Threads.cpp revision 054d9d3dea1390294650ac704acb4aa0a0731217
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::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 2937 audio_patch_handle_t *handle) 2938{ 2939 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2940 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2941 if (mFastMixer != 0) { 2942 FastMixerStateQueue *sq = mFastMixer->sq(); 2943 FastMixerState *state = sq->begin(); 2944 if (!(state->mCommand & FastMixerState::IDLE)) { 2945 previousCommand = state->mCommand; 2946 state->mCommand = FastMixerState::HOT_IDLE; 2947 sq->end(); 2948 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2949 } else { 2950 sq->end(false /*didModify*/); 2951 } 2952 } 2953 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 2954 2955 if (!(previousCommand & FastMixerState::IDLE)) { 2956 ALOG_ASSERT(mFastMixer != 0); 2957 FastMixerStateQueue *sq = mFastMixer->sq(); 2958 FastMixerState *state = sq->begin(); 2959 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 2960 state->mCommand = previousCommand; 2961 sq->end(); 2962 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2963 } 2964 2965 return status; 2966} 2967 2968status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2969 audio_patch_handle_t *handle) 2970{ 2971 status_t status = NO_ERROR; 2972 2973 // store new device and send to effects 2974 audio_devices_t type = AUDIO_DEVICE_NONE; 2975 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2976 type |= patch->sinks[i].ext.device.type; 2977 } 2978 2979#ifdef ADD_BATTERY_DATA 2980 // when changing the audio output device, call addBatteryData to notify 2981 // the change 2982 if (mOutDevice != type) { 2983 uint32_t params = 0; 2984 // check whether speaker is on 2985 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 2986 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2987 } 2988 2989 audio_devices_t deviceWithoutSpeaker 2990 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2991 // check if any other device (except speaker) is on 2992 if (type & deviceWithoutSpeaker) { 2993 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2994 } 2995 2996 if (params != 0) { 2997 addBatteryData(params); 2998 } 2999 } 3000#endif 3001 3002 for (size_t i = 0; i < mEffectChains.size(); i++) { 3003 mEffectChains[i]->setDevice_l(type); 3004 } 3005 mOutDevice = type; 3006 3007 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3008 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3009 status = hwDevice->create_audio_patch(hwDevice, 3010 patch->num_sources, 3011 patch->sources, 3012 patch->num_sinks, 3013 patch->sinks, 3014 handle); 3015 } else { 3016 char *address; 3017 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3018 //FIXME: we only support address on first sink with HAL version < 3.0 3019 address = audio_device_address_to_parameter( 3020 patch->sinks[0].ext.device.type, 3021 patch->sinks[0].ext.device.address); 3022 } else { 3023 address = (char *)calloc(1, 1); 3024 } 3025 AudioParameter param = AudioParameter(String8(address)); 3026 free(address); 3027 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3028 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3029 param.toString().string()); 3030 *handle = AUDIO_PATCH_HANDLE_NONE; 3031 } 3032 return status; 3033} 3034 3035status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3036{ 3037 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3038 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3039 if (mFastMixer != 0) { 3040 FastMixerStateQueue *sq = mFastMixer->sq(); 3041 FastMixerState *state = sq->begin(); 3042 if (!(state->mCommand & FastMixerState::IDLE)) { 3043 previousCommand = state->mCommand; 3044 state->mCommand = FastMixerState::HOT_IDLE; 3045 sq->end(); 3046 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3047 } else { 3048 sq->end(false /*didModify*/); 3049 } 3050 } 3051 3052 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3053 3054 if (!(previousCommand & FastMixerState::IDLE)) { 3055 ALOG_ASSERT(mFastMixer != 0); 3056 FastMixerStateQueue *sq = mFastMixer->sq(); 3057 FastMixerState *state = sq->begin(); 3058 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3059 state->mCommand = previousCommand; 3060 sq->end(); 3061 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3062 } 3063 3064 return status; 3065} 3066 3067status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3068{ 3069 status_t status = NO_ERROR; 3070 3071 mOutDevice = AUDIO_DEVICE_NONE; 3072 3073 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3074 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3075 status = hwDevice->release_audio_patch(hwDevice, handle); 3076 } else { 3077 AudioParameter param; 3078 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3079 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3080 param.toString().string()); 3081 } 3082 return status; 3083} 3084 3085void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3086{ 3087 Mutex::Autolock _l(mLock); 3088 mTracks.add(track); 3089} 3090 3091void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3092{ 3093 Mutex::Autolock _l(mLock); 3094 destroyTrack_l(track); 3095} 3096 3097void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3098{ 3099 ThreadBase::getAudioPortConfig(config); 3100 config->role = AUDIO_PORT_ROLE_SOURCE; 3101 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3102 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3103} 3104 3105// ---------------------------------------------------------------------------- 3106 3107AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3108 audio_io_handle_t id, audio_devices_t device, type_t type) 3109 : PlaybackThread(audioFlinger, output, id, device, type), 3110 // mAudioMixer below 3111 // mFastMixer below 3112 mFastMixerFutex(0) 3113 // mOutputSink below 3114 // mPipeSink below 3115 // mNormalSink below 3116{ 3117 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3118 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3119 "mFrameCount=%d, mNormalFrameCount=%d", 3120 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3121 mNormalFrameCount); 3122 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3123 3124 if (type == DUPLICATING) { 3125 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3126 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3127 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3128 return; 3129 } 3130 // create an NBAIO sink for the HAL output stream, and negotiate 3131 mOutputSink = new AudioStreamOutSink(output->stream); 3132 size_t numCounterOffers = 0; 3133 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3134 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3135 ALOG_ASSERT(index == 0); 3136 3137 // initialize fast mixer depending on configuration 3138 bool initFastMixer; 3139 switch (kUseFastMixer) { 3140 case FastMixer_Never: 3141 initFastMixer = false; 3142 break; 3143 case FastMixer_Always: 3144 initFastMixer = true; 3145 break; 3146 case FastMixer_Static: 3147 case FastMixer_Dynamic: 3148 initFastMixer = mFrameCount < mNormalFrameCount; 3149 break; 3150 } 3151 if (initFastMixer) { 3152 audio_format_t fastMixerFormat; 3153 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3154 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3155 } else { 3156 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3157 } 3158 if (mFormat != fastMixerFormat) { 3159 // change our Sink format to accept our intermediate precision 3160 mFormat = fastMixerFormat; 3161 free(mSinkBuffer); 3162 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3163 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3164 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3165 } 3166 3167 // create a MonoPipe to connect our submix to FastMixer 3168 NBAIO_Format format = mOutputSink->format(); 3169 NBAIO_Format origformat = format; 3170 // adjust format to match that of the Fast Mixer 3171 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3172 format.mFormat = fastMixerFormat; 3173 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3174 3175 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3176 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3177 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3178 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3179 const NBAIO_Format offers[1] = {format}; 3180 size_t numCounterOffers = 0; 3181 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3182 ALOG_ASSERT(index == 0); 3183 monoPipe->setAvgFrames((mScreenState & 1) ? 3184 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3185 mPipeSink = monoPipe; 3186 3187#ifdef TEE_SINK 3188 if (mTeeSinkOutputEnabled) { 3189 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3190 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3191 const NBAIO_Format offers2[1] = {origformat}; 3192 numCounterOffers = 0; 3193 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3194 ALOG_ASSERT(index == 0); 3195 mTeeSink = teeSink; 3196 PipeReader *teeSource = new PipeReader(*teeSink); 3197 numCounterOffers = 0; 3198 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3199 ALOG_ASSERT(index == 0); 3200 mTeeSource = teeSource; 3201 } 3202#endif 3203 3204 // create fast mixer and configure it initially with just one fast track for our submix 3205 mFastMixer = new FastMixer(); 3206 FastMixerStateQueue *sq = mFastMixer->sq(); 3207#ifdef STATE_QUEUE_DUMP 3208 sq->setObserverDump(&mStateQueueObserverDump); 3209 sq->setMutatorDump(&mStateQueueMutatorDump); 3210#endif 3211 FastMixerState *state = sq->begin(); 3212 FastTrack *fastTrack = &state->mFastTracks[0]; 3213 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3214 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3215 fastTrack->mVolumeProvider = NULL; 3216 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3217 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3218 fastTrack->mGeneration++; 3219 state->mFastTracksGen++; 3220 state->mTrackMask = 1; 3221 // fast mixer will use the HAL output sink 3222 state->mOutputSink = mOutputSink.get(); 3223 state->mOutputSinkGen++; 3224 state->mFrameCount = mFrameCount; 3225 state->mCommand = FastMixerState::COLD_IDLE; 3226 // already done in constructor initialization list 3227 //mFastMixerFutex = 0; 3228 state->mColdFutexAddr = &mFastMixerFutex; 3229 state->mColdGen++; 3230 state->mDumpState = &mFastMixerDumpState; 3231#ifdef TEE_SINK 3232 state->mTeeSink = mTeeSink.get(); 3233#endif 3234 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3235 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3236 sq->end(); 3237 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3238 3239 // start the fast mixer 3240 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3241 pid_t tid = mFastMixer->getTid(); 3242 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3243 if (err != 0) { 3244 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3245 kPriorityFastMixer, getpid_cached, tid, err); 3246 } 3247 3248#ifdef AUDIO_WATCHDOG 3249 // create and start the watchdog 3250 mAudioWatchdog = new AudioWatchdog(); 3251 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3252 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3253 tid = mAudioWatchdog->getTid(); 3254 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3255 if (err != 0) { 3256 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3257 kPriorityFastMixer, getpid_cached, tid, err); 3258 } 3259#endif 3260 3261 } 3262 3263 switch (kUseFastMixer) { 3264 case FastMixer_Never: 3265 case FastMixer_Dynamic: 3266 mNormalSink = mOutputSink; 3267 break; 3268 case FastMixer_Always: 3269 mNormalSink = mPipeSink; 3270 break; 3271 case FastMixer_Static: 3272 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3273 break; 3274 } 3275} 3276 3277AudioFlinger::MixerThread::~MixerThread() 3278{ 3279 if (mFastMixer != 0) { 3280 FastMixerStateQueue *sq = mFastMixer->sq(); 3281 FastMixerState *state = sq->begin(); 3282 if (state->mCommand == FastMixerState::COLD_IDLE) { 3283 int32_t old = android_atomic_inc(&mFastMixerFutex); 3284 if (old == -1) { 3285 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3286 } 3287 } 3288 state->mCommand = FastMixerState::EXIT; 3289 sq->end(); 3290 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3291 mFastMixer->join(); 3292 // Though the fast mixer thread has exited, it's state queue is still valid. 3293 // We'll use that extract the final state which contains one remaining fast track 3294 // corresponding to our sub-mix. 3295 state = sq->begin(); 3296 ALOG_ASSERT(state->mTrackMask == 1); 3297 FastTrack *fastTrack = &state->mFastTracks[0]; 3298 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3299 delete fastTrack->mBufferProvider; 3300 sq->end(false /*didModify*/); 3301 mFastMixer.clear(); 3302#ifdef AUDIO_WATCHDOG 3303 if (mAudioWatchdog != 0) { 3304 mAudioWatchdog->requestExit(); 3305 mAudioWatchdog->requestExitAndWait(); 3306 mAudioWatchdog.clear(); 3307 } 3308#endif 3309 } 3310 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3311 delete mAudioMixer; 3312} 3313 3314 3315uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3316{ 3317 if (mFastMixer != 0) { 3318 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3319 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3320 } 3321 return latency; 3322} 3323 3324 3325void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3326{ 3327 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3328} 3329 3330ssize_t AudioFlinger::MixerThread::threadLoop_write() 3331{ 3332 // FIXME we should only do one push per cycle; confirm this is true 3333 // Start the fast mixer if it's not already running 3334 if (mFastMixer != 0) { 3335 FastMixerStateQueue *sq = mFastMixer->sq(); 3336 FastMixerState *state = sq->begin(); 3337 if (state->mCommand != FastMixerState::MIX_WRITE && 3338 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3339 if (state->mCommand == FastMixerState::COLD_IDLE) { 3340 int32_t old = android_atomic_inc(&mFastMixerFutex); 3341 if (old == -1) { 3342 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3343 } 3344#ifdef AUDIO_WATCHDOG 3345 if (mAudioWatchdog != 0) { 3346 mAudioWatchdog->resume(); 3347 } 3348#endif 3349 } 3350 state->mCommand = FastMixerState::MIX_WRITE; 3351#ifdef FAST_THREAD_STATISTICS 3352 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3353 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3354#endif 3355 sq->end(); 3356 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3357 if (kUseFastMixer == FastMixer_Dynamic) { 3358 mNormalSink = mPipeSink; 3359 } 3360 } else { 3361 sq->end(false /*didModify*/); 3362 } 3363 } 3364 return PlaybackThread::threadLoop_write(); 3365} 3366 3367void AudioFlinger::MixerThread::threadLoop_standby() 3368{ 3369 // Idle the fast mixer if it's currently running 3370 if (mFastMixer != 0) { 3371 FastMixerStateQueue *sq = mFastMixer->sq(); 3372 FastMixerState *state = sq->begin(); 3373 if (!(state->mCommand & FastMixerState::IDLE)) { 3374 state->mCommand = FastMixerState::COLD_IDLE; 3375 state->mColdFutexAddr = &mFastMixerFutex; 3376 state->mColdGen++; 3377 mFastMixerFutex = 0; 3378 sq->end(); 3379 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3380 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3381 if (kUseFastMixer == FastMixer_Dynamic) { 3382 mNormalSink = mOutputSink; 3383 } 3384#ifdef AUDIO_WATCHDOG 3385 if (mAudioWatchdog != 0) { 3386 mAudioWatchdog->pause(); 3387 } 3388#endif 3389 } else { 3390 sq->end(false /*didModify*/); 3391 } 3392 } 3393 PlaybackThread::threadLoop_standby(); 3394} 3395 3396bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3397{ 3398 return false; 3399} 3400 3401bool AudioFlinger::PlaybackThread::shouldStandby_l() 3402{ 3403 return !mStandby; 3404} 3405 3406bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3407{ 3408 Mutex::Autolock _l(mLock); 3409 return waitingAsyncCallback_l(); 3410} 3411 3412// shared by MIXER and DIRECT, overridden by DUPLICATING 3413void AudioFlinger::PlaybackThread::threadLoop_standby() 3414{ 3415 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3416 mOutput->standby(); 3417 if (mUseAsyncWrite != 0) { 3418 // discard any pending drain or write ack by incrementing sequence 3419 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3420 mDrainSequence = (mDrainSequence + 2) & ~1; 3421 ALOG_ASSERT(mCallbackThread != 0); 3422 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3423 mCallbackThread->setDraining(mDrainSequence); 3424 } 3425 mHwPaused = false; 3426} 3427 3428void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3429{ 3430 ALOGV("signal playback thread"); 3431 broadcast_l(); 3432} 3433 3434void AudioFlinger::MixerThread::threadLoop_mix() 3435{ 3436 // obtain the presentation timestamp of the next output buffer 3437 int64_t pts; 3438 status_t status = INVALID_OPERATION; 3439 3440 if (mNormalSink != 0) { 3441 status = mNormalSink->getNextWriteTimestamp(&pts); 3442 } else { 3443 status = mOutputSink->getNextWriteTimestamp(&pts); 3444 } 3445 3446 if (status != NO_ERROR) { 3447 pts = AudioBufferProvider::kInvalidPTS; 3448 } 3449 3450 // mix buffers... 