Threads.cpp revision 8edb8dc44b8a2f81bdb5db645b6b708548771a31
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/primitives.h> 40#include <audio_utils/format.h> 41#include <audio_utils/minifloat.h> 42 43// NBAIO implementations 44#include <media/nbaio/AudioStreamInSource.h> 45#include <media/nbaio/AudioStreamOutSink.h> 46#include <media/nbaio/MonoPipe.h> 47#include <media/nbaio/MonoPipeReader.h> 48#include <media/nbaio/Pipe.h> 49#include <media/nbaio/PipeReader.h> 50#include <media/nbaio/SourceAudioBufferProvider.h> 51 52#include <powermanager/PowerManager.h> 53 54#include <common_time/cc_helper.h> 55#include <common_time/local_clock.h> 56 57#include "AudioFlinger.h" 58#include "AudioMixer.h" 59#include "FastMixer.h" 60#include "FastCapture.h" 61#include "ServiceUtilities.h" 62#include "SchedulingPolicyService.h" 63 64#ifdef ADD_BATTERY_DATA 65#include <media/IMediaPlayerService.h> 66#include <media/IMediaDeathNotifier.h> 67#endif 68 69#ifdef DEBUG_CPU_USAGE 70#include <cpustats/CentralTendencyStatistics.h> 71#include <cpustats/ThreadCpuUsage.h> 72#endif 73 74// ---------------------------------------------------------------------------- 75 76// Note: the following macro is used for extremely verbose logging message. In 77// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80// turned on. Do not uncomment the #def below unless you really know what you 81// are doing and want to see all of the extremely verbose messages. 82//#define VERY_VERY_VERBOSE_LOGGING 83#ifdef VERY_VERY_VERBOSE_LOGGING 84#define ALOGVV ALOGV 85#else 86#define ALOGVV(a...) do { } while(0) 87#endif 88 89// TODO: Move these macro/inlines to a header file. 90#define max(a, b) ((a) > (b) ? (a) : (b)) 91template <typename T> 92static inline T min(const T& a, const T& b) 93{ 94 return a < b ? a : b; 95} 96 97namespace android { 98 99// retry counts for buffer fill timeout 100// 50 * ~20msecs = 1 second 101static const int8_t kMaxTrackRetries = 50; 102static const int8_t kMaxTrackStartupRetries = 50; 103// allow less retry attempts on direct output thread. 104// direct outputs can be a scarce resource in audio hardware and should 105// be released as quickly as possible. 106static const int8_t kMaxTrackRetriesDirect = 2; 107 108// don't warn about blocked writes or record buffer overflows more often than this 109static const nsecs_t kWarningThrottleNs = seconds(5); 110 111// RecordThread loop sleep time upon application overrun or audio HAL read error 112static const int kRecordThreadSleepUs = 5000; 113 114// maximum time to wait in sendConfigEvent_l() for a status to be received 115static const nsecs_t kConfigEventTimeoutNs = seconds(2); 116 117// minimum sleep time for the mixer thread loop when tracks are active but in underrun 118static const uint32_t kMinThreadSleepTimeUs = 5000; 119// maximum divider applied to the active sleep time in the mixer thread loop 120static const uint32_t kMaxThreadSleepTimeShift = 2; 121 122// minimum normal sink buffer size, expressed in milliseconds rather than frames 123static const uint32_t kMinNormalSinkBufferSizeMs = 20; 124// maximum normal sink buffer size 125static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 126 127// Offloaded output thread standby delay: allows track transition without going to standby 128static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 129 130// Whether to use fast mixer 131static const enum { 132 FastMixer_Never, // never initialize or use: for debugging only 133 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 134 // normal mixer multiplier is 1 135 FastMixer_Static, // initialize if needed, then use all the time if initialized, 136 // multiplier is calculated based on min & max normal mixer buffer size 137 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 138 // multiplier is calculated based on min & max normal mixer buffer size 139 // FIXME for FastMixer_Dynamic: 140 // Supporting this option will require fixing HALs that can't handle large writes. 141 // For example, one HAL implementation returns an error from a large write, 142 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 143 // We could either fix the HAL implementations, or provide a wrapper that breaks 144 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 145} kUseFastMixer = FastMixer_Static; 146 147// Whether to use fast capture 148static const enum { 149 FastCapture_Never, // never initialize or use: for debugging only 150 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 151 FastCapture_Static, // initialize if needed, then use all the time if initialized 152} kUseFastCapture = FastCapture_Static; 153 154// Priorities for requestPriority 155static const int kPriorityAudioApp = 2; 156static const int kPriorityFastMixer = 3; 157static const int kPriorityFastCapture = 3; 158 159// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 160// for the track. The client then sub-divides this into smaller buffers for its use. 161// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 162// So for now we just assume that client is double-buffered for fast tracks. 163// FIXME It would be better for client to tell AudioFlinger the value of N, 164// so AudioFlinger could allocate the right amount of memory. 165// See the client's minBufCount and mNotificationFramesAct calculations for details. 166 167// This is the default value, if not specified by property. 168static const int kFastTrackMultiplier = 2; 169 170// The minimum and maximum allowed values 171static const int kFastTrackMultiplierMin = 1; 172static const int kFastTrackMultiplierMax = 2; 173 174// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 175static int sFastTrackMultiplier = kFastTrackMultiplier; 176 177// See Thread::readOnlyHeap(). 178// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 179// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 180// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 181static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 182 183// ---------------------------------------------------------------------------- 184 185static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 186 187static void sFastTrackMultiplierInit() 188{ 189 char value[PROPERTY_VALUE_MAX]; 190 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 191 char *endptr; 192 unsigned long ul = strtoul(value, &endptr, 0); 193 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 194 sFastTrackMultiplier = (int) ul; 195 } 196 } 197} 198 199// ---------------------------------------------------------------------------- 200 201#ifdef ADD_BATTERY_DATA 202// To collect the amplifier usage 203static void addBatteryData(uint32_t params) { 204 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 205 if (service == NULL) { 206 // it already logged 207 return; 208 } 209 210 service->addBatteryData(params); 211} 212#endif 213 214 215// ---------------------------------------------------------------------------- 216// CPU Stats 217// ---------------------------------------------------------------------------- 218 219class CpuStats { 220public: 221 CpuStats(); 222 void sample(const String8 &title); 223#ifdef DEBUG_CPU_USAGE 224private: 225 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 226 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 227 228 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 229 230 int mCpuNum; // thread's current CPU number 231 int mCpukHz; // frequency of thread's current CPU in kHz 232#endif 233}; 234 235CpuStats::CpuStats() 236#ifdef DEBUG_CPU_USAGE 237 : mCpuNum(-1), mCpukHz(-1) 238#endif 239{ 240} 241 242void CpuStats::sample(const String8 &title 243#ifndef DEBUG_CPU_USAGE 244 __unused 245#endif 246 ) { 247#ifdef DEBUG_CPU_USAGE 248 // get current thread's delta CPU time in wall clock ns 249 double wcNs; 250 bool valid = mCpuUsage.sampleAndEnable(wcNs); 251 252 // record sample for wall clock statistics 253 if (valid) { 254 mWcStats.sample(wcNs); 255 } 256 257 // get the current CPU number 258 int cpuNum = sched_getcpu(); 259 260 // get the current CPU frequency in kHz 261 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 262 263 // check if either CPU number or frequency changed 264 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 265 mCpuNum = cpuNum; 266 mCpukHz = cpukHz; 267 // ignore sample for purposes of cycles 268 valid = false; 269 } 270 271 // if no change in CPU number or frequency, then record sample for cycle statistics 272 if (valid && mCpukHz > 0) { 273 double cycles = wcNs * cpukHz * 0.000001; 274 mHzStats.sample(cycles); 275 } 276 277 unsigned n = mWcStats.n(); 278 // mCpuUsage.elapsed() is expensive, so don't call it every loop 279 if ((n & 127) == 1) { 280 long long elapsed = mCpuUsage.elapsed(); 281 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 282 double perLoop = elapsed / (double) n; 283 double perLoop100 = perLoop * 0.01; 284 double perLoop1k = perLoop * 0.001; 285 double mean = mWcStats.mean(); 286 double stddev = mWcStats.stddev(); 287 double minimum = mWcStats.minimum(); 288 double maximum = mWcStats.maximum(); 289 double meanCycles = mHzStats.mean(); 290 double stddevCycles = mHzStats.stddev(); 291 double minCycles = mHzStats.minimum(); 292 double maxCycles = mHzStats.maximum(); 293 mCpuUsage.resetElapsed(); 294 mWcStats.reset(); 295 mHzStats.reset(); 296 ALOGD("CPU usage for %s over past %.1f secs\n" 297 " (%u mixer loops at %.1f mean ms per loop):\n" 298 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 299 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 300 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 301 title.string(), 302 elapsed * .000000001, n, perLoop * .000001, 303 mean * .001, 304 stddev * .001, 305 minimum * .001, 306 maximum * .001, 307 mean / perLoop100, 308 stddev / perLoop100, 309 minimum / perLoop100, 310 maximum / perLoop100, 311 meanCycles / perLoop1k, 312 stddevCycles / perLoop1k, 313 minCycles / perLoop1k, 314 maxCycles / perLoop1k); 315 316 } 317 } 318#endif 319}; 320 321// ---------------------------------------------------------------------------- 322// ThreadBase 323// ---------------------------------------------------------------------------- 324 325// static 326const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 327{ 328 switch (type) { 329 case MIXER: 330 return "MIXER"; 331 case DIRECT: 332 return "DIRECT"; 333 case DUPLICATING: 334 return "DUPLICATING"; 335 case RECORD: 336 return "RECORD"; 337 case OFFLOAD: 338 return "OFFLOAD"; 339 default: 340 return "unknown"; 341 } 342} 343 344String8 devicesToString(audio_devices_t devices) 345{ 346 static const struct mapping { 347 audio_devices_t mDevices; 348 const char * mString; 349 } mappingsOut[] = { 350 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 351 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 352 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 353 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 354 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 355 AUDIO_DEVICE_NONE, "NONE", // must be last 356 }, mappingsIn[] = { 357 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 358 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 359 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 360 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 361 AUDIO_DEVICE_NONE, "NONE", // must be last 362 }; 363 String8 result; 364 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 365 const mapping *entry; 366 if (devices & AUDIO_DEVICE_BIT_IN) { 367 devices &= ~AUDIO_DEVICE_BIT_IN; 368 entry = mappingsIn; 369 } else { 370 entry = mappingsOut; 371 } 372 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 373 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 374 if (devices & entry->mDevices) { 375 if (!result.isEmpty()) { 376 result.append("|"); 377 } 378 result.append(entry->mString); 379 } 380 } 381 if (devices & ~allDevices) { 382 if (!result.isEmpty()) { 383 result.append("|"); 384 } 385 result.appendFormat("0x%X", devices & ~allDevices); 386 } 387 if (result.isEmpty()) { 388 result.append(entry->mString); 389 } 390 return result; 391} 392 393String8 inputFlagsToString(audio_input_flags_t flags) 394{ 395 static const struct mapping { 396 audio_input_flags_t mFlag; 397 const char * mString; 398 } mappings[] = { 399 AUDIO_INPUT_FLAG_FAST, "FAST", 400 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 401 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 402 }; 403 String8 result; 404 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 405 const mapping *entry; 406 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 407 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 408 if (flags & entry->mFlag) { 409 if (!result.isEmpty()) { 410 result.append("|"); 411 } 412 result.append(entry->mString); 413 } 414 } 415 if (flags & ~allFlags) { 416 if (!result.isEmpty()) { 417 result.append("|"); 418 } 419 result.appendFormat("0x%X", flags & ~allFlags); 420 } 421 if (result.isEmpty()) { 422 result.append(entry->mString); 423 } 424 return result; 425} 426 427String8 outputFlagsToString(audio_output_flags_t flags) 428{ 429 static const struct mapping { 430 audio_output_flags_t mFlag; 431 const char * mString; 432 } mappings[] = { 433 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 434 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 435 AUDIO_OUTPUT_FLAG_FAST, "FAST", 436 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 437 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 438 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 439 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 440 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 441 }; 442 String8 result; 443 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 444 const mapping *entry; 445 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 446 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 447 if (flags & entry->mFlag) { 448 if (!result.isEmpty()) { 449 result.append("|"); 450 } 451 result.append(entry->mString); 452 } 453 } 454 if (flags & ~allFlags) { 455 if (!result.isEmpty()) { 456 result.append("|"); 457 } 458 result.appendFormat("0x%X", flags & ~allFlags); 459 } 460 if (result.isEmpty()) { 461 result.append(entry->mString); 462 } 463 return result; 464} 465 466const char *sourceToString(audio_source_t source) 467{ 468 switch (source) { 469 case AUDIO_SOURCE_DEFAULT: return "default"; 470 case AUDIO_SOURCE_MIC: return "mic"; 471 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 472 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 473 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 474 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 475 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 476 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 477 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 478 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 479 case AUDIO_SOURCE_HOTWORD: return "hotword"; 480 default: return "unknown"; 481 } 482} 483 484AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 485 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 486 : Thread(false /*canCallJava*/), 487 mType(type), 488 mAudioFlinger(audioFlinger), 489 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 490 // are set by PlaybackThread::readOutputParameters_l() or 491 // RecordThread::readInputParameters_l() 492 //FIXME: mStandby should be true here. Is this some kind of hack? 493 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 494 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 495 // mName will be set by concrete (non-virtual) subclass 496 mDeathRecipient(new PMDeathRecipient(this)) 497{ 498} 499 500AudioFlinger::ThreadBase::~ThreadBase() 501{ 502 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 503 mConfigEvents.clear(); 504 505 // do not lock the mutex in destructor 506 releaseWakeLock_l(); 507 if (mPowerManager != 0) { 508 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 509 binder->unlinkToDeath(mDeathRecipient); 510 } 511} 512 513status_t AudioFlinger::ThreadBase::readyToRun() 514{ 515 status_t status = initCheck(); 516 if (status == NO_ERROR) { 517 ALOGI("AudioFlinger's thread %p ready to run", this); 518 } else { 519 ALOGE("No working audio driver found."); 520 } 521 return status; 522} 523 524void AudioFlinger::ThreadBase::exit() 525{ 526 ALOGV("ThreadBase::exit"); 527 // do any cleanup required for exit to succeed 528 preExit(); 529 { 530 // This lock prevents the following race in thread (uniprocessor for illustration): 531 // if (!exitPending()) { 532 // // context switch from here to exit() 533 // // exit() calls requestExit(), what exitPending() observes 534 // // exit() calls signal(), which is dropped since no waiters 535 // // context switch back from exit() to here 536 // mWaitWorkCV.wait(...); 537 // // now thread is hung 538 // } 539 AutoMutex lock(mLock); 540 requestExit(); 541 mWaitWorkCV.broadcast(); 542 } 543 // When Thread::requestExitAndWait is made virtual and this method is renamed to 544 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 545 requestExitAndWait(); 546} 547 548status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 549{ 550 status_t status; 551 552 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 553 Mutex::Autolock _l(mLock); 554 555 return sendSetParameterConfigEvent_l(keyValuePairs); 556} 557 558// sendConfigEvent_l() must be called with ThreadBase::mLock held 559// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 560status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 561{ 562 status_t status = NO_ERROR; 563 564 mConfigEvents.add(event); 565 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 566 mWaitWorkCV.signal(); 567 mLock.unlock(); 568 { 569 Mutex::Autolock _l(event->mLock); 570 while (event->mWaitStatus) { 571 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 572 event->mStatus = TIMED_OUT; 573 event->mWaitStatus = false; 574 } 575 } 576 status = event->mStatus; 577 } 578 mLock.lock(); 579 return status; 580} 581 582void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 583{ 584 Mutex::Autolock _l(mLock); 585 sendIoConfigEvent_l(event, param); 586} 587 588// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 589void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 590{ 591 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 592 sendConfigEvent_l(configEvent); 593} 594 595// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 596void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 597{ 598 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 599 sendConfigEvent_l(configEvent); 600} 601 602// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 603status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 604{ 605 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 606 return sendConfigEvent_l(configEvent); 607} 608 609status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 610 const struct audio_patch *patch, 611 audio_patch_handle_t *handle) 612{ 613 Mutex::Autolock _l(mLock); 614 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 615 status_t status = sendConfigEvent_l(configEvent); 616 if (status == NO_ERROR) { 617 CreateAudioPatchConfigEventData *data = 618 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 619 *handle = data->mHandle; 620 } 621 return status; 622} 623 624status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 625 const audio_patch_handle_t handle) 626{ 627 Mutex::Autolock _l(mLock); 628 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 629 return sendConfigEvent_l(configEvent); 630} 631 632 633// post condition: mConfigEvents.isEmpty() 634void AudioFlinger::ThreadBase::processConfigEvents_l() 635{ 636 bool configChanged = false; 637 638 while (!mConfigEvents.isEmpty()) { 639 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 640 sp<ConfigEvent> event = mConfigEvents[0]; 641 mConfigEvents.removeAt(0); 642 switch (event->mType) { 643 case CFG_EVENT_PRIO: { 644 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 645 // FIXME Need to understand why this has to be done asynchronously 646 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 647 true /*asynchronous*/); 648 if (err != 0) { 649 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 650 data->mPrio, data->mPid, data->mTid, err); 651 } 652 } break; 653 case CFG_EVENT_IO: { 654 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 655 audioConfigChanged(data->mEvent, data->mParam); 656 } break; 657 case CFG_EVENT_SET_PARAMETER: { 658 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 659 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 660 configChanged = true; 661 } 662 } break; 663 case CFG_EVENT_CREATE_AUDIO_PATCH: { 664 CreateAudioPatchConfigEventData *data = 665 (CreateAudioPatchConfigEventData *)event->mData.get(); 666 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 667 } break; 668 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 669 ReleaseAudioPatchConfigEventData *data = 670 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 671 event->mStatus = releaseAudioPatch_l(data->mHandle); 672 } break; 673 default: 674 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 675 break; 676 } 677 { 678 Mutex::Autolock _l(event->mLock); 679 if (event->mWaitStatus) { 680 event->mWaitStatus = false; 681 event->mCond.signal(); 682 } 683 } 684 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 685 } 686 687 if (configChanged) { 688 cacheParameters_l(); 689 } 690} 691 692String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 693 String8 s; 694 if (output) { 695 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 696 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 697 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 698 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 699 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 700 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 701 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 702 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 703 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 704 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 705 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 706 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 707 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 708 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 709 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 710 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 711 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 712 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 713 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 714 } else { 715 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 716 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 717 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 718 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 719 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 720 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 721 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 722 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 723 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 724 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 725 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 726 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 727 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 728 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 729 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 730 } 731 int len = s.length(); 732 if (s.length() > 2) { 733 char *str = s.lockBuffer(len); 734 s.unlockBuffer(len - 2); 735 } 736 return s; 737} 738 739void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 740{ 741 const size_t SIZE = 256; 742 char buffer[SIZE]; 743 String8 result; 744 745 bool locked = AudioFlinger::dumpTryLock(mLock); 746 if (!locked) { 747 dprintf(fd, "thread %p may be deadlocked\n", this); 748 } 749 750 dprintf(fd, " Thread name: %s\n", mThreadName); 751 dprintf(fd, " I/O handle: %d\n", mId); 752 dprintf(fd, " TID: %d\n", getTid()); 753 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 754 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 755 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 756 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 757 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 758 dprintf(fd, " Channel count: %u\n", mChannelCount); 759 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 760 channelMaskToString(mChannelMask, mType != RECORD).string()); 761 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 762 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 763 dprintf(fd, " Pending config events:"); 764 size_t numConfig = mConfigEvents.size(); 765 if (numConfig) { 766 for (size_t i = 0; i < numConfig; i++) { 767 mConfigEvents[i]->dump(buffer, SIZE); 768 dprintf(fd, "\n %s", buffer); 769 } 770 dprintf(fd, "\n"); 771 } else { 772 dprintf(fd, " none\n"); 773 } 774 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 775 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 776 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 777 778 if (locked) { 779 mLock.unlock(); 780 } 781} 782 783void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 784{ 785 const size_t SIZE = 256; 786 char buffer[SIZE]; 787 String8 result; 788 789 size_t numEffectChains = mEffectChains.size(); 790 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 791 write(fd, buffer, strlen(buffer)); 792 793 for (size_t i = 0; i < numEffectChains; ++i) { 794 sp<EffectChain> chain = mEffectChains[i]; 795 if (chain != 0) { 796 chain->dump(fd, args); 797 } 798 } 799} 800 801void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 802{ 803 Mutex::Autolock _l(mLock); 804 acquireWakeLock_l(uid); 805} 806 807String16 AudioFlinger::ThreadBase::getWakeLockTag() 808{ 809 switch (mType) { 810 case MIXER: 811 return String16("AudioMix"); 812 case DIRECT: 813 return String16("AudioDirectOut"); 814 case DUPLICATING: 815 return String16("AudioDup"); 816 case RECORD: 817 return String16("AudioIn"); 818 case OFFLOAD: 819 return String16("AudioOffload"); 820 default: 821 ALOG_ASSERT(false); 822 return String16("AudioUnknown"); 823 } 824} 825 826void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 827{ 828 getPowerManager_l(); 829 if (mPowerManager != 0) { 830 sp<IBinder> binder = new BBinder(); 831 status_t status; 832 if (uid >= 0) { 833 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 834 binder, 835 getWakeLockTag(), 836 String16("media"), 837 uid, 838 true /* FIXME force oneway contrary to .aidl */); 839 } else { 840 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 841 binder, 842 getWakeLockTag(), 843 String16("media"), 844 true /* FIXME force oneway contrary to .aidl */); 845 } 846 if (status == NO_ERROR) { 847 mWakeLockToken = binder; 848 } 849 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 850 } 851} 852 853void AudioFlinger::ThreadBase::releaseWakeLock() 854{ 855 Mutex::Autolock _l(mLock); 856 releaseWakeLock_l(); 857} 858 859void AudioFlinger::ThreadBase::releaseWakeLock_l() 860{ 861 if (mWakeLockToken != 0) { 862 ALOGV("releaseWakeLock_l() %s", mThreadName); 863 if (mPowerManager != 0) { 864 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 865 true /* FIXME force oneway contrary to .aidl */); 866 } 867 mWakeLockToken.clear(); 868 } 869} 870 871void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 872 Mutex::Autolock _l(mLock); 873 updateWakeLockUids_l(uids); 874} 875 876void AudioFlinger::ThreadBase::getPowerManager_l() { 877 878 if (mPowerManager == 0) { 879 // use checkService() to avoid blocking if power service is not up yet 880 sp<IBinder> binder = 881 defaultServiceManager()->checkService(String16("power")); 882 if (binder == 0) { 883 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 884 } else { 885 mPowerManager = interface_cast<IPowerManager>(binder); 886 binder->linkToDeath(mDeathRecipient); 887 } 888 } 889} 890 891void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 892 893 getPowerManager_l(); 894 if (mWakeLockToken == NULL) { 895 ALOGE("no wake lock to update!"); 896 return; 897 } 898 if (mPowerManager != 0) { 899 sp<IBinder> binder = new BBinder(); 900 status_t status; 901 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 902 true /* FIXME force oneway contrary to .aidl */); 903 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 904 } 905} 906 907void AudioFlinger::ThreadBase::clearPowerManager() 908{ 909 Mutex::Autolock _l(mLock); 910 releaseWakeLock_l(); 911 mPowerManager.clear(); 912} 913 914void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 915{ 916 sp<ThreadBase> thread = mThread.promote(); 917 if (thread != 0) { 918 thread->clearPowerManager(); 919 } 920 ALOGW("power manager service died !!!"); 921} 922 923void AudioFlinger::ThreadBase::setEffectSuspended( 924 const effect_uuid_t *type, bool suspend, int sessionId) 925{ 926 Mutex::Autolock _l(mLock); 927 setEffectSuspended_l(type, suspend, sessionId); 928} 929 930void AudioFlinger::ThreadBase::setEffectSuspended_l( 931 const effect_uuid_t *type, bool suspend, int sessionId) 932{ 933 sp<EffectChain> chain = getEffectChain_l(sessionId); 934 if (chain != 0) { 935 if (type != NULL) { 936 chain->setEffectSuspended_l(type, suspend); 937 } else { 938 chain->setEffectSuspendedAll_l(suspend); 939 } 940 } 941 942 updateSuspendedSessions_l(type, suspend, sessionId); 943} 944 945void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 946{ 947 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 948 if (index < 0) { 949 return; 950 } 951 952 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 953 mSuspendedSessions.valueAt(index); 954 955 for (size_t i = 0; i < sessionEffects.size(); i++) { 956 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 957 for (int j = 0; j < desc->mRefCount; j++) { 958 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 959 chain->setEffectSuspendedAll_l(true); 960 } else { 961 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 962 desc->mType.timeLow); 963 chain->setEffectSuspended_l(&desc->mType, true); 964 } 965 } 966 } 967} 968 969void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 970 bool suspend, 971 int sessionId) 972{ 973 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 974 975 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 976 977 if (suspend) { 978 if (index >= 0) { 979 sessionEffects = mSuspendedSessions.valueAt(index); 980 } else { 981 mSuspendedSessions.add(sessionId, sessionEffects); 982 } 983 } else { 984 if (index < 0) { 985 return; 986 } 987 sessionEffects = mSuspendedSessions.valueAt(index); 988 } 989 990 991 int key = EffectChain::kKeyForSuspendAll; 992 if (type != NULL) { 993 key = type->timeLow; 994 } 995 index = sessionEffects.indexOfKey(key); 996 997 sp<SuspendedSessionDesc> desc; 998 if (suspend) { 999 if (index >= 0) { 1000 desc = sessionEffects.valueAt(index); 1001 } else { 1002 desc = new SuspendedSessionDesc(); 1003 if (type != NULL) { 1004 desc->mType = *type; 1005 } 1006 sessionEffects.add(key, desc); 1007 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1008 } 1009 desc->mRefCount++; 1010 } else { 1011 if (index < 0) { 1012 return; 1013 } 1014 desc = sessionEffects.valueAt(index); 1015 if (--desc->mRefCount == 0) { 1016 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1017 sessionEffects.removeItemsAt(index); 1018 if (sessionEffects.isEmpty()) { 1019 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1020 sessionId); 1021 mSuspendedSessions.removeItem(sessionId); 1022 } 1023 } 1024 } 1025 if (!sessionEffects.isEmpty()) { 1026 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1027 } 1028} 1029 1030void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1031 bool enabled, 1032 int sessionId) 1033{ 1034 Mutex::Autolock _l(mLock); 1035 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1036} 1037 1038void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1039 bool enabled, 1040 int sessionId) 1041{ 1042 if (mType != RECORD) { 1043 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1044 // another session. This gives the priority to well behaved effect control panels 1045 // and applications not using global effects. 1046 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1047 // global effects 1048 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1049 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1050 } 1051 } 1052 1053 sp<EffectChain> chain = getEffectChain_l(sessionId); 1054 if (chain != 0) { 1055 chain->checkSuspendOnEffectEnabled(effect, enabled); 1056 } 1057} 1058 1059// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1060sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1061 const sp<AudioFlinger::Client>& client, 1062 const sp<IEffectClient>& effectClient, 1063 int32_t priority, 1064 int sessionId, 1065 effect_descriptor_t *desc, 1066 int *enabled, 1067 status_t *status) 1068{ 1069 sp<EffectModule> effect; 1070 sp<EffectHandle> handle; 1071 status_t lStatus; 1072 sp<EffectChain> chain; 1073 bool chainCreated = false; 1074 bool effectCreated = false; 1075 bool effectRegistered = false; 1076 1077 lStatus = initCheck(); 1078 if (lStatus != NO_ERROR) { 1079 ALOGW("createEffect_l() Audio driver not initialized."); 1080 goto Exit; 1081 } 1082 1083 // Reject any effect on Direct output threads for now, since the format of 1084 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1085 if (mType == DIRECT) { 1086 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1087 desc->name, mThreadName); 1088 lStatus = BAD_VALUE; 1089 goto Exit; 1090 } 1091 1092 // Reject any effect on mixer or duplicating multichannel sinks. 1093 // TODO: fix both format and multichannel issues with effects. 1094 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1095 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1096 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1097 lStatus = BAD_VALUE; 1098 goto Exit; 1099 } 1100 1101 // Allow global effects only on offloaded and mixer threads 1102 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1103 switch (mType) { 1104 case MIXER: 1105 case OFFLOAD: 1106 break; 1107 case DIRECT: 1108 case DUPLICATING: 1109 case RECORD: 1110 default: 1111 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1112 desc->name, mThreadName); 1113 lStatus = BAD_VALUE; 1114 goto Exit; 1115 } 1116 } 1117 1118 // Only Pre processor effects are allowed on input threads and only on input threads 1119 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1120 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1121 desc->name, desc->flags, mType); 1122 lStatus = BAD_VALUE; 1123 goto Exit; 1124 } 1125 1126 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1127 1128 { // scope for mLock 1129 Mutex::Autolock _l(mLock); 1130 1131 // check for existing effect chain with the requested audio session 1132 chain = getEffectChain_l(sessionId); 1133 if (chain == 0) { 1134 // create a new chain for this session 1135 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1136 chain = new EffectChain(this, sessionId); 1137 addEffectChain_l(chain); 1138 chain->setStrategy(getStrategyForSession_l(sessionId)); 1139 chainCreated = true; 1140 } else { 1141 effect = chain->getEffectFromDesc_l(desc); 1142 } 1143 1144 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1145 1146 if (effect == 0) { 1147 int id = mAudioFlinger->nextUniqueId(); 1148 // Check CPU and memory usage 1149 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1150 if (lStatus != NO_ERROR) { 1151 goto Exit; 1152 } 1153 effectRegistered = true; 1154 // create a new effect module if none present in the chain 1155 effect = new EffectModule(this, chain, desc, id, sessionId); 1156 lStatus = effect->status(); 1157 if (lStatus != NO_ERROR) { 1158 goto Exit; 1159 } 1160 effect->setOffloaded(mType == OFFLOAD, mId); 1161 1162 lStatus = chain->addEffect_l(effect); 1163 if (lStatus != NO_ERROR) { 1164 goto Exit; 1165 } 1166 effectCreated = true; 1167 1168 effect->setDevice(mOutDevice); 1169 effect->setDevice(mInDevice); 1170 effect->setMode(mAudioFlinger->getMode()); 1171 effect->setAudioSource(mAudioSource); 1172 } 1173 // create effect handle and connect it to effect module 1174 handle = new EffectHandle(effect, client, effectClient, priority); 1175 lStatus = handle->initCheck(); 1176 if (lStatus == OK) { 1177 lStatus = effect->addHandle(handle.get()); 1178 } 1179 if (enabled != NULL) { 1180 *enabled = (int)effect->isEnabled(); 1181 } 1182 } 1183 1184Exit: 1185 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1186 Mutex::Autolock _l(mLock); 1187 if (effectCreated) { 1188 chain->removeEffect_l(effect); 1189 } 1190 if (effectRegistered) { 1191 AudioSystem::unregisterEffect(effect->id()); 1192 } 1193 if (chainCreated) { 1194 removeEffectChain_l(chain); 1195 } 1196 handle.clear(); 1197 } 1198 1199 *status = lStatus; 1200 return handle; 1201} 1202 1203sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1204{ 1205 Mutex::Autolock _l(mLock); 1206 return getEffect_l(sessionId, effectId); 1207} 1208 1209sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1210{ 1211 sp<EffectChain> chain = getEffectChain_l(sessionId); 1212 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1213} 1214 1215// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1216// PlaybackThread::mLock held 1217status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1218{ 1219 // check for existing effect chain with the requested audio session 1220 int sessionId = effect->sessionId(); 1221 sp<EffectChain> chain = getEffectChain_l(sessionId); 1222 bool chainCreated = false; 1223 1224 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1225 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1226 this, effect->desc().name, effect->desc().flags); 1227 1228 if (chain == 0) { 1229 // create a new chain for this session 1230 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1231 chain = new EffectChain(this, sessionId); 1232 addEffectChain_l(chain); 1233 chain->setStrategy(getStrategyForSession_l(sessionId)); 1234 chainCreated = true; 1235 } 1236 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1237 1238 if (chain->getEffectFromId_l(effect->id()) != 0) { 1239 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1240 this, effect->desc().name, chain.get()); 1241 return BAD_VALUE; 1242 } 1243 1244 effect->setOffloaded(mType == OFFLOAD, mId); 1245 1246 status_t status = chain->addEffect_l(effect); 1247 if (status != NO_ERROR) { 1248 if (chainCreated) { 1249 removeEffectChain_l(chain); 1250 } 1251 return status; 1252 } 1253 1254 effect->setDevice(mOutDevice); 1255 effect->setDevice(mInDevice); 1256 effect->setMode(mAudioFlinger->getMode()); 1257 effect->setAudioSource(mAudioSource); 1258 return NO_ERROR; 1259} 1260 1261void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1262 1263 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1264 effect_descriptor_t desc = effect->desc(); 1265 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1266 detachAuxEffect_l(effect->id()); 1267 } 1268 1269 sp<EffectChain> chain = effect->chain().promote(); 1270 if (chain != 0) { 1271 // remove effect chain if removing last effect 1272 if (chain->removeEffect_l(effect) == 0) { 1273 removeEffectChain_l(chain); 1274 } 1275 } else { 1276 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::lockEffectChains_l( 1281 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1282{ 1283 effectChains = mEffectChains; 1284 for (size_t i = 0; i < mEffectChains.size(); i++) { 1285 mEffectChains[i]->lock(); 1286 } 1287} 1288 1289void AudioFlinger::ThreadBase::unlockEffectChains( 1290 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1291{ 1292 for (size_t i = 0; i < effectChains.size(); i++) { 1293 effectChains[i]->unlock(); 1294 } 1295} 1296 1297sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1298{ 1299 Mutex::Autolock _l(mLock); 1300 return getEffectChain_l(sessionId); 1301} 1302 1303sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1304{ 1305 size_t size = mEffectChains.size(); 1306 for (size_t i = 0; i < size; i++) { 1307 if (mEffectChains[i]->sessionId() == sessionId) { 1308 return mEffectChains[i]; 1309 } 1310 } 1311 return 0; 1312} 1313 1314void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 size_t size = mEffectChains.size(); 1318 for (size_t i = 0; i < size; i++) { 1319 mEffectChains[i]->setMode_l(mode); 1320 } 1321} 1322 1323void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1324{ 1325 config->type = AUDIO_PORT_TYPE_MIX; 1326 config->ext.mix.handle = mId; 1327 config->sample_rate = mSampleRate; 1328 config->format = mFormat; 1329 config->channel_mask = mChannelMask; 1330 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1331 AUDIO_PORT_CONFIG_FORMAT; 1332} 1333 1334 1335// ---------------------------------------------------------------------------- 1336// Playback 1337// ---------------------------------------------------------------------------- 1338 1339AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1340 AudioStreamOut* output, 1341 audio_io_handle_t id, 1342 audio_devices_t device, 1343 type_t type) 1344 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1345 mNormalFrameCount(0), mSinkBuffer(NULL), 1346 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1347 mMixerBuffer(NULL), 1348 mMixerBufferSize(0), 1349 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1350 mMixerBufferValid(false), 1351 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1352 mEffectBuffer(NULL), 1353 mEffectBufferSize(0), 1354 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1355 mEffectBufferValid(false), 1356 mSuspended(0), mBytesWritten(0), 1357 mActiveTracksGeneration(0), 1358 // mStreamTypes[] initialized in constructor body 1359 mOutput(output), 1360 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1361 mMixerStatus(MIXER_IDLE), 1362 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1363 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1364 mBytesRemaining(0), 1365 mCurrentWriteLength(0), 1366 mUseAsyncWrite(false), 1367 mWriteAckSequence(0), 1368 mDrainSequence(0), 1369 mSignalPending(false), 1370 mScreenState(AudioFlinger::mScreenState), 1371 // index 0 is reserved for normal mixer's submix 1372 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1373 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1374 // mLatchD, mLatchQ, 1375 mLatchDValid(false), mLatchQValid(false) 1376{ 1377 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1378 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1379 1380 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1381 // it would be safer to explicitly pass initial masterVolume/masterMute as 1382 // parameter. 1383 // 1384 // If the HAL we are using has support for master volume or master mute, 1385 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1386 // and the mute set to false). 1387 mMasterVolume = audioFlinger->masterVolume_l(); 1388 mMasterMute = audioFlinger->masterMute_l(); 1389 if (mOutput && mOutput->audioHwDev) { 1390 if (mOutput->audioHwDev->canSetMasterVolume()) { 1391 mMasterVolume = 1.0; 1392 } 1393 1394 if (mOutput->audioHwDev->canSetMasterMute()) { 1395 mMasterMute = false; 1396 } 1397 } 1398 1399 readOutputParameters_l(); 1400 1401 // ++ operator does not compile 1402 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1403 stream = (audio_stream_type_t) (stream + 1)) { 1404 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1405 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1406 } 1407} 1408 1409AudioFlinger::PlaybackThread::~PlaybackThread() 1410{ 1411 mAudioFlinger->unregisterWriter(mNBLogWriter); 1412 free(mSinkBuffer); 1413 free(mMixerBuffer); 1414 free(mEffectBuffer); 1415} 1416 1417void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1418{ 1419 dumpInternals(fd, args); 1420 dumpTracks(fd, args); 1421 dumpEffectChains(fd, args); 1422} 1423 1424void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1425{ 1426 const size_t SIZE = 256; 1427 char buffer[SIZE]; 1428 String8 result; 1429 1430 result.appendFormat(" Stream volumes in dB: "); 1431 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1432 const stream_type_t *st = &mStreamTypes[i]; 1433 if (i > 0) { 1434 result.appendFormat(", "); 1435 } 1436 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1437 if (st->mute) { 1438 result.append("M"); 1439 } 1440 } 1441 result.append("\n"); 1442 write(fd, result.string(), result.length()); 1443 result.clear(); 1444 1445 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1446 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1447 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1448 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1449 1450 size_t numtracks = mTracks.size(); 1451 size_t numactive = mActiveTracks.size(); 1452 dprintf(fd, " %d Tracks", numtracks); 1453 size_t numactiveseen = 0; 1454 if (numtracks) { 1455 dprintf(fd, " of which %d are active\n", numactive); 1456 Track::appendDumpHeader(result); 1457 for (size_t i = 0; i < numtracks; ++i) { 1458 sp<Track> track = mTracks[i]; 1459 if (track != 0) { 1460 bool active = mActiveTracks.indexOf(track) >= 0; 1461 if (active) { 1462 numactiveseen++; 1463 } 1464 track->dump(buffer, SIZE, active); 1465 result.append(buffer); 1466 } 1467 } 1468 } else { 1469 result.append("\n"); 1470 } 1471 if (numactiveseen != numactive) { 1472 // some tracks in the active list were not in the tracks list 1473 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1474 " not in the track list\n"); 1475 result.append(buffer); 1476 Track::appendDumpHeader(result); 1477 for (size_t i = 0; i < numactive; ++i) { 1478 sp<Track> track = mActiveTracks[i].promote(); 1479 if (track != 0 && mTracks.indexOf(track) < 0) { 1480 track->dump(buffer, SIZE, true); 1481 result.append(buffer); 1482 } 1483 } 1484 } 1485 1486 write(fd, result.string(), result.size()); 1487} 1488 1489void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1490{ 1491 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1492 1493 dumpBase(fd, args); 1494 1495 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1496 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1497 dprintf(fd, " Total writes: %d\n", mNumWrites); 1498 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1499 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1500 dprintf(fd, " Suspend count: %d\n", mSuspended); 1501 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1502 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1503 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1504 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1505 AudioStreamOut *output = mOutput; 1506 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1507 String8 flagsAsString = outputFlagsToString(flags); 1508 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1509} 1510 1511// Thread virtuals 1512 1513void AudioFlinger::PlaybackThread::onFirstRef() 1514{ 1515 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1516} 1517 1518// ThreadBase virtuals 1519void AudioFlinger::PlaybackThread::preExit() 1520{ 1521 ALOGV(" preExit()"); 1522 // FIXME this is using hard-coded strings but in the future, this functionality will be 1523 // converted to use audio HAL extensions required to support tunneling 1524 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1525} 1526 1527// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1528sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1529 const sp<AudioFlinger::Client>& client, 1530 audio_stream_type_t streamType, 1531 uint32_t sampleRate, 1532 audio_format_t format, 1533 audio_channel_mask_t channelMask, 1534 size_t *pFrameCount, 1535 const sp<IMemory>& sharedBuffer, 1536 int sessionId, 1537 IAudioFlinger::track_flags_t *flags, 1538 pid_t tid, 1539 int uid, 1540 status_t *status) 1541{ 1542 size_t frameCount = *pFrameCount; 1543 sp<Track> track; 1544 status_t lStatus; 1545 1546 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1547 1548 // client expresses a preference for FAST, but we get the final say 1549 if (*flags & IAudioFlinger::TRACK_FAST) { 1550 if ( 1551 // not timed 1552 (!isTimed) && 1553 // either of these use cases: 1554 ( 1555 // use case 1: shared buffer with any frame count 1556 ( 1557 (sharedBuffer != 0) 1558 ) || 1559 // use case 2: frame count is default or at least as large as HAL 1560 ( 1561 // we formerly checked for a callback handler (non-0 tid), 1562 // but that is no longer required for TRANSFER_OBTAIN mode 1563 ((frameCount == 0) || 1564 (frameCount >= mFrameCount)) 1565 ) 1566 ) && 1567 // PCM data 1568 audio_is_linear_pcm(format) && 1569 // identical channel mask to sink, or mono in and stereo sink 1570 (channelMask == mChannelMask || 1571 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1572 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1573 // hardware sample rate 1574 (sampleRate == mSampleRate) && 1575 // normal mixer has an associated fast mixer 1576 hasFastMixer() && 1577 // there are sufficient fast track slots available 1578 (mFastTrackAvailMask != 0) 1579 // FIXME test that MixerThread for this fast track has a capable output HAL 1580 // FIXME add a permission test also? 1581 ) { 1582 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1583 if (frameCount == 0) { 1584 // read the fast track multiplier property the first time it is needed 1585 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1586 if (ok != 0) { 1587 ALOGE("%s pthread_once failed: %d", __func__, ok); 1588 } 1589 frameCount = mFrameCount * sFastTrackMultiplier; 1590 } 1591 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1592 frameCount, mFrameCount); 1593 } else { 1594 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1595 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1596 "sampleRate=%u mSampleRate=%u " 1597 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1598 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1599 audio_is_linear_pcm(format), 1600 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1601 *flags &= ~IAudioFlinger::TRACK_FAST; 1602 } 1603 } 1604 // For normal PCM streaming tracks, update minimum frame count. 1605 // For compatibility with AudioTrack calculation, buffer depth is forced 1606 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1607 // This is probably too conservative, but legacy application code may depend on it. 1608 // If you change this calculation, also review the start threshold which is related. 1609 if (!(*flags & IAudioFlinger::TRACK_FAST) 1610 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1611 // this must match AudioTrack.cpp calculateMinFrameCount(). 1612 // TODO: Move to a common library 1613 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1614 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1615 if (minBufCount < 2) { 1616 minBufCount = 2; 1617 } 1618 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1619 // or the client should compute and pass in a larger buffer request. 1620 size_t minFrameCount = 1621 minBufCount * sourceFramesNeededWithTimestretch( 1622 sampleRate, mNormalFrameCount, 1623 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1624 if (frameCount < minFrameCount) { // including frameCount == 0 1625 frameCount = minFrameCount; 1626 } 1627 } 1628 *pFrameCount = frameCount; 1629 1630 switch (mType) { 1631 1632 case DIRECT: 1633 if (audio_is_linear_pcm(format)) { 1634 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1635 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1636 "for output %p with format %#x", 1637 sampleRate, format, channelMask, mOutput, mFormat); 1638 lStatus = BAD_VALUE; 1639 goto Exit; 1640 } 1641 } 1642 break; 1643 1644 case OFFLOAD: 1645 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1646 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1647 "for output %p with format %#x", 1648 sampleRate, format, channelMask, mOutput, mFormat); 1649 lStatus = BAD_VALUE; 1650 goto Exit; 1651 } 1652 break; 1653 1654 default: 1655 if (!audio_is_linear_pcm(format)) { 1656 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1657 "for output %p with format %#x", 1658 format, mOutput, mFormat); 1659 lStatus = BAD_VALUE; 1660 goto Exit; 1661 } 1662 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1663 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1664 lStatus = BAD_VALUE; 1665 goto Exit; 1666 } 1667 break; 1668 1669 } 1670 1671 lStatus = initCheck(); 1672 if (lStatus != NO_ERROR) { 1673 ALOGE("createTrack_l() audio driver not initialized"); 1674 goto Exit; 1675 } 1676 1677 { // scope for mLock 1678 Mutex::Autolock _l(mLock); 1679 1680 // all tracks in same audio session must share the same routing strategy otherwise 1681 // conflicts will happen when tracks are moved from one output to another by audio policy 1682 // manager 1683 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1684 for (size_t i = 0; i < mTracks.size(); ++i) { 1685 sp<Track> t = mTracks[i]; 1686 if (t != 0 && t->isExternalTrack()) { 1687 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1688 if (sessionId == t->sessionId() && strategy != actual) { 1689 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1690 strategy, actual); 1691 lStatus = BAD_VALUE; 1692 goto Exit; 1693 } 1694 } 1695 } 1696 1697 if (!isTimed) { 1698 track = new Track(this, client, streamType, sampleRate, format, 1699 channelMask, frameCount, NULL, sharedBuffer, 1700 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1701 } else { 1702 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1703 channelMask, frameCount, sharedBuffer, sessionId, uid); 1704 } 1705 1706 // new Track always returns non-NULL, 1707 // but TimedTrack::create() is a factory that could fail by returning NULL 1708 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1709 if (lStatus != NO_ERROR) { 1710 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1711 // track must be cleared from the caller as the caller has the AF lock 1712 goto Exit; 1713 } 1714 mTracks.add(track); 1715 1716 sp<EffectChain> chain = getEffectChain_l(sessionId); 1717 if (chain != 0) { 1718 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1719 track->setMainBuffer(chain->inBuffer()); 1720 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1721 chain->incTrackCnt(); 1722 } 1723 1724 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1725 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1726 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1727 // so ask activity manager to do this on our behalf 1728 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1729 } 1730 } 1731 1732 lStatus = NO_ERROR; 1733 1734Exit: 1735 *status = lStatus; 1736 return track; 1737} 1738 1739uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1740{ 1741 return latency; 1742} 1743 1744uint32_t AudioFlinger::PlaybackThread::latency() const 1745{ 1746 Mutex::Autolock _l(mLock); 1747 return latency_l(); 1748} 1749uint32_t AudioFlinger::PlaybackThread::latency_l() const 1750{ 1751 if (initCheck() == NO_ERROR) { 1752 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1753 } else { 1754 return 0; 1755 } 1756} 1757 1758void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1759{ 1760 Mutex::Autolock _l(mLock); 1761 // Don't apply master volume in SW if our HAL can do it for us. 1762 if (mOutput && mOutput->audioHwDev && 1763 mOutput->audioHwDev->canSetMasterVolume()) { 1764 mMasterVolume = 1.0; 1765 } else { 1766 mMasterVolume = value; 1767 } 1768} 1769 1770void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1771{ 1772 Mutex::Autolock _l(mLock); 1773 // Don't apply master mute in SW if our HAL can do it for us. 1774 if (mOutput && mOutput->audioHwDev && 1775 mOutput->audioHwDev->canSetMasterMute()) { 1776 mMasterMute = false; 1777 } else { 1778 mMasterMute = muted; 1779 } 1780} 1781 1782void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1783{ 1784 Mutex::Autolock _l(mLock); 1785 mStreamTypes[stream].volume = value; 1786 broadcast_l(); 1787} 1788 1789void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1790{ 1791 Mutex::Autolock _l(mLock); 1792 mStreamTypes[stream].mute = muted; 1793 broadcast_l(); 1794} 1795 1796float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1797{ 1798 Mutex::Autolock _l(mLock); 1799 return mStreamTypes[stream].volume; 1800} 1801 1802// addTrack_l() must be called with ThreadBase::mLock held 1803status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1804{ 1805 status_t status = ALREADY_EXISTS; 1806 1807 // set retry count for buffer fill 1808 track->mRetryCount = kMaxTrackStartupRetries; 1809 if (mActiveTracks.indexOf(track) < 0) { 1810 // the track is newly added, make sure it fills up all its 1811 // buffers before playing. This is to ensure the client will 1812 // effectively get the latency it requested. 1813 if (track->isExternalTrack()) { 1814 TrackBase::track_state state = track->mState; 1815 mLock.unlock(); 1816 status = AudioSystem::startOutput(mId, track->streamType(), 1817 (audio_session_t)track->sessionId()); 1818 mLock.lock(); 1819 // abort track was stopped/paused while we released the lock 1820 if (state != track->mState) { 1821 if (status == NO_ERROR) { 1822 mLock.unlock(); 1823 AudioSystem::stopOutput(mId, track->streamType(), 1824 (audio_session_t)track->sessionId()); 1825 mLock.lock(); 1826 } 1827 return INVALID_OPERATION; 1828 } 1829 // abort if start is rejected by audio policy manager 1830 if (status != NO_ERROR) { 1831 return PERMISSION_DENIED; 1832 } 1833#ifdef ADD_BATTERY_DATA 1834 // to track the speaker usage 1835 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1836#endif 1837 } 1838 1839 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1840 track->mResetDone = false; 1841 track->mPresentationCompleteFrames = 0; 1842 mActiveTracks.add(track); 1843 mWakeLockUids.add(track->uid()); 1844 mActiveTracksGeneration++; 1845 mLatestActiveTrack = track; 1846 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1847 if (chain != 0) { 1848 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1849 track->sessionId()); 1850 chain->incActiveTrackCnt(); 1851 } 1852 1853 status = NO_ERROR; 1854 } 1855 1856 onAddNewTrack_l(); 1857 return status; 1858} 1859 1860bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1861{ 1862 track->terminate(); 1863 // active tracks are removed by threadLoop() 1864 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1865 track->mState = TrackBase::STOPPED; 1866 if (!trackActive) { 1867 removeTrack_l(track); 1868 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1869 track->mState = TrackBase::STOPPING_1; 1870 } 1871 1872 return trackActive; 1873} 1874 1875void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1876{ 1877 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1878 mTracks.remove(track); 1879 deleteTrackName_l(track->name()); 1880 // redundant as track is about to be destroyed, for dumpsys only 1881 track->mName = -1; 1882 if (track->isFastTrack()) { 1883 int index = track->mFastIndex; 1884 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1885 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1886 mFastTrackAvailMask |= 1 << index; 1887 // redundant as track is about to be destroyed, for dumpsys only 1888 track->mFastIndex = -1; 1889 } 1890 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1891 if (chain != 0) { 1892 chain->decTrackCnt(); 1893 } 1894} 1895 1896void AudioFlinger::PlaybackThread::broadcast_l() 1897{ 1898 // Thread could be blocked waiting for async 1899 // so signal it to handle state changes immediately 1900 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1901 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1902 mSignalPending = true; 1903 mWaitWorkCV.broadcast(); 1904} 1905 1906String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1907{ 1908 Mutex::Autolock _l(mLock); 1909 if (initCheck() != NO_ERROR) { 1910 return String8(); 1911 } 1912 1913 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1914 const String8 out_s8(s); 1915 free(s); 1916 return out_s8; 1917} 1918 1919void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1920 AudioSystem::OutputDescriptor desc; 1921 void *param2 = NULL; 1922 1923 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1924 param); 1925 1926 switch (event) { 1927 case AudioSystem::OUTPUT_OPENED: 1928 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1929 desc.channelMask = mChannelMask; 1930 desc.samplingRate = mSampleRate; 1931 desc.format = mFormat; 1932 desc.frameCount = mNormalFrameCount; // FIXME see 1933 // AudioFlinger::frameCount(audio_io_handle_t) 1934 desc.latency = latency_l(); 1935 param2 = &desc; 1936 break; 1937 1938 case AudioSystem::STREAM_CONFIG_CHANGED: 1939 param2 = ¶m; 1940 case AudioSystem::OUTPUT_CLOSED: 1941 default: 1942 break; 1943 } 1944 mAudioFlinger->audioConfigChanged(event, mId, param2); 1945} 1946 1947void AudioFlinger::PlaybackThread::writeCallback() 1948{ 1949 ALOG_ASSERT(mCallbackThread != 0); 1950 mCallbackThread->resetWriteBlocked(); 1951} 1952 1953void AudioFlinger::PlaybackThread::drainCallback() 1954{ 1955 ALOG_ASSERT(mCallbackThread != 0); 1956 mCallbackThread->resetDraining(); 1957} 1958 1959void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1960{ 1961 Mutex::Autolock _l(mLock); 1962 // reject out of sequence requests 1963 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1964 mWriteAckSequence &= ~1; 1965 mWaitWorkCV.signal(); 1966 } 1967} 1968 1969void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1970{ 1971 Mutex::Autolock _l(mLock); 1972 // reject out of sequence requests 1973 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1974 mDrainSequence &= ~1; 1975 mWaitWorkCV.signal(); 1976 } 1977} 1978 1979// static 1980int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1981 void *param __unused, 1982 void *cookie) 1983{ 1984 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1985 ALOGV("asyncCallback() event %d", event); 1986 switch (event) { 1987 case STREAM_CBK_EVENT_WRITE_READY: 1988 me->writeCallback(); 1989 break; 1990 case STREAM_CBK_EVENT_DRAIN_READY: 1991 me->drainCallback(); 1992 break; 1993 default: 1994 ALOGW("asyncCallback() unknown event %d", event); 1995 break; 1996 } 1997 return 0; 1998} 1999 2000void AudioFlinger::PlaybackThread::readOutputParameters_l() 2001{ 2002 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2003 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2004 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2005 if (!audio_is_output_channel(mChannelMask)) { 2006 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2007 } 2008 if ((mType == MIXER || mType == DUPLICATING) 2009 && !isValidPcmSinkChannelMask(mChannelMask)) { 2010 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2011 mChannelMask); 2012 } 2013 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2014 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2015 mFormat = mHALFormat; 2016 if (!audio_is_valid_format(mFormat)) { 2017 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2018 } 2019 if ((mType == MIXER || mType == DUPLICATING) 2020 && !isValidPcmSinkFormat(mFormat)) { 2021 LOG_FATAL("HAL format %#x not supported for mixed output", 2022 mFormat); 2023 } 2024 mFrameSize = mOutput->getFrameSize(); 2025 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2026 mFrameCount = mBufferSize / mFrameSize; 2027 if (mFrameCount & 15) { 2028 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2029 mFrameCount); 2030 } 2031 2032 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2033 (mOutput->stream->set_callback != NULL)) { 2034 if (mOutput->stream->set_callback(mOutput->stream, 2035 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2036 mUseAsyncWrite = true; 2037 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2038 } 2039 } 2040 2041 mHwSupportsPause = false; 2042 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2043 if (mOutput->stream->pause != NULL) { 2044 if (mOutput->stream->resume != NULL) { 2045 mHwSupportsPause = true; 2046 } else { 2047 ALOGW("direct output implements pause but not resume"); 2048 } 2049 } else if (mOutput->stream->resume != NULL) { 2050 ALOGW("direct output implements resume but not pause"); 2051 } 2052 } 2053 2054 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2055 // For best precision, we use float instead of the associated output 2056 // device format (typically PCM 16 bit). 2057 2058 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2059 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2060 mBufferSize = mFrameSize * mFrameCount; 2061 2062 // TODO: We currently use the associated output device channel mask and sample rate. 2063 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2064 // (if a valid mask) to avoid premature downmix. 2065 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2066 // instead of the output device sample rate to avoid loss of high frequency information. 2067 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2068 } 2069 2070 // Calculate size of normal sink buffer relative to the HAL output buffer size 2071 double multiplier = 1.0; 2072 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2073 kUseFastMixer == FastMixer_Dynamic)) { 2074 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2075 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2076 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2077 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2078 maxNormalFrameCount = maxNormalFrameCount & ~15; 2079 if (maxNormalFrameCount < minNormalFrameCount) { 2080 maxNormalFrameCount = minNormalFrameCount; 2081 } 2082 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2083 if (multiplier <= 1.0) { 2084 multiplier = 1.0; 2085 } else if (multiplier <= 2.0) { 2086 if (2 * mFrameCount <= maxNormalFrameCount) { 2087 multiplier = 2.0; 2088 } else { 2089 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2090 } 2091 } else { 2092 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2093 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2094 // track, but we sometimes have to do this to satisfy the maximum frame count 2095 // constraint) 2096 // FIXME this rounding up should not be done if no HAL SRC 2097 uint32_t truncMult = (uint32_t) multiplier; 2098 if ((truncMult & 1)) { 2099 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2100 ++truncMult; 2101 } 2102 } 2103 multiplier = (double) truncMult; 2104 } 2105 } 2106 mNormalFrameCount = multiplier * mFrameCount; 2107 // round up to nearest 16 frames to satisfy AudioMixer 2108 if (mType == MIXER || mType == DUPLICATING) { 2109 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2110 } 2111 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2112 mNormalFrameCount); 2113 2114 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2115 // Originally this was int16_t[] array, need to remove legacy implications. 2116 free(mSinkBuffer); 2117 mSinkBuffer = NULL; 2118 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2119 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2120 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2121 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2122 2123 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2124 // drives the output. 2125 free(mMixerBuffer); 2126 mMixerBuffer = NULL; 2127 if (mMixerBufferEnabled) { 2128 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2129 mMixerBufferSize = mNormalFrameCount * mChannelCount 2130 * audio_bytes_per_sample(mMixerBufferFormat); 2131 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2132 } 2133 free(mEffectBuffer); 2134 mEffectBuffer = NULL; 2135 if (mEffectBufferEnabled) { 2136 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2137 mEffectBufferSize = mNormalFrameCount * mChannelCount 2138 * audio_bytes_per_sample(mEffectBufferFormat); 2139 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2140 } 2141 2142 // force reconfiguration of effect chains and engines to take new buffer size and audio 2143 // parameters into account 2144 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2145 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2146 // matter. 2147 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2148 Vector< sp<EffectChain> > effectChains = mEffectChains; 2149 for (size_t i = 0; i < effectChains.size(); i ++) { 2150 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2151 } 2152} 2153 2154 2155status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2156{ 2157 if (halFrames == NULL || dspFrames == NULL) { 2158 return BAD_VALUE; 2159 } 2160 Mutex::Autolock _l(mLock); 2161 if (initCheck() != NO_ERROR) { 2162 return INVALID_OPERATION; 2163 } 2164 size_t framesWritten = mBytesWritten / mFrameSize; 2165 *halFrames = framesWritten; 2166 2167 if (isSuspended()) { 2168 // return an estimation of rendered frames when the output is suspended 2169 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2170 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2171 return NO_ERROR; 2172 } else { 2173 status_t status; 2174 uint32_t frames; 2175 status = mOutput->getRenderPosition(&frames); 2176 *dspFrames = (size_t)frames; 2177 return status; 2178 } 2179} 2180 2181uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2182{ 2183 Mutex::Autolock _l(mLock); 2184 uint32_t result = 0; 2185 if (getEffectChain_l(sessionId) != 0) { 2186 result = EFFECT_SESSION; 2187 } 2188 2189 for (size_t i = 0; i < mTracks.size(); ++i) { 2190 sp<Track> track = mTracks[i]; 2191 if (sessionId == track->sessionId() && !track->isInvalid()) { 2192 result |= TRACK_SESSION; 2193 break; 2194 } 2195 } 2196 2197 return result; 2198} 2199 2200uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2201{ 2202 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2203 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2204 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2205 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2206 } 2207 for (size_t i = 0; i < mTracks.size(); i++) { 2208 sp<Track> track = mTracks[i]; 2209 if (sessionId == track->sessionId() && !track->isInvalid()) { 2210 return AudioSystem::getStrategyForStream(track->streamType()); 2211 } 2212 } 2213 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2214} 2215 2216 2217AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2218{ 2219 Mutex::Autolock _l(mLock); 2220 return mOutput; 2221} 2222 2223AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2224{ 2225 Mutex::Autolock _l(mLock); 2226 AudioStreamOut *output = mOutput; 2227 mOutput = NULL; 2228 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2229 // must push a NULL and wait for ack 2230 mOutputSink.clear(); 2231 mPipeSink.clear(); 2232 mNormalSink.clear(); 2233 return output; 2234} 2235 2236// this method must always be called either with ThreadBase mLock held or inside the thread loop 2237audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2238{ 2239 if (mOutput == NULL) { 2240 return NULL; 2241 } 2242 return &mOutput->stream->common; 2243} 2244 2245uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2246{ 2247 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2248} 2249 2250status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2251{ 2252 if (!isValidSyncEvent(event)) { 2253 return BAD_VALUE; 2254 } 2255 2256 Mutex::Autolock _l(mLock); 2257 2258 for (size_t i = 0; i < mTracks.size(); ++i) { 2259 sp<Track> track = mTracks[i]; 2260 if (event->triggerSession() == track->sessionId()) { 2261 (void) track->setSyncEvent(event); 2262 return NO_ERROR; 2263 } 2264 } 2265 2266 return NAME_NOT_FOUND; 2267} 2268 2269bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2270{ 2271 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2272} 2273 2274void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2275 const Vector< sp<Track> >& tracksToRemove) 2276{ 2277 size_t count = tracksToRemove.size(); 2278 if (count > 0) { 2279 for (size_t i = 0 ; i < count ; i++) { 2280 const sp<Track>& track = tracksToRemove.itemAt(i); 2281 if (track->isExternalTrack()) { 2282 AudioSystem::stopOutput(mId, track->streamType(), 2283 (audio_session_t)track->sessionId()); 2284#ifdef ADD_BATTERY_DATA 2285 // to track the speaker usage 2286 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2287#endif 2288 if (track->isTerminated()) { 2289 AudioSystem::releaseOutput(mId, track->streamType(), 2290 (audio_session_t)track->sessionId()); 2291 } 2292 } 2293 } 2294 } 2295} 2296 2297void AudioFlinger::PlaybackThread::checkSilentMode_l() 2298{ 2299 if (!mMasterMute) { 2300 char value[PROPERTY_VALUE_MAX]; 2301 if (property_get("ro.audio.silent", value, "0") > 0) { 2302 char *endptr; 2303 unsigned long ul = strtoul(value, &endptr, 0); 2304 if (*endptr == '\0' && ul != 0) { 2305 ALOGD("Silence is golden"); 2306 // The setprop command will not allow a property to be changed after 2307 // the first time it is set, so we don't have to worry about un-muting. 2308 setMasterMute_l(true); 2309 } 2310 } 2311 } 2312} 2313 2314// shared by MIXER and DIRECT, overridden by DUPLICATING 2315ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2316{ 2317 // FIXME rewrite to reduce number of system calls 2318 mLastWriteTime = systemTime(); 2319 mInWrite = true; 2320 ssize_t bytesWritten; 2321 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2322 2323 // If an NBAIO sink is present, use it to write the normal mixer's submix 2324 if (mNormalSink != 0) { 2325 2326 const size_t count = mBytesRemaining / mFrameSize; 2327 2328 ATRACE_BEGIN("write"); 2329 // update the setpoint when AudioFlinger::mScreenState changes 2330 uint32_t screenState = AudioFlinger::mScreenState; 2331 if (screenState != mScreenState) { 2332 mScreenState = screenState; 2333 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2334 if (pipe != NULL) { 2335 pipe->setAvgFrames((mScreenState & 1) ? 2336 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2337 } 2338 } 2339 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2340 ATRACE_END(); 2341 if (framesWritten > 0) { 2342 bytesWritten = framesWritten * mFrameSize; 2343 } else { 2344 bytesWritten = framesWritten; 2345 } 2346 mLatchDValid = false; 2347 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2348 if (status == NO_ERROR) { 2349 size_t totalFramesWritten = mNormalSink->framesWritten(); 2350 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2351 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2352 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2353 mLatchDValid = true; 2354 } 2355 } 2356 // otherwise use the HAL / AudioStreamOut directly 2357 } else { 2358 // Direct output and offload threads 2359 2360 if (mUseAsyncWrite) { 2361 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2362 mWriteAckSequence += 2; 2363 mWriteAckSequence |= 1; 2364 ALOG_ASSERT(mCallbackThread != 0); 2365 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2366 } 2367 // FIXME We should have an implementation of timestamps for direct output threads. 2368 // They are used e.g for multichannel PCM playback over HDMI. 2369 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2370 if (mUseAsyncWrite && 2371 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2372 // do not wait for async callback in case of error of full write 2373 mWriteAckSequence &= ~1; 2374 ALOG_ASSERT(mCallbackThread != 0); 2375 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2376 } 2377 } 2378 2379 mNumWrites++; 2380 mInWrite = false; 2381 mStandby = false; 2382 return bytesWritten; 2383} 2384 2385void AudioFlinger::PlaybackThread::threadLoop_drain() 2386{ 2387 if (mOutput->stream->drain) { 2388 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2389 if (mUseAsyncWrite) { 2390 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2391 mDrainSequence |= 1; 2392 ALOG_ASSERT(mCallbackThread != 0); 2393 mCallbackThread->setDraining(mDrainSequence); 2394 } 2395 mOutput->stream->drain(mOutput->stream, 2396 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2397 : AUDIO_DRAIN_ALL); 2398 } 2399} 2400 2401void AudioFlinger::PlaybackThread::threadLoop_exit() 2402{ 2403 { 2404 Mutex::Autolock _l(mLock); 2405 for (size_t i = 0; i < mTracks.size(); i++) { 2406 sp<Track> track = mTracks[i]; 2407 track->invalidate(); 2408 } 2409 } 2410} 2411 2412/* 2413The derived values that are cached: 2414 - mSinkBufferSize from frame count * frame size 2415 - activeSleepTime from activeSleepTimeUs() 2416 - idleSleepTime from idleSleepTimeUs() 2417 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2418 - maxPeriod from frame count and sample rate (MIXER only) 2419 2420The parameters that affect these derived values are: 2421 - frame count 2422 - frame size 2423 - sample rate 2424 - device type: A2DP or not 2425 - device latency 2426 - format: PCM or not 2427 - active sleep time 2428 - idle sleep time 2429*/ 2430 2431void AudioFlinger::PlaybackThread::cacheParameters_l() 2432{ 2433 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2434 activeSleepTime = activeSleepTimeUs(); 2435 idleSleepTime = idleSleepTimeUs(); 2436} 2437 2438void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2439{ 2440 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2441 this, streamType, mTracks.size()); 2442 Mutex::Autolock _l(mLock); 2443 2444 size_t size = mTracks.size(); 2445 for (size_t i = 0; i < size; i++) { 2446 sp<Track> t = mTracks[i]; 2447 if (t->streamType() == streamType) { 2448 t->invalidate(); 2449 } 2450 } 2451} 2452 2453status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2454{ 2455 int session = chain->sessionId(); 2456 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2457 ? mEffectBuffer : mSinkBuffer); 2458 bool ownsBuffer = false; 2459 2460 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2461 if (session > 0) { 2462 // Only one effect chain can be present in direct output thread and it uses 2463 // the sink buffer as input 2464 if (mType != DIRECT) { 2465 size_t numSamples = mNormalFrameCount * mChannelCount; 2466 buffer = new int16_t[numSamples]; 2467 memset(buffer, 0, numSamples * sizeof(int16_t)); 2468 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2469 ownsBuffer = true; 2470 } 2471 2472 // Attach all tracks with same session ID to this chain. 2473 for (size_t i = 0; i < mTracks.size(); ++i) { 2474 sp<Track> track = mTracks[i]; 2475 if (session == track->sessionId()) { 2476 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2477 buffer); 2478 track->setMainBuffer(buffer); 2479 chain->incTrackCnt(); 2480 } 2481 } 2482 2483 // indicate all active tracks in the chain 2484 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2485 sp<Track> track = mActiveTracks[i].promote(); 2486 if (track == 0) { 2487 continue; 2488 } 2489 if (session == track->sessionId()) { 2490 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2491 chain->incActiveTrackCnt(); 2492 } 2493 } 2494 } 2495 chain->setThread(this); 2496 chain->setInBuffer(buffer, ownsBuffer); 2497 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2498 ? mEffectBuffer : mSinkBuffer)); 2499 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2500 // chains list in order to be processed last as it contains output stage effects 2501 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2502 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2503 // after track specific effects and before output stage 2504 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2505 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2506 // Effect chain for other sessions are inserted at beginning of effect 2507 // chains list to be processed before output mix effects. Relative order between other 2508 // sessions is not important 2509 size_t size = mEffectChains.size(); 2510 size_t i = 0; 2511 for (i = 0; i < size; i++) { 2512 if (mEffectChains[i]->sessionId() < session) { 2513 break; 2514 } 2515 } 2516 mEffectChains.insertAt(chain, i); 2517 checkSuspendOnAddEffectChain_l(chain); 2518 2519 return NO_ERROR; 2520} 2521 2522size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2523{ 2524 int session = chain->sessionId(); 2525 2526 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2527 2528 for (size_t i = 0; i < mEffectChains.size(); i++) { 2529 if (chain == mEffectChains[i]) { 2530 mEffectChains.removeAt(i); 2531 // detach all active tracks from the chain 2532 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2533 sp<Track> track = mActiveTracks[i].promote(); 2534 if (track == 0) { 2535 continue; 2536 } 2537 if (session == track->sessionId()) { 2538 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2539 chain.get(), session); 2540 chain->decActiveTrackCnt(); 2541 } 2542 } 2543 2544 // detach all tracks with same session ID from this chain 2545 for (size_t i = 0; i < mTracks.size(); ++i) { 2546 sp<Track> track = mTracks[i]; 2547 if (session == track->sessionId()) { 2548 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2549 chain->decTrackCnt(); 2550 } 2551 } 2552 break; 2553 } 2554 } 2555 return mEffectChains.size(); 2556} 2557 2558status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2559 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2560{ 2561 Mutex::Autolock _l(mLock); 2562 return attachAuxEffect_l(track, EffectId); 2563} 2564 2565status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2566 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2567{ 2568 status_t status = NO_ERROR; 2569 2570 if (EffectId == 0) { 2571 track->setAuxBuffer(0, NULL); 2572 } else { 2573 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2574 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2575 if (effect != 0) { 2576 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2577 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2578 } else { 2579 status = INVALID_OPERATION; 2580 } 2581 } else { 2582 status = BAD_VALUE; 2583 } 2584 } 2585 return status; 2586} 2587 2588void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2589{ 2590 for (size_t i = 0; i < mTracks.size(); ++i) { 2591 sp<Track> track = mTracks[i]; 2592 if (track->auxEffectId() == effectId) { 2593 attachAuxEffect_l(track, 0); 2594 } 2595 } 2596} 2597 2598bool AudioFlinger::PlaybackThread::threadLoop() 2599{ 2600 Vector< sp<Track> > tracksToRemove; 2601 2602 standbyTime = systemTime(); 2603 2604 // MIXER 2605 nsecs_t lastWarning = 0; 2606 2607 // DUPLICATING 2608 // FIXME could this be made local to while loop? 2609 writeFrames = 0; 2610 2611 int lastGeneration = 0; 2612 2613 cacheParameters_l(); 2614 sleepTime = idleSleepTime; 2615 2616 if (mType == MIXER) { 2617 sleepTimeShift = 0; 2618 } 2619 2620 CpuStats cpuStats; 2621 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2622 2623 acquireWakeLock(); 2624 2625 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2626 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2627 // and then that string will be logged at the next convenient opportunity. 2628 const char *logString = NULL; 2629 2630 checkSilentMode_l(); 2631 2632 while (!exitPending()) 2633 { 2634 cpuStats.sample(myName); 2635 2636 Vector< sp<EffectChain> > effectChains; 2637 2638 { // scope for mLock 2639 2640 Mutex::Autolock _l(mLock); 2641 2642 processConfigEvents_l(); 2643 2644 if (logString != NULL) { 2645 mNBLogWriter->logTimestamp(); 2646 mNBLogWriter->log(logString); 2647 logString = NULL; 2648 } 2649 2650 // Gather the framesReleased counters for all active tracks, 2651 // and latch them atomically with the timestamp. 2652 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2653 mLatchD.mFramesReleased.clear(); 2654 size_t size = mActiveTracks.size(); 2655 for (size_t i = 0; i < size; i++) { 2656 sp<Track> t = mActiveTracks[i].promote(); 2657 if (t != 0) { 2658 mLatchD.mFramesReleased.add(t.get(), 2659 t->mAudioTrackServerProxy->framesReleased()); 2660 } 2661 } 2662 if (mLatchDValid) { 2663 mLatchQ = mLatchD; 2664 mLatchDValid = false; 2665 mLatchQValid = true; 2666 } 2667 2668 saveOutputTracks(); 2669 if (mSignalPending) { 2670 // A signal was raised while we were unlocked 2671 mSignalPending = false; 2672 } else if (waitingAsyncCallback_l()) { 2673 if (exitPending()) { 2674 break; 2675 } 2676 releaseWakeLock_l(); 2677 mWakeLockUids.clear(); 2678 mActiveTracksGeneration++; 2679 ALOGV("wait async completion"); 2680 mWaitWorkCV.wait(mLock); 2681 ALOGV("async completion/wake"); 2682 acquireWakeLock_l(); 2683 standbyTime = systemTime() + standbyDelay; 2684 sleepTime = 0; 2685 2686 continue; 2687 } 2688 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2689 isSuspended()) { 2690 // put audio hardware into standby after short delay 2691 if (shouldStandby_l()) { 2692 2693 threadLoop_standby(); 2694 2695 mStandby = true; 2696 } 2697 2698 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2699 // we're about to wait, flush the binder command buffer 2700 IPCThreadState::self()->flushCommands(); 2701 2702 clearOutputTracks(); 2703 2704 if (exitPending()) { 2705 break; 2706 } 2707 2708 releaseWakeLock_l(); 2709 mWakeLockUids.clear(); 2710 mActiveTracksGeneration++; 2711 // wait until we have something to do... 2712 ALOGV("%s going to sleep", myName.string()); 2713 mWaitWorkCV.wait(mLock); 2714 ALOGV("%s waking up", myName.string()); 2715 acquireWakeLock_l(); 2716 2717 mMixerStatus = MIXER_IDLE; 2718 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2719 mBytesWritten = 0; 2720 mBytesRemaining = 0; 2721 checkSilentMode_l(); 2722 2723 standbyTime = systemTime() + standbyDelay; 2724 sleepTime = idleSleepTime; 2725 if (mType == MIXER) { 2726 sleepTimeShift = 0; 2727 } 2728 2729 continue; 2730 } 2731 } 2732 // mMixerStatusIgnoringFastTracks is also updated internally 2733 mMixerStatus = prepareTracks_l(&tracksToRemove); 2734 2735 // compare with previously applied list 2736 if (lastGeneration != mActiveTracksGeneration) { 2737 // update wakelock 2738 updateWakeLockUids_l(mWakeLockUids); 2739 lastGeneration = mActiveTracksGeneration; 2740 } 2741 2742 // prevent any changes in effect chain list and in each effect chain 2743 // during mixing and effect process as the audio buffers could be deleted 2744 // or modified if an effect is created or deleted 2745 lockEffectChains_l(effectChains); 2746 } // mLock scope ends 2747 2748 if (mBytesRemaining == 0) { 2749 mCurrentWriteLength = 0; 2750 if (mMixerStatus == MIXER_TRACKS_READY) { 2751 // threadLoop_mix() sets mCurrentWriteLength 2752 threadLoop_mix(); 2753 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2754 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2755 // threadLoop_sleepTime sets sleepTime to 0 if data 2756 // must be written to HAL 2757 threadLoop_sleepTime(); 2758 if (sleepTime == 0) { 2759 mCurrentWriteLength = mSinkBufferSize; 2760 } 2761 } 2762 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2763 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2764 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2765 // or mSinkBuffer (if there are no effects). 2766 // 2767 // This is done pre-effects computation; if effects change to 2768 // support higher precision, this needs to move. 2769 // 2770 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2771 // TODO use sleepTime == 0 as an additional condition. 2772 if (mMixerBufferValid) { 2773 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2774 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2775 2776 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2777 mNormalFrameCount * mChannelCount); 2778 } 2779 2780 mBytesRemaining = mCurrentWriteLength; 2781 if (isSuspended()) { 2782 sleepTime = suspendSleepTimeUs(); 2783 // simulate write to HAL when suspended 2784 mBytesWritten += mSinkBufferSize; 2785 mBytesRemaining = 0; 2786 } 2787 2788 // only process effects if we're going to write 2789 if (sleepTime == 0 && mType != OFFLOAD) { 2790 for (size_t i = 0; i < effectChains.size(); i ++) { 2791 effectChains[i]->process_l(); 2792 } 2793 } 2794 } 2795 // Process effect chains for offloaded thread even if no audio 2796 // was read from audio track: process only updates effect state 2797 // and thus does have to be synchronized with audio writes but may have 2798 // to be called while waiting for async write callback 2799 if (mType == OFFLOAD) { 2800 for (size_t i = 0; i < effectChains.size(); i ++) { 2801 effectChains[i]->process_l(); 2802 } 2803 } 2804 2805 // Only if the Effects buffer is enabled and there is data in the 2806 // Effects buffer (buffer valid), we need to 2807 // copy into the sink buffer. 2808 // TODO use sleepTime == 0 as an additional condition. 2809 if (mEffectBufferValid) { 2810 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2811 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2812 mNormalFrameCount * mChannelCount); 2813 } 2814 2815 // enable changes in effect chain 2816 unlockEffectChains(effectChains); 2817 2818 if (!waitingAsyncCallback()) { 2819 // sleepTime == 0 means we must write to audio hardware 2820 if (sleepTime == 0) { 2821 if (mBytesRemaining) { 2822 ssize_t ret = threadLoop_write(); 2823 if (ret < 0) { 2824 mBytesRemaining = 0; 2825 } else { 2826 mBytesWritten += ret; 2827 mBytesRemaining -= ret; 2828 } 2829 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2830 (mMixerStatus == MIXER_DRAIN_ALL)) { 2831 threadLoop_drain(); 2832 } 2833 if (mType == MIXER) { 2834 // write blocked detection 2835 nsecs_t now = systemTime(); 2836 nsecs_t delta = now - mLastWriteTime; 2837 if (!mStandby && delta > maxPeriod) { 2838 mNumDelayedWrites++; 2839 if ((now - lastWarning) > kWarningThrottleNs) { 2840 ATRACE_NAME("underrun"); 2841 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2842 ns2ms(delta), mNumDelayedWrites, this); 2843 lastWarning = now; 2844 } 2845 } 2846 } 2847 2848 } else { 2849 ATRACE_BEGIN("sleep"); 2850 usleep(sleepTime); 2851 ATRACE_END(); 2852 } 2853 } 2854 2855 // Finally let go of removed track(s), without the lock held 2856 // since we can't guarantee the destructors won't acquire that 2857 // same lock. This will also mutate and push a new fast mixer state. 2858 threadLoop_removeTracks(tracksToRemove); 2859 tracksToRemove.clear(); 2860 2861 // FIXME I don't understand the need for this here; 2862 // it was in the original code but maybe the 2863 // assignment in saveOutputTracks() makes this unnecessary? 2864 clearOutputTracks(); 2865 2866 // Effect chains will be actually deleted here if they were removed from 2867 // mEffectChains list during mixing or effects processing 2868 effectChains.clear(); 2869 2870 // FIXME Note that the above .clear() is no longer necessary since effectChains 2871 // is now local to this block, but will keep it for now (at least until merge done). 2872 } 2873 2874 threadLoop_exit(); 2875 2876 if (!mStandby) { 2877 threadLoop_standby(); 2878 mStandby = true; 2879 } 2880 2881 releaseWakeLock(); 2882 mWakeLockUids.clear(); 2883 mActiveTracksGeneration++; 2884 2885 ALOGV("Thread %p type %d exiting", this, mType); 2886 return false; 2887} 2888 2889// removeTracks_l() must be called with ThreadBase::mLock held 2890void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2891{ 2892 size_t count = tracksToRemove.size(); 2893 if (count > 0) { 2894 for (size_t i=0 ; i<count ; i++) { 2895 const sp<Track>& track = tracksToRemove.itemAt(i); 2896 mActiveTracks.remove(track); 2897 mWakeLockUids.remove(track->uid()); 2898 mActiveTracksGeneration++; 2899 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2900 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2901 if (chain != 0) { 2902 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2903 track->sessionId()); 2904 chain->decActiveTrackCnt(); 2905 } 2906 if (track->isTerminated()) { 2907 removeTrack_l(track); 2908 } 2909 } 2910 } 2911 2912} 2913 2914status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2915{ 2916 if (mNormalSink != 0) { 2917 return mNormalSink->getTimestamp(timestamp); 2918 } 2919 if ((mType == OFFLOAD || mType == DIRECT) 2920 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2921 uint64_t position64; 2922 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 2923 if (ret == 0) { 2924 timestamp.mPosition = (uint32_t)position64; 2925 return NO_ERROR; 2926 } 2927 } 2928 return INVALID_OPERATION; 2929} 2930 2931status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2932 audio_patch_handle_t *handle) 2933{ 2934 status_t status = NO_ERROR; 2935 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2936 // store new device and send to effects 2937 audio_devices_t type = AUDIO_DEVICE_NONE; 2938 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2939 type |= patch->sinks[i].ext.device.type; 2940 } 2941 mOutDevice = type; 2942 for (size_t i = 0; i < mEffectChains.size(); i++) { 2943 mEffectChains[i]->setDevice_l(mOutDevice); 2944 } 2945 2946 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2947 status = hwDevice->create_audio_patch(hwDevice, 2948 patch->num_sources, 2949 patch->sources, 2950 patch->num_sinks, 2951 patch->sinks, 2952 handle); 2953 } else { 2954 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2955 } 2956 return status; 2957} 2958 2959status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2960{ 2961 status_t status = NO_ERROR; 2962 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2963 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2964 status = hwDevice->release_audio_patch(hwDevice, handle); 2965 } else { 2966 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2967 } 2968 return status; 2969} 2970 2971void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2972{ 2973 Mutex::Autolock _l(mLock); 2974 mTracks.add(track); 2975} 2976 2977void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2978{ 2979 Mutex::Autolock _l(mLock); 2980 destroyTrack_l(track); 2981} 2982 2983void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2984{ 2985 ThreadBase::getAudioPortConfig(config); 2986 config->role = AUDIO_PORT_ROLE_SOURCE; 2987 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2988 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2989} 2990 2991// ---------------------------------------------------------------------------- 2992 2993AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2994 audio_io_handle_t id, audio_devices_t device, type_t type) 2995 : PlaybackThread(audioFlinger, output, id, device, type), 2996 // mAudioMixer below 2997 // mFastMixer below 2998 mFastMixerFutex(0) 2999 // mOutputSink below 3000 // mPipeSink below 3001 // mNormalSink below 3002{ 3003 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3004 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3005 "mFrameCount=%d, mNormalFrameCount=%d", 3006 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3007 mNormalFrameCount); 3008 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3009 3010 if (type == DUPLICATING) { 3011 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3012 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3013 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3014 return; 3015 } 3016 // create an NBAIO sink for the HAL output stream, and negotiate 3017 mOutputSink = new AudioStreamOutSink(output->stream); 3018 size_t numCounterOffers = 0; 3019 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3020 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3021 ALOG_ASSERT(index == 0); 3022 3023 // initialize fast mixer depending on configuration 3024 bool initFastMixer; 3025 switch (kUseFastMixer) { 3026 case FastMixer_Never: 3027 initFastMixer = false; 3028 break; 3029 case FastMixer_Always: 3030 initFastMixer = true; 3031 break; 3032 case FastMixer_Static: 3033 case FastMixer_Dynamic: 3034 initFastMixer = mFrameCount < mNormalFrameCount; 3035 break; 3036 } 3037 if (initFastMixer) { 3038 audio_format_t fastMixerFormat; 3039 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3040 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3041 } else { 3042 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3043 } 3044 if (mFormat != fastMixerFormat) { 3045 // change our Sink format to accept our intermediate precision 3046 mFormat = fastMixerFormat; 3047 free(mSinkBuffer); 3048 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3049 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3050 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3051 } 3052 3053 // create a MonoPipe to connect our submix to FastMixer 3054 NBAIO_Format format = mOutputSink->format(); 3055 NBAIO_Format origformat = format; 3056 // adjust format to match that of the Fast Mixer 3057 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3058 format.mFormat = fastMixerFormat; 3059 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3060 3061 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3062 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3063 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3064 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3065 const NBAIO_Format offers[1] = {format}; 3066 size_t numCounterOffers = 0; 3067 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3068 ALOG_ASSERT(index == 0); 3069 monoPipe->setAvgFrames((mScreenState & 1) ? 3070 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3071 mPipeSink = monoPipe; 3072 3073#ifdef TEE_SINK 3074 if (mTeeSinkOutputEnabled) { 3075 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3076 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3077 const NBAIO_Format offers2[1] = {origformat}; 3078 numCounterOffers = 0; 3079 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3080 ALOG_ASSERT(index == 0); 3081 mTeeSink = teeSink; 3082 PipeReader *teeSource = new PipeReader(*teeSink); 3083 numCounterOffers = 0; 3084 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3085 ALOG_ASSERT(index == 0); 3086 mTeeSource = teeSource; 3087 } 3088#endif 3089 3090 // create fast mixer and configure it initially with just one fast track for our submix 3091 mFastMixer = new FastMixer(); 3092 FastMixerStateQueue *sq = mFastMixer->sq(); 3093#ifdef STATE_QUEUE_DUMP 3094 sq->setObserverDump(&mStateQueueObserverDump); 3095 sq->setMutatorDump(&mStateQueueMutatorDump); 3096#endif 3097 FastMixerState *state = sq->begin(); 3098 FastTrack *fastTrack = &state->mFastTracks[0]; 3099 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3100 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3101 fastTrack->mVolumeProvider = NULL; 3102 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3103 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3104 fastTrack->mGeneration++; 3105 state->mFastTracksGen++; 3106 state->mTrackMask = 1; 3107 // fast mixer will use the HAL output sink 3108 state->mOutputSink = mOutputSink.get(); 3109 state->mOutputSinkGen++; 3110 state->mFrameCount = mFrameCount; 3111 state->mCommand = FastMixerState::COLD_IDLE; 3112 // already done in constructor initialization list 3113 //mFastMixerFutex = 0; 3114 state->mColdFutexAddr = &mFastMixerFutex; 3115 state->mColdGen++; 3116 state->mDumpState = &mFastMixerDumpState; 3117#ifdef TEE_SINK 3118 state->mTeeSink = mTeeSink.get(); 3119#endif 3120 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3121 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3122 sq->end(); 3123 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3124 3125 // start the fast mixer 3126 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3127 pid_t tid = mFastMixer->getTid(); 3128 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3129 if (err != 0) { 3130 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3131 kPriorityFastMixer, getpid_cached, tid, err); 3132 } 3133 3134#ifdef AUDIO_WATCHDOG 3135 // create and start the watchdog 3136 mAudioWatchdog = new AudioWatchdog(); 3137 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3138 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3139 tid = mAudioWatchdog->getTid(); 3140 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3141 if (err != 0) { 3142 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3143 kPriorityFastMixer, getpid_cached, tid, err); 3144 } 3145#endif 3146 3147 } 3148 3149 switch (kUseFastMixer) { 3150 case FastMixer_Never: 3151 case FastMixer_Dynamic: 3152 mNormalSink = mOutputSink; 3153 break; 3154 case FastMixer_Always: 3155 mNormalSink = mPipeSink; 3156 break; 3157 case FastMixer_Static: 3158 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3159 break; 3160 } 3161} 3162 3163AudioFlinger::MixerThread::~MixerThread() 3164{ 3165 if (mFastMixer != 0) { 3166 FastMixerStateQueue *sq = mFastMixer->sq(); 3167 FastMixerState *state = sq->begin(); 3168 if (state->mCommand == FastMixerState::COLD_IDLE) { 3169 int32_t old = android_atomic_inc(&mFastMixerFutex); 3170 if (old == -1) { 3171 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3172 } 3173 } 3174 state->mCommand = FastMixerState::EXIT; 3175 sq->end(); 3176 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3177 mFastMixer->join(); 3178 // Though the fast mixer thread has exited, it's state queue is still valid. 3179 // We'll use that extract the final state which contains one remaining fast track 3180 // corresponding to our sub-mix. 3181 state = sq->begin(); 3182 ALOG_ASSERT(state->mTrackMask == 1); 3183 FastTrack *fastTrack = &state->mFastTracks[0]; 3184 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3185 delete fastTrack->mBufferProvider; 3186 sq->end(false /*didModify*/); 3187 mFastMixer.clear(); 3188#ifdef AUDIO_WATCHDOG 3189 if (mAudioWatchdog != 0) { 3190 mAudioWatchdog->requestExit(); 3191 mAudioWatchdog->requestExitAndWait(); 3192 mAudioWatchdog.clear(); 3193 } 3194#endif 3195 } 3196 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3197 delete mAudioMixer; 3198} 3199 3200 3201uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3202{ 3203 if (mFastMixer != 0) { 3204 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3205 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3206 } 3207 return latency; 3208} 3209 3210 3211void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3212{ 3213 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3214} 3215 3216ssize_t AudioFlinger::MixerThread::threadLoop_write() 3217{ 3218 // FIXME we should only do one push per cycle; confirm this is true 3219 // Start the fast mixer if it's not already running 3220 if (mFastMixer != 0) { 3221 FastMixerStateQueue *sq = mFastMixer->sq(); 3222 FastMixerState *state = sq->begin(); 3223 if (state->mCommand != FastMixerState::MIX_WRITE && 3224 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3225 if (state->mCommand == FastMixerState::COLD_IDLE) { 3226 int32_t old = android_atomic_inc(&mFastMixerFutex); 3227 if (old == -1) { 3228 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3229 } 3230#ifdef AUDIO_WATCHDOG 3231 if (mAudioWatchdog != 0) { 3232 mAudioWatchdog->resume(); 3233 } 3234#endif 3235 } 3236 state->mCommand = FastMixerState::MIX_WRITE; 3237#ifdef FAST_THREAD_STATISTICS 3238 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3239 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3240#endif 3241 sq->end(); 3242 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3243 if (kUseFastMixer == FastMixer_Dynamic) { 3244 mNormalSink = mPipeSink; 3245 } 3246 } else { 3247 sq->end(false /*didModify*/); 3248 } 3249 } 3250 return PlaybackThread::threadLoop_write(); 3251} 3252 3253void AudioFlinger::MixerThread::threadLoop_standby() 3254{ 3255 // Idle the fast mixer if it's currently running 3256 if (mFastMixer != 0) { 3257 FastMixerStateQueue *sq = mFastMixer->sq(); 3258 FastMixerState *state = sq->begin(); 3259 if (!(state->mCommand & FastMixerState::IDLE)) { 3260 state->mCommand = FastMixerState::COLD_IDLE; 3261 state->mColdFutexAddr = &mFastMixerFutex; 3262 state->mColdGen++; 3263 mFastMixerFutex = 0; 3264 sq->end(); 3265 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3266 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3267 if (kUseFastMixer == FastMixer_Dynamic) { 3268 mNormalSink = mOutputSink; 3269 } 3270#ifdef AUDIO_WATCHDOG 3271 if (mAudioWatchdog != 0) { 3272 mAudioWatchdog->pause(); 3273 } 3274#endif 3275 } else { 3276 sq->end(false /*didModify*/); 3277 } 3278 } 3279 PlaybackThread::threadLoop_standby(); 3280} 3281 3282bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3283{ 3284 return false; 3285} 3286 3287bool AudioFlinger::PlaybackThread::shouldStandby_l() 3288{ 3289 return !mStandby; 3290} 3291 3292bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3293{ 3294 Mutex::Autolock _l(mLock); 3295 return waitingAsyncCallback_l(); 3296} 3297 3298// shared by MIXER and DIRECT, overridden by DUPLICATING 3299void AudioFlinger::PlaybackThread::threadLoop_standby() 3300{ 3301 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3302 mOutput->standby(); 3303 if (mUseAsyncWrite != 0) { 3304 // discard any pending drain or write ack by incrementing sequence 3305 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3306 mDrainSequence = (mDrainSequence + 2) & ~1; 3307 ALOG_ASSERT(mCallbackThread != 0); 3308 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3309 mCallbackThread->setDraining(mDrainSequence); 3310 } 3311 mHwPaused = false; 3312} 3313 3314void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3315{ 3316 ALOGV("signal playback thread"); 3317 broadcast_l(); 3318} 3319 3320void AudioFlinger::MixerThread::threadLoop_mix() 3321{ 3322 // obtain the presentation timestamp of the next output buffer 3323 int64_t pts; 3324 status_t status = INVALID_OPERATION; 3325 3326 if (mNormalSink != 0) { 3327 status = mNormalSink->getNextWriteTimestamp(&pts); 3328 } else { 3329 status = mOutputSink->getNextWriteTimestamp(&pts); 3330 } 3331 3332 if (status != NO_ERROR) { 3333 pts = AudioBufferProvider::kInvalidPTS; 3334 } 3335 3336 // mix buffers... 3337 mAudioMixer->process(pts); 3338 mCurrentWriteLength = mSinkBufferSize; 3339 // increase sleep time progressively when application underrun condition clears. 3340 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3341 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3342 // such that we would underrun the audio HAL. 3343 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3344 sleepTimeShift--; 3345 } 3346 sleepTime = 0; 3347 standbyTime = systemTime() + standbyDelay; 3348 //TODO: delay standby when effects have a tail 3349 3350} 3351 3352void AudioFlinger::MixerThread::threadLoop_sleepTime() 3353{ 3354 // If no tracks are ready, sleep once for the duration of an output 3355 // buffer size, then write 0s to the output 3356 if (sleepTime == 0) { 3357 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3358 sleepTime = activeSleepTime >> sleepTimeShift; 3359 if (sleepTime < kMinThreadSleepTimeUs) { 3360 sleepTime = kMinThreadSleepTimeUs; 3361 } 3362 // reduce sleep time in case of consecutive application underruns to avoid 3363 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3364 // duration we would end up writing less data than needed by the audio HAL if 3365 // the condition persists. 3366 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3367 sleepTimeShift++; 3368 } 3369 } else { 3370 sleepTime = idleSleepTime; 3371 } 3372 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3373 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3374 // before effects processing or output. 3375 if (mMixerBufferValid) { 3376 memset(mMixerBuffer, 0, mMixerBufferSize); 3377 } else { 3378 memset(mSinkBuffer, 0, mSinkBufferSize); 3379 } 3380 sleepTime = 0; 3381 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3382 "anticipated start"); 3383 } 3384 // TODO add standby time extension fct of effect tail 3385} 3386 3387// prepareTracks_l() must be called with ThreadBase::mLock held 3388AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3389 Vector< sp<Track> > *tracksToRemove) 3390{ 3391 3392 mixer_state mixerStatus = MIXER_IDLE; 3393 // find out which tracks need to be processed 3394 size_t count = mActiveTracks.size(); 3395 size_t mixedTracks = 0; 3396 size_t tracksWithEffect = 0; 3397 // counts only _active_ fast tracks 3398 size_t fastTracks = 0; 3399 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3400 3401 float masterVolume = mMasterVolume; 3402 bool masterMute = mMasterMute; 3403 3404 if (masterMute) { 3405 masterVolume = 0; 3406 } 3407 // Delegate master volume control to effect in output mix effect chain if needed 3408 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3409 if (chain != 0) { 3410 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3411 chain->setVolume_l(&v, &v); 3412 masterVolume = (float)((v + (1 << 23)) >> 24); 3413 chain.clear(); 3414 } 3415 3416 // prepare a new state to push 3417 FastMixerStateQueue *sq = NULL; 3418 FastMixerState *state = NULL; 3419 bool didModify = false; 3420 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3421 if (mFastMixer != 0) { 3422 sq = mFastMixer->sq(); 3423 state = sq->begin(); 3424 } 3425 3426 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3427 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3428 3429 for (size_t i=0 ; i<count ; i++) { 3430 const sp<Track> t = mActiveTracks[i].promote(); 3431 if (t == 0) { 3432 continue; 3433 } 3434 3435 // this const just means the local variable doesn't change 3436 Track* const track = t.get(); 3437 3438 // process fast tracks 3439 if (track->isFastTrack()) { 3440 3441 // It's theoretically possible (though unlikely) for a fast track to be created 3442 // and then removed within the same normal mix cycle. This is not a problem, as 3443 // the track never becomes active so it's fast mixer slot is never touched. 3444 // The converse, of removing an (active) track and then creating a new track 3445 // at the identical fast mixer slot within the same normal mix cycle, 3446 // is impossible because the slot isn't marked available until the end of each cycle. 3447 int j = track->mFastIndex; 3448 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3449 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3450 FastTrack *fastTrack = &state->mFastTracks[j]; 3451 3452 // Determine whether the track is currently in underrun condition, 3453 // and whether it had a recent underrun. 3454 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3455 FastTrackUnderruns underruns = ftDump->mUnderruns; 3456 uint32_t recentFull = (underruns.mBitFields.mFull - 3457 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3458 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3459 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3460 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3461 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3462 uint32_t recentUnderruns = recentPartial + recentEmpty; 3463 track->mObservedUnderruns = underruns; 3464 // don't count underruns that occur while stopping or pausing 3465 // or stopped which can occur when flush() is called while active 3466 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3467 recentUnderruns > 0) { 3468 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3469 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3470 } 3471 3472 // This is similar to the state machine for normal tracks, 3473 // with a few modifications for fast tracks. 3474 bool isActive = true; 3475 switch (track->mState) { 3476 case TrackBase::STOPPING_1: 3477 // track stays active in STOPPING_1 state until first underrun 3478 if (recentUnderruns > 0 || track->isTerminated()) { 3479 track->mState = TrackBase::STOPPING_2; 3480 } 3481 break; 3482 case TrackBase::PAUSING: 3483 // ramp down is not yet implemented 3484 track->setPaused(); 3485 break; 3486 case TrackBase::RESUMING: 3487 // ramp up is not yet implemented 3488 track->mState = TrackBase::ACTIVE; 3489 break; 3490 case TrackBase::ACTIVE: 3491 if (recentFull > 0 || recentPartial > 0) { 3492 // track has provided at least some frames recently: reset retry count 3493 track->mRetryCount = kMaxTrackRetries; 3494 } 3495 if (recentUnderruns == 0) { 3496 // no recent underruns: stay active 3497 break; 3498 } 3499 // there has recently been an underrun of some kind 3500 if (track->sharedBuffer() == 0) { 3501 // were any of the recent underruns "empty" (no frames available)? 3502 if (recentEmpty == 0) { 3503 // no, then ignore the partial underruns as they are allowed indefinitely 3504 break; 3505 } 3506 // there has recently been an "empty" underrun: decrement the retry counter 3507 if (--(track->mRetryCount) > 0) { 3508 break; 3509 } 3510 // indicate to client process that the track was disabled because of underrun; 3511 // it will then automatically call start() when data is available 3512 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3513 // remove from active list, but state remains ACTIVE [confusing but true] 3514 isActive = false; 3515 break; 3516 } 3517 // fall through 3518 case TrackBase::STOPPING_2: 3519 case TrackBase::PAUSED: 3520 case TrackBase::STOPPED: 3521 case TrackBase::FLUSHED: // flush() while active 3522 // Check for presentation complete if track is inactive 3523 // We have consumed all the buffers of this track. 3524 // This would be incomplete if we auto-paused on underrun 3525 { 3526 size_t audioHALFrames = 3527 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3528 size_t framesWritten = mBytesWritten / mFrameSize; 3529 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3530 // track stays in active list until presentation is complete 3531 break; 3532 } 3533 } 3534 if (track->isStopping_2()) { 3535 track->mState = TrackBase::STOPPED; 3536 } 3537 if (track->isStopped()) { 3538 // Can't reset directly, as fast mixer is still polling this track 3539 // track->reset(); 3540 // So instead mark this track as needing to be reset after push with ack 3541 resetMask |= 1 << i; 3542 } 3543 isActive = false; 3544 break; 3545 case TrackBase::IDLE: 3546 default: 3547 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3548 } 3549 3550 if (isActive) { 3551 // was it previously inactive? 3552 if (!(state->mTrackMask & (1 << j))) { 3553 ExtendedAudioBufferProvider *eabp = track; 3554 VolumeProvider *vp = track; 3555 fastTrack->mBufferProvider = eabp; 3556 fastTrack->mVolumeProvider = vp; 3557 fastTrack->mChannelMask = track->mChannelMask; 3558 fastTrack->mFormat = track->mFormat; 3559 fastTrack->mGeneration++; 3560 state->mTrackMask |= 1 << j; 3561 didModify = true; 3562 // no acknowledgement required for newly active tracks 3563 } 3564 // cache the combined master volume and stream type volume for fast mixer; this 3565 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3566 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3567 ++fastTracks; 3568 } else { 3569 // was it previously active? 3570 if (state->mTrackMask & (1 << j)) { 3571 fastTrack->mBufferProvider = NULL; 3572 fastTrack->mGeneration++; 3573 state->mTrackMask &= ~(1 << j); 3574 didModify = true; 3575 // If any fast tracks were removed, we must wait for acknowledgement 3576 // because we're about to decrement the last sp<> on those tracks. 3577 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3578 } else { 3579 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3580 } 3581 tracksToRemove->add(track); 3582 // Avoids a misleading display in dumpsys 3583 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3584 } 3585 continue; 3586 } 3587 3588 { // local variable scope to avoid goto warning 3589 3590 audio_track_cblk_t* cblk = track->cblk(); 3591 3592 // The first time a track is added we wait 3593 // for all its buffers to be filled before processing it 3594 int name = track->name(); 3595 // make sure that we have enough frames to mix one full buffer. 3596 // enforce this condition only once to enable draining the buffer in case the client 3597 // app does not call stop() and relies on underrun to stop: 3598 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3599 // during last round 3600 size_t desiredFrames; 3601 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3602 float speed, pitch; 3603 track->mAudioTrackServerProxy->getPlaybackRate(&speed, &pitch); 3604 3605 desiredFrames = sourceFramesNeededWithTimestretch( 3606 sampleRate, mNormalFrameCount, mSampleRate, speed); 3607 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3608 // add frames already consumed but not yet released by the resampler 3609 // because mAudioTrackServerProxy->framesReady() will include these frames 3610 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3611 3612 uint32_t minFrames = 1; 3613 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3614 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3615 minFrames = desiredFrames; 3616 } 3617 3618 size_t framesReady = track->framesReady(); 3619 if (ATRACE_ENABLED()) { 3620 // I wish we had formatted trace names 3621 char traceName[16]; 3622 strcpy(traceName, "nRdy"); 3623 int name = track->name(); 3624 if (AudioMixer::TRACK0 <= name && 3625 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3626 name -= AudioMixer::TRACK0; 3627 traceName[4] = (name / 10) + '0'; 3628 traceName[5] = (name % 10) + '0'; 3629 } else { 3630 traceName[4] = '?'; 3631 traceName[5] = '?'; 3632 } 3633 traceName[6] = '\0'; 3634 ATRACE_INT(traceName, framesReady); 3635 } 3636 if ((framesReady >= minFrames) && track->isReady() && 3637 !track->isPaused() && !track->isTerminated()) 3638 { 3639 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3640 3641 mixedTracks++; 3642 3643 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3644 // there is an effect chain connected to the track 3645 chain.clear(); 3646 if (track->mainBuffer() != mSinkBuffer && 3647 track->mainBuffer() != mMixerBuffer) { 3648 if (mEffectBufferEnabled) { 3649 mEffectBufferValid = true; // Later can set directly. 3650 } 3651 chain = getEffectChain_l(track->sessionId()); 3652 // Delegate volume control to effect in track effect chain if needed 3653 if (chain != 0) { 3654 tracksWithEffect++; 3655 } else { 3656 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3657 "session %d", 3658 name, track->sessionId()); 3659 } 3660 } 3661 3662 3663 int param = AudioMixer::VOLUME; 3664 if (track->mFillingUpStatus == Track::FS_FILLED) { 3665 // no ramp for the first volume setting 3666 track->mFillingUpStatus = Track::FS_ACTIVE; 3667 if (track->mState == TrackBase::RESUMING) { 3668 track->mState = TrackBase::ACTIVE; 3669 param = AudioMixer::RAMP_VOLUME; 3670 } 3671 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3672 // FIXME should not make a decision based on mServer 3673 } else if (cblk->mServer != 0) { 3674 // If the track is stopped before the first frame was mixed, 3675 // do not apply ramp 3676 param = AudioMixer::RAMP_VOLUME; 3677 } 3678 3679 // compute volume for this track 3680 uint32_t vl, vr; // in U8.24 integer format 3681 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3682 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3683 vl = vr = 0; 3684 vlf = vrf = vaf = 0.; 3685 if (track->isPausing()) { 3686 track->setPaused(); 3687 } 3688 } else { 3689 3690 // read original volumes with volume control 3691 float typeVolume = mStreamTypes[track->streamType()].volume; 3692 float v = masterVolume * typeVolume; 3693 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3694 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3695 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3696 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3697 // track volumes come from shared memory, so can't be trusted and must be clamped 3698 if (vlf > GAIN_FLOAT_UNITY) { 3699 ALOGV("Track left volume out of range: %.3g", vlf); 3700 vlf = GAIN_FLOAT_UNITY; 3701 } 3702 if (vrf > GAIN_FLOAT_UNITY) { 3703 ALOGV("Track right volume out of range: %.3g", vrf); 3704 vrf = GAIN_FLOAT_UNITY; 3705 } 3706 // now apply the master volume and stream type volume 3707 vlf *= v; 3708 vrf *= v; 3709 // assuming master volume and stream type volume each go up to 1.0, 3710 // then derive vl and vr as U8.24 versions for the effect chain 3711 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3712 vl = (uint32_t) (scaleto8_24 * vlf); 3713 vr = (uint32_t) (scaleto8_24 * vrf); 3714 // vl and vr are now in U8.24 format 3715 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3716 // send level comes from shared memory and so may be corrupt 3717 if (sendLevel > MAX_GAIN_INT) { 3718 ALOGV("Track send level out of range: %04X", sendLevel); 3719 sendLevel = MAX_GAIN_INT; 3720 } 3721 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3722 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3723 } 3724 3725 // Delegate volume control to effect in track effect chain if needed 3726 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3727 // Do not ramp volume if volume is controlled by effect 3728 param = AudioMixer::VOLUME; 3729 // Update remaining floating point volume levels 3730 vlf = (float)vl / (1 << 24); 3731 vrf = (float)vr / (1 << 24); 3732 track->mHasVolumeController = true; 3733 } else { 3734 // force no volume ramp when volume controller was just disabled or removed 3735 // from effect chain to avoid volume spike 3736 if (track->mHasVolumeController) { 3737 param = AudioMixer::VOLUME; 3738 } 3739 track->mHasVolumeController = false; 3740 } 3741 3742 // XXX: these things DON'T need to be done each time 3743 mAudioMixer->setBufferProvider(name, track); 3744 mAudioMixer->enable(name); 3745 3746 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3747 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3748 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3749 mAudioMixer->setParameter( 3750 name, 3751 AudioMixer::TRACK, 3752 AudioMixer::FORMAT, (void *)track->format()); 3753 mAudioMixer->setParameter( 3754 name, 3755 AudioMixer::TRACK, 3756 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3757 mAudioMixer->setParameter( 3758 name, 3759 AudioMixer::TRACK, 3760 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3761 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3762 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3763 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3764 if (reqSampleRate == 0) { 3765 reqSampleRate = mSampleRate; 3766 } else if (reqSampleRate > maxSampleRate) { 3767 reqSampleRate = maxSampleRate; 3768 } 3769 mAudioMixer->setParameter( 3770 name, 3771 AudioMixer::RESAMPLE, 3772 AudioMixer::SAMPLE_RATE, 3773 (void *)(uintptr_t)reqSampleRate); 3774 3775 // set the playback rate as an float array {speed, pitch} 3776 float playbackRate[2]; 3777 track->mAudioTrackServerProxy->getPlaybackRate( 3778 &playbackRate[0] /*speed*/, &playbackRate[1] /*pitch*/); 3779 mAudioMixer->setParameter( 3780 name, 3781 AudioMixer::TIMESTRETCH, 3782 AudioMixer::PLAYBACK_RATE, 3783 playbackRate); 3784 3785 /* 3786 * Select the appropriate output buffer for the track. 3787 * 3788 * Tracks with effects go into their own effects chain buffer 3789 * and from there into either mEffectBuffer or mSinkBuffer. 3790 * 3791 * Other tracks can use mMixerBuffer for higher precision 3792 * channel accumulation. If this buffer is enabled 3793 * (mMixerBufferEnabled true), then selected tracks will accumulate 3794 * into it. 3795 * 3796 */ 3797 if (mMixerBufferEnabled 3798 && (track->mainBuffer() == mSinkBuffer 3799 || track->mainBuffer() == mMixerBuffer)) { 3800 mAudioMixer->setParameter( 3801 name, 3802 AudioMixer::TRACK, 3803 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3804 mAudioMixer->setParameter( 3805 name, 3806 AudioMixer::TRACK, 3807 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3808 // TODO: override track->mainBuffer()? 3809 mMixerBufferValid = true; 3810 } else { 3811 mAudioMixer->setParameter( 3812 name, 3813 AudioMixer::TRACK, 3814 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3815 mAudioMixer->setParameter( 3816 name, 3817 AudioMixer::TRACK, 3818 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3819 } 3820 mAudioMixer->setParameter( 3821 name, 3822 AudioMixer::TRACK, 3823 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3824 3825 // reset retry count 3826 track->mRetryCount = kMaxTrackRetries; 3827 3828 // If one track is ready, set the mixer ready if: 3829 // - the mixer was not ready during previous round OR 3830 // - no other track is not ready 3831 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3832 mixerStatus != MIXER_TRACKS_ENABLED) { 3833 mixerStatus = MIXER_TRACKS_READY; 3834 } 3835 } else { 3836 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3837 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3838 } 3839 // clear effect chain input buffer if an active track underruns to avoid sending 3840 // previous audio buffer again to effects 3841 chain = getEffectChain_l(track->sessionId()); 3842 if (chain != 0) { 3843 chain->clearInputBuffer(); 3844 } 3845 3846 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3847 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3848 track->isStopped() || track->isPaused()) { 3849 // We have consumed all the buffers of this track. 3850 // Remove it from the list of active tracks. 3851 // TODO: use actual buffer filling status instead of latency when available from 3852 // audio HAL 3853 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3854 size_t framesWritten = mBytesWritten / mFrameSize; 3855 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3856 if (track->isStopped()) { 3857 track->reset(); 3858 } 3859 tracksToRemove->add(track); 3860 } 3861 } else { 3862 // No buffers for this track. Give it a few chances to 3863 // fill a buffer, then remove it from active list. 3864 if (--(track->mRetryCount) <= 0) { 3865 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3866 tracksToRemove->add(track); 3867 // indicate to client process that the track was disabled because of underrun; 3868 // it will then automatically call start() when data is available 3869 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3870 // If one track is not ready, mark the mixer also not ready if: 3871 // - the mixer was ready during previous round OR 3872 // - no other track is ready 3873 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3874 mixerStatus != MIXER_TRACKS_READY) { 3875 mixerStatus = MIXER_TRACKS_ENABLED; 3876 } 3877 } 3878 mAudioMixer->disable(name); 3879 } 3880 3881 } // local variable scope to avoid goto warning 3882track_is_ready: ; 3883 3884 } 3885 3886 // Push the new FastMixer state if necessary 3887 bool pauseAudioWatchdog = false; 3888 if (didModify) { 3889 state->mFastTracksGen++; 3890 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3891 if (kUseFastMixer == FastMixer_Dynamic && 3892 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3893 state->mCommand = FastMixerState::COLD_IDLE; 3894 state->mColdFutexAddr = &mFastMixerFutex; 3895 state->mColdGen++; 3896 mFastMixerFutex = 0; 3897 if (kUseFastMixer == FastMixer_Dynamic) { 3898 mNormalSink = mOutputSink; 3899 } 3900 // If we go into cold idle, need to wait for acknowledgement 3901 // so that fast mixer stops doing I/O. 3902 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3903 pauseAudioWatchdog = true; 3904 } 3905 } 3906 if (sq != NULL) { 3907 sq->end(didModify); 3908 sq->push(block); 3909 } 3910#ifdef AUDIO_WATCHDOG 3911 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3912 mAudioWatchdog->pause(); 3913 } 3914#endif 3915 3916 // Now perform the deferred reset on fast tracks that have stopped 3917 while (resetMask != 0) { 3918 size_t i = __builtin_ctz(resetMask); 3919 ALOG_ASSERT(i < count); 3920 resetMask &= ~(1 << i); 3921 sp<Track> t = mActiveTracks[i].promote(); 3922 if (t == 0) { 3923 continue; 3924 } 3925 Track* track = t.get(); 3926 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3927 track->reset(); 3928 } 3929 3930 // remove all the tracks that need to be... 3931 removeTracks_l(*tracksToRemove); 3932 3933 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3934 mEffectBufferValid = true; 3935 } 3936 3937 if (mEffectBufferValid) { 3938 // as long as there are effects we should clear the effects buffer, to avoid 3939 // passing a non-clean buffer to the effect chain 3940 memset(mEffectBuffer, 0, mEffectBufferSize); 3941 } 3942 // sink or mix buffer must be cleared if all tracks are connected to an 3943 // effect chain as in this case the mixer will not write to the sink or mix buffer 3944 // and track effects will accumulate into it 3945 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3946 (mixedTracks == 0 && fastTracks > 0))) { 3947 // FIXME as a performance optimization, should remember previous zero status 3948 if (mMixerBufferValid) { 3949 memset(mMixerBuffer, 0, mMixerBufferSize); 3950 // TODO: In testing, mSinkBuffer below need not be cleared because 3951 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3952 // after mixing. 3953 // 3954 // To enforce this guarantee: 3955 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3956 // (mixedTracks == 0 && fastTracks > 0)) 3957 // must imply MIXER_TRACKS_READY. 3958 // Later, we may clear buffers regardless, and skip much of this logic. 3959 } 3960 // FIXME as a performance optimization, should remember previous zero status 3961 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3962 } 3963 3964 // if any fast tracks, then status is ready 3965 mMixerStatusIgnoringFastTracks = mixerStatus; 3966 if (fastTracks > 0) { 3967 mixerStatus = MIXER_TRACKS_READY; 3968 } 3969 return mixerStatus; 3970} 3971 3972// getTrackName_l() must be called with ThreadBase::mLock held 3973int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3974 audio_format_t format, int sessionId) 3975{ 3976 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3977} 3978 3979// deleteTrackName_l() must be called with ThreadBase::mLock held 3980void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3981{ 3982 ALOGV("remove track (%d) and delete from mixer", name); 3983 mAudioMixer->deleteTrackName(name); 3984} 3985 3986// checkForNewParameter_l() must be called with ThreadBase::mLock held 3987bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3988 status_t& status) 3989{ 3990 bool reconfig = false; 3991 3992 status = NO_ERROR; 3993 3994 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3995 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3996 if (mFastMixer != 0) { 3997 FastMixerStateQueue *sq = mFastMixer->sq(); 3998 FastMixerState *state = sq->begin(); 3999 if (!(state->mCommand & FastMixerState::IDLE)) { 4000 previousCommand = state->mCommand; 4001 state->mCommand = FastMixerState::HOT_IDLE; 4002 sq->end(); 4003 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4004 } else { 4005 sq->end(false /*didModify*/); 4006 } 4007 } 4008 4009 AudioParameter param = AudioParameter(keyValuePair); 4010 int value; 4011 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4012 reconfig = true; 4013 } 4014 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4015 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4016 status = BAD_VALUE; 4017 } else { 4018 // no need to save value, since it's constant 4019 reconfig = true; 4020 } 4021 } 4022 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4023 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4024 status = BAD_VALUE; 4025 } else { 4026 // no need to save value, since it's constant 4027 reconfig = true; 4028 } 4029 } 4030 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4031 // do not accept frame count changes if tracks are open as the track buffer 4032 // size depends on frame count and correct behavior would not be guaranteed 4033 // if frame count is changed after track creation 4034 if (!mTracks.isEmpty()) { 4035 status = INVALID_OPERATION; 4036 } else { 4037 reconfig = true; 4038 } 4039 } 4040 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4041#ifdef ADD_BATTERY_DATA 4042 // when changing the audio output device, call addBatteryData to notify 4043 // the change 4044 if (mOutDevice != value) { 4045 uint32_t params = 0; 4046 // check whether speaker is on 4047 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4048 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4049 } 4050 4051 audio_devices_t deviceWithoutSpeaker 4052 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4053 // check if any other device (except speaker) is on 4054 if (value & deviceWithoutSpeaker ) { 4055 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4056 } 4057 4058 if (params != 0) { 4059 addBatteryData(params); 4060 } 4061 } 4062#endif 4063 4064 // forward device change to effects that have requested to be 4065 // aware of attached audio device. 4066 if (value != AUDIO_DEVICE_NONE) { 4067 mOutDevice = value; 4068 for (size_t i = 0; i < mEffectChains.size(); i++) { 4069 mEffectChains[i]->setDevice_l(mOutDevice); 4070 } 4071 } 4072 } 4073 4074 if (status == NO_ERROR) { 4075 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4076 keyValuePair.string()); 4077 if (!mStandby && status == INVALID_OPERATION) { 4078 mOutput->standby(); 4079 mStandby = true; 4080 mBytesWritten = 0; 4081 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4082 keyValuePair.string()); 4083 } 4084 if (status == NO_ERROR && reconfig) { 4085 readOutputParameters_l(); 4086 delete mAudioMixer; 4087 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4088 for (size_t i = 0; i < mTracks.size() ; i++) { 4089 int name = getTrackName_l(mTracks[i]->mChannelMask, 4090 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4091 if (name < 0) { 4092 break; 4093 } 4094 mTracks[i]->mName = name; 4095 } 4096 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4097 } 4098 } 4099 4100 if (!(previousCommand & FastMixerState::IDLE)) { 4101 ALOG_ASSERT(mFastMixer != 0); 4102 FastMixerStateQueue *sq = mFastMixer->sq(); 4103 FastMixerState *state = sq->begin(); 4104 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4105 state->mCommand = previousCommand; 4106 sq->end(); 4107 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4108 } 4109 4110 return reconfig; 4111} 4112 4113 4114void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4115{ 4116 const size_t SIZE = 256; 4117 char buffer[SIZE]; 4118 String8 result; 4119 4120 PlaybackThread::dumpInternals(fd, args); 4121 4122 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4123 4124 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4125 const FastMixerDumpState copy(mFastMixerDumpState); 4126 copy.dump(fd); 4127 4128#ifdef STATE_QUEUE_DUMP 4129 // Similar for state queue 4130 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4131 observerCopy.dump(fd); 4132 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4133 mutatorCopy.dump(fd); 4134#endif 4135 4136#ifdef TEE_SINK 4137 // Write the tee output to a .wav file 4138 dumpTee(fd, mTeeSource, mId); 4139#endif 4140 4141#ifdef AUDIO_WATCHDOG 4142 if (mAudioWatchdog != 0) { 4143 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4144 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4145 wdCopy.dump(fd); 4146 } 4147#endif 4148} 4149 4150uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4151{ 4152 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4153} 4154 4155uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4156{ 4157 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4158} 4159 4160void AudioFlinger::MixerThread::cacheParameters_l() 4161{ 4162 PlaybackThread::cacheParameters_l(); 4163 4164 // FIXME: Relaxed timing because of a certain device that can't meet latency 4165 // Should be reduced to 2x after the vendor fixes the driver issue 4166 // increase threshold again due to low power audio mode. The way this warning 4167 // threshold is calculated and its usefulness should be reconsidered anyway. 4168 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4169} 4170 4171// ---------------------------------------------------------------------------- 4172 4173AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4174 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4175 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4176 // mLeftVolFloat, mRightVolFloat 4177{ 4178} 4179 4180AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4181 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4182 ThreadBase::type_t type) 4183 : PlaybackThread(audioFlinger, output, id, device, type) 4184 // mLeftVolFloat, mRightVolFloat 4185{ 4186} 4187 4188AudioFlinger::DirectOutputThread::~DirectOutputThread() 4189{ 4190} 4191 4192void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4193{ 4194 audio_track_cblk_t* cblk = track->cblk(); 4195 float left, right; 4196 4197 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4198 left = right = 0; 4199 } else { 4200 float typeVolume = mStreamTypes[track->streamType()].volume; 4201 float v = mMasterVolume * typeVolume; 4202 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4203 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4204 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4205 if (left > GAIN_FLOAT_UNITY) { 4206 left = GAIN_FLOAT_UNITY; 4207 } 4208 left *= v; 4209 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4210 if (right > GAIN_FLOAT_UNITY) { 4211 right = GAIN_FLOAT_UNITY; 4212 } 4213 right *= v; 4214 } 4215 4216 if (lastTrack) { 4217 if (left != mLeftVolFloat || right != mRightVolFloat) { 4218 mLeftVolFloat = left; 4219 mRightVolFloat = right; 4220 4221 // Convert volumes from float to 8.24 4222 uint32_t vl = (uint32_t)(left * (1 << 24)); 4223 uint32_t vr = (uint32_t)(right * (1 << 24)); 4224 4225 // Delegate volume control to effect in track effect chain if needed 4226 // only one effect chain can be present on DirectOutputThread, so if 4227 // there is one, the track is connected to it 4228 if (!mEffectChains.isEmpty()) { 4229 mEffectChains[0]->setVolume_l(&vl, &vr); 4230 left = (float)vl / (1 << 24); 4231 right = (float)vr / (1 << 24); 4232 } 4233 if (mOutput->stream->set_volume) { 4234 mOutput->stream->set_volume(mOutput->stream, left, right); 4235 } 4236 } 4237 } 4238} 4239 4240 4241AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4242 Vector< sp<Track> > *tracksToRemove 4243) 4244{ 4245 size_t count = mActiveTracks.size(); 4246 mixer_state mixerStatus = MIXER_IDLE; 4247 bool doHwPause = false; 4248 bool doHwResume = false; 4249 bool flushPending = false; 4250 4251 // find out which tracks need to be processed 4252 for (size_t i = 0; i < count; i++) { 4253 sp<Track> t = mActiveTracks[i].promote(); 4254 // The track died recently 4255 if (t == 0) { 4256 continue; 4257 } 4258 4259 Track* const track = t.get(); 4260 audio_track_cblk_t* cblk = track->cblk(); 4261 // Only consider last track started for volume and mixer state control. 4262 // In theory an older track could underrun and restart after the new one starts 4263 // but as we only care about the transition phase between two tracks on a 4264 // direct output, it is not a problem to ignore the underrun case. 4265 sp<Track> l = mLatestActiveTrack.promote(); 4266 bool last = l.get() == track; 4267 4268 if (mHwSupportsPause && track->isPausing()) { 4269 track->setPaused(); 4270 if (last && !mHwPaused) { 4271 doHwPause = true; 4272 mHwPaused = true; 4273 } 4274 tracksToRemove->add(track); 4275 } else if (track->isFlushPending()) { 4276 track->flushAck(); 4277 if (last) { 4278 flushPending = true; 4279 } 4280 } else if (mHwSupportsPause && track->isResumePending()){ 4281 track->resumeAck(); 4282 if (last) { 4283 if (mHwPaused) { 4284 doHwResume = true; 4285 mHwPaused = false; 4286 } 4287 } 4288 } 4289 4290 // The first time a track is added we wait 4291 // for all its buffers to be filled before processing it. 4292 // Allow draining the buffer in case the client 4293 // app does not call stop() and relies on underrun to stop: 4294 // hence the test on (track->mRetryCount > 1). 4295 // If retryCount<=1 then track is about to underrun and be removed. 4296 uint32_t minFrames; 4297 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4298 && (track->mRetryCount > 1)) { 4299 minFrames = mNormalFrameCount; 4300 } else { 4301 minFrames = 1; 4302 } 4303 4304 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4305 !track->isStopping_2() && !track->isStopped()) 4306 { 4307 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4308 4309 if (track->mFillingUpStatus == Track::FS_FILLED) { 4310 track->mFillingUpStatus = Track::FS_ACTIVE; 4311 // make sure processVolume_l() will apply new volume even if 0 4312 mLeftVolFloat = mRightVolFloat = -1.0; 4313 if (!mHwSupportsPause) { 4314 track->resumeAck(); 4315 } 4316 } 4317 4318 // compute volume for this track 4319 processVolume_l(track, last); 4320 if (last) { 4321 // reset retry count 4322 track->mRetryCount = kMaxTrackRetriesDirect; 4323 mActiveTrack = t; 4324 mixerStatus = MIXER_TRACKS_READY; 4325 if (usesHwAvSync() && mHwPaused) { 4326 doHwResume = true; 4327 mHwPaused = false; 4328 } 4329 } 4330 } else { 4331 // clear effect chain input buffer if the last active track started underruns 4332 // to avoid sending previous audio buffer again to effects 4333 if (!mEffectChains.isEmpty() && last) { 4334 mEffectChains[0]->clearInputBuffer(); 4335 } 4336 if (track->isStopping_1()) { 4337 track->mState = TrackBase::STOPPING_2; 4338 if (last && mHwPaused) { 4339 doHwResume = true; 4340 mHwPaused = false; 4341 } 4342 } 4343 if ((track->sharedBuffer() != 0) || track->isStopped() || 4344 track->isStopping_2() || track->isPaused()) { 4345 // We have consumed all the buffers of this track. 4346 // Remove it from the list of active tracks. 4347 size_t audioHALFrames; 4348 if (audio_is_linear_pcm(mFormat)) { 4349 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4350 } else { 4351 audioHALFrames = 0; 4352 } 4353 4354 size_t framesWritten = mBytesWritten / mFrameSize; 4355 if (mStandby || !last || 4356 track->presentationComplete(framesWritten, audioHALFrames)) { 4357 if (track->isStopping_2()) { 4358 track->mState = TrackBase::STOPPED; 4359 } 4360 if (track->isStopped()) { 4361 track->reset(); 4362 } 4363 tracksToRemove->add(track); 4364 } 4365 } else { 4366 // No buffers for this track. Give it a few chances to 4367 // fill a buffer, then remove it from active list. 4368 // Only consider last track started for mixer state control 4369 if (--(track->mRetryCount) <= 0) { 4370 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4371 tracksToRemove->add(track); 4372 // indicate to client process that the track was disabled because of underrun; 4373 // it will then automatically call start() when data is available 4374 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4375 } else if (last) { 4376 mixerStatus = MIXER_TRACKS_ENABLED; 4377 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4378 doHwPause = true; 4379 mHwPaused = true; 4380 } 4381 } 4382 } 4383 } 4384 } 4385 4386 // if an active track did not command a flush, check for pending flush on stopped tracks 4387 if (!flushPending) { 4388 for (size_t i = 0; i < mTracks.size(); i++) { 4389 if (mTracks[i]->isFlushPending()) { 4390 mTracks[i]->flushAck(); 4391 flushPending = true; 4392 } 4393 } 4394 } 4395 4396 // make sure the pause/flush/resume sequence is executed in the right order. 4397 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4398 // before flush and then resume HW. This can happen in case of pause/flush/resume 4399 // if resume is received before pause is executed. 4400 if (mHwSupportsPause && !mStandby && 4401 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4402 mOutput->stream->pause(mOutput->stream); 4403 } 4404 if (flushPending) { 4405 flushHw_l(); 4406 } 4407 if (mHwSupportsPause && !mStandby && doHwResume) { 4408 mOutput->stream->resume(mOutput->stream); 4409 } 4410 // remove all the tracks that need to be... 4411 removeTracks_l(*tracksToRemove); 4412 4413 return mixerStatus; 4414} 4415 4416void AudioFlinger::DirectOutputThread::threadLoop_mix() 4417{ 4418 size_t frameCount = mFrameCount; 4419 int8_t *curBuf = (int8_t *)mSinkBuffer; 4420 // output audio to hardware 4421 while (frameCount) { 4422 AudioBufferProvider::Buffer buffer; 4423 buffer.frameCount = frameCount; 4424 status_t status = mActiveTrack->getNextBuffer(&buffer); 4425 if (status != NO_ERROR || buffer.raw == NULL) { 4426 memset(curBuf, 0, frameCount * mFrameSize); 4427 break; 4428 } 4429 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4430 frameCount -= buffer.frameCount; 4431 curBuf += buffer.frameCount * mFrameSize; 4432 mActiveTrack->releaseBuffer(&buffer); 4433 } 4434 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4435 sleepTime = 0; 4436 standbyTime = systemTime() + standbyDelay; 4437 mActiveTrack.clear(); 4438} 4439 4440void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4441{ 4442 // do not write to HAL when paused 4443 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4444 sleepTime = idleSleepTime; 4445 return; 4446 } 4447 if (sleepTime == 0) { 4448 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4449 sleepTime = activeSleepTime; 4450 } else { 4451 sleepTime = idleSleepTime; 4452 } 4453 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4454 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4455 sleepTime = 0; 4456 } 4457} 4458 4459void AudioFlinger::DirectOutputThread::threadLoop_exit() 4460{ 4461 { 4462 Mutex::Autolock _l(mLock); 4463 bool flushPending = false; 4464 for (size_t i = 0; i < mTracks.size(); i++) { 4465 if (mTracks[i]->isFlushPending()) { 4466 mTracks[i]->flushAck(); 4467 flushPending = true; 4468 } 4469 } 4470 if (flushPending) { 4471 flushHw_l(); 4472 } 4473 } 4474 PlaybackThread::threadLoop_exit(); 4475} 4476 4477// must be called with thread mutex locked 4478bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4479{ 4480 bool trackPaused = false; 4481 bool trackStopped = false; 4482 4483 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4484 // after a timeout and we will enter standby then. 4485 if (mTracks.size() > 0) { 4486 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4487 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4488 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4489 } 4490 4491 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped)); 4492} 4493 4494// getTrackName_l() must be called with ThreadBase::mLock held 4495int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4496 audio_format_t format __unused, int sessionId __unused) 4497{ 4498 return 0; 4499} 4500 4501// deleteTrackName_l() must be called with ThreadBase::mLock held 4502void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4503{ 4504} 4505 4506// checkForNewParameter_l() must be called with ThreadBase::mLock held 4507bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4508 status_t& status) 4509{ 4510 bool reconfig = false; 4511 4512 status = NO_ERROR; 4513 4514 AudioParameter param = AudioParameter(keyValuePair); 4515 int value; 4516 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4517 // forward device change to effects that have requested to be 4518 // aware of attached audio device. 4519 if (value != AUDIO_DEVICE_NONE) { 4520 mOutDevice = value; 4521 for (size_t i = 0; i < mEffectChains.size(); i++) { 4522 mEffectChains[i]->setDevice_l(mOutDevice); 4523 } 4524 } 4525 } 4526 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4527 // do not accept frame count changes if tracks are open as the track buffer 4528 // size depends on frame count and correct behavior would not be garantied 4529 // if frame count is changed after track creation 4530 if (!mTracks.isEmpty()) { 4531 status = INVALID_OPERATION; 4532 } else { 4533 reconfig = true; 4534 } 4535 } 4536 if (status == NO_ERROR) { 4537 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4538 keyValuePair.string()); 4539 if (!mStandby && status == INVALID_OPERATION) { 4540 mOutput->standby(); 4541 mStandby = true; 4542 mBytesWritten = 0; 4543 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4544 keyValuePair.string()); 4545 } 4546 if (status == NO_ERROR && reconfig) { 4547 readOutputParameters_l(); 4548 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4549 } 4550 } 4551 4552 return reconfig; 4553} 4554 4555uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4556{ 4557 uint32_t time; 4558 if (audio_is_linear_pcm(mFormat)) { 4559 time = PlaybackThread::activeSleepTimeUs(); 4560 } else { 4561 time = 10000; 4562 } 4563 return time; 4564} 4565 4566uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4567{ 4568 uint32_t time; 4569 if (audio_is_linear_pcm(mFormat)) { 4570 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4571 } else { 4572 time = 10000; 4573 } 4574 return time; 4575} 4576 4577uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4578{ 4579 uint32_t time; 4580 if (audio_is_linear_pcm(mFormat)) { 4581 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4582 } else { 4583 time = 10000; 4584 } 4585 return time; 4586} 4587 4588void AudioFlinger::DirectOutputThread::cacheParameters_l() 4589{ 4590 PlaybackThread::cacheParameters_l(); 4591 4592 // use shorter standby delay as on normal output to release 4593 // hardware resources as soon as possible 4594 // no delay on outputs with HW A/V sync 4595 if (usesHwAvSync()) { 4596 standbyDelay = 0; 4597 } else if (audio_is_linear_pcm(mFormat)) { 4598 standbyDelay = microseconds(activeSleepTime*2); 4599 } else { 4600 standbyDelay = kOffloadStandbyDelayNs; 4601 } 4602} 4603 4604void AudioFlinger::DirectOutputThread::flushHw_l() 4605{ 4606 mOutput->flush(); 4607 mHwPaused = false; 4608} 4609 4610// ---------------------------------------------------------------------------- 4611 4612AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4613 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4614 : Thread(false /*canCallJava*/), 4615 mPlaybackThread(playbackThread), 4616 mWriteAckSequence(0), 4617 mDrainSequence(0) 4618{ 4619} 4620 4621AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4622{ 4623} 4624 4625void AudioFlinger::AsyncCallbackThread::onFirstRef() 4626{ 4627 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4628} 4629 4630bool AudioFlinger::AsyncCallbackThread::threadLoop() 4631{ 4632 while (!exitPending()) { 4633 uint32_t writeAckSequence; 4634 uint32_t drainSequence; 4635 4636 { 4637 Mutex::Autolock _l(mLock); 4638 while (!((mWriteAckSequence & 1) || 4639 (mDrainSequence & 1) || 4640 exitPending())) { 4641 mWaitWorkCV.wait(mLock); 4642 } 4643 4644 if (exitPending()) { 4645 break; 4646 } 4647 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4648 mWriteAckSequence, mDrainSequence); 4649 writeAckSequence = mWriteAckSequence; 4650 mWriteAckSequence &= ~1; 4651 drainSequence = mDrainSequence; 4652 mDrainSequence &= ~1; 4653 } 4654 { 4655 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4656 if (playbackThread != 0) { 4657 if (writeAckSequence & 1) { 4658 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4659 } 4660 if (drainSequence & 1) { 4661 playbackThread->resetDraining(drainSequence >> 1); 4662 } 4663 } 4664 } 4665 } 4666 return false; 4667} 4668 4669void AudioFlinger::AsyncCallbackThread::exit() 4670{ 4671 ALOGV("AsyncCallbackThread::exit"); 4672 Mutex::Autolock _l(mLock); 4673 requestExit(); 4674 mWaitWorkCV.broadcast(); 4675} 4676 4677void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4678{ 4679 Mutex::Autolock _l(mLock); 4680 // bit 0 is cleared 4681 mWriteAckSequence = sequence << 1; 4682} 4683 4684void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4685{ 4686 Mutex::Autolock _l(mLock); 4687 // ignore unexpected callbacks 4688 if (mWriteAckSequence & 2) { 4689 mWriteAckSequence |= 1; 4690 mWaitWorkCV.signal(); 4691 } 4692} 4693 4694void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4695{ 4696 Mutex::Autolock _l(mLock); 4697 // bit 0 is cleared 4698 mDrainSequence = sequence << 1; 4699} 4700 4701void AudioFlinger::AsyncCallbackThread::resetDraining() 4702{ 4703 Mutex::Autolock _l(mLock); 4704 // ignore unexpected callbacks 4705 if (mDrainSequence & 2) { 4706 mDrainSequence |= 1; 4707 mWaitWorkCV.signal(); 4708 } 4709} 4710 4711 4712// ---------------------------------------------------------------------------- 4713AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4714 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4715 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4716 mPausedBytesRemaining(0) 4717{ 4718 //FIXME: mStandby should be set to true by ThreadBase constructor 4719 mStandby = true; 4720} 4721 4722void AudioFlinger::OffloadThread::threadLoop_exit() 4723{ 4724 if (mFlushPending || mHwPaused) { 4725 // If a flush is pending or track was paused, just discard buffered data 4726 flushHw_l(); 4727 } else { 4728 mMixerStatus = MIXER_DRAIN_ALL; 4729 threadLoop_drain(); 4730 } 4731 if (mUseAsyncWrite) { 4732 ALOG_ASSERT(mCallbackThread != 0); 4733 mCallbackThread->exit(); 4734 } 4735 PlaybackThread::threadLoop_exit(); 4736} 4737 4738AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4739 Vector< sp<Track> > *tracksToRemove 4740) 4741{ 4742 size_t count = mActiveTracks.size(); 4743 4744 mixer_state mixerStatus = MIXER_IDLE; 4745 bool doHwPause = false; 4746 bool doHwResume = false; 4747 4748 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4749 4750 // find out which tracks need to be processed 4751 for (size_t i = 0; i < count; i++) { 4752 sp<Track> t = mActiveTracks[i].promote(); 4753 // The track died recently 4754 if (t == 0) { 4755 continue; 4756 } 4757 Track* const track = t.get(); 4758 audio_track_cblk_t* cblk = track->cblk(); 4759 // Only consider last track started for volume and mixer state control. 4760 // In theory an older track could underrun and restart after the new one starts 4761 // but as we only care about the transition phase between two tracks on a 4762 // direct output, it is not a problem to ignore the underrun case. 4763 sp<Track> l = mLatestActiveTrack.promote(); 4764 bool last = l.get() == track; 4765 4766 if (track->isInvalid()) { 4767 ALOGW("An invalidated track shouldn't be in active list"); 4768 tracksToRemove->add(track); 4769 continue; 4770 } 4771 4772 if (track->mState == TrackBase::IDLE) { 4773 ALOGW("An idle track shouldn't be in active list"); 4774 continue; 4775 } 4776 4777 if (track->isPausing()) { 4778 track->setPaused(); 4779 if (last) { 4780 if (!mHwPaused) { 4781 doHwPause = true; 4782 mHwPaused = true; 4783 } 4784 // If we were part way through writing the mixbuffer to 4785 // the HAL we must save this until we resume 4786 // BUG - this will be wrong if a different track is made active, 4787 // in that case we want to discard the pending data in the 4788 // mixbuffer and tell the client to present it again when the 4789 // track is resumed 4790 mPausedWriteLength = mCurrentWriteLength; 4791 mPausedBytesRemaining = mBytesRemaining; 4792 mBytesRemaining = 0; // stop writing 4793 } 4794 tracksToRemove->add(track); 4795 } else if (track->isFlushPending()) { 4796 track->flushAck(); 4797 if (last) { 4798 mFlushPending = true; 4799 } 4800 } else if (track->isResumePending()){ 4801 track->resumeAck(); 4802 if (last) { 4803 if (mPausedBytesRemaining) { 4804 // Need to continue write that was interrupted 4805 mCurrentWriteLength = mPausedWriteLength; 4806 mBytesRemaining = mPausedBytesRemaining; 4807 mPausedBytesRemaining = 0; 4808 } 4809 if (mHwPaused) { 4810 doHwResume = true; 4811 mHwPaused = false; 4812 // threadLoop_mix() will handle the case that we need to 4813 // resume an interrupted write 4814 } 4815 // enable write to audio HAL 4816 sleepTime = 0; 4817 4818 // Do not handle new data in this iteration even if track->framesReady() 4819 mixerStatus = MIXER_TRACKS_ENABLED; 4820 } 4821 } else if (track->framesReady() && track->isReady() && 4822 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4823 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4824 if (track->mFillingUpStatus == Track::FS_FILLED) { 4825 track->mFillingUpStatus = Track::FS_ACTIVE; 4826 // make sure processVolume_l() will apply new volume even if 0 4827 mLeftVolFloat = mRightVolFloat = -1.0; 4828 } 4829 4830 if (last) { 4831 sp<Track> previousTrack = mPreviousTrack.promote(); 4832 if (previousTrack != 0) { 4833 if (track != previousTrack.get()) { 4834 // Flush any data still being written from last track 4835 mBytesRemaining = 0; 4836 if (mPausedBytesRemaining) { 4837 // Last track was paused so we also need to flush saved 4838 // mixbuffer state and invalidate track so that it will 4839 // re-submit that unwritten data when it is next resumed 4840 mPausedBytesRemaining = 0; 4841 // Invalidate is a bit drastic - would be more efficient 4842 // to have a flag to tell client that some of the 4843 // previously written data was lost 4844 previousTrack->invalidate(); 4845 } 4846 // flush data already sent to the DSP if changing audio session as audio 4847 // comes from a different source. Also invalidate previous track to force a 4848 // seek when resuming. 4849 if (previousTrack->sessionId() != track->sessionId()) { 4850 previousTrack->invalidate(); 4851 } 4852 } 4853 } 4854 mPreviousTrack = track; 4855 // reset retry count 4856 track->mRetryCount = kMaxTrackRetriesOffload; 4857 mActiveTrack = t; 4858 mixerStatus = MIXER_TRACKS_READY; 4859 } 4860 } else { 4861 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4862 if (track->isStopping_1()) { 4863 // Hardware buffer can hold a large amount of audio so we must 4864 // wait for all current track's data to drain before we say 4865 // that the track is stopped. 4866 if (mBytesRemaining == 0) { 4867 // Only start draining when all data in mixbuffer 4868 // has been written 4869 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4870 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4871 // do not drain if no data was ever sent to HAL (mStandby == true) 4872 if (last && !mStandby) { 4873 // do not modify drain sequence if we are already draining. This happens 4874 // when resuming from pause after drain. 4875 if ((mDrainSequence & 1) == 0) { 4876 sleepTime = 0; 4877 standbyTime = systemTime() + standbyDelay; 4878 mixerStatus = MIXER_DRAIN_TRACK; 4879 mDrainSequence += 2; 4880 } 4881 if (mHwPaused) { 4882 // It is possible to move from PAUSED to STOPPING_1 without 4883 // a resume so we must ensure hardware is running 4884 doHwResume = true; 4885 mHwPaused = false; 4886 } 4887 } 4888 } 4889 } else if (track->isStopping_2()) { 4890 // Drain has completed or we are in standby, signal presentation complete 4891 if (!(mDrainSequence & 1) || !last || mStandby) { 4892 track->mState = TrackBase::STOPPED; 4893 size_t audioHALFrames = 4894 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4895 size_t framesWritten = 4896 mBytesWritten / mOutput->getFrameSize(); 4897 track->presentationComplete(framesWritten, audioHALFrames); 4898 track->reset(); 4899 tracksToRemove->add(track); 4900 } 4901 } else { 4902 // No buffers for this track. Give it a few chances to 4903 // fill a buffer, then remove it from active list. 4904 if (--(track->mRetryCount) <= 0) { 4905 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4906 track->name()); 4907 tracksToRemove->add(track); 4908 // indicate to client process that the track was disabled because of underrun; 4909 // it will then automatically call start() when data is available 4910 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4911 } else if (last){ 4912 mixerStatus = MIXER_TRACKS_ENABLED; 4913 } 4914 } 4915 } 4916 // compute volume for this track 4917 processVolume_l(track, last); 4918 } 4919 4920 // make sure the pause/flush/resume sequence is executed in the right order. 4921 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4922 // before flush and then resume HW. This can happen in case of pause/flush/resume 4923 // if resume is received before pause is executed. 4924 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4925 mOutput->stream->pause(mOutput->stream); 4926 } 4927 if (mFlushPending) { 4928 flushHw_l(); 4929 mFlushPending = false; 4930 } 4931 if (!mStandby && doHwResume) { 4932 mOutput->stream->resume(mOutput->stream); 4933 } 4934 4935 // remove all the tracks that need to be... 4936 removeTracks_l(*tracksToRemove); 4937 4938 return mixerStatus; 4939} 4940 4941// must be called with thread mutex locked 4942bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4943{ 4944 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4945 mWriteAckSequence, mDrainSequence); 4946 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4947 return true; 4948 } 4949 return false; 4950} 4951 4952bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4953{ 4954 Mutex::Autolock _l(mLock); 4955 return waitingAsyncCallback_l(); 4956} 4957 4958void AudioFlinger::OffloadThread::flushHw_l() 4959{ 4960 DirectOutputThread::flushHw_l(); 4961 // Flush anything still waiting in the mixbuffer 4962 mCurrentWriteLength = 0; 4963 mBytesRemaining = 0; 4964 mPausedWriteLength = 0; 4965 mPausedBytesRemaining = 0; 4966 4967 if (mUseAsyncWrite) { 4968 // discard any pending drain or write ack by incrementing sequence 4969 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4970 mDrainSequence = (mDrainSequence + 2) & ~1; 4971 ALOG_ASSERT(mCallbackThread != 0); 4972 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4973 mCallbackThread->setDraining(mDrainSequence); 4974 } 4975} 4976 4977void AudioFlinger::OffloadThread::onAddNewTrack_l() 4978{ 4979 sp<Track> previousTrack = mPreviousTrack.promote(); 4980 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4981 4982 if (previousTrack != 0 && latestTrack != 0 && 4983 (previousTrack->sessionId() != latestTrack->sessionId())) { 4984 mFlushPending = true; 4985 } 4986 PlaybackThread::onAddNewTrack_l(); 4987} 4988 4989// ---------------------------------------------------------------------------- 4990 4991AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4992 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4993 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4994 DUPLICATING), 4995 mWaitTimeMs(UINT_MAX) 4996{ 4997 addOutputTrack(mainThread); 4998} 4999 5000AudioFlinger::DuplicatingThread::~DuplicatingThread() 5001{ 5002 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5003 mOutputTracks[i]->destroy(); 5004 } 5005} 5006 5007void AudioFlinger::DuplicatingThread::threadLoop_mix() 5008{ 5009 // mix buffers... 5010 if (outputsReady(outputTracks)) { 5011 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5012 } else { 5013 if (mMixerBufferValid) { 5014 memset(mMixerBuffer, 0, mMixerBufferSize); 5015 } else { 5016 memset(mSinkBuffer, 0, mSinkBufferSize); 5017 } 5018 } 5019 sleepTime = 0; 5020 writeFrames = mNormalFrameCount; 5021 mCurrentWriteLength = mSinkBufferSize; 5022 standbyTime = systemTime() + standbyDelay; 5023} 5024 5025void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5026{ 5027 if (sleepTime == 0) { 5028 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5029 sleepTime = activeSleepTime; 5030 } else { 5031 sleepTime = idleSleepTime; 5032 } 5033 } else if (mBytesWritten != 0) { 5034 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5035 writeFrames = mNormalFrameCount; 5036 memset(mSinkBuffer, 0, mSinkBufferSize); 5037 } else { 5038 // flush remaining overflow buffers in output tracks 5039 writeFrames = 0; 5040 } 5041 sleepTime = 0; 5042 } 5043} 5044 5045ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5046{ 5047 for (size_t i = 0; i < outputTracks.size(); i++) { 5048 outputTracks[i]->write(mSinkBuffer, writeFrames); 5049 } 5050 mStandby = false; 5051 return (ssize_t)mSinkBufferSize; 5052} 5053 5054void AudioFlinger::DuplicatingThread::threadLoop_standby() 5055{ 5056 // DuplicatingThread implements standby by stopping all tracks 5057 for (size_t i = 0; i < outputTracks.size(); i++) { 5058 outputTracks[i]->stop(); 5059 } 5060} 5061 5062void AudioFlinger::DuplicatingThread::saveOutputTracks() 5063{ 5064 outputTracks = mOutputTracks; 5065} 5066 5067void AudioFlinger::DuplicatingThread::clearOutputTracks() 5068{ 5069 outputTracks.clear(); 5070} 5071 5072void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5073{ 5074 Mutex::Autolock _l(mLock); 5075 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5076 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5077 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5078 const size_t frameCount = 5079 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5080 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5081 // from different OutputTracks and their associated MixerThreads (e.g. one may 5082 // nearly empty and the other may be dropping data). 5083 5084 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5085 this, 5086 mSampleRate, 5087 mFormat, 5088 mChannelMask, 5089 frameCount, 5090 IPCThreadState::self()->getCallingUid()); 5091 if (outputTrack->cblk() != NULL) { 5092 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5093 mOutputTracks.add(outputTrack); 5094 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5095 updateWaitTime_l(); 5096 } 5097} 5098 5099void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5100{ 5101 Mutex::Autolock _l(mLock); 5102 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5103 if (mOutputTracks[i]->thread() == thread) { 5104 mOutputTracks[i]->destroy(); 5105 mOutputTracks.removeAt(i); 5106 updateWaitTime_l(); 5107 return; 5108 } 5109 } 5110 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5111} 5112 5113// caller must hold mLock 5114void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5115{ 5116 mWaitTimeMs = UINT_MAX; 5117 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5118 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5119 if (strong != 0) { 5120 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5121 if (waitTimeMs < mWaitTimeMs) { 5122 mWaitTimeMs = waitTimeMs; 5123 } 5124 } 5125 } 5126} 5127 5128 5129bool AudioFlinger::DuplicatingThread::outputsReady( 5130 const SortedVector< sp<OutputTrack> > &outputTracks) 5131{ 5132 for (size_t i = 0; i < outputTracks.size(); i++) { 5133 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5134 if (thread == 0) { 5135 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5136 outputTracks[i].get()); 5137 return false; 5138 } 5139 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5140 // see note at standby() declaration 5141 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5142 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5143 thread.get()); 5144 return false; 5145 } 5146 } 5147 return true; 5148} 5149 5150uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5151{ 5152 return (mWaitTimeMs * 1000) / 2; 5153} 5154 5155void AudioFlinger::DuplicatingThread::cacheParameters_l() 5156{ 5157 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5158 updateWaitTime_l(); 5159 5160 MixerThread::cacheParameters_l(); 5161} 5162 5163// ---------------------------------------------------------------------------- 5164// Record 5165// ---------------------------------------------------------------------------- 5166 5167AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5168 AudioStreamIn *input, 5169 audio_io_handle_t id, 5170 audio_devices_t outDevice, 5171 audio_devices_t inDevice 5172#ifdef TEE_SINK 5173 , const sp<NBAIO_Sink>& teeSink 5174#endif 5175 ) : 5176 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5177 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5178 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5179 mRsmpInRear(0) 5180#ifdef TEE_SINK 5181 , mTeeSink(teeSink) 5182#endif 5183 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5184 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5185 // mFastCapture below 5186 , mFastCaptureFutex(0) 5187 // mInputSource 5188 // mPipeSink 5189 // mPipeSource 5190 , mPipeFramesP2(0) 5191 // mPipeMemory 5192 // mFastCaptureNBLogWriter 5193 , mFastTrackAvail(false) 5194{ 5195 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5196 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5197 5198 readInputParameters_l(); 5199 5200 // create an NBAIO source for the HAL input stream, and negotiate 5201 mInputSource = new AudioStreamInSource(input->stream); 5202 size_t numCounterOffers = 0; 5203 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5204 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5205 ALOG_ASSERT(index == 0); 5206 5207 // initialize fast capture depending on configuration 5208 bool initFastCapture; 5209 switch (kUseFastCapture) { 5210 case FastCapture_Never: 5211 initFastCapture = false; 5212 break; 5213 case FastCapture_Always: 5214 initFastCapture = true; 5215 break; 5216 case FastCapture_Static: 5217 uint32_t primaryOutputSampleRate; 5218 { 5219 AutoMutex _l(audioFlinger->mHardwareLock); 5220 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5221 } 5222 initFastCapture = 5223 // either capture sample rate is same as (a reasonable) primary output sample rate 5224 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5225 (mSampleRate == primaryOutputSampleRate)) || 5226 // or primary output sample rate is unknown, and capture sample rate is reasonable 5227 ((primaryOutputSampleRate == 0) && 5228 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5229 // and the buffer size is < 12 ms 5230 (mFrameCount * 1000) / mSampleRate < 12; 5231 break; 5232 // case FastCapture_Dynamic: 5233 } 5234 5235 if (initFastCapture) { 5236 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5237 NBAIO_Format format = mInputSource->format(); 5238 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5239 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5240 void *pipeBuffer; 5241 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5242 sp<IMemory> pipeMemory; 5243 if ((roHeap == 0) || 5244 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5245 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5246 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5247 goto failed; 5248 } 5249 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5250 memset(pipeBuffer, 0, pipeSize); 5251 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5252 const NBAIO_Format offers[1] = {format}; 5253 size_t numCounterOffers = 0; 5254 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5255 ALOG_ASSERT(index == 0); 5256 mPipeSink = pipe; 5257 PipeReader *pipeReader = new PipeReader(*pipe); 5258 numCounterOffers = 0; 5259 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5260 ALOG_ASSERT(index == 0); 5261 mPipeSource = pipeReader; 5262 mPipeFramesP2 = pipeFramesP2; 5263 mPipeMemory = pipeMemory; 5264 5265 // create fast capture 5266 mFastCapture = new FastCapture(); 5267 FastCaptureStateQueue *sq = mFastCapture->sq(); 5268#ifdef STATE_QUEUE_DUMP 5269 // FIXME 5270#endif 5271 FastCaptureState *state = sq->begin(); 5272 state->mCblk = NULL; 5273 state->mInputSource = mInputSource.get(); 5274 state->mInputSourceGen++; 5275 state->mPipeSink = pipe; 5276 state->mPipeSinkGen++; 5277 state->mFrameCount = mFrameCount; 5278 state->mCommand = FastCaptureState::COLD_IDLE; 5279 // already done in constructor initialization list 5280 //mFastCaptureFutex = 0; 5281 state->mColdFutexAddr = &mFastCaptureFutex; 5282 state->mColdGen++; 5283 state->mDumpState = &mFastCaptureDumpState; 5284#ifdef TEE_SINK 5285 // FIXME 5286#endif 5287 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5288 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5289 sq->end(); 5290 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5291 5292 // start the fast capture 5293 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5294 pid_t tid = mFastCapture->getTid(); 5295 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5296 if (err != 0) { 5297 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5298 kPriorityFastCapture, getpid_cached, tid, err); 5299 } 5300 5301#ifdef AUDIO_WATCHDOG 5302 // FIXME 5303#endif 5304 5305 mFastTrackAvail = true; 5306 } 5307failed: ; 5308 5309 // FIXME mNormalSource 5310} 5311 5312AudioFlinger::RecordThread::~RecordThread() 5313{ 5314 if (mFastCapture != 0) { 5315 FastCaptureStateQueue *sq = mFastCapture->sq(); 5316 FastCaptureState *state = sq->begin(); 5317 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5318 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5319 if (old == -1) { 5320 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5321 } 5322 } 5323 state->mCommand = FastCaptureState::EXIT; 5324 sq->end(); 5325 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5326 mFastCapture->join(); 5327 mFastCapture.clear(); 5328 } 5329 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5330 mAudioFlinger->unregisterWriter(mNBLogWriter); 5331 delete[] mRsmpInBuffer; 5332} 5333 5334void AudioFlinger::RecordThread::onFirstRef() 5335{ 5336 run(mThreadName, PRIORITY_URGENT_AUDIO); 5337} 5338 5339bool AudioFlinger::RecordThread::threadLoop() 5340{ 5341 nsecs_t lastWarning = 0; 5342 5343 inputStandBy(); 5344 5345reacquire_wakelock: 5346 sp<RecordTrack> activeTrack; 5347 int activeTracksGen; 5348 { 5349 Mutex::Autolock _l(mLock); 5350 size_t size = mActiveTracks.size(); 5351 activeTracksGen = mActiveTracksGen; 5352 if (size > 0) { 5353 // FIXME an arbitrary choice 5354 activeTrack = mActiveTracks[0]; 5355 acquireWakeLock_l(activeTrack->uid()); 5356 if (size > 1) { 5357 SortedVector<int> tmp; 5358 for (size_t i = 0; i < size; i++) { 5359 tmp.add(mActiveTracks[i]->uid()); 5360 } 5361 updateWakeLockUids_l(tmp); 5362 } 5363 } else { 5364 acquireWakeLock_l(-1); 5365 } 5366 } 5367 5368 // used to request a deferred sleep, to be executed later while mutex is unlocked 5369 uint32_t sleepUs = 0; 5370 5371 // loop while there is work to do 5372 for (;;) { 5373 Vector< sp<EffectChain> > effectChains; 5374 5375 // sleep with mutex unlocked 5376 if (sleepUs > 0) { 5377 ATRACE_BEGIN("sleep"); 5378 usleep(sleepUs); 5379 ATRACE_END(); 5380 sleepUs = 0; 5381 } 5382 5383 // activeTracks accumulates a copy of a subset of mActiveTracks 5384 Vector< sp<RecordTrack> > activeTracks; 5385 5386 // reference to the (first and only) active fast track 5387 sp<RecordTrack> fastTrack; 5388 5389 // reference to a fast track which is about to be removed 5390 sp<RecordTrack> fastTrackToRemove; 5391 5392 { // scope for mLock 5393 Mutex::Autolock _l(mLock); 5394 5395 processConfigEvents_l(); 5396 5397 // check exitPending here because checkForNewParameters_l() and 5398 // checkForNewParameters_l() can temporarily release mLock 5399 if (exitPending()) { 5400 break; 5401 } 5402 5403 // if no active track(s), then standby and release wakelock 5404 size_t size = mActiveTracks.size(); 5405 if (size == 0) { 5406 standbyIfNotAlreadyInStandby(); 5407 // exitPending() can't become true here 5408 releaseWakeLock_l(); 5409 ALOGV("RecordThread: loop stopping"); 5410 // go to sleep 5411 mWaitWorkCV.wait(mLock); 5412 ALOGV("RecordThread: loop starting"); 5413 goto reacquire_wakelock; 5414 } 5415 5416 if (mActiveTracksGen != activeTracksGen) { 5417 activeTracksGen = mActiveTracksGen; 5418 SortedVector<int> tmp; 5419 for (size_t i = 0; i < size; i++) { 5420 tmp.add(mActiveTracks[i]->uid()); 5421 } 5422 updateWakeLockUids_l(tmp); 5423 } 5424 5425 bool doBroadcast = false; 5426 for (size_t i = 0; i < size; ) { 5427 5428 activeTrack = mActiveTracks[i]; 5429 if (activeTrack->isTerminated()) { 5430 if (activeTrack->isFastTrack()) { 5431 ALOG_ASSERT(fastTrackToRemove == 0); 5432 fastTrackToRemove = activeTrack; 5433 } 5434 removeTrack_l(activeTrack); 5435 mActiveTracks.remove(activeTrack); 5436 mActiveTracksGen++; 5437 size--; 5438 continue; 5439 } 5440 5441 TrackBase::track_state activeTrackState = activeTrack->mState; 5442 switch (activeTrackState) { 5443 5444 case TrackBase::PAUSING: 5445 mActiveTracks.remove(activeTrack); 5446 mActiveTracksGen++; 5447 doBroadcast = true; 5448 size--; 5449 continue; 5450 5451 case TrackBase::STARTING_1: 5452 sleepUs = 10000; 5453 i++; 5454 continue; 5455 5456 case TrackBase::STARTING_2: 5457 doBroadcast = true; 5458 mStandby = false; 5459 activeTrack->mState = TrackBase::ACTIVE; 5460 break; 5461 5462 case TrackBase::ACTIVE: 5463 break; 5464 5465 case TrackBase::IDLE: 5466 i++; 5467 continue; 5468 5469 default: 5470 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5471 } 5472 5473 activeTracks.add(activeTrack); 5474 i++; 5475 5476 if (activeTrack->isFastTrack()) { 5477 ALOG_ASSERT(!mFastTrackAvail); 5478 ALOG_ASSERT(fastTrack == 0); 5479 fastTrack = activeTrack; 5480 } 5481 } 5482 if (doBroadcast) { 5483 mStartStopCond.broadcast(); 5484 } 5485 5486 // sleep if there are no active tracks to process 5487 if (activeTracks.size() == 0) { 5488 if (sleepUs == 0) { 5489 sleepUs = kRecordThreadSleepUs; 5490 } 5491 continue; 5492 } 5493 sleepUs = 0; 5494 5495 lockEffectChains_l(effectChains); 5496 } 5497 5498 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5499 5500 size_t size = effectChains.size(); 5501 for (size_t i = 0; i < size; i++) { 5502 // thread mutex is not locked, but effect chain is locked 5503 effectChains[i]->process_l(); 5504 } 5505 5506 // Push a new fast capture state if fast capture is not already running, or cblk change 5507 if (mFastCapture != 0) { 5508 FastCaptureStateQueue *sq = mFastCapture->sq(); 5509 FastCaptureState *state = sq->begin(); 5510 bool didModify = false; 5511 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5512 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5513 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5514 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5515 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5516 if (old == -1) { 5517 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5518 } 5519 } 5520 state->mCommand = FastCaptureState::READ_WRITE; 5521#if 0 // FIXME 5522 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5523 FastThreadDumpState::kSamplingNforLowRamDevice : 5524 FastThreadDumpState::kSamplingN); 5525#endif 5526 didModify = true; 5527 } 5528 audio_track_cblk_t *cblkOld = state->mCblk; 5529 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5530 if (cblkNew != cblkOld) { 5531 state->mCblk = cblkNew; 5532 // block until acked if removing a fast track 5533 if (cblkOld != NULL) { 5534 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5535 } 5536 didModify = true; 5537 } 5538 sq->end(didModify); 5539 if (didModify) { 5540 sq->push(block); 5541#if 0 5542 if (kUseFastCapture == FastCapture_Dynamic) { 5543 mNormalSource = mPipeSource; 5544 } 5545#endif 5546 } 5547 } 5548 5549 // now run the fast track destructor with thread mutex unlocked 5550 fastTrackToRemove.clear(); 5551 5552 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5553 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5554 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5555 // If destination is non-contiguous, first read past the nominal end of buffer, then 5556 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5557 5558 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5559 ssize_t framesRead; 5560 5561 // If an NBAIO source is present, use it to read the normal capture's data 5562 if (mPipeSource != 0) { 5563 size_t framesToRead = mBufferSize / mFrameSize; 5564 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5565 framesToRead, AudioBufferProvider::kInvalidPTS); 5566 if (framesRead == 0) { 5567 // since pipe is non-blocking, simulate blocking input 5568 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5569 } 5570 // otherwise use the HAL / AudioStreamIn directly 5571 } else { 5572 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5573 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5574 if (bytesRead < 0) { 5575 framesRead = bytesRead; 5576 } else { 5577 framesRead = bytesRead / mFrameSize; 5578 } 5579 } 5580 5581 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5582 ALOGE("read failed: framesRead=%d", framesRead); 5583 // Force input into standby so that it tries to recover at next read attempt 5584 inputStandBy(); 5585 sleepUs = kRecordThreadSleepUs; 5586 } 5587 if (framesRead <= 0) { 5588 goto unlock; 5589 } 5590 ALOG_ASSERT(framesRead > 0); 5591 5592 if (mTeeSink != 0) { 5593 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5594 } 5595 // If destination is non-contiguous, we now correct for reading past end of buffer. 5596 { 5597 size_t part1 = mRsmpInFramesP2 - rear; 5598 if ((size_t) framesRead > part1) { 5599 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5600 (framesRead - part1) * mFrameSize); 5601 } 5602 } 5603 rear = mRsmpInRear += framesRead; 5604 5605 size = activeTracks.size(); 5606 // loop over each active track 5607 for (size_t i = 0; i < size; i++) { 5608 activeTrack = activeTracks[i]; 5609 5610 // skip fast tracks, as those are handled directly by FastCapture 5611 if (activeTrack->isFastTrack()) { 5612 continue; 5613 } 5614 5615 // TODO: This code probably should be moved to RecordTrack. 5616 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5617 5618 enum { 5619 OVERRUN_UNKNOWN, 5620 OVERRUN_TRUE, 5621 OVERRUN_FALSE 5622 } overrun = OVERRUN_UNKNOWN; 5623 5624 // loop over getNextBuffer to handle circular sink 5625 for (;;) { 5626 5627 activeTrack->mSink.frameCount = ~0; 5628 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5629 size_t framesOut = activeTrack->mSink.frameCount; 5630 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5631 5632 // check available frames and handle overrun conditions 5633 // if the record track isn't draining fast enough. 5634 bool hasOverrun; 5635 size_t framesIn; 5636 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5637 if (hasOverrun) { 5638 overrun = OVERRUN_TRUE; 5639 } 5640 if (framesOut == 0 || framesIn == 0) { 5641 break; 5642 } 5643 5644 // Don't allow framesOut to be larger than what is possible with resampling 5645 // from framesIn. 5646 // This isn't strictly necessary but helps limit buffer resizing in 5647 // RecordBufferConverter. TODO: remove when no longer needed. 5648 framesOut = min(framesOut, 5649 destinationFramesPossible( 5650 framesIn, mSampleRate, activeTrack->mSampleRate)); 5651 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5652 framesOut = activeTrack->mRecordBufferConverter->convert( 5653 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5654 5655 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5656 overrun = OVERRUN_FALSE; 5657 } 5658 5659 if (activeTrack->mFramesToDrop == 0) { 5660 if (framesOut > 0) { 5661 activeTrack->mSink.frameCount = framesOut; 5662 activeTrack->releaseBuffer(&activeTrack->mSink); 5663 } 5664 } else { 5665 // FIXME could do a partial drop of framesOut 5666 if (activeTrack->mFramesToDrop > 0) { 5667 activeTrack->mFramesToDrop -= framesOut; 5668 if (activeTrack->mFramesToDrop <= 0) { 5669 activeTrack->clearSyncStartEvent(); 5670 } 5671 } else { 5672 activeTrack->mFramesToDrop += framesOut; 5673 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5674 activeTrack->mSyncStartEvent->isCancelled()) { 5675 ALOGW("Synced record %s, session %d, trigger session %d", 5676 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5677 activeTrack->sessionId(), 5678 (activeTrack->mSyncStartEvent != 0) ? 5679 activeTrack->mSyncStartEvent->triggerSession() : 0); 5680 activeTrack->clearSyncStartEvent(); 5681 } 5682 } 5683 } 5684 5685 if (framesOut == 0) { 5686 break; 5687 } 5688 } 5689 5690 switch (overrun) { 5691 case OVERRUN_TRUE: 5692 // client isn't retrieving buffers fast enough 5693 if (!activeTrack->setOverflow()) { 5694 nsecs_t now = systemTime(); 5695 // FIXME should lastWarning per track? 5696 if ((now - lastWarning) > kWarningThrottleNs) { 5697 ALOGW("RecordThread: buffer overflow"); 5698 lastWarning = now; 5699 } 5700 } 5701 break; 5702 case OVERRUN_FALSE: 5703 activeTrack->clearOverflow(); 5704 break; 5705 case OVERRUN_UNKNOWN: 5706 break; 5707 } 5708 5709 } 5710 5711unlock: 5712 // enable changes in effect chain 5713 unlockEffectChains(effectChains); 5714 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5715 } 5716 5717 standbyIfNotAlreadyInStandby(); 5718 5719 { 5720 Mutex::Autolock _l(mLock); 5721 for (size_t i = 0; i < mTracks.size(); i++) { 5722 sp<RecordTrack> track = mTracks[i]; 5723 track->invalidate(); 5724 } 5725 mActiveTracks.clear(); 5726 mActiveTracksGen++; 5727 mStartStopCond.broadcast(); 5728 } 5729 5730 releaseWakeLock(); 5731 5732 ALOGV("RecordThread %p exiting", this); 5733 return false; 5734} 5735 5736void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5737{ 5738 if (!mStandby) { 5739 inputStandBy(); 5740 mStandby = true; 5741 } 5742} 5743 5744void AudioFlinger::RecordThread::inputStandBy() 5745{ 5746 // Idle the fast capture if it's currently running 5747 if (mFastCapture != 0) { 5748 FastCaptureStateQueue *sq = mFastCapture->sq(); 5749 FastCaptureState *state = sq->begin(); 5750 if (!(state->mCommand & FastCaptureState::IDLE)) { 5751 state->mCommand = FastCaptureState::COLD_IDLE; 5752 state->mColdFutexAddr = &mFastCaptureFutex; 5753 state->mColdGen++; 5754 mFastCaptureFutex = 0; 5755 sq->end(); 5756 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5757 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5758#if 0 5759 if (kUseFastCapture == FastCapture_Dynamic) { 5760 // FIXME 5761 } 5762#endif 5763#ifdef AUDIO_WATCHDOG 5764 // FIXME 5765#endif 5766 } else { 5767 sq->end(false /*didModify*/); 5768 } 5769 } 5770 mInput->stream->common.standby(&mInput->stream->common); 5771} 5772 5773// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5774sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5775 const sp<AudioFlinger::Client>& client, 5776 uint32_t sampleRate, 5777 audio_format_t format, 5778 audio_channel_mask_t channelMask, 5779 size_t *pFrameCount, 5780 int sessionId, 5781 size_t *notificationFrames, 5782 int uid, 5783 IAudioFlinger::track_flags_t *flags, 5784 pid_t tid, 5785 status_t *status) 5786{ 5787 size_t frameCount = *pFrameCount; 5788 sp<RecordTrack> track; 5789 status_t lStatus; 5790 5791 // client expresses a preference for FAST, but we get the final say 5792 if (*flags & IAudioFlinger::TRACK_FAST) { 5793 if ( 5794 // we formerly checked for a callback handler (non-0 tid), 5795 // but that is no longer required for TRANSFER_OBTAIN mode 5796 // 5797 // frame count is not specified, or is exactly the pipe depth 5798 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5799 // PCM data 5800 audio_is_linear_pcm(format) && 5801 // native format 5802 (format == mFormat) && 5803 // native channel mask 5804 (channelMask == mChannelMask) && 5805 // native hardware sample rate 5806 (sampleRate == mSampleRate) && 5807 // record thread has an associated fast capture 5808 hasFastCapture() && 5809 // there are sufficient fast track slots available 5810 mFastTrackAvail 5811 ) { 5812 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5813 frameCount, mFrameCount); 5814 } else { 5815 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5816 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5817 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5818 frameCount, mFrameCount, mPipeFramesP2, 5819 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5820 hasFastCapture(), tid, mFastTrackAvail); 5821 *flags &= ~IAudioFlinger::TRACK_FAST; 5822 } 5823 } 5824 5825 // compute track buffer size in frames, and suggest the notification frame count 5826 if (*flags & IAudioFlinger::TRACK_FAST) { 5827 // fast track: frame count is exactly the pipe depth 5828 frameCount = mPipeFramesP2; 5829 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5830 *notificationFrames = mFrameCount; 5831 } else { 5832 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5833 // or 20 ms if there is a fast capture 5834 // TODO This could be a roundupRatio inline, and const 5835 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5836 * sampleRate + mSampleRate - 1) / mSampleRate; 5837 // minimum number of notification periods is at least kMinNotifications, 5838 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5839 static const size_t kMinNotifications = 3; 5840 static const uint32_t kMinMs = 30; 5841 // TODO This could be a roundupRatio inline 5842 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5843 // TODO This could be a roundupRatio inline 5844 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5845 maxNotificationFrames; 5846 const size_t minFrameCount = maxNotificationFrames * 5847 max(kMinNotifications, minNotificationsByMs); 5848 frameCount = max(frameCount, minFrameCount); 5849 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5850 *notificationFrames = maxNotificationFrames; 5851 } 5852 } 5853 *pFrameCount = frameCount; 5854 5855 lStatus = initCheck(); 5856 if (lStatus != NO_ERROR) { 5857 ALOGE("createRecordTrack_l() audio driver not initialized"); 5858 goto Exit; 5859 } 5860 5861 { // scope for mLock 5862 Mutex::Autolock _l(mLock); 5863 5864 track = new RecordTrack(this, client, sampleRate, 5865 format, channelMask, frameCount, NULL, sessionId, uid, 5866 *flags, TrackBase::TYPE_DEFAULT); 5867 5868 lStatus = track->initCheck(); 5869 if (lStatus != NO_ERROR) { 5870 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5871 // track must be cleared from the caller as the caller has the AF lock 5872 goto Exit; 5873 } 5874 mTracks.add(track); 5875 5876 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5877 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5878 mAudioFlinger->btNrecIsOff(); 5879 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5880 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5881 5882 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5883 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5884 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5885 // so ask activity manager to do this on our behalf 5886 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5887 } 5888 } 5889 5890 lStatus = NO_ERROR; 5891 5892Exit: 5893 *status = lStatus; 5894 return track; 5895} 5896 5897status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5898 AudioSystem::sync_event_t event, 5899 int triggerSession) 5900{ 5901 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5902 sp<ThreadBase> strongMe = this; 5903 status_t status = NO_ERROR; 5904 5905 if (event == AudioSystem::SYNC_EVENT_NONE) { 5906 recordTrack->clearSyncStartEvent(); 5907 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5908 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5909 triggerSession, 5910 recordTrack->sessionId(), 5911 syncStartEventCallback, 5912 recordTrack); 5913 // Sync event can be cancelled by the trigger session if the track is not in a 5914 // compatible state in which case we start record immediately 5915 if (recordTrack->mSyncStartEvent->isCancelled()) { 5916 recordTrack->clearSyncStartEvent(); 5917 } else { 5918 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5919 recordTrack->mFramesToDrop = - 5920 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5921 } 5922 } 5923 5924 { 5925 // This section is a rendezvous between binder thread executing start() and RecordThread 5926 AutoMutex lock(mLock); 5927 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5928 if (recordTrack->mState == TrackBase::PAUSING) { 5929 ALOGV("active record track PAUSING -> ACTIVE"); 5930 recordTrack->mState = TrackBase::ACTIVE; 5931 } else { 5932 ALOGV("active record track state %d", recordTrack->mState); 5933 } 5934 return status; 5935 } 5936 5937 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5938 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5939 // or using a separate command thread 5940 recordTrack->mState = TrackBase::STARTING_1; 5941 mActiveTracks.add(recordTrack); 5942 mActiveTracksGen++; 5943 status_t status = NO_ERROR; 5944 if (recordTrack->isExternalTrack()) { 5945 mLock.unlock(); 5946 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5947 mLock.lock(); 5948 // FIXME should verify that recordTrack is still in mActiveTracks 5949 if (status != NO_ERROR) { 5950 mActiveTracks.remove(recordTrack); 5951 mActiveTracksGen++; 5952 recordTrack->clearSyncStartEvent(); 5953 ALOGV("RecordThread::start error %d", status); 5954 return status; 5955 } 5956 } 5957 // Catch up with current buffer indices if thread is already running. 5958 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5959 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5960 // see previously buffered data before it called start(), but with greater risk of overrun. 5961 5962 recordTrack->mResamplerBufferProvider->reset(); 5963 // clear any converter state as new data will be discontinuous 5964 recordTrack->mRecordBufferConverter->reset(); 5965 recordTrack->mState = TrackBase::STARTING_2; 5966 // signal thread to start 5967 mWaitWorkCV.broadcast(); 5968 if (mActiveTracks.indexOf(recordTrack) < 0) { 5969 ALOGV("Record failed to start"); 5970 status = BAD_VALUE; 5971 goto startError; 5972 } 5973 return status; 5974 } 5975 5976startError: 5977 if (recordTrack->isExternalTrack()) { 5978 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5979 } 5980 recordTrack->clearSyncStartEvent(); 5981 // FIXME I wonder why we do not reset the state here? 5982 return status; 5983} 5984 5985void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5986{ 5987 sp<SyncEvent> strongEvent = event.promote(); 5988 5989 if (strongEvent != 0) { 5990 sp<RefBase> ptr = strongEvent->cookie().promote(); 5991 if (ptr != 0) { 5992 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5993 recordTrack->handleSyncStartEvent(strongEvent); 5994 } 5995 } 5996} 5997 5998bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5999 ALOGV("RecordThread::stop"); 6000 AutoMutex _l(mLock); 6001 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6002 return false; 6003 } 6004 // note that threadLoop may still be processing the track at this point [without lock] 6005 recordTrack->mState = TrackBase::PAUSING; 6006 // do not wait for mStartStopCond if exiting 6007 if (exitPending()) { 6008 return true; 6009 } 6010 // FIXME incorrect usage of wait: no explicit predicate or loop 6011 mStartStopCond.wait(mLock); 6012 // if we have been restarted, recordTrack is in mActiveTracks here 6013 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6014 ALOGV("Record stopped OK"); 6015 return true; 6016 } 6017 return false; 6018} 6019 6020bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6021{ 6022 return false; 6023} 6024 6025status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6026{ 6027#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6028 if (!isValidSyncEvent(event)) { 6029 return BAD_VALUE; 6030 } 6031 6032 int eventSession = event->triggerSession(); 6033 status_t ret = NAME_NOT_FOUND; 6034 6035 Mutex::Autolock _l(mLock); 6036 6037 for (size_t i = 0; i < mTracks.size(); i++) { 6038 sp<RecordTrack> track = mTracks[i]; 6039 if (eventSession == track->sessionId()) { 6040 (void) track->setSyncEvent(event); 6041 ret = NO_ERROR; 6042 } 6043 } 6044 return ret; 6045#else 6046 return BAD_VALUE; 6047#endif 6048} 6049 6050// destroyTrack_l() must be called with ThreadBase::mLock held 6051void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6052{ 6053 track->terminate(); 6054 track->mState = TrackBase::STOPPED; 6055 // active tracks are removed by threadLoop() 6056 if (mActiveTracks.indexOf(track) < 0) { 6057 removeTrack_l(track); 6058 } 6059} 6060 6061void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6062{ 6063 mTracks.remove(track); 6064 // need anything related to effects here? 6065 if (track->isFastTrack()) { 6066 ALOG_ASSERT(!mFastTrackAvail); 6067 mFastTrackAvail = true; 6068 } 6069} 6070 6071void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6072{ 6073 dumpInternals(fd, args); 6074 dumpTracks(fd, args); 6075 dumpEffectChains(fd, args); 6076} 6077 6078void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6079{ 6080 dprintf(fd, "\nInput thread %p:\n", this); 6081 6082 dumpBase(fd, args); 6083 6084 if (mActiveTracks.size() == 0) { 6085 dprintf(fd, " No active record clients\n"); 6086 } 6087 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6088 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6089 6090 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6091 const FastCaptureDumpState copy(mFastCaptureDumpState); 6092 copy.dump(fd); 6093} 6094 6095void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6096{ 6097 const size_t SIZE = 256; 6098 char buffer[SIZE]; 6099 String8 result; 6100 6101 size_t numtracks = mTracks.size(); 6102 size_t numactive = mActiveTracks.size(); 6103 size_t numactiveseen = 0; 6104 dprintf(fd, " %d Tracks", numtracks); 6105 if (numtracks) { 6106 dprintf(fd, " of which %d are active\n", numactive); 6107 RecordTrack::appendDumpHeader(result); 6108 for (size_t i = 0; i < numtracks ; ++i) { 6109 sp<RecordTrack> track = mTracks[i]; 6110 if (track != 0) { 6111 bool active = mActiveTracks.indexOf(track) >= 0; 6112 if (active) { 6113 numactiveseen++; 6114 } 6115 track->dump(buffer, SIZE, active); 6116 result.append(buffer); 6117 } 6118 } 6119 } else { 6120 dprintf(fd, "\n"); 6121 } 6122 6123 if (numactiveseen != numactive) { 6124 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6125 " not in the track list\n"); 6126 result.append(buffer); 6127 RecordTrack::appendDumpHeader(result); 6128 for (size_t i = 0; i < numactive; ++i) { 6129 sp<RecordTrack> track = mActiveTracks[i]; 6130 if (mTracks.indexOf(track) < 0) { 6131 track->dump(buffer, SIZE, true); 6132 result.append(buffer); 6133 } 6134 } 6135 6136 } 6137 write(fd, result.string(), result.size()); 6138} 6139 6140 6141void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6142{ 6143 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6144 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6145 mRsmpInFront = recordThread->mRsmpInRear; 6146 mRsmpInUnrel = 0; 6147} 6148 6149void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6150 size_t *framesAvailable, bool *hasOverrun) 6151{ 6152 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6153 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6154 const int32_t rear = recordThread->mRsmpInRear; 6155 const int32_t front = mRsmpInFront; 6156 const ssize_t filled = rear - front; 6157 6158 size_t framesIn; 6159 bool overrun = false; 6160 if (filled < 0) { 6161 // should not happen, but treat like a massive overrun and re-sync 6162 framesIn = 0; 6163 mRsmpInFront = rear; 6164 overrun = true; 6165 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6166 framesIn = (size_t) filled; 6167 } else { 6168 // client is not keeping up with server, but give it latest data 6169 framesIn = recordThread->mRsmpInFrames; 6170 mRsmpInFront = /* front = */ rear - framesIn; 6171 overrun = true; 6172 } 6173 if (framesAvailable != NULL) { 6174 *framesAvailable = framesIn; 6175 } 6176 if (hasOverrun != NULL) { 6177 *hasOverrun = overrun; 6178 } 6179} 6180 6181// AudioBufferProvider interface 6182status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6183 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6184{ 6185 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6186 if (threadBase == 0) { 6187 buffer->frameCount = 0; 6188 buffer->raw = NULL; 6189 return NOT_ENOUGH_DATA; 6190 } 6191 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6192 int32_t rear = recordThread->mRsmpInRear; 6193 int32_t front = mRsmpInFront; 6194 ssize_t filled = rear - front; 6195 // FIXME should not be P2 (don't want to increase latency) 6196 // FIXME if client not keeping up, discard 6197 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6198 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6199 front &= recordThread->mRsmpInFramesP2 - 1; 6200 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6201 if (part1 > (size_t) filled) { 6202 part1 = filled; 6203 } 6204 size_t ask = buffer->frameCount; 6205 ALOG_ASSERT(ask > 0); 6206 if (part1 > ask) { 6207 part1 = ask; 6208 } 6209 if (part1 == 0) { 6210 // out of data is fine since the resampler will return a short-count. 6211 buffer->raw = NULL; 6212 buffer->frameCount = 0; 6213 mRsmpInUnrel = 0; 6214 return NOT_ENOUGH_DATA; 6215 } 6216 6217 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6218 buffer->frameCount = part1; 6219 mRsmpInUnrel = part1; 6220 return NO_ERROR; 6221} 6222 6223// AudioBufferProvider interface 6224void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6225 AudioBufferProvider::Buffer* buffer) 6226{ 6227 size_t stepCount = buffer->frameCount; 6228 if (stepCount == 0) { 6229 return; 6230 } 6231 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6232 mRsmpInUnrel -= stepCount; 6233 mRsmpInFront += stepCount; 6234 buffer->raw = NULL; 6235 buffer->frameCount = 0; 6236} 6237 6238AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6239 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6240 uint32_t srcSampleRate, 6241 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6242 uint32_t dstSampleRate) : 6243 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6244 // mSrcFormat 6245 // mSrcSampleRate 6246 // mDstChannelMask 6247 // mDstFormat 6248 // mDstSampleRate 6249 // mSrcChannelCount 6250 // mDstChannelCount 6251 // mDstFrameSize 6252 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6253 mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0) 6254{ 6255 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6256 dstChannelMask, dstFormat, dstSampleRate); 6257} 6258 6259AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6260 free(mBuf); 6261 delete mResampler; 6262 free(mRsmpOutBuffer); 6263} 6264 6265size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6266 AudioBufferProvider *provider, size_t frames) 6267{ 6268 if (mSrcSampleRate == mDstSampleRate) { 6269 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6270 mSrcSampleRate, mSrcFormat, mDstFormat); 6271 6272 AudioBufferProvider::Buffer buffer; 6273 for (size_t i = frames; i > 0; ) { 6274 buffer.frameCount = i; 6275 status_t status = provider->getNextBuffer(&buffer, 0); 6276 if (status != OK || buffer.frameCount == 0) { 6277 frames -= i; // cannot fill request. 6278 break; 6279 } 6280 // convert to destination buffer 6281 convert(dst, buffer.raw, buffer.frameCount); 6282 6283 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6284 i -= buffer.frameCount; 6285 provider->releaseBuffer(&buffer); 6286 } 6287 } else { 6288 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6289 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6290 6291 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 6292 if (mRsmpOutFrameCount < frames) { 6293 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 6294 free(mRsmpOutBuffer); 6295 // resampler always outputs stereo (FOR NOW) 6296 (void)posix_memalign(&mRsmpOutBuffer, 32, frames * FCC_2 * sizeof(int32_t) /*Q4.27*/); 6297 mRsmpOutFrameCount = frames; 6298 } 6299 // resampler accumulates, but we only have one source track 6300 memset(mRsmpOutBuffer, 0, frames * FCC_2 * sizeof(int32_t)); 6301 frames = mResampler->resample((int32_t*)mRsmpOutBuffer, frames, provider); 6302 6303 // convert to destination buffer 6304 convert(dst, mRsmpOutBuffer, frames); 6305 } 6306 return frames; 6307} 6308 6309status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6310 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6311 uint32_t srcSampleRate, 6312 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6313 uint32_t dstSampleRate) 6314{ 6315 // quick evaluation if there is any change. 6316 if (mSrcFormat == srcFormat 6317 && mSrcChannelMask == srcChannelMask 6318 && mSrcSampleRate == srcSampleRate 6319 && mDstFormat == dstFormat 6320 && mDstChannelMask == dstChannelMask 6321 && mDstSampleRate == dstSampleRate) { 6322 return NO_ERROR; 6323 } 6324 6325 const bool valid = 6326 audio_is_input_channel(srcChannelMask) 6327 && audio_is_input_channel(dstChannelMask) 6328 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6329 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6330 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6331 ; // no upsampling checks for now 6332 if (!valid) { 6333 return BAD_VALUE; 6334 } 6335 6336 mSrcFormat = srcFormat; 6337 mSrcChannelMask = srcChannelMask; 6338 mSrcSampleRate = srcSampleRate; 6339 mDstFormat = dstFormat; 6340 mDstChannelMask = dstChannelMask; 6341 mDstSampleRate = dstSampleRate; 6342 6343 // compute derived parameters 6344 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6345 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6346 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6347 6348 // do we need a format buffer? 6349 if (mSrcFormat != mDstFormat && mDstChannelCount != mSrcChannelCount) { 6350 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6351 } else { 6352 mBufFrameSize = 0; 6353 } 6354 mBufFrames = 0; // force the buffer to be resized. 6355 6356 // do we need to resample? 6357 if (mSrcSampleRate != mDstSampleRate) { 6358 if (mResampler != NULL) { 6359 delete mResampler; 6360 } 6361 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, 6362 mSrcChannelCount, mDstSampleRate); // may seem confusing... 6363 mResampler->setSampleRate(mSrcSampleRate); 6364 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6365 } 6366 return NO_ERROR; 6367} 6368 6369void AudioFlinger::RecordThread::RecordBufferConverter::convert( 6370 void *dst, /*const*/ void *src, size_t frames) 6371{ 6372 // check if a memcpy will do 6373 if (mResampler == NULL 6374 && mSrcChannelCount == mDstChannelCount 6375 && mSrcFormat == mDstFormat) { 6376 memