if (chain == 0) { 1139 // create a new chain for this session 1140 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1141 chain = new EffectChain(this, sessionId); 1142 addEffectChain_l(chain); 1143 chain->setStrategy(getStrategyForSession_l(sessionId)); 1144 chainCreated = true; 1145 } else { 1146 effect = chain->getEffectFromDesc_l(desc); 1147 } 1148 1149 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1150 1151 if (effect == 0) { 1152 int id = mAudioFlinger->nextUniqueId(); 1153 // Check CPU and memory usage 1154 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1155 if (lStatus != NO_ERROR) { 1156 goto Exit; 1157 } 1158 effectRegistered = true; 1159 // create a new effect module if none present in the chain 1160 effect = new EffectModule(this, chain, desc, id, sessionId); 1161 lStatus = effect->status(); 1162 if (lStatus != NO_ERROR) { 1163 goto Exit; 1164 } 1165 effect->setOffloaded(mType == OFFLOAD, mId); 1166 1167 lStatus = chain->addEffect_l(effect); 1168 if (lStatus != NO_ERROR) { 1169 goto Exit; 1170 } 1171 effectCreated = true; 1172 1173 effect->setDevice(mOutDevice); 1174 effect->setDevice(mInDevice); 1175 effect->setMode(mAudioFlinger->getMode()); 1176 effect->setAudioSource(mAudioSource); 1177 } 1178 // create effect handle and connect it to effect module 1179 handle = new EffectHandle(effect, client, effectClient, priority); 1180 lStatus = handle->initCheck(); 1181 if (lStatus == OK) { 1182 lStatus = effect->addHandle(handle.get()); 1183 } 1184 if (enabled != NULL) { 1185 *enabled = (int)effect->isEnabled(); 1186 } 1187 } 1188 1189Exit: 1190 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1191 Mutex::Autolock _l(mLock); 1192 if (effectCreated) { 1193 chain->removeEffect_l(effect); 1194 } 1195 if (effectRegistered) { 1196 AudioSystem::unregisterEffect(effect->id()); 1197 } 1198 if (chainCreated) { 1199 removeEffectChain_l(chain); 1200 } 1201 handle.clear(); 1202 } 1203 1204 *status = lStatus; 1205 return handle; 1206} 1207 1208sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1209{ 1210 Mutex::Autolock _l(mLock); 1211 return getEffect_l(sessionId, effectId); 1212} 1213 1214sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1215{ 1216 sp<EffectChain> chain = getEffectChain_l(sessionId); 1217 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1218} 1219 1220// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1221// PlaybackThread::mLock held 1222status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1223{ 1224 // check for existing effect chain with the requested audio session 1225 int sessionId = effect->sessionId(); 1226 sp<EffectChain> chain = getEffectChain_l(sessionId); 1227 bool chainCreated = false; 1228 1229 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1230 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1231 this, effect->desc().name, effect->desc().flags); 1232 1233 if (chain == 0) { 1234 // create a new chain for this session 1235 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1236 chain = new EffectChain(this, sessionId); 1237 addEffectChain_l(chain); 1238 chain->setStrategy(getStrategyForSession_l(sessionId)); 1239 chainCreated = true; 1240 } 1241 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1242 1243 if (chain->getEffectFromId_l(effect->id()) != 0) { 1244 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1245 this, effect->desc().name, chain.get()); 1246 return BAD_VALUE; 1247 } 1248 1249 effect->setOffloaded(mType == OFFLOAD, mId); 1250 1251 status_t status = chain->addEffect_l(effect); 1252 if (status != NO_ERROR) { 1253 if (chainCreated) { 1254 removeEffectChain_l(chain); 1255 } 1256 return status; 1257 } 1258 1259 effect->setDevice(mOutDevice); 1260 effect->setDevice(mInDevice); 1261 effect->setMode(mAudioFlinger->getMode()); 1262 effect->setAudioSource(mAudioSource); 1263 return NO_ERROR; 1264} 1265 1266void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1267 1268 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1269 effect_descriptor_t desc = effect->desc(); 1270 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1271 detachAuxEffect_l(effect->id()); 1272 } 1273 1274 sp<EffectChain> chain = effect->chain().promote(); 1275 if (chain != 0) { 1276 // remove effect chain if removing last effect 1277 if (chain->removeEffect_l(effect) == 0) { 1278 removeEffectChain_l(chain); 1279 } 1280 } else { 1281 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1282 } 1283} 1284 1285void AudioFlinger::ThreadBase::lockEffectChains_l( 1286 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1287{ 1288 effectChains = mEffectChains; 1289 for (size_t i = 0; i < mEffectChains.size(); i++) { 1290 mEffectChains[i]->lock(); 1291 } 1292} 1293 1294void AudioFlinger::ThreadBase::unlockEffectChains( 1295 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1296{ 1297 for (size_t i = 0; i < effectChains.size(); i++) { 1298 effectChains[i]->unlock(); 1299 } 1300} 1301 1302sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1303{ 1304 Mutex::Autolock _l(mLock); 1305 return getEffectChain_l(sessionId); 1306} 1307 1308sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1309{ 1310 size_t size = mEffectChains.size(); 1311 for (size_t i = 0; i < size; i++) { 1312 if (mEffectChains[i]->sessionId() == sessionId) { 1313 return mEffectChains[i]; 1314 } 1315 } 1316 return 0; 1317} 1318 1319void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1320{ 1321 Mutex::Autolock _l(mLock); 1322 size_t size = mEffectChains.size(); 1323 for (size_t i = 0; i < size; i++) { 1324 mEffectChains[i]->setMode_l(mode); 1325 } 1326} 1327 1328void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1329{ 1330 config->type = AUDIO_PORT_TYPE_MIX; 1331 config->ext.mix.handle = mId; 1332 config->sample_rate = mSampleRate; 1333 config->format = mFormat; 1334 config->channel_mask = mChannelMask; 1335 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1336 AUDIO_PORT_CONFIG_FORMAT; 1337} 1338 1339 1340// ---------------------------------------------------------------------------- 1341// Playback 1342// ---------------------------------------------------------------------------- 1343 1344AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1345 AudioStreamOut* output, 1346 audio_io_handle_t id, 1347 audio_devices_t device, 1348 type_t type) 1349 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1350 mNormalFrameCount(0), mSinkBuffer(NULL), 1351 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1352 mMixerBuffer(NULL), 1353 mMixerBufferSize(0), 1354 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1355 mMixerBufferValid(false), 1356 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1357 mEffectBuffer(NULL), 1358 mEffectBufferSize(0), 1359 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1360 mEffectBufferValid(false), 1361 mSuspended(0), mBytesWritten(0), 1362 mActiveTracksGeneration(0), 1363 // mStreamTypes[] initialized in constructor body 1364 mOutput(output), 1365 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1366 mMixerStatus(MIXER_IDLE), 1367 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1368 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1369 mBytesRemaining(0), 1370 mCurrentWriteLength(0), 1371 mUseAsyncWrite(false), 1372 mWriteAckSequence(0), 1373 mDrainSequence(0), 1374 mSignalPending(false), 1375 mScreenState(AudioFlinger::mScreenState), 1376 // index 0 is reserved for normal mixer's submix 1377 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1378 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1379 // mLatchD, mLatchQ, 1380 mLatchDValid(false), mLatchQValid(false) 1381{ 1382 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1383 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1384 1385 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1386 // it would be safer to explicitly pass initial masterVolume/masterMute as 1387 // parameter. 1388 // 1389 // If the HAL we are using has support for master volume or master mute, 1390 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1391 // and the mute set to false). 1392 mMasterVolume = audioFlinger->masterVolume_l(); 1393 mMasterMute = audioFlinger->masterMute_l(); 1394 if (mOutput && mOutput->audioHwDev) { 1395 if (mOutput->audioHwDev->canSetMasterVolume()) { 1396 mMasterVolume = 1.0; 1397 } 1398 1399 if (mOutput->audioHwDev->canSetMasterMute()) { 1400 mMasterMute = false; 1401 } 1402 } 1403 1404 readOutputParameters_l(); 1405 1406 // ++ operator does not compile 1407 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1408 stream = (audio_stream_type_t) (stream + 1)) { 1409 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1410 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1411 } 1412} 1413 1414AudioFlinger::PlaybackThread::~PlaybackThread() 1415{ 1416 mAudioFlinger->unregisterWriter(mNBLogWriter); 1417 free(mSinkBuffer); 1418 free(mMixerBuffer); 1419 free(mEffectBuffer); 1420} 1421 1422void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1423{ 1424 dumpInternals(fd, args); 1425 dumpTracks(fd, args); 1426 dumpEffectChains(fd, args); 1427} 1428 1429void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1430{ 1431 const size_t SIZE = 256; 1432 char buffer[SIZE]; 1433 String8 result; 1434 1435 result.appendFormat(" Stream volumes in dB: "); 1436 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1437 const stream_type_t *st = &mStreamTypes[i]; 1438 if (i > 0) { 1439 result.appendFormat(", "); 1440 } 1441 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1442 if (st->mute) { 1443 result.append("M"); 1444 } 1445 } 1446 result.append("\n"); 1447 write(fd, result.string(), result.length()); 1448 result.clear(); 1449 1450 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1451 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1452 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1453 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1454 1455 size_t numtracks = mTracks.size(); 1456 size_t numactive = mActiveTracks.size(); 1457 dprintf(fd, " %d Tracks", numtracks); 1458 size_t numactiveseen = 0; 1459 if (numtracks) { 1460 dprintf(fd, " of which %d are active\n", numactive); 1461 Track::appendDumpHeader(result); 1462 for (size_t i = 0; i < numtracks; ++i) { 1463 sp<Track> track = mTracks[i]; 1464 if (track != 0) { 1465 bool active = mActiveTracks.indexOf(track) >= 0; 1466 if (active) { 1467 numactiveseen++; 1468 } 1469 track->dump(buffer, SIZE, active); 1470 result.append(buffer); 1471 } 1472 } 1473 } else { 1474 result.append("\n"); 1475 } 1476 if (numactiveseen != numactive) { 1477 // some tracks in the active list were not in the tracks list 1478 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1479 " not in the track list\n"); 1480 result.append(buffer); 1481 Track::appendDumpHeader(result); 1482 for (size_t i = 0; i < numactive; ++i) { 1483 sp<Track> track = mActiveTracks[i].promote(); 1484 if (track != 0 && mTracks.indexOf(track) < 0) { 1485 track->dump(buffer, SIZE, true); 1486 result.append(buffer); 1487 } 1488 } 1489 } 1490 1491 write(fd, result.string(), result.size()); 1492} 1493 1494void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1495{ 1496 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1497 1498 dumpBase(fd, args); 1499 1500 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1501 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1502 dprintf(fd, " Total writes: %d\n", mNumWrites); 1503 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1504 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1505 dprintf(fd, " Suspend count: %d\n", mSuspended); 1506 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1507 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1508 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1509 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1510 AudioStreamOut *output = mOutput; 1511 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1512 String8 flagsAsString = outputFlagsToString(flags); 1513 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1514} 1515 1516// Thread virtuals 1517 1518void AudioFlinger::PlaybackThread::onFirstRef() 1519{ 1520 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1521} 1522 1523// ThreadBase virtuals 1524void AudioFlinger::PlaybackThread::preExit() 1525{ 1526 ALOGV(" preExit()"); 1527 // FIXME this is using hard-coded strings but in the future, this functionality will be 1528 // converted to use audio HAL extensions required to support tunneling 1529 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1530} 1531 1532// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1533sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1534 const sp<AudioFlinger::Client>& client, 1535 audio_stream_type_t streamType, 1536 uint32_t sampleRate, 1537 audio_format_t format, 1538 audio_channel_mask_t channelMask, 1539 size_t *pFrameCount, 1540 const sp<IMemory>& sharedBuffer, 1541 int sessionId, 1542 IAudioFlinger::track_flags_t *flags, 1543 pid_t tid, 1544 int uid, 1545 status_t *status) 1546{ 1547 size_t frameCount = *pFrameCount; 1548 sp<Track> track; 1549 status_t lStatus; 1550 1551 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1552 1553 // client expresses a preference for FAST, but we get the final say 1554 if (*flags & IAudioFlinger::TRACK_FAST) { 1555 if ( 1556 // not timed 1557 (!isTimed) && 1558 // either of these use cases: 1559 ( 1560 // use case 1: shared buffer with any frame count 1561 ( 1562 (sharedBuffer != 0) 1563 ) || 1564 // use case 2: frame count is default or at least as large as HAL 1565 ( 1566 // we formerly checked for a callback handler (non-0 tid), 1567 // but that is no longer required for TRANSFER_OBTAIN mode 1568 ((frameCount == 0) || 1569 (frameCount >= mFrameCount)) 1570 ) 1571 ) && 1572 // PCM data 1573 audio_is_linear_pcm(format) && 1574 // identical channel mask to sink, or mono in and stereo sink 1575 (channelMask == mChannelMask || 1576 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1577 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1578 // hardware sample rate 1579 (sampleRate == mSampleRate) && 1580 // normal mixer has an associated fast mixer 1581 hasFastMixer() && 1582 // there are sufficient fast track slots available 1583 (mFastTrackAvailMask != 0) 1584 // FIXME test that MixerThread for this fast track has a capable output HAL 1585 // FIXME add a permission test also? 1586 ) { 1587 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1588 if (frameCount == 0) { 1589 // read the fast track multiplier property the first time it is needed 1590 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1591 if (ok != 0) { 1592 ALOGE("%s pthread_once failed: %d", __func__, ok); 1593 } 1594 frameCount = mFrameCount * sFastTrackMultiplier; 1595 } 1596 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1597 frameCount, mFrameCount); 1598 } else { 1599 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1600 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1601 "sampleRate=%u mSampleRate=%u " 1602 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1603 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1604 audio_is_linear_pcm(format), 1605 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1606 *flags &= ~IAudioFlinger::TRACK_FAST; 1607 } 1608 } 1609 // For normal PCM streaming tracks, update minimum frame count. 1610 // For compatibility with AudioTrack calculation, buffer depth is forced 1611 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1612 // This is probably too conservative, but legacy application code may depend on it. 1613 // If you change this calculation, also review the start threshold which is related. 1614 if (!(*flags & IAudioFlinger::TRACK_FAST) 1615 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1616 // this must match AudioTrack.cpp calculateMinFrameCount(). 1617 // TODO: Move to a common library 1618 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1619 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1620 if (minBufCount < 2) { 1621 minBufCount = 2; 1622 } 1623 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1624 // or the client should compute and pass in a larger buffer request. 1625 size_t minFrameCount = 1626 minBufCount * sourceFramesNeededWithTimestretch( 1627 sampleRate, mNormalFrameCount, 1628 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1629 if (frameCount < minFrameCount) { // including frameCount == 0 1630 frameCount = minFrameCount; 1631 } 1632 } 1633 *pFrameCount = frameCount; 1634 1635 switch (mType) { 1636 1637 case DIRECT: 1638 if (audio_is_linear_pcm(format)) { 1639 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1640 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1641 "for output %p with format %#x", 1642 sampleRate, format, channelMask, mOutput, mFormat); 1643 lStatus = BAD_VALUE; 1644 goto Exit; 1645 } 1646 } 1647 break; 1648 1649 case OFFLOAD: 1650 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1651 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1652 "for output %p with format %#x", 1653 sampleRate, format, channelMask, mOutput, mFormat); 1654 lStatus = BAD_VALUE; 1655 goto Exit; 1656 } 1657 break; 1658 1659 default: 1660 if (!audio_is_linear_pcm(format)) { 1661 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1662 "for output %p with format %#x", 1663 format, mOutput, mFormat); 1664 lStatus = BAD_VALUE; 1665 goto Exit; 1666 } 1667 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1668 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1669 lStatus = BAD_VALUE; 1670 goto Exit; 1671 } 1672 break; 1673 1674 } 1675 1676 lStatus = initCheck(); 1677 if (lStatus != NO_ERROR) { 1678 ALOGE("createTrack_l() audio driver not initialized"); 1679 goto Exit; 1680 } 1681 1682 { // scope for mLock 1683 Mutex::Autolock _l(mLock); 1684 1685 // all tracks in same audio session must share the same routing strategy otherwise 1686 // conflicts will happen when tracks are moved from one output to another by audio policy 1687 // manager 1688 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1689 for (size_t i = 0; i < mTracks.size(); ++i) { 1690 sp<Track> t = mTracks[i]; 1691 if (t != 0 && t->isExternalTrack()) { 1692 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1693 if (sessionId == t->sessionId() && strategy != actual) { 1694 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1695 strategy, actual); 1696 lStatus = BAD_VALUE; 1697 goto Exit; 1698 } 1699 } 1700 } 1701 1702 if (!isTimed) { 1703 track = new Track(this, client, streamType, sampleRate, format, 1704 channelMask, frameCount, NULL, sharedBuffer, 1705 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1706 } else { 1707 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1708 channelMask, frameCount, sharedBuffer, sessionId, uid); 1709 } 1710 1711 // new Track always returns non-NULL, 1712 // but TimedTrack::create() is a factory that could fail by returning NULL 1713 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1714 if (lStatus != NO_ERROR) { 1715 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1716 // track must be cleared from the caller as the caller has the AF lock 1717 goto Exit; 1718 } 1719 mTracks.add(track); 1720 1721 sp<EffectChain> chain = getEffectChain_l(sessionId); 1722 if (chain != 0) { 1723 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1724 track->setMainBuffer(chain->inBuffer()); 1725 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1726 chain->incTrackCnt(); 1727 } 1728 1729 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1730 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1731 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1732 // so ask activity manager to do this on our behalf 1733 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1734 } 1735 } 1736 1737 lStatus = NO_ERROR; 1738 1739Exit: 1740 *status = lStatus; 1741 return track; 1742} 1743 1744uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1745{ 1746 return latency; 1747} 1748 1749uint32_t AudioFlinger::PlaybackThread::latency() const 1750{ 1751 Mutex::Autolock _l(mLock); 1752 return latency_l(); 1753} 1754uint32_t AudioFlinger::PlaybackThread::latency_l() const 1755{ 1756 if (initCheck() == NO_ERROR) { 1757 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1758 } else { 1759 return 0; 1760 } 1761} 1762 1763void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1764{ 1765 Mutex::Autolock _l(mLock); 1766 // Don't apply master volume in SW if our HAL can do it for us. 1767 if (mOutput && mOutput->audioHwDev && 1768 mOutput->audioHwDev->canSetMasterVolume()) { 1769 mMasterVolume = 1.0; 1770 } else { 1771 mMasterVolume = value; 1772 } 1773} 1774 1775void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1776{ 1777 Mutex::Autolock _l(mLock); 1778 // Don't apply master mute in SW if our HAL can do it for us. 1779 if (mOutput && mOutput->audioHwDev && 1780 mOutput->audioHwDev->canSetMasterMute()) { 1781 mMasterMute = false; 1782 } else { 1783 mMasterMute = muted; 1784 } 1785} 1786 1787void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1788{ 1789 Mutex::Autolock _l(mLock); 1790 mStreamTypes[stream].volume = value; 1791 broadcast_l(); 1792} 1793 1794void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1795{ 1796 Mutex::Autolock _l(mLock); 1797 mStreamTypes[stream].mute = muted; 1798 broadcast_l(); 1799} 1800 1801float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1802{ 1803 Mutex::Autolock _l(mLock); 1804 return mStreamTypes[stream].volume; 1805} 1806 1807// addTrack_l() must be called with ThreadBase::mLock held 1808status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1809{ 1810 status_t status = ALREADY_EXISTS; 1811 1812 // set retry count for buffer fill 1813 track->mRetryCount = kMaxTrackStartupRetries; 1814 if (mActiveTracks.indexOf(track) < 0) { 1815 // the track is newly added, make sure it fills up all its 1816 // buffers before playing. This is to ensure the client will 1817 // effectively get the latency it requested. 1818 if (track->isExternalTrack()) { 1819 TrackBase::track_state state = track->mState; 1820 mLock.unlock(); 1821 status = AudioSystem::startOutput(mId, track->streamType(), 1822 (audio_session_t)track->sessionId()); 1823 mLock.lock(); 1824 // abort track was stopped/paused while we released the lock 1825 if (state != track->mState) { 1826 if (status == NO_ERROR) { 1827 mLock.unlock(); 1828 AudioSystem::stopOutput(mId, track->streamType(), 1829 (audio_session_t)track->sessionId()); 1830 mLock.lock(); 1831 } 1832 return INVALID_OPERATION; 1833 } 1834 // abort if start is rejected by audio policy manager 1835 if (status != NO_ERROR) { 1836 return PERMISSION_DENIED; 1837 } 1838#ifdef ADD_BATTERY_DATA 1839 // to track the speaker usage 1840 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1841#endif 1842 } 1843 1844 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1845 track->mResetDone = false; 1846 track->mPresentationCompleteFrames = 0; 1847 mActiveTracks.add(track); 1848 mWakeLockUids.add(track->uid()); 1849 mActiveTracksGeneration++; 1850 mLatestActiveTrack = track; 1851 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1852 if (chain != 0) { 1853 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1854 track->sessionId()); 1855 chain->incActiveTrackCnt(); 1856 } 1857 1858 status = NO_ERROR; 1859 } 1860 1861 onAddNewTrack_l(); 1862 return status; 1863} 1864 1865bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1866{ 1867 track->terminate(); 1868 // active tracks are removed by threadLoop() 1869 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1870 track->mState = TrackBase::STOPPED; 1871 if (!trackActive) { 1872 removeTrack_l(track); 1873 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1874 track->mState = TrackBase::STOPPING_1; 1875 } 1876 1877 return trackActive; 1878} 1879 1880void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1881{ 1882 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1883 mTracks.remove(track); 1884 deleteTrackName_l(track->name()); 1885 // redundant as track is about to be destroyed, for dumpsys only 1886 track->mName = -1; 1887 if (track->isFastTrack()) { 1888 int index = track->mFastIndex; 1889 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1890 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1891 mFastTrackAvailMask |= 1 << index; 1892 // redundant as track is about to be destroyed, for dumpsys only 1893 track->mFastIndex = -1; 1894 } 1895 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1896 if (chain != 0) { 1897 chain->decTrackCnt(); 1898 } 1899} 1900 1901void AudioFlinger::PlaybackThread::broadcast_l() 1902{ 1903 // Thread could be blocked waiting for async 1904 // so signal it to handle state changes immediately 1905 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1906 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1907 mSignalPending = true; 1908 mWaitWorkCV.broadcast(); 1909} 1910 1911String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1912{ 1913 Mutex::Autolock _l(mLock); 1914 if (initCheck() != NO_ERROR) { 1915 return String8(); 1916 } 1917 1918 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1919 const String8 out_s8(s); 1920 free(s); 1921 return out_s8; 1922} 1923 1924void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event) { 1925 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 1926 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 1927 1928 desc->mIoHandle = mId; 1929 1930 switch (event) { 1931 case AUDIO_OUTPUT_OPENED: 1932 case AUDIO_OUTPUT_CONFIG_CHANGED: 1933 desc->mChannelMask = mChannelMask; 1934 desc->mSamplingRate = mSampleRate; 1935 desc->mFormat = mFormat; 1936 desc->mFrameCount = mNormalFrameCount; // FIXME see 1937 // AudioFlinger::frameCount(audio_io_handle_t) 1938 desc->mLatency = latency_l(); 1939 break; 1940 1941 case AUDIO_OUTPUT_CLOSED: 1942 default: 1943 break; 1944 } 1945 mAudioFlinger->ioConfigChanged(event, desc); 1946} 1947 1948void AudioFlinger::PlaybackThread::writeCallback() 1949{ 1950 ALOG_ASSERT(mCallbackThread != 0); 1951 mCallbackThread->resetWriteBlocked(); 1952} 1953 1954void AudioFlinger::PlaybackThread::drainCallback() 1955{ 1956 ALOG_ASSERT(mCallbackThread != 0); 1957 mCallbackThread->resetDraining(); 1958} 1959 1960void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1961{ 1962 Mutex::Autolock _l(mLock); 1963 // reject out of sequence requests 1964 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1965 mWriteAckSequence &= ~1; 1966 mWaitWorkCV.signal(); 1967 } 1968} 1969 1970void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1971{ 1972 Mutex::Autolock _l(mLock); 1973 // reject out of sequence requests 1974 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1975 mDrainSequence &= ~1; 1976 mWaitWorkCV.signal(); 1977 } 1978} 1979 1980// static 1981int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1982 void *param __unused, 1983 void *cookie) 1984{ 1985 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1986 ALOGV("asyncCallback() event %d", event); 1987 switch (event) { 1988 case STREAM_CBK_EVENT_WRITE_READY: 1989 me->writeCallback(); 1990 break; 1991 case STREAM_CBK_EVENT_DRAIN_READY: 1992 me->drainCallback(); 1993 break; 1994 default: 1995 ALOGW("asyncCallback() unknown event %d", event); 1996 break; 1997 } 1998 return 0; 1999} 2000 2001void AudioFlinger::PlaybackThread::readOutputParameters_l() 2002{ 2003 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2004 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2005 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2006 if (!audio_is_output_channel(mChannelMask)) { 2007 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2008 } 2009 if ((mType == MIXER || mType == DUPLICATING) 2010 && !isValidPcmSinkChannelMask(mChannelMask)) { 2011 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2012 mChannelMask); 2013 } 2014 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2015 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2016 mFormat = mHALFormat; 2017 if (!audio_is_valid_format(mFormat)) { 2018 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2019 } 2020 if ((mType == MIXER || mType == DUPLICATING) 2021 && !isValidPcmSinkFormat(mFormat)) { 2022 LOG_FATAL("HAL format %#x not supported for mixed output", 2023 mFormat); 2024 } 2025 mFrameSize = mOutput->getFrameSize(); 2026 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2027 mFrameCount = mBufferSize / mFrameSize; 2028 if (mFrameCount & 15) { 2029 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2030 mFrameCount); 2031 } 2032 2033 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2034 (mOutput->stream->set_callback != NULL)) { 2035 if (mOutput->stream->set_callback(mOutput->stream, 2036 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2037 mUseAsyncWrite = true; 2038 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2039 } 2040 } 2041 2042 mHwSupportsPause = false; 2043 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2044 if (mOutput->stream->pause != NULL) { 2045 if (mOutput->stream->resume != NULL) { 2046 mHwSupportsPause = true; 2047 } else { 2048 ALOGW("direct output implements pause but not resume"); 2049 } 2050 } else if (mOutput->stream->resume != NULL) { 2051 ALOGW("direct output implements resume but not pause"); 2052 } 2053 } 2054 2055 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2056 // For best precision, we use float instead of the associated output 2057 // device format (typically PCM 16 bit). 2058 2059 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2060 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2061 mBufferSize = mFrameSize * mFrameCount; 2062 2063 // TODO: We currently use the associated output device channel mask and sample rate. 2064 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2065 // (if a valid mask) to avoid premature downmix. 2066 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2067 // instead of the output device sample rate to avoid loss of high frequency information. 2068 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2069 } 2070 2071 // Calculate size of normal sink buffer relative to the HAL output buffer size 2072 double multiplier = 1.0; 2073 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2074 kUseFastMixer == FastMixer_Dynamic)) { 2075 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2076 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2077 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2078 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2079 maxNormalFrameCount = maxNormalFrameCount & ~15; 2080 if (maxNormalFrameCount < minNormalFrameCount) { 2081 maxNormalFrameCount = minNormalFrameCount; 2082 } 2083 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2084 if (multiplier <= 1.0) { 2085 multiplier = 1.0; 2086 } else if (multiplier <= 2.0) { 2087 if (2 * mFrameCount <= maxNormalFrameCount) { 2088 multiplier = 2.0; 2089 } else { 2090 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2091 } 2092 } else { 2093 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2094 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2095 // track, but we sometimes have to do this to satisfy the maximum frame count 2096 // constraint) 2097 // FIXME this rounding up should not be done if no HAL SRC 2098 uint32_t truncMult = (uint32_t) multiplier; 2099 if ((truncMult & 1)) { 2100 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2101 ++truncMult; 2102 } 2103 } 2104 multiplier = (double) truncMult; 2105 } 2106 } 2107 mNormalFrameCount = multiplier * mFrameCount; 2108 // round up to nearest 16 frames to satisfy AudioMixer 2109 if (mType == MIXER || mType == DUPLICATING) { 2110 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2111 } 2112 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2113 mNormalFrameCount); 2114 2115 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2116 // Originally this was int16_t[] array, need to remove legacy implications. 2117 free(mSinkBuffer); 2118 mSinkBuffer = NULL; 2119 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2120 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2121 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2122 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2123 2124 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2125 // drives the output. 2126 free(mMixerBuffer); 2127 mMixerBuffer = NULL; 2128 if (mMixerBufferEnabled) { 2129 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2130 mMixerBufferSize = mNormalFrameCount * mChannelCount 2131 * audio_bytes_per_sample(mMixerBufferFormat); 2132 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2133 } 2134 free(mEffectBuffer); 2135 mEffectBuffer = NULL; 2136 if (mEffectBufferEnabled) { 2137 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2138 mEffectBufferSize = mNormalFrameCount * mChannelCount 2139 * audio_bytes_per_sample(mEffectBufferFormat); 2140 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2141 } 2142 2143 // force reconfiguration of effect chains and engines to take new buffer size and audio 2144 // parameters into account 2145 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2146 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2147 // matter. 2148 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2149 Vector< sp<EffectChain> > effectChains = mEffectChains; 2150 for (size_t i = 0; i < effectChains.size(); i ++) { 2151 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2152 } 2153} 2154 2155 2156status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2157{ 2158 if (halFrames == NULL || dspFrames == NULL) { 2159 return BAD_VALUE; 2160 } 2161 Mutex::Autolock _l(mLock); 2162 if (initCheck() != NO_ERROR) { 2163 return INVALID_OPERATION; 2164 } 2165 size_t framesWritten = mBytesWritten / mFrameSize; 2166 *halFrames = framesWritten; 2167 2168 if (isSuspended()) { 2169 // return an estimation of rendered frames when the output is suspended 2170 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2171 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2172 return NO_ERROR; 2173 } else { 2174 status_t status; 2175 uint32_t frames; 2176 status = mOutput->getRenderPosition(&frames); 2177 *dspFrames = (size_t)frames; 2178 return status; 2179 } 2180} 2181 2182uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2183{ 2184 Mutex::Autolock _l(mLock); 2185 uint32_t result = 0; 2186 if (getEffectChain_l(sessionId) != 0) { 2187 result = EFFECT_SESSION; 2188 } 2189 2190 for (size_t i = 0; i < mTracks.size(); ++i) { 2191 sp<Track> track = mTracks[i]; 2192 if (sessionId == track->sessionId() && !track->isInvalid()) { 2193 result |= TRACK_SESSION; 2194 break; 2195 } 2196 } 2197 2198 return result; 2199} 2200 2201uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2202{ 2203 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2204 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2205 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2206 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2207 } 2208 for (size_t i = 0; i < mTracks.size(); i++) { 2209 sp<Track> track = mTracks[i]; 2210 if (sessionId == track->sessionId() && !track->isInvalid()) { 2211 return AudioSystem::getStrategyForStream(track->streamType()); 2212 } 2213 } 2214 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2215} 2216 2217 2218AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2219{ 2220 Mutex::Autolock _l(mLock); 2221 return mOutput; 2222} 2223 2224AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2225{ 2226 Mutex::Autolock _l(mLock); 2227 AudioStreamOut *output = mOutput; 2228 mOutput = NULL; 2229 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2230 // must push a NULL and wait for ack 2231 mOutputSink.clear(); 2232 mPipeSink.clear(); 2233 mNormalSink.clear(); 2234 return output; 2235} 2236 2237// this method must always be called either with ThreadBase mLock held or inside the thread loop 2238audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2239{ 2240 if (mOutput == NULL) { 2241 return NULL; 2242 } 2243 return &mOutput->stream->common; 2244} 2245 2246uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2247{ 2248 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2249} 2250 2251status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2252{ 2253 if (!isValidSyncEvent(event)) { 2254 return BAD_VALUE; 2255 } 2256 2257 Mutex::Autolock _l(mLock); 2258 2259 for (size_t i = 0; i < mTracks.size(); ++i) { 2260 sp<Track> track = mTracks[i]; 2261 if (event->triggerSession() == track->sessionId()) { 2262 (void) track->setSyncEvent(event); 2263 return NO_ERROR; 2264 } 2265 } 2266 2267 return NAME_NOT_FOUND; 2268} 2269 2270bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2271{ 2272 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2273} 2274 2275void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2276 const Vector< sp<Track> >& tracksToRemove) 2277{ 2278 size_t count = tracksToRemove.size(); 2279 if (count > 0) { 2280 for (size_t i = 0 ; i < count ; i++) { 2281 const sp<Track>& track = tracksToRemove.itemAt(i); 2282 if (track->isExternalTrack()) { 2283 AudioSystem::stopOutput(mId, track->streamType(), 2284 (audio_session_t)track->sessionId()); 2285#ifdef ADD_BATTERY_DATA 2286 // to track the speaker usage 2287 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2288#endif 2289 if (track->isTerminated()) { 2290 AudioSystem::releaseOutput(mId, track->streamType(), 2291 (audio_session_t)track->sessionId()); 2292 } 2293 } 2294 } 2295 } 2296} 2297 2298void AudioFlinger::PlaybackThread::checkSilentMode_l() 2299{ 2300 if (!mMasterMute) { 2301 char value[PROPERTY_VALUE_MAX]; 2302 if (property_get("ro.audio.silent", value, "0") > 0) { 2303 char *endptr; 2304 unsigned long ul = strtoul(value, &endptr, 0); 2305 if (*endptr == '\0' && ul != 0) { 2306 ALOGD("Silence is golden"); 2307 // The setprop command will not allow a property to be changed after 2308 // the first time it is set, so we don't have to worry about un-muting. 2309 setMasterMute_l(true); 2310 } 2311 } 2312 } 2313} 2314 2315// shared by MIXER and DIRECT, overridden by DUPLICATING 2316ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2317{ 2318 // FIXME rewrite to reduce number of system calls 2319 mLastWriteTime = systemTime(); 2320 mInWrite = true; 2321 ssize_t bytesWritten; 2322 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2323 2324 // If an NBAIO sink is present, use it to write the normal mixer's submix 2325 if (mNormalSink != 0) { 2326 2327 const size_t count = mBytesRemaining / mFrameSize; 2328 2329 ATRACE_BEGIN("write"); 2330 // update the setpoint when AudioFlinger::mScreenState changes 2331 uint32_t screenState = AudioFlinger::mScreenState; 2332 if (screenState != mScreenState) { 2333 mScreenState = screenState; 2334 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2335 if (pipe != NULL) { 2336 pipe->setAvgFrames((mScreenState & 1) ? 2337 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2338 } 2339 } 2340 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2341 ATRACE_END(); 2342 if (framesWritten > 0) { 2343 bytesWritten = framesWritten * mFrameSize; 2344 } else { 2345 bytesWritten = framesWritten; 2346 } 2347 mLatchDValid = false; 2348 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2349 if (status == NO_ERROR) { 2350 size_t totalFramesWritten = mNormalSink->framesWritten(); 2351 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2352 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2353 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2354 mLatchDValid = true; 2355 } 2356 } 2357 // otherwise use the HAL / AudioStreamOut directly 2358 } else { 2359 // Direct output and offload threads 2360 2361 if (mUseAsyncWrite) { 2362 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2363 mWriteAckSequence += 2; 2364 mWriteAckSequence |= 1; 2365 ALOG_ASSERT(mCallbackThread != 0); 2366 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2367 } 2368 // FIXME We should have an implementation of timestamps for direct output threads. 2369 // They are used e.g for multichannel PCM playback over HDMI. 2370 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2371 if (mUseAsyncWrite && 2372 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2373 // do not wait for async callback in case of error of full write 2374 mWriteAckSequence &= ~1; 2375 ALOG_ASSERT(mCallbackThread != 0); 2376 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2377 } 2378 } 2379 2380 mNumWrites++; 2381 mInWrite = false; 2382 mStandby = false; 2383 return bytesWritten; 2384} 2385 2386void AudioFlinger::PlaybackThread::threadLoop_drain() 2387{ 2388 if (mOutput->stream->drain) { 2389 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2390 if (mUseAsyncWrite) { 2391 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2392 mDrainSequence |= 1; 2393 ALOG_ASSERT(mCallbackThread != 0); 2394 mCallbackThread->setDraining(mDrainSequence); 2395 } 2396 mOutput->stream->drain(mOutput->stream, 2397 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2398 : AUDIO_DRAIN_ALL); 2399 } 2400} 2401 2402void AudioFlinger::PlaybackThread::threadLoop_exit() 2403{ 2404 { 2405 Mutex::Autolock _l(mLock); 2406 for (size_t i = 0; i < mTracks.size(); i++) { 2407 sp<Track> track = mTracks[i]; 2408 track->invalidate(); 2409 } 2410 } 2411} 2412 2413/* 2414The derived values that are cached: 2415 - mSinkBufferSize from frame count * frame size 2416 - activeSleepTime from activeSleepTimeUs() 2417 - idleSleepTime from idleSleepTimeUs() 2418 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2419 - maxPeriod from frame count and sample rate (MIXER only) 2420 2421The parameters that affect these derived values are: 2422 - frame count 2423 - frame size 2424 - sample rate 2425 - device type: A2DP or not 2426 - device latency 2427 - format: PCM or not 2428 - active sleep time 2429 - idle sleep time 2430*/ 2431 2432void AudioFlinger::PlaybackThread::cacheParameters_l() 2433{ 2434 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2435 activeSleepTime = activeSleepTimeUs(); 2436 idleSleepTime = idleSleepTimeUs(); 2437} 2438 2439void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2440{ 2441 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2442 this, streamType, mTracks.size()); 2443 Mutex::Autolock _l(mLock); 2444 2445 size_t size = mTracks.size(); 2446 for (size_t i = 0; i < size; i++) { 2447 sp<Track> t = mTracks[i]; 2448 if (t->streamType() == streamType) { 2449 t->invalidate(); 2450 } 2451 } 2452} 2453 2454status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2455{ 2456 int session = chain->sessionId(); 2457 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2458 ? mEffectBuffer : mSinkBuffer); 2459 bool ownsBuffer = false; 2460 2461 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2462 if (session > 0) { 2463 // Only one effect chain can be present in direct output thread and it uses 2464 // the sink buffer as input 2465 if (mType != DIRECT) { 2466 size_t numSamples = mNormalFrameCount * mChannelCount; 2467 buffer = new int16_t[numSamples]; 2468 memset(buffer, 0, numSamples * sizeof(int16_t)); 2469 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2470 ownsBuffer = true; 2471 } 2472 2473 // Attach all tracks with same session ID to this chain. 2474 for (size_t i = 0; i < mTracks.size(); ++i) { 2475 sp<Track> track = mTracks[i]; 2476 if (session == track->sessionId()) { 2477 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2478 buffer); 2479 track->setMainBuffer(buffer); 2480 chain->incTrackCnt(); 2481 } 2482 } 2483 2484 // indicate all active tracks in the chain 2485 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2486 sp<Track> track = mActiveTracks[i].promote(); 2487 if (track == 0) { 2488 continue; 2489 } 2490 if (session == track->sessionId()) { 2491 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2492 chain->incActiveTrackCnt(); 2493 } 2494 } 2495 } 2496 chain->setThread(this); 2497 chain->setInBuffer(buffer, ownsBuffer); 2498 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2499 ? mEffectBuffer : mSinkBuffer)); 2500 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2501 // chains list in order to be processed last as it contains output stage effects 2502 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2503 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2504 // after track specific effects and before output stage 2505 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2506 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2507 // Effect chain for other sessions are inserted at beginning of effect 2508 // chains list to be processed before output mix effects. Relative order between other 2509 // sessions is not important 2510 size_t size = mEffectChains.size(); 2511 size_t i = 0; 2512 for (i = 0; i < size; i++) { 2513 if (mEffectChains[i]->sessionId() < session) { 2514 break; 2515 } 2516 } 2517 mEffectChains.insertAt(chain, i); 2518 checkSuspendOnAddEffectChain_l(chain); 2519 2520 return NO_ERROR; 2521} 2522 2523size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2524{ 2525 int session = chain->sessionId(); 2526 2527 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2528 2529 for (size_t i = 0; i < mEffectChains.size(); i++) { 2530 if (chain == mEffectChains[i]) { 2531 mEffectChains.removeAt(i); 2532 // detach all active tracks from the chain 2533 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2534 sp<Track> track = mActiveTracks[i].promote(); 2535 if (track == 0) { 2536 continue; 2537 } 2538 if (session == track->sessionId()) { 2539 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2540 chain.get(), session); 2541 chain->decActiveTrackCnt(); 2542 } 2543 } 2544 2545 // detach all tracks with same session ID from this chain 2546 for (size_t i = 0; i < mTracks.size(); ++i) { 2547 sp<Track> track = mTracks[i]; 2548 if (session == track->sessionId()) { 2549 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2550 chain->decTrackCnt(); 2551 } 2552 } 2553 break; 2554 } 2555 } 2556 return mEffectChains.size(); 2557} 2558 2559status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2560 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2561{ 2562 Mutex::Autolock _l(mLock); 2563 return attachAuxEffect_l(track, EffectId); 2564} 2565 2566status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2567 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2568{ 2569 status_t status = NO_ERROR; 2570 2571 if (EffectId == 0) { 2572 track->setAuxBuffer(0, NULL); 2573 } else { 2574 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2575 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2576 if (effect != 0) { 2577 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2578 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2579 } else { 2580 status = INVALID_OPERATION; 2581 } 2582 } else { 2583 status = BAD_VALUE; 2584 } 2585 } 2586 return status; 2587} 2588 2589void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2590{ 2591 for (size_t i = 0; i < mTracks.size(); ++i) { 2592 sp<Track> track = mTracks[i]; 2593 if (track->auxEffectId() == effectId) { 2594 attachAuxEffect_l(track, 0); 2595 } 2596 } 2597} 2598 2599bool AudioFlinger::PlaybackThread::threadLoop() 2600{ 2601 Vector< sp<Track> > tracksToRemove; 2602 2603 standbyTime = systemTime(); 2604 2605 // MIXER 2606 nsecs_t lastWarning = 0; 2607 2608 // DUPLICATING 2609 // FIXME could this be made local to while loop? 2610 writeFrames = 0; 2611 2612 int lastGeneration = 0; 2613 2614 cacheParameters_l(); 2615 sleepTime = idleSleepTime; 2616 2617 if (mType == MIXER) { 2618 sleepTimeShift = 0; 2619 } 2620 2621 CpuStats cpuStats; 2622 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2623 2624 acquireWakeLock(); 2625 2626 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2627 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2628 // and then that string will be logged at the next convenient opportunity. 