3451 mAudioMixer->process(pts); 3452 mCurrentWriteLength = mSinkBufferSize; 3453 // increase sleep time progressively when application underrun condition clears. 3454 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3455 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3456 // such that we would underrun the audio HAL. 3457 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3458 sleepTimeShift--; 3459 } 3460 sleepTime = 0; 3461 standbyTime = systemTime() + standbyDelay; 3462 //TODO: delay standby when effects have a tail 3463 3464} 3465 3466void AudioFlinger::MixerThread::threadLoop_sleepTime() 3467{ 3468 // If no tracks are ready, sleep once for the duration of an output 3469 // buffer size, then write 0s to the output 3470 if (sleepTime == 0) { 3471 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3472 sleepTime = activeSleepTime >> sleepTimeShift; 3473 if (sleepTime < kMinThreadSleepTimeUs) { 3474 sleepTime = kMinThreadSleepTimeUs; 3475 } 3476 // reduce sleep time in case of consecutive application underruns to avoid 3477 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3478 // duration we would end up writing less data than needed by the audio HAL if 3479 // the condition persists. 3480 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3481 sleepTimeShift++; 3482 } 3483 } else { 3484 sleepTime = idleSleepTime; 3485 } 3486 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3487 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3488 // before effects processing or output. 3489 if (mMixerBufferValid) { 3490 memset(mMixerBuffer, 0, mMixerBufferSize); 3491 } else { 3492 memset(mSinkBuffer, 0, mSinkBufferSize); 3493 } 3494 sleepTime = 0; 3495 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3496 "anticipated start"); 3497 } 3498 // TODO add standby time extension fct of effect tail 3499} 3500 3501// prepareTracks_l() must be called with ThreadBase::mLock held 3502AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3503 Vector< sp<Track> > *tracksToRemove) 3504{ 3505 3506 mixer_state mixerStatus = MIXER_IDLE; 3507 // find out which tracks need to be processed 3508 size_t count = mActiveTracks.size(); 3509 size_t mixedTracks = 0; 3510 size_t tracksWithEffect = 0; 3511 // counts only _active_ fast tracks 3512 size_t fastTracks = 0; 3513 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3514 3515 float masterVolume = mMasterVolume; 3516 bool masterMute = mMasterMute; 3517 3518 if (masterMute) { 3519 masterVolume = 0; 3520 } 3521 // Delegate master volume control to effect in output mix effect chain if needed 3522 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3523 if (chain != 0) { 3524 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3525 chain->setVolume_l(&v, &v); 3526 masterVolume = (float)((v + (1 << 23)) >> 24); 3527 chain.clear(); 3528 } 3529 3530 // prepare a new state to push 3531 FastMixerStateQueue *sq = NULL; 3532 FastMixerState *state = NULL; 3533 bool didModify = false; 3534 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3535 if (mFastMixer != 0) { 3536 sq = mFastMixer->sq(); 3537 state = sq->begin(); 3538 } 3539 3540 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3541 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3542 3543 for (size_t i=0 ; i<count ; i++) { 3544 const sp<Track> t = mActiveTracks[i].promote(); 3545 if (t == 0) { 3546 continue; 3547 } 3548 3549 // this const just means the local variable doesn't change 3550 Track* const track = t.get(); 3551 3552 // process fast tracks 3553 if (track->isFastTrack()) { 3554 3555 // It's theoretically possible (though unlikely) for a fast track to be created 3556 // and then removed within the same normal mix cycle. This is not a problem, as 3557 // the track never becomes active so it's fast mixer slot is never touched. 3558 // The converse, of removing an (active) track and then creating a new track 3559 // at the identical fast mixer slot within the same normal mix cycle, 3560 // is impossible because the slot isn't marked available until the end of each cycle. 3561 int j = track->mFastIndex; 3562 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3563 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3564 FastTrack *fastTrack = &state->mFastTracks[j]; 3565 3566 // Determine whether the track is currently in underrun condition, 3567 // and whether it had a recent underrun. 3568 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3569 FastTrackUnderruns underruns = ftDump->mUnderruns; 3570 uint32_t recentFull = (underruns.mBitFields.mFull - 3571 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3572 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3573 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3574 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3575 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3576 uint32_t recentUnderruns = recentPartial + recentEmpty; 3577 track->mObservedUnderruns = underruns; 3578 // don't count underruns that occur while stopping or pausing 3579 // or stopped which can occur when flush() is called while active 3580 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3581 recentUnderruns > 0) { 3582 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3583 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3584 } 3585 3586 // This is similar to the state machine for normal tracks, 3587 // with a few modifications for fast tracks. 3588 bool isActive = true; 3589 switch (track->mState) { 3590 case TrackBase::STOPPING_1: 3591 // track stays active in STOPPING_1 state until first underrun 3592 if (recentUnderruns > 0 || track->isTerminated()) { 3593 track->mState = TrackBase::STOPPING_2; 3594 } 3595 break; 3596 case TrackBase::PAUSING: 3597 // ramp down is not yet implemented 3598 track->setPaused(); 3599 break; 3600 case TrackBase::RESUMING: 3601 // ramp up is not yet implemented 3602 track->mState = TrackBase::ACTIVE; 3603 break; 3604 case TrackBase::ACTIVE: 3605 if (recentFull > 0 || recentPartial > 0) { 3606 // track has provided at least some frames recently: reset retry count 3607 track->mRetryCount = kMaxTrackRetries; 3608 } 3609 if (recentUnderruns == 0) { 3610 // no recent underruns: stay active 3611 break; 3612 } 3613 // there has recently been an underrun of some kind 3614 if (track->sharedBuffer() == 0) { 3615 // were any of the recent underruns "empty" (no frames available)? 3616 if (recentEmpty == 0) { 3617 // no, then ignore the partial underruns as they are allowed indefinitely 3618 break; 3619 } 3620 // there has recently been an "empty" underrun: decrement the retry counter 3621 if (--(track->mRetryCount) > 0) { 3622 break; 3623 } 3624 // indicate to client process that the track was disabled because of underrun; 3625 // it will then automatically call start() when data is available 3626 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3627 // remove from active list, but state remains ACTIVE [confusing but true] 3628 isActive = false; 3629 break; 3630 } 3631 // fall through 3632 case TrackBase::STOPPING_2: 3633 case TrackBase::PAUSED: 3634 case TrackBase::STOPPED: 3635 case TrackBase::FLUSHED: // flush() while active 3636 // Check for presentation complete if track is inactive 3637 // We have consumed all the buffers of this track. 3638 // This would be incomplete if we auto-paused on underrun 3639 { 3640 size_t audioHALFrames = 3641 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3642 size_t framesWritten = mBytesWritten / mFrameSize; 3643 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3644 // track stays in active list until presentation is complete 3645 break; 3646 } 3647 } 3648 if (track->isStopping_2()) { 3649 track->mState = TrackBase::STOPPED; 3650 } 3651 if (track->isStopped()) { 3652 // Can't reset directly, as fast mixer is still polling this track 3653 // track->reset(); 3654 // So instead mark this track as needing to be reset after push with ack 3655 resetMask |= 1 << i; 3656 } 3657 isActive = false; 3658 break; 3659 case TrackBase::IDLE: 3660 default: 3661 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3662 } 3663 3664 if (isActive) { 3665 // was it previously inactive? 3666 if (!(state->mTrackMask & (1 << j))) { 3667 ExtendedAudioBufferProvider *eabp = track; 3668 VolumeProvider *vp = track; 3669 fastTrack->mBufferProvider = eabp; 3670 fastTrack->mVolumeProvider = vp; 3671 fastTrack->mChannelMask = track->mChannelMask; 3672 fastTrack->mFormat = track->mFormat; 3673 fastTrack->mGeneration++; 3674 state->mTrackMask |= 1 << j; 3675 didModify = true; 3676 // no acknowledgement required for newly active tracks 3677 } 3678 // cache the combined master volume and stream type volume for fast mixer; this 3679 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3680 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3681 ++fastTracks; 3682 } else { 3683 // was it previously active? 3684 if (state->mTrackMask & (1 << j)) { 3685 fastTrack->mBufferProvider = NULL; 3686 fastTrack->mGeneration++; 3687 state->mTrackMask &= ~(1 << j); 3688 didModify = true; 3689 // If any fast tracks were removed, we must wait for acknowledgement 3690 // because we're about to decrement the last sp<> on those tracks. 3691 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3692 } else { 3693 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3694 } 3695 tracksToRemove->add(track); 3696 // Avoids a misleading display in dumpsys 3697 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3698 } 3699 continue; 3700 } 3701 3702 { // local variable scope to avoid goto warning 3703 3704 audio_track_cblk_t* cblk = track->cblk(); 3705 3706 // The first time a track is added we wait 3707 // for all its buffers to be filled before processing it 3708 int name = track->name(); 3709 // make sure that we have enough frames to mix one full buffer. 3710 // enforce this condition only once to enable draining the buffer in case the client 3711 // app does not call stop() and relies on underrun to stop: 3712 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3713 // during last round 3714 size_t desiredFrames; 3715 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3716 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3717 3718 desiredFrames = sourceFramesNeededWithTimestretch( 3719 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3720 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3721 // add frames already consumed but not yet released by the resampler 3722 // because mAudioTrackServerProxy->framesReady() will include these frames 3723 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3724 3725 uint32_t minFrames = 1; 3726 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3727 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3728 minFrames = desiredFrames; 3729 } 3730 3731 size_t framesReady = track->framesReady(); 3732 if (ATRACE_ENABLED()) { 3733 // I wish we had formatted trace names 3734 char traceName[16]; 3735 strcpy(traceName, "nRdy"); 3736 int name = track->name(); 3737 if (AudioMixer::TRACK0 <= name && 3738 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3739 name -= AudioMixer::TRACK0; 3740 traceName[4] = (name / 10) + '0'; 3741 traceName[5] = (name % 10) + '0'; 3742 } else { 3743 traceName[4] = '?'; 3744 traceName[5] = '?'; 3745 } 3746 traceName[6] = '\0'; 3747 ATRACE_INT(traceName, framesReady); 3748 } 3749 if ((framesReady >= minFrames) && track->isReady() && 3750 !track->isPaused() && !track->isTerminated()) 3751 { 3752 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3753 3754 mixedTracks++; 3755 3756 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3757 // there is an effect chain connected to the track 3758 chain.clear(); 3759 if (track->mainBuffer() != mSinkBuffer && 3760 track->mainBuffer() != mMixerBuffer) { 3761 if (mEffectBufferEnabled) { 3762 mEffectBufferValid = true; // Later can set directly. 3763 } 3764 chain = getEffectChain_l(track->sessionId()); 3765 // Delegate volume control to effect in track effect chain if needed 3766 if (chain != 0) { 3767 tracksWithEffect++; 3768 } else { 3769 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3770 "session %d", 3771 name, track->sessionId()); 3772 } 3773 } 3774 3775 3776 int param = AudioMixer::VOLUME; 3777 if (track->mFillingUpStatus == Track::FS_FILLED) { 3778 // no ramp for the first volume setting 3779 track->mFillingUpStatus = Track::FS_ACTIVE; 3780 if (track->mState == TrackBase::RESUMING) { 3781 track->mState = TrackBase::ACTIVE; 3782 param = AudioMixer::RAMP_VOLUME; 3783 } 3784 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3785 // FIXME should not make a decision based on mServer 3786 } else if (cblk->mServer != 0) { 3787 // If the track is stopped before the first frame was mixed, 3788 // do not apply ramp 3789 param = AudioMixer::RAMP_VOLUME; 3790 } 3791 3792 // compute volume for this track 3793 uint32_t vl, vr; // in U8.24 integer format 3794 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3795 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3796 vl = vr = 0; 3797 vlf = vrf = vaf = 0.; 3798 if (track->isPausing()) { 3799 track->setPaused(); 3800 } 3801 } else { 3802 3803 // read original volumes with volume control 3804 float typeVolume = mStreamTypes[track->streamType()].volume; 3805 float v = masterVolume * typeVolume; 3806 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3807 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3808 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3809 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3810 // track volumes come from shared memory, so can't be trusted and must be clamped 3811 if (vlf > GAIN_FLOAT_UNITY) { 3812 ALOGV("Track left volume out of range: %.3g", vlf); 3813 vlf = GAIN_FLOAT_UNITY; 3814 } 3815 if (vrf > GAIN_FLOAT_UNITY) { 3816 ALOGV("Track right volume out of range: %.3g", vrf); 3817 vrf = GAIN_FLOAT_UNITY; 3818 } 3819 // now apply the master volume and stream type volume 3820 vlf *= v; 3821 vrf *= v; 3822 // assuming master volume and stream type volume each go up to 1.0, 3823 // then derive vl and vr as U8.24 versions for the effect chain 3824 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3825 vl = (uint32_t) (scaleto8_24 * vlf); 3826 vr = (uint32_t) (scaleto8_24 * vrf); 3827 // vl and vr are now in U8.24 format 3828 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3829 // send level comes from shared memory and so may be corrupt 3830 if (sendLevel > MAX_GAIN_INT) { 3831 ALOGV("Track send level out of range: %04X", sendLevel); 3832 sendLevel = MAX_GAIN_INT; 3833 } 3834 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3835 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3836 } 3837 3838 // Delegate volume control to effect in track effect chain if needed 3839 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3840 // Do not ramp volume if volume is controlled by effect 3841 param = AudioMixer::VOLUME; 3842 // Update remaining floating point volume levels 3843 vlf = (float)vl / (1 << 24); 3844 vrf = (float)vr / (1 << 24); 3845 track->mHasVolumeController = true; 3846 } else { 3847 // force no volume ramp when volume controller was just disabled or removed 3848 // from effect chain to avoid volume spike 3849 if (track->mHasVolumeController) { 3850 param = AudioMixer::VOLUME; 3851 } 3852 track->mHasVolumeController = false; 3853 } 3854 3855 // XXX: these things DON'T need to be done each time 3856 mAudioMixer->setBufferProvider(name, track); 3857 mAudioMixer->enable(name); 3858 3859 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3860 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3861 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3862 mAudioMixer->setParameter( 3863 name, 3864 AudioMixer::TRACK, 3865 AudioMixer::FORMAT, (void *)track->format()); 3866 mAudioMixer->setParameter( 3867 name, 3868 AudioMixer::TRACK, 3869 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3870 mAudioMixer->setParameter( 3871 name, 3872 AudioMixer::TRACK, 3873 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3874 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3875 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3876 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3877 if (reqSampleRate == 0) { 3878 reqSampleRate = mSampleRate; 3879 } else if (reqSampleRate > maxSampleRate) { 3880 reqSampleRate = maxSampleRate; 3881 } 3882 mAudioMixer->setParameter( 3883 name, 3884 AudioMixer::RESAMPLE, 3885 AudioMixer::SAMPLE_RATE, 3886 (void *)(uintptr_t)reqSampleRate); 3887 3888 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3889 mAudioMixer->setParameter( 3890 name, 3891 AudioMixer::TIMESTRETCH, 3892 AudioMixer::PLAYBACK_RATE, 3893 &playbackRate); 3894 3895 /* 3896 * Select the appropriate output buffer for the track. 3897 * 3898 * Tracks with effects go into their own effects chain buffer 3899 * and from there into either mEffectBuffer or mSinkBuffer. 3900 * 3901 * Other tracks can use mMixerBuffer for higher precision 3902 * channel accumulation. If this buffer is enabled 3903 * (mMixerBufferEnabled true), then selected tracks will accumulate 3904 * into it. 3905 * 3906 */ 3907 if (mMixerBufferEnabled 3908 && (track->mainBuffer() == mSinkBuffer 3909 || track->mainBuffer() == mMixerBuffer)) { 3910 mAudioMixer->setParameter( 3911 name, 3912 AudioMixer::TRACK, 3913 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3914 mAudioMixer->setParameter( 3915 name, 3916 AudioMixer::TRACK, 3917 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3918 // TODO: override track->mainBuffer()? 3919 mMixerBufferValid = true; 3920 } else { 3921 mAudioMixer->setParameter( 3922 name, 3923 AudioMixer::TRACK, 3924 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3925 mAudioMixer->setParameter( 3926 name, 3927 AudioMixer::TRACK, 3928 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3929 } 3930 mAudioMixer->setParameter( 3931 name, 3932 AudioMixer::TRACK, 3933 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3934 3935 // reset retry count 3936 track->mRetryCount = kMaxTrackRetries; 3937 3938 // If one track is ready, set the mixer ready if: 3939 // - the mixer was not ready during previous round OR 3940 // - no other track is not ready 3941 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3942 mixerStatus != MIXER_TRACKS_ENABLED) { 3943 mixerStatus = MIXER_TRACKS_READY; 3944 } 3945 } else { 3946 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3947 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3948 } 3949 // clear effect chain input buffer if an active track underruns to avoid sending 3950 // previous audio buffer again to effects 3951 chain = getEffectChain_l(track->sessionId()); 3952 if (chain != 0) { 3953 chain->clearInputBuffer(); 3954 } 3955 3956 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3957 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3958 track->isStopped() || track->isPaused()) { 3959 // We have consumed all the buffers of this track. 3960 // Remove it from the list of active tracks. 3961 // TODO: use actual buffer filling status instead of latency when available from 3962 // audio HAL 3963 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3964 size_t framesWritten = mBytesWritten / mFrameSize; 3965 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3966 if (track->isStopped()) { 3967 track->reset(); 3968 } 3969 tracksToRemove->add(track); 3970 } 3971 } else { 3972 // No buffers for this track. Give it a few chances to 3973 // fill a buffer, then remove it from active list. 3974 if (--(track->mRetryCount) <= 0) { 3975 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3976 tracksToRemove->add(track); 3977 // indicate to client process that the track was disabled because of underrun; 3978 // it will then automatically call start() when data is available 3979 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3980 // If one track is not ready, mark the mixer also not ready if: 3981 // - the mixer was ready during previous round OR 3982 // - no other track is ready 3983 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3984 mixerStatus != MIXER_TRACKS_READY) { 3985 mixerStatus = MIXER_TRACKS_ENABLED; 3986 } 3987 } 3988 mAudioMixer->disable(name); 3989 } 3990 3991 } // local variable scope to avoid goto warning 3992track_is_ready: ; 3993 3994 } 3995 3996 // Push the new FastMixer state if necessary 3997 bool pauseAudioWatchdog = false; 3998 if (didModify) { 3999 state->mFastTracksGen++; 4000 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4001 if (kUseFastMixer == FastMixer_Dynamic && 4002 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4003 state->mCommand = FastMixerState::COLD_IDLE; 4004 state->mColdFutexAddr = &mFastMixerFutex; 4005 state->mColdGen++; 4006 mFastMixerFutex = 0; 4007 if (kUseFastMixer == FastMixer_Dynamic) { 4008 mNormalSink = mOutputSink; 4009 } 4010 // If we go into cold idle, need to wait for acknowledgement 4011 // so that fast mixer stops doing I/O. 4012 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4013 pauseAudioWatchdog = true; 4014 } 4015 } 4016 if (sq != NULL) { 4017 sq->end(didModify); 4018 sq->push(block); 4019 } 4020#ifdef AUDIO_WATCHDOG 4021 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4022 mAudioWatchdog->pause(); 4023 } 4024#endif 4025 4026 // Now perform the deferred reset on fast tracks that have stopped 4027 while (resetMask != 0) { 4028 size_t i = __builtin_ctz(resetMask); 4029 ALOG_ASSERT(i < count); 4030 resetMask &= ~(1 << i); 4031 sp<Track> t = mActiveTracks[i].promote(); 4032 if (t == 0) { 4033 continue; 4034 } 4035 Track* track = t.get(); 4036 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4037 track->reset(); 4038 } 4039 4040 // remove all the tracks that need to be... 4041 removeTracks_l(*tracksToRemove); 4042 4043 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4044 mEffectBufferValid = true; 4045 } 4046 4047 if (mEffectBufferValid) { 4048 // as long as there are effects we should clear the effects buffer, to avoid 4049 // passing a non-clean buffer to the effect chain 4050 memset(mEffectBuffer, 0, mEffectBufferSize); 4051 } 4052 // sink or mix buffer must be cleared if all tracks are connected to an 4053 // effect chain as in this case the mixer will not write to the sink or mix buffer 4054 // and track effects will accumulate into it 4055 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4056 (mixedTracks == 0 && fastTracks > 0))) { 4057 // FIXME as a performance optimization, should remember previous zero status 4058 if (mMixerBufferValid) { 4059 memset(mMixerBuffer, 0, mMixerBufferSize); 4060 // TODO: In testing, mSinkBuffer below need not be cleared because 4061 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4062 // after mixing. 4063 // 4064 // To enforce this guarantee: 4065 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4066 // (mixedTracks == 0 && fastTracks > 0)) 4067 // must imply MIXER_TRACKS_READY. 4068 // Later, we may clear buffers regardless, and skip much of this logic. 4069 } 4070 // FIXME as a performance optimization, should remember previous zero status 4071 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4072 } 4073 4074 // if any fast tracks, then status is ready 4075 mMixerStatusIgnoringFastTracks = mixerStatus; 4076 if (fastTracks > 0) { 4077 mixerStatus = MIXER_TRACKS_READY; 4078 } 4079 return mixerStatus; 4080} 4081 4082// getTrackName_l() must be called with ThreadBase::mLock held 4083int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4084 audio_format_t format, int sessionId) 4085{ 4086 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4087} 4088 4089// deleteTrackName_l() must be called with ThreadBase::mLock held 4090void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4091{ 4092 ALOGV("remove track (%d) and delete from mixer", name); 4093 mAudioMixer->deleteTrackName(name); 4094} 4095 4096// checkForNewParameter_l() must be called with ThreadBase::mLock held 4097bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4098 status_t& status) 4099{ 4100 bool reconfig = false; 4101 4102 status = NO_ERROR; 4103 4104 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4105 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4106 if (mFastMixer != 0) { 4107 FastMixerStateQueue *sq = mFastMixer->sq(); 4108 FastMixerState *state = sq->begin(); 4109 if (!(state->mCommand & FastMixerState::IDLE)) { 4110 previousCommand = state->mCommand; 4111 state->mCommand = FastMixerState::HOT_IDLE; 4112 sq->end(); 4113 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4114 } else { 4115 sq->end(false /*didModify*/); 4116 } 4117 } 4118 4119 AudioParameter param = AudioParameter(keyValuePair); 4120 int value; 4121 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4122 reconfig = true; 4123 } 4124 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4125 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4126 status = BAD_VALUE; 4127 } else { 4128 // no need to save value, since it's constant 4129 reconfig = true; 4130 } 4131 } 4132 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4133 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4134 status = BAD_VALUE; 4135 } else { 4136 // no need to save value, since it's constant 4137 reconfig = true; 4138 } 4139 } 4140 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4141 // do not accept frame count changes if tracks are open as the track buffer 4142 // size depends on frame count and correct behavior would not be guaranteed 4143 // if frame count is changed after track creation 4144 if (!mTracks.isEmpty()) { 4145 status = INVALID_OPERATION; 4146 } else { 4147 reconfig = true; 4148 } 4149 } 4150 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4151#ifdef ADD_BATTERY_DATA 4152 // when changing the audio output device, call addBatteryData to notify 4153 // the change 4154 if (mOutDevice != value) { 4155 uint32_t params = 0; 4156 // check whether speaker is on 4157 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4158 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4159 } 4160 4161 audio_devices_t deviceWithoutSpeaker 4162 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4163 // check if any other device (except speaker) is on 4164 if (value & deviceWithoutSpeaker) { 4165 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4166 } 4167 4168 if (params != 0) { 4169 addBatteryData(params); 4170 } 4171 } 4172#endif 4173 4174 // forward device change to effects that have requested to be 4175 // aware of attached audio device. 4176 if (value != AUDIO_DEVICE_NONE) { 4177 mOutDevice = value; 4178 for (size_t i = 0; i < mEffectChains.size(); i++) { 4179 mEffectChains[i]->setDevice_l(mOutDevice); 4180 } 4181 } 4182 } 4183 4184 if (status == NO_ERROR) { 4185 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4186 keyValuePair.string()); 4187 if (!mStandby && status == INVALID_OPERATION) { 4188 mOutput->standby(); 4189 mStandby = true; 4190 mBytesWritten = 0; 4191 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4192 keyValuePair.string()); 4193 } 4194 if (status == NO_ERROR && reconfig) { 4195 readOutputParameters_l(); 4196 delete mAudioMixer; 4197 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4198 for (size_t i = 0; i < mTracks.size() ; i++) { 4199 int name = getTrackName_l(mTracks[i]->mChannelMask, 4200 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4201 if (name < 0) { 4202 break; 4203 } 4204 mTracks[i]->mName = name; 4205 } 4206 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4207 } 4208 } 4209 4210 if (!(previousCommand & FastMixerState::IDLE)) { 4211 ALOG_ASSERT(mFastMixer != 0); 4212 FastMixerStateQueue *sq = mFastMixer->sq(); 4213 FastMixerState *state = sq->begin(); 4214 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4215 state->mCommand = previousCommand; 4216 sq->end(); 4217 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4218 } 4219 4220 return reconfig; 4221} 4222 4223 4224void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4225{ 4226 const size_t SIZE = 256; 4227 char buffer[SIZE]; 4228 String8 result; 4229 4230 PlaybackThread::dumpInternals(fd, args); 4231 4232 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4233 4234 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4235 const FastMixerDumpState copy(mFastMixerDumpState); 4236 copy.dump(fd); 4237 4238#ifdef STATE_QUEUE_DUMP 4239 // Similar for state queue 4240 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4241 observerCopy.dump(fd); 4242 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4243 mutatorCopy.dump(fd); 4244#endif 4245 4246#ifdef TEE_SINK 4247 // Write the tee output to a .wav file 4248 dumpTee(fd, mTeeSource, mId); 4249#endif 4250 4251#ifdef AUDIO_WATCHDOG 4252 if (mAudioWatchdog != 0) { 4253 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4254 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4255 wdCopy.dump(fd); 4256 } 4257#endif 4258} 4259 4260uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4261{ 4262 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4263} 4264 4265uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4266{ 4267 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4268} 4269 4270void AudioFlinger::MixerThread::cacheParameters_l() 4271{ 4272 PlaybackThread::cacheParameters_l(); 4273 4274 // FIXME: Relaxed timing because of a certain device that can't meet latency 4275 // Should be reduced to 2x after the vendor fixes the driver issue 4276 // increase threshold again due to low power audio mode. The way this warning 4277 // threshold is calculated and its usefulness should be reconsidered anyway. 4278 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4279} 4280 4281// ---------------------------------------------------------------------------- 4282 4283AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4284 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4285 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4286 // mLeftVolFloat, mRightVolFloat 4287{ 4288} 4289 4290AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4291 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4292 ThreadBase::type_t type) 4293 : PlaybackThread(audioFlinger, output, id, device, type) 4294 // mLeftVolFloat, mRightVolFloat 4295{ 4296} 4297 4298AudioFlinger::DirectOutputThread::~DirectOutputThread() 4299{ 4300} 4301 4302void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4303{ 4304 audio_track_cblk_t* cblk = track->cblk(); 4305 float left, right; 4306 4307 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4308 left = right = 0; 4309 } else { 4310 float typeVolume = mStreamTypes[track->streamType()].volume; 4311 float v = mMasterVolume * typeVolume; 4312 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4313 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4314 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4315 if (left > GAIN_FLOAT_UNITY) { 4316 left = GAIN_FLOAT_UNITY; 4317 } 4318 left *= v; 4319 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4320 if (right > GAIN_FLOAT_UNITY) { 4321 right = GAIN_FLOAT_UNITY; 4322 } 4323 right *= v; 4324 } 4325 4326 if (lastTrack) { 4327 if (left != mLeftVolFloat || right != mRightVolFloat) { 4328 mLeftVolFloat = left; 4329 mRightVolFloat = right; 4330 4331 // Convert volumes from float to 8.24 4332 uint32_t vl = (uint32_t)(left * (1 << 24)); 4333 uint32_t vr = (uint32_t)(right * (1 << 24)); 4334 4335 // Delegate volume control to effect in track effect chain if needed 4336 // only one effect chain can be present on DirectOutputThread, so if 4337 // there is one, the track is connected to it 4338 if (!mEffectChains.isEmpty()) { 4339 mEffectChains[0]->setVolume_l(&vl, &vr); 4340 left = (float)vl / (1 << 24); 4341 right = (float)vr / (1 << 24); 4342 } 4343 if (mOutput->stream->set_volume) { 4344 mOutput->stream->set_volume(mOutput->stream, left, right); 4345 } 4346 } 4347 } 4348} 4349 4350 4351AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4352 Vector< sp<Track> > *tracksToRemove 4353) 4354{ 4355 size_t count = mActiveTracks.size(); 4356 mixer_state mixerStatus = MIXER_IDLE; 4357 bool doHwPause = false; 4358 bool doHwResume = false; 4359 bool flushPending = false; 4360 4361 // find out which tracks need to be processed 4362 for (size_t i = 0; i < count; i++) { 4363 sp<Track> t = mActiveTracks[i].promote(); 4364 // The track died recently 4365 if (t == 0) { 4366 continue; 4367 } 4368 4369 Track* const track = t.get(); 4370 audio_track_cblk_t* cblk = track->cblk(); 4371 // Only consider last track started for volume and mixer state control. 4372 // In theory an older track could underrun and restart after the new one starts 4373 // but as we only care about the transition phase between two tracks on a 4374 // direct output, it is not a problem to ignore the underrun case. 4375 sp<Track> l = mLatestActiveTrack.promote(); 4376 bool last = l.get() == track; 4377 4378 if (mHwSupportsPause && track->isPausing()) { 4379 track->setPaused(); 4380 if (last && !mHwPaused) { 4381 doHwPause = true; 4382 mHwPaused = true; 4383 } 4384 tracksToRemove->add(track); 4385 } else if (track->isFlushPending()) { 4386 track->flushAck(); 4387 if (last) { 4388 flushPending = true; 4389 } 4390 } else if (mHwSupportsPause && track->isResumePending()){ 4391 track->resumeAck(); 4392 if (last) { 4393 if (mHwPaused) { 4394 doHwResume = true; 4395 mHwPaused = false; 4396 } 4397 } 4398 } 4399 4400 // The first time a track is added we wait 4401 // for all its buffers to be filled before processing it. 4402 // Allow draining the buffer in case the client 4403 // app does not call stop() and relies on underrun to stop: 4404 // hence the test on (track->mRetryCount > 1). 4405 // If retryCount<=1 then track is about to underrun and be removed. 4406 uint32_t minFrames; 4407 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4408 && (track->mRetryCount > 1)) { 4409 minFrames = mNormalFrameCount; 4410 } else { 4411 minFrames = 1; 4412 } 4413 4414 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4415 !track->isStopping_2() && !track->isStopped()) 4416 { 4417 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4418 4419 if (track->mFillingUpStatus == Track::FS_FILLED) { 4420 track->mFillingUpStatus = Track::FS_ACTIVE; 4421 // make sure processVolume_l() will apply new volume even if 0 4422 mLeftVolFloat = mRightVolFloat = -1.0; 4423 if (!mHwSupportsPause) { 4424 track->resumeAck(); 4425 } 4426 } 4427 4428 // compute volume for this track 4429 processVolume_l(track, last); 4430 if (last) { 4431 // reset retry count 4432 track->mRetryCount = kMaxTrackRetriesDirect; 4433 mActiveTrack = t; 4434 mixerStatus = MIXER_TRACKS_READY; 4435 if (usesHwAvSync() && mHwPaused) { 4436 doHwResume = true; 4437 mHwPaused = false; 4438 } 4439 } 4440 } else { 4441 // clear effect chain input buffer if the last active track started underruns 4442 // to avoid sending previous audio buffer again to effects 4443 if (!mEffectChains.isEmpty() && last) { 4444 mEffectChains[0]->clearInputBuffer(); 4445 } 4446 if (track->isStopping_1()) { 4447 track->mState = TrackBase::STOPPING_2; 4448 if (last && mHwPaused) { 4449 doHwResume = true; 4450 mHwPaused = false; 4451 } 4452 } 4453 if ((track->sharedBuffer() != 0) || track->isStopped() || 4454 track->isStopping_2() || track->isPaused()) { 4455 // We have consumed all the buffers of this track. 4456 // Remove it from the list of active tracks. 4457 size_t audioHALFrames; 4458 if (audio_is_linear_pcm(mFormat)) { 4459 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4460 } else { 4461 audioHALFrames = 0; 4462 } 4463 4464 size_t framesWritten = mBytesWritten / mFrameSize; 4465 if (mStandby || !last || 4466 track->presentationComplete(framesWritten, audioHALFrames)) { 4467 if (track->isStopping_2()) { 4468 track->mState = TrackBase::STOPPED; 4469 } 4470 if (track->isStopped()) { 4471 track->reset(); 4472 } 4473 tracksToRemove->add(track); 4474 } 4475 } else { 4476 // No buffers for this track. Give it a few chances to 4477 // fill a buffer, then remove it from active list. 4478 // Only consider last track started for mixer state control 4479 if (--(track->mRetryCount) <= 0) { 4480 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4481 tracksToRemove->add(track); 4482 // indicate to client process that the track was disabled because of underrun; 4483 // it will then automatically call start() when data is available 4484 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4485 } else if (last) { 4486 mixerStatus = MIXER_TRACKS_ENABLED; 4487 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4488 doHwPause = true; 4489 mHwPaused = true; 4490 } 4491 } 4492 } 4493 } 4494 } 4495 4496 // if an active track did not command a flush, check for pending flush on stopped tracks 4497 if (!flushPending) { 4498 for (size_t i = 0; i < mTracks.size(); i++) { 4499 if (mTracks[i]->isFlushPending()) { 4500 mTracks[i]->flushAck(); 4501 flushPending = true; 4502 } 4503 } 4504 } 4505 4506 // make sure the pause/flush/resume sequence is executed in the right order. 4507 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4508 // before flush and then resume HW. This can happen in case of pause/flush/resume 4509 // if resume is received before pause is executed. 4510 if (mHwSupportsPause && !mStandby && 4511 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4512 mOutput->stream->pause(mOutput->stream); 4513 } 4514 if (flushPending) { 4515 flushHw_l(); 4516 } 4517 if (mHwSupportsPause && !mStandby && doHwResume) { 4518 mOutput->stream->resume(mOutput->stream); 4519 } 4520 // remove all the tracks that need to be... 4521 removeTracks_l(*tracksToRemove); 4522 4523 return mixerStatus; 4524} 4525 4526void AudioFlinger::DirectOutputThread::threadLoop_mix() 4527{ 4528 size_t frameCount = mFrameCount; 4529 int8_t *curBuf = (int8_t *)mSinkBuffer; 4530 // output audio to hardware 4531 while (frameCount) { 4532 AudioBufferProvider::Buffer buffer; 4533 buffer.frameCount = frameCount; 4534 status_t status = mActiveTrack->getNextBuffer(&buffer); 4535 if (status != NO_ERROR || buffer.raw == NULL) { 4536 memset(curBuf, 0, frameCount * mFrameSize); 4537 break; 4538 } 4539 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4540 frameCount -= buffer.frameCount; 4541 curBuf += buffer.frameCount * mFrameSize; 4542 mActiveTrack->releaseBuffer(&buffer); 4543 } 4544 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4545 sleepTime = 0; 4546 standbyTime = systemTime() + standbyDelay; 4547 mActiveTrack.clear(); 4548} 4549 4550void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4551{ 4552 // do not write to HAL when paused 4553 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4554 sleepTime = idleSleepTime; 4555 return; 4556 } 4557 if (sleepTime == 0) { 4558 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4559 sleepTime = activeSleepTime; 4560 } else { 4561 sleepTime = idleSleepTime; 4562 } 4563 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4564 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4565 sleepTime = 0; 4566 } 4567} 4568 4569void AudioFlinger::DirectOutputThread::threadLoop_exit() 4570{ 4571 { 4572 Mutex::Autolock _l(mLock); 4573 bool flushPending = false; 4574 for (size_t i = 0; i < mTracks.size(); i++) { 4575 if (mTracks[i]->isFlushPending()) { 4576 mTracks[i]->flushAck(); 4577 flushPending = true; 4578 } 4579 } 4580 if (flushPending) { 4581 flushHw_l(); 4582 } 4583 } 4584 PlaybackThread::threadLoop_exit(); 4585} 4586 4587// must be called with thread mutex locked 4588bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4589{ 4590 bool trackPaused = false; 4591 bool trackStopped = false; 4592 4593 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4594 // after a timeout and we will enter standby then. 4595 if (mTracks.size() > 0) { 4596 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4597 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4598 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4599 } 4600 4601 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped)); 4602} 4603 4604// getTrackName_l() must be called with ThreadBase::mLock held 4605int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4606 audio_format_t format __unused, int sessionId __unused) 4607{ 4608 return 0; 4609} 4610 4611// deleteTrackName_l() must be called with ThreadBase::mLock held 4612void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4613{ 4614} 4615 4616// checkForNewParameter_l() must be called with ThreadBase::mLock held 4617bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4618 status_t& status) 4619{ 4620 bool reconfig = false; 4621 4622 status = NO_ERROR; 4623 4624 AudioParameter param = AudioParameter(keyValuePair); 4625 int value; 4626 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4627 // forward device change to effects that have requested to be 4628 // aware of attached audio device. 4629 if (value != AUDIO_DEVICE_NONE) { 4630 mOutDevice = value; 4631 for (size_t i = 0; i < mEffectChains.size(); i++) { 4632 mEffectChains[i]->setDevice_l(mOutDevice); 4633 } 4634 } 4635 } 4636 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4637 // do not accept frame count changes if tracks are open as the track buffer 4638 // size depends on frame count and correct behavior would not be garantied 4639 // if frame count is changed after track creation 4640 if (!mTracks.isEmpty()) { 4641 status = INVALID_OPERATION; 4642 } else { 4643 reconfig = true; 4644 } 4645 } 4646 if (status == NO_ERROR) { 4647 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4648 keyValuePair.string()); 4649 if (!mStandby && status == INVALID_OPERATION) { 4650 mOutput->standby(); 4651 mStandby = true; 4652 mBytesWritten = 0; 4653 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4654 keyValuePair.string()); 4655 } 4656 if (status == NO_ERROR && reconfig) { 4657 readOutputParameters_l(); 4658 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4659 } 4660 } 4661 4662 return reconfig; 4663} 4664 4665uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4666{ 4667 uint32_t time; 4668 if (audio_is_linear_pcm(mFormat)) { 4669 time = PlaybackThread::activeSleepTimeUs(); 4670 } else { 4671 time = 10000; 4672 } 4673 return time; 4674} 4675 4676uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4677{ 4678 uint32_t time; 4679 if (audio_is_linear_pcm(mFormat)) { 4680 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4681 } else { 4682 time = 10000; 4683 } 4684 return time; 4685} 4686 4687uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4688{ 4689 uint32_t time; 4690 if (audio_is_linear_pcm(mFormat)) { 4691 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4692 } else { 4693 time = 10000; 4694 } 4695 return time; 4696} 4697 4698void AudioFlinger::DirectOutputThread::cacheParameters_l() 4699{ 4700 PlaybackThread::cacheParameters_l(); 4701 4702 // use shorter standby delay as on normal output to release 4703 // hardware resources as soon as possible 4704 // no delay on outputs with HW A/V sync 4705 if (usesHwAvSync()) { 4706 standbyDelay = 0; 4707 } else if (audio_is_linear_pcm(mFormat)) { 4708 standbyDelay = microseconds(activeSleepTime*2); 4709 } else { 4710 standbyDelay = kOffloadStandbyDelayNs; 4711 } 4712} 4713 4714void AudioFlinger::DirectOutputThread::flushHw_l() 4715{ 4716 mOutput->flush(); 4717 mHwPaused = false; 4718} 4719 4720// ---------------------------------------------------------------------------- 4721 4722AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4723 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4724 : Thread(false /*canCallJava*/), 4725 mPlaybackThread(playbackThread), 4726 mWriteAckSequence(0), 4727 mDrainSequence(0) 4728{ 4729} 4730 4731AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4732{ 4733} 4734 4735void AudioFlinger::AsyncCallbackThread::onFirstRef() 4736{ 4737 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4738} 4739 4740bool AudioFlinger::AsyncCallbackThread::threadLoop() 4741{ 4742 while (!exitPending()) { 4743 uint32_t writeAckSequence; 4744 uint32_t drainSequence; 4745 4746 { 4747 Mutex::Autolock _l(mLock); 4748 while (!((mWriteAckSequence & 1) || 4749 (mDrainSequence & 1) || 4750 exitPending())) { 4751 mWaitWorkCV.wait(mLock); 4752 } 4753 4754 if (exitPending()) { 4755 break; 4756 } 4757 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4758 mWriteAckSequence, mDrainSequence); 4759 writeAckSequence = mWriteAckSequence; 4760 mWriteAckSequence &= ~1; 4761 drainSequence = mDrainSequence; 4762 mDrainSequence &= ~1; 4763 } 4764 { 4765 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4766 if (playbackThread != 0) { 4767 if (writeAckSequence & 1) { 4768 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4769 } 4770 if (drainSequence & 1) { 4771 playbackThread->resetDraining(drainSequence >> 1); 4772 } 4773 } 4774 } 4775 } 4776 return false; 4777} 4778 4779void AudioFlinger::AsyncCallbackThread::exit() 4780{ 4781 ALOGV("AsyncCallbackThread::exit"); 4782 Mutex::Autolock _l(mLock); 4783 requestExit(); 4784 mWaitWorkCV.broadcast(); 4785} 4786 4787void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4788{ 4789 Mutex::Autolock _l(mLock); 4790 // bit 0 is cleared 4791 mWriteAckSequence = sequence << 1; 4792} 4793 4794void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4795{ 4796 Mutex::Autolock _l(mLock); 4797 // ignore unexpected callbacks 4798 if (mWriteAckSequence & 2) { 4799 mWriteAckSequence |= 1; 4800 mWaitWorkCV.signal(); 4801 } 4802} 4803 4804void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4805{ 4806 Mutex::Autolock _l(mLock); 4807 // bit 0 is cleared 4808 mDrainSequence = sequence << 1; 4809} 4810 4811void AudioFlinger::AsyncCallbackThread::resetDraining() 4812{ 4813 Mutex::Autolock _l(mLock); 4814 // ignore unexpected callbacks 4815 if (mDrainSequence & 2) { 4816 mDrainSequence |= 1; 4817 mWaitWorkCV.signal(); 4818 } 4819} 4820 4821 4822// ---------------------------------------------------------------------------- 4823AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4824 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4825 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4826 mPausedBytesRemaining(0) 4827{ 4828 //FIXME: mStandby should be set to true by ThreadBase constructor 4829 mStandby = true; 4830} 4831 4832void AudioFlinger::OffloadThread::threadLoop_exit() 4833{ 4834 if (mFlushPending || mHwPaused) { 4835 // If a flush is pending or track was paused, just discard buffered data 4836 flushHw_l(); 4837 } else { 4838 mMixerStatus = MIXER_DRAIN_ALL; 4839 threadLoop_drain(); 4840 } 4841 if (mUseAsyncWrite) { 4842 ALOG_ASSERT(mCallbackThread != 0); 4843 mCallbackThread->exit(); 4844 } 4845 PlaybackThread::threadLoop_exit(); 4846} 4847 4848AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4849 Vector< sp<Track> > *tracksToRemove 4850) 4851{ 4852 size_t count = mActiveTracks.size(); 4853 4854 mixer_state mixerStatus = MIXER_IDLE; 4855 bool doHwPause = false; 4856 bool doHwResume = false; 4857 4858 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4859 4860 // find out which tracks need to be processed 4861 for (size_t i = 0; i < count; i++) { 4862 sp<Track> t = mActiveTracks[i].promote(); 4863 // The track died recently 4864 if (t == 0) { 4865 continue; 4866 } 4867 Track* const track = t.get(); 4868 audio_track_cblk_t* cblk = track->cblk(); 4869 // Only consider last track started for volume and mixer state control. 4870 // In theory an older track could underrun and restart after the new one starts 4871 // but as we only care about the transition phase between two tracks on a 4872 // direct output, it is not a problem to ignore the underrun case. 4873 sp<Track> l = mLatestActiveTrack.promote(); 4874 bool last = l.get() == track; 4875 4876 if (track->isInvalid()) { 4877 ALOGW("An invalidated track shouldn't be in active list"); 4878 tracksToRemove->add(track); 4879 continue; 4880 } 4881 4882 if (track->mState == TrackBase::IDLE) { 4883 ALOGW("An idle track shouldn't be in active list"); 4884 continue; 4885 } 4886 4887 if (track->isPausing()) { 4888 track->setPaused(); 4889 if (last) { 4890 if (!mHwPaused) { 4891 doHwPause = true; 4892 mHwPaused = true; 4893 } 4894 // If we were part way through writing the mixbuffer to 4895 // the HAL we must save this until we resume 4896 // BUG - this will be wrong if a different track is made active, 4897 // in that case we want to discard the pending data in the 4898 // mixbuffer and tell the client to present it again when the 4899 // track is resumed 4900 mPausedWriteLength = mCurrentWriteLength; 4901 mPausedBytesRemaining = mBytesRemaining; 4902 mBytesRemaining = 0; // stop writing 4903 } 4904 tracksToRemove->add(track); 4905 } else if (track->isFlushPending()) { 4906 track->flushAck(); 4907 if (last) { 4908 mFlushPending = true; 4909 } 4910 } else if (track->isResumePending()){ 4911 track->resumeAck(); 4912 if (last) { 4913 if (mPausedBytesRemaining) { 4914 // Need to continue write that was interrupted 4915 mCurrentWriteLength = mPausedWriteLength; 4916 mBytesRemaining = mPausedBytesRemaining; 4917 mPausedBytesRemaining = 0; 4918 } 4919 if (mHwPaused) { 4920 doHwResume = true; 4921 mHwPaused = false; 4922 // threadLoop_mix() will handle the case that we need to 4923 // resume an interrupted write 4924 } 4925 // enable write to audio HAL 4926 sleepTime = 0; 4927 4928 // Do not handle new data in this iteration even if track->framesReady() 4929 mixerStatus = MIXER_TRACKS_ENABLED; 4930 } 4931 } else if (track->framesReady() && track->isReady() && 4932 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4933 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4934 if (track->mFillingUpStatus == Track::FS_FILLED) { 4935 track->mFillingUpStatus = Track::FS_ACTIVE; 4936 // make sure processVolume_l() will apply new volume even if 0 4937 mLeftVolFloat = mRightVolFloat = -1.0; 4938 } 4939 4940 if (last) { 4941 sp<Track> previousTrack = mPreviousTrack.promote(); 4942 if (previousTrack != 0) { 4943 if (track != previousTrack.get()) { 4944 // Flush any data still being written from last track 4945 mBytesRemaining = 0; 4946 if (mPausedBytesRemaining) { 4947 // Last track was paused so we also need to flush saved 4948 // mixbuffer state and invalidate track so that it will 4949 // re-submit that unwritten data when it is next resumed 4950 mPausedBytesRemaining = 0; 4951 // Invalidate is a bit drastic - would be more efficient 4952 // to have a flag to tell client that some of the 4953 // previously written data was lost 4954 previousTrack->invalidate(); 4955 } 4956 // flush data already sent to the DSP if changing audio session as audio 4957 // comes from a different source. Also invalidate previous track to force a 4958 // seek when resuming. 