Threads.cpp revision d330ee46022f34da76d14d0c4d2910526ecc2321
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(int event, int param) 588{ 589 Mutex::Autolock _l(mLock); 590 sendIoConfigEvent_l(event, param); 591} 592 593// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 594void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 595{ 596 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 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 audioConfigChanged(data->mEvent, data->mParam); 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::audioConfigChanged(int event, int param) { 1925 AudioSystem::OutputDescriptor desc; 1926 void *param2 = NULL; 1927 1928 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1929 param); 1930 1931 switch (event) { 1932 case AudioSystem::OUTPUT_OPENED: 1933 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1934 desc.channelMask = mChannelMask; 1935 desc.samplingRate = mSampleRate; 1936 desc.format = mFormat; 1937 desc.frameCount = mNormalFrameCount; // FIXME see 1938 // AudioFlinger::frameCount(audio_io_handle_t) 1939 desc.latency = latency_l(); 1940 param2 = &desc; 1941 break; 1942 1943 case AudioSystem::STREAM_CONFIG_CHANGED: 1944 param2 = ¶m; 1945 case AudioSystem::OUTPUT_CLOSED: 1946 default: 1947 break; 1948 } 1949 mAudioFlinger->audioConfigChanged(event, mId, param2); 1950} 1951 1952void AudioFlinger::PlaybackThread::writeCallback() 1953{ 1954 ALOG_ASSERT(mCallbackThread != 0); 1955 mCallbackThread->resetWriteBlocked(); 1956} 1957 1958void AudioFlinger::PlaybackThread::drainCallback() 1959{ 1960 ALOG_ASSERT(mCallbackThread != 0); 1961 mCallbackThread->resetDraining(); 1962} 1963 1964void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1965{ 1966 Mutex::Autolock _l(mLock); 1967 // reject out of sequence requests 1968 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1969 mWriteAckSequence &= ~1; 1970 mWaitWorkCV.signal(); 1971 } 1972} 1973 1974void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1975{ 1976 Mutex::Autolock _l(mLock); 1977 // reject out of sequence requests 1978 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1979 mDrainSequence &= ~1; 1980 mWaitWorkCV.signal(); 1981 } 1982} 1983 1984// static 1985int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1986 void *param __unused, 1987 void *cookie) 1988{ 1989 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1990 ALOGV("asyncCallback() event %d", event); 1991 switch (event) { 1992 case STREAM_CBK_EVENT_WRITE_READY: 1993 me->writeCallback(); 1994 break; 1995 case STREAM_CBK_EVENT_DRAIN_READY: 1996 me->drainCallback(); 1997 break; 1998 default: 1999 ALOGW("asyncCallback() unknown event %d", event); 2000 break; 2001 } 2002 return 0; 2003} 2004 2005void AudioFlinger::PlaybackThread::readOutputParameters_l() 2006{ 2007 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2008 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2009 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2010 if (!audio_is_output_channel(mChannelMask)) { 2011 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2012 } 2013 if ((mType == MIXER || mType == DUPLICATING) 2014 && !isValidPcmSinkChannelMask(mChannelMask)) { 2015 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2016 mChannelMask); 2017 } 2018 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2019 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2020 mFormat = mHALFormat; 2021 if (!audio_is_valid_format(mFormat)) { 2022 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2023 } 2024 if ((mType == MIXER || mType == DUPLICATING) 2025 && !isValidPcmSinkFormat(mFormat)) { 2026 LOG_FATAL("HAL format %#x not supported for mixed output", 2027 mFormat); 2028 } 2029 mFrameSize = mOutput->getFrameSize(); 2030 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2031 mFrameCount = mBufferSize / mFrameSize; 2032 if (mFrameCount & 15) { 2033 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2034 mFrameCount); 2035 } 2036 2037 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2038 (mOutput->stream->set_callback != NULL)) { 2039 if (mOutput->stream->set_callback(mOutput->stream, 2040 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2041 mUseAsyncWrite = true; 2042 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2043 } 2044 } 2045 2046 mHwSupportsPause = false; 2047 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2048 if (mOutput->stream->pause != NULL) { 2049 if (mOutput->stream->resume != NULL) { 2050 mHwSupportsPause = true; 2051 } else { 2052 ALOGW("direct output implements pause but not resume"); 2053 } 2054 } else if (mOutput->stream->resume != NULL) { 2055 ALOGW("direct output implements resume but not pause"); 2056 } 2057 } 2058 2059 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2060 // For best precision, we use float instead of the associated output 2061 // device format (typically PCM 16 bit). 2062 2063 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2064 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2065 mBufferSize = mFrameSize * mFrameCount; 2066 2067 // TODO: We currently use the associated output device channel mask and sample rate. 2068 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2069 // (if a valid mask) to avoid premature downmix. 2070 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2071 // instead of the output device sample rate to avoid loss of high frequency information. 2072 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2073 } 2074 2075 // Calculate size of normal sink buffer relative to the HAL output buffer size 2076 double multiplier = 1.0; 2077 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2078 kUseFastMixer == FastMixer_Dynamic)) { 2079 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2080 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2081 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2082 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2083 maxNormalFrameCount = maxNormalFrameCount & ~15; 2084 if (maxNormalFrameCount < minNormalFrameCount) { 2085 maxNormalFrameCount = minNormalFrameCount; 2086 } 2087 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2088 if (multiplier <= 1.0) { 2089 multiplier = 1.0; 2090 } else if (multiplier <= 2.0) { 2091 if (2 * mFrameCount <= maxNormalFrameCount) { 2092 multiplier = 2.0; 2093 } else { 2094 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2095 } 2096 } else { 2097 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2098 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2099 // track, but we sometimes have to do this to satisfy the maximum frame count 2100 // constraint) 2101 // FIXME this rounding up should not be done if no HAL SRC 2102 uint32_t truncMult = (uint32_t) multiplier; 2103 if ((truncMult & 1)) { 2104 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2105 ++truncMult; 2106 } 2107 } 2108 multiplier = (double) truncMult; 2109 } 2110 } 2111 mNormalFrameCount = multiplier * mFrameCount; 2112 // round up to nearest 16 frames to satisfy AudioMixer 2113 if (mType == MIXER || mType == DUPLICATING) { 2114 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2115 } 2116 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2117 mNormalFrameCount); 2118 2119 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2120 // Originally this was int16_t[] array, need to remove legacy implications. 2121 free(mSinkBuffer); 2122 mSinkBuffer = NULL; 2123 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2124 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2125 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2126 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2127 2128 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2129 // drives the output. 2130 free(mMixerBuffer); 2131 mMixerBuffer = NULL; 2132 if (mMixerBufferEnabled) { 2133 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2134 mMixerBufferSize = mNormalFrameCount * mChannelCount 2135 * audio_bytes_per_sample(mMixerBufferFormat); 2136 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2137 } 2138 free(mEffectBuffer); 2139 mEffectBuffer = NULL; 2140 if (mEffectBufferEnabled) { 2141 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2142 mEffectBufferSize = mNormalFrameCount * mChannelCount 2143 * audio_bytes_per_sample(mEffectBufferFormat); 2144 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2145 } 2146 2147 // force reconfiguration of effect chains and engines to take new buffer size and audio 2148 // parameters into account 2149 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2150 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2151 // matter. 2152 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2153 Vector< sp<EffectChain> > effectChains = mEffectChains; 2154 for (size_t i = 0; i < effectChains.size(); i ++) { 2155 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2156 } 2157} 2158 2159 2160status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2161{ 2162 if (halFrames == NULL || dspFrames == NULL) { 2163 return BAD_VALUE; 2164 } 2165 Mutex::Autolock _l(mLock); 2166 if (initCheck() != NO_ERROR) { 2167 return INVALID_OPERATION; 2168 } 2169 size_t framesWritten = mBytesWritten / mFrameSize; 2170 *halFrames = framesWritten; 2171 2172 if (isSuspended()) { 2173 // return an estimation of rendered frames when the output is suspended 2174 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2175 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2176 return NO_ERROR; 2177 } else { 2178 status_t status; 2179 uint32_t frames; 2180 status = mOutput->getRenderPosition(&frames); 2181 *dspFrames = (size_t)frames; 2182 return status; 2183 } 2184} 2185 2186uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2187{ 2188 Mutex::Autolock _l(mLock); 2189 uint32_t result = 0; 2190 if (getEffectChain_l(sessionId) != 0) { 2191 result = EFFECT_SESSION; 2192 } 2193 2194 for (size_t i = 0; i < mTracks.size(); ++i) { 2195 sp<Track> track = mTracks[i]; 2196 if (sessionId == track->sessionId() && !track->isInvalid()) { 2197 result |= TRACK_SESSION; 2198 break; 2199 } 2200 } 2201 2202 return result; 2203} 2204 2205uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2206{ 2207 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2208 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2209 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2210 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2211 } 2212 for (size_t i = 0; i < mTracks.size(); i++) { 2213 sp<Track> track = mTracks[i]; 2214 if (sessionId == track->sessionId() && !track->isInvalid()) { 2215 return AudioSystem::getStrategyForStream(track->streamType()); 2216 } 2217 } 2218 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2219} 2220 2221 2222AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2223{ 2224 Mutex::Autolock _l(mLock); 2225 return mOutput; 2226} 2227 2228AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2229{ 2230 Mutex::Autolock _l(mLock); 2231 AudioStreamOut *output = mOutput; 2232 mOutput = NULL; 2233 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2234 // must push a NULL and wait for ack 2235 mOutputSink.clear(); 2236 mPipeSink.clear(); 2237 mNormalSink.clear(); 2238 return output; 2239} 2240 2241// this method must always be called either with ThreadBase mLock held or inside the thread loop 2242audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2243{ 2244 if (mOutput == NULL) { 2245 return NULL; 2246 } 2247 return &mOutput->stream->common; 2248} 2249 2250uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2251{ 2252 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2253} 2254 2255status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2256{ 2257 if (!isValidSyncEvent(event)) { 2258 return BAD_VALUE; 2259 } 2260 2261 Mutex::Autolock _l(mLock); 2262 2263 for (size_t i = 0; i < mTracks.size(); ++i) { 2264 sp<Track> track = mTracks[i]; 2265 if (event->triggerSession() == track->sessionId()) { 2266 (void) track->setSyncEvent(event); 2267 return NO_ERROR; 2268 } 2269 } 2270 2271 return NAME_NOT_FOUND; 2272} 2273 2274bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2275{ 2276 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2277} 2278 2279void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2280 const Vector< sp<Track> >& tracksToRemove) 2281{ 2282 size_t count = tracksToRemove.size(); 2283 if (count > 0) { 2284 for (size_t i = 0 ; i < count ; i++) { 2285 const sp<Track>& track = tracksToRemove.itemAt(i); 2286 if (track->isExternalTrack()) { 2287 AudioSystem::stopOutput(mId, track->streamType(), 2288 (audio_session_t)track->sessionId()); 2289#ifdef ADD_BATTERY_DATA 2290 // to track the speaker usage 2291 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2292#endif 2293 if (track->isTerminated()) { 2294 AudioSystem::releaseOutput(mId, track->streamType(), 2295 (audio_session_t)track->sessionId()); 2296 } 2297 } 2298 } 2299 } 2300} 2301 2302void AudioFlinger::PlaybackThread::checkSilentMode_l() 2303{ 2304 if (!mMasterMute) { 2305 char value[PROPERTY_VALUE_MAX]; 2306 if (property_get("ro.audio.silent", value, "0") > 0) { 2307 char *endptr; 2308 unsigned long ul = strtoul(value, &endptr, 0); 2309 if (*endptr == '\0' && ul != 0) { 2310 ALOGD("Silence is golden"); 2311 // The setprop command will not allow a property to be changed after 2312 // the first time it is set, so we don't have to worry about un-muting. 2313 setMasterMute_l(true); 2314 } 2315 } 2316 } 2317} 2318 2319// shared by MIXER and DIRECT, overridden by DUPLICATING 2320ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2321{ 2322 // FIXME rewrite to reduce number of system calls 2323 mLastWriteTime = systemTime(); 2324 mInWrite = true; 2325 ssize_t bytesWritten; 2326 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2327 2328 // If an NBAIO sink is present, use it to write the normal mixer's submix 2329 if (mNormalSink != 0) { 2330 2331 const size_t count = mBytesRemaining / mFrameSize; 2332 2333 ATRACE_BEGIN("write"); 2334 // update the setpoint when AudioFlinger::mScreenState changes 2335 uint32_t screenState = AudioFlinger::mScreenState; 2336 if (screenState != mScreenState) { 2337 mScreenState = screenState; 2338 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2339 if (pipe != NULL) { 2340 pipe->setAvgFrames((mScreenState & 1) ? 2341 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2342 } 2343 } 2344 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2345 ATRACE_END(); 2346 if (framesWritten > 0) { 2347 bytesWritten = framesWritten * mFrameSize; 2348 } else { 2349 bytesWritten = framesWritten; 2350 } 2351 mLatchDValid = false; 2352 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2353 if (status == NO_ERROR) { 2354 size_t totalFramesWritten = mNormalSink->framesWritten(); 2355 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2356 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2357 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2358 mLatchDValid = true; 2359 } 2360 } 2361 // otherwise use the HAL / AudioStreamOut directly 2362 } else { 2363 // Direct output and offload threads 2364 2365 if (mUseAsyncWrite) { 2366 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2367 mWriteAckSequence += 2; 2368 mWriteAckSequence |= 1; 2369 ALOG_ASSERT(mCallbackThread != 0); 2370 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2371 } 2372 // FIXME We should have an implementation of timestamps for direct output threads. 2373 // They are used e.g for multichannel PCM playback over HDMI. 2374 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2375 if (mUseAsyncWrite && 2376 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2377 // do not wait for async callback in case of error of full write 2378 mWriteAckSequence &= ~1; 2379 ALOG_ASSERT(mCallbackThread != 0); 2380 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2381 } 2382 } 2383 2384 mNumWrites++; 2385 mInWrite = false; 2386 mStandby = false; 2387 return bytesWritten; 2388} 2389 2390void AudioFlinger::PlaybackThread::threadLoop_drain() 2391{ 2392 if (mOutput->stream->drain) { 2393 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2394 if (mUseAsyncWrite) { 2395 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2396 mDrainSequence |= 1; 2397 ALOG_ASSERT(mCallbackThread != 0); 2398 mCallbackThread->setDraining(mDrainSequence); 2399 } 2400 mOutput->stream->drain(mOutput->stream, 2401 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2402 : AUDIO_DRAIN_ALL); 2403 } 2404} 2405 2406void AudioFlinger::PlaybackThread::threadLoop_exit() 2407{ 2408 { 2409 Mutex::Autolock _l(mLock); 2410 for (size_t i = 0; i < mTracks.size(); i++) { 2411 sp<Track> track = mTracks[i]; 2412 track->invalidate(); 2413 } 2414 } 2415} 2416 2417/* 2418The derived values that are cached: 2419 - mSinkBufferSize from frame count * frame size 2420 - activeSleepTime from activeSleepTimeUs() 2421 - idleSleepTime from idleSleepTimeUs() 2422 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2423 - maxPeriod from frame count and sample rate (MIXER only) 2424 2425The parameters that affect these derived values are: 2426 - frame count 2427 - frame size 2428 - sample rate 2429 - device type: A2DP or not 2430 - device latency 2431 - format: PCM or not 2432 - active sleep time 2433 - idle sleep time 2434*/ 2435 2436void AudioFlinger::PlaybackThread::cacheParameters_l() 2437{ 2438 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2439 activeSleepTime = activeSleepTimeUs(); 2440 idleSleepTime = idleSleepTimeUs(); 2441} 2442 2443void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2444{ 2445 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2446 this, streamType, mTracks.size()); 2447 Mutex::Autolock _l(mLock); 2448 2449 size_t size = mTracks.size(); 2450 for (size_t i = 0; i < size; i++) { 2451 sp<Track> t = mTracks[i]; 2452 if (t->streamType() == streamType) { 2453 t->invalidate(); 2454 } 2455 } 2456} 2457 2458status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2459{ 2460 int session = chain->sessionId(); 2461 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2462 ? mEffectBuffer : mSinkBuffer); 2463 bool ownsBuffer = false; 2464 2465 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2466 if (session > 0) { 2467 // Only one effect chain can be present in direct output thread and it uses 2468 // the sink buffer as input 2469 if (mType != DIRECT) { 2470 size_t numSamples = mNormalFrameCount * mChannelCount; 2471 buffer = new int16_t[numSamples]; 2472 memset(buffer, 0, numSamples * sizeof(int16_t)); 2473 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2474 ownsBuffer = true; 2475 } 2476 2477 // Attach all tracks with same session ID to this chain. 2478 for (size_t i = 0; i < mTracks.size(); ++i) { 2479 sp<Track> track = mTracks[i]; 2480 if (session == track->sessionId()) { 2481 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2482 buffer); 2483 track->setMainBuffer(buffer); 2484 chain->incTrackCnt(); 2485 } 2486 } 2487 2488 // indicate all active tracks in the chain 2489 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2490 sp<Track> track = mActiveTracks[i].promote(); 2491 if (track == 0) { 2492 continue; 2493 } 2494 if (session == track->sessionId()) { 2495 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2496 chain->incActiveTrackCnt(); 2497 } 2498 } 2499 } 2500 chain->setThread(this); 2501 chain->setInBuffer(buffer, ownsBuffer); 2502 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2503 ? mEffectBuffer : mSinkBuffer)); 2504 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2505 // chains list in order to be processed last as it contains output stage effects 2506 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2507 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2508 // after track specific effects and before output stage 2509 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2510 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2511 // Effect chain for other sessions are inserted at beginning of effect 2512 // chains list to be processed before output mix effects. Relative order between other 2513 // sessions is not important 2514 size_t size = mEffectChains.size(); 2515 size_t i = 0; 2516 for (i = 0; i < size; i++) { 2517 if (mEffectChains[i]->sessionId() < session) { 2518 break; 2519 } 2520 } 2521 mEffectChains.insertAt(chain, i); 2522 checkSuspendOnAddEffectChain_l(chain); 2523 2524 return NO_ERROR; 2525} 2526 2527size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2528{ 2529 int session = chain->sessionId(); 2530 2531 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2532 2533 for (size_t i = 0; i < mEffectChains.size(); i++) { 2534 if (chain == mEffectChains[i]) { 2535 mEffectChains.removeAt(i); 2536 // detach all active tracks from the chain 2537 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2538 sp<Track> track = mActiveTracks[i].promote(); 2539 if (track == 0) { 2540 continue; 2541 } 2542 if (session == track->sessionId()) { 2543 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2544 chain.get(), session); 2545 chain->decActiveTrackCnt(); 2546 } 2547 } 2548 2549 // detach all tracks with same session ID from this chain 2550 for (size_t i = 0; i < mTracks.size(); ++i) { 2551 sp<Track> track = mTracks[i]; 2552 if (session == track->sessionId()) { 2553 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2554 chain->decTrackCnt(); 2555 } 2556 } 2557 break; 2558 } 2559 } 2560 return mEffectChains.size(); 2561} 2562 2563status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2564 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2565{ 2566 Mutex::Autolock _l(mLock); 2567 return attachAuxEffect_l(track, EffectId); 2568} 2569 2570status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2571 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2572{ 2573 status_t status = NO_ERROR; 2574 2575 if (EffectId == 0) { 2576 track->setAuxBuffer(0, NULL); 2577 } else { 2578 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2579 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2580 if (effect != 0) { 2581 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2582 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2583 } else { 2584 status = INVALID_OPERATION; 2585 } 2586 } else { 2587 status = BAD_VALUE; 2588 } 2589 } 2590 return status; 2591} 2592 2593void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2594{ 2595 for (size_t i = 0; i < mTracks.size(); ++i) { 2596 sp<Track> track = mTracks[i]; 2597 if (track->auxEffectId() == effectId) { 2598 attachAuxEffect_l(track, 0); 2599 } 2600 } 2601} 2602 2603bool AudioFlinger::PlaybackThread::threadLoop() 2604{ 2605 Vector< sp<Track> > tracksToRemove; 2606 2607 standbyTime = systemTime(); 2608 2609 // MIXER 2610 nsecs_t lastWarning = 0; 2611 2612 // DUPLICATING 2613 // FIXME could this be made local to while loop? 2614 writeFrames = 0; 2615 2616 int lastGeneration = 0; 2617 2618 cacheParameters_l(); 2619 sleepTime = idleSleepTime; 2620 2621 if (mType == MIXER) { 2622 sleepTimeShift = 0; 2623 } 2624 2625 CpuStats cpuStats; 2626 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2627 2628 acquireWakeLock(); 2629 2630 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2631 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2632 // and then that string will be logged at the next convenient opportunity. 2633 const char *logString = NULL; 2634 2635 checkSilentMode_l(); 2636 2637 while (!exitPending()) 2638 { 2639 cpuStats.sample(myName); 2640 2641 Vector< sp<EffectChain> > effectChains; 2642 2643 { // scope for mLock 2644 2645 Mutex::Autolock _l(mLock); 2646 2647 processConfigEvents_l(); 2648 2649 if (logString != NULL) { 2650 mNBLogWriter->logTimestamp(); 2651 mNBLogWriter->log(logString); 2652 logString = NULL; 2653 } 2654 2655 // Gather the framesReleased counters for all active tracks, 2656 // and latch them atomically with the timestamp. 2657 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2658 mLatchD.mFramesReleased.clear(); 2659 size_t size = mActiveTracks.size(); 2660 for (size_t i = 0; i < size; i++) { 2661 sp<Track> t = mActiveTracks[i].promote(); 2662 if (t != 0) { 2663 mLatchD.mFramesReleased.add(t.get(), 2664 t->mAudioTrackServerProxy->framesReleased()); 2665 } 2666 } 2667 if (mLatchDValid) { 2668 mLatchQ = mLatchD; 2669 mLatchDValid = false; 2670 mLatchQValid = true; 2671 } 2672 2673 saveOutputTracks(); 2674 if (mSignalPending) { 2675 // A signal was raised while we were unlocked 2676 mSignalPending = false; 2677 } else if (waitingAsyncCallback_l()) { 2678 if (exitPending()) { 2679 break; 2680 } 2681 releaseWakeLock_l(); 2682 mWakeLockUids.clear(); 2683 mActiveTracksGeneration++; 2684 ALOGV("wait async completion"); 2685 mWaitWorkCV.wait(mLock); 2686 ALOGV("async completion/wake"); 2687 acquireWakeLock_l(); 2688 standbyTime = systemTime() + standbyDelay; 2689 sleepTime = 0; 2690 2691 continue; 2692 } 2693 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2694 isSuspended()) { 2695 // put audio hardware into standby after short delay 2696 if (shouldStandby_l()) { 2697 2698 threadLoop_standby(); 2699 2700 mStandby = true; 2701 } 2702 2703 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2704 // we're about to wait, flush the binder command buffer 2705 IPCThreadState::self()->flushCommands(); 2706 2707 clearOutputTracks(); 2708 2709 if (exitPending()) { 2710 break; 2711 } 2712 2713 releaseWakeLock_l(); 2714 mWakeLockUids.clear(); 2715 mActiveTracksGeneration++; 2716 // wait until we have something to do... 2717 ALOGV("%s going to sleep", myName.string()); 2718 mWaitWorkCV.wait(mLock); 2719 ALOGV("%s waking up", myName.string()); 2720 acquireWakeLock_l(); 2721 2722 mMixerStatus = MIXER_IDLE; 2723 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2724 mBytesWritten = 0; 2725 mBytesRemaining = 0; 2726 checkSilentMode_l(); 2727 2728 standbyTime = systemTime() + standbyDelay; 2729 sleepTime = idleSleepTime; 2730 if (mType == MIXER) { 2731 sleepTimeShift = 0; 2732 } 2733 2734 continue; 2735 } 2736 } 2737 // mMixerStatusIgnoringFastTracks is also updated internally 2738 mMixerStatus = prepareTracks_l(&tracksToRemove); 2739 2740 // compare with previously applied list 2741 if (lastGeneration != mActiveTracksGeneration) { 2742 // update wakelock 2743 updateWakeLockUids_l(mWakeLockUids); 2744 lastGeneration = mActiveTracksGeneration; 2745 } 2746 2747 // prevent any changes in effect chain list and in each effect chain 2748 // during mixing and effect process as the audio buffers could be deleted 2749 // or modified if an effect is created or deleted 2750 lockEffectChains_l(effectChains); 2751 } // mLock scope ends 2752 2753 if (mBytesRemaining == 0) { 2754 mCurrentWriteLength = 0; 2755 if (mMixerStatus == MIXER_TRACKS_READY) { 2756 // threadLoop_mix() sets mCurrentWriteLength 2757 threadLoop_mix(); 2758 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2759 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2760 // threadLoop_sleepTime sets sleepTime to 0 if data 2761 // must be written to HAL 2762 threadLoop_sleepTime(); 2763 if (sleepTime == 0) { 2764 mCurrentWriteLength = mSinkBufferSize; 2765 } 2766 } 2767 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2768 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2769 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2770 // or mSinkBuffer (if there are no effects). 2771 // 2772 // This is done pre-effects computation; if effects change to 2773 // support higher precision, this needs to move. 2774 // 2775 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2776 // TODO use sleepTime == 0 as an additional condition. 2777 if (mMixerBufferValid) { 2778 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2779 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2780 2781 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2782 mNormalFrameCount * mChannelCount); 2783 } 2784 2785 mBytesRemaining = mCurrentWriteLength; 2786 if (isSuspended()) { 2787 sleepTime = suspendSleepTimeUs(); 2788 // simulate write to HAL when suspended 2789 mBytesWritten += mSinkBufferSize; 2790 mBytesRemaining = 0; 2791 } 2792 2793 // only process effects if we're going to write 2794 if (sleepTime == 0 && mType != OFFLOAD) { 2795 for (size_t i = 0; i < effectChains.size(); i ++) { 2796 effectChains[i]->process_l(); 2797 } 2798 } 2799 } 2800 // Process effect chains for offloaded thread even if no audio 2801 // was read from audio track: process only updates effect state 2802 // and thus does have to be synchronized with audio writes but may have 2803 // to be called while waiting for async write callback 2804 if (mType == OFFLOAD) { 2805 for (size_t i = 0; i < effectChains.size(); i ++) { 2806 effectChains[i]->process_l(); 2807 } 2808 } 2809 2810 // Only if the Effects buffer is enabled and there is data in the 2811 // Effects buffer (buffer valid), we need to 2812 // copy into the sink buffer. 2813 // TODO use sleepTime == 0 as an additional condition. 2814 if (mEffectBufferValid) { 2815 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2816 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2817 mNormalFrameCount * mChannelCount); 2818 } 2819 2820 // enable changes in effect chain 2821 unlockEffectChains(effectChains); 2822 2823 if (!waitingAsyncCallback()) { 2824 // sleepTime == 0 means we must write to audio hardware 2825 if (sleepTime == 0) { 2826 if (mBytesRemaining) { 2827 ssize_t ret = threadLoop_write(); 2828 if (ret < 0) { 2829 mBytesRemaining = 0; 2830 } else { 2831 mBytesWritten += ret; 2832 mBytesRemaining -= ret; 2833 } 2834 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2835 (mMixerStatus == MIXER_DRAIN_ALL)) { 2836 threadLoop_drain(); 2837 } 2838 if (mType == MIXER) { 2839 // write blocked detection 2840 nsecs_t now = systemTime(); 2841 nsecs_t delta = now - mLastWriteTime; 2842 if (!mStandby && delta > maxPeriod) { 2843 mNumDelayedWrites++; 2844 if ((now - lastWarning) > kWarningThrottleNs) { 2845 ATRACE_NAME("underrun"); 2846 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2847 ns2ms(delta), mNumDelayedWrites, this); 2848 lastWarning = now; 2849 } 2850 } 2851 } 2852 2853 } else { 2854 ATRACE_BEGIN("sleep"); 2855 usleep(sleepTime); 2856 ATRACE_END(); 2857 } 2858 } 2859 2860 // Finally let go of removed track(s), without the lock held 2861 // since we can't guarantee the destructors won't acquire that 2862 // same lock. This will also mutate and push a new fast mixer state. 2863 threadLoop_removeTracks(tracksToRemove); 2864 tracksToRemove.clear(); 2865 2866 // FIXME I don't understand the need for this here; 2867 // it was in the original code but maybe the 2868 // assignment in saveOutputTracks() makes this unnecessary? 2869 clearOutputTracks(); 2870 2871 // Effect chains will be actually deleted here if they were removed from 2872 // mEffectChains list during mixing or effects processing 2873 effectChains.clear(); 2874 2875 // FIXME Note that the above .clear() is no longer necessary since effectChains 2876 // is now local to this block, but will keep it for now (at least until merge done). 2877 } 2878 2879 threadLoop_exit(); 2880 2881 if (!mStandby) { 2882 threadLoop_standby(); 2883 mStandby = true; 2884 } 2885 2886 releaseWakeLock(); 2887 mWakeLockUids.clear(); 2888 mActiveTracksGeneration++; 2889 2890 ALOGV("Thread %p type %d exiting", this, mType); 2891 return false; 2892} 2893 2894// removeTracks_l() must be called with ThreadBase::mLock held 2895void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2896{ 2897 size_t count = tracksToRemove.size(); 2898 if (count > 0) { 2899 for (size_t i=0 ; i<count ; i++) { 2900 const sp<Track>& track = tracksToRemove.itemAt(i); 2901 mActiveTracks.remove(track); 2902 mWakeLockUids.remove(track->uid()); 2903 mActiveTracksGeneration++; 2904 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2905 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2906 if (chain != 0) { 2907 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2908 track->sessionId()); 2909 chain->decActiveTrackCnt(); 2910 } 2911 if (track->isTerminated()) { 2912 removeTrack_l(track); 2913 } 2914 } 2915 } 2916 2917} 2918 2919status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2920{ 2921 if (mNormalSink != 0) { 2922 return mNormalSink->getTimestamp(timestamp); 2923 } 2924 if ((mType == OFFLOAD || mType == DIRECT) 2925 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2926 uint64_t position64; 2927 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 2928 if (ret == 0) { 2929 timestamp.mPosition = (uint32_t)position64; 2930 return NO_ERROR; 2931 } 2932 } 2933 return INVALID_OPERATION; 2934} 2935 2936status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2937 audio_patch_handle_t *handle) 2938{ 2939 status_t status = NO_ERROR; 2940 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2941 // store new device and send to effects 2942 audio_devices_t type = AUDIO_DEVICE_NONE; 2943 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2944 type |= patch->sinks[i].ext.device.type; 2945 } 2946 mOutDevice = type; 2947 for (size_t i = 0; i < mEffectChains.size(); i++) { 2948 mEffectChains[i]->setDevice_l(mOutDevice); 2949 } 2950 2951 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2952 status = hwDevice->create_audio_patch(hwDevice, 2953 patch->num_sources, 2954 patch->sources, 2955 patch->num_sinks, 2956 patch->sinks, 2957 handle); 2958 } else { 2959 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2960 } 2961 return status; 2962} 2963 2964status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2965{ 2966 status_t status = NO_ERROR; 2967 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2968 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2969 status = hwDevice->release_audio_patch(hwDevice, handle); 2970 } else { 2971 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2972 } 2973 return status; 2974} 2975 2976void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2977{ 2978 Mutex::Autolock _l(mLock); 2979 mTracks.add(track); 2980} 2981 2982void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2983{ 2984 Mutex::Autolock _l(mLock); 2985 destroyTrack_l(track); 2986} 2987 2988void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2989{ 2990 ThreadBase::getAudioPortConfig(config); 2991 config->role = AUDIO_PORT_ROLE_SOURCE; 2992 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2993 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2994} 2995 2996// ---------------------------------------------------------------------------- 2997 2998AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2999 audio_io_handle_t id, audio_devices_t device, type_t type) 3000 : PlaybackThread(audioFlinger, output, id, device, type), 3001 // mAudioMixer below 3002 // mFastMixer below 3003 mFastMixerFutex(0) 3004 // mOutputSink below 3005 // mPipeSink below 3006 // mNormalSink below 3007{ 3008 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3009 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3010 "mFrameCount=%d, mNormalFrameCount=%d", 3011 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3012 mNormalFrameCount); 3013 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3014 3015 if (type == DUPLICATING) { 3016 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3017 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3018 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3019 return; 3020 } 3021 // create an NBAIO sink for the HAL output stream, and negotiate 3022 mOutputSink = new AudioStreamOutSink(output->stream); 3023 size_t numCounterOffers = 0; 3024 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3025 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3026 ALOG_ASSERT(index == 0); 3027 3028 // initialize fast mixer depending on configuration 3029 bool initFastMixer; 3030 switch (kUseFastMixer) { 3031 case FastMixer_Never: 3032 initFastMixer = false; 3033 break; 3034 case FastMixer_Always: 3035 initFastMixer = true; 3036 break; 3037 case FastMixer_Static: 3038 case FastMixer_Dynamic: 3039 initFastMixer = mFrameCount < mNormalFrameCount; 3040 break; 3041 } 3042 if (initFastMixer) { 3043 audio_format_t fastMixerFormat; 3044 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3045 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3046 } else { 3047 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3048 } 3049 if (mFormat != fastMixerFormat) { 3050 // change our Sink format to accept our intermediate precision 3051 mFormat = fastMixerFormat; 3052 free(mSinkBuffer); 3053 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3054 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3055 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3056 } 3057 3058 // create a MonoPipe to connect our submix to FastMixer 3059 NBAIO_Format format = mOutputSink->format(); 3060 NBAIO_Format origformat = format; 3061 // adjust format to match that of the Fast Mixer 3062 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3063 format.mFormat = fastMixerFormat; 3064 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3065 3066 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3067 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3068 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3069 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3070 const NBAIO_Format offers[1] = {format}; 3071 size_t numCounterOffers = 0; 3072 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3073 ALOG_ASSERT(index == 0); 3074 monoPipe->setAvgFrames((mScreenState & 1) ? 3075 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3076 mPipeSink = monoPipe; 3077 3078#ifdef TEE_SINK 3079 if (mTeeSinkOutputEnabled) { 3080 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3081 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3082 const NBAIO_Format offers2[1] = {origformat}; 3083 numCounterOffers = 0; 3084 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3085 ALOG_ASSERT(index == 0); 3086 mTeeSink = teeSink; 3087 PipeReader *teeSource = new PipeReader(*teeSink); 3088 numCounterOffers = 0; 3089 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3090 ALOG_ASSERT(index == 0); 3091 mTeeSource = teeSource; 3092 } 3093#endif 3094 3095 // create fast mixer and configure it initially with just one fast track for our submix 3096 mFastMixer = new FastMixer(); 3097 FastMixerStateQueue *sq = mFastMixer->sq(); 3098#ifdef STATE_QUEUE_DUMP 3099 sq->setObserverDump(&mStateQueueObserverDump); 3100 sq->setMutatorDump(&mStateQueueMutatorDump); 3101#endif 3102 FastMixerState *state = sq->begin(); 3103 FastTrack *fastTrack = &state->mFastTracks[0]; 3104 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3105 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3106 fastTrack->mVolumeProvider = NULL; 3107 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3108 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3109 fastTrack->mGeneration++; 3110 state->mFastTracksGen++; 3111 state->mTrackMask = 1; 3112 // fast mixer will use the HAL output sink 3113 state->mOutputSink = mOutputSink.get(); 3114 state->mOutputSinkGen++; 3115 state->mFrameCount = mFrameCount; 3116 state->mCommand = FastMixerState::COLD_IDLE; 3117 // already done in constructor initialization list 3118 //mFastMixerFutex = 0; 3119 state->mColdFutexAddr = &mFastMixerFutex; 3120 state->mColdGen++; 3121 state->mDumpState = &mFastMixerDumpState; 3122#ifdef TEE_SINK 3123 state->mTeeSink = mTeeSink.get(); 3124#endif 3125 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3126 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3127 sq->end(); 3128 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3129 3130 // start the fast mixer 3131 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3132 pid_t tid = mFastMixer->getTid(); 3133 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3134 if (err != 0) { 3135 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3136 kPriorityFastMixer, getpid_cached, tid, err); 3137 } 3138 3139#ifdef AUDIO_WATCHDOG 3140 // create and start the watchdog 3141 mAudioWatchdog = new AudioWatchdog(); 3142 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3143 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3144 tid = mAudioWatchdog->getTid(); 3145 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3146 if (err != 0) { 3147 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3148 kPriorityFastMixer, getpid_cached, tid, err); 3149 } 3150#endif 3151 3152 } 3153 3154 switch (kUseFastMixer) { 3155 case FastMixer_Never: 3156 case FastMixer_Dynamic: 3157 mNormalSink = mOutputSink; 3158 break; 3159 case FastMixer_Always: 3160 mNormalSink = mPipeSink; 3161 break; 3162 case FastMixer_Static: 3163 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3164 break; 3165 } 3166} 3167 3168AudioFlinger::MixerThread::~MixerThread() 3169{ 3170 if (mFastMixer != 0) { 3171 FastMixerStateQueue *sq = mFastMixer->sq(); 3172 FastMixerState *state = sq->begin(); 3173 if (state->mCommand == FastMixerState::COLD_IDLE) { 3174 int32_t old = android_atomic_inc(&mFastMixerFutex); 3175 if (old == -1) { 3176 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3177 } 3178 } 3179 state->mCommand = FastMixerState::EXIT; 3180 sq->end(); 3181 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3182 mFastMixer->join(); 3183 // Though the fast mixer thread has exited, it's state queue is still valid. 3184 // We'll use that extract the final state which contains one remaining fast track 3185 // corresponding to our sub-mix. 3186 state = sq->begin(); 3187 ALOG_ASSERT(state->mTrackMask == 1); 3188 FastTrack *fastTrack = &state->mFastTracks[0]; 3189 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3190 delete fastTrack->mBufferProvider; 3191 sq->end(false /*didModify*/); 3192 mFastMixer.clear(); 3193#ifdef AUDIO_WATCHDOG 3194 if (mAudioWatchdog != 0) { 3195 mAudioWatchdog->requestExit(); 3196 mAudioWatchdog->requestExitAndWait(); 3197 mAudioWatchdog.clear(); 3198 } 3199#endif 3200 } 3201 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3202 delete mAudioMixer; 3203} 3204 3205 3206uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3207{ 3208 if (mFastMixer != 0) { 3209 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3210 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3211 } 3212 return latency; 3213} 3214 3215 3216void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3217{ 3218 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3219} 3220 3221ssize_t AudioFlinger::MixerThread::threadLoop_write() 3222{ 3223 // FIXME we should only do one push per cycle; confirm this is true 3224 // Start the fast mixer if it's not already running 3225 if (mFastMixer != 0) { 3226 FastMixerStateQueue *sq = mFastMixer->sq(); 3227 FastMixerState *state = sq->begin(); 3228 if (state->mCommand != FastMixerState::MIX_WRITE && 3229 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3230 if (state->mCommand == FastMixerState::COLD_IDLE) { 3231 int32_t old = android_atomic_inc(&mFastMixerFutex); 3232 if (old == -1) { 3233 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3234 } 3235#ifdef AUDIO_WATCHDOG 3236 if (mAudioWatchdog != 0) { 3237 mAudioWatchdog->resume(); 3238 } 3239#endif 3240 } 3241 state->mCommand = FastMixerState::MIX_WRITE; 3242#ifdef FAST_THREAD_STATISTICS 3243 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3244 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3245#endif 3246 sq->end(); 3247 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3248 if (kUseFastMixer == FastMixer_Dynamic) { 3249 mNormalSink = mPipeSink; 3250 } 3251 } else { 3252 sq->end(false /*didModify*/); 3253 } 3254 } 3255 return PlaybackThread::threadLoop_write(); 3256} 3257 3258void AudioFlinger::MixerThread::threadLoop_standby() 3259{ 3260 // Idle the fast mixer if it's currently running 3261 if (mFastMixer != 0) { 3262 FastMixerStateQueue *sq = mFastMixer->sq(); 3263 FastMixerState *state = sq->begin(); 3264 if (!(state->mCommand & FastMixerState::IDLE)) { 3265 state->mCommand = FastMixerState::COLD_IDLE; 3266 state->mColdFutexAddr = &mFastMixerFutex; 3267 state->mColdGen++; 3268 mFastMixerFutex = 0; 3269 sq->end(); 3270 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3271 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3272 if (kUseFastMixer == FastMixer_Dynamic) { 3273 mNormalSink = mOutputSink; 3274 } 3275#ifdef AUDIO_WATCHDOG 3276 if (mAudioWatchdog != 0) { 3277 mAudioWatchdog->pause(); 3278 } 3279#endif 3280 } else { 3281 sq->end(false /*didModify*/); 3282 } 3283 } 3284 PlaybackThread::threadLoop_standby(); 3285} 3286 3287bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3288{ 3289 return false; 3290} 3291 3292bool AudioFlinger::PlaybackThread::shouldStandby_l() 3293{ 3294 return !mStandby; 3295} 3296 3297bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3298{ 3299 Mutex::Autolock _l(mLock); 3300 return waitingAsyncCallback_l(); 3301} 3302 3303// shared by MIXER and DIRECT, overridden by DUPLICATING 3304void AudioFlinger::PlaybackThread::threadLoop_standby() 3305{ 3306 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3307 mOutput->standby(); 3308 if (mUseAsyncWrite != 0) { 3309 // discard any pending drain or write ack by incrementing sequence 3310 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3311 mDrainSequence = (mDrainSequence + 2) & ~1; 3312 ALOG_ASSERT(mCallbackThread != 0); 3313 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3314 mCallbackThread->setDraining(mDrainSequence); 3315 } 3316 mHwPaused = false; 3317} 3318 3319void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3320{ 3321 ALOGV("signal playback thread"); 3322 broadcast_l(); 3323} 3324 3325void AudioFlinger::MixerThread::threadLoop_mix() 3326{ 3327 // obtain the presentation timestamp of the next output buffer 3328 int64_t pts; 3329 status_t status = INVALID_OPERATION; 3330 3331 if (mNormalSink != 0) { 3332 status = mNormalSink->getNextWriteTimestamp(&pts); 3333 } else { 3334 status = mOutputSink->getNextWriteTimestamp(&pts); 3335 } 3336 3337 if (status != NO_ERROR) { 3338 pts = AudioBufferProvider::kInvalidPTS; 3339 } 3340 3341 // mix buffers... 3342 mAudioMixer->process(pts); 3343 mCurrentWriteLength = mSinkBufferSize; 3344 // increase sleep time progressively when application underrun condition clears. 3345 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3346 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3347 // such that we would underrun the audio HAL. 3348 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3349 sleepTimeShift--; 3350 } 3351 sleepTime = 0; 3352 standbyTime = systemTime() + standbyDelay; 3353 //TODO: delay standby when effects have a tail 3354 3355} 3356 3357void AudioFlinger::MixerThread::threadLoop_sleepTime() 3358{ 3359 // If no tracks are ready, sleep once for the duration of an output 3360 // buffer size, then write 0s to the output 3361 if (sleepTime == 0) { 3362 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3363 sleepTime = activeSleepTime >> sleepTimeShift; 3364 if (sleepTime < kMinThreadSleepTimeUs) { 3365 sleepTime = kMinThreadSleepTimeUs; 3366 } 3367 // reduce sleep time in case of consecutive application underruns to avoid 3368 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3369 // duration we would end up writing less data than needed by the audio HAL if 3370 // the condition persists. 3371 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3372 sleepTimeShift++; 3373 } 3374 } else { 3375 sleepTime = idleSleepTime; 3376 } 3377 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3378 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3379 // before effects processing or output. 3380 if (mMixerBufferValid) { 3381 memset(mMixerBuffer, 0, mMixerBufferSize); 3382 } else { 3383 memset(mSinkBuffer, 0, mSinkBufferSize); 3384 } 3385 sleepTime = 0; 3386 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3387 "anticipated start"); 3388 } 3389 // TODO add standby time extension fct of effect tail 3390} 3391 3392// prepareTracks_l() must be called with ThreadBase::mLock held 3393AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3394 Vector< sp<Track> > *tracksToRemove) 3395{ 3396 3397 mixer_state mixerStatus = MIXER_IDLE; 3398 // find out which tracks need to be processed 3399 size_t count = mActiveTracks.size(); 3400 size_t mixedTracks = 0; 3401 size_t tracksWithEffect = 0; 3402 // counts only _active_ fast tracks 3403 size_t fastTracks = 0; 3404 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3405 3406 float masterVolume = mMasterVolume; 3407 bool masterMute = mMasterMute; 3408 3409 if (masterMute) { 3410 masterVolume = 0; 3411 } 3412 // Delegate master volume control to effect in output mix effect chain if needed 3413 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3414 if (chain != 0) { 3415 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3416 chain->setVolume_l(&v, &v); 3417 masterVolume = (float)((v + (1 << 23)) >> 24); 3418 chain.clear(); 3419 } 3420 3421 // prepare a new state to push 3422 FastMixerStateQueue *sq = NULL; 3423 FastMixerState *state = NULL; 3424 bool didModify = false; 3425 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3426 if (mFastMixer != 0) { 3427 sq = mFastMixer->sq(); 3428 state = sq->begin(); 3429 } 3430 3431 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3432 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3433 3434 for (size_t i=0 ; i<count ; i++) { 3435 const sp<Track> t = mActiveTracks[i].promote(); 3436 if (t == 0) { 3437 continue; 3438 } 3439 3440 // this const just means the local variable doesn't change 3441 Track* const track = t.get(); 3442 3443 // process fast tracks 3444 if (track->isFastTrack()) { 3445 3446 // It's theoretically possible (though unlikely) for a fast track to be created 3447 // and then removed within the same normal mix cycle. This is not a problem, as 3448 // the track never becomes active so it's fast mixer slot is never touched. 3449 // The converse, of removing an (active) track and then creating a new track 3450 // at the identical fast mixer slot within the same normal mix cycle, 3451 // is impossible because the slot isn't marked available until the end of each cycle. 3452 int j = track->mFastIndex; 3453 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3454 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3455 FastTrack *fastTrack = &state->mFastTracks[j]; 3456 3457 // Determine whether the track is currently in underrun condition, 3458 // and whether it had a recent underrun. 3459 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3460 FastTrackUnderruns underruns = ftDump->mUnderruns; 3461 uint32_t recentFull = (underruns.mBitFields.mFull - 3462 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3463 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3464 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3465 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3466 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3467 uint32_t recentUnderruns = recentPartial + recentEmpty; 3468 track->mObservedUnderruns = underruns; 3469 // don't count underruns that occur while stopping or pausing 3470 // or stopped which can occur when flush() is called while active 3471 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3472 recentUnderruns > 0) { 3473 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3474 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3475 } 3476 3477 // This is similar to the state machine for normal tracks, 3478 // with a few modifications for fast tracks. 3479 bool isActive = true; 3480 switch (track->mState) { 3481 case TrackBase::STOPPING_1: 3482 // track stays active in STOPPING_1 state until first underrun 3483 if (recentUnderruns > 0 || track->isTerminated()) { 3484 track->mState = TrackBase::STOPPING_2; 3485 } 3486 break; 3487 case TrackBase::PAUSING: 3488 // ramp down is not yet implemented 3489 track->setPaused(); 3490 break; 3491 case TrackBase::RESUMING: 3492 // ramp up is not yet implemented 3493 track->mState = TrackBase::ACTIVE; 3494 break; 3495 case TrackBase::ACTIVE: 3496 if (recentFull > 0 || recentPartial > 0) { 3497 // track has provided at least some frames recently: reset retry count 3498 track->mRetryCount = kMaxTrackRetries; 3499 } 3500 if (recentUnderruns == 0) { 3501 // no recent underruns: stay active 3502 break; 3503 } 3504 // there has recently been an underrun of some kind 3505 if (track->sharedBuffer() == 0) { 3506 // were any of the recent underruns "empty" (no frames available)? 3507 if (recentEmpty == 0) { 3508 // no, then ignore the partial underruns as they are allowed indefinitely 3509 break; 3510 } 3511 // there has recently been an "empty" underrun: decrement the retry counter 3512 if (--(track->mRetryCount) > 0) { 3513 break; 3514 } 3515 // indicate to client process that the track was disabled because of underrun; 3516 // it will then automatically call start() when data is available 3517 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3518 // remove from active list, but state remains ACTIVE [confusing but true] 3519 isActive = false; 3520 break; 3521 } 3522 // fall through 3523 case TrackBase::STOPPING_2: 3524 case TrackBase::PAUSED: 3525 case TrackBase::STOPPED: 3526 case TrackBase::FLUSHED: // flush() while active 3527 // Check for presentation complete if track is inactive 3528 // We have consumed all the buffers of this track. 3529 // This would be incomplete if we auto-paused on underrun 3530 { 3531 size_t audioHALFrames = 3532 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3533 size_t framesWritten = mBytesWritten / mFrameSize; 3534 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3535 // track stays in active list until presentation is complete 3536 break; 3537 } 3538 } 3539 if (track->isStopping_2()) { 3540 track->mState = TrackBase::STOPPED; 3541 } 3542 if (track->isStopped()) { 3543 // Can't reset directly, as fast mixer is still polling this track 3544 // track->reset(); 3545 // So instead mark this track as needing to be reset after push with ack 3546 resetMask |= 1 << i; 3547 } 3548 isActive = false; 3549 break; 3550 case TrackBase::IDLE: 3551 default: 3552 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3553 } 3554 3555 if (isActive) { 3556 // was it previously inactive? 3557 if (!(state->mTrackMask & (1 << j))) { 3558 ExtendedAudioBufferProvider *eabp = track; 3559 VolumeProvider *vp = track; 3560 fastTrack->mBufferProvider = eabp; 3561 fastTrack->mVolumeProvider = vp; 3562 fastTrack->mChannelMask = track->mChannelMask; 3563 fastTrack->mFormat = track->mFormat; 3564 fastTrack->mGeneration++; 3565 state->mTrackMask |= 1 << j; 3566 didModify = true; 3567 // no acknowledgement required for newly active tracks 3568 } 3569 // cache the combined master volume and stream type volume for fast mixer; this 3570 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3571 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3572 ++fastTracks; 3573 } else { 3574 // was it previously active? 3575 if (state->mTrackMask & (1 << j)) { 3576 fastTrack->mBufferProvider = NULL; 3577 fastTrack->mGeneration++; 3578 state->mTrackMask &= ~(1 << j); 3579 didModify = true; 3580 // If any fast tracks were removed, we must wait for acknowledgement 3581 // because we're about to decrement the last sp<> on those tracks. 3582 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3583 } else { 3584 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3585 } 3586 tracksToRemove->add(track); 3587 // Avoids a misleading display in dumpsys 3588 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3589 } 3590 continue; 3591 } 3592 3593 { // local variable scope to avoid goto warning 3594 3595 audio_track_cblk_t* cblk = track->cblk(); 3596 3597 // The first time a track is added we wait 3598 // for all its buffers to be filled before processing it 3599 int name = track->name(); 3600 // make sure that we have enough frames to mix one full buffer. 3601 // enforce this condition only once to enable draining the buffer in case the client 3602 // app does not call stop() and relies on underrun to stop: 3603 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3604 // during last round 3605 size_t desiredFrames; 3606 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3607 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3608 3609 desiredFrames = sourceFramesNeededWithTimestretch( 3610 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3611 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3612 // add frames already consumed but not yet released by the resampler 3613 // because mAudioTrackServerProxy->framesReady() will include these frames 3614 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3615 3616 uint32_t minFrames = 1; 3617 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3618 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3619 minFrames = desiredFrames; 3620 } 3621 3622 size_t framesReady = track->framesReady(); 3623 if (ATRACE_ENABLED()) { 3624 // I wish we had formatted trace names 3625 char traceName[16]; 3626 strcpy(traceName, "nRdy"); 3627 int name = track->name(); 3628 if (AudioMixer::TRACK0 <= name && 3629 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3630 name -= AudioMixer::TRACK0; 3631 traceName[4] = (name / 10) + '0'; 3632 traceName[5] = (name % 10) + '0'; 3633 } else { 3634 traceName[4] = '?'; 3635 traceName[5] = '?'; 3636 } 3637 traceName[6] = '\0'; 3638 ATRACE_INT(traceName, framesReady); 3639 } 3640 if ((framesReady >= minFrames) && track->isReady() && 3641 !track->isPaused() && !track->isTerminated()) 3642 { 3643 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3644 3645 mixedTracks++; 3646 3647 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3648 // there is an effect chain connected to the track 3649 chain.clear(); 3650 if (track->mainBuffer() != mSinkBuffer && 3651 track->mainBuffer() != mMixerBuffer) { 3652 if (mEffectBufferEnabled) { 3653 mEffectBufferValid = true; // Later can set directly. 3654 } 3655 chain = getEffectChain_l(track->sessionId()); 3656 // Delegate volume control to effect in track effect chain if needed 3657 if (chain != 0) { 3658 tracksWithEffect++; 3659 } else { 3660 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3661 "session %d", 3662 name, track->sessionId()); 3663 } 3664 } 3665 3666 3667 int param = AudioMixer::VOLUME; 3668 if (track->mFillingUpStatus == Track::FS_FILLED) { 3669 // no ramp for the first volume setting 3670 track->mFillingUpStatus = Track::FS_ACTIVE; 3671 if (track->mState == TrackBase::RESUMING) { 3672 track->mState = TrackBase::ACTIVE; 3673 param = AudioMixer::RAMP_VOLUME; 3674 } 3675 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3676 // FIXME should not make a decision based on mServer 3677 } else if (cblk->mServer != 0) { 3678 // If the track is stopped before the first frame was mixed, 3679 // do not apply ramp 3680 param = AudioMixer::RAMP_VOLUME; 3681 } 3682 3683 // compute volume for this track 3684 uint32_t vl, vr; // in U8.24 integer format 3685 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3686 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3687 vl = vr = 0; 3688 vlf = vrf = vaf = 0.; 3689 if (track->isPausing()) { 3690 track->setPaused(); 3691 } 3692 } else { 3693 3694 // read original volumes with volume control 3695 float typeVolume = mStreamTypes[track->streamType()].volume; 3696 float v = masterVolume * typeVolume; 3697 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3698 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3699 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3700 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3701 // track volumes come from shared memory, so can't be trusted and must be clamped 3702 if (vlf > GAIN_FLOAT_UNITY) { 3703 ALOGV("Track left volume out of range: %.3g", vlf); 3704 vlf = GAIN_FLOAT_UNITY; 3705 } 3706 if (vrf > GAIN_FLOAT_UNITY) { 3707 ALOGV("Track right volume out of range: %.3g", vrf); 3708 vrf = GAIN_FLOAT_UNITY; 3709 } 3710 // now apply the master volume and stream type volume 3711 vlf *= v; 3712 vrf *= v; 3713 // assuming master volume and stream type volume each go up to 1.0, 3714 // then derive vl and vr as U8.24 versions for the effect chain 3715 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3716 vl = (uint32_t) (scaleto8_24 * vlf); 3717 vr = (uint32_t) (scaleto8_24 * vrf); 3718 // vl and vr are now in U8.24 format 3719 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3720 // send level comes from shared memory and so may be corrupt 3721 if (sendLevel > MAX_GAIN_INT) { 3722 ALOGV("Track send level out of range: %04X", sendLevel); 3723 sendLevel = MAX_GAIN_INT; 3724 } 3725 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3726 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3727 } 3728 3729 // Delegate volume control to effect in track effect chain if needed 3730 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3731 // Do not ramp volume if volume is controlled by effect 3732 param = AudioMixer::VOLUME; 3733 // Update remaining floating point volume levels 3734 vlf = (float)vl / (1 << 24); 3735 vrf = (float)vr / (1 << 24); 3736 track->mHasVolumeController = true; 3737 } else { 3738 // force no volume ramp when volume controller was just disabled or removed 3739 // from effect chain to avoid volume spike 3740 if (track->mHasVolumeController) { 3741 param = AudioMixer::VOLUME; 3742 } 3743 track->mHasVolumeController = false; 3744 } 3745 3746 // XXX: these things DON'T need to be done each time 3747 mAudioMixer->setBufferProvider(name, track); 3748 mAudioMixer->enable(name); 3749 3750 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3751 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3752 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3753 mAudioMixer->setParameter( 3754 name, 3755 AudioMixer::TRACK, 3756 AudioMixer::FORMAT, (void *)track->format()); 3757 mAudioMixer->setParameter( 3758 name, 3759 AudioMixer::TRACK, 3760 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3761 mAudioMixer->setParameter( 3762 name, 3763 AudioMixer::TRACK, 3764 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3765 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3766 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3767 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3768 if (reqSampleRate == 0) { 3769 reqSampleRate = mSampleRate; 3770 } else if (reqSampleRate > maxSampleRate) { 3771 reqSampleRate = maxSampleRate; 3772 } 3773 mAudioMixer->setParameter( 3774 name, 3775 AudioMixer::RESAMPLE, 3776 AudioMixer::SAMPLE_RATE, 3777 (void *)(uintptr_t)reqSampleRate); 3778 3779 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3780 mAudioMixer->setParameter( 3781 name, 3782 AudioMixer::TIMESTRETCH, 3783 AudioMixer::PLAYBACK_RATE, 3784 &playbackRate); 3785 3786 /* 3787 * Select the appropriate output buffer for the track. 3788 * 3789 * Tracks with effects go into their own effects chain buffer 3790 * and from there into either mEffectBuffer or mSinkBuffer. 3791 * 3792 * Other tracks can use mMixerBuffer for higher precision 3793 * channel accumulation. If this buffer is enabled 3794 * (mMixerBufferEnabled true), then selected tracks will accumulate 3795 * into it. 3796 * 3797 */ 3798 if (mMixerBufferEnabled 3799 && (track->mainBuffer() == mSinkBuffer 3800 || track->mainBuffer() == mMixerBuffer)) { 3801 mAudioMixer->setParameter( 3802 name, 3803 AudioMixer::TRACK, 3804 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3805 mAudioMixer->setParameter( 3806 name, 3807 AudioMixer::TRACK, 3808 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3809 // TODO: override track->mainBuffer()? 3810 mMixerBufferValid = true; 3811 } else { 3812 mAudioMixer->setParameter( 3813 name, 3814 AudioMixer::TRACK, 3815 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3816 mAudioMixer->setParameter( 3817 name, 3818 AudioMixer::TRACK, 3819 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3820 } 3821 mAudioMixer->setParameter( 3822 name, 3823 AudioMixer::TRACK, 3824 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3825 3826 // reset retry count 3827 track->mRetryCount = kMaxTrackRetries; 3828 3829 // If one track is ready, set the mixer ready if: 3830 // - the mixer was not ready during previous round OR 3831 // - no other track is not ready 3832 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3833 mixerStatus != MIXER_TRACKS_ENABLED) { 3834 mixerStatus = MIXER_TRACKS_READY; 3835 } 3836 } else { 3837 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3838 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3839 } 3840 // clear effect chain input buffer if an active track underruns to avoid sending 3841 // previous audio buffer again to effects 3842 chain = getEffectChain_l(track->sessionId()); 3843 if (chain != 0) { 3844 chain->clearInputBuffer(); 3845 } 3846 3847 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3848 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3849 track->isStopped() || track->isPaused()) { 3850 // We have consumed all the buffers of this track. 3851 // Remove it from the list of active tracks. 3852 // TODO: use actual buffer filling status instead of latency when available from 3853 // audio HAL 3854 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3855 size_t framesWritten = mBytesWritten / mFrameSize; 3856 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3857 if (track->isStopped()) { 3858 track->reset(); 3859 } 3860 tracksToRemove->add(track); 3861 } 3862 } else { 3863 // No buffers for this track. Give it a few chances to 3864 // fill a buffer, then remove it from active list. 3865 if (--(track->mRetryCount) <= 0) { 3866 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3867 tracksToRemove->add(track); 3868 // indicate to client process that the track was disabled because of underrun; 3869 // it will then automatically call start() when data is available 3870 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3871 // If one track is not ready, mark the mixer also not ready if: 3872 // - the mixer was ready during previous round OR 3873 // - no other track is ready 3874 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3875 mixerStatus != MIXER_TRACKS_READY) { 3876 mixerStatus = MIXER_TRACKS_ENABLED; 3877 } 3878 } 3879 mAudioMixer->disable(name); 3880 } 3881 3882 } // local variable scope to avoid goto warning 3883track_is_ready: ; 3884 3885 } 3886 3887 // Push the new FastMixer state if necessary 3888 bool pauseAudioWatchdog = false; 3889 if (didModify) { 3890 state->mFastTracksGen++; 3891 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3892 if (kUseFastMixer == FastMixer_Dynamic && 3893 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3894 state->mCommand = FastMixerState::COLD_IDLE; 3895 state->mColdFutexAddr = &mFastMixerFutex; 3896 state->mColdGen++; 3897 mFastMixerFutex = 0; 3898 if (kUseFastMixer == FastMixer_Dynamic) { 3899 mNormalSink = mOutputSink; 3900 } 3901 // If we go into cold idle, need to wait for acknowledgement 3902 // so that fast mixer stops doing I/O. 3903 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3904 pauseAudioWatchdog = true; 3905 } 3906 } 3907 if (sq != NULL) { 3908 sq->end(didModify); 3909 sq->push(block); 3910 } 3911#ifdef AUDIO_WATCHDOG 3912 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3913 mAudioWatchdog->pause(); 3914 } 3915#endif 3916 3917 // Now perform the deferred reset on fast tracks that have stopped 3918 while (resetMask != 0) { 3919 size_t i = __builtin_ctz(resetMask); 3920 ALOG_ASSERT(i < count); 3921 resetMask &= ~(1 << i); 3922 sp<Track> t = mActiveTracks[i].promote(); 3923 if (t == 0) { 3924 continue; 3925 } 3926 Track* track = t.get(); 3927 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3928 track->reset(); 3929 } 3930 3931 // remove all the tracks that need to be... 3932 removeTracks_l(*tracksToRemove); 3933 3934 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3935 mEffectBufferValid = true; 3936 } 3937 3938 if (mEffectBufferValid) { 3939 // as long as there are effects we should clear the effects buffer, to avoid 3940 // passing a non-clean buffer to the effect chain 3941 memset(mEffectBuffer, 0, mEffectBufferSize); 3942 } 3943 // sink or mix buffer must be cleared if all tracks are connected to an 3944 // effect chain as in this case the mixer will not write to the sink or mix buffer 3945 // and track effects will accumulate into it 3946 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3947 (mixedTracks == 0 && fastTracks > 0))) { 3948 // FIXME as a performance optimization, should remember previous zero status 3949 if (mMixerBufferValid) { 3950 memset(mMixerBuffer, 0, mMixerBufferSize); 3951 // TODO: In testing, mSinkBuffer below need not be cleared because 3952 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3953 // after mixing. 3954 // 3955 // To enforce this guarantee: 3956 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3957 // (mixedTracks == 0 && fastTracks > 0)) 3958 // must imply MIXER_TRACKS_READY. 3959 // Later, we may clear buffers regardless, and skip much of this logic. 3960 } 3961 // FIXME as a performance optimization, should remember previous zero status 3962 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3963 } 3964 3965 // if any fast tracks, then status is ready 3966 mMixerStatusIgnoringFastTracks = mixerStatus; 3967 if (fastTracks > 0) { 3968 mixerStatus = MIXER_TRACKS_READY; 3969 } 3970 return mixerStatus; 3971} 3972 3973// getTrackName_l() must be called with ThreadBase::mLock held 3974int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3975 audio_format_t format, int sessionId) 3976{ 3977 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3978} 3979 3980// deleteTrackName_l() must be called with ThreadBase::mLock held 3981void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3982{ 3983 ALOGV("remove track (%d) and delete from mixer", name); 3984 mAudioMixer->deleteTrackName(name); 3985} 3986 3987// checkForNewParameter_l() must be called with ThreadBase::mLock held 3988bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3989 status_t& status) 3990{ 3991 bool reconfig = false; 3992 3993 status = NO_ERROR; 3994 3995 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3996 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3997 if (mFastMixer != 0) { 3998 FastMixerStateQueue *sq = mFastMixer->sq(); 3999 FastMixerState *state = sq->begin(); 4000 if (!(state->mCommand & FastMixerState::IDLE)) { 4001 previousCommand = state->mCommand; 4002 state->mCommand = FastMixerState::HOT_IDLE; 4003 sq->end(); 4004 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4005 } else { 4006 sq->end(false /*didModify*/); 4007 } 4008 } 4009 4010 AudioParameter param = AudioParameter(keyValuePair); 4011 int value; 4012 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4013 reconfig = true; 4014 } 4015 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4016 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4017 status = BAD_VALUE; 4018 } else { 4019 // no need to save value, since it's constant 4020 reconfig = true; 4021 } 4022 } 4023 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4024 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4025 status = BAD_VALUE; 4026 } else { 4027 // no need to save value, since it's constant 4028 reconfig = true; 4029 } 4030 } 4031 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4032 // do not accept frame count changes if tracks are open as the track buffer 4033 // size depends on frame count and correct behavior would not be guaranteed 4034 // if frame count is changed after track creation 4035 if (!mTracks.isEmpty()) { 4036 status = INVALID_OPERATION; 4037 } else { 4038 reconfig = true; 4039 } 4040 } 4041 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4042#ifdef ADD_BATTERY_DATA 4043 // when changing the audio output device, call addBatteryData to notify 4044 // the change 4045 if (mOutDevice != value) { 4046 uint32_t params = 0; 4047 // check whether speaker is on 4048 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4049 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4050 } 4051 4052 audio_devices_t deviceWithoutSpeaker 4053 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4054 // check if any other device (except speaker) is on 4055 if (value & deviceWithoutSpeaker ) { 4056 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4057 } 4058 4059 if (params != 0) { 4060 addBatteryData(params); 4061 } 4062 } 4063#endif 4064 4065 // forward device change to effects that have requested to be 4066 // aware of attached audio device. 4067 if (value != AUDIO_DEVICE_NONE) { 4068 mOutDevice = value; 4069 for (size_t i = 0; i < mEffectChains.size(); i++) { 4070 mEffectChains[i]->setDevice_l(mOutDevice); 4071 } 4072 } 4073 } 4074 4075 if (status == NO_ERROR) { 4076 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4077 keyValuePair.string()); 4078 if (!mStandby && status == INVALID_OPERATION) { 4079 mOutput->standby(); 4080 mStandby = true; 4081 mBytesWritten = 0; 4082 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4083 keyValuePair.string()); 4084 } 4085 if (status == NO_ERROR && reconfig) { 4086 readOutputParameters_l(); 4087 delete mAudioMixer; 4088 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4089 for (size_t i = 0; i < mTracks.size() ; i++) { 4090 int name = getTrackName_l(mTracks[i]->mChannelMask, 4091 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4092 if (name < 0) { 4093 break; 4094 } 4095 mTracks[i]->mName = name; 4096 } 4097 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4098 } 4099 } 4100 4101 if (!(previousCommand & FastMixerState::IDLE)) { 4102 ALOG_ASSERT(mFastMixer != 0); 4103 FastMixerStateQueue *sq = mFastMixer->sq(); 4104 FastMixerState *state = sq->begin(); 4105 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4106 state->mCommand = previousCommand; 4107 sq->end(); 4108 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4109 } 4110 4111 return reconfig; 4112} 4113 4114 4115void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4116{ 4117 const size_t SIZE = 256; 4118 char buffer[SIZE]; 4119 String8 result; 4120 4121 PlaybackThread::dumpInternals(fd, args); 4122 4123 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4124 4125 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4126 const FastMixerDumpState copy(mFastMixerDumpState); 4127 copy.dump(fd); 4128 4129#ifdef STATE_QUEUE_DUMP 4130 // Similar for state queue 4131 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4132 observerCopy.dump(fd); 4133 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4134 mutatorCopy.dump(fd); 4135#endif 4136 4137#ifdef TEE_SINK 4138 // Write the tee output to a .wav file 4139 dumpTee(fd, mTeeSource, mId); 4140#endif 4141 4142#ifdef AUDIO_WATCHDOG 4143 if (mAudioWatchdog != 0) { 4144 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4145 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4146 wdCopy.dump(fd); 4147 } 4148#endif 4149} 4150 4151uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4152{ 4153 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4154} 4155 4156uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4157{ 4158 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4159} 4160 4161void AudioFlinger::MixerThread::cacheParameters_l() 4162{ 4163 PlaybackThread::cacheParameters_l(); 4164 4165 // FIXME: Relaxed timing because of a certain device that can't meet latency 4166 // Should be reduced to 2x after the vendor fixes the driver issue 4167 // increase threshold again due to low power audio mode. The way this warning 4168 // threshold is calculated and its usefulness should be reconsidered anyway. 4169 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4170} 4171 4172// ---------------------------------------------------------------------------- 4173 4174AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4175 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4176 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4177 // mLeftVolFloat, mRightVolFloat 4178{ 4179} 4180 4181AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4182 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4183 ThreadBase::type_t type) 4184 : PlaybackThread(audioFlinger, output, id, device, type) 4185 // mLeftVolFloat, mRightVolFloat 4186{ 4187} 4188 4189AudioFlinger::DirectOutputThread::~DirectOutputThread() 4190{ 4191} 4192 4193void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4194{ 4195 audio_track_cblk_t* cblk = track->cblk(); 4196 float left, right; 4197 4198 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4199 left = right = 0; 4200 } else { 4201 float typeVolume = mStreamTypes[track->streamType()].volume; 4202 float v = mMasterVolume * typeVolume; 4203 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4204 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4205 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4206 if (left > GAIN_FLOAT_UNITY) { 4207 left = GAIN_FLOAT_UNITY; 4208 } 4209 left *= v; 4210 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4211 if (right > GAIN_FLOAT_UNITY) { 4212 right = GAIN_FLOAT_UNITY; 4213 } 4214 right *= v; 4215 } 4216 4217 if (lastTrack) { 4218 if (left != mLeftVolFloat || right != mRightVolFloat) { 4219 mLeftVolFloat = left; 4220 mRightVolFloat = right; 4221 4222 // Convert volumes from float to 8.24 4223 uint32_t vl = (uint32_t)(left * (1 << 24)); 4224 uint32_t vr = (uint32_t)(right * (1 << 24)); 4225 4226 // Delegate volume control to effect in track effect chain if needed 4227 // only one effect chain can be present on DirectOutputThread, so if 4228 // there is one, the track is connected to it 4229 if (!mEffectChains.isEmpty()) { 4230 mEffectChains[0]->setVolume_l(&vl, &vr); 4231 left = (float)vl / (1 << 24); 4232 right = (float)vr / (1 << 24); 4233 } 4234 if (mOutput->stream->set_volume) { 4235 mOutput->stream->set_volume(mOutput->stream, left, right); 4236 } 4237 } 4238 } 4239} 4240 4241 4242AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4243 Vector< sp<Track> > *tracksToRemove 4244) 4245{ 4246 size_t count = mActiveTracks.size(); 4247 mixer_state mixerStatus = MIXER_IDLE; 4248 bool doHwPause = false; 4249 bool doHwResume = false; 4250 bool flushPending = false; 4251 4252 // find out which tracks need to be processed 4253 for (size_t i = 0; i < count; i++) { 4254 sp<Track> t = mActiveTracks[i].promote(); 4255 // The track died recently 4256 if (t == 0) { 4257 continue; 4258 } 4259 4260 Track* const track = t.get(); 4261 audio_track_cblk_t* cblk = track->cblk(); 4262 // Only consider last track started for volume and mixer state control. 4263 // In theory an older track could underrun and restart after the new one starts 4264 // but as we only care about the transition phase between two tracks on a 4265 // direct output, it is not a problem to ignore the underrun case. 4266 sp<Track> l = mLatestActiveTrack.promote(); 4267 bool last = l.get() == track; 4268 4269 if (mHwSupportsPause && track->isPausing()) { 4270 track->setPaused(); 4271 if (last && !mHwPaused) { 4272 doHwPause = true; 4273 mHwPaused = true; 4274 } 4275 tracksToRemove->add(track); 4276 } else if (track->isFlushPending()) { 4277 track->flushAck(); 4278 if (last) { 4279 flushPending = true; 4280 } 4281 } else if (mHwSupportsPause && track->isResumePending()){ 4282 track->resumeAck(); 4283 if (last) { 4284 if (mHwPaused) { 4285 doHwResume = true; 4286 mHwPaused = false; 4287 } 4288 } 4289 } 4290 4291 // The first time a track is added we wait 4292 // for all its buffers to be filled before processing it. 4293 // Allow draining the buffer in case the client 4294 // app does not call stop() and relies on underrun to stop: 4295 // hence the test on (track->mRetryCount > 1). 4296 // If retryCount<=1 then track is about to underrun and be removed. 4297 uint32_t minFrames; 4298 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4299 && (track->mRetryCount > 1)) { 4300 minFrames = mNormalFrameCount; 4301 } else { 4302 minFrames = 1; 4303 } 4304 4305 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4306 !track->isStopping_2() && !track->isStopped()) 4307 { 4308 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4309 4310 if (track->mFillingUpStatus == Track::FS_FILLED) { 4311 track->mFillingUpStatus = Track::FS_ACTIVE; 4312 // make sure processVolume_l() will apply new volume even if 0 4313 mLeftVolFloat = mRightVolFloat = -1.0; 4314 if (!mHwSupportsPause) { 4315 track->resumeAck(); 4316 } 4317 } 4318 4319 // compute volume for this track 4320 processVolume_l(track, last); 4321 if (last) { 4322 // reset retry count 4323 track->mRetryCount = kMaxTrackRetriesDirect; 4324 mActiveTrack = t; 4325 mixerStatus = MIXER_TRACKS_READY; 4326 if (usesHwAvSync() && mHwPaused) { 4327 doHwResume = true; 4328 mHwPaused = false; 4329 } 4330 } 4331 } else { 4332 // clear effect chain input buffer if the last active track started underruns 4333 // to avoid sending previous audio buffer again to effects 4334 if (!mEffectChains.isEmpty() && last) { 4335 mEffectChains[0]->clearInputBuffer(); 4336 } 4337 if (track->isStopping_1()) { 4338 track->mState = TrackBase::STOPPING_2; 4339 if (last && mHwPaused) { 4340 doHwResume = true; 4341 mHwPaused = false; 4342 } 4343 } 4344 if ((track->sharedBuffer() != 0) || track->isStopped() || 4345 track->isStopping_2() || track->isPaused()) { 4346 // We have consumed all the buffers of this track. 4347 // Remove it from the list of active tracks. 4348 size_t audioHALFrames; 4349 if (audio_is_linear_pcm(mFormat)) { 4350 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4351 } else { 4352 audioHALFrames = 0; 4353 } 4354 4355 size_t framesWritten = mBytesWritten / mFrameSize; 4356 if (mStandby || !last || 4357 track->presentationComplete(framesWritten, audioHALFrames)) { 4358 if (track->isStopping_2()) { 4359 track->mState = TrackBase::STOPPED; 4360 } 4361 if (track->isStopped()) { 4362 track->reset(); 4363 } 4364 tracksToRemove->add(track); 4365 } 4366 } else { 4367 // No buffers for this track. Give it a few chances to 4368 // fill a buffer, then remove it from active list. 4369 // Only consider last track started for mixer state control 4370 if (--(track->mRetryCount) <= 0) { 4371 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4372 tracksToRemove->add(track); 4373 // indicate to client process that the track was disabled because of underrun; 4374 // it will then automatically call start() when data is available 4375 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4376 } else if (last) { 4377 mixerStatus = MIXER_TRACKS_ENABLED; 4378 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4379 doHwPause = true; 4380 mHwPaused = true; 4381 } 4382 } 4383 } 4384 } 4385 } 4386 4387 // if an active track did not command a flush, check for pending flush on stopped tracks 4388 if (!flushPending) { 4389 for (size_t i = 0; i < mTracks.size(); i++) { 4390 if (mTracks[i]->isFlushPending()) { 4391 mTracks[i]->flushAck(); 4392 flushPending = true; 4393 } 4394 } 4395 } 4396 4397 // make sure the pause/flush/resume sequence is executed in the right order. 4398 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4399 // before flush and then resume HW. This can happen in case of pause/flush/resume 4400 // if resume is received before pause is executed. 4401 if (mHwSupportsPause && !mStandby && 4402 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4403 mOutput->stream->pause(mOutput->stream); 4404 } 4405 if (flushPending) { 4406 flushHw_l(); 4407 } 4408 if (mHwSupportsPause && !mStandby && doHwResume) { 4409 mOutput->stream->resume(mOutput->stream); 4410 } 4411 // remove all the tracks that need to be... 4412 removeTracks_l(*tracksToRemove); 4413 4414 return mixerStatus; 4415} 4416 4417void AudioFlinger::DirectOutputThread::threadLoop_mix() 4418{ 4419 size_t frameCount = mFrameCount; 4420 int8_t *curBuf = (int8_t *)mSinkBuffer; 4421 // output audio to hardware 4422 while (frameCount) { 4423 AudioBufferProvider::Buffer buffer; 4424 buffer.frameCount = frameCount; 4425 status_t status = mActiveTrack->getNextBuffer(&buffer); 4426 if (status != NO_ERROR || buffer.raw == NULL) { 4427 memset(curBuf, 0, frameCount * mFrameSize); 4428 break; 4429 } 4430 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4431 frameCount -= buffer.frameCount; 4432 curBuf += buffer.frameCount * mFrameSize; 4433 mActiveTrack->releaseBuffer(&buffer); 4434 } 4435 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4436 sleepTime = 0; 4437 standbyTime = systemTime() + standbyDelay; 4438 mActiveTrack.clear(); 4439} 4440 4441void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4442{ 4443 // do not write to HAL when paused 4444 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4445 sleepTime = idleSleepTime; 4446 return; 4447 } 4448 if (sleepTime == 0) { 4449 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4450 sleepTime = activeSleepTime; 4451 } else { 4452 sleepTime = idleSleepTime; 4453 } 4454 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4455 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4456 sleepTime = 0; 4457 } 4458} 4459 4460void AudioFlinger::DirectOutputThread::threadLoop_exit() 4461{ 4462 { 4463 Mutex::Autolock _l(mLock); 4464 bool flushPending = false; 4465 for (size_t i = 0; i < mTracks.size(); i++) { 4466 if (mTracks[i]->isFlushPending()) { 4467 mTracks[i]->flushAck(); 4468 flushPending = true; 4469 } 4470 } 4471 if (flushPending) { 4472 flushHw_l(); 4473 } 4474 } 4475 PlaybackThread::threadLoop_exit(); 4476} 4477 4478// must be called with thread mutex locked 4479bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4480{ 4481 bool trackPaused = false; 4482 bool trackStopped = false; 4483 4484 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4485 // after a timeout and we will enter standby then. 4486 if (mTracks.size() > 0) { 4487 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4488 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4489 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4490 } 4491 4492 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped)); 4493} 4494 4495// getTrackName_l() must be called with ThreadBase::mLock held 4496int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4497 audio_format_t format __unused, int sessionId __unused) 4498{ 4499 return 0; 4500} 4501 4502// deleteTrackName_l() must be called with ThreadBase::mLock held 4503void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4504{ 4505} 4506 4507// checkForNewParameter_l() must be called with ThreadBase::mLock held 4508bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4509 status_t& status) 4510{ 4511 bool reconfig = false; 4512 4513 status = NO_ERROR; 4514 4515 AudioParameter param = AudioParameter(keyValuePair); 4516 int value; 4517 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4518 // forward device change to effects that have requested to be 4519 // aware of attached audio device. 4520 if (value != AUDIO_DEVICE_NONE) { 4521 mOutDevice = value; 4522 for (size_t i = 0; i < mEffectChains.size(); i++) { 4523 mEffectChains[i]->setDevice_l(mOutDevice); 4524 } 4525 } 4526 } 4527 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4528 // do not accept frame count changes if tracks are open as the track buffer 4529 // size depends on frame count and correct behavior would not be garantied 4530 // if frame count is changed after track creation 4531 if (!mTracks.isEmpty()) { 4532 status = INVALID_OPERATION; 4533 } else { 4534 reconfig = true; 4535 } 4536 } 4537 if (status == NO_ERROR) { 4538 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4539 keyValuePair.string()); 4540 if (!mStandby && status == INVALID_OPERATION) { 4541 mOutput->standby(); 4542 mStandby = true; 4543 mBytesWritten = 0; 4544 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4545 keyValuePair.string()); 4546 } 4547 if (status == NO_ERROR && reconfig) { 4548 readOutputParameters_l(); 4549 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4550 } 4551 } 4552 4553 return reconfig; 4554} 4555 4556uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4557{ 4558 uint32_t time; 4559 if (audio_is_linear_pcm(mFormat)) { 4560 time = PlaybackThread::activeSleepTimeUs(); 4561 } else { 4562 time = 10000; 4563 } 4564 return time; 4565} 4566 4567uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4568{ 4569 uint32_t time; 4570 if (audio_is_linear_pcm(mFormat)) { 4571 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4572 } else { 4573 time = 10000; 4574 } 4575 return time; 4576} 4577 4578uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4579{ 4580 uint32_t time; 4581 if (audio_is_linear_pcm(mFormat)) { 4582 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4583 } else { 4584 time = 10000; 4585 } 4586 return time; 4587} 4588 4589void AudioFlinger::DirectOutputThread::cacheParameters_l() 4590{ 4591 PlaybackThread::cacheParameters_l(); 4592 4593 // use shorter standby delay as on normal output to release 4594 // hardware resources as soon as possible 4595 // no delay on outputs with HW A/V sync 4596 if (usesHwAvSync()) { 4597 standbyDelay = 0; 4598 } else if (audio_is_linear_pcm(mFormat)) { 4599 standbyDelay = microseconds(activeSleepTime*2); 4600 } else { 4601 standbyDelay = kOffloadStandbyDelayNs; 4602 } 4603} 4604 4605void AudioFlinger::DirectOutputThread::flushHw_l() 4606{ 4607 mOutput->flush(); 4608 mHwPaused = false; 4609} 4610 4611// ---------------------------------------------------------------------------- 4612 4613AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4614 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4615 : Thread(false /*canCallJava*/), 4616 mPlaybackThread(playbackThread), 4617 mWriteAckSequence(0), 4618 mDrainSequence(0) 4619{ 4620} 4621 4622AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4623{ 4624} 4625 4626void AudioFlinger::AsyncCallbackThread::onFirstRef() 4627{ 4628 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4629} 4630 4631bool AudioFlinger::AsyncCallbackThread::threadLoop() 4632{ 4633 while (!exitPending()) { 4634 uint32_t writeAckSequence; 4635 uint32_t drainSequence; 4636 4637 { 4638 Mutex::Autolock _l(mLock); 4639 while (!((mWriteAckSequence & 1) || 4640 (mDrainSequence & 1) || 4641 exitPending())) { 4642 mWaitWorkCV.wait(mLock); 4643 } 4644 4645 if (exitPending()) { 4646 break; 4647 } 4648 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4649 mWriteAckSequence, mDrainSequence); 4650 writeAckSequence = mWriteAckSequence; 4651 mWriteAckSequence &= ~1; 4652 drainSequence = mDrainSequence; 4653 mDrainSequence &= ~1; 4654 } 4655 { 4656 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4657 if (playbackThread != 0) { 4658 if (writeAckSequence & 1) { 4659 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4660 } 4661 if (drainSequence & 1) { 4662 playbackThread->resetDraining(drainSequence >> 1); 4663 } 4664 } 4665 } 4666 } 4667 return false; 4668} 4669 4670void AudioFlinger::AsyncCallbackThread::exit() 4671{ 4672 ALOGV("AsyncCallbackThread::exit"); 4673 Mutex::Autolock _l(mLock); 4674 requestExit(); 4675 mWaitWorkCV.broadcast(); 4676} 4677 4678void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4679{ 4680 Mutex::Autolock _l(mLock); 4681 // bit 0 is cleared 4682 mWriteAckSequence = sequence << 1; 4683} 4684 4685void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4686{ 4687 Mutex::Autolock _l(mLock); 4688 // ignore unexpected callbacks 4689 if (mWriteAckSequence & 2) { 4690 mWriteAckSequence |= 1; 4691 mWaitWorkCV.signal(); 4692 } 4693} 4694 4695void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4696{ 4697 Mutex::Autolock _l(mLock); 4698 // bit 0 is cleared 4699 mDrainSequence = sequence << 1; 4700} 4701 4702void AudioFlinger::AsyncCallbackThread::resetDraining() 4703{ 4704 Mutex::Autolock _l(mLock); 4705 // ignore unexpected callbacks 4706 if (mDrainSequence & 2) { 4707 mDrainSequence |= 1; 4708 mWaitWorkCV.signal(); 4709 } 4710} 4711 4712 4713// ---------------------------------------------------------------------------- 4714AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4715 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4716 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4717 mPausedBytesRemaining(0) 4718{ 4719 //FIXME: mStandby should be set to true by ThreadBase constructor 4720 mStandby = true; 4721} 4722 4723void AudioFlinger::OffloadThread::threadLoop_exit() 4724{ 4725 if (mFlushPending || mHwPaused) { 4726 // If a flush is pending or track was paused, just discard buffered data 4727 flushHw_l(); 4728 } else { 4729 mMixerStatus = MIXER_DRAIN_ALL; 4730 threadLoop_drain(); 4731 } 4732 if (mUseAsyncWrite) { 4733 ALOG_ASSERT(mCallbackThread != 0); 4734 mCallbackThread->exit(); 4735 } 4736 PlaybackThread::threadLoop_exit(); 4737} 4738 4739AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4740 Vector< sp<Track> > *tracksToRemove 4741) 4742{ 4743 size_t count = mActiveTracks.size(); 4744 4745 mixer_state mixerStatus = MIXER_IDLE; 4746 bool doHwPause = false; 4747 bool doHwResume = false; 4748 4749 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4750 4751 // find out which tracks need to be processed 4752 for (size_t i = 0; i < count; i++) { 4753 sp<Track> t = mActiveTracks[i].promote(); 4754 // The track died recently 4755 if (t == 0) { 4756 continue; 4757 } 4758 Track* const track = t.get(); 4759 audio_track_cblk_t* cblk = track->cblk(); 4760 // Only consider last track started for volume and mixer state control. 4761 // In theory an older track could underrun and restart after the new one starts 4762 // but as we only care about the transition phase between two tracks on a 4763 // direct output, it is not a problem to ignore the underrun case. 4764 sp<Track> l = mLatestActiveTrack.promote(); 4765 bool last = l.get() == track; 4766 4767 if (track->isInvalid()) { 4768 ALOGW("An invalidated track shouldn't be in active list"); 4769 tracksToRemove->add(track); 4770 continue; 4771 } 4772 4773 if (track->mState == TrackBase::IDLE) { 4774 ALOGW("An idle track shouldn't be in active list"); 4775 continue; 4776 } 4777 4778 if (track->isPausing()) { 4779 track->setPaused(); 4780 if (last) { 4781 if (!mHwPaused) { 4782 doHwPause = true; 4783 mHwPaused = true; 4784 } 4785 // If we were part way through writing the mixbuffer to 4786 // the HAL we must save this until we resume 4787 // BUG - this will be wrong if a different track is made active, 4788 // in that case we want to discard the pending data in the 4789 // mixbuffer and tell the client to present it again when the 4790 // track is resumed 4791 mPausedWriteLength = mCurrentWriteLength; 4792 mPausedBytesRemaining = mBytesRemaining; 4793 mBytesRemaining = 0; // stop writing 4794 } 4795 tracksToRemove->add(track); 4796 } else if (track->isFlushPending()) { 4797 track->flushAck(); 4798 if (last) { 4799 mFlushPending = true; 4800 } 4801 } else if (track->isResumePending()){ 4802 track->resumeAck(); 4803 if (last) { 4804 if (mPausedBytesRemaining) { 4805 // Need to continue write that was interrupted 4806 mCurrentWriteLength = mPausedWriteLength; 4807 mBytesRemaining = mPausedBytesRemaining; 4808 mPausedBytesRemaining = 0; 4809 } 4810 if (mHwPaused) { 4811 doHwResume = true; 4812 mHwPaused = false; 4813 // threadLoop_mix() will handle the case that we need to 4814 // resume an interrupted write 4815 } 4816 // enable write to audio HAL 4817 sleepTime = 0; 4818 4819 // Do not handle new data in this iteration even if track->framesReady() 4820 mixerStatus = MIXER_TRACKS_ENABLED; 4821 } 4822 } else if (track->framesReady() && track->isReady() && 4823 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4824 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4825 if (track->mFillingUpStatus == Track::FS_FILLED) { 4826 track->mFillingUpStatus = Track::FS_ACTIVE; 4827 // make sure processVolume_l() will apply new volume even if 0 4828 mLeftVolFloat = mRightVolFloat = -1.0; 4829 } 4830 4831 if (last) { 4832 sp<Track> previousTrack = mPreviousTrack.promote(); 4833 if (previousTrack != 0) { 4834 if (track != previousTrack.get()) { 4835 // Flush any data still being written from last track 4836 mBytesRemaining = 0; 4837 if (mPausedBytesRemaining) { 4838 // Last track was paused so we also need to flush saved 4839 // mixbuffer state and invalidate track so that it will 4840 // re-submit that unwritten data when it is next resumed 4841 mPausedBytesRemaining = 0; 4842 // Invalidate is a bit drastic - would be more efficient 4843 // to have a flag to tell client that some of the 4844 // previously written data was lost 4845 previousTrack->invalidate(); 4846 } 4847 // flush data already sent to the DSP if changing audio session as audio 4848 // comes from a different source. Also invalidate previous track to force a 4849 // seek when resuming. 4850 if (previousTrack->sessionId() != track->sessionId()) { 4851 previousTrack->invalidate(); 4852 } 4853 } 4854 } 4855 mPreviousTrack = track; 4856 // reset retry count 4857 track->mRetryCount = kMaxTrackRetriesOffload; 4858 mActiveTrack = t; 4859 mixerStatus = MIXER_TRACKS_READY; 4860 } 4861 } else { 4862 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4863 if (track->isStopping_1()) { 4864 // Hardware buffer can hold a large amount of audio so we must 4865 // wait for all current track's data to drain before we say 4866 // that the track is stopped. 4867 if (mBytesRemaining == 0) { 4868 // Only start draining when all data in mixbuffer 4869 // has been written 4870 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4871 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4872 // do not drain if no data was ever sent to HAL (mStandby == true) 4873 if (last && !mStandby) { 4874 // do not modify drain sequence if we are already draining. This happens 4875 // when resuming from pause after drain. 4876 if ((mDrainSequence & 1) == 0) { 4877 sleepTime = 0; 4878 standbyTime = systemTime() + standbyDelay; 4879 mixerStatus = MIXER_DRAIN_TRACK; 4880 mDrainSequence += 2; 4881 } 4882 if (mHwPaused) { 4883 // It is possible to move from PAUSED to STOPPING_1 without 4884 // a resume so we must ensure hardware is running 4885 doHwResume = true; 4886 mHwPaused = false; 4887 } 4888 } 4889 } 4890 } else if (track->isStopping_2()) { 4891 // Drain has completed or we are in standby, signal presentation complete 4892 if (!(mDrainSequence & 1) || !last || mStandby) { 4893 track->mState = TrackBase::STOPPED; 4894 size_t audioHALFrames = 4895 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4896 size_t framesWritten = 4897 mBytesWritten / mOutput->getFrameSize(); 4898 track->presentationComplete(framesWritten, audioHALFrames); 4899 track->reset(); 4900 tracksToRemove->add(track); 4901 } 4902 } else { 4903 // No buffers for this track. Give it a few chances to 4904 // fill a buffer, then remove it from active list. 4905 if (--(track->mRetryCount) <= 0) { 4906 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4907 track->name()); 4908 tracksToRemove->add(track); 4909 // indicate to client process that the track was disabled because of underrun; 4910 // it will then automatically call start() when data is available 4911 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4912 } else if (last){ 4913 mixerStatus = MIXER_TRACKS_ENABLED; 4914 } 4915 } 4916 } 4917 // compute volume for this track 4918 processVolume_l(track, last); 4919 } 4920 4921 // make sure the pause/flush/resume sequence is executed in the right order. 4922 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4923 // before flush and then resume HW. This can happen in case of pause/flush/resume 4924 // if resume is received before pause is executed. 4925 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4926 mOutput->stream->pause(mOutput->stream); 4927 } 4928 if (mFlushPending) { 4929 flushHw_l(); 4930 mFlushPending = false; 4931 } 4932 if (!mStandby && doHwResume) { 4933 mOutput->stream->resume(mOutput->stream); 4934 } 4935 4936 // remove all the tracks that need to be... 4937 removeTracks_l(*tracksToRemove); 4938 4939 return mixerStatus; 4940} 4941 4942// must be called with thread mutex locked 4943bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4944{ 4945 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4946 mWriteAckSequence, mDrainSequence); 4947 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4948 return true; 4949 } 4950 return false; 4951} 4952 4953bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4954{ 4955 Mutex::Autolock _l(mLock); 4956 return waitingAsyncCallback_l(); 4957} 4958 4959void AudioFlinger::OffloadThread::flushHw_l() 4960{ 4961 DirectOutputThread::flushHw_l(); 4962 // Flush anything still waiting in the mixbuffer 4963 mCurrentWriteLength = 0; 4964 mBytesRemaining = 0; 4965 mPausedWriteLength = 0; 4966 mPausedBytesRemaining = 0; 4967 4968 if (mUseAsyncWrite) { 4969 // discard any pending drain or write ack by incrementing sequence 4970 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4971 mDrainSequence = (mDrainSequence + 2) & ~1; 4972 ALOG_ASSERT(mCallbackThread != 0); 4973 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4974 mCallbackThread->setDraining(mDrainSequence); 4975 } 4976} 4977 4978void AudioFlinger::OffloadThread::onAddNewTrack_l() 4979{ 4980 sp<Track> previousTrack = mPreviousTrack.promote(); 4981 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4982 4983 if (previousTrack != 0 && latestTrack != 0 && 4984 (previousTrack->sessionId() != latestTrack->sessionId())) { 4985 mFlushPending = true; 4986 } 4987 PlaybackThread::onAddNewTrack_l(); 4988} 4989 4990// ---------------------------------------------------------------------------- 4991 4992AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4993 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4994 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4995 DUPLICATING), 4996 mWaitTimeMs(UINT_MAX) 4997{ 4998 addOutputTrack(mainThread); 4999} 5000 5001AudioFlinger::DuplicatingThread::~DuplicatingThread() 5002{ 5003 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5004 mOutputTracks[i]->destroy(); 5005 } 5006} 5007 5008void AudioFlinger::DuplicatingThread::threadLoop_mix() 5009{ 5010 // mix buffers... 5011 if (outputsReady(outputTracks)) { 5012 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5013 } else { 5014 if (mMixerBufferValid) { 5015 memset(mMixerBuffer, 0, mMixerBufferSize); 5016 } else { 5017 memset(mSinkBuffer, 0, mSinkBufferSize); 5018 } 5019 } 5020 sleepTime = 0; 5021 writeFrames = mNormalFrameCount; 5022 mCurrentWriteLength = mSinkBufferSize; 5023 standbyTime = systemTime() + standbyDelay; 5024} 5025 5026void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5027{ 5028 if (sleepTime == 0) { 5029 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5030 sleepTime = activeSleepTime; 5031 } else { 5032 sleepTime = idleSleepTime; 5033 } 5034 } else if (mBytesWritten != 0) { 5035 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5036 writeFrames = mNormalFrameCount; 5037 memset(mSinkBuffer, 0, mSinkBufferSize); 5038 } else { 5039 // flush remaining overflow buffers in output tracks 5040 writeFrames = 0; 5041 } 5042 sleepTime = 0; 5043 } 5044} 5045 5046ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5047{ 5048 for (size_t i = 0; i < outputTracks.size(); i++) { 5049 outputTracks[i]->write(mSinkBuffer, writeFrames); 5050 } 5051 mStandby = false; 5052 return (ssize_t)mSinkBufferSize; 5053} 5054 5055void AudioFlinger::DuplicatingThread::threadLoop_standby() 5056{ 5057 // DuplicatingThread implements standby by stopping all tracks 5058 for (size_t i = 0; i < outputTracks.size(); i++) { 5059 outputTracks[i]->stop(); 5060 } 5061} 5062 5063void AudioFlinger::DuplicatingThread::saveOutputTracks() 5064{ 5065 outputTracks = mOutputTracks; 5066} 5067 5068void AudioFlinger::DuplicatingThread::clearOutputTracks() 5069{ 5070 outputTracks.clear(); 5071} 5072 5073void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5074{ 5075 Mutex::Autolock _l(mLock); 5076 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5077 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5078 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5079 const size_t frameCount = 5080 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5081 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5082 // from different OutputTracks and their associated MixerThreads (e.g. one may 5083 // nearly empty and the other may be dropping data). 5084 5085 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5086 this, 5087 mSampleRate, 5088 mFormat, 5089 mChannelMask, 5090 frameCount, 5091 IPCThreadState::self()->getCallingUid()); 5092 if (outputTrack->cblk() != NULL) { 5093 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5094 mOutputTracks.add(outputTrack); 5095 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5096 updateWaitTime_l(); 5097 } 5098} 5099 5100void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5101{ 5102 Mutex::Autolock _l(mLock); 5103 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5104 if (mOutputTracks[i]->thread() == thread) { 5105 mOutputTracks[i]->destroy(); 5106 mOutputTracks.removeAt(i); 5107 updateWaitTime_l(); 5108 return; 5109 } 5110 } 5111 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5112} 5113 5114// caller must hold mLock 5115void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5116{ 5117 mWaitTimeMs = UINT_MAX; 5118 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5119 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5120 if (strong != 0) { 5121 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5122 if (waitTimeMs < mWaitTimeMs) { 5123 mWaitTimeMs = waitTimeMs; 5124 } 5125 } 5126 } 5127} 5128 5129 5130bool AudioFlinger::DuplicatingThread::outputsReady( 5131 const SortedVector< sp<OutputTrack> > &outputTracks) 5132{ 5133 for (size_t i = 0; i < outputTracks.size(); i++) { 5134 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5135 if (thread == 0) { 5136 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5137 outputTracks[i].get()); 5138 return false; 5139 } 5140 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5141 // see note at standby() declaration 5142 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5143 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5144 thread.get()); 5145 return false; 5146 } 5147 } 5148 return true; 5149} 5150 5151uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5152{ 5153 return (mWaitTimeMs * 1000) / 2; 5154} 5155 5156void AudioFlinger::DuplicatingThread::cacheParameters_l() 5157{ 5158 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5159 updateWaitTime_l(); 5160 5161 MixerThread::cacheParameters_l(); 5162} 5163 5164// ---------------------------------------------------------------------------- 5165// Record 5166// ---------------------------------------------------------------------------- 5167 5168AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5169 AudioStreamIn *input, 5170 audio_io_handle_t id, 5171 audio_devices_t outDevice, 5172 audio_devices_t inDevice 5173#ifdef TEE_SINK 5174 , const sp<NBAIO_Sink>& teeSink 5175#endif 5176 ) : 5177 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5178 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5179 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5180 mRsmpInRear(0) 5181#ifdef TEE_SINK 5182 , mTeeSink(teeSink) 5183#endif 5184 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5185 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5186 // mFastCapture below 5187 , mFastCaptureFutex(0) 5188 // mInputSource 5189 // mPipeSink 5190 // mPipeSource 5191 , mPipeFramesP2(0) 5192 // mPipeMemory 5193 // mFastCaptureNBLogWriter 5194 , mFastTrackAvail(false) 5195{ 5196 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5197 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5198 5199 readInputParameters_l(); 5200 5201 // create an NBAIO source for the HAL input stream, and negotiate 5202 mInputSource = new AudioStreamInSource(input->stream); 5203 size_t numCounterOffers = 0; 5204 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5205 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5206 ALOG_ASSERT(index == 0); 5207 5208 // initialize fast capture depending on configuration 5209 bool initFastCapture; 5210 switch (kUseFastCapture) { 5211 case FastCapture_Never: 5212 initFastCapture = false; 5213 break; 5214 case FastCapture_Always: 5215 initFastCapture = true; 5216 break; 5217 case FastCapture_Static: 5218 uint32_t primaryOutputSampleRate; 5219 { 5220 AutoMutex _l(audioFlinger->mHardwareLock); 5221 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5222 } 5223 initFastCapture = 5224 // either capture sample rate is same as (a reasonable) primary output sample rate 5225 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5226 (mSampleRate == primaryOutputSampleRate)) || 5227 // or primary output sample rate is unknown, and capture sample rate is reasonable 5228 ((primaryOutputSampleRate == 0) && 5229 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5230 // and the buffer size is < 12 ms 5231 (mFrameCount * 1000) / mSampleRate < 12; 5232 break; 5233 // case FastCapture_Dynamic: 5234 } 5235 5236 if (initFastCapture) { 5237 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5238 NBAIO_Format format = mInputSource->format(); 5239 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5240 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5241 void *pipeBuffer; 5242 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5243 sp<IMemory> pipeMemory; 5244 if ((roHeap == 0) || 5245 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5246 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5247 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5248 goto failed; 5249 } 5250 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5251 memset(pipeBuffer, 0, pipeSize); 5252 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5253 const NBAIO_Format offers[1] = {format}; 5254 size_t numCounterOffers = 0; 5255 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5256 ALOG_ASSERT(index == 0); 5257 mPipeSink = pipe; 5258 PipeReader *pipeReader = new PipeReader(*pipe); 5259 numCounterOffers = 0; 5260 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5261 ALOG_ASSERT(index == 0); 5262 mPipeSource = pipeReader; 5263 mPipeFramesP2 = pipeFramesP2; 5264 mPipeMemory = pipeMemory; 5265 5266 // create fast capture 5267 mFastCapture = new FastCapture(); 5268 FastCaptureStateQueue *sq = mFastCapture->sq(); 5269#ifdef STATE_QUEUE_DUMP 5270 // FIXME 5271#endif 5272 FastCaptureState *state = sq->begin(); 5273 state->mCblk = NULL; 5274 state->mInputSource = mInputSource.get(); 5275 state->mInputSourceGen++; 5276 state->mPipeSink = pipe; 5277 state->mPipeSinkGen++; 5278 state->mFrameCount = mFrameCount; 5279 state->mCommand = FastCaptureState::COLD_IDLE; 5280 // already done in constructor initialization list 5281 //mFastCaptureFutex = 0; 5282 state->mColdFutexAddr = &mFastCaptureFutex; 5283 state->mColdGen++; 5284 state->mDumpState = &mFastCaptureDumpState; 5285#ifdef TEE_SINK 5286 // FIXME 5287#endif 5288 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5289 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5290 sq->end(); 5291 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5292 5293 // start the fast capture 5294 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5295 pid_t tid = mFastCapture->getTid(); 5296 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5297 if (err != 0) { 5298 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5299 kPriorityFastCapture, getpid_cached, tid, err); 5300 } 5301 5302#ifdef AUDIO_WATCHDOG 5303 // FIXME 5304#endif 5305 5306 mFastTrackAvail = true; 5307 } 5308failed: ; 5309 5310 // FIXME mNormalSource 5311} 5312 5313AudioFlinger::RecordThread::~RecordThread() 5314{ 5315 if (mFastCapture != 0) { 5316 FastCaptureStateQueue *sq = mFastCapture->sq(); 5317 FastCaptureState *state = sq->begin(); 5318 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5319 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5320 if (old == -1) { 5321 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5322 } 5323 } 5324 state->mCommand = FastCaptureState::EXIT; 5325 sq->end(); 5326 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5327 mFastCapture->join(); 5328 mFastCapture.clear(); 5329 } 5330 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5331 mAudioFlinger->unregisterWriter(mNBLogWriter); 5332 free(mRsmpInBuffer); 5333} 5334 5335void AudioFlinger::RecordThread::onFirstRef() 5336{ 5337 run(mThreadName, PRIORITY_URGENT_AUDIO); 5338} 5339 5340bool AudioFlinger::RecordThread::threadLoop() 5341{ 5342 nsecs_t lastWarning = 0; 5343 5344 inputStandBy(); 5345 5346reacquire_wakelock: 5347 sp<RecordTrack> activeTrack; 5348 int activeTracksGen; 5349 { 5350 Mutex::Autolock _l(mLock); 5351 size_t size = mActiveTracks.size(); 5352 activeTracksGen = mActiveTracksGen; 5353 if (size > 0) { 5354 // FIXME an arbitrary choice 5355 activeTrack = mActiveTracks[0]; 5356 acquireWakeLock_l(activeTrack->uid()); 5357 if (size > 1) { 5358 SortedVector<int> tmp; 5359 for (size_t i = 0; i < size; i++) { 5360 tmp.add(mActiveTracks[i]->uid()); 5361 } 5362 updateWakeLockUids_l(tmp); 5363 } 5364 } else { 5365 acquireWakeLock_l(-1); 5366 } 5367 } 5368 5369 // used to request a deferred sleep, to be executed later while mutex is unlocked 5370 uint32_t sleepUs = 0; 5371 5372 // loop while there is work to do 5373 for (;;) { 5374 Vector< sp<EffectChain> > effectChains; 5375 5376 // sleep with mutex unlocked 5377 if (sleepUs > 0) { 5378 ATRACE_BEGIN("sleep"); 5379 usleep(sleepUs); 5380 ATRACE_END(); 5381 sleepUs = 0; 5382 } 5383 5384 // activeTracks accumulates a copy of a subset of mActiveTracks 5385 Vector< sp<RecordTrack> > activeTracks; 5386 5387 // reference to the (first and only) active fast track 5388 sp<RecordTrack> fastTrack; 5389 5390 // reference to a fast track which is about to be removed 5391 sp<RecordTrack> fastTrackToRemove; 5392 5393 { // scope for mLock 5394 Mutex::Autolock _l(mLock); 5395 5396 processConfigEvents_l(); 5397 5398 // check exitPending here because checkForNewParameters_l() and 5399 // checkForNewParameters_l() can temporarily release mLock 5400 if (exitPending()) { 5401 break; 5402 } 5403 5404 // if no active track(s), then standby and release wakelock 5405 size_t size = mActiveTracks.size(); 5406 if (size == 0) { 5407 standbyIfNotAlreadyInStandby(); 5408 // exitPending() can't become true here 5409 releaseWakeLock_l(); 5410 ALOGV("RecordThread: loop stopping"); 5411 // go to sleep 5412 mWaitWorkCV.wait(mLock); 5413 ALOGV("RecordThread: loop starting"); 5414 goto reacquire_wakelock; 5415 } 5416 5417 if (mActiveTracksGen != activeTracksGen) { 5418 activeTracksGen = mActiveTracksGen; 5419 SortedVector<int> tmp; 5420 for (size_t i = 0; i < size; i++) { 5421 tmp.add(mActiveTracks[i]->uid()); 5422 } 5423 updateWakeLockUids_l(tmp); 5424 } 5425 5426 bool doBroadcast = false; 5427 for (size_t i = 0; i < size; ) { 5428 5429 activeTrack = mActiveTracks[i]; 5430 if (activeTrack->isTerminated()) { 5431 if (activeTrack->isFastTrack()) { 5432 ALOG_ASSERT(fastTrackToRemove == 0); 5433 fastTrackToRemove = activeTrack; 5434 } 5435 removeTrack_l(activeTrack); 5436 mActiveTracks.remove(activeTrack); 5437 mActiveTracksGen++; 5438 size--; 5439 continue; 5440 } 5441 5442 TrackBase::track_state activeTrackState = activeTrack->mState; 5443 switch (activeTrackState) { 5444 5445 case TrackBase::PAUSING: 5446 mActiveTracks.remove(activeTrack); 5447 mActiveTracksGen++; 5448 doBroadcast = true; 5449 size--; 5450 continue; 5451 5452 case TrackBase::STARTING_1: 5453 sleepUs = 10000; 5454 i++; 5455 continue; 5456 5457 case TrackBase::STARTING_2: 5458 doBroadcast = true; 5459 mStandby = false; 5460 activeTrack->mState = TrackBase::ACTIVE; 5461 break; 5462 5463 case TrackBase::ACTIVE: 5464 break; 5465 5466 case TrackBase::IDLE: 5467 i++; 5468 continue; 5469 5470 default: 5471 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5472 } 5473 5474 activeTracks.add(activeTrack); 5475 i++; 5476 5477 if (activeTrack->isFastTrack()) { 5478 ALOG_ASSERT(!mFastTrackAvail); 5479 ALOG_ASSERT(fastTrack == 0); 5480 fastTrack = activeTrack; 5481 } 5482 } 5483 if (doBroadcast) { 5484 mStartStopCond.broadcast(); 5485 } 5486 5487 // sleep if there are no active tracks to process 5488 if (activeTracks.size() == 0) { 5489 if (sleepUs == 0) { 5490 sleepUs = kRecordThreadSleepUs; 5491 } 5492 continue; 5493 } 5494 sleepUs = 0; 5495 5496 lockEffectChains_l(effectChains); 5497 } 5498 5499 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5500 5501 size_t size = effectChains.size(); 5502 for (size_t i = 0; i < size; i++) { 5503 // thread mutex is not locked, but effect chain is locked 5504 effectChains[i]->process_l(); 5505 } 5506 5507 // Push a new fast capture state if fast capture is not already running, or cblk change 5508 if (mFastCapture != 0) { 5509 FastCaptureStateQueue *sq = mFastCapture->sq(); 5510 FastCaptureState *state = sq->begin(); 5511 bool didModify = false; 5512 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5513 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5514 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5515 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5516 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5517 if (old == -1) { 5518 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5519 } 5520 } 5521 state->mCommand = FastCaptureState::READ_WRITE; 5522#if 0 // FIXME 5523 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5524 FastThreadDumpState::kSamplingNforLowRamDevice : 5525 FastThreadDumpState::kSamplingN); 5526#endif 5527 didModify = true; 5528 } 5529 audio_track_cblk_t *cblkOld = state->mCblk; 5530 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5531 if (cblkNew != cblkOld) { 5532 state->mCblk = cblkNew; 5533 // block until acked if removing a fast track 5534 if (cblkOld != NULL) { 5535 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5536 } 5537 didModify = true; 5538 } 5539 sq->end(didModify); 5540 if (didModify) { 5541 sq->push(block); 5542#if 0 5543 if (kUseFastCapture == FastCapture_Dynamic) { 5544 mNormalSource = mPipeSource; 5545 } 5546#endif 5547 } 5548 } 5549 5550 // now run the fast track destructor with thread mutex unlocked 5551 fastTrackToRemove.clear(); 5552 5553 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5554 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5555 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5556 // If destination is non-contiguous, first read past the nominal end of buffer, then 5557 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5558 5559 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5560 ssize_t framesRead; 5561 5562 // If an NBAIO source is present, use it to read the normal capture's data 5563 if (mPipeSource != 0) { 5564 size_t framesToRead = mBufferSize / mFrameSize; 5565 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5566 framesToRead, AudioBufferProvider::kInvalidPTS); 5567 if (framesRead == 0) { 5568 // since pipe is non-blocking, simulate blocking input 5569 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5570 } 5571 // otherwise use the HAL / AudioStreamIn directly 5572 } else { 5573 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5574 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5575 if (bytesRead < 0) { 5576 framesRead = bytesRead; 5577 } else { 5578 framesRead = bytesRead / mFrameSize; 5579 } 5580 } 5581 5582 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5583 ALOGE("read failed: framesRead=%d", framesRead); 5584 // Force input into standby so that it tries to recover at next read attempt 5585 inputStandBy(); 5586 sleepUs = kRecordThreadSleepUs; 5587 } 5588 if (framesRead <= 0) { 5589 goto unlock; 5590 } 5591 ALOG_ASSERT(framesRead > 0); 5592 5593 if (mTeeSink != 0) { 5594 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5595 } 5596 // If destination is non-contiguous, we now correct for reading past end of buffer. 5597 { 5598 size_t part1 = mRsmpInFramesP2 - rear; 5599 if ((size_t) framesRead > part1) { 5600 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5601 (framesRead - part1) * mFrameSize); 5602 } 5603 } 5604 rear = mRsmpInRear += framesRead; 5605 5606 size = activeTracks.size(); 5607 // loop over each active track 5608 for (size_t i = 0; i < size; i++) { 5609 activeTrack = activeTracks[i]; 5610 5611 // skip fast tracks, as those are handled directly by FastCapture 5612 if (activeTrack->isFastTrack()) { 5613 continue; 5614 } 5615 5616 // TODO: This code probably should be moved to RecordTrack. 5617 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5618 5619 enum { 5620 OVERRUN_UNKNOWN, 5621 OVERRUN_TRUE, 5622 OVERRUN_FALSE 5623 } overrun = OVERRUN_UNKNOWN; 5624 5625 // loop over getNextBuffer to handle circular sink 5626 for (;;) { 5627 5628 activeTrack->mSink.frameCount = ~0; 5629 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5630 size_t framesOut = activeTrack->mSink.frameCount; 5631 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5632 5633 // check available frames and handle overrun conditions 5634 // if the record track isn't draining fast enough. 5635 bool hasOverrun; 5636 size_t framesIn; 5637 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5638 if (hasOverrun) { 5639 overrun = OVERRUN_TRUE; 5640 } 5641 if (framesOut == 0 || framesIn == 0) { 5642 break; 5643 } 5644 5645 // Don't allow framesOut to be larger than what is possible with resampling 5646 // from framesIn. 5647 // This isn't strictly necessary but helps limit buffer resizing in 5648 // RecordBufferConverter. TODO: remove when no longer needed. 5649 framesOut = min(framesOut, 5650 destinationFramesPossible( 5651 framesIn, mSampleRate, activeTrack->mSampleRate)); 5652 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5653 framesOut = activeTrack->mRecordBufferConverter->convert( 5654 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5655 5656 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5657 overrun = OVERRUN_FALSE; 5658 } 5659 5660 if (activeTrack->mFramesToDrop == 0) { 5661 if (framesOut > 0) { 5662 activeTrack->mSink.frameCount = framesOut; 5663 activeTrack->releaseBuffer(&activeTrack->mSink); 5664 } 5665 } else { 5666 // FIXME could do a partial drop of framesOut 5667 if (activeTrack->mFramesToDrop > 0) { 5668 activeTrack->mFramesToDrop -= framesOut; 5669 if (activeTrack->mFramesToDrop <= 0) { 5670 activeTrack->clearSyncStartEvent(); 5671 } 5672 } else { 5673 activeTrack->mFramesToDrop += framesOut; 5674 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5675 activeTrack->mSyncStartEvent->isCancelled()) { 5676 ALOGW("Synced record %s, session %d, trigger session %d", 5677 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5678 activeTrack->sessionId(), 5679 (activeTrack->mSyncStartEvent != 0) ? 5680 activeTrack->mSyncStartEvent->triggerSession() : 0); 5681 activeTrack->clearSyncStartEvent(); 5682 } 5683 } 5684 } 5685 5686 if (framesOut == 0) { 5687 break; 5688 } 5689 } 5690 5691 switch (overrun) { 5692 case OVERRUN_TRUE: 5693 // client isn't retrieving buffers fast enough 5694 if (!activeTrack->setOverflow()) { 5695 nsecs_t now = systemTime(); 5696 // FIXME should lastWarning per track? 5697 if ((now - lastWarning) > kWarningThrottleNs) { 5698 ALOGW("RecordThread: buffer overflow"); 5699 lastWarning = now; 5700 } 5701 } 5702 break; 5703 case OVERRUN_FALSE: 5704 activeTrack->clearOverflow(); 5705 break; 5706 case OVERRUN_UNKNOWN: 5707 break; 5708 } 5709 5710 } 5711 5712unlock: 5713 // enable changes in effect chain 5714 unlockEffectChains(effectChains); 5715 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5716 } 5717 5718 standbyIfNotAlreadyInStandby(); 5719 5720 { 5721 Mutex::Autolock _l(mLock); 5722 for (size_t i = 0; i < mTracks.size(); i++) { 5723 sp<RecordTrack> track = mTracks[i]; 5724 track->invalidate(); 5725 } 5726 mActiveTracks.clear(); 5727 mActiveTracksGen++; 5728 mStartStopCond.broadcast(); 5729 } 5730 5731 releaseWakeLock(); 5732 5733 ALOGV("RecordThread %p exiting", this); 5734 return false; 5735} 5736 5737void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5738{ 5739 if (!mStandby) { 5740 inputStandBy(); 5741 mStandby = true; 5742 } 5743} 5744 5745void AudioFlinger::RecordThread::inputStandBy() 5746{ 5747 // Idle the fast capture if it's currently running 5748 if (mFastCapture != 0) { 5749 FastCaptureStateQueue *sq = mFastCapture->sq(); 5750 FastCaptureState *state = sq->begin(); 5751 if (!(state->mCommand & FastCaptureState::IDLE)) { 5752 state->mCommand = FastCaptureState::COLD_IDLE; 5753 state->mColdFutexAddr = &mFastCaptureFutex; 5754 state->mColdGen++; 5755 mFastCaptureFutex = 0; 5756 sq->end(); 5757 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5758 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5759#if 0 5760 if (kUseFastCapture == FastCapture_Dynamic) { 5761 // FIXME 5762 } 5763#endif 5764#ifdef AUDIO_WATCHDOG 5765 // FIXME 5766#endif 5767 } else { 5768 sq->end(false /*didModify*/); 5769 } 5770 } 5771 mInput->stream->common.standby(&mInput->stream->common); 5772} 5773 5774// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5775sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5776 const sp<AudioFlinger::Client>& client, 5777 uint32_t sampleRate, 5778 audio_format_t format, 5779 audio_channel_mask_t channelMask, 5780 size_t *pFrameCount, 5781 int sessionId, 5782 size_t *notificationFrames, 5783 int uid, 5784 IAudioFlinger::track_flags_t *flags, 5785 pid_t tid, 5786 status_t *status) 5787{ 5788 size_t frameCount = *pFrameCount; 5789 sp<RecordTrack> track; 5790 status_t lStatus; 5791 5792 // client expresses a preference for FAST, but we get the final say 5793 if (*flags & IAudioFlinger::TRACK_FAST) { 5794 if ( 5795 // we formerly checked for a callback handler (non-0 tid), 5796 // but that is no longer required for TRANSFER_OBTAIN mode 5797 // 5798 // frame count is not specified, or is exactly the pipe depth 5799 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5800 // PCM data 5801 audio_is_linear_pcm(format) && 5802 // native format 5803 (format == mFormat) && 5804 // native channel mask 5805 (channelMask == mChannelMask) && 5806 // native hardware sample rate 5807 (sampleRate == mSampleRate) && 5808 // record thread has an associated fast capture 5809 hasFastCapture() && 5810 // there are sufficient fast track slots available 5811 mFastTrackAvail 5812 ) { 5813 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5814 frameCount, mFrameCount); 5815 } else { 5816 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5817 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5818 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5819 frameCount, mFrameCount, mPipeFramesP2, 5820 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5821 hasFastCapture(), tid, mFastTrackAvail); 5822 *flags &= ~IAudioFlinger::TRACK_FAST; 5823 } 5824 } 5825 5826 // compute track buffer size in frames, and suggest the notification frame count 5827 if (*flags & IAudioFlinger::TRACK_FAST) { 5828 // fast track: frame count is exactly the pipe depth 5829 frameCount = mPipeFramesP2; 5830 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5831 *notificationFrames = mFrameCount; 5832 } else { 5833 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5834 // or 20 ms if there is a fast capture 5835 // TODO This could be a roundupRatio inline, and const 5836 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5837 * sampleRate + mSampleRate - 1) / mSampleRate; 5838 // minimum number of notification periods is at least kMinNotifications, 5839 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5840 static const size_t kMinNotifications = 3; 5841 static const uint32_t kMinMs = 30; 5842 // TODO This could be a roundupRatio inline 5843 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5844 // TODO This could be a roundupRatio inline 5845 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5846 maxNotificationFrames; 5847 const size_t minFrameCount = maxNotificationFrames * 5848 max(kMinNotifications, minNotificationsByMs); 5849 frameCount = max(frameCount, minFrameCount); 5850 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5851 *notificationFrames = maxNotificationFrames; 5852 } 5853 } 5854 *pFrameCount = frameCount; 5855 5856 lStatus = initCheck(); 5857 if (lStatus != NO_ERROR) { 5858 ALOGE("createRecordTrack_l() audio driver not initialized"); 5859 goto Exit; 5860 } 5861 5862 { // scope for mLock 5863 Mutex::Autolock _l(mLock); 5864 5865 track = new RecordTrack(this, client, sampleRate, 5866 format, channelMask, frameCount, NULL, sessionId, uid, 5867 *flags, TrackBase::TYPE_DEFAULT); 5868 5869 lStatus = track->initCheck(); 5870 if (lStatus != NO_ERROR) { 5871 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5872 // track must be cleared from the caller as the caller has the AF lock 5873 goto Exit; 5874 } 5875 mTracks.add(track); 5876 5877 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5878 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5879 mAudioFlinger->btNrecIsOff(); 5880 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5881 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5882 5883 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5884 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5885 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5886 // so ask activity manager to do this on our behalf 5887 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5888 } 5889 } 5890 5891 lStatus = NO_ERROR; 5892 5893Exit: 5894 *status = lStatus; 5895 return track; 5896} 5897 5898status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5899 AudioSystem::sync_event_t event, 5900 int triggerSession) 5901{ 5902 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5903 sp<ThreadBase> strongMe = this; 5904 status_t status = NO_ERROR; 5905 5906 if (event == AudioSystem::SYNC_EVENT_NONE) { 5907 recordTrack->clearSyncStartEvent(); 5908 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5909 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5910 triggerSession, 5911 recordTrack->sessionId(), 5912 syncStartEventCallback, 5913 recordTrack); 5914 // Sync event can be cancelled by the trigger session if the track is not in a 5915 // compatible state in which case we start record immediately 5916 if (recordTrack->mSyncStartEvent->isCancelled()) { 5917 recordTrack->clearSyncStartEvent(); 5918 } else { 5919 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5920 recordTrack->mFramesToDrop = - 5921 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5922 } 5923 } 5924 5925 { 5926 // This section is a rendezvous between binder thread executing start() and RecordThread 5927 AutoMutex lock(mLock); 5928 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5929 if (recordTrack->mState == TrackBase::PAUSING) { 5930 ALOGV("active record track PAUSING -> ACTIVE"); 5931 recordTrack->mState = TrackBase::ACTIVE; 5932 } else { 5933 ALOGV("active record track state %d", recordTrack->mState); 5934 } 5935 return status; 5936 } 5937 5938 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5939 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5940 // or using a separate command thread 5941 recordTrack->mState = TrackBase::STARTING_1; 5942 mActiveTracks.add(recordTrack); 5943 mActiveTracksGen++; 5944 status_t status = NO_ERROR; 5945 if (recordTrack->isExternalTrack()) { 5946 mLock.unlock(); 5947 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5948 mLock.lock(); 5949 // FIXME should verify that recordTrack is still in mActiveTracks 5950 if (status != NO_ERROR) { 5951 mActiveTracks.remove(recordTrack); 5952 mActiveTracksGen++; 5953 recordTrack->clearSyncStartEvent(); 5954 ALOGV("RecordThread::start error %d", status); 5955 return status; 5956 } 5957 } 5958 // Catch up with current buffer indices if thread is already running. 5959 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5960 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5961 // see previously buffered data before it called start(), but with greater risk of overrun. 5962 5963 recordTrack->mResamplerBufferProvider->reset(); 5964 // clear any converter state as new data will be discontinuous 5965 recordTrack->mRecordBufferConverter->reset(); 5966 recordTrack->mState = TrackBase::STARTING_2; 5967 // signal thread to start 5968 mWaitWorkCV.broadcast(); 5969 if (mActiveTracks.indexOf(recordTrack) < 0) { 5970 ALOGV("Record failed to start"); 5971 status = BAD_VALUE; 5972 goto startError; 5973 } 5974 return status; 5975 } 5976 5977startError: 5978 if (recordTrack->isExternalTrack()) { 5979 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5980 } 5981 recordTrack->clearSyncStartEvent(); 5982 // FIXME I wonder why we do not reset the state here? 5983 return status; 5984} 5985 5986void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5987{ 5988 sp<SyncEvent> strongEvent = event.promote(); 5989 5990 if (strongEvent != 0) { 5991 sp<RefBase> ptr = strongEvent->cookie().promote(); 5992 if (ptr != 0) { 5993 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5994 recordTrack->handleSyncStartEvent(strongEvent); 5995 } 5996 } 5997} 5998 5999bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6000 ALOGV("RecordThread::stop"); 6001 AutoMutex _l(mLock); 6002 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6003 return false; 6004 } 6005 // note that threadLoop may still be processing the track at this point [without lock] 6006 recordTrack->mState = TrackBase::PAUSING; 6007 // do not wait for mStartStopCond if exiting 6008 if (exitPending()) { 6009 return true; 6010 } 6011 // FIXME incorrect usage of wait: no explicit predicate or loop 6012 mStartStopCond.wait(mLock); 6013 // if we have been restarted, recordTrack is in mActiveTracks here 6014 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6015 ALOGV("Record stopped OK"); 6016 return true; 6017 } 6018 return false; 6019} 6020 6021bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6022{ 6023 return false; 6024} 6025 6026status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6027{ 6028#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6029 if (!isValidSyncEvent(event)) { 6030 return BAD_VALUE; 6031 } 6032 6033 int eventSession = event->triggerSession(); 6034 status_t ret = NAME_NOT_FOUND; 6035 6036 Mutex::Autolock _l(mLock); 6037 6038 for (size_t i = 0; i < mTracks.size(); i++) { 6039 sp<RecordTrack> track = mTracks[i]; 6040 if (eventSession == track->sessionId()) { 6041 (void) track->setSyncEvent(event); 6042 ret = NO_ERROR; 6043 } 6044 } 6045 return ret; 6046#else 6047 return BAD_VALUE; 6048#endif 6049} 6050 6051// destroyTrack_l() must be called with ThreadBase::mLock held 6052void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6053{ 6054 track->terminate(); 6055 track->mState = TrackBase::STOPPED; 6056 // active tracks are removed by threadLoop() 6057 if (mActiveTracks.indexOf(track) < 0) { 6058 removeTrack_l(track); 6059 } 6060} 6061 6062void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6063{ 6064 mTracks.remove(track); 6065 // need anything related to effects here? 6066 if (track->isFastTrack()) { 6067 ALOG_ASSERT(!mFastTrackAvail); 6068 mFastTrackAvail = true; 6069 } 6070} 6071 6072void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6073{ 6074 dumpInternals(fd, args); 6075 dumpTracks(fd, args); 6076 dumpEffectChains(fd, args); 6077} 6078 6079void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6080{ 6081 dprintf(fd, "\nInput thread %p:\n", this); 6082 6083 dumpBase(fd, args); 6084 6085 if (mActiveTracks.size() == 0) { 6086 dprintf(fd, " No active record clients\n"); 6087 } 6088 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6089 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6090 6091 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6092 const FastCaptureDumpState copy(mFastCaptureDumpState); 6093 copy.dump(fd); 6094} 6095 6096void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6097{ 6098 const size_t SIZE = 256; 6099 char buffer[SIZE]; 6100 String8 result; 6101 6102 size_t numtracks = mTracks.size(); 6103 size_t numactive = mActiveTracks.size(); 6104 size_t numactiveseen = 0; 6105 dprintf(fd, " %d Tracks", numtracks); 6106 if (numtracks) { 6107 dprintf(fd, " of which %d are active\n", numactive); 6108 RecordTrack::appendDumpHeader(result); 6109 for (size_t i = 0; i < numtracks ; ++i) { 6110 sp<RecordTrack> track = mTracks[i]; 6111 if (track != 0) { 6112 bool active = mActiveTracks.indexOf(track) >= 0; 6113 if (active) { 6114 numactiveseen++; 6115 } 6116 track->dump(buffer, SIZE, active); 6117 result.append(buffer); 6118 } 6119 } 6120 } else { 6121 dprintf(fd, "\n"); 6122 } 6123 6124 if (numactiveseen != numactive) { 6125 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6126 " not in the track list\n"); 6127 result.append(buffer); 6128 RecordTrack::appendDumpHeader(result); 6129 for (size_t i = 0; i < numactive; ++i) { 6130 sp<RecordTrack> track = mActiveTracks[i]; 6131 if (mTracks.indexOf(track) < 0) { 6132 track->dump(buffer, SIZE, true); 6133 result.append(buffer); 6134 } 6135 } 6136 6137 } 6138 write(fd, result.string(), result.size()); 6139} 6140 6141 6142void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6143{ 6144 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6145 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6146 mRsmpInFront = recordThread->mRsmpInRear; 6147 mRsmpInUnrel = 0; 6148} 6149 6150void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6151 size_t *framesAvailable, bool *hasOverrun) 6152{ 6153 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6154 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6155 const int32_t rear = recordThread->mRsmpInRear; 6156 const int32_t front = mRsmpInFront; 6157 const ssize_t filled = rear - front; 6158 6159 size_t framesIn; 6160 bool overrun = false; 6161 if (filled < 0) { 6162 // should not happen, but treat like a massive overrun and re-sync 6163 framesIn = 0; 6164 mRsmpInFront = rear; 6165 overrun = true; 6166 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6167 framesIn = (size_t) filled; 6168 } else { 6169 // client is not keeping up with server, but give it latest data 6170 framesIn = recordThread->mRsmpInFrames; 6171 mRsmpInFront = /* front = */ rear - framesIn; 6172 overrun = true; 6173 } 6174 if (framesAvailable != NULL) { 6175 *framesAvailable = framesIn; 6176 } 6177 if (hasOverrun != NULL) { 6178 *hasOverrun = overrun; 6179 } 6180} 6181 6182// AudioBufferProvider interface 6183status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6184 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6185{ 6186 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6187 if (threadBase == 0) { 6188 buffer->frameCount = 0; 6189 buffer->raw = NULL; 6190 return NOT_ENOUGH_DATA; 6191 } 6192 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6193 int32_t rear = recordThread->mRsmpInRear; 6194 int32_t front = mRsmpInFront; 6195 ssize_t filled = rear - front; 6196 // FIXME should not be P2 (don't want to increase latency) 6197 // FIXME if client not keeping up, discard 6198 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6199 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6200 front &= recordThread->mRsmpInFramesP2 - 1; 6201 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6202 if (part1 > (size_t) filled) { 6203 part1 = filled; 6204 } 6205 size_t ask = buffer->frameCount; 6206 ALOG_ASSERT(ask > 0); 6207 if (part1 > ask) { 6208 part1 = ask; 6209 } 6210 if (part1 == 0) { 6211 // out of data is fine since the resampler will return a short-count. 6212 buffer->raw = NULL; 6213 buffer->frameCount = 0; 6214 mRsmpInUnrel = 0; 6215 return NOT_ENOUGH_DATA; 6216 } 6217 6218 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6219 buffer->frameCount = part1; 6220 mRsmpInUnrel = part1; 6221 return NO_ERROR; 6222} 6223 6224// AudioBufferProvider interface 6225void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6226 AudioBufferProvider::Buffer* buffer) 6227{ 6228 size_t stepCount = buffer->frameCount; 6229 if (stepCount == 0) { 6230 return; 6231 } 6232 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6233 mRsmpInUnrel -= stepCount; 6234 mRsmpInFront += stepCount; 6235 buffer->raw = NULL; 6236 buffer->frameCount = 0; 6237} 6238 6239AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6240 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6241 uint32_t srcSampleRate, 6242 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6243 uint32_t dstSampleRate) : 6244 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6245 // mSrcFormat 6246 // mSrcSampleRate 6247 // mDstChannelMask 6248 // mDstFormat 6249 // mDstSampleRate 6250 // mSrcChannelCount 6251 // mDstChannelCount 6252 // mDstFrameSize 6253 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6254 mResampler(NULL), 6255 mIsLegacyDownmix(false), 6256 mIsLegacyUpmix(false), 6257 mRequiresFloat(false), 6258 mInputConverterProvider(NULL) 6259{ 6260 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6261 dstChannelMask, dstFormat, dstSampleRate); 6262} 6263 6264AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6265 free(mBuf); 6266 delete mResampler; 6267 delete mInputConverterProvider; 6268} 6269 6270size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6271 AudioBufferProvider *provider, size_t frames) 6272{ 6273 if (mInputConverterProvider != NULL) { 6274 mInputConverterProvider->setBufferProvider(provider); 6275 provider = mInputConverterProvider; 6276 } 6277 6278 if (mResampler == NULL) { 6279 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6280 mSrcSampleRate, mSrcFormat, mDstFormat); 6281 6282 AudioBufferProvider::Buffer buffer; 6283 for (size_t i = frames; i > 0; ) { 6284 buffer.frameCount = i; 6285 status_t status = provider->getNextBuffer(&buffer, 0); 6286 if (status != OK || buffer.frameCount == 0) { 6287 frames -= i; // cannot fill request. 6288 break; 6289 } 6290 // format convert to destination buffer 6291 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6292 6293 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6294 i -= buffer.frameCount; 6295 provider->releaseBuffer(&buffer); 6296 } 6297 } else { 6298 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6299 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6300 6301 // reallocate buffer if needed 6302 if (mBufFrameSize != 0 && mBufFrames < frames) { 6303 free(mBuf); 6304 mBufFrames = frames; 6305 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6306 } 6307 // resampler accumulates, but we only have one source track 6308 memset(mBuf, 0, frames * mBufFrameSize); 6309 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6310 // format convert to destination buffer 6311 convertResampler(dst, mBuf, frames); 6312 } 6313 return frames; 6314} 6315 6316status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6317 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6318 uint32_t srcSampleRate, 6319 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6320 uint32_t dstSampleRate) 6321{ 6322 // quick evaluation if there is any change. 6323 if (mSrcFormat == srcFormat 6324 && mSrcChannelMask == srcChannelMask 6325 && mSrcSampleRate == srcSampleRate 6326 && mDstFormat == dstFormat 6327 && mDstChannelMask == dstChannelMask 6328 && mDstSampleRate == dstSampleRate) { 6329 return NO_ERROR; 6330 } 6331 6332 const bool valid = 6333 audio_is_input_channel(srcChannelMask) 6334 && audio_is_input_channel(dstChannelMask) 6335 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6336 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6337 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6338 ; // no upsampling checks for now 6339 if (!valid) { 6340 return BAD_VALUE; 6341 } 6342 6343 mSrcFormat = srcFormat; 6344 mSrcChannelMask = srcChannelMask; 6345 mSrcSampleRate = srcSampleRate; 6346 mDstFormat = dstFormat; 6347 mDstChannelMask = dstChannelMask; 6348 mDstSampleRate = dstSampleRate; 6349 6350 // compute derived parameters 6351 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6352 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6353 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6354 6355 // do we need to resample? 6356 delete mResampler; 6357 mResampler = NULL; 6358 if (mSrcSampleRate != mDstSampleRate) { 6359 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6360 mSrcChannelCount, mDstSampleRate); 6361 mResampler->setSampleRate(mSrcSampleRate); 6362 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6363 } 6364 6365 // are we running legacy channel conversion modes? 6366 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6367 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6368 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6369 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6370 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6371 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6372 6373 // do we need to process in float? 6374 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6375 6376 // do we need a staging buffer to convert for destination (we can still optimize this)? 6377 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6378 if (mResampler != NULL) { 6379 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6380 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6381 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6382 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6383 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6384 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6385 } else { 6386 mBufFrameSize = 0; 6387 } 6388 mBufFrames = 0; // force the buffer to be resized. 6389 6390 // do we need an input converter buffer provider to give us float? 6391 delete mInputConverterProvider; 6392 mInputConverterProvider = NULL; 6393 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6394 mInputConverterProvider = new ReformatBufferProvider( 6395 audio_channel_count_from_in_mask(mSrcChannelMask), 6396 mSrcFormat, 6397 AUDIO_FORMAT_PCM_FLOAT, 6398 256 /* provider buffer frame count */); 6399 } 6400 6401 // do we need a remixer to do channel mask conversion 6402 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6403 (void) memcpy_by_index_array_initialization_from_channel_mask( 6404 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6405 } 6406 return NO_ERROR; 6407} 6408 6409void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6410 void *dst, const void *src, size_t frames) 6411{ 6412 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6413 if (mBufFrameSize != 0 && mBufFrames < frames) { 6414 free(mBuf); 6415 mBufFrames = frames; 6416 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6417 } 6418 // do we need to do legacy upmix and downmix? 6419 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6420 void *dstBuf = mBuf != NULL ? mBuf : dst; 6421 if (mIsLegacyUpmix) { 6422 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6423 (const float *)src, frames); 6424 } else /*mIsLegacyDownmix */ { 6425 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6426 (const float *)src, frames); 6427 } 6428 if (mBuf != NULL) { 6429 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6430 frames * mDstChannelCount); 6431 } 6432 return; 6433 } 6434 // do we need to do channel mask conversion? 6435 if (mSrcChannelMask != mDstChannelMask) { 6436 void *dstBuf = mBuf != NULL ? mBuf : dst; 6437 memcpy_by_index_array(dstBuf, mDstChannelCount, 6438 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6439 if (dstBuf == dst) { 6440 return; // format is the same 6441 } 6442 } 6443 // convert to destination buffer 6444 const void *convertBuf = mBuf != NULL ? mBuf : src; 6445 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6446 frames * mDstChannelCount); 6447} 6448 6449void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6450 void *dst, /*not-a-const*/ void *src, size_t frames) 6451{ 6452 // src buffer format is ALWAYS float when entering this routine 6453 if (mIsLegacyUpmix) { 6454 ; // mono to stereo already handled by resampler 6455 } else if (mIsLegacyDownmix 6456 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6457 // the resampler outputs stereo for mono input channel (a feature?) 6458 // must convert to mono 6459 downmix_to_mono_float_from_stereo_float((float *)src, 6460 (const float *)src, frames); 6461 } else if (mSrcChannelMask != mDstChannelMask) { 6462 // convert to mono channel again for channel mask conversion (could be skipped 6463 // with further optimization). 6464 if (mSrcChannelCount == 1) { 6465 downmix_to_mono_float_from_stereo_float((float *)src, 6466 (const float *)src, frames); 6467 } 6468 // convert to destination format (in place, OK as float is larger than other types) 6469 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6470 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6471 frames * mSrcChannelCount); 6472 } 6473 // channel convert and save to dst 6474 memcpy_by_index_array(dst, mDstChannelCount, 6475 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6476 return; 6477 } 6478 // convert to destination format and save to dst 6479 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6480 frames * mDstChannelCount); 6481} 6482 6483bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6484 status_t& status) 6485{ 6486 bool reconfig = false; 6487 6488 status = NO_ERROR; 6489 6490 audio_format_t reqFormat = mFormat; 6491 uint32_t samplingRate = mSampleRate; 6492 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6493 // possible that we are > 2 channels, use channel index mask 6494 if (channelMask == AUDIO_CHANNEL_INVALID && mChannelCount <= FCC_8) { 6495 audio_channel_mask_for_index_assignment_from_count(mChannelCount); 6496 } 6497 6498 AudioParameter param = AudioParameter(keyValuePair); 6499 int value; 6500 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6501 // channel count change can be requested. Do we mandate the first client defines the 6502 // HAL sampling rate and channel count or do we allow changes on the fly? 6503 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6504 samplingRate = value; 6505 reconfig = true; 6506 } 6507 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6508 if (!audio_is_linear_pcm((audio_format_t) value)) { 6509 status = BAD_VALUE; 6510 } else { 6511 reqFormat = (audio_format_t) value; 6512 reconfig = true; 6513 } 6514 } 6515 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6516 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6517 if (!audio_is_input_channel(mask) || 6518 audio_channel_count_from_in_mask(mask) > FCC_8) { 6519 status = BAD_VALUE; 6520 } else { 6521 channelMask = mask; 6522 reconfig = true; 6523 } 6524 } 6525 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6526 // do not accept frame count changes if tracks are open as the track buffer 6527 // size depends on frame count and correct behavior would not be guaranteed 6528 // if frame count is changed after track creation 6529 if (mActiveTracks.size() > 0) { 6530 status = INVALID_OPERATION; 6531 } else { 6532 reconfig = true; 6533 } 6534 } 6535 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6536 // forward device change to effects that have requested to be 6537 // aware of attached audio device. 6538 for (size_t i = 0; i < mEffectChains.size(); i++) { 6539 mEffectChains[i]->setDevice_l(value); 6540 } 6541 6542 // store input device and output device but do not forward output device to audio HAL. 6543 // Note that status is ignored by the caller for output device 6544 // (see AudioFlinger::setParameters() 6545 if (audio_is_output_devices(value)) { 6546 mOutDevice = value; 6547 status = BAD_VALUE; 6548 } else { 6549 mInDevice = value; 6550 // disable AEC and NS if the device is a BT SCO headset supporting those 6551 // pre processings 6552 if (mTracks.size() > 0) { 6553 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6554 mAudioFlinger->btNrecIsOff(); 6555 for (size_t i = 0; i < mTracks.size(); i++) { 6556 sp<RecordTrack> track = mTracks[i]; 6557 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6558 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6559 } 6560 } 6561 } 6562 } 6563 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6564 mAudioSource != (audio_source_t)value) { 6565 // forward device change to effects that have requested to be 6566 // aware of attached audio device. 6567 for (size_t i = 0; i < mEffectChains.size(); i++) { 6568 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6569 } 6570 mAudioSource = (audio_source_t)value; 6571 } 6572 6573 if (status == NO_ERROR) { 6574 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6575 keyValuePair.string()); 6576 if (status == INVALID_OPERATION) { 6577 inputStandBy(); 6578 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6579 keyValuePair.string()); 6580 } 6581 if (reconfig) { 6582 if (status == BAD_VALUE && 6583 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6584 audio_is_linear_pcm(reqFormat) && 6585 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6586 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6587 audio_channel_count_from_in_mask( 6588 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6589 (channelMask == AUDIO_CHANNEL_IN_MONO || 6590 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6591 status = NO_ERROR; 6592 } 6593 if (status == NO_ERROR) { 6594 readInputParameters_l(); 6595 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6596 } 6597 } 6598 } 6599 6600 return reconfig; 6601} 6602 6603String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6604{ 6605 Mutex::Autolock _l(mLock); 6606 if (initCheck() != NO_ERROR) { 6607 return String8(); 6608 } 6609 6610 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6611 const String8 out_s8(s); 6612 free(s); 6613 return out_s8; 6614} 6615 6616void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6617 AudioSystem::OutputDescriptor desc; 6618 const void *param2 = NULL; 6619 6620 switch (event) { 6621 case AudioSystem::INPUT_OPENED: 6622 case AudioSystem::INPUT_CONFIG_CHANGED: 6623 desc.channelMask = mChannelMask; 6624 desc.samplingRate = mSampleRate; 6625 desc.format = mFormat; 6626 desc.frameCount = mFrameCount; 6627 desc.latency = 0; 6628 param2 = &desc; 6629 break; 6630 6631 case AudioSystem::INPUT_CLOSED: 6632 default: 6633 break; 6634 } 6635 mAudioFlinger->audioConfigChanged(event, mId, param2); 6636} 6637 6638void AudioFlinger::RecordThread::readInputParameters_l() 6639{ 6640 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6641 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6642 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6643 if (mChannelCount > FCC_8) { 6644 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6645 } 6646 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6647 mFormat = mHALFormat; 6648 if (!audio_is_linear_pcm(mFormat)) { 6649 ALOGE("HAL format %#x is not linear pcm", mFormat); 6650 } 6651 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6652 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6653 mFrameCount = mBufferSize / mFrameSize; 6654 // This is the formula for calculating the temporary buffer size. 6655 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6656 // 1 full output buffer, regardless of the alignment of the available input. 6657 // The value is somewhat arbitrary, and could probably be even larger. 6658 // A larger value should allow more old data to be read after a track calls start(), 6659 // without increasing latency. 6660 // 6661 // Note this is independent of the maximum downsampling ratio permitted for capture. 6662 mRsmpInFrames = mFrameCount * 7; 6663 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6664 free(mRsmpInBuffer); 6665 6666 // TODO optimize audio capture buffer sizes ... 6667 // Here we calculate the size of the sliding buffer used as a source 6668 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6669 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6670 // be better to have it derived from the pipe depth in the long term. 6671 // The current value is higher than necessary. However it should not add to latency. 6672 6673 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6674 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6675 6676 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6677 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6678} 6679 6680uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6681{ 6682 Mutex::Autolock _l(mLock); 6683 if (initCheck() != NO_ERROR) { 6684 return 0; 6685 } 6686 6687 return mInput->stream->get_input_frames_lost(mInput->stream); 6688} 6689 6690uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6691{ 6692 Mutex::Autolock _l(mLock); 6693 uint32_t result = 0; 6694 if (getEffectChain_l(sessionId) != 0) { 6695 result = EFFECT_SESSION; 6696 } 6697 6698 for (size_t i = 0; i < mTracks.size(); ++i) { 6699 if (sessionId == mTracks[i]->sessionId()) { 6700 result |= TRACK_SESSION; 6701 break; 6702 } 6703 } 6704 6705 return result; 6706} 6707 6708KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6709{ 6710 KeyedVector<int, bool> ids; 6711 Mutex::Autolock _l(mLock); 6712 for (size_t j = 0; j < mTracks.size(); ++j) { 6713 sp<RecordThread::RecordTrack> track = mTracks[j]; 6714 int sessionId = track->sessionId(); 6715 if (ids.indexOfKey(sessionId) < 0) { 6716 ids.add(sessionId, true); 6717 } 6718 } 6719 return ids; 6720} 6721 6722AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6723{ 6724 Mutex::Autolock _l(mLock); 6725 AudioStreamIn *input = mInput; 6726 mInput = NULL; 6727 return input; 6728} 6729 6730// this method must always be called either with ThreadBase mLock held or inside the thread loop 6731audio_stream_t* AudioFlinger::RecordThread::stream() const 6732{ 6733 if (mInput == NULL) { 6734 return NULL; 6735 } 6736 return &mInput->stream->common; 6737} 6738 6739status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6740{ 6741 // only one chain per input thread 6742 if (mEffectChains.size() != 0) { 6743 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6744 return INVALID_OPERATION; 6745 } 6746 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6747 chain->setThread(this); 6748 chain->setInBuffer(NULL); 6749 chain->setOutBuffer(NULL); 6750 6751 checkSuspendOnAddEffectChain_l(chain); 6752 6753 // make sure enabled pre processing effects state is communicated to the HAL as we 6754 // just moved them to a new input stream. 6755 chain->syncHalEffectsState(); 6756 6757 mEffectChains.add(chain); 6758 6759 return NO_ERROR; 6760} 6761 6762size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6763{ 6764 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6765 ALOGW_IF(mEffectChains.size() != 1, 6766 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6767 chain.get(), mEffectChains.size(), this); 6768 if (mEffectChains.size() == 1) { 6769 mEffectChains.removeAt(0); 6770 } 6771 return 0; 6772} 6773 6774status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6775 audio_patch_handle_t *handle) 6776{ 6777 status_t status = NO_ERROR; 6778 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6779 // store new device and send to effects 6780 mInDevice = patch->sources[0].ext.device.type; 6781 for (size_t i = 0; i < mEffectChains.size(); i++) { 6782 mEffectChains[i]->setDevice_l(mInDevice); 6783 } 6784 6785 // disable AEC and NS if the device is a BT SCO headset supporting those 6786 // pre processings 6787 if (mTracks.size() > 0) { 6788 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6789 mAudioFlinger->btNrecIsOff(); 6790 for (size_t i = 0; i < mTracks.size(); i++) { 6791 sp<RecordTrack> track = mTracks[i]; 6792 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6793 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6794 } 6795 } 6796 6797 // store new source and send to effects 6798 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6799 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6800 for (size_t i = 0; i < mEffectChains.size(); i++) { 6801 mEffectChains[i]->setAudioSource_l(mAudioSource); 6802 } 6803 } 6804 6805 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6806 status = hwDevice->create_audio_patch(hwDevice, 6807 patch->num_sources, 6808 patch->sources, 6809 patch->num_sinks, 6810 patch->sinks, 6811 handle); 6812 } else { 6813 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6814 } 6815 return status; 6816} 6817 6818status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6819{ 6820 status_t status = NO_ERROR; 6821 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6822 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6823 status = hwDevice->release_audio_patch(hwDevice, handle); 6824 } else { 6825 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6826 } 6827 return status; 6828} 6829 6830void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6831{ 6832 Mutex::Autolock _l(mLock); 6833 mTracks.add(record); 6834} 6835 6836void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6837{ 6838 Mutex::Autolock _l(mLock); 6839 destroyTrack_l(record); 6840} 6841 6842void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6843{ 6844 ThreadBase::getAudioPortConfig(config); 6845 config->role = AUDIO_PORT_ROLE_SINK; 6846 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6847 config->ext.mix.usecase.source = mAudioSource; 6848} 6849 6850} // namespace android 6851