2629 const char *logString = NULL; 2630 2631 checkSilentMode_l(); 2632 2633 while (!exitPending()) 2634 { 2635 cpuStats.sample(myName); 2636 2637 Vector< sp<EffectChain> > effectChains; 2638 2639 { // scope for mLock 2640 2641 Mutex::Autolock _l(mLock); 2642 2643 processConfigEvents_l(); 2644 2645 if (logString != NULL) { 2646 mNBLogWriter->logTimestamp(); 2647 mNBLogWriter->log(logString); 2648 logString = NULL; 2649 } 2650 2651 // Gather the framesReleased counters for all active tracks, 2652 // and latch them atomically with the timestamp. 2653 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2654 mLatchD.mFramesReleased.clear(); 2655 size_t size = mActiveTracks.size(); 2656 for (size_t i = 0; i < size; i++) { 2657 sp<Track> t = mActiveTracks[i].promote(); 2658 if (t != 0) { 2659 mLatchD.mFramesReleased.add(t.get(), 2660 t->mAudioTrackServerProxy->framesReleased()); 2661 } 2662 } 2663 if (mLatchDValid) { 2664 mLatchQ = mLatchD; 2665 mLatchDValid = false; 2666 mLatchQValid = true; 2667 } 2668 2669 saveOutputTracks(); 2670 if (mSignalPending) { 2671 // A signal was raised while we were unlocked 2672 mSignalPending = false; 2673 } else if (waitingAsyncCallback_l()) { 2674 if (exitPending()) { 2675 break; 2676 } 2677 releaseWakeLock_l(); 2678 mWakeLockUids.clear(); 2679 mActiveTracksGeneration++; 2680 ALOGV("wait async completion"); 2681 mWaitWorkCV.wait(mLock); 2682 ALOGV("async completion/wake"); 2683 acquireWakeLock_l(); 2684 standbyTime = systemTime() + standbyDelay; 2685 sleepTime = 0; 2686 2687 continue; 2688 } 2689 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2690 isSuspended()) { 2691 // put audio hardware into standby after short delay 2692 if (shouldStandby_l()) { 2693 2694 threadLoop_standby(); 2695 2696 mStandby = true; 2697 } 2698 2699 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2700 // we're about to wait, flush the binder command buffer 2701 IPCThreadState::self()->flushCommands(); 2702 2703 clearOutputTracks(); 2704 2705 if (exitPending()) { 2706 break; 2707 } 2708 2709 releaseWakeLock_l(); 2710 mWakeLockUids.clear(); 2711 mActiveTracksGeneration++; 2712 // wait until we have something to do... 2713 ALOGV("%s going to sleep", myName.string()); 2714 mWaitWorkCV.wait(mLock); 2715 ALOGV("%s waking up", myName.string()); 2716 acquireWakeLock_l(); 2717 2718 mMixerStatus = MIXER_IDLE; 2719 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2720 mBytesWritten = 0; 2721 mBytesRemaining = 0; 2722 checkSilentMode_l(); 2723 2724 standbyTime = systemTime() + standbyDelay; 2725 sleepTime = idleSleepTime; 2726 if (mType == MIXER) { 2727 sleepTimeShift = 0; 2728 } 2729 2730 continue; 2731 } 2732 } 2733 // mMixerStatusIgnoringFastTracks is also updated internally 2734 mMixerStatus = prepareTracks_l(&tracksToRemove); 2735 2736 // compare with previously applied list 2737 if (lastGeneration != mActiveTracksGeneration) { 2738 // update wakelock 2739 updateWakeLockUids_l(mWakeLockUids); 2740 lastGeneration = mActiveTracksGeneration; 2741 } 2742 2743 // prevent any changes in effect chain list and in each effect chain 2744 // during mixing and effect process as the audio buffers could be deleted 2745 // or modified if an effect is created or deleted 2746 lockEffectChains_l(effectChains); 2747 } // mLock scope ends 2748 2749 if (mBytesRemaining == 0) { 2750 mCurrentWriteLength = 0; 2751 if (mMixerStatus == MIXER_TRACKS_READY) { 2752 // threadLoop_mix() sets mCurrentWriteLength 2753 threadLoop_mix(); 2754 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2755 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2756 // threadLoop_sleepTime sets sleepTime to 0 if data 2757 // must be written to HAL 2758 threadLoop_sleepTime(); 2759 if (sleepTime == 0) { 2760 mCurrentWriteLength = mSinkBufferSize; 2761 } 2762 } 2763 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2764 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2765 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2766 // or mSinkBuffer (if there are no effects). 2767 // 2768 // This is done pre-effects computation; if effects change to 2769 // support higher precision, this needs to move. 2770 // 2771 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2772 // TODO use sleepTime == 0 as an additional condition. 2773 if (mMixerBufferValid) { 2774 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2775 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2776 2777 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2778 mNormalFrameCount * mChannelCount); 2779 } 2780 2781 mBytesRemaining = mCurrentWriteLength; 2782 if (isSuspended()) { 2783 sleepTime = suspendSleepTimeUs(); 2784 // simulate write to HAL when suspended 2785 mBytesWritten += mSinkBufferSize; 2786 mBytesRemaining = 0; 2787 } 2788 2789 // only process effects if we're going to write 2790 if (sleepTime == 0 && mType != OFFLOAD) { 2791 for (size_t i = 0; i < effectChains.size(); i ++) { 2792 effectChains[i]->process_l(); 2793 } 2794 } 2795 } 2796 // Process effect chains for offloaded thread even if no audio 2797 // was read from audio track: process only updates effect state 2798 // and thus does have to be synchronized with audio writes but may have 2799 // to be called while waiting for async write callback 2800 if (mType == OFFLOAD) { 2801 for (size_t i = 0; i < effectChains.size(); i ++) { 2802 effectChains[i]->process_l(); 2803 } 2804 } 2805 2806 // Only if the Effects buffer is enabled and there is data in the 2807 // Effects buffer (buffer valid), we need to 2808 // copy into the sink buffer. 2809 // TODO use sleepTime == 0 as an additional condition. 2810 if (mEffectBufferValid) { 2811 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2812 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2813 mNormalFrameCount * mChannelCount); 2814 } 2815 2816 // enable changes in effect chain 2817 unlockEffectChains(effectChains); 2818 2819 if (!waitingAsyncCallback()) { 2820 // sleepTime == 0 means we must write to audio hardware 2821 if (sleepTime == 0) { 2822 if (mBytesRemaining) { 2823 ssize_t ret = threadLoop_write(); 2824 if (ret < 0) { 2825 mBytesRemaining = 0; 2826 } else { 2827 mBytesWritten += ret; 2828 mBytesRemaining -= ret; 2829 } 2830 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2831 (mMixerStatus == MIXER_DRAIN_ALL)) { 2832 threadLoop_drain(); 2833 } 2834 if (mType == MIXER) { 2835 // write blocked detection 2836 nsecs_t now = systemTime(); 2837 nsecs_t delta = now - mLastWriteTime; 2838 if (!mStandby && delta > maxPeriod) { 2839 mNumDelayedWrites++; 2840 if ((now - lastWarning) > kWarningThrottleNs) { 2841 ATRACE_NAME("underrun"); 2842 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2843 ns2ms(delta), mNumDelayedWrites, this); 2844 lastWarning = now; 2845 } 2846 } 2847 } 2848 2849 } else { 2850 ATRACE_BEGIN("sleep"); 2851 usleep(sleepTime); 2852 ATRACE_END(); 2853 } 2854 } 2855 2856 // Finally let go of removed track(s), without the lock held 2857 // since we can't guarantee the destructors won't acquire that 2858 // same lock. This will also mutate and push a new fast mixer state. 2859 threadLoop_removeTracks(tracksToRemove); 2860 tracksToRemove.clear(); 2861 2862 // FIXME I don't understand the need for this here; 2863 // it was in the original code but maybe the 2864 // assignment in saveOutputTracks() makes this unnecessary? 2865 clearOutputTracks(); 2866 2867 // Effect chains will be actually deleted here if they were removed from 2868 // mEffectChains list during mixing or effects processing 2869 effectChains.clear(); 2870 2871 // FIXME Note that the above .clear() is no longer necessary since effectChains 2872 // is now local to this block, but will keep it for now (at least until merge done). 2873 } 2874 2875 threadLoop_exit(); 2876 2877 if (!mStandby) { 2878 threadLoop_standby(); 2879 mStandby = true; 2880 } 2881 2882 releaseWakeLock(); 2883 mWakeLockUids.clear(); 2884 mActiveTracksGeneration++; 2885 2886 ALOGV("Thread %p type %d exiting", this, mType); 2887 return false; 2888} 2889 2890// removeTracks_l() must be called with ThreadBase::mLock held 2891void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2892{ 2893 size_t count = tracksToRemove.size(); 2894 if (count > 0) { 2895 for (size_t i=0 ; i<count ; i++) { 2896 const sp<Track>& track = tracksToRemove.itemAt(i); 2897 mActiveTracks.remove(track); 2898 mWakeLockUids.remove(track->uid()); 2899 mActiveTracksGeneration++; 2900 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2901 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2902 if (chain != 0) { 2903 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2904 track->sessionId()); 2905 chain->decActiveTrackCnt(); 2906 } 2907 if (track->isTerminated()) { 2908 removeTrack_l(track); 2909 } 2910 } 2911 } 2912 2913} 2914 2915status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2916{ 2917 if (mNormalSink != 0) { 2918 return mNormalSink->getTimestamp(timestamp); 2919 } 2920 if ((mType == OFFLOAD || mType == DIRECT) 2921 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2922 uint64_t position64; 2923 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 2924 if (ret == 0) { 2925 timestamp.mPosition = (uint32_t)position64; 2926 return NO_ERROR; 2927 } 2928 } 2929 return INVALID_OPERATION; 2930} 2931 2932status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 2933 audio_patch_handle_t *handle) 2934{ 2935 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2936 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2937 if (mFastMixer != 0) { 2938 FastMixerStateQueue *sq = mFastMixer->sq(); 2939 FastMixerState *state = sq->begin(); 2940 if (!(state->mCommand & FastMixerState::IDLE)) { 2941 previousCommand = state->mCommand; 2942 state->mCommand = FastMixerState::HOT_IDLE; 2943 sq->end(); 2944 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2945 } else { 2946 sq->end(false /*didModify*/); 2947 } 2948 } 2949 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 2950 2951 if (!(previousCommand & FastMixerState::IDLE)) { 2952 ALOG_ASSERT(mFastMixer != 0); 2953 FastMixerStateQueue *sq = mFastMixer->sq(); 2954 FastMixerState *state = sq->begin(); 2955 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 2956 state->mCommand = previousCommand; 2957 sq->end(); 2958 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2959 } 2960 2961 return status; 2962} 2963 2964status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2965 audio_patch_handle_t *handle) 2966{ 2967 status_t status = NO_ERROR; 2968 2969 // store new device and send to effects 2970 audio_devices_t type = AUDIO_DEVICE_NONE; 2971 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2972 type |= patch->sinks[i].ext.device.type; 2973 } 2974 2975#ifdef ADD_BATTERY_DATA 2976 // when changing the audio output device, call addBatteryData to notify 2977 // the change 2978 if (mOutDevice != type) { 2979 uint32_t params = 0; 2980 // check whether speaker is on 2981 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 2982 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2983 } 2984 2985 audio_devices_t deviceWithoutSpeaker 2986 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2987 // check if any other device (except speaker) is on 2988 if (type & deviceWithoutSpeaker) { 2989 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2990 } 2991 2992 if (params != 0) { 2993 addBatteryData(params); 2994 } 2995 } 2996#endif 2997 2998 for (size_t i = 0; i < mEffectChains.size(); i++) { 2999 mEffectChains[i]->setDevice_l(type); 3000 } 3001 mOutDevice = type; 3002 3003 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3004 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3005 status = hwDevice->create_audio_patch(hwDevice, 3006 patch->num_sources, 3007 patch->sources, 3008 patch->num_sinks, 3009 patch->sinks, 3010 handle); 3011 } else { 3012 char *address; 3013 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3014 //FIXME: we only support address on first sink with HAL version < 3.0 3015 address = audio_device_address_to_parameter( 3016 patch->sinks[0].ext.device.type, 3017 patch->sinks[0].ext.device.address); 3018 } else { 3019 address = (char *)calloc(1, 1); 3020 } 3021 AudioParameter param = AudioParameter(String8(address)); 3022 free(address); 3023 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3024 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3025 param.toString().string()); 3026 *handle = AUDIO_PATCH_HANDLE_NONE; 3027 } 3028 return status; 3029} 3030 3031status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3032{ 3033 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3034 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3035 if (mFastMixer != 0) { 3036 FastMixerStateQueue *sq = mFastMixer->sq(); 3037 FastMixerState *state = sq->begin(); 3038 if (!(state->mCommand & FastMixerState::IDLE)) { 3039 previousCommand = state->mCommand; 3040 state->mCommand = FastMixerState::HOT_IDLE; 3041 sq->end(); 3042 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3043 } else { 3044 sq->end(false /*didModify*/); 3045 } 3046 } 3047 3048 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3049 3050 if (!(previousCommand & FastMixerState::IDLE)) { 3051 ALOG_ASSERT(mFastMixer != 0); 3052 FastMixerStateQueue *sq = mFastMixer->sq(); 3053 FastMixerState *state = sq->begin(); 3054 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3055 state->mCommand = previousCommand; 3056 sq->end(); 3057 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3058 } 3059 3060 return status; 3061} 3062 3063status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3064{ 3065 status_t status = NO_ERROR; 3066 3067 mOutDevice = AUDIO_DEVICE_NONE; 3068 3069 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3070 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3071 status = hwDevice->release_audio_patch(hwDevice, handle); 3072 } else { 3073 AudioParameter param; 3074 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3075 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3076 param.toString().string()); 3077 } 3078 return status; 3079} 3080 3081void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3082{ 3083 Mutex::Autolock _l(mLock); 3084 mTracks.add(track); 3085} 3086 3087void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3088{ 3089 Mutex::Autolock _l(mLock); 3090 destroyTrack_l(track); 3091} 3092 3093void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3094{ 3095 ThreadBase::getAudioPortConfig(config); 3096 config->role = AUDIO_PORT_ROLE_SOURCE; 3097 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3098 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3099} 3100 3101// ---------------------------------------------------------------------------- 3102 3103AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3104 audio_io_handle_t id, audio_devices_t device, type_t type) 3105 : PlaybackThread(audioFlinger, output, id, device, type), 3106 // mAudioMixer below 3107 // mFastMixer below 3108 mFastMixerFutex(0) 3109 // mOutputSink below 3110 // mPipeSink below 3111 // mNormalSink below 3112{ 3113 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3114 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3115 "mFrameCount=%d, mNormalFrameCount=%d", 3116 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3117 mNormalFrameCount); 3118 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3119 3120 if (type == DUPLICATING) { 3121 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3122 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3123 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3124 return; 3125 } 3126 // create an NBAIO sink for the HAL output stream, and negotiate 3127 mOutputSink = new AudioStreamOutSink(output->stream); 3128 size_t numCounterOffers = 0; 3129 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3130 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3131 ALOG_ASSERT(index == 0); 3132 3133 // initialize fast mixer depending on configuration 3134 bool initFastMixer; 3135 switch (kUseFastMixer) { 3136 case FastMixer_Never: 3137 initFastMixer = false; 3138 break; 3139 case FastMixer_Always: 3140 initFastMixer = true; 3141 break; 3142 case FastMixer_Static: 3143 case FastMixer_Dynamic: 3144 initFastMixer = mFrameCount < mNormalFrameCount; 3145 break; 3146 } 3147 if (initFastMixer) { 3148 audio_format_t fastMixerFormat; 3149 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3150 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3151 } else { 3152 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3153 } 3154 if (mFormat != fastMixerFormat) { 3155 // change our Sink format to accept our intermediate precision 3156 mFormat = fastMixerFormat; 3157 free(mSinkBuffer); 3158 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3159 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3160 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3161 } 3162 3163 // create a MonoPipe to connect our submix to FastMixer 3164 NBAIO_Format format = mOutputSink->format(); 3165 NBAIO_Format origformat = format; 3166 // adjust format to match that of the Fast Mixer 3167 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3168 format.mFormat = fastMixerFormat; 3169 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3170 3171 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3172 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3173 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3174 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3175 const NBAIO_Format offers[1] = {format}; 3176 size_t numCounterOffers = 0; 3177 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3178 ALOG_ASSERT(index == 0); 3179 monoPipe->setAvgFrames((mScreenState & 1) ? 3180 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3181 mPipeSink = monoPipe; 3182 3183#ifdef TEE_SINK 3184 if (mTeeSinkOutputEnabled) { 3185 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3186 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3187 const NBAIO_Format offers2[1] = {origformat}; 3188 numCounterOffers = 0; 3189 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3190 ALOG_ASSERT(index == 0); 3191 mTeeSink = teeSink; 3192 PipeReader *teeSource = new PipeReader(*teeSink); 3193 numCounterOffers = 0; 3194 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3195 ALOG_ASSERT(index == 0); 3196 mTeeSource = teeSource; 3197 } 3198#endif 3199 3200 // create fast mixer and configure it initially with just one fast track for our submix 3201 mFastMixer = new FastMixer(); 3202 FastMixerStateQueue *sq = mFastMixer->sq(); 3203#ifdef STATE_QUEUE_DUMP 3204 sq->setObserverDump(&mStateQueueObserverDump); 3205 sq->setMutatorDump(&mStateQueueMutatorDump); 3206#endif 3207 FastMixerState *state = sq->begin(); 3208 FastTrack *fastTrack = &state->mFastTracks[0]; 3209 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3210 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3211 fastTrack->mVolumeProvider = NULL; 3212 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3213 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3214 fastTrack->mGeneration++; 3215 state->mFastTracksGen++; 3216 state->mTrackMask = 1; 3217 // fast mixer will use the HAL output sink 3218 state->mOutputSink = mOutputSink.get(); 3219 state->mOutputSinkGen++; 3220 state->mFrameCount = mFrameCount; 3221 state->mCommand = FastMixerState::COLD_IDLE; 3222 // already done in constructor initialization list 3223 //mFastMixerFutex = 0; 3224 state->mColdFutexAddr = &mFastMixerFutex; 3225 state->mColdGen++; 3226 state->mDumpState = &mFastMixerDumpState; 3227#ifdef TEE_SINK 3228 state->mTeeSink = mTeeSink.get(); 3229#endif 3230 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3231 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3232 sq->end(); 3233 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3234 3235 // start the fast mixer 3236 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3237 pid_t tid = mFastMixer->getTid(); 3238 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3239 if (err != 0) { 3240 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3241 kPriorityFastMixer, getpid_cached, tid, err); 3242 } 3243 3244#ifdef AUDIO_WATCHDOG 3245 // create and start the watchdog 3246 mAudioWatchdog = new AudioWatchdog(); 3247 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3248 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3249 tid = mAudioWatchdog->getTid(); 3250 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3251 if (err != 0) { 3252 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3253 kPriorityFastMixer, getpid_cached, tid, err); 3254 } 3255#endif 3256 3257 } 3258 3259 switch (kUseFastMixer) { 3260 case FastMixer_Never: 3261 case FastMixer_Dynamic: 3262 mNormalSink = mOutputSink; 3263 break; 3264 case FastMixer_Always: 3265 mNormalSink = mPipeSink; 3266 break; 3267 case FastMixer_Static: 3268 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3269 break; 3270 } 3271} 3272 3273AudioFlinger::MixerThread::~MixerThread() 3274{ 3275 if (mFastMixer != 0) { 3276 FastMixerStateQueue *sq = mFastMixer->sq(); 3277 FastMixerState *state = sq->begin(); 3278 if (state->mCommand == FastMixerState::COLD_IDLE) { 3279 int32_t old = android_atomic_inc(&mFastMixerFutex); 3280 if (old == -1) { 3281 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3282 } 3283 } 3284 state->mCommand = FastMixerState::EXIT; 3285 sq->end(); 3286 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3287 mFastMixer->join(); 3288 // Though the fast mixer thread has exited, it's state queue is still valid. 3289 // We'll use that extract the final state which contains one remaining fast track 3290 // corresponding to our sub-mix. 3291 state = sq->begin(); 3292 ALOG_ASSERT(state->mTrackMask == 1); 3293 FastTrack *fastTrack = &state->mFastTracks[0]; 3294 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3295 delete fastTrack->mBufferProvider; 3296 sq->end(false /*didModify*/); 3297 mFastMixer.clear(); 3298#ifdef AUDIO_WATCHDOG 3299 if (mAudioWatchdog != 0) { 3300 mAudioWatchdog->requestExit(); 3301 mAudioWatchdog->requestExitAndWait(); 3302 mAudioWatchdog.clear(); 3303 } 3304#endif 3305 } 3306 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3307 delete mAudioMixer; 3308} 3309 3310 3311uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3312{ 3313 if (mFastMixer != 0) { 3314 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3315 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3316 } 3317 return latency; 3318} 3319 3320 3321void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3322{ 3323 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3324} 3325 3326ssize_t AudioFlinger::MixerThread::threadLoop_write() 3327{ 3328 // FIXME we should only do one push per cycle; confirm this is true 3329 // Start the fast mixer if it's not already running 3330 if (mFastMixer != 0) { 3331 FastMixerStateQueue *sq = mFastMixer->sq(); 3332 FastMixerState *state = sq->begin(); 3333 if (state->mCommand != FastMixerState::MIX_WRITE && 3334 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3335 if (state->mCommand == FastMixerState::COLD_IDLE) { 3336 int32_t old = android_atomic_inc(&mFastMixerFutex); 3337 if (old == -1) { 3338 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3339 } 3340#ifdef AUDIO_WATCHDOG 3341 if (mAudioWatchdog != 0) { 3342 mAudioWatchdog->resume(); 3343 } 3344#endif 3345 } 3346 state->mCommand = FastMixerState::MIX_WRITE; 3347#ifdef FAST_THREAD_STATISTICS 3348 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3349 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3350#endif 3351 sq->end(); 3352 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3353 if (kUseFastMixer == FastMixer_Dynamic) { 3354 mNormalSink = mPipeSink; 3355 } 3356 } else { 3357 sq->end(false /*didModify*/); 3358 } 3359 } 3360 return PlaybackThread::threadLoop_write(); 3361} 3362 3363void AudioFlinger::MixerThread::threadLoop_standby() 3364{ 3365 // Idle the fast mixer if it's currently running 3366 if (mFastMixer != 0) { 3367 FastMixerStateQueue *sq = mFastMixer->sq(); 3368 FastMixerState *state = sq->begin(); 3369 if (!(state->mCommand & FastMixerState::IDLE)) { 3370 state->mCommand = FastMixerState::COLD_IDLE; 3371 state->mColdFutexAddr = &mFastMixerFutex; 3372 state->mColdGen++; 3373 mFastMixerFutex = 0; 3374 sq->end(); 3375 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3376 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3377 if (kUseFastMixer == FastMixer_Dynamic) { 3378 mNormalSink = mOutputSink; 3379 } 3380#ifdef AUDIO_WATCHDOG 3381 if (mAudioWatchdog != 0) { 3382 mAudioWatchdog->pause(); 3383 } 3384#endif 3385 } else { 3386 sq->end(false /*didModify*/); 3387 } 3388 } 3389 PlaybackThread::threadLoop_standby(); 3390} 3391 3392bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3393{ 3394 return false; 3395} 3396 3397bool AudioFlinger::PlaybackThread::shouldStandby_l() 3398{ 3399 return !mStandby; 3400} 3401 3402bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3403{ 3404 Mutex::Autolock _l(mLock); 3405 return waitingAsyncCallback_l(); 3406} 3407 3408// shared by MIXER and DIRECT, overridden by DUPLICATING 3409void AudioFlinger::PlaybackThread::threadLoop_standby() 3410{ 3411 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3412 mOutput->standby(); 3413 if (mUseAsyncWrite != 0) { 3414 // discard any pending drain or write ack by incrementing sequence 3415 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3416 mDrainSequence = (mDrainSequence + 2) & ~1; 3417 ALOG_ASSERT(mCallbackThread != 0); 3418 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3419 mCallbackThread->setDraining(mDrainSequence); 3420 } 3421 mHwPaused = false; 3422} 3423 3424void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3425{ 3426 ALOGV("signal playback thread"); 3427 broadcast_l(); 3428} 3429 3430void AudioFlinger::MixerThread::threadLoop_mix() 3431{ 3432 // obtain the presentation timestamp of the next output buffer 3433 int64_t pts; 3434 status_t status = INVALID_OPERATION; 3435 3436 if (mNormalSink != 0) { 3437 status = mNormalSink->getNextWriteTimestamp(&pts); 3438 } else { 3439 status = mOutputSink->getNextWriteTimestamp(&pts); 3440 } 3441 3442 if (status != NO_ERROR) { 3443 pts = AudioBufferProvider::kInvalidPTS; 3444 } 3445 3446 // mix buffers... 3447 mAudioMixer->process(pts); 3448 mCurrentWriteLength = mSinkBufferSize; 3449 // increase sleep time progressively when application underrun condition clears. 3450 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3451 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3452 // such that we would underrun the audio HAL. 3453 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3454 sleepTimeShift--; 3455 } 3456 sleepTime = 0; 3457 standbyTime = systemTime() + standbyDelay; 3458 //TODO: delay standby when effects have a tail 3459 3460} 3461 3462void AudioFlinger::MixerThread::threadLoop_sleepTime() 3463{ 3464 // If no tracks are ready, sleep once for the duration of an output 3465 // buffer size, then write 0s to the output 3466 if (sleepTime == 0) { 3467 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3468 sleepTime = activeSleepTime >> sleepTimeShift; 3469 if (sleepTime < kMinThreadSleepTimeUs) { 3470 sleepTime = kMinThreadSleepTimeUs; 3471 } 3472 // reduce sleep time in case of consecutive application underruns to avoid 3473 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3474 // duration we would end up writing less data than needed by the audio HAL if 3475 // the condition persists. 3476 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3477 sleepTimeShift++; 3478 } 3479 } else { 3480 sleepTime = idleSleepTime; 3481 } 3482 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3483 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3484 // before effects processing or output. 3485 if (mMixerBufferValid) { 3486 memset(mMixerBuffer, 0, mMixerBufferSize); 3487 } else { 3488 memset(mSinkBuffer, 0, mSinkBufferSize); 3489 } 3490 sleepTime = 0; 3491 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3492 "anticipated start"); 3493 } 3494 // TODO add standby time extension fct of effect tail 3495} 3496 3497// prepareTracks_l() must be called with ThreadBase::mLock held 3498AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3499 Vector< sp<Track> > *tracksToRemove) 3500{ 3501 3502 mixer_state mixerStatus = MIXER_IDLE; 3503 // find out which tracks need to be processed 3504 size_t count = mActiveTracks.size(); 3505 size_t mixedTracks = 0; 3506 size_t tracksWithEffect = 0; 3507 // counts only _active_ fast tracks 3508 size_t fastTracks = 0; 3509 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3510 3511 float masterVolume = mMasterVolume; 3512 bool masterMute = mMasterMute; 3513 3514 if (masterMute) { 3515 masterVolume = 0; 3516 } 3517 // Delegate master volume control to effect in output mix effect chain if needed 3518 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3519 if (chain != 0) { 3520 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3521 chain->setVolume_l(&v, &v); 3522 masterVolume = (float)((v + (1 << 23)) >> 24); 3523 chain.clear(); 3524 } 3525 3526 // prepare a new state to push 3527 FastMixerStateQueue *sq = NULL; 3528 FastMixerState *state = NULL; 3529 bool didModify = false; 3530 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3531 if (mFastMixer != 0) { 3532 sq = mFastMixer->sq(); 3533 state = sq->begin(); 3534 } 3535 3536 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3537 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3538 3539 for (size_t i=0 ; i<count ; i++) { 3540 const sp<Track> t = mActiveTracks[i].promote(); 3541 if (t == 0) { 3542 continue; 3543 } 3544 3545 // this const just means the local variable doesn't change 3546 Track* const track = t.get(); 3547 3548 // process fast tracks 3549 if (track->isFastTrack()) { 3550 3551 // It's theoretically possible (though unlikely) for a fast track to be created 3552 // and then removed within the same normal mix cycle. This is not a problem, as 3553 // the track never becomes active so it's fast mixer slot is never touched. 3554 // The converse, of removing an (active) track and then creating a new track 3555 // at the identical fast mixer slot within the same normal mix cycle, 3556 // is impossible because the slot isn't marked available until the end of each cycle. 3557 int j = track->mFastIndex; 3558 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3559 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3560 FastTrack *fastTrack = &state->mFastTracks[j]; 3561 3562 // Determine whether the track is currently in underrun condition, 3563 // and whether it had a recent underrun. 3564 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3565 FastTrackUnderruns underruns = ftDump->mUnderruns; 3566 uint32_t recentFull = (underruns.mBitFields.mFull - 3567 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3568 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3569 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3570 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3571 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3572 uint32_t recentUnderruns = recentPartial + recentEmpty; 3573 track->mObservedUnderruns = underruns; 3574 // don't count underruns that occur while stopping or pausing 3575 // or stopped which can occur when flush() is called while active 3576 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3577 recentUnderruns > 0) { 3578 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3579 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3580 } 3581 3582 // This is similar to the state machine for normal tracks, 3583 // with a few modifications for fast tracks. 3584 bool isActive = true; 3585 switch (track->mState) { 3586 case TrackBase::STOPPING_1: 3587 // track stays active in STOPPING_1 state until first underrun 3588 if (recentUnderruns > 0 || track->isTerminated()) { 3589 track->mState = TrackBase::STOPPING_2; 3590 } 3591 break; 3592 case TrackBase::PAUSING: 3593 // ramp down is not yet implemented 3594 track->setPaused(); 3595 break; 3596 case TrackBase::RESUMING: 3597 // ramp up is not yet implemented 3598 track->mState = TrackBase::ACTIVE; 3599 break; 3600 case TrackBase::ACTIVE: 3601 if (recentFull > 0 || recentPartial > 0) { 3602 // track has provided at least some frames recently: reset retry count 3603 track->mRetryCount = kMaxTrackRetries; 3604 } 3605 if (recentUnderruns == 0) { 3606 // no recent underruns: stay active 3607 break; 3608 } 3609 // there has recently been an underrun of some kind 3610 if (track->sharedBuffer() == 0) { 3611 // were any of the recent underruns "empty" (no frames available)? 3612 if (recentEmpty == 0) { 3613 // no, then ignore the partial underruns as they are allowed indefinitely 3614 break; 3615 } 3616 // there has recently been an "empty" underrun: decrement the retry counter 3617 if (--(track->mRetryCount) > 0) { 3618 break; 3619 } 3620 // indicate to client process that the track was disabled because of underrun; 3621 // it will then automatically call start() when data is available 3622 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3623 // remove from active list, but state remains ACTIVE [confusing but true] 3624 isActive = false; 3625 break; 3626 } 3627 // fall through 3628 case TrackBase::STOPPING_2: 3629 case TrackBase::PAUSED: 3630 case TrackBase::STOPPED: 3631 case TrackBase::FLUSHED: // flush() while active 3632 // Check for presentation complete if track is inactive 3633 // We have consumed all the buffers of this track. 3634 // This would be incomplete if we auto-paused on underrun 3635 { 3636 size_t audioHALFrames = 3637 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3638 size_t framesWritten = mBytesWritten / mFrameSize; 3639 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3640 // track stays in active list until presentation is complete 3641 break; 3642 } 3643 } 3644 if (track->isStopping_2()) { 3645 track->mState = TrackBase::STOPPED; 3646 } 3647 if (track->isStopped()) { 3648 // Can't reset directly, as fast mixer is still polling this track 3649 // track->reset(); 3650 // So instead mark this track as needing to be reset after push with ack 3651 resetMask |= 1 << i; 3652 } 3653 isActive = false; 3654 break; 3655 case TrackBase::IDLE: 3656 default: 3657 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3658 } 3659 3660 if (isActive) { 3661 // was it previously inactive? 3662 if (!(state->mTrackMask & (1 << j))) { 3663 ExtendedAudioBufferProvider *eabp = track; 3664 VolumeProvider *vp = track; 3665 fastTrack->mBufferProvider = eabp; 3666 fastTrack->mVolumeProvider = vp; 3667 fastTrack->mChannelMask = track->mChannelMask; 3668 fastTrack->mFormat = track->mFormat; 3669 fastTrack->mGeneration++; 3670 state->mTrackMask |= 1 << j; 3671 didModify = true; 3672 // no acknowledgement required for newly active tracks 3673 } 3674 // cache the combined master volume and stream type volume for fast mixer; this 3675 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3676 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3677 ++fastTracks; 3678 } else { 3679 // was it previously active? 3680 if (state->mTrackMask & (1 << j)) { 3681 fastTrack->mBufferProvider = NULL; 3682 fastTrack->mGeneration++; 3683 state->mTrackMask &= ~(1 << j); 3684 didModify = true; 3685 // If any fast tracks were removed, we must wait for acknowledgement 3686 // because we're about to decrement the last sp<> on those tracks. 3687 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3688 } else { 3689 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3690 } 3691 tracksToRemove->add(track); 3692 // Avoids a misleading display in dumpsys 3693 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3694 } 3695 continue; 3696 } 3697 3698 { // local variable scope to avoid goto warning 3699 3700 audio_track_cblk_t* cblk = track->cblk(); 3701 3702 // The first time a track is added we wait 3703 // for all its buffers to be filled before processing it 3704 int name = track->name(); 3705 // make sure that we have enough frames to mix one full buffer. 3706 // enforce this condition only once to enable draining the buffer in case the client 3707 // app does not call stop() and relies on underrun to stop: 3708 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3709 // during last round 3710 size_t desiredFrames; 3711 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3712 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3713 3714 desiredFrames = sourceFramesNeededWithTimestretch( 3715 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3716 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3717 // add frames already consumed but not yet released by the resampler 3718 // because mAudioTrackServerProxy->framesReady() will include these frames 3719 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3720 3721 uint32_t minFrames = 1; 3722 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3723 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3724 minFrames = desiredFrames; 3725 } 3726 3727 size_t framesReady = track->framesReady(); 3728 if (ATRACE_ENABLED()) { 3729 // I wish we had formatted trace names 3730 char traceName[16]; 3731 strcpy(traceName, "nRdy"); 3732 int name = track->name(); 3733 if (AudioMixer::TRACK0 <= name && 3734 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3735 name -= AudioMixer::TRACK0; 3736 traceName[4] = (name / 10) + '0'; 3737 traceName[5] = (name % 10) + '0'; 3738 } else { 3739 traceName[4] = '?'; 3740 traceName[5] = '?'; 3741 } 3742 traceName[6] = '\0'; 3743 ATRACE_INT(traceName, framesReady); 3744 } 3745 if ((framesReady >= minFrames) && track->isReady() && 3746 !track->isPaused() && !track->isTerminated()) 3747 { 3748 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3749 3750 mixedTracks++; 3751 3752 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3753 // there is an effect chain connected to the track 3754 chain.clear(); 3755 if (track->mainBuffer() != mSinkBuffer && 3756 track->mainBuffer() != mMixerBuffer) { 3757 if (mEffectBufferEnabled) { 3758 mEffectBufferValid = true; // Later can set directly. 3759 } 3760 chain = getEffectChain_l(track->sessionId()); 3761 // Delegate volume control to effect in track effect chain if needed 3762 if (chain != 0) { 3763 tracksWithEffect++; 3764 } else { 3765 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3766 "session %d", 3767 name, track->sessionId()); 3768 } 3769 } 3770 3771 3772 int param = AudioMixer::VOLUME; 3773 if (track->mFillingUpStatus == Track::FS_FILLED) { 3774 // no ramp for the first volume setting 3775 track->mFillingUpStatus = Track::FS_ACTIVE; 3776 if (track->mState == TrackBase::RESUMING) { 3777 track->mState = TrackBase::ACTIVE; 3778 param = AudioMixer::RAMP_VOLUME; 3779 } 3780 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3781 // FIXME should not make a decision based on mServer 3782 } else if (cblk->mServer != 0) { 3783 // If the track is stopped before the first frame was mixed, 3784 // do not apply ramp 3785 param = AudioMixer::RAMP_VOLUME; 3786 } 3787 3788 // compute volume for this track 3789 uint32_t vl, vr; // in U8.24 integer format 3790 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3791 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3792 vl = vr = 0; 3793 vlf = vrf = vaf = 0.; 3794 if (track->isPausing()) { 3795 track->setPaused(); 3796 } 3797 } else { 3798 3799 // read original volumes with volume control 3800 float typeVolume = mStreamTypes[track->streamType()].volume; 3801 float v = masterVolume * typeVolume; 3802 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3803 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3804 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3805 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3806 // track volumes come from shared memory, so can't be trusted and must be clamped 3807 if (vlf > GAIN_FLOAT_UNITY) { 3808 ALOGV("Track left volume out of range: %.3g", vlf); 3809 vlf = GAIN_FLOAT_UNITY; 3810 } 3811 if (vrf > GAIN_FLOAT_UNITY) { 3812 ALOGV("Track right volume out of range: %.3g", vrf); 3813 vrf = GAIN_FLOAT_UNITY; 3814 } 3815 // now apply the master volume and stream type volume 3816 vlf *= v; 3817 vrf *= v; 3818 // assuming master volume and stream type volume each go up to 1.0, 3819 // then derive vl and vr as U8.24 versions for the effect chain 3820 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3821 vl = (uint32_t) (scaleto8_24 * vlf); 3822 vr = (uint32_t) (scaleto8_24 * vrf); 3823 // vl and vr are now in U8.24 format 3824 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3825 // send level comes from shared memory and so may be corrupt 3826 if (sendLevel > MAX_GAIN_INT) { 3827 ALOGV("Track send level out of range: %04X", sendLevel); 3828 sendLevel = MAX_GAIN_INT; 3829 } 3830 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3831 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3832 } 3833 3834 // Delegate volume control to effect in track effect chain if needed 3835 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3836 // Do not ramp volume if volume is controlled by effect 3837 param = AudioMixer::VOLUME; 3838 // Update remaining floating point volume levels 3839 vlf = (float)vl / (1 << 24); 3840 vrf = (float)vr / (1 << 24); 3841 track->mHasVolumeController = true; 3842 } else { 3843 // force no volume ramp when volume controller was just disabled or removed 3844 // from effect chain to avoid volume spike 3845 if (track->mHasVolumeController) { 3846 param = AudioMixer::VOLUME; 3847 } 3848 track->mHasVolumeController = false; 3849 } 3850 3851 // XXX: these things DON'T need to be done each time 3852 mAudioMixer->setBufferProvider(name, track); 3853 mAudioMixer->enable(name); 3854 3855 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3856 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3857 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3858 mAudioMixer->setParameter( 3859 name, 3860 AudioMixer::TRACK, 3861 AudioMixer::FORMAT, (void *)track->format()); 3862 mAudioMixer->setParameter( 3863 name, 3864 AudioMixer::TRACK, 3865 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3866 mAudioMixer->setParameter( 3867 name, 3868 AudioMixer::TRACK, 3869 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3870 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3871 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3872 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3873 if (reqSampleRate == 0) { 3874 reqSampleRate = mSampleRate; 3875 } else if (reqSampleRate > maxSampleRate) { 3876 reqSampleRate = maxSampleRate; 3877 } 3878 mAudioMixer->setParameter( 3879 name, 3880 AudioMixer::RESAMPLE, 3881 AudioMixer::SAMPLE_RATE, 3882 (void *)(uintptr_t)reqSampleRate); 3883 3884 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3885 mAudioMixer->setParameter( 3886 name, 3887 AudioMixer::TIMESTRETCH, 3888 AudioMixer::PLAYBACK_RATE, 3889 &playbackRate); 3890 3891 /* 3892 * Select the appropriate output buffer for the track. 3893 * 3894 * Tracks with effects go into their own effects chain buffer 3895 * and from there into either mEffectBuffer or mSinkBuffer. 3896 * 3897 * Other tracks can use mMixerBuffer for higher precision 3898 * channel accumulation. If this buffer is enabled 3899 * (mMixerBufferEnabled true), then selected tracks will accumulate 3900 * into it. 3901 * 3902 */ 3903 if (mMixerBufferEnabled 3904 && (track->mainBuffer() == mSinkBuffer 3905 || track->mainBuffer() == mMixerBuffer)) { 3906 mAudioMixer->setParameter( 3907 name, 3908 AudioMixer::TRACK, 3909 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3910 mAudioMixer->setParameter( 3911 name, 3912 AudioMixer::TRACK, 3913 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3914 // TODO: override track->mainBuffer()? 3915 mMixerBufferValid = true; 3916 } else { 3917 mAudioMixer->setParameter( 3918 name, 3919 AudioMixer::TRACK, 3920 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3921 mAudioMixer->setParameter( 3922 name, 3923 AudioMixer::TRACK, 3924 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3925 } 3926 mAudioMixer->setParameter( 3927 name, 3928 AudioMixer::TRACK, 3929 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3930 3931 // reset retry count 3932 track->mRetryCount = kMaxTrackRetries; 3933 3934 // If one track is ready, set the mixer ready if: 3935 // - the mixer was not ready during previous round OR 3936 // - no other track is not ready 3937 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3938 mixerStatus != MIXER_TRACKS_ENABLED) { 3939 mixerStatus = MIXER_TRACKS_READY; 3940 } 3941 } else { 3942 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3943 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3944 } 3945 // clear effect chain input buffer if an active track underruns to avoid sending 3946 // previous audio buffer again to effects 3947 chain = getEffectChain_l(track->sessionId()); 3948 if (chain != 0) { 3949 chain->clearInputBuffer(); 3950 } 3951 3952 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3953 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3954 track->isStopped() || track->isPaused()) { 3955 // We have consumed all the buffers of this track. 3956 // Remove it from the list of active tracks. 3957 // TODO: use actual buffer filling status instead of latency when available from 3958 // audio HAL 3959 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3960 size_t framesWritten = mBytesWritten / mFrameSize; 3961 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3962 if (track->isStopped()) { 3963 track->reset(); 3964 } 3965 tracksToRemove->add(track); 3966 } 3967 } else { 3968 // No buffers for this track. Give it a few chances to 3969 // fill a buffer, then remove it from active list. 3970 if (--(track->mRetryCount) <= 0) { 3971 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3972 tracksToRemove->add(track); 3973 // indicate to client process that the track was disabled because of underrun; 3974 // it will then automatically call start() when data is available 3975 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3976 // If one track is not ready, mark the mixer also not ready if: 3977 // - the mixer was ready during previous round OR 3978 // - no other track is ready 3979 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3980 mixerStatus != MIXER_TRACKS_READY) { 3981 mixerStatus = MIXER_TRACKS_ENABLED; 3982 } 3983 } 3984 mAudioMixer->disable(name); 3985 } 3986 3987 } // local variable scope to avoid goto warning 3988track_is_ready: ; 3989 3990 } 3991 3992 // Push the new FastMixer state if necessary 3993 bool pauseAudioWatchdog = false; 3994 if (didModify) { 3995 state->mFastTracksGen++; 3996 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3997 if (kUseFastMixer == FastMixer_Dynamic && 3998 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3999 state->mCommand = FastMixerState::COLD_IDLE; 4000 state->mColdFutexAddr = &mFastMixerFutex; 4001 state->mColdGen++; 4002 mFastMixerFutex = 0; 4003 if (kUseFastMixer == FastMixer_Dynamic) { 4004 mNormalSink = mOutputSink; 4005 } 4006 // If we go into cold idle, need to wait for acknowledgement 4007 // so that fast mixer stops doing I/O. 4008 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4009 pauseAudioWatchdog = true; 4010 } 4011 } 4012 if (sq != NULL) { 4013 sq->end(didModify); 4014 sq->push(block); 4015 } 4016#ifdef AUDIO_WATCHDOG 4017 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4018 mAudioWatchdog->pause(); 4019 } 4020#endif 4021 4022 // Now perform the deferred reset on fast tracks that have stopped 4023 while (resetMask != 0) { 4024 size_t i = __builtin_ctz(resetMask); 4025 ALOG_ASSERT(i < count); 4026 resetMask &= ~(1 << i); 4027 sp<Track> t = mActiveTracks[i].promote(); 4028 if (t == 0) { 4029 continue; 4030 } 4031 Track* track = t.get(); 4032 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4033 track->reset(); 4034 } 4035 4036 // remove all the tracks that need to be... 4037 removeTracks_l(*tracksToRemove); 4038 4039 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4040 mEffectBufferValid = true; 4041 } 4042 4043 if (mEffectBufferValid) { 4044 // as long as there are effects we should clear the effects buffer, to avoid 4045 // passing a non-clean buffer to the effect chain 4046 memset(mEffectBuffer, 0, mEffectBufferSize); 4047 } 4048 // sink or mix buffer must be cleared if all tracks are connected to an 4049 // effect chain as in this case the mixer will not write to the sink or mix buffer 4050 // and track effects will accumulate into it 4051 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4052 (mixedTracks == 0 && fastTracks > 0))) { 4053 // FIXME as a performance optimization, should remember previous zero status 4054 if (mMixerBufferValid) { 4055 memset(mMixerBuffer, 0, mMixerBufferSize); 4056 // TODO: In testing, mSinkBuffer below need not be cleared because 4057 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4058 // after mixing. 4059 // 4060 // To enforce this guarantee: 4061 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4062 // (mixedTracks == 0 && fastTracks > 0)) 4063 // must imply MIXER_TRACKS_READY. 4064 // Later, we may clear buffers regardless, and skip much of this logic. 4065 } 4066 // FIXME as a performance optimization, should remember previous zero status 4067 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4068 } 4069 4070 // if any fast tracks, then status is ready 4071 mMixerStatusIgnoringFastTracks = mixerStatus; 4072 if (fastTracks > 0) { 4073 mixerStatus = MIXER_TRACKS_READY; 4074 } 4075 return mixerStatus; 4076} 4077 4078// getTrackName_l() must be called with ThreadBase::mLock held 4079int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4080 audio_format_t format, int sessionId) 4081{ 4082 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4083} 4084 4085// deleteTrackName_l() must be called with ThreadBase::mLock held 4086void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4087{ 4088 ALOGV("remove track (%d) and delete from mixer", name); 4089 mAudioMixer->deleteTrackName(name); 4090} 4091 4092// checkForNewParameter_l() must be called with ThreadBase::mLock held 4093bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4094 status_t& status) 4095{ 4096 bool reconfig = false; 4097 4098 status = NO_ERROR; 4099 4100 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4101 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4102 if (mFastMixer != 0) { 4103 FastMixerStateQueue *sq = mFastMixer->sq(); 4104 FastMixerState *state = sq->begin(); 4105 if (!(state->mCommand & FastMixerState::IDLE)) { 4106 previousCommand = state->mCommand; 4107 state->mCommand = FastMixerState::HOT_IDLE; 4108 sq->end(); 4109 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4110 } else { 4111 sq->end(false /*didModify*/); 4112 } 4113 } 4114 4115 AudioParameter param = AudioParameter(keyValuePair); 4116 int value; 4117 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4118 reconfig = true; 4119 } 4120 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4121 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4122 status = BAD_VALUE; 4123 } else { 4124 // no need to save value, since it's constant 4125 reconfig = true; 4126 } 4127 } 4128 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4129 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4130 status = BAD_VALUE; 4131 } else { 4132 // no need to save value, since it's constant 4133 reconfig = true; 4134 } 4135 } 4136 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4137 // do not accept frame count changes if tracks are open as the track buffer 4138 // size depends on frame count and correct behavior would not be guaranteed 4139 // if frame count is changed after track creation 4140 if (!mTracks.isEmpty()) { 4141 status = INVALID_OPERATION; 4142 } else { 4143 reconfig = true; 4144 } 4145 } 4146 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4147#ifdef ADD_BATTERY_DATA 4148 // when changing the audio output device, call addBatteryData to notify 4149 // the change 4150 if (mOutDevice != value) { 4151 uint32_t params = 0; 4152 // check whether speaker is on 4153 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4154 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4155 } 4156 4157 audio_devices_t deviceWithoutSpeaker 4158 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4159 // check if any other device (except speaker) is on 4160 if (value & deviceWithoutSpeaker) { 4161 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4162 } 4163 4164 if (params != 0) { 4165 addBatteryData(params); 4166 } 4167 } 4168#endif 4169 4170 // forward device change to effects that have requested to be 4171 // aware of attached audio device. 4172 if (value != AUDIO_DEVICE_NONE) { 4173 mOutDevice = value; 4174 for (size_t i = 0; i < mEffectChains.size(); i++) { 4175 mEffectChains[i]->setDevice_l(mOutDevice); 4176 } 4177 } 4178 } 4179 4180 if (status == NO_ERROR) { 4181 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4182 keyValuePair.string()); 4183 if (!mStandby && status == INVALID_OPERATION) { 4184 mOutput->standby(); 4185 mStandby = true; 4186 mBytesWritten = 0; 4187 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4188 keyValuePair.string()); 4189 } 4190 if (status == NO_ERROR && reconfig) { 4191 readOutputParameters_l(); 4192 delete mAudioMixer; 4193 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4194 for (size_t i = 0; i < mTracks.size() ; i++) { 4195 int name = getTrackName_l(mTracks[i]->mChannelMask, 4196 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4197 if (name < 0) { 4198 break; 4199 } 4200 mTracks[i]->mName = name; 4201 } 4202 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4203 } 4204 } 4205 4206 if (!(previousCommand & FastMixerState::IDLE)) { 4207 ALOG_ASSERT(mFastMixer != 0); 4208 FastMixerStateQueue *sq = mFastMixer->sq(); 4209 FastMixerState *state = sq->begin(); 4210 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4211 state->mCommand = previousCommand; 4212 sq->end(); 4213 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4214 } 4215 4216 return reconfig; 4217} 4218 4219 4220void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4221{ 4222 const size_t SIZE = 256; 4223 char buffer[SIZE]; 4224 String8 result; 4225 4226 PlaybackThread::dumpInternals(fd, args); 4227 4228 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4229 4230 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4231 const FastMixerDumpState copy(mFastMixerDumpState); 4232 copy.dump(fd); 4233 4234#ifdef STATE_QUEUE_DUMP 4235 // Similar for state queue 4236 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4237 observerCopy.dump(fd); 4238 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4239 mutatorCopy.dump(fd); 4240#endif 4241 4242#ifdef TEE_SINK 4243 // Write the tee output to a .wav file 4244 dumpTee(fd, mTeeSource, mId); 4245#endif 4246 4247#ifdef AUDIO_WATCHDOG 4248 if (mAudioWatchdog != 0) { 4249 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4250 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4251 wdCopy.dump(fd); 4252 } 4253#endif 4254} 4255 4256uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4257{ 4258 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4259} 4260 4261uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4262{ 4263 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4264} 4265 4266void AudioFlinger::MixerThread::cacheParameters_l() 4267{ 4268 PlaybackThread::cacheParameters_l(); 4269 4270 // FIXME: Relaxed timing because of a certain device that can't meet latency 4271 // Should be reduced to 2x after the vendor fixes the driver issue 4272 // increase threshold again due to low power audio mode. The way this warning 4273 // threshold is calculated and its usefulness should be reconsidered anyway. 4274 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4275} 4276 4277// ---------------------------------------------------------------------------- 4278 4279AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4280 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4281 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4282 // mLeftVolFloat, mRightVolFloat 4283{ 4284} 4285 4286AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4287 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4288 ThreadBase::type_t type) 4289 : PlaybackThread(audioFlinger, output, id, device, type) 4290 // mLeftVolFloat, mRightVolFloat 4291{ 4292} 4293 4294AudioFlinger::DirectOutputThread::~DirectOutputThread() 4295{ 4296} 4297 4298void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4299{ 4300 audio_track_cblk_t* cblk = track->cblk(); 4301 float left, right; 4302 4303 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4304 left = right = 0; 4305 } else { 4306 float typeVolume = mStreamTypes[track->streamType()].volume; 4307 float v = mMasterVolume * typeVolume; 4308 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4309 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4310 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4311 if (left > GAIN_FLOAT_UNITY) { 4312 left = GAIN_FLOAT_UNITY; 4313 } 4314 left *= v; 4315 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4316 if (right > GAIN_FLOAT_UNITY) { 4317 right = GAIN_FLOAT_UNITY; 4318 } 4319 right *= v; 4320 } 4321 4322 if (lastTrack) { 4323 if (left != mLeftVolFloat || right != mRightVolFloat) { 4324 mLeftVolFloat = left; 4325 mRightVolFloat = right; 4326 4327 // Convert volumes from float to 8.24 4328 uint32_t vl = (uint32_t)(left * (1 << 24)); 4329 uint32_t vr = (uint32_t)(right * (1 << 24)); 4330 4331 // Delegate volume control to effect in track effect chain if needed 4332 // only one effect chain can be present on DirectOutputThread, so if 4333 // there is one, the track is connected to it 4334 if (!mEffectChains.isEmpty()) { 4335 mEffectChains[0]->setVolume_l(&vl, &vr); 4336 left = (float)vl / (1 << 24); 4337 right = (float)vr / (1 << 24); 4338 } 4339 if (mOutput->stream->set_volume) { 4340 mOutput->stream->set_volume(mOutput->stream, left, right); 4341 } 4342 } 4343 } 4344} 4345 4346 4347AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4348 Vector< sp<Track> > *tracksToRemove 4349) 4350{ 4351 size_t count = mActiveTracks.size(); 4352 mixer_state mixerStatus = MIXER_IDLE; 4353 bool doHwPause = false; 4354 bool doHwResume = false; 4355 bool flushPending = false; 4356 4357 // find out which tracks need to be processed 4358 for (size_t i = 0; i < count; i++) { 4359 sp<Track> t = mActiveTracks[i].promote(); 4360 // The track died recently 4361 if (t == 0) { 4362 continue; 4363 } 4364 4365 Track* const track = t.get(); 4366 audio_track_cblk_t* cblk = track->cblk(); 4367 // Only consider last track started for volume and mixer state control. 4368 // In theory an older track could underrun and restart after the new one starts 4369 // but as we only care about the transition phase between two tracks on a 4370 // direct output, it is not a problem to ignore the underrun case. 4371 sp<Track> l = mLatestActiveTrack.promote(); 4372 bool last = l.get() == track; 4373 4374 if (mHwSupportsPause && track->isPausing()) { 4375 track->setPaused(); 4376 if (last && !mHwPaused) { 4377 doHwPause = true; 4378 mHwPaused = true; 4379 } 4380 tracksToRemove->add(track); 4381 } else if (track->isFlushPending()) { 4382 track->flushAck(); 4383 if (last) { 4384 flushPending = true; 4385 } 4386 } else if (mHwSupportsPause && track->isResumePending()){ 4387 track->resumeAck(); 4388 if (last) { 4389 if (mHwPaused) { 4390 doHwResume = true; 4391 mHwPaused = false; 4392 } 4393 } 4394 } 4395 4396 // The first time a track is added we wait 4397 // for all its buffers to be filled before processing it. 4398 // Allow draining the buffer in case the client 4399 // app does not call stop() and relies on underrun to stop: 4400 // hence the test on (track->mRetryCount > 1). 4401 // If retryCount<=1 then track is about to underrun and be removed. 4402 uint32_t minFrames; 4403 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4404 && (track->mRetryCount > 1)) { 4405 minFrames = mNormalFrameCount; 4406 } else { 4407 minFrames = 1; 4408 } 4409 4410 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4411 !track->isStopping_2() && !track->isStopped()) 4412 { 4413 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4414 4415 if (track->mFillingUpStatus == Track::FS_FILLED) { 4416 track->mFillingUpStatus = Track::FS_ACTIVE; 4417 // make sure processVolume_l() will apply new volume even if 0 4418 mLeftVolFloat = mRightVolFloat = -1.0; 4419 if (!mHwSupportsPause) { 4420 track->resumeAck(); 4421 } 4422 } 4423 4424 // compute volume for this track 4425 processVolume_l(track, last); 4426 if (last) { 4427 // reset retry count 4428 track->mRetryCount = kMaxTrackRetriesDirect; 4429 mActiveTrack = t; 4430 mixerStatus = MIXER_TRACKS_READY; 4431 if (usesHwAvSync() && mHwPaused) { 4432 doHwResume = true; 4433 mHwPaused = false; 4434 } 4435 } 4436 } else { 4437 // clear effect chain input buffer if the last active track started underruns 4438 // to avoid sending previous audio buffer again to effects 4439 if (!mEffectChains.isEmpty() && last) { 4440 mEffectChains[0]->clearInputBuffer(); 4441 } 4442 if (track->isStopping_1()) { 4443 track->mState = TrackBase::STOPPING_2; 4444 if (last && mHwPaused) { 4445 doHwResume = true; 4446 mHwPaused = false; 4447 } 4448 } 4449 if ((track->sharedBuffer() != 0) || track->isStopped() || 4450 track->isStopping_2() || track->isPaused()) { 4451 // We have consumed all the buffers of this track. 4452 // Remove it from the list of active tracks. 4453 size_t audioHALFrames; 4454 if (audio_is_linear_pcm(mFormat)) { 4455 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4456 } else { 4457 audioHALFrames = 0; 4458 } 4459 4460 size_t framesWritten = mBytesWritten / mFrameSize; 4461 if (mStandby || !last || 4462 track->presentationComplete(framesWritten, audioHALFrames)) { 4463 if (track->isStopping_2()) { 4464 track->mState = TrackBase::STOPPED; 4465 } 4466 if (track->isStopped()) { 4467 track->reset(); 4468 } 4469 tracksToRemove->add(track); 4470 } 4471 } else { 4472 // No buffers for this track. Give it a few chances to 4473 // fill a buffer, then remove it from active list. 4474 // Only consider last track started for mixer state control 4475 if (--(track->mRetryCount) <= 0) { 4476 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4477 tracksToRemove->add(track); 4478 // indicate to client process that the track was disabled because of underrun; 4479 // it will then automatically call start() when data is available 4480 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4481 } else if (last) { 4482 mixerStatus = MIXER_TRACKS_ENABLED; 4483 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4484 doHwPause = true; 4485 mHwPaused = true; 4486 } 4487 } 4488 } 4489 } 4490 } 4491 4492 // if an active track did not command a flush, check for pending flush on stopped tracks 4493 if (!flushPending) { 4494 for (size_t i = 0; i < mTracks.size(); i++) { 4495 if (mTracks[i]->isFlushPending()) { 4496 mTracks[i]->flushAck(); 4497 flushPending = true; 4498 } 4499 } 4500 } 4501 4502 // make sure the pause/flush/resume sequence is executed in the right order. 4503 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4504 // before flush and then resume HW. This can happen in case of pause/flush/resume 4505 // if resume is received before pause is executed. 4506 if (mHwSupportsPause && !mStandby && 4507 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4508 mOutput->stream->pause(mOutput->stream); 4509 } 4510 if (flushPending) { 4511 flushHw_l(); 4512 } 4513 if (mHwSupportsPause && !mStandby && doHwResume) { 4514 mOutput->stream->resume(mOutput->stream); 4515 } 4516 // remove all the tracks that need to be... 4517 removeTracks_l(*tracksToRemove); 4518 4519 return mixerStatus; 4520} 4521 4522void AudioFlinger::DirectOutputThread::threadLoop_mix() 4523{ 4524 size_t frameCount = mFrameCount; 4525 int8_t *curBuf = (int8_t *)mSinkBuffer; 4526 // output audio to hardware 4527 while (frameCount) { 4528 AudioBufferProvider::Buffer buffer; 4529 buffer.frameCount = frameCount; 4530 status_t status = mActiveTrack->getNextBuffer(&buffer); 4531 if (status != NO_ERROR || buffer.raw == NULL) { 4532 memset(curBuf, 0, frameCount * mFrameSize); 4533 break; 4534 } 4535 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4536 frameCount -= buffer.frameCount; 4537 curBuf += buffer.frameCount * mFrameSize; 4538 mActiveTrack->releaseBuffer(&buffer); 4539 } 4540 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4541 sleepTime = 0; 4542 standbyTime = systemTime() + standbyDelay; 4543 mActiveTrack.clear(); 4544} 4545 4546void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4547{ 4548 // do not write to HAL when paused 4549 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4550 sleepTime = idleSleepTime; 4551 return; 4552 } 4553 if (sleepTime == 0) { 4554 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4555 sleepTime = activeSleepTime; 4556 } else { 4557 sleepTime = idleSleepTime; 4558 } 4559 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4560 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4561 sleepTime = 0; 4562 } 4563} 4564 4565void AudioFlinger::DirectOutputThread::threadLoop_exit() 4566{ 4567 { 4568 Mutex::Autolock _l(mLock); 4569 bool flushPending = false; 4570 for (size_t i = 0; i < mTracks.size(); i++) { 4571 if (mTracks[i]->isFlushPending()) { 4572 mTracks[i]->flushAck(); 4573 flushPending = true; 4574 } 4575 } 4576 if (flushPending) { 4577 flushHw_l(); 4578 } 4579 } 4580 PlaybackThread::threadLoop_exit(); 4581} 4582 4583// must be called with thread mutex locked 4584bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4585{ 4586 bool trackPaused = false; 4587 bool trackStopped = false; 4588 4589 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4590 // after a timeout and we will enter standby then. 4591 if (mTracks.size() > 0) { 4592 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4593 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4594 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4595 } 4596 4597 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped)); 4598} 4599 4600// getTrackName_l() must be called with ThreadBase::mLock held 4601int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4602 audio_format_t format __unused, int sessionId __unused) 4603{ 4604 return 0; 4605} 4606 4607// deleteTrackName_l() must be called with ThreadBase::mLock held 4608void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4609{ 4610} 4611 4612// checkForNewParameter_l() must be called with ThreadBase::mLock held 4613bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4614 status_t& status) 4615{ 4616 bool reconfig = false; 4617 4618 status = NO_ERROR; 4619 4620 AudioParameter param = AudioParameter(keyValuePair); 4621 int value; 4622 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4623 // forward device change to effects that have requested to be 4624 // aware of attached audio device. 4625 if (value != AUDIO_DEVICE_NONE) { 4626 mOutDevice = value; 4627 for (size_t i = 0; i < mEffectChains.size(); i++) { 4628 mEffectChains[i]->setDevice_l(mOutDevice); 4629 } 4630 } 4631 } 4632 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4633 // do not accept frame count changes if tracks are open as the track buffer 4634 // size depends on frame count and correct behavior would not be garantied 4635 // if frame count is changed after track creation 4636 if (!mTracks.isEmpty()) { 4637 status = INVALID_OPERATION; 4638 } else { 4639 reconfig = true; 4640 } 4641 } 4642 if (status == NO_ERROR) { 4643 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4644 keyValuePair.string()); 4645 if (!mStandby && status == INVALID_OPERATION) { 4646 mOutput->standby(); 4647 mStandby = true; 4648 mBytesWritten = 0; 4649 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4650 keyValuePair.string()); 4651 } 4652 if (status == NO_ERROR && reconfig) { 4653 readOutputParameters_l(); 4654 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4655 } 4656 } 4657 4658 return reconfig; 4659} 4660 4661uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4662{ 4663 uint32_t time; 4664 if (audio_is_linear_pcm(mFormat)) { 4665 time = PlaybackThread::activeSleepTimeUs(); 4666 } else { 4667 time = 10000; 4668 } 4669 return time; 4670} 4671 4672uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4673{ 4674 uint32_t time; 4675 if (audio_is_linear_pcm(mFormat)) { 4676 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4677 } else { 4678 time = 10000; 4679 } 4680 return time; 4681} 4682 4683uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4684{ 4685 uint32_t time; 4686 if (audio_is_linear_pcm(mFormat)) { 4687 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4688 } else { 4689 time = 10000; 4690 } 4691 return time; 4692} 4693 4694void AudioFlinger::DirectOutputThread::cacheParameters_l() 4695{ 4696 PlaybackThread::cacheParameters_l(); 4697 4698 // use shorter standby delay as on normal output to release 4699 // hardware resources as soon as possible 4700 // no delay on outputs with HW A/V sync 4701 if (usesHwAvSync()) { 4702 standbyDelay = 0; 4703 } else if (audio_is_linear_pcm(mFormat)) { 4704 standbyDelay = microseconds(activeSleepTime*2); 4705 } else { 4706 standbyDelay = kOffloadStandbyDelayNs; 4707 } 4708} 4709 4710void AudioFlinger::DirectOutputThread::flushHw_l() 4711{ 4712 mOutput->flush(); 4713 mHwPaused = false; 4714} 4715 4716// ---------------------------------------------------------------------------- 4717 4718AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4719 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4720 : Thread(false /*canCallJava*/), 4721 mPlaybackThread(playbackThread), 4722 mWriteAckSequence(0), 4723 mDrainSequence(0) 4724{ 4725} 4726 4727AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4728{ 4729} 4730 4731void AudioFlinger::AsyncCallbackThread::onFirstRef() 4732{ 4733 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4734} 4735 4736bool AudioFlinger::AsyncCallbackThread::threadLoop() 4737{ 4738 while (!exitPending()) { 4739 uint32_t writeAckSequence; 4740 uint32_t drainSequence; 4741 4742 { 4743 Mutex::Autolock _l(mLock); 4744 while (!((mWriteAckSequence & 1) || 4745 (mDrainSequence & 1) || 4746 exitPending())) { 4747 mWaitWorkCV.wait(mLock); 4748 } 4749 4750 if (exitPending()) { 4751 break; 4752 } 4753 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4754 mWriteAckSequence, mDrainSequence); 4755 writeAckSequence = mWriteAckSequence; 4756 mWriteAckSequence &= ~1; 4757 drainSequence = mDrainSequence; 4758 mDrainSequence &= ~1; 4759 } 4760 { 4761 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4762 if (playbackThread != 0) { 4763 if (writeAckSequence & 1) { 4764 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4765 } 4766 if (drainSequence & 1) { 4767 playbackThread->resetDraining(drainSequence >> 1); 4768 } 4769 } 4770 } 4771 } 4772 return false; 4773} 4774 4775void AudioFlinger::AsyncCallbackThread::exit() 4776{ 4777 ALOGV("AsyncCallbackThread::exit"); 4778 Mutex::Autolock _l(mLock); 4779 requestExit(); 4780 mWaitWorkCV.broadcast(); 4781} 4782 4783void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4784{ 4785 Mutex::Autolock _l(mLock); 4786 // bit 0 is cleared 4787 mWriteAckSequence = sequence << 1; 4788} 4789 4790void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4791{ 4792 Mutex::Autolock _l(mLock); 4793 // ignore unexpected callbacks 4794 if (mWriteAckSequence & 2) { 4795 mWriteAckSequence |= 1; 4796 mWaitWorkCV.signal(); 4797 } 4798} 4799 4800void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4801{ 4802 Mutex::Autolock _l(mLock); 4803 // bit 0 is cleared 4804 mDrainSequence = sequence << 1; 4805} 4806 4807void AudioFlinger::AsyncCallbackThread::resetDraining() 4808{ 4809 Mutex::Autolock _l(mLock); 4810 // ignore unexpected callbacks 4811 if (mDrainSequence & 2) { 4812 mDrainSequence |= 1; 4813 mWaitWorkCV.signal(); 4814 } 4815} 4816 4817 4818// ---------------------------------------------------------------------------- 4819AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4820 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4821 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4822 mPausedBytesRemaining(0) 4823{ 4824 //FIXME: mStandby should be set to true by ThreadBase constructor 4825 mStandby = true; 4826} 4827 4828void AudioFlinger::OffloadThread::threadLoop_exit() 4829{ 4830 if (mFlushPending || mHwPaused) { 4831 // If a flush is pending or track was paused, just discard buffered data 4832 flushHw_l(); 4833 } else { 4834 mMixerStatus = MIXER_DRAIN_ALL; 4835 threadLoop_drain(); 4836 } 4837 if (mUseAsyncWrite) { 4838 ALOG_ASSERT(mCallbackThread != 0); 4839 mCallbackThread->exit(); 4840 } 4841 PlaybackThread::threadLoop_exit(); 4842} 4843 4844AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4845 Vector< sp<Track> > *tracksToRemove 4846) 4847{ 4848 size_t count = mActiveTracks.size(); 4849 4850 mixer_state mixerStatus = MIXER_IDLE; 4851 bool doHwPause = false; 4852 bool doHwResume = false; 4853 4854 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4855 4856 // find out which tracks need to be processed 4857 for (size_t i = 0; i < count; i++) { 4858 sp<Track> t = mActiveTracks[i].promote(); 4859 // The track died recently 4860 if (t == 0) { 4861 continue; 4862 } 4863 Track* const track = t.get(); 4864 audio_track_cblk_t* cblk = track->cblk(); 4865 // Only consider last track started for volume and mixer state control. 4866 // In theory an older track could underrun and restart after the new one starts 4867 // but as we only care about the transition phase between two tracks on a 4868 // direct output, it is not a problem to ignore the underrun case. 4869 sp<Track> l = mLatestActiveTrack.promote(); 4870 bool last = l.get() == track; 4871 4872 if (track->isInvalid()) { 4873 ALOGW("An invalidated track shouldn't be in active list"); 4874 tracksToRemove->add(track); 4875 continue; 4876 } 4877 4878 if (track->mState == TrackBase::IDLE) { 4879 ALOGW("An idle track shouldn't be in active list"); 4880 continue; 4881 } 4882 4883 if (track->isPausing()) { 4884 track->setPaused(); 4885 if (last) { 4886 if (!mHwPaused) { 4887 doHwPause = true; 4888 mHwPaused = true; 4889 } 4890 // If we were part way through writing the mixbuffer to 4891 // the HAL we must save this until we resume 4892 // BUG - this will be wrong if a different track is made active, 4893 // in that case we want to discard the pending data in the 4894 // mixbuffer and tell the client to present it again when the 4895 // track is resumed 4896 mPausedWriteLength = mCurrentWriteLength; 4897 mPausedBytesRemaining = mBytesRemaining; 4898 mBytesRemaining = 0; // stop writing 4899 } 4900 tracksToRemove->add(track); 4901 } else if (track->isFlushPending()) { 4902 track->flushAck(); 4903 if (last) { 4904 mFlushPending = true; 4905 } 4906 } else if (track->isResumePending()){ 4907 track->resumeAck(); 4908 if (last) { 4909 if (mPausedBytesRemaining) { 4910 // Need to continue write that was interrupted 4911 mCurrentWriteLength = mPausedWriteLength; 4912 mBytesRemaining = mPausedBytesRemaining; 4913 mPausedBytesRemaining = 0; 4914 } 4915 if (mHwPaused) { 4916 doHwResume = true; 4917 mHwPaused = false; 4918 // threadLoop_mix() will handle the case that we need to 4919 // resume an interrupted write 4920 } 4921 // enable write to audio HAL 4922 sleepTime = 0; 4923 4924 // Do not handle new data in this iteration even if track->framesReady() 4925 mixerStatus = MIXER_TRACKS_ENABLED; 4926 } 4927 } else if (track->framesReady() && track->isReady() && 4928 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4929 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4930 if (track->mFillingUpStatus == Track::FS_FILLED) { 4931 track->mFillingUpStatus = Track::FS_ACTIVE; 4932 // make sure processVolume_l() will apply new volume even if 0 4933 mLeftVolFloat = mRightVolFloat = -1.0; 4934 } 4935 4936 if (last) { 4937 sp<Track> previousTrack = mPreviousTrack.promote(); 4938 if (previousTrack != 0) { 4939 if (track != previousTrack.get()) { 4940 // Flush any data still being written from last track 4941 mBytesRemaining = 0; 4942 if (mPausedBytesRemaining) { 4943 // Last track was paused so we also need to flush saved 4944 // mixbuffer state and invalidate track so that it will 4945 // re-submit that unwritten data when it is next resumed 4946 mPausedBytesRemaining = 0; 4947 // Invalidate is a bit drastic - would be more efficient 4948 // to have a flag to tell client that some of the 4949 // previously written data was lost 4950 previousTrack->invalidate(); 4951 } 4952 // flush data already sent to the DSP if changing audio session as audio 4953 // comes from a different source. Also invalidate previous track to force a 4954 // seek when resuming. 4955 if (previousTrack->sessionId() != track->sessionId()) { 4956 previousTrack->invalidate(); 4957 } 4958 } 4959 } 4960 mPreviousTrack = track; 4961 // reset retry count 4962 track->mRetryCount = kMaxTrackRetriesOffload; 4963 mActiveTrack = t; 4964 mixerStatus = MIXER_TRACKS_READY; 4965 } 4966 } else { 4967 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4968 if (track->isStopping_1()) { 4969 // Hardware buffer can hold a large amount of audio so we must 4970 // wait for all current track's data to drain before we say 4971 // that the track is stopped. 4972 if (mBytesRemaining == 0) { 4973 // Only start draining when all data in mixbuffer 4974 // has been written 4975 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4976 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4977 // do not drain if no data was ever sent to HAL (mStandby == true) 4978 if (last && !mStandby) { 4979 // do not modify drain sequence if we are already draining. This happens 4980 // when resuming from pause after drain. 4981 if ((mDrainSequence & 1) == 0) { 4982 sleepTime = 0; 4983 standbyTime = systemTime() + standbyDelay; 4984 mixerStatus = MIXER_DRAIN_TRACK; 4985 mDrainSequence += 2; 4986 } 4987 if (mHwPaused) { 4988 // It is possible to move from PAUSED to STOPPING_1 without 4989 // a resume so we must ensure hardware is running 4990 doHwResume = true; 4991 mHwPaused = false; 4992 } 4993 } 4994 } 4995 } else if (track->isStopping_2()) { 4996 // Drain has completed or we are in standby, signal presentation complete 4997 if (!(mDrainSequence & 1) || !last || mStandby) { 4998 track->mState = TrackBase::STOPPED; 4999 size_t audioHALFrames = 5000 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5001 size_t framesWritten = 5002 mBytesWritten / mOutput->getFrameSize(); 5003 track->presentationComplete(framesWritten, audioHALFrames); 5004 track->reset(); 5005 tracksToRemove->add(track); 5006 } 5007 } else { 5008 // No buffers for this track. Give it a few chances to 5009 // fill a buffer, then remove it from active list. 5010 if (--(track->mRetryCount) <= 0) { 5011 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5012 track->name()); 5013 tracksToRemove->add(track); 5014 // indicate to client process that the track was disabled because of underrun; 5015 // it will then automatically call start() when data is available 5016 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5017 } else if (last){ 5018 mixerStatus = MIXER_TRACKS_ENABLED; 5019 } 5020 } 5021 } 5022 // compute volume for this track 5023 processVolume_l(track, last); 5024 } 5025 5026 // make sure the pause/flush/resume sequence is executed in the right order. 5027 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5028 // before flush and then resume HW. This can happen in case of pause/flush/resume 5029 // if resume is received before pause is executed. 5030 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5031 mOutput->stream->pause(mOutput->stream); 5032 } 5033 if (mFlushPending) { 5034 flushHw_l(); 5035 mFlushPending = false; 5036 } 5037 if (!mStandby && doHwResume) { 5038 mOutput->stream->resume(mOutput->stream); 5039 } 5040 5041 // remove all the tracks that need to be... 5042 removeTracks_l(*tracksToRemove); 5043 5044 return mixerStatus; 5045} 5046 5047// must be called with thread mutex locked 5048bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5049{ 5050 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5051 mWriteAckSequence, mDrainSequence); 5052 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5053 return true; 5054 } 5055 return false; 5056} 5057 5058bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5059{ 5060 Mutex::Autolock _l(mLock); 5061 return waitingAsyncCallback_l(); 5062} 5063 5064void AudioFlinger::OffloadThread::flushHw_l() 5065{ 5066 DirectOutputThread::flushHw_l(); 5067 // Flush anything still waiting in the mixbuffer 5068 mCurrentWriteLength = 0; 5069 mBytesRemaining = 0; 5070 mPausedWriteLength = 0; 5071 mPausedBytesRemaining = 0; 5072 5073 if (mUseAsyncWrite) { 5074 // discard any pending drain or write ack by incrementing sequence 5075 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5076 mDrainSequence = (mDrainSequence + 2) & ~1; 5077 ALOG_ASSERT(mCallbackThread != 0); 5078 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5079 mCallbackThread->setDraining(mDrainSequence); 5080 } 5081} 5082 5083void AudioFlinger::OffloadThread::onAddNewTrack_l() 5084{ 5085 sp<Track> previousTrack = mPreviousTrack.promote(); 5086 sp<Track> latestTrack = mLatestActiveTrack.promote(); 5087 5088 if (previousTrack != 0 && latestTrack != 0 && 5089 (previousTrack->sessionId() != latestTrack->sessionId())) { 5090 mFlushPending = true; 5091 } 5092 PlaybackThread::onAddNewTrack_l(); 5093} 5094 5095// ---------------------------------------------------------------------------- 5096 5097AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5098 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 5099 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5100 DUPLICATING), 5101 mWaitTimeMs(UINT_MAX) 5102{ 5103 addOutputTrack(mainThread); 5104} 5105 5106AudioFlinger::DuplicatingThread::~DuplicatingThread() 5107{ 5108 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5109 mOutputTracks[i]->destroy(); 5110 } 5111} 5112 5113void AudioFlinger::DuplicatingThread::threadLoop_mix() 5114{ 5115 // mix buffers... 5116 if (outputsReady(outputTracks)) { 5117 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5118 } else { 5119 if (mMixerBufferValid) { 5120 memset(mMixerBuffer, 0, mMixerBufferSize); 5121 } else { 5122 memset(mSinkBuffer, 0, mSinkBufferSize); 5123 } 5124 } 5125 sleepTime = 0; 5126 writeFrames = mNormalFrameCount; 5127 mCurrentWriteLength = mSinkBufferSize; 5128 standbyTime = systemTime() + standbyDelay; 5129} 5130 5131void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5132{ 5133 if (sleepTime == 0) { 5134 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5135 sleepTime = activeSleepTime; 5136 } else { 5137 sleepTime = idleSleepTime; 5138 } 5139 } else if (mBytesWritten != 0) { 5140 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5141 writeFrames = mNormalFrameCount; 5142 memset(mSinkBuffer, 0, mSinkBufferSize); 5143 } else { 5144 // flush remaining overflow buffers in output tracks 5145 writeFrames = 0; 5146 } 5147 sleepTime = 0; 5148 } 5149} 5150 5151ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5152{ 5153 for (size_t i = 0; i < outputTracks.size(); i++) { 5154 outputTracks[i]->write(mSinkBuffer, writeFrames); 5155 } 5156 mStandby = false; 5157 return (ssize_t)mSinkBufferSize; 5158} 5159 5160void AudioFlinger::DuplicatingThread::threadLoop_standby() 5161{ 5162 // DuplicatingThread implements standby by stopping all tracks 5163 for (size_t i = 0; i < outputTracks.size(); i++) { 5164 outputTracks[i]->stop(); 5165 } 5166} 5167 5168void AudioFlinger::DuplicatingThread::saveOutputTracks() 5169{ 5170 outputTracks = mOutputTracks; 5171} 5172 5173void AudioFlinger::DuplicatingThread::clearOutputTracks() 5174{ 5175 outputTracks.clear(); 5176} 5177 5178void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5179{ 5180 Mutex::Autolock _l(mLock); 5181 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5182 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5183 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5184 const size_t frameCount = 5185 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5186 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5187 // from different OutputTracks and their associated MixerThreads (e.g. one may 5188 // nearly empty and the other may be dropping data). 5189 5190 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5191 this, 5192 mSampleRate, 5193 mFormat, 5194 mChannelMask, 5195 frameCount, 5196 IPCThreadState::self()->getCallingUid()); 5197 if (outputTrack->cblk() != NULL) { 5198 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5199 mOutputTracks.add(outputTrack); 5200 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5201 updateWaitTime_l(); 5202 } 5203} 5204 5205void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5206{ 5207 Mutex::Autolock _l(mLock); 5208 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5209 if (mOutputTracks[i]->thread() == thread) { 5210 mOutputTracks[i]->destroy(); 5211 mOutputTracks.removeAt(i); 5212 updateWaitTime_l(); 5213 return; 5214 } 5215 } 5216 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5217} 5218 5219// caller must hold mLock 5220void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5221{ 5222 mWaitTimeMs = UINT_MAX; 5223 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5224 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5225 if (strong != 0) { 5226 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5227 if (waitTimeMs < mWaitTimeMs) { 5228 mWaitTimeMs = waitTimeMs; 5229 } 5230 } 5231 } 5232} 5233 5234 5235bool AudioFlinger::DuplicatingThread::outputsReady( 5236 const SortedVector< sp<OutputTrack> > &outputTracks) 5237{ 5238 for (size_t i = 0; i < outputTracks.size(); i++) { 5239 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5240 if (thread == 0) { 5241 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5242 outputTracks[i].get()); 5243 return false; 5244 } 5245 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5246 // see note at standby() declaration 5247 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5248 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5249 thread.get()); 5250 return false; 5251 } 5252 } 5253 return true; 5254} 5255 5256uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5257{ 5258 return (mWaitTimeMs * 1000) / 2; 5259} 5260 5261void AudioFlinger::DuplicatingThread::cacheParameters_l() 5262{ 5263 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5264 updateWaitTime_l(); 5265 5266 MixerThread::cacheParameters_l(); 5267} 5268 5269// ---------------------------------------------------------------------------- 5270// Record 5271// ---------------------------------------------------------------------------- 5272 5273AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5274 AudioStreamIn *input, 5275 audio_io_handle_t id, 5276 audio_devices_t outDevice, 5277 audio_devices_t inDevice 5278#ifdef TEE_SINK 5279 , const sp<NBAIO_Sink>& teeSink 5280#endif 5281 ) : 5282 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5283 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5284 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5285 mRsmpInRear(0) 5286#ifdef TEE_SINK 5287 , mTeeSink(teeSink) 5288#endif 5289 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5290 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5291 // mFastCapture below 5292 , mFastCaptureFutex(0) 5293 // mInputSource 5294 // mPipeSink 5295 // mPipeSource 5296 , mPipeFramesP2(0) 5297 // mPipeMemory 5298 // mFastCaptureNBLogWriter 5299 , mFastTrackAvail(false) 5300{ 5301 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5302 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5303 5304 readInputParameters_l(); 5305 5306 // create an NBAIO source for the HAL input stream, and negotiate 5307 mInputSource = new AudioStreamInSource(input->stream); 5308 size_t numCounterOffers = 0; 5309 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5310 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5311 ALOG_ASSERT(index == 0); 5312 5313 // initialize fast capture depending on configuration 5314 bool initFastCapture; 5315 switch (kUseFastCapture) { 5316 case FastCapture_Never: 5317 initFastCapture = false; 5318 break; 5319 case FastCapture_Always: 5320 initFastCapture = true; 5321 break; 5322 case FastCapture_Static: 5323 uint32_t primaryOutputSampleRate; 5324 { 5325 AutoMutex _l(audioFlinger->mHardwareLock); 5326 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5327 } 5328 initFastCapture = 5329 // either capture sample rate is same as (a reasonable) primary output sample rate 5330 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5331 (mSampleRate == primaryOutputSampleRate)) || 5332 // or primary output sample rate is unknown, and capture sample rate is reasonable 5333 ((primaryOutputSampleRate == 0) && 5334 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5335 // and the buffer size is < 12 ms 5336 (mFrameCount * 1000) / mSampleRate < 12; 5337 break; 5338 // case FastCapture_Dynamic: 5339 } 5340 5341 if (initFastCapture) { 5342 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5343 NBAIO_Format format = mInputSource->format(); 5344 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5345 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5346 void *pipeBuffer; 5347 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5348 sp<IMemory> pipeMemory; 5349 if ((roHeap == 0) || 5350 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5351 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5352 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5353 goto failed; 5354 } 5355 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5356 memset(pipeBuffer, 0, pipeSize); 5357 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5358 const NBAIO_Format offers[1] = {format}; 5359 size_t numCounterOffers = 0; 5360 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5361 ALOG_ASSERT(index == 0); 5362 mPipeSink = pipe; 5363 PipeReader *pipeReader = new PipeReader(*pipe); 5364 numCounterOffers = 0; 5365 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5366 ALOG_ASSERT(index == 0); 5367 mPipeSource = pipeReader; 5368 mPipeFramesP2 = pipeFramesP2; 5369 mPipeMemory = pipeMemory; 5370 5371 // create fast capture 5372 mFastCapture = new FastCapture(); 5373 FastCaptureStateQueue *sq = mFastCapture->sq(); 5374#ifdef STATE_QUEUE_DUMP 5375 // FIXME 5376#endif 5377 FastCaptureState *state = sq->begin(); 5378 state->mCblk = NULL; 5379 state->mInputSource = mInputSource.get(); 5380 state->mInputSourceGen++; 5381 state->mPipeSink = pipe; 5382 state->mPipeSinkGen++; 5383 state->mFrameCount = mFrameCount; 5384 state->mCommand = FastCaptureState::COLD_IDLE; 5385 // already done in constructor initialization list 5386 //mFastCaptureFutex = 0; 5387 state->mColdFutexAddr = &mFastCaptureFutex; 5388 state->mColdGen++; 5389 state->mDumpState = &mFastCaptureDumpState; 5390#ifdef TEE_SINK 5391 // FIXME 5392#endif 5393 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5394 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5395 sq->end(); 5396 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5397 5398 // start the fast capture 5399 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5400 pid_t tid = mFastCapture->getTid(); 5401 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5402 if (err != 0) { 5403 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5404 kPriorityFastCapture, getpid_cached, tid, err); 5405 } 5406 5407#ifdef AUDIO_WATCHDOG 5408 // FIXME 5409#endif 5410 5411 mFastTrackAvail = true; 5412 } 5413failed: ; 5414 5415 // FIXME mNormalSource 5416} 5417 5418AudioFlinger::RecordThread::~RecordThread() 5419{ 5420 if (mFastCapture != 0) { 5421 FastCaptureStateQueue *sq = mFastCapture->sq(); 5422 FastCaptureState *state = sq->begin(); 5423 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5424 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5425 if (old == -1) { 5426 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5427 } 5428 } 5429 state->mCommand = FastCaptureState::EXIT; 5430 sq->end(); 5431 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5432 mFastCapture->join(); 5433 mFastCapture.clear(); 5434 } 5435 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5436 mAudioFlinger->unregisterWriter(mNBLogWriter); 5437 free(mRsmpInBuffer); 5438} 5439 5440void AudioFlinger::RecordThread::onFirstRef() 5441{ 5442 run(mThreadName, PRIORITY_URGENT_AUDIO); 5443} 5444 5445bool AudioFlinger::RecordThread::threadLoop() 5446{ 5447 nsecs_t lastWarning = 0; 5448 5449 inputStandBy(); 5450 5451reacquire_wakelock: 5452 sp<RecordTrack> activeTrack; 5453 int activeTracksGen; 5454 { 5455 Mutex::Autolock _l(mLock); 5456 size_t size = mActiveTracks.size(); 5457 activeTracksGen = mActiveTracksGen; 5458 if (size > 0) { 5459 // FIXME an arbitrary choice 5460 activeTrack = mActiveTracks[0]; 5461 acquireWakeLock_l(activeTrack->uid()); 5462 if (size > 1) { 5463 SortedVector<int> tmp; 5464 for (size_t i = 0; i < size; i++) { 5465 tmp.add(mActiveTracks[i]->uid()); 5466 } 5467 updateWakeLockUids_l(tmp); 5468 } 5469 } else { 5470 acquireWakeLock_l(-1); 5471 } 5472 } 5473 5474 // used to request a deferred sleep, to be executed later while mutex is unlocked 5475 uint32_t sleepUs = 0; 5476 5477 // loop while there is work to do 5478 for (;;) { 5479 Vector< sp<EffectChain> > effectChains; 5480 5481 // sleep with mutex unlocked 5482 if (sleepUs > 0) { 5483 ATRACE_BEGIN("sleep"); 5484 usleep(sleepUs); 5485 ATRACE_END(); 5486 sleepUs = 0; 5487 } 5488 5489 // activeTracks accumulates a copy of a subset of mActiveTracks 5490 Vector< sp<RecordTrack> > activeTracks; 5491 5492 // reference to the (first and only) active fast track 5493 sp<RecordTrack> fastTrack; 5494 5495 // reference to a fast track which is about to be removed 5496 sp<RecordTrack> fastTrackToRemove; 5497 5498 { // scope for mLock 5499 Mutex::Autolock _l(mLock); 5500 5501 processConfigEvents_l(); 5502 5503 // check exitPending here because checkForNewParameters_l() and 5504 // checkForNewParameters_l() can temporarily release mLock 5505 if (exitPending()) { 5506 break; 5507 } 5508 5509 // if no active track(s), then standby and release wakelock 5510 size_t size = mActiveTracks.size(); 5511 if (size == 0) { 5512 standbyIfNotAlreadyInStandby(); 5513 // exitPending() can't become true here 5514 releaseWakeLock_l(); 5515 ALOGV("RecordThread: loop stopping"); 5516 // go to sleep 5517 mWaitWorkCV.wait(mLock); 5518 ALOGV("RecordThread: loop starting"); 5519 goto reacquire_wakelock; 5520 } 5521 5522 if (mActiveTracksGen != activeTracksGen) { 5523 activeTracksGen = mActiveTracksGen; 5524 SortedVector<int> tmp; 5525 for (size_t i = 0; i < size; i++) { 5526 tmp.add(mActiveTracks[i]->uid()); 5527 } 5528 updateWakeLockUids_l(tmp); 5529 } 5530 5531 bool doBroadcast = false; 5532 for (size_t i = 0; i < size; ) { 5533 5534 activeTrack = mActiveTracks[i]; 5535 if (activeTrack->isTerminated()) { 5536 if (activeTrack->isFastTrack()) { 5537 ALOG_ASSERT(fastTrackToRemove == 0); 5538 fastTrackToRemove = activeTrack; 5539 } 5540 removeTrack_l(activeTrack); 5541 mActiveTracks.remove(activeTrack); 5542 mActiveTracksGen++; 5543 size--; 5544 continue; 5545 } 5546 5547 TrackBase::track_state activeTrackState = activeTrack->mState; 5548 switch (activeTrackState) { 5549 5550 case TrackBase::PAUSING: 5551 mActiveTracks.remove(activeTrack); 5552 mActiveTracksGen++; 5553 doBroadcast = true; 5554 size--; 5555 continue; 5556 5557 case TrackBase::STARTING_1: 5558 sleepUs = 10000; 5559 i++; 5560 continue; 5561 5562 case TrackBase::STARTING_2: 5563 doBroadcast = true; 5564 mStandby = false; 5565 activeTrack->mState = TrackBase::ACTIVE; 5566 break; 5567 5568 case TrackBase::ACTIVE: 5569 break; 5570 5571 case TrackBase::IDLE: 5572 i++; 5573 continue; 5574 5575 default: 5576 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5577 } 5578 5579 activeTracks.add(activeTrack); 5580 i++; 5581 5582 if (activeTrack->isFastTrack()) { 5583 ALOG_ASSERT(!mFastTrackAvail); 5584 ALOG_ASSERT(fastTrack == 0); 5585 fastTrack = activeTrack; 5586 } 5587 } 5588 if (doBroadcast) { 5589 mStartStopCond.broadcast(); 5590 } 5591 5592 // sleep if there are no active tracks to process 5593 if (activeTracks.size() == 0) { 5594 if (sleepUs == 0) { 5595 sleepUs = kRecordThreadSleepUs; 5596 } 5597 continue; 5598 } 5599 sleepUs = 0; 5600 5601 lockEffectChains_l(effectChains); 5602 } 5603 5604 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5605 5606 size_t size = effectChains.size(); 5607 for (size_t i = 0; i < size; i++) { 5608 // thread mutex is not locked, but effect chain is locked 5609 effectChains[i]->process_l(); 5610 } 5611 5612 // Push a new fast capture state if fast capture is not already running, or cblk change 5613 if (mFastCapture != 0) { 5614 FastCaptureStateQueue *sq = mFastCapture->sq(); 5615 FastCaptureState *state = sq->begin(); 5616 bool didModify = false; 5617 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5618 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5619 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5620 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5621 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5622 if (old == -1) { 5623 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5624 } 5625 } 5626 state->mCommand = FastCaptureState::READ_WRITE; 5627#if 0 // FIXME 5628 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5629 FastThreadDumpState::kSamplingNforLowRamDevice : 5630 FastThreadDumpState::kSamplingN); 5631#endif 5632 didModify = true; 5633 } 5634 audio_track_cblk_t *cblkOld = state->mCblk; 5635 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5636 if (cblkNew != cblkOld) { 5637 state->mCblk = cblkNew; 5638 // block until acked if removing a fast track 5639 if (cblkOld != NULL) { 5640 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5641 } 5642 didModify = true; 5643 } 5644 sq->end(didModify); 5645 if (didModify) { 5646 sq->push(block); 5647#if 0 5648 if (kUseFastCapture == FastCapture_Dynamic) { 5649 mNormalSource = mPipeSource; 5650 } 5651#endif 5652 } 5653 } 5654 5655 // now run the fast track destructor with thread mutex unlocked 5656 fastTrackToRemove.clear(); 5657 5658 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5659 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5660 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5661 // If destination is non-contiguous, first read past the nominal end of buffer, then 5662 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5663 5664 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5665 ssize_t framesRead; 5666 5667 // If an NBAIO source is present, use it to read the normal capture's data 5668 if (mPipeSource != 0) { 5669 size_t framesToRead = mBufferSize / mFrameSize; 5670 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5671 framesToRead, AudioBufferProvider::kInvalidPTS); 5672 if (framesRead == 0) { 5673 // since pipe is non-blocking, simulate blocking input 5674 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5675 } 5676 // otherwise use the HAL / AudioStreamIn directly 5677 } else { 5678 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5679 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5680 if (bytesRead < 0) { 5681 framesRead = bytesRead; 5682 } else { 5683 framesRead = bytesRead / mFrameSize; 5684 } 5685 } 5686 5687 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5688 ALOGE("read failed: framesRead=%d", framesRead); 5689 // Force input into standby so that it tries to recover at next read attempt 5690 inputStandBy(); 5691 sleepUs = kRecordThreadSleepUs; 5692 } 5693 if (framesRead <= 0) { 5694 goto unlock; 5695 } 5696 ALOG_ASSERT(framesRead > 0); 5697 5698 if (mTeeSink != 0) { 5699 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5700 } 5701 // If destination is non-contiguous, we now correct for reading past end of buffer. 5702 { 5703 size_t part1 = mRsmpInFramesP2 - rear; 5704 if ((size_t) framesRead > part1) { 5705 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5706 (framesRead - part1) * mFrameSize); 5707 } 5708 } 5709 rear = mRsmpInRear += framesRead; 5710 5711 size = activeTracks.size(); 5712 // loop over each active track 5713 for (size_t i = 0; i < size; i++) { 5714 activeTrack = activeTracks[i]; 5715 5716 // skip fast tracks, as those are handled directly by FastCapture 5717 if (activeTrack->isFastTrack()) { 5718 continue; 5719 } 5720 5721 // TODO: This code probably should be moved to RecordTrack. 5722 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5723 5724 enum { 5725 OVERRUN_UNKNOWN, 5726 OVERRUN_TRUE, 5727 OVERRUN_FALSE 5728 } overrun = OVERRUN_UNKNOWN; 5729 5730 // loop over getNextBuffer to handle circular sink 5731 for (;;) { 5732 5733 activeTrack->mSink.frameCount = ~0; 5734 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5735 size_t framesOut = activeTrack->mSink.frameCount; 5736 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5737 5738 // check available frames and handle overrun conditions 5739 // if the record track isn't draining fast enough. 5740 bool hasOverrun; 5741 size_t framesIn; 5742 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5743 if (hasOverrun) { 5744 overrun = OVERRUN_TRUE; 5745 } 5746 if (framesOut == 0 || framesIn == 0) { 5747 break; 5748 } 5749 5750 // Don't allow framesOut to be larger than what is possible with resampling 5751 // from framesIn. 5752 // This isn't strictly necessary but helps limit buffer resizing in 5753 // RecordBufferConverter. TODO: remove when no longer needed. 5754 framesOut = min(framesOut, 5755 destinationFramesPossible( 5756 framesIn, mSampleRate, activeTrack->mSampleRate)); 5757 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5758 framesOut = activeTrack->mRecordBufferConverter->convert( 5759 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5760 5761 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5762 overrun = OVERRUN_FALSE; 5763 } 5764 5765 if (activeTrack->mFramesToDrop == 0) { 5766 if (framesOut > 0) { 5767 activeTrack->mSink.frameCount = framesOut; 5768 activeTrack->releaseBuffer(&activeTrack->mSink); 5769 } 5770 } else { 5771 // FIXME could do a partial drop of framesOut 5772 if (activeTrack->mFramesToDrop > 0) { 5773 activeTrack->mFramesToDrop -= framesOut; 5774 if (activeTrack->mFramesToDrop <= 0) { 5775 activeTrack->clearSyncStartEvent(); 5776 } 5777 } else { 5778 activeTrack->mFramesToDrop += framesOut; 5779 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5780 activeTrack->mSyncStartEvent->isCancelled()) { 5781 ALOGW("Synced record %s, session %d, trigger session %d", 5782 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5783 activeTrack->sessionId(), 5784 (activeTrack->mSyncStartEvent != 0) ? 5785 activeTrack->mSyncStartEvent->triggerSession() : 0); 5786 activeTrack->clearSyncStartEvent(); 5787 } 5788 } 5789 } 5790 5791 if (framesOut == 0) { 5792 break; 5793 } 5794 } 5795 5796 switch (overrun) { 5797 case OVERRUN_TRUE: 5798 // client isn't retrieving buffers fast enough 5799 if (!activeTrack->setOverflow()) { 5800 nsecs_t now = systemTime(); 5801 // FIXME should lastWarning per track? 5802 if ((now - lastWarning) > kWarningThrottleNs) { 5803 ALOGW("RecordThread: buffer overflow"); 5804 lastWarning = now; 5805 } 5806 } 5807 break; 5808 case OVERRUN_FALSE: 5809 activeTrack->clearOverflow(); 5810 break; 5811 case OVERRUN_UNKNOWN: 5812 break; 5813 } 5814 5815 } 5816 5817unlock: 5818 // enable changes in effect chain 5819 unlockEffectChains(effectChains); 5820 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5821 } 5822 5823 standbyIfNotAlreadyInStandby(); 5824 5825 { 5826 Mutex::Autolock _l(mLock); 5827 for (size_t i = 0; i < mTracks.size(); i++) { 5828 sp<RecordTrack> track = mTracks[i]; 5829 track->invalidate(); 5830 } 5831 mActiveTracks.clear(); 5832 mActiveTracksGen++; 5833 mStartStopCond.broadcast(); 5834 } 5835 5836 releaseWakeLock(); 5837 5838 ALOGV("RecordThread %p exiting", this); 5839 return false; 5840} 5841 5842void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5843{ 5844 if (!mStandby) { 5845 inputStandBy(); 5846 mStandby = true; 5847 } 5848} 5849 5850void AudioFlinger::RecordThread::inputStandBy() 5851{ 5852 // Idle the fast capture if it's currently running 5853 if (mFastCapture != 0) { 5854 FastCaptureStateQueue *sq = mFastCapture->sq(); 5855 FastCaptureState *state = sq->begin(); 5856 if (!(state->mCommand & FastCaptureState::IDLE)) { 5857 state->mCommand = FastCaptureState::COLD_IDLE; 5858 state->mColdFutexAddr = &mFastCaptureFutex; 5859 state->mColdGen++; 5860 mFastCaptureFutex = 0; 5861 sq->end(); 5862 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5863 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5864#if 0 5865 if (kUseFastCapture == FastCapture_Dynamic) { 5866 // FIXME 5867 } 5868#endif 5869#ifdef AUDIO_WATCHDOG 5870 // FIXME 5871#endif 5872 } else { 5873 sq->end(false /*didModify*/); 5874 } 5875 } 5876 mInput->stream->common.standby(&mInput->stream->common); 5877} 5878 5879// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5880sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5881 const sp<AudioFlinger::Client>& client, 5882 uint32_t sampleRate, 5883 audio_format_t format, 5884 audio_channel_mask_t channelMask, 5885 size_t *pFrameCount, 5886 int sessionId, 5887 size_t *notificationFrames, 5888 int uid, 5889 IAudioFlinger::track_flags_t *flags, 5890 pid_t tid, 5891 status_t *status) 5892{ 5893 size_t frameCount = *pFrameCount; 5894 sp<RecordTrack> track; 5895 status_t lStatus; 5896 5897 // client expresses a preference for FAST, but we get the final say 5898 if (*flags & IAudioFlinger::TRACK_FAST) { 5899 if ( 5900 // we formerly checked for a callback handler (non-0 tid), 5901 // but that is no longer required for TRANSFER_OBTAIN mode 5902 // 5903 // frame count is not specified, or is exactly the pipe depth 5904 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5905 // PCM data 5906 audio_is_linear_pcm(format) && 5907 // native format 5908 (format == mFormat) && 5909 // native channel mask 5910 (channelMask == mChannelMask) && 5911 // native hardware sample rate 5912 (sampleRate == mSampleRate) && 5913 // record thread has an associated fast capture 5914 hasFastCapture() && 5915 // there are sufficient fast track slots available 5916 mFastTrackAvail 5917 ) { 5918 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5919 frameCount, mFrameCount); 5920 } else { 5921 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5922 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5923 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5924 frameCount, mFrameCount, mPipeFramesP2, 5925 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5926 hasFastCapture(), tid, mFastTrackAvail); 5927 *flags &= ~IAudioFlinger::TRACK_FAST; 5928 } 5929 } 5930 5931 // compute track buffer size in frames, and suggest the notification frame count 5932 if (*flags & IAudioFlinger::TRACK_FAST) { 5933 // fast track: frame count is exactly the pipe depth 5934 frameCount = mPipeFramesP2; 5935 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5936 *notificationFrames = mFrameCount; 5937 } else { 5938 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5939 // or 20 ms if there is a fast capture 5940 // TODO This could be a roundupRatio inline, and const 5941 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5942 * sampleRate + mSampleRate - 1) / mSampleRate; 5943 // minimum number of notification periods is at least kMinNotifications, 5944 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5945 static const size_t kMinNotifications = 3; 5946 static const uint32_t kMinMs = 30; 5947 // TODO This could be a roundupRatio inline 5948 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5949 // TODO This could be a roundupRatio inline 5950 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5951 maxNotificationFrames; 5952 const size_t minFrameCount = maxNotificationFrames * 5953 max(kMinNotifications, minNotificationsByMs); 5954 frameCount = max(frameCount, minFrameCount); 5955 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5956 *notificationFrames = maxNotificationFrames; 5957 } 5958 } 5959 *pFrameCount = frameCount; 5960 5961 lStatus = initCheck(); 5962 if (lStatus != NO_ERROR) { 5963 ALOGE("createRecordTrack_l() audio driver not initialized"); 5964 goto Exit; 5965 } 5966 5967 { // scope for mLock 5968 Mutex::Autolock _l(mLock); 5969 5970 track = new RecordTrack(this, client, sampleRate, 5971 format, channelMask, frameCount, NULL, sessionId, uid, 5972 *flags, TrackBase::TYPE_DEFAULT); 5973 5974 lStatus = track->initCheck(); 5975 if (lStatus != NO_ERROR) { 5976 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5977 // track must be cleared from the caller as the caller has the AF lock 5978 goto Exit; 5979 } 5980 mTracks.add(track); 5981 5982 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5983 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5984 mAudioFlinger->btNrecIsOff(); 5985 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5986 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5987 5988 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5989 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5990 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5991 // so ask activity manager to do this on our behalf 5992 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5993 } 5994 } 5995 5996 lStatus = NO_ERROR; 5997 5998Exit: 5999 *status = lStatus; 6000 return track; 6001} 6002 6003status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6004 AudioSystem::sync_event_t event, 6005 int triggerSession) 6006{ 6007 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6008 sp<ThreadBase> strongMe = this; 6009 status_t status = NO_ERROR; 6010 6011 if (event == AudioSystem::SYNC_EVENT_NONE) { 6012 recordTrack->clearSyncStartEvent(); 6013 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6014 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6015 triggerSession, 6016 recordTrack->sessionId(), 6017 syncStartEventCallback, 6018 recordTrack); 6019 // Sync event can be cancelled by the trigger session if the track is not in a 6020 // compatible state in which case we start record immediately 6021 if (recordTrack->mSyncStartEvent->isCancelled()) { 6022 recordTrack->clearSyncStartEvent(); 6023 } else { 6024 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6025 recordTrack->mFramesToDrop = - 6026 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6027 } 6028 } 6029 6030 { 6031 // This section is a rendezvous between binder thread executing start() and RecordThread 6032 AutoMutex lock(mLock); 6033 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6034 if (recordTrack->mState == TrackBase::PAUSING) { 6035 ALOGV("active record track PAUSING -> ACTIVE"); 6036 recordTrack->mState = TrackBase::ACTIVE; 6037 } else { 6038 ALOGV("active record track state %d", recordTrack->mState); 6039 } 6040 return status; 6041 } 6042 6043 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6044 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6045 // or using a separate command thread 6046 recordTrack->mState = TrackBase::STARTING_1; 6047 mActiveTracks.add(recordTrack); 6048 mActiveTracksGen++; 6049 status_t status = NO_ERROR; 6050 if (recordTrack->isExternalTrack()) { 6051 mLock.unlock(); 6052 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6053 mLock.lock(); 6054 // FIXME should verify that recordTrack is still in mActiveTracks 6055 if (status != NO_ERROR) { 6056 mActiveTracks.remove(recordTrack); 6057 mActiveTracksGen++; 6058 recordTrack->clearSyncStartEvent(); 6059 ALOGV("RecordThread::start error %d", status); 6060 return status; 6061 } 6062 } 6063 // Catch up with current buffer indices if thread is already running. 6064 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6065 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6066 // see previously buffered data before it called start(), but with greater risk of overrun. 6067 6068 recordTrack->mResamplerBufferProvider->reset(); 6069 // clear any converter state as new data will be discontinuous 6070 recordTrack->mRecordBufferConverter->reset(); 6071 recordTrack->mState = TrackBase::STARTING_2; 6072 // signal thread to start 6073 mWaitWorkCV.broadcast(); 6074 if (mActiveTracks.indexOf(recordTrack) < 0) { 6075 ALOGV("Record failed to start"); 6076 status = BAD_VALUE; 6077 goto startError; 6078 } 6079 return status; 6080 } 6081 6082startError: 6083 if (recordTrack->isExternalTrack()) { 6084 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6085 } 6086 recordTrack->clearSyncStartEvent(); 6087 // FIXME I wonder why we do not reset the state here? 6088 return status; 6089} 6090 6091void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6092{ 6093 sp<SyncEvent> strongEvent = event.promote(); 6094 6095 if (strongEvent != 0) { 6096 sp<RefBase> ptr = strongEvent->cookie().promote(); 6097 if (ptr != 0) { 6098 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6099 recordTrack->handleSyncStartEvent(strongEvent); 6100 } 6101 } 6102} 6103 6104bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6105 ALOGV("RecordThread::stop"); 6106 AutoMutex _l(mLock); 6107 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6108 return false; 6109 } 6110 // note that threadLoop may still be processing the track at this point [without lock] 6111 recordTrack->mState = TrackBase::PAUSING; 6112 // do not wait for mStartStopCond if exiting 6113 if (exitPending()) { 6114 return true; 6115 } 6116 // FIXME incorrect usage of wait: no explicit predicate or loop 6117 mStartStopCond.wait(mLock); 6118 // if we have been restarted, recordTrack is in mActiveTracks here 6119 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6120 ALOGV("Record stopped OK"); 6121 return true; 6122 } 6123 return false; 6124} 6125 6126bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6127{ 6128 return false; 6129} 6130 6131status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6132{ 6133#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6134 if (!isValidSyncEvent(event)) { 6135 return BAD_VALUE; 6136 } 6137 6138 int eventSession = event->triggerSession(); 6139 status_t ret = NAME_NOT_FOUND; 6140 6141 Mutex::Autolock _l(mLock); 6142 6143 for (size_t i = 0; i < mTracks.size(); i++) { 6144 sp<RecordTrack> track = mTracks[i]; 6145 if (eventSession == track->sessionId()) { 6146 (void) track->setSyncEvent(event); 6147 ret = NO_ERROR; 6148 } 6149 } 6150 return ret; 6151#else 6152 return BAD_VALUE; 6153#endif 6154} 6155 6156// destroyTrack_l() must be called with ThreadBase::mLock held 6157void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6158{ 6159 track->terminate(); 6160 track->mState = TrackBase::STOPPED; 6161 // active tracks are removed by threadLoop() 6162 if (mActiveTracks.indexOf(track) < 0) { 6163 removeTrack_l(track); 6164 } 6165} 6166 6167void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6168{ 6169 mTracks.remove(track); 6170 // need anything related to effects here? 6171 if (track->isFastTrack()) { 6172 ALOG_ASSERT(!mFastTrackAvail); 6173 mFastTrackAvail = true; 6174 } 6175} 6176 6177void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6178{ 6179 dumpInternals(fd, args); 6180 dumpTracks(fd, args); 6181 dumpEffectChains(fd, args); 6182} 6183 6184void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6185{ 6186 dprintf(fd, "\nInput thread %p:\n", this); 6187 6188 dumpBase(fd, args); 6189 6190 if (mActiveTracks.size() == 0) { 6191 dprintf(fd, " No active record clients\n"); 6192 } 6193 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6194 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6195 6196 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6197 const FastCaptureDumpState copy(mFastCaptureDumpState); 6198 copy.dump(fd); 6199} 6200 6201void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6202{ 6203 const size_t SIZE = 256; 6204 char buffer[SIZE]; 6205 String8 result; 6206 6207 size_t numtracks = mTracks.size(); 6208 size_t numactive = mActiveTracks.size(); 6209 size_t numactiveseen = 0; 6210 dprintf(fd, " %d Tracks", numtracks); 6211 if (numtracks) { 6212 dprintf(fd, " of which %d are active\n", numactive); 6213 RecordTrack::appendDumpHeader(result); 6214 for (size_t i = 0; i < numtracks ; ++i) { 6215 sp<RecordTrack> track = mTracks[i]; 6216 if (track != 0) { 6217 bool active = mActiveTracks.indexOf(track) >= 0; 6218 if (active) { 6219 numactiveseen++; 6220 } 6221 track->dump(buffer, SIZE, active); 6222 result.append(buffer); 6223 } 6224 } 6225 } else { 6226 dprintf(fd, "\n"); 6227 } 6228 6229 if (numactiveseen != numactive) { 6230 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6231 " not in the track list\n"); 6232 result.append(buffer); 6233 RecordTrack::appendDumpHeader(result); 6234 for (size_t i = 0; i < numactive; ++i) { 6235 sp<RecordTrack> track = mActiveTracks[i]; 6236 if (mTracks.indexOf(track) < 0) { 6237 track->dump(buffer, SIZE, true); 6238 result.append(buffer); 6239 } 6240 } 6241 6242 } 6243 write(fd, result.string(), result.size()); 6244} 6245 6246 6247void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6248{ 6249 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6250 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6251 mRsmpInFront = recordThread->mRsmpInRear; 6252 mRsmpInUnrel = 0; 6253} 6254 6255void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6256 size_t *framesAvailable, bool *hasOverrun) 6257{ 6258 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6259 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6260 const int32_t rear = recordThread->mRsmpInRear; 6261 const int32_t front = mRsmpInFront; 6262 const ssize_t filled = rear - front; 6263 6264 size_t framesIn; 6265 bool overrun = false; 6266 if (filled < 0) { 6267 // should not happen, but treat like a massive overrun and re-sync 6268 framesIn = 0; 6269 mRsmpInFront = rear; 6270 overrun = true; 6271 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6272 framesIn = (size_t) filled; 6273 } else { 6274 // client is not keeping up with server, but give it latest data 6275 framesIn = recordThread->mRsmpInFrames; 6276 mRsmpInFront = /* front = */ rear - framesIn; 6277 overrun = true; 6278 } 6279 if (framesAvailable != NULL) { 6280 *framesAvailable = framesIn; 6281 } 6282 if (hasOverrun != NULL) { 6283 *hasOverrun = overrun; 6284 } 6285} 6286 6287// AudioBufferProvider interface 6288status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6289 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6290{ 6291 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6292 if (threadBase == 0) { 6293 buffer->frameCount = 0; 6294 buffer->raw = NULL; 6295 return NOT_ENOUGH_DATA; 6296 } 6297 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6298 int32_t rear = recordThread->mRsmpInRear; 6299 int32_t front = mRsmpInFront; 6300 ssize_t filled = rear - front; 6301 // FIXME should not be P2 (don't want to increase latency) 6302 // FIXME if client not keeping up, discard 6303 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6304 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6305 front &= recordThread->mRsmpInFramesP2 - 1; 6306 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6307 if (part1 > (size_t) filled) { 6308 part1 = filled; 6309 } 6310 size_t ask = buffer->frameCount; 6311 ALOG_ASSERT(ask > 0); 6312 if (part1 > ask) { 6313 part1 = ask; 6314 } 6315 if (part1 == 0) { 6316 // out of data is fine since the resampler will return a short-count. 6317 buffer->raw = NULL; 6318 buffer->frameCount = 0; 6319 mRsmpInUnrel = 0; 6320 return NOT_ENOUGH_DATA; 6321 } 6322 6323 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6324 buffer->frameCount = part1; 6325 mRsmpInUnrel = part1; 6326 return NO_ERROR; 6327} 6328 6329// AudioBufferProvider interface 6330void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6331 AudioBufferProvider::Buffer* buffer) 6332{ 6333 size_t stepCount = buffer->frameCount; 6334 if (stepCount == 0) { 6335 return; 6336 } 6337 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6338 mRsmpInUnrel -= stepCount; 6339 mRsmpInFront += stepCount; 6340 buffer->raw = NULL; 6341 buffer->frameCount = 0; 6342} 6343 6344AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6345 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6346 uint32_t srcSampleRate, 6347 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6348 uint32_t dstSampleRate) : 6349 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6350 // mSrcFormat 6351 // mSrcSampleRate 6352 // mDstChannelMask 6353 // mDstFormat 6354 // mDstSampleRate 6355 // mSrcChannelCount 6356 // mDstChannelCount 6357 // mDstFrameSize 6358 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6359 mResampler(NULL), 6360 mIsLegacyDownmix(false), 6361 mIsLegacyUpmix(false), 6362 mRequiresFloat(false), 6363 mInputConverterProvider(NULL) 6364{ 6365 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6366 dstChannelMask, dstFormat, dstSampleRate); 6367} 6368 6369AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6370 free(mBuf); 6371 delete mResampler; 6372 delete mInputConverterProvider; 6373} 6374 6375size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6376 AudioBufferProvider *provider, size_t frames) 6377{ 6378 if (mInputConverterProvider != NULL) { 6379 mInputConverterProvider->setBufferProvider(provider); 6380 provider = mInputConverterProvider; 6381 } 6382 6383 if (mResampler == NULL) { 6384 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6385 mSrcSampleRate, mSrcFormat, mDstFormat); 6386 6387 AudioBufferProvider::Buffer buffer; 6388 for (size_t i = frames; i > 0; ) { 6389 buffer.frameCount = i; 6390 status_t status = provider->getNextBuffer(&buffer, 0); 6391 if (status != OK || buffer.frameCount == 0) { 6392 frames -= i; // cannot fill request. 6393 break; 6394 } 6395 // format convert to destination buffer 6396 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6397 6398 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6399 i -= buffer.frameCount; 6400 provider->releaseBuffer(&buffer); 6401 } 6402 } else { 6403 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6404 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6405 6406 // reallocate buffer if needed 6407 if (mBufFrameSize != 0 && mBufFrames < frames) { 6408 free(mBuf); 6409 mBufFrames = frames; 6410 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6411 } 6412 // resampler accumulates, but we only have one source track 6413 memset(mBuf, 0, frames * mBufFrameSize); 6414 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6415 // format convert to destination buffer 6416 convertResampler(dst, mBuf, frames); 6417 } 6418 return frames; 6419} 6420 6421status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6422 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6423 uint32_t srcSampleRate, 6424 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6425 uint32_t dstSampleRate) 6426{ 6427 // quick evaluation if there is any change. 6428 if (mSrcFormat == srcFormat 6429 && mSrcChannelMask == srcChannelMask 6430 && mSrcSampleRate == srcSampleRate 6431 && mDstFormat == dstFormat 6432 && mDstChannelMask == dstChannelMask 6433 && mDstSampleRate == dstSampleRate) { 6434 return NO_ERROR; 6435 } 6436 6437 const bool valid = 6438 audio_is_input_channel(srcChannelMask) 6439 && audio_is_input_channel(dstChannelMask) 6440 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6441 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6442 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6443 ; // no upsampling checks for now 6444 if (!valid) { 6445 return BAD_VALUE; 6446 } 6447 6448 mSrcFormat = srcFormat; 6449 mSrcChannelMask = srcChannelMask; 6450 mSrcSampleRate = srcSampleRate; 6451 mDstFormat = dstFormat; 6452 mDstChannelMask = dstChannelMask; 6453 mDstSampleRate = dstSampleRate; 6454 6455 // compute derived parameters 6456 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6457 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6458 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6459 6460 // do we need to resample? 6461 delete mResampler; 6462 mResampler = NULL; 6463 if (mSrcSampleRate != mDstSampleRate) { 6464 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6465 mSrcChannelCount, mDstSampleRate); 6466 mResampler->setSampleRate(mSrcSampleRate); 6467 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6468 } 6469 6470 // are we running legacy channel conversion modes? 6471 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6472 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6473 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6474 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6475 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6476 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6477 6478 // do we need to process in float? 6479 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6480 6481 // do we need a staging buffer to convert for destination (we can still optimize this)? 6482 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6483 if (mResampler != NULL) { 6484 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6485 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6486 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6487 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6488 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6489 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6490 } else { 6491 mBufFrameSize = 0; 6492 } 6493 mBufFrames = 0; // force the buffer to be resized. 6494 6495 // do we need an input converter buffer provider to give us float? 6496 delete mInputConverterProvider; 6497 mInputConverterProvider = NULL; 6498 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6499 mInputConverterProvider = new ReformatBufferProvider( 6500 audio_channel_count_from_in_mask(mSrcChannelMask), 6501 mSrcFormat, 6502 AUDIO_FORMAT_PCM_FLOAT, 6503 256 /* provider buffer frame count */); 6504 } 6505 6506 // do we need a remixer to do channel mask conversion 6507 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6508 (void) memcpy_by_index_array_initialization_from_channel_mask( 6509 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6510 } 6511 return NO_ERROR; 6512} 6513 6514void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6515 void *dst, const void *src, size_t frames) 6516{ 6517 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6518 if (mBufFrameSize != 0 && mBufFrames < frames) { 6519 free(mBuf); 6520 mBufFrames = frames; 6521 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6522 } 6523 // do we need to do legacy upmix and downmix? 6524 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6525 void *dstBuf = mBuf != NULL ? mBuf : dst; 6526 if (mIsLegacyUpmix) { 6527 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6528 (const float *)src, frames); 6529 } else /*mIsLegacyDownmix */ { 6530 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6531 (const float *)src, frames); 6532 } 6533 if (mBuf != NULL) { 6534 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6535 frames * mDstChannelCount); 6536 } 6537 return; 6538 } 6539 // do we need to do channel mask conversion? 6540 if (mSrcChannelMask != mDstChannelMask) { 6541 void *dstBuf = mBuf != NULL ? mBuf : dst; 6542 memcpy_by_index_array(dstBuf, mDstChannelCount, 6543 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6544 if (dstBuf == dst) { 6545 return; // format is the same 6546 } 6547 } 6548 // convert to destination buffer 6549 const void *convertBuf = mBuf != NULL ? mBuf : src; 6550 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6551 frames * mDstChannelCount); 6552} 6553 6554void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6555 void *dst, /*not-a-const*/ void *src, size_t frames) 6556{ 6557 // src buffer format is ALWAYS float when entering this routine 6558 if (mIsLegacyUpmix) { 6559 ; // mono to stereo already handled by resampler 6560 } else if (mIsLegacyDownmix 6561 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6562 // the resampler outputs stereo for mono input channel (a feature?) 6563 // must convert to mono 6564 downmix_to_mono_float_from_stereo_float((float *)src, 6565 (const float *)src, frames); 6566 } else if (mSrcChannelMask != mDstChannelMask) { 6567 // convert to mono channel again for channel mask conversion (could be skipped 6568 // with further optimization). 6569 if (mSrcChannelCount == 1) { 6570 downmix_to_mono_float_from_stereo_float((float *)src, 6571 (const float *)src, frames); 6572 } 6573 // convert to destination format (in place, OK as float is larger than other types) 6574 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6575 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6576 frames * mSrcChannelCount); 6577 } 6578 // channel convert and save to dst 6579 memcpy_by_index_array(dst, mDstChannelCount, 6580 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6581 return; 6582 } 6583 // convert to destination format and save to dst 6584 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6585 frames * mDstChannelCount); 6586} 6587 6588bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6589 status_t& status) 6590{ 6591 bool reconfig = false; 6592 6593 status = NO_ERROR; 6594 6595 audio_format_t reqFormat = mFormat; 6596 uint32_t samplingRate = mSampleRate; 6597 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6598 // possible that we are > 2 channels, use channel index mask 6599 if (channelMask == AUDIO_CHANNEL_INVALID && mChannelCount <= FCC_8) { 6600 audio_channel_mask_for_index_assignment_from_count(mChannelCount); 6601 } 6602 6603 AudioParameter param = AudioParameter(keyValuePair); 6604 int value; 6605 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6606 // channel count change can be requested. Do we mandate the first client defines the 6607 // HAL sampling rate and channel count or do we allow changes on the fly? 6608 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6609 samplingRate = value; 6610 reconfig = true; 6611 } 6612 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6613 if (!audio_is_linear_pcm((audio_format_t) value)) { 6614 status = BAD_VALUE; 6615 } else { 6616 reqFormat = (audio_format_t) value; 6617 reconfig = true; 6618 } 6619 } 6620 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6621 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6622 if (!audio_is_input_channel(mask) || 6623 audio_channel_count_from_in_mask(mask) > FCC_8) { 6624 status = BAD_VALUE; 6625 } else { 6626 channelMask = mask; 6627 reconfig = true; 6628 } 6629 } 6630 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6631 // do not accept frame count changes if tracks are open as the track buffer 6632 // size depends on frame count and correct behavior would not be guaranteed 6633 // if frame count is changed after track creation 6634 if (mActiveTracks.size() > 0) { 6635 status = INVALID_OPERATION; 6636 } else { 6637 reconfig = true; 6638 } 6639 } 6640 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6641 // forward device change to effects that have requested to be 6642 // aware of attached audio device. 6643 for (size_t i = 0; i < mEffectChains.size(); i++) { 6644 mEffectChains[i]->setDevice_l(value); 6645 } 6646 6647 // store input device and output device but do not forward output device to audio HAL. 6648 // Note that status is ignored by the caller for output device 6649 // (see AudioFlinger::setParameters() 6650 if (audio_is_output_devices(value)) { 6651 mOutDevice = value; 6652 status = BAD_VALUE; 6653 } else { 6654 mInDevice = value; 6655 // disable AEC and NS if the device is a BT SCO headset supporting those 6656 // pre processings 6657 if (mTracks.size() > 0) { 6658 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6659 mAudioFlinger->btNrecIsOff(); 6660 for (size_t i = 0; i < mTracks.size(); i++) { 6661 sp<RecordTrack> track = mTracks[i]; 6662 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6663 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6664 } 6665 } 6666 } 6667 } 6668 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6669 mAudioSource != (audio_source_t)value) { 6670 // forward device change to effects that have requested to be 6671 // aware of attached audio device. 6672 for (size_t i = 0; i < mEffectChains.size(); i++) { 6673 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6674 } 6675 mAudioSource = (audio_source_t)value; 6676 } 6677 6678 if (status == NO_ERROR) { 6679 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6680 keyValuePair.string()); 6681 if (status == INVALID_OPERATION) { 6682 inputStandBy(); 6683 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6684 keyValuePair.string()); 6685 } 6686 if (reconfig) { 6687 if (status == BAD_VALUE && 6688 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6689 audio_is_linear_pcm(reqFormat) && 6690 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6691 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6692 audio_channel_count_from_in_mask( 6693 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6694 (channelMask == AUDIO_CHANNEL_IN_MONO || 6695 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6696 status = NO_ERROR; 6697 } 6698 if (status == NO_ERROR) { 6699 readInputParameters_l(); 6700 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6701 } 6702 } 6703 } 6704 6705 return reconfig; 6706} 6707 6708String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6709{ 6710 Mutex::Autolock _l(mLock); 6711 if (initCheck() != NO_ERROR) { 6712 return String8(); 6713 } 6714 6715 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6716 const String8 out_s8(s); 6717 free(s); 6718 return out_s8; 6719} 6720 6721void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event) { 6722 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6723 6724 desc->mIoHandle = mId; 6725 6726 switch (event) { 6727 case AUDIO_INPUT_OPENED: 6728 case AUDIO_INPUT_CONFIG_CHANGED: 6729 desc->mChannelMask = mChannelMask; 6730 desc->mSamplingRate = mSampleRate; 6731 desc->mFormat = mFormat; 6732 desc->mFrameCount = mFrameCount; 6733 desc->mLatency = 0; 6734 break; 6735 6736 case AUDIO_INPUT_CLOSED: 6737 default: 6738 break; 6739 } 6740 mAudioFlinger->ioConfigChanged(event, desc); 6741} 6742 6743void AudioFlinger::RecordThread::readInputParameters_l() 6744{ 6745 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6746 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6747 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6748 if (mChannelCount > FCC_8) { 6749 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6750 } 6751 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6752 mFormat = mHALFormat; 6753 if (!audio_is_linear_pcm(mFormat)) { 6754 ALOGE("HAL format %#x is not linear pcm", mFormat); 6755 } 6756 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6757 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6758 mFrameCount = mBufferSize / mFrameSize; 6759 // This is the formula for calculating the temporary buffer size. 6760 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6761 // 1 full output buffer, regardless of the alignment of the available input. 6762 // The value is somewhat arbitrary, and could probably be even larger. 6763 // A larger value should allow more old data to be read after a track calls start(), 6764 // without increasing latency. 6765 // 6766 // Note this is independent of the maximum downsampling ratio permitted for capture. 6767 mRsmpInFrames = mFrameCount * 7; 6768 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6769 free(mRsmpInBuffer); 6770 6771 // TODO optimize audio capture buffer sizes ... 6772 // Here we calculate the size of the sliding buffer used as a source 6773 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6774 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6775 // be better to have it derived from the pipe depth in the long term. 6776 // The current value is higher than necessary. However it should not add to latency. 6777 6778 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6779 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6780 6781 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6782 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6783} 6784 6785uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6786{ 6787 Mutex::Autolock _l(mLock); 6788 if (initCheck() != NO_ERROR) { 6789 return 0; 6790 } 6791 6792 return mInput->stream->get_input_frames_lost(mInput->stream); 6793} 6794 6795uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6796{ 6797 Mutex::Autolock _l(mLock); 6798 uint32_t result = 0; 6799 if (getEffectChain_l(sessionId) != 0) { 6800 result = EFFECT_SESSION; 6801 } 6802 6803 for (size_t i = 0; i < mTracks.size(); ++i) { 6804 if (sessionId == mTracks[i]->sessionId()) { 6805 result |= TRACK_SESSION; 6806 break; 6807 } 6808 } 6809 6810 return result; 6811} 6812 6813KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6814{ 6815 KeyedVector<int, bool> ids; 6816 Mutex::Autolock _l(mLock); 6817 for (size_t j = 0; j < mTracks.size(); ++j) { 6818 sp<RecordThread::RecordTrack> track = mTracks[j]; 6819 int sessionId = track->sessionId(); 6820 if (ids.indexOfKey(sessionId) < 0) { 6821 ids.add(sessionId, true); 6822 } 6823 } 6824 return ids; 6825} 6826 6827AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6828{ 6829 Mutex::Autolock _l(mLock); 6830 AudioStreamIn *input = mInput; 6831 mInput = NULL; 6832 return input; 6833} 6834 6835// this method must always be called either with ThreadBase mLock held or inside the thread loop 6836audio_stream_t* AudioFlinger::RecordThread::stream() const 6837{ 6838 if (mInput == NULL) { 6839 return NULL; 6840 } 6841 return &mInput->stream->common; 6842} 6843 6844status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6845{ 6846 // only one chain per input thread 6847 if (mEffectChains.size() != 0) { 6848 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6849 return INVALID_OPERATION; 6850 } 6851 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6852 chain->setThread(this); 6853 chain->setInBuffer(NULL); 6854 chain->setOutBuffer(NULL); 6855 6856 checkSuspendOnAddEffectChain_l(chain); 6857 6858 // make sure enabled pre processing effects state is communicated to the HAL as we 6859 // just moved them to a new input stream. 6860 chain->syncHalEffectsState(); 6861 6862 mEffectChains.add(chain); 6863 6864 return NO_ERROR; 6865} 6866 6867size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6868{ 6869 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6870 ALOGW_IF(mEffectChains.size() != 1, 6871 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6872 chain.get(), mEffectChains.size(), this); 6873 if (mEffectChains.size() == 1) { 6874 mEffectChains.removeAt(0); 6875 } 6876 return 0; 6877} 6878 6879status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6880 audio_patch_handle_t *handle) 6881{ 6882 status_t status = NO_ERROR; 6883 6884 // store new device and send to effects 6885 mInDevice = patch->sources[0].ext.device.type; 6886 for (size_t i = 0; i < mEffectChains.size(); i++) { 6887 mEffectChains[i]->setDevice_l(mInDevice); 6888 } 6889 6890 // disable AEC and NS if the device is a BT SCO headset supporting those 6891 // pre processings 6892 if (mTracks.size() > 0) { 6893 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6894 mAudioFlinger->btNrecIsOff(); 6895 for (size_t i = 0; i < mTracks.size(); i++) { 6896 sp<RecordTrack> track = mTracks[i]; 6897 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6898 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6899 } 6900 } 6901 6902 // store new source and send to effects 6903 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6904 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6905 for (size_t i = 0; i < mEffectChains.size(); i++) { 6906 mEffectChains[i]->setAudioSource_l(mAudioSource); 6907 } 6908 } 6909 6910 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6911 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6912 status = hwDevice->create_audio_patch(hwDevice, 6913 patch->num_sources, 6914 patch->sources, 6915 patch->num_sinks, 6916 patch->sinks, 6917 handle); 6918 } else { 6919 char *address; 6920 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 6921 address = audio_device_address_to_parameter( 6922 patch->sources[0].ext.device.type, 6923 patch->sources[0].ext.device.address); 6924 } else { 6925 address = (char *)calloc(1, 1); 6926 } 6927 AudioParameter param = AudioParameter(String8(address)); 6928 free(address); 6929 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 6930 (int)patch->sources[0].ext.device.type); 6931 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 6932 (int)patch->sinks[0].ext.mix.usecase.source); 6933 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6934 param.toString().string()); 6935 *handle = AUDIO_PATCH_HANDLE_NONE; 6936 } 6937 6938 return status; 6939} 6940 6941status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6942{ 6943 status_t status = NO_ERROR; 6944 6945 mInDevice = AUDIO_DEVICE_NONE; 6946 6947 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6948 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6949 status = hwDevice->release_audio_patch(hwDevice, handle); 6950 } else { 6951 AudioParameter param; 6952 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 6953 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6954 param.toString().string()); 6955 } 6956 return status; 6957} 6958 6959void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6960{ 6961 Mutex::Autolock _l(mLock); 6962 mTracks.add(record); 6963} 6964 6965void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6966{ 6967 Mutex::Autolock _l(mLock); 6968 destroyTrack_l(record); 6969} 6970 6971void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6972{ 6973 ThreadBase::getAudioPortConfig(config); 6974 config->role = AUDIO_PORT_ROLE_SINK; 6975 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6976 config->ext.mix.usecase.source = mAudioSource; 6977} 6978 6979} // namespace android 6980