4959 if (previousTrack->sessionId() != track->sessionId()) { 4960 previousTrack->invalidate(); 4961 } 4962 } 4963 } 4964 mPreviousTrack = track; 4965 // reset retry count 4966 track->mRetryCount = kMaxTrackRetriesOffload; 4967 mActiveTrack = t; 4968 mixerStatus = MIXER_TRACKS_READY; 4969 } 4970 } else { 4971 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4972 if (track->isStopping_1()) { 4973 // Hardware buffer can hold a large amount of audio so we must 4974 // wait for all current track's data to drain before we say 4975 // that the track is stopped. 4976 if (mBytesRemaining == 0) { 4977 // Only start draining when all data in mixbuffer 4978 // has been written 4979 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4980 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4981 // do not drain if no data was ever sent to HAL (mStandby == true) 4982 if (last && !mStandby) { 4983 // do not modify drain sequence if we are already draining. This happens 4984 // when resuming from pause after drain. 4985 if ((mDrainSequence & 1) == 0) { 4986 sleepTime = 0; 4987 standbyTime = systemTime() + standbyDelay; 4988 mixerStatus = MIXER_DRAIN_TRACK; 4989 mDrainSequence += 2; 4990 } 4991 if (mHwPaused) { 4992 // It is possible to move from PAUSED to STOPPING_1 without 4993 // a resume so we must ensure hardware is running 4994 doHwResume = true; 4995 mHwPaused = false; 4996 } 4997 } 4998 } 4999 } else if (track->isStopping_2()) { 5000 // Drain has completed or we are in standby, signal presentation complete 5001 if (!(mDrainSequence & 1) || !last || mStandby) { 5002 track->mState = TrackBase::STOPPED; 5003 size_t audioHALFrames = 5004 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5005 size_t framesWritten = 5006 mBytesWritten / mOutput->getFrameSize(); 5007 track->presentationComplete(framesWritten, audioHALFrames); 5008 track->reset(); 5009 tracksToRemove->add(track); 5010 } 5011 } else { 5012 // No buffers for this track. Give it a few chances to 5013 // fill a buffer, then remove it from active list. 5014 if (--(track->mRetryCount) <= 0) { 5015 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5016 track->name()); 5017 tracksToRemove->add(track); 5018 // indicate to client process that the track was disabled because of underrun; 5019 // it will then automatically call start() when data is available 5020 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5021 } else if (last){ 5022 mixerStatus = MIXER_TRACKS_ENABLED; 5023 } 5024 } 5025 } 5026 // compute volume for this track 5027 processVolume_l(track, last); 5028 } 5029 5030 // make sure the pause/flush/resume sequence is executed in the right order. 5031 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5032 // before flush and then resume HW. This can happen in case of pause/flush/resume 5033 // if resume is received before pause is executed. 5034 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5035 mOutput->stream->pause(mOutput->stream); 5036 } 5037 if (mFlushPending) { 5038 flushHw_l(); 5039 mFlushPending = false; 5040 } 5041 if (!mStandby && doHwResume) { 5042 mOutput->stream->resume(mOutput->stream); 5043 } 5044 5045 // remove all the tracks that need to be... 5046 removeTracks_l(*tracksToRemove); 5047 5048 return mixerStatus; 5049} 5050 5051// must be called with thread mutex locked 5052bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5053{ 5054 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5055 mWriteAckSequence, mDrainSequence); 5056 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5057 return true; 5058 } 5059 return false; 5060} 5061 5062bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5063{ 5064 Mutex::Autolock _l(mLock); 5065 return waitingAsyncCallback_l(); 5066} 5067 5068void AudioFlinger::OffloadThread::flushHw_l() 5069{ 5070 DirectOutputThread::flushHw_l(); 5071 // Flush anything still waiting in the mixbuffer 5072 mCurrentWriteLength = 0; 5073 mBytesRemaining = 0; 5074 mPausedWriteLength = 0; 5075 mPausedBytesRemaining = 0; 5076 5077 if (mUseAsyncWrite) { 5078 // discard any pending drain or write ack by incrementing sequence 5079 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5080 mDrainSequence = (mDrainSequence + 2) & ~1; 5081 ALOG_ASSERT(mCallbackThread != 0); 5082 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5083 mCallbackThread->setDraining(mDrainSequence); 5084 } 5085} 5086 5087void AudioFlinger::OffloadThread::onAddNewTrack_l() 5088{ 5089 sp<Track> previousTrack = mPreviousTrack.promote(); 5090 sp<Track> latestTrack = mLatestActiveTrack.promote(); 5091 5092 if (previousTrack != 0 && latestTrack != 0 && 5093 (previousTrack->sessionId() != latestTrack->sessionId())) { 5094 mFlushPending = true; 5095 } 5096 PlaybackThread::onAddNewTrack_l(); 5097} 5098 5099// ---------------------------------------------------------------------------- 5100 5101AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5102 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 5103 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5104 DUPLICATING), 5105 mWaitTimeMs(UINT_MAX) 5106{ 5107 addOutputTrack(mainThread); 5108} 5109 5110AudioFlinger::DuplicatingThread::~DuplicatingThread() 5111{ 5112 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5113 mOutputTracks[i]->destroy(); 5114 } 5115} 5116 5117void AudioFlinger::DuplicatingThread::threadLoop_mix() 5118{ 5119 // mix buffers... 5120 if (outputsReady(outputTracks)) { 5121 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5122 } else { 5123 if (mMixerBufferValid) { 5124 memset(mMixerBuffer, 0, mMixerBufferSize); 5125 } else { 5126 memset(mSinkBuffer, 0, mSinkBufferSize); 5127 } 5128 } 5129 sleepTime = 0; 5130 writeFrames = mNormalFrameCount; 5131 mCurrentWriteLength = mSinkBufferSize; 5132 standbyTime = systemTime() + standbyDelay; 5133} 5134 5135void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5136{ 5137 if (sleepTime == 0) { 5138 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5139 sleepTime = activeSleepTime; 5140 } else { 5141 sleepTime = idleSleepTime; 5142 } 5143 } else if (mBytesWritten != 0) { 5144 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5145 writeFrames = mNormalFrameCount; 5146 memset(mSinkBuffer, 0, mSinkBufferSize); 5147 } else { 5148 // flush remaining overflow buffers in output tracks 5149 writeFrames = 0; 5150 } 5151 sleepTime = 0; 5152 } 5153} 5154 5155ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5156{ 5157 for (size_t i = 0; i < outputTracks.size(); i++) { 5158 outputTracks[i]->write(mSinkBuffer, writeFrames); 5159 } 5160 mStandby = false; 5161 return (ssize_t)mSinkBufferSize; 5162} 5163 5164void AudioFlinger::DuplicatingThread::threadLoop_standby() 5165{ 5166 // DuplicatingThread implements standby by stopping all tracks 5167 for (size_t i = 0; i < outputTracks.size(); i++) { 5168 outputTracks[i]->stop(); 5169 } 5170} 5171 5172void AudioFlinger::DuplicatingThread::saveOutputTracks() 5173{ 5174 outputTracks = mOutputTracks; 5175} 5176 5177void AudioFlinger::DuplicatingThread::clearOutputTracks() 5178{ 5179 outputTracks.clear(); 5180} 5181 5182void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5183{ 5184 Mutex::Autolock _l(mLock); 5185 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5186 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5187 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5188 const size_t frameCount = 5189 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5190 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5191 // from different OutputTracks and their associated MixerThreads (e.g. one may 5192 // nearly empty and the other may be dropping data). 5193 5194 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5195 this, 5196 mSampleRate, 5197 mFormat, 5198 mChannelMask, 5199 frameCount, 5200 IPCThreadState::self()->getCallingUid()); 5201 if (outputTrack->cblk() != NULL) { 5202 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5203 mOutputTracks.add(outputTrack); 5204 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5205 updateWaitTime_l(); 5206 } 5207} 5208 5209void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5210{ 5211 Mutex::Autolock _l(mLock); 5212 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5213 if (mOutputTracks[i]->thread() == thread) { 5214 mOutputTracks[i]->destroy(); 5215 mOutputTracks.removeAt(i); 5216 updateWaitTime_l(); 5217 return; 5218 } 5219 } 5220 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5221} 5222 5223// caller must hold mLock 5224void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5225{ 5226 mWaitTimeMs = UINT_MAX; 5227 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5228 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5229 if (strong != 0) { 5230 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5231 if (waitTimeMs < mWaitTimeMs) { 5232 mWaitTimeMs = waitTimeMs; 5233 } 5234 } 5235 } 5236} 5237 5238 5239bool AudioFlinger::DuplicatingThread::outputsReady( 5240 const SortedVector< sp<OutputTrack> > &outputTracks) 5241{ 5242 for (size_t i = 0; i < outputTracks.size(); i++) { 5243 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5244 if (thread == 0) { 5245 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5246 outputTracks[i].get()); 5247 return false; 5248 } 5249 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5250 // see note at standby() declaration 5251 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5252 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5253 thread.get()); 5254 return false; 5255 } 5256 } 5257 return true; 5258} 5259 5260uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5261{ 5262 return (mWaitTimeMs * 1000) / 2; 5263} 5264 5265void AudioFlinger::DuplicatingThread::cacheParameters_l() 5266{ 5267 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5268 updateWaitTime_l(); 5269 5270 MixerThread::cacheParameters_l(); 5271} 5272 5273// ---------------------------------------------------------------------------- 5274// Record 5275// ---------------------------------------------------------------------------- 5276 5277AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5278 AudioStreamIn *input, 5279 audio_io_handle_t id, 5280 audio_devices_t outDevice, 5281 audio_devices_t inDevice 5282#ifdef TEE_SINK 5283 , const sp<NBAIO_Sink>& teeSink 5284#endif 5285 ) : 5286 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5287 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5288 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5289 mRsmpInRear(0) 5290#ifdef TEE_SINK 5291 , mTeeSink(teeSink) 5292#endif 5293 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5294 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5295 // mFastCapture below 5296 , mFastCaptureFutex(0) 5297 // mInputSource 5298 // mPipeSink 5299 // mPipeSource 5300 , mPipeFramesP2(0) 5301 // mPipeMemory 5302 // mFastCaptureNBLogWriter 5303 , mFastTrackAvail(false) 5304{ 5305 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5306 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5307 5308 readInputParameters_l(); 5309 5310 // create an NBAIO source for the HAL input stream, and negotiate 5311 mInputSource = new AudioStreamInSource(input->stream); 5312 size_t numCounterOffers = 0; 5313 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5314 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5315 ALOG_ASSERT(index == 0); 5316 5317 // initialize fast capture depending on configuration 5318 bool initFastCapture; 5319 switch (kUseFastCapture) { 5320 case FastCapture_Never: 5321 initFastCapture = false; 5322 break; 5323 case FastCapture_Always: 5324 initFastCapture = true; 5325 break; 5326 case FastCapture_Static: 5327 uint32_t primaryOutputSampleRate; 5328 { 5329 AutoMutex _l(audioFlinger->mHardwareLock); 5330 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5331 } 5332 initFastCapture = 5333 // either capture sample rate is same as (a reasonable) primary output sample rate 5334 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5335 (mSampleRate == primaryOutputSampleRate)) || 5336 // or primary output sample rate is unknown, and capture sample rate is reasonable 5337 ((primaryOutputSampleRate == 0) && 5338 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5339 // and the buffer size is < 12 ms 5340 (mFrameCount * 1000) / mSampleRate < 12; 5341 break; 5342 // case FastCapture_Dynamic: 5343 } 5344 5345 if (initFastCapture) { 5346 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5347 NBAIO_Format format = mInputSource->format(); 5348 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5349 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5350 void *pipeBuffer; 5351 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5352 sp<IMemory> pipeMemory; 5353 if ((roHeap == 0) || 5354 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5355 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5356 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5357 goto failed; 5358 } 5359 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5360 memset(pipeBuffer, 0, pipeSize); 5361 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5362 const NBAIO_Format offers[1] = {format}; 5363 size_t numCounterOffers = 0; 5364 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5365 ALOG_ASSERT(index == 0); 5366 mPipeSink = pipe; 5367 PipeReader *pipeReader = new PipeReader(*pipe); 5368 numCounterOffers = 0; 5369 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5370 ALOG_ASSERT(index == 0); 5371 mPipeSource = pipeReader; 5372 mPipeFramesP2 = pipeFramesP2; 5373 mPipeMemory = pipeMemory; 5374 5375 // create fast capture 5376 mFastCapture = new FastCapture(); 5377 FastCaptureStateQueue *sq = mFastCapture->sq(); 5378#ifdef STATE_QUEUE_DUMP 5379 // FIXME 5380#endif 5381 FastCaptureState *state = sq->begin(); 5382 state->mCblk = NULL; 5383 state->mInputSource = mInputSource.get(); 5384 state->mInputSourceGen++; 5385 state->mPipeSink = pipe; 5386 state->mPipeSinkGen++; 5387 state->mFrameCount = mFrameCount; 5388 state->mCommand = FastCaptureState::COLD_IDLE; 5389 // already done in constructor initialization list 5390 //mFastCaptureFutex = 0; 5391 state->mColdFutexAddr = &mFastCaptureFutex; 5392 state->mColdGen++; 5393 state->mDumpState = &mFastCaptureDumpState; 5394#ifdef TEE_SINK 5395 // FIXME 5396#endif 5397 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5398 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5399 sq->end(); 5400 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5401 5402 // start the fast capture 5403 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5404 pid_t tid = mFastCapture->getTid(); 5405 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5406 if (err != 0) { 5407 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5408 kPriorityFastCapture, getpid_cached, tid, err); 5409 } 5410 5411#ifdef AUDIO_WATCHDOG 5412 // FIXME 5413#endif 5414 5415 mFastTrackAvail = true; 5416 } 5417failed: ; 5418 5419 // FIXME mNormalSource 5420} 5421 5422AudioFlinger::RecordThread::~RecordThread() 5423{ 5424 if (mFastCapture != 0) { 5425 FastCaptureStateQueue *sq = mFastCapture->sq(); 5426 FastCaptureState *state = sq->begin(); 5427 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5428 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5429 if (old == -1) { 5430 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5431 } 5432 } 5433 state->mCommand = FastCaptureState::EXIT; 5434 sq->end(); 5435 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5436 mFastCapture->join(); 5437 mFastCapture.clear(); 5438 } 5439 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5440 mAudioFlinger->unregisterWriter(mNBLogWriter); 5441 free(mRsmpInBuffer); 5442} 5443 5444void AudioFlinger::RecordThread::onFirstRef() 5445{ 5446 run(mThreadName, PRIORITY_URGENT_AUDIO); 5447} 5448 5449bool AudioFlinger::RecordThread::threadLoop() 5450{ 5451 nsecs_t lastWarning = 0; 5452 5453 inputStandBy(); 5454 5455reacquire_wakelock: 5456 sp<RecordTrack> activeTrack; 5457 int activeTracksGen; 5458 { 5459 Mutex::Autolock _l(mLock); 5460 size_t size = mActiveTracks.size(); 5461 activeTracksGen = mActiveTracksGen; 5462 if (size > 0) { 5463 // FIXME an arbitrary choice 5464 activeTrack = mActiveTracks[0]; 5465 acquireWakeLock_l(activeTrack->uid()); 5466 if (size > 1) { 5467 SortedVector<int> tmp; 5468 for (size_t i = 0; i < size; i++) { 5469 tmp.add(mActiveTracks[i]->uid()); 5470 } 5471 updateWakeLockUids_l(tmp); 5472 } 5473 } else { 5474 acquireWakeLock_l(-1); 5475 } 5476 } 5477 5478 // used to request a deferred sleep, to be executed later while mutex is unlocked 5479 uint32_t sleepUs = 0; 5480 5481 // loop while there is work to do 5482 for (;;) { 5483 Vector< sp<EffectChain> > effectChains; 5484 5485 // sleep with mutex unlocked 5486 if (sleepUs > 0) { 5487 ATRACE_BEGIN("sleep"); 5488 usleep(sleepUs); 5489 ATRACE_END(); 5490 sleepUs = 0; 5491 } 5492 5493 // activeTracks accumulates a copy of a subset of mActiveTracks 5494 Vector< sp<RecordTrack> > activeTracks; 5495 5496 // reference to the (first and only) active fast track 5497 sp<RecordTrack> fastTrack; 5498 5499 // reference to a fast track which is about to be removed 5500 sp<RecordTrack> fastTrackToRemove; 5501 5502 { // scope for mLock 5503 Mutex::Autolock _l(mLock); 5504 5505 processConfigEvents_l(); 5506 5507 // check exitPending here because checkForNewParameters_l() and 5508 // checkForNewParameters_l() can temporarily release mLock 5509 if (exitPending()) { 5510 break; 5511 } 5512 5513 // if no active track(s), then standby and release wakelock 5514 size_t size = mActiveTracks.size(); 5515 if (size == 0) { 5516 standbyIfNotAlreadyInStandby(); 5517 // exitPending() can't become true here 5518 releaseWakeLock_l(); 5519 ALOGV("RecordThread: loop stopping"); 5520 // go to sleep 5521 mWaitWorkCV.wait(mLock); 5522 ALOGV("RecordThread: loop starting"); 5523 goto reacquire_wakelock; 5524 } 5525 5526 if (mActiveTracksGen != activeTracksGen) { 5527 activeTracksGen = mActiveTracksGen; 5528 SortedVector<int> tmp; 5529 for (size_t i = 0; i < size; i++) { 5530 tmp.add(mActiveTracks[i]->uid()); 5531 } 5532 updateWakeLockUids_l(tmp); 5533 } 5534 5535 bool doBroadcast = false; 5536 for (size_t i = 0; i < size; ) { 5537 5538 activeTrack = mActiveTracks[i]; 5539 if (activeTrack->isTerminated()) { 5540 if (activeTrack->isFastTrack()) { 5541 ALOG_ASSERT(fastTrackToRemove == 0); 5542 fastTrackToRemove = activeTrack; 5543 } 5544 removeTrack_l(activeTrack); 5545 mActiveTracks.remove(activeTrack); 5546 mActiveTracksGen++; 5547 size--; 5548 continue; 5549 } 5550 5551 TrackBase::track_state activeTrackState = activeTrack->mState; 5552 switch (activeTrackState) { 5553 5554 case TrackBase::PAUSING: 5555 mActiveTracks.remove(activeTrack); 5556 mActiveTracksGen++; 5557 doBroadcast = true; 5558 size--; 5559 continue; 5560 5561 case TrackBase::STARTING_1: 5562 sleepUs = 10000; 5563 i++; 5564 continue; 5565 5566 case TrackBase::STARTING_2: 5567 doBroadcast = true; 5568 mStandby = false; 5569 activeTrack->mState = TrackBase::ACTIVE; 5570 break; 5571 5572 case TrackBase::ACTIVE: 5573 break; 5574 5575 case TrackBase::IDLE: 5576 i++; 5577 continue; 5578 5579 default: 5580 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5581 } 5582 5583 activeTracks.add(activeTrack); 5584 i++; 5585 5586 if (activeTrack->isFastTrack()) { 5587 ALOG_ASSERT(!mFastTrackAvail); 5588 ALOG_ASSERT(fastTrack == 0); 5589 fastTrack = activeTrack; 5590 } 5591 } 5592 if (doBroadcast) { 5593 mStartStopCond.broadcast(); 5594 } 5595 5596 // sleep if there are no active tracks to process 5597 if (activeTracks.size() == 0) { 5598 if (sleepUs == 0) { 5599 sleepUs = kRecordThreadSleepUs; 5600 } 5601 continue; 5602 } 5603 sleepUs = 0; 5604 5605 lockEffectChains_l(effectChains); 5606 } 5607 5608 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5609 5610 size_t size = effectChains.size(); 5611 for (size_t i = 0; i < size; i++) { 5612 // thread mutex is not locked, but effect chain is locked 5613 effectChains[i]->process_l(); 5614 } 5615 5616 // Push a new fast capture state if fast capture is not already running, or cblk change 5617 if (mFastCapture != 0) { 5618 FastCaptureStateQueue *sq = mFastCapture->sq(); 5619 FastCaptureState *state = sq->begin(); 5620 bool didModify = false; 5621 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5622 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5623 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5624 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5625 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5626 if (old == -1) { 5627 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5628 } 5629 } 5630 state->mCommand = FastCaptureState::READ_WRITE; 5631#if 0 // FIXME 5632 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5633 FastThreadDumpState::kSamplingNforLowRamDevice : 5634 FastThreadDumpState::kSamplingN); 5635#endif 5636 didModify = true; 5637 } 5638 audio_track_cblk_t *cblkOld = state->mCblk; 5639 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5640 if (cblkNew != cblkOld) { 5641 state->mCblk = cblkNew; 5642 // block until acked if removing a fast track 5643 if (cblkOld != NULL) { 5644 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5645 } 5646 didModify = true; 5647 } 5648 sq->end(didModify); 5649 if (didModify) { 5650 sq->push(block); 5651#if 0 5652 if (kUseFastCapture == FastCapture_Dynamic) { 5653 mNormalSource = mPipeSource; 5654 } 5655#endif 5656 } 5657 } 5658 5659 // now run the fast track destructor with thread mutex unlocked 5660 fastTrackToRemove.clear(); 5661 5662 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5663 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5664 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5665 // If destination is non-contiguous, first read past the nominal end of buffer, then 5666 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5667 5668 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5669 ssize_t framesRead; 5670 5671 // If an NBAIO source is present, use it to read the normal capture's data 5672 if (mPipeSource != 0) { 5673 size_t framesToRead = mBufferSize / mFrameSize; 5674 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5675 framesToRead, AudioBufferProvider::kInvalidPTS); 5676 if (framesRead == 0) { 5677 // since pipe is non-blocking, simulate blocking input 5678 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5679 } 5680 // otherwise use the HAL / AudioStreamIn directly 5681 } else { 5682 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5683 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5684 if (bytesRead < 0) { 5685 framesRead = bytesRead; 5686 } else { 5687 framesRead = bytesRead / mFrameSize; 5688 } 5689 } 5690 5691 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5692 ALOGE("read failed: framesRead=%d", framesRead); 5693 // Force input into standby so that it tries to recover at next read attempt 5694 inputStandBy(); 5695 sleepUs = kRecordThreadSleepUs; 5696 } 5697 if (framesRead <= 0) { 5698 goto unlock; 5699 } 5700 ALOG_ASSERT(framesRead > 0); 5701 5702 if (mTeeSink != 0) { 5703 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5704 } 5705 // If destination is non-contiguous, we now correct for reading past end of buffer. 5706 { 5707 size_t part1 = mRsmpInFramesP2 - rear; 5708 if ((size_t) framesRead > part1) { 5709 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5710 (framesRead - part1) * mFrameSize); 5711 } 5712 } 5713 rear = mRsmpInRear += framesRead; 5714 5715 size = activeTracks.size(); 5716 // loop over each active track 5717 for (size_t i = 0; i < size; i++) { 5718 activeTrack = activeTracks[i]; 5719 5720 // skip fast tracks, as those are handled directly by FastCapture 5721 if (activeTrack->isFastTrack()) { 5722 continue; 5723 } 5724 5725 // TODO: This code probably should be moved to RecordTrack. 5726 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5727 5728 enum { 5729 OVERRUN_UNKNOWN, 5730 OVERRUN_TRUE, 5731 OVERRUN_FALSE 5732 } overrun = OVERRUN_UNKNOWN; 5733 5734 // loop over getNextBuffer to handle circular sink 5735 for (;;) { 5736 5737 activeTrack->mSink.frameCount = ~0; 5738 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5739 size_t framesOut = activeTrack->mSink.frameCount; 5740 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5741 5742 // check available frames and handle overrun conditions 5743 // if the record track isn't draining fast enough. 5744 bool hasOverrun; 5745 size_t framesIn; 5746 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5747 if (hasOverrun) { 5748 overrun = OVERRUN_TRUE; 5749 } 5750 if (framesOut == 0 || framesIn == 0) { 5751 break; 5752 } 5753 5754 // Don't allow framesOut to be larger than what is possible with resampling 5755 // from framesIn. 5756 // This isn't strictly necessary but helps limit buffer resizing in 5757 // RecordBufferConverter. TODO: remove when no longer needed. 5758 framesOut = min(framesOut, 5759 destinationFramesPossible( 5760 framesIn, mSampleRate, activeTrack->mSampleRate)); 5761 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5762 framesOut = activeTrack->mRecordBufferConverter->convert( 5763 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5764 5765 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5766 overrun = OVERRUN_FALSE; 5767 } 5768 5769 if (activeTrack->mFramesToDrop == 0) { 5770 if (framesOut > 0) { 5771 activeTrack->mSink.frameCount = framesOut; 5772 activeTrack->releaseBuffer(&activeTrack->mSink); 5773 } 5774 } else { 5775 // FIXME could do a partial drop of framesOut 5776 if (activeTrack->mFramesToDrop > 0) { 5777 activeTrack->mFramesToDrop -= framesOut; 5778 if (activeTrack->mFramesToDrop <= 0) { 5779 activeTrack->clearSyncStartEvent(); 5780 } 5781 } else { 5782 activeTrack->mFramesToDrop += framesOut; 5783 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5784 activeTrack->mSyncStartEvent->isCancelled()) { 5785 ALOGW("Synced record %s, session %d, trigger session %d", 5786 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5787 activeTrack->sessionId(), 5788 (activeTrack->mSyncStartEvent != 0) ? 5789 activeTrack->mSyncStartEvent->triggerSession() : 0); 5790 activeTrack->clearSyncStartEvent(); 5791 } 5792 } 5793 } 5794 5795 if (framesOut == 0) { 5796 break; 5797 } 5798 } 5799 5800 switch (overrun) { 5801 case OVERRUN_TRUE: 5802 // client isn't retrieving buffers fast enough 5803 if (!activeTrack->setOverflow()) { 5804 nsecs_t now = systemTime(); 5805 // FIXME should lastWarning per track? 5806 if ((now - lastWarning) > kWarningThrottleNs) { 5807 ALOGW("RecordThread: buffer overflow"); 5808 lastWarning = now; 5809 } 5810 } 5811 break; 5812 case OVERRUN_FALSE: 5813 activeTrack->clearOverflow(); 5814 break; 5815 case OVERRUN_UNKNOWN: 5816 break; 5817 } 5818 5819 } 5820 5821unlock: 5822 // enable changes in effect chain 5823 unlockEffectChains(effectChains); 5824 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5825 } 5826 5827 standbyIfNotAlreadyInStandby(); 5828 5829 { 5830 Mutex::Autolock _l(mLock); 5831 for (size_t i = 0; i < mTracks.size(); i++) { 5832 sp<RecordTrack> track = mTracks[i]; 5833 track->invalidate(); 5834 } 5835 mActiveTracks.clear(); 5836 mActiveTracksGen++; 5837 mStartStopCond.broadcast(); 5838 } 5839 5840 releaseWakeLock(); 5841 5842 ALOGV("RecordThread %p exiting", this); 5843 return false; 5844} 5845 5846void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5847{ 5848 if (!mStandby) { 5849 inputStandBy(); 5850 mStandby = true; 5851 } 5852} 5853 5854void AudioFlinger::RecordThread::inputStandBy() 5855{ 5856 // Idle the fast capture if it's currently running 5857 if (mFastCapture != 0) { 5858 FastCaptureStateQueue *sq = mFastCapture->sq(); 5859 FastCaptureState *state = sq->begin(); 5860 if (!(state->mCommand & FastCaptureState::IDLE)) { 5861 state->mCommand = FastCaptureState::COLD_IDLE; 5862 state->mColdFutexAddr = &mFastCaptureFutex; 5863 state->mColdGen++; 5864 mFastCaptureFutex = 0; 5865 sq->end(); 5866 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5867 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5868#if 0 5869 if (kUseFastCapture == FastCapture_Dynamic) { 5870 // FIXME 5871 } 5872#endif 5873#ifdef AUDIO_WATCHDOG 5874 // FIXME 5875#endif 5876 } else { 5877 sq->end(false /*didModify*/); 5878 } 5879 } 5880 mInput->stream->common.standby(&mInput->stream->common); 5881} 5882 5883// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5884sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5885 const sp<AudioFlinger::Client>& client, 5886 uint32_t sampleRate, 5887 audio_format_t format, 5888 audio_channel_mask_t channelMask, 5889 size_t *pFrameCount, 5890 int sessionId, 5891 size_t *notificationFrames, 5892 int uid, 5893 IAudioFlinger::track_flags_t *flags, 5894 pid_t tid, 5895 status_t *status) 5896{ 5897 size_t frameCount = *pFrameCount; 5898 sp<RecordTrack> track; 5899 status_t lStatus; 5900 5901 // client expresses a preference for FAST, but we get the final say 5902 if (*flags & IAudioFlinger::TRACK_FAST) { 5903 if ( 5904 // we formerly checked for a callback handler (non-0 tid), 5905 // but that is no longer required for TRANSFER_OBTAIN mode 5906 // 5907 // frame count is not specified, or is exactly the pipe depth 5908 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5909 // PCM data 5910 audio_is_linear_pcm(format) && 5911 // native format 5912 (format == mFormat) && 5913 // native channel mask 5914 (channelMask == mChannelMask) && 5915 // native hardware sample rate 5916 (sampleRate == mSampleRate) && 5917 // record thread has an associated fast capture 5918 hasFastCapture() && 5919 // there are sufficient fast track slots available 5920 mFastTrackAvail 5921 ) { 5922 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5923 frameCount, mFrameCount); 5924 } else { 5925 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5926 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5927 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5928 frameCount, mFrameCount, mPipeFramesP2, 5929 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5930 hasFastCapture(), tid, mFastTrackAvail); 5931 *flags &= ~IAudioFlinger::TRACK_FAST; 5932 } 5933 } 5934 5935 // compute track buffer size in frames, and suggest the notification frame count 5936 if (*flags & IAudioFlinger::TRACK_FAST) { 5937 // fast track: frame count is exactly the pipe depth 5938 frameCount = mPipeFramesP2; 5939 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5940 *notificationFrames = mFrameCount; 5941 } else { 5942 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5943 // or 20 ms if there is a fast capture 5944 // TODO This could be a roundupRatio inline, and const 5945 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5946 * sampleRate + mSampleRate - 1) / mSampleRate; 5947 // minimum number of notification periods is at least kMinNotifications, 5948 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5949 static const size_t kMinNotifications = 3; 5950 static const uint32_t kMinMs = 30; 5951 // TODO This could be a roundupRatio inline 5952 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5953 // TODO This could be a roundupRatio inline 5954 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5955 maxNotificationFrames; 5956 const size_t minFrameCount = maxNotificationFrames * 5957 max(kMinNotifications, minNotificationsByMs); 5958 frameCount = max(frameCount, minFrameCount); 5959 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5960 *notificationFrames = maxNotificationFrames; 5961 } 5962 } 5963 *pFrameCount = frameCount; 5964 5965 lStatus = initCheck(); 5966 if (lStatus != NO_ERROR) { 5967 ALOGE("createRecordTrack_l() audio driver not initialized"); 5968 goto Exit; 5969 } 5970 5971 { // scope for mLock 5972 Mutex::Autolock _l(mLock); 5973 5974 track = new RecordTrack(this, client, sampleRate, 5975 format, channelMask, frameCount, NULL, sessionId, uid, 5976 *flags, TrackBase::TYPE_DEFAULT); 5977 5978 lStatus = track->initCheck(); 5979 if (lStatus != NO_ERROR) { 5980 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5981 // track must be cleared from the caller as the caller has the AF lock 5982 goto Exit; 5983 } 5984 mTracks.add(track); 5985 5986 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5987 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5988 mAudioFlinger->btNrecIsOff(); 5989 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5990 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5991 5992 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5993 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5994 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5995 // so ask activity manager to do this on our behalf 5996 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5997 } 5998 } 5999 6000 lStatus = NO_ERROR; 6001 6002Exit: 6003 *status = lStatus; 6004 return track; 6005} 6006 6007status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6008 AudioSystem::sync_event_t event, 6009 int triggerSession) 6010{ 6011 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6012 sp<ThreadBase> strongMe = this; 6013 status_t status = NO_ERROR; 6014 6015 if (event == AudioSystem::SYNC_EVENT_NONE) { 6016 recordTrack->clearSyncStartEvent(); 6017 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6018 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6019 triggerSession, 6020 recordTrack->sessionId(), 6021 syncStartEventCallback, 6022 recordTrack); 6023 // Sync event can be cancelled by the trigger session if the track is not in a 6024 // compatible state in which case we start record immediately 6025 if (recordTrack->mSyncStartEvent->isCancelled()) { 6026 recordTrack->clearSyncStartEvent(); 6027 } else { 6028 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6029 recordTrack->mFramesToDrop = - 6030 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6031 } 6032 } 6033 6034 { 6035 // This section is a rendezvous between binder thread executing start() and RecordThread 6036 AutoMutex lock(mLock); 6037 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6038 if (recordTrack->mState == TrackBase::PAUSING) { 6039 ALOGV("active record track PAUSING -> ACTIVE"); 6040 recordTrack->mState = TrackBase::ACTIVE; 6041 } else { 6042 ALOGV("active record track state %d", recordTrack->mState); 6043 } 6044 return status; 6045 } 6046 6047 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6048 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6049 // or using a separate command thread 6050 recordTrack->mState = TrackBase::STARTING_1; 6051 mActiveTracks.add(recordTrack); 6052 mActiveTracksGen++; 6053 status_t status = NO_ERROR; 6054 if (recordTrack->isExternalTrack()) { 6055 mLock.unlock(); 6056 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6057 mLock.lock(); 6058 // FIXME should verify that recordTrack is still in mActiveTracks 6059 if (status != NO_ERROR) { 6060 mActiveTracks.remove(recordTrack); 6061 mActiveTracksGen++; 6062 recordTrack->clearSyncStartEvent(); 6063 ALOGV("RecordThread::start error %d", status); 6064 return status; 6065 } 6066 } 6067 // Catch up with current buffer indices if thread is already running. 6068 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6069 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6070 // see previously buffered data before it called start(), but with greater risk of overrun. 6071 6072 recordTrack->mResamplerBufferProvider->reset(); 6073 // clear any converter state as new data will be discontinuous 6074 recordTrack->mRecordBufferConverter->reset(); 6075 recordTrack->mState = TrackBase::STARTING_2; 6076 // signal thread to start 6077 mWaitWorkCV.broadcast(); 6078 if (mActiveTracks.indexOf(recordTrack) < 0) { 6079 ALOGV("Record failed to start"); 6080 status = BAD_VALUE; 6081 goto startError; 6082 } 6083 return status; 6084 } 6085 6086startError: 6087 if (recordTrack->isExternalTrack()) { 6088 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6089 } 6090 recordTrack->clearSyncStartEvent(); 6091 // FIXME I wonder why we do not reset the state here? 6092 return status; 6093} 6094 6095void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6096{ 6097 sp<SyncEvent> strongEvent = event.promote(); 6098 6099 if (strongEvent != 0) { 6100 sp<RefBase> ptr = strongEvent->cookie().promote(); 6101 if (ptr != 0) { 6102 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6103 recordTrack->handleSyncStartEvent(strongEvent); 6104 } 6105 } 6106} 6107 6108bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6109 ALOGV("RecordThread::stop"); 6110 AutoMutex _l(mLock); 6111 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6112 return false; 6113 } 6114 // note that threadLoop may still be processing the track at this point [without lock] 6115 recordTrack->mState = TrackBase::PAUSING; 6116 // do not wait for mStartStopCond if exiting 6117 if (exitPending()) { 6118 return true; 6119 } 6120 // FIXME incorrect usage of wait: no explicit predicate or loop 6121 mStartStopCond.wait(mLock); 6122 // if we have been restarted, recordTrack is in mActiveTracks here 6123 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6124 ALOGV("Record stopped OK"); 6125 return true; 6126 } 6127 return false; 6128} 6129 6130bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6131{ 6132 return false; 6133} 6134 6135status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6136{ 6137#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6138 if (!isValidSyncEvent(event)) { 6139 return BAD_VALUE; 6140 } 6141 6142 int eventSession = event->triggerSession(); 6143 status_t ret = NAME_NOT_FOUND; 6144 6145 Mutex::Autolock _l(mLock); 6146 6147 for (size_t i = 0; i < mTracks.size(); i++) { 6148 sp<RecordTrack> track = mTracks[i]; 6149 if (eventSession == track->sessionId()) { 6150 (void) track->setSyncEvent(event); 6151 ret = NO_ERROR; 6152 } 6153 } 6154 return ret; 6155#else 6156 return BAD_VALUE; 6157#endif 6158} 6159 6160// destroyTrack_l() must be called with ThreadBase::mLock held 6161void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6162{ 6163 track->terminate(); 6164 track->mState = TrackBase::STOPPED; 6165 // active tracks are removed by threadLoop() 6166 if (mActiveTracks.indexOf(track) < 0) { 6167 removeTrack_l(track); 6168 } 6169} 6170 6171void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6172{ 6173 mTracks.remove(track); 6174 // need anything related to effects here? 6175 if (track->isFastTrack()) { 6176 ALOG_ASSERT(!mFastTrackAvail); 6177 mFastTrackAvail = true; 6178 } 6179} 6180 6181void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6182{ 6183 dumpInternals(fd, args); 6184 dumpTracks(fd, args); 6185 dumpEffectChains(fd, args); 6186} 6187 6188void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6189{ 6190 dprintf(fd, "\nInput thread %p:\n", this); 6191 6192 dumpBase(fd, args); 6193 6194 if (mActiveTracks.size() == 0) { 6195 dprintf(fd, " No active record clients\n"); 6196 } 6197 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6198 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6199 6200 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6201 const FastCaptureDumpState copy(mFastCaptureDumpState); 6202 copy.dump(fd); 6203} 6204 6205void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6206{ 6207 const size_t SIZE = 256; 6208 char buffer[SIZE]; 6209 String8 result; 6210 6211 size_t numtracks = mTracks.size(); 6212 size_t numactive = mActiveTracks.size(); 6213 size_t numactiveseen = 0; 6214 dprintf(fd, " %d Tracks", numtracks); 6215 if (numtracks) { 6216 dprintf(fd, " of which %d are active\n", numactive); 6217 RecordTrack::appendDumpHeader(result); 6218 for (size_t i = 0; i < numtracks ; ++i) { 6219 sp<RecordTrack> track = mTracks[i]; 6220 if (track != 0) { 6221 bool active = mActiveTracks.indexOf(track) >= 0; 6222 if (active) { 6223 numactiveseen++; 6224 } 6225 track->dump(buffer, SIZE, active); 6226 result.append(buffer); 6227 } 6228 } 6229 } else { 6230 dprintf(fd, "\n"); 6231 } 6232 6233 if (numactiveseen != numactive) { 6234 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6235 " not in the track list\n"); 6236 result.append(buffer); 6237 RecordTrack::appendDumpHeader(result); 6238 for (size_t i = 0; i < numactive; ++i) { 6239 sp<RecordTrack> track = mActiveTracks[i]; 6240 if (mTracks.indexOf(track) < 0) { 6241 track->dump(buffer, SIZE, true); 6242 result.append(buffer); 6243 } 6244 } 6245 6246 } 6247 write(fd, result.string(), result.size()); 6248} 6249 6250 6251void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6252{ 6253 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6254 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6255 mRsmpInFront = recordThread->mRsmpInRear; 6256 mRsmpInUnrel = 0; 6257} 6258 6259void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6260 size_t *framesAvailable, bool *hasOverrun) 6261{ 6262 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6263 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6264 const int32_t rear = recordThread->mRsmpInRear; 6265 const int32_t front = mRsmpInFront; 6266 const ssize_t filled = rear - front; 6267 6268 size_t framesIn; 6269 bool overrun = false; 6270 if (filled < 0) { 6271 // should not happen, but treat like a massive overrun and re-sync 6272 framesIn = 0; 6273 mRsmpInFront = rear; 6274 overrun = true; 6275 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6276 framesIn = (size_t) filled; 6277 } else { 6278 // client is not keeping up with server, but give it latest data 6279 framesIn = recordThread->mRsmpInFrames; 6280 mRsmpInFront = /* front = */ rear - framesIn; 6281 overrun = true; 6282 } 6283 if (framesAvailable != NULL) { 6284 *framesAvailable = framesIn; 6285 } 6286 if (hasOverrun != NULL) { 6287 *hasOverrun = overrun; 6288 } 6289} 6290 6291// AudioBufferProvider interface 6292status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6293 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6294{ 6295 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6296 if (threadBase == 0) { 6297 buffer->frameCount = 0; 6298 buffer->raw = NULL; 6299 return NOT_ENOUGH_DATA; 6300 } 6301 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6302 int32_t rear = recordThread->mRsmpInRear; 6303 int32_t front = mRsmpInFront; 6304 ssize_t filled = rear - front; 6305 // FIXME should not be P2 (don't want to increase latency) 6306 // FIXME if client not keeping up, discard 6307 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6308 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6309 front &= recordThread->mRsmpInFramesP2 - 1; 6310 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6311 if (part1 > (size_t) filled) { 6312 part1 = filled; 6313 } 6314 size_t ask = buffer->frameCount; 6315 ALOG_ASSERT(ask > 0); 6316 if (part1 > ask) { 6317 part1 = ask; 6318 } 6319 if (part1 == 0) { 6320 // out of data is fine since the resampler will return a short-count. 6321 buffer->raw = NULL; 6322 buffer->frameCount = 0; 6323 mRsmpInUnrel = 0; 6324 return NOT_ENOUGH_DATA; 6325 } 6326 6327 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6328 buffer->frameCount = part1; 6329 mRsmpInUnrel = part1; 6330 return NO_ERROR; 6331} 6332 6333// AudioBufferProvider interface 6334void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6335 AudioBufferProvider::Buffer* buffer) 6336{ 6337 size_t stepCount = buffer->frameCount; 6338 if (stepCount == 0) { 6339 return; 6340 } 6341 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6342 mRsmpInUnrel -= stepCount; 6343 mRsmpInFront += stepCount; 6344 buffer->raw = NULL; 6345 buffer->frameCount = 0; 6346} 6347 6348AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6349 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6350 uint32_t srcSampleRate, 6351 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6352 uint32_t dstSampleRate) : 6353 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6354 // mSrcFormat 6355 // mSrcSampleRate 6356 // mDstChannelMask 6357 // mDstFormat 6358 // mDstSampleRate 6359 // mSrcChannelCount 6360 // mDstChannelCount 6361 // mDstFrameSize 6362 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6363 mResampler(NULL), 6364 mIsLegacyDownmix(false), 6365 mIsLegacyUpmix(false), 6366 mRequiresFloat(false), 6367 mInputConverterProvider(NULL) 6368{ 6369 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6370 dstChannelMask, dstFormat, dstSampleRate); 6371} 6372 6373AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6374 free(mBuf); 6375 delete mResampler; 6376 delete mInputConverterProvider; 6377} 6378 6379size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6380 AudioBufferProvider *provider, size_t frames) 6381{ 6382 if (mInputConverterProvider != NULL) { 6383 mInputConverterProvider->setBufferProvider(provider); 6384 provider = mInputConverterProvider; 6385 } 6386 6387 if (mResampler == NULL) { 6388 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6389 mSrcSampleRate, mSrcFormat, mDstFormat); 6390 6391 AudioBufferProvider::Buffer buffer; 6392 for (size_t i = frames; i > 0; ) { 6393 buffer.frameCount = i; 6394 status_t status = provider->getNextBuffer(&buffer, 0); 6395 if (status != OK || buffer.frameCount == 0) { 6396 frames -= i; // cannot fill request. 6397 break; 6398 } 6399 // format convert to destination buffer 6400 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6401 6402 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6403 i -= buffer.frameCount; 6404 provider->releaseBuffer(&buffer); 6405 } 6406 } else { 6407 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6408 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6409 6410 // reallocate buffer if needed 6411 if (mBufFrameSize != 0 && mBufFrames < frames) { 6412 free(mBuf); 6413 mBufFrames = frames; 6414 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6415 } 6416 // resampler accumulates, but we only have one source track 6417 memset(mBuf, 0, frames * mBufFrameSize); 6418 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6419 // format convert to destination buffer 6420 convertResampler(dst, mBuf, frames); 6421 } 6422 return frames; 6423} 6424 6425status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6426 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6427 uint32_t srcSampleRate, 6428 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6429 uint32_t dstSampleRate) 6430{ 6431 // quick evaluation if there is any change. 6432 if (mSrcFormat == srcFormat 6433 && mSrcChannelMask == srcChannelMask 6434 && mSrcSampleRate == srcSampleRate 6435 && mDstFormat == dstFormat 6436 && mDstChannelMask == dstChannelMask 6437 && mDstSampleRate == dstSampleRate) { 6438 return NO_ERROR; 6439 } 6440 6441 const bool valid = 6442 audio_is_input_channel(srcChannelMask) 6443 && audio_is_input_channel(dstChannelMask) 6444 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6445 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6446 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6447 ; // no upsampling checks for now 6448 if (!valid) { 6449 return BAD_VALUE; 6450 } 6451 6452 mSrcFormat = srcFormat; 6453 mSrcChannelMask = srcChannelMask; 6454 mSrcSampleRate = srcSampleRate; 6455 mDstFormat = dstFormat; 6456 mDstChannelMask = dstChannelMask; 6457 mDstSampleRate = dstSampleRate; 6458 6459 // compute derived parameters 6460 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6461 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6462 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6463 6464 // do we need to resample? 6465 delete mResampler; 6466 mResampler = NULL; 6467 if (mSrcSampleRate != mDstSampleRate) { 6468 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6469 mSrcChannelCount, mDstSampleRate); 6470 mResampler->setSampleRate(mSrcSampleRate); 6471 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6472 } 6473 6474 // are we running legacy channel conversion modes? 6475 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6476 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6477 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6478 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6479 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6480 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6481 6482 // do we need to process in float? 6483 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6484 6485 // do we need a staging buffer to convert for destination (we can still optimize this)? 6486 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6487 if (mResampler != NULL) { 6488 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6489 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6490 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6491 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6492 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6493 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6494 } else { 6495 mBufFrameSize = 0; 6496 } 6497 mBufFrames = 0; // force the buffer to be resized. 6498 6499 // do we need an input converter buffer provider to give us float? 6500 delete mInputConverterProvider; 6501 mInputConverterProvider = NULL; 6502 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6503 mInputConverterProvider = new ReformatBufferProvider( 6504 audio_channel_count_from_in_mask(mSrcChannelMask), 6505 mSrcFormat, 6506 AUDIO_FORMAT_PCM_FLOAT, 6507 256 /* provider buffer frame count */); 6508 } 6509 6510 // do we need a remixer to do channel mask conversion 6511 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6512 (void) memcpy_by_index_array_initialization_from_channel_mask( 6513 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6514 } 6515 return NO_ERROR; 6516} 6517 6518void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6519 void *dst, const void *src, size_t frames) 6520{ 6521 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6522 if (mBufFrameSize != 0 && mBufFrames < frames) { 6523 free(mBuf); 6524 mBufFrames = frames; 6525 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6526 } 6527 // do we need to do legacy upmix and downmix? 6528 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6529 void *dstBuf = mBuf != NULL ? mBuf : dst; 6530 if (mIsLegacyUpmix) { 6531 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6532 (const float *)src, frames); 6533 } else /*mIsLegacyDownmix */ { 6534 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6535 (const float *)src, frames); 6536 } 6537 if (mBuf != NULL) { 6538 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6539 frames * mDstChannelCount); 6540 } 6541 return; 6542 } 6543 // do we need to do channel mask conversion? 6544 if (mSrcChannelMask != mDstChannelMask) { 6545 void *dstBuf = mBuf != NULL ? mBuf : dst; 6546 memcpy_by_index_array(dstBuf, mDstChannelCount, 6547 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6548 if (dstBuf == dst) { 6549 return; // format is the same 6550 } 6551 } 6552 // convert to destination buffer 6553 const void *convertBuf = mBuf != NULL ? mBuf : src; 6554 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6555 frames * mDstChannelCount); 6556} 6557 6558void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6559 void *dst, /*not-a-const*/ void *src, size_t frames) 6560{ 6561 // src buffer format is ALWAYS float when entering this routine 6562 if (mIsLegacyUpmix) { 6563 ; // mono to stereo already handled by resampler 6564 } else if (mIsLegacyDownmix 6565 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6566 // the resampler outputs stereo for mono input channel (a feature?) 6567 // must convert to mono 6568 downmix_to_mono_float_from_stereo_float((float *)src, 6569 (const float *)src, frames); 6570 } else if (mSrcChannelMask != mDstChannelMask) { 6571 // convert to mono channel again for channel mask conversion (could be skipped 6572 // with further optimization). 6573 if (mSrcChannelCount == 1) { 6574 downmix_to_mono_float_from_stereo_float((float *)src, 6575 (const float *)src, frames); 6576 } 6577 // convert to destination format (in place, OK as float is larger than other types) 6578 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6579 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6580 frames * mSrcChannelCount); 6581 } 6582 // channel convert and save to dst 6583 memcpy_by_index_array(dst, mDstChannelCount, 6584 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6585 return; 6586 } 6587 // convert to destination format and save to dst 6588 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6589 frames * mDstChannelCount); 6590} 6591 6592bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6593 status_t& status) 6594{ 6595 bool reconfig = false; 6596 6597 status = NO_ERROR; 6598 6599 audio_format_t reqFormat = mFormat; 6600 uint32_t samplingRate = mSampleRate; 6601 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6602 // possible that we are > 2 channels, use channel index mask 6603 if (channelMask == AUDIO_CHANNEL_INVALID && mChannelCount <= FCC_8) { 6604 audio_channel_mask_for_index_assignment_from_count(mChannelCount); 6605 } 6606 6607 AudioParameter param = AudioParameter(keyValuePair); 6608 int value; 6609 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6610 // channel count change can be requested. Do we mandate the first client defines the 6611 // HAL sampling rate and channel count or do we allow changes on the fly? 6612 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6613 samplingRate = value; 6614 reconfig = true; 6615 } 6616 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6617 if (!audio_is_linear_pcm((audio_format_t) value)) { 6618 status = BAD_VALUE; 6619 } else { 6620 reqFormat = (audio_format_t) value; 6621 reconfig = true; 6622 } 6623 } 6624 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6625 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6626 if (!audio_is_input_channel(mask) || 6627 audio_channel_count_from_in_mask(mask) > FCC_8) { 6628 status = BAD_VALUE; 6629 } else { 6630 channelMask = mask; 6631 reconfig = true; 6632 } 6633 } 6634 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6635 // do not accept frame count changes if tracks are open as the track buffer 6636 // size depends on frame count and correct behavior would not be guaranteed 6637 // if frame count is changed after track creation 6638 if (mActiveTracks.size() > 0) { 6639 status = INVALID_OPERATION; 6640 } else { 6641 reconfig = true; 6642 } 6643 } 6644 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6645 // forward device change to effects that have requested to be 6646 // aware of attached audio device. 6647 for (size_t i = 0; i < mEffectChains.size(); i++) { 6648 mEffectChains[i]->setDevice_l(value); 6649 } 6650 6651 // store input device and output device but do not forward output device to audio HAL. 6652 // Note that status is ignored by the caller for output device 6653 // (see AudioFlinger::setParameters() 6654 if (audio_is_output_devices(value)) { 6655 mOutDevice = value; 6656 status = BAD_VALUE; 6657 } else { 6658 mInDevice = value; 6659 // disable AEC and NS if the device is a BT SCO headset supporting those 6660 // pre processings 6661 if (mTracks.size() > 0) { 6662 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6663 mAudioFlinger->btNrecIsOff(); 6664 for (size_t i = 0; i < mTracks.size(); i++) { 6665 sp<RecordTrack> track = mTracks[i]; 6666 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6667 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6668 } 6669 } 6670 } 6671 } 6672 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6673 mAudioSource != (audio_source_t)value) { 6674 // forward device change to effects that have requested to be 6675 // aware of attached audio device. 6676 for (size_t i = 0; i < mEffectChains.size(); i++) { 6677 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6678 } 6679 mAudioSource = (audio_source_t)value; 6680 } 6681 6682 if (status == NO_ERROR) { 6683 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6684 keyValuePair.string()); 6685 if (status == INVALID_OPERATION) { 6686 inputStandBy(); 6687 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6688 keyValuePair.string()); 6689 } 6690 if (reconfig) { 6691 if (status == BAD_VALUE && 6692 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6693 audio_is_linear_pcm(reqFormat) && 6694 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6695 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6696 audio_channel_count_from_in_mask( 6697 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6698 (channelMask == AUDIO_CHANNEL_IN_MONO || 6699 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6700 status = NO_ERROR; 6701 } 6702 if (status == NO_ERROR) { 6703 readInputParameters_l(); 6704 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6705 } 6706 } 6707 } 6708 6709 return reconfig; 6710} 6711 6712String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6713{ 6714 Mutex::Autolock _l(mLock); 6715 if (initCheck() != NO_ERROR) { 6716 return String8(); 6717 } 6718 6719 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6720 const String8 out_s8(s); 6721 free(s); 6722 return out_s8; 6723} 6724 6725void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6726 AudioSystem::OutputDescriptor desc; 6727 const void *param2 = NULL; 6728 6729 switch (event) { 6730 case AudioSystem::INPUT_OPENED: 6731 case AudioSystem::INPUT_CONFIG_CHANGED: 6732 desc.channelMask = mChannelMask; 6733 desc.samplingRate = mSampleRate; 6734 desc.format = mFormat; 6735 desc.frameCount = mFrameCount; 6736 desc.latency = 0; 6737 param2 = &desc; 6738 break; 6739 6740 case AudioSystem::INPUT_CLOSED: 6741 default: 6742 break; 6743 } 6744 mAudioFlinger->audioConfigChanged(event, mId, param2); 6745} 6746 6747void AudioFlinger::RecordThread::readInputParameters_l() 6748{ 6749 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6750 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6751 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6752 if (mChannelCount > FCC_8) { 6753 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6754 } 6755 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6756 mFormat = mHALFormat; 6757 if (!audio_is_linear_pcm(mFormat)) { 6758 ALOGE("HAL format %#x is not linear pcm", mFormat); 6759 } 6760 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6761 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6762 mFrameCount = mBufferSize / mFrameSize; 6763 // This is the formula for calculating the temporary buffer size. 6764 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6765 // 1 full output buffer, regardless of the alignment of the available input. 6766 // The value is somewhat arbitrary, and could probably be even larger. 6767 // A larger value should allow more old data to be read after a track calls start(), 6768 // without increasing latency. 6769 // 6770 // Note this is independent of the maximum downsampling ratio permitted for capture. 6771 mRsmpInFrames = mFrameCount * 7; 6772 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6773 free(mRsmpInBuffer); 6774 6775 // TODO optimize audio capture buffer sizes ... 6776 // Here we calculate the size of the sliding buffer used as a source 6777 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6778 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6779 // be better to have it derived from the pipe depth in the long term. 6780 // The current value is higher than necessary. However it should not add to latency. 6781 6782 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6783 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6784 6785 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6786 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6787} 6788 6789uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6790{ 6791 Mutex::Autolock _l(mLock); 6792 if (initCheck() != NO_ERROR) { 6793 return 0; 6794 } 6795 6796 return mInput->stream->get_input_frames_lost(mInput->stream); 6797} 6798 6799uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6800{ 6801 Mutex::Autolock _l(mLock); 6802 uint32_t result = 0; 6803 if (getEffectChain_l(sessionId) != 0) { 6804 result = EFFECT_SESSION; 6805 } 6806 6807 for (size_t i = 0; i < mTracks.size(); ++i) { 6808 if (sessionId == mTracks[i]->sessionId()) { 6809 result |= TRACK_SESSION; 6810 break; 6811 } 6812 } 6813 6814 return result; 6815} 6816 6817KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6818{ 6819 KeyedVector<int, bool> ids; 6820 Mutex::Autolock _l(mLock); 6821 for (size_t j = 0; j < mTracks.size(); ++j) { 6822 sp<RecordThread::RecordTrack> track = mTracks[j]; 6823 int sessionId = track->sessionId(); 6824 if (ids.indexOfKey(sessionId) < 0) { 6825 ids.add(sessionId, true); 6826 } 6827 } 6828 return ids; 6829} 6830 6831AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6832{ 6833 Mutex::Autolock _l(mLock); 6834 AudioStreamIn *input = mInput; 6835 mInput = NULL; 6836 return input; 6837} 6838 6839// this method must always be called either with ThreadBase mLock held or inside the thread loop 6840audio_stream_t* AudioFlinger::RecordThread::stream() const 6841{ 6842 if (mInput == NULL) { 6843 return NULL; 6844 } 6845 return &mInput->stream->common; 6846} 6847 6848status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6849{ 6850 // only one chain per input thread 6851 if (mEffectChains.size() != 0) { 6852 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6853 return INVALID_OPERATION; 6854 } 6855 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6856 chain->setThread(this); 6857 chain->setInBuffer(NULL); 6858 chain->setOutBuffer(NULL); 6859 6860 checkSuspendOnAddEffectChain_l(chain); 6861 6862 // make sure enabled pre processing effects state is communicated to the HAL as we 6863 // just moved them to a new input stream. 6864 chain->syncHalEffectsState(); 6865 6866 mEffectChains.add(chain); 6867 6868 return NO_ERROR; 6869} 6870 6871size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6872{ 6873 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6874 ALOGW_IF(mEffectChains.size() != 1, 6875 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6876 chain.get(), mEffectChains.size(), this); 6877 if (mEffectChains.size() == 1) { 6878 mEffectChains.removeAt(0); 6879 } 6880 return 0; 6881} 6882 6883status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6884 audio_patch_handle_t *handle) 6885{ 6886 status_t status = NO_ERROR; 6887 6888 // store new device and send to effects 6889 mInDevice = patch->sources[0].ext.device.type; 6890 for (size_t i = 0; i < mEffectChains.size(); i++) { 6891 mEffectChains[i]->setDevice_l(mInDevice); 6892 } 6893 6894 // disable AEC and NS if the device is a BT SCO headset supporting those 6895 // pre processings 6896 if (mTracks.size() > 0) { 6897 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6898 mAudioFlinger->btNrecIsOff(); 6899 for (size_t i = 0; i < mTracks.size(); i++) { 6900 sp<RecordTrack> track = mTracks[i]; 6901 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6902 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6903 } 6904 } 6905 6906 // store new source and send to effects 6907 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6908 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6909 for (size_t i = 0; i < mEffectChains.size(); i++) { 6910 mEffectChains[i]->setAudioSource_l(mAudioSource); 6911 } 6912 } 6913 6914 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6915 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6916 status = hwDevice->create_audio_patch(hwDevice, 6917 patch->num_sources, 6918 patch->sources, 6919 patch->num_sinks, 6920 patch->sinks, 6921 handle); 6922 } else { 6923 char *address; 6924 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 6925 address = audio_device_address_to_parameter( 6926 patch->sources[0].ext.device.type, 6927 patch->sources[0].ext.device.address); 6928 } else { 6929 address = (char *)calloc(1, 1); 6930 } 6931 AudioParameter param = AudioParameter(String8(address)); 6932 free(address); 6933 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 6934 (int)patch->sources[0].ext.device.type); 6935 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 6936 (int)patch->sinks[0].ext.mix.usecase.source); 6937 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6938 param.toString().string()); 6939 *handle = AUDIO_PATCH_HANDLE_NONE; 6940 } 6941 6942 return status; 6943} 6944 6945status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6946{ 6947 status_t status = NO_ERROR; 6948 6949 mInDevice = AUDIO_DEVICE_NONE; 6950 6951 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6952 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6953 status = hwDevice->release_audio_patch(hwDevice, handle); 6954 } else { 6955 AudioParameter param; 6956 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 6957 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6958 param.toString().string()); 6959 } 6960 return status; 6961} 6962 6963void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6964{ 6965 Mutex::Autolock _l(mLock); 6966 mTracks.add(record); 6967} 6968 6969void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6970{ 6971 Mutex::Autolock _l(mLock); 6972 destroyTrack_l(record); 6973} 6974 6975void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6976{ 6977 ThreadBase::getAudioPortConfig(config); 6978 config->role = AUDIO_PORT_ROLE_SINK; 6979 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6980 config->ext.mix.usecase.source = mAudioSource; 6981} 6982 6983} // namespace android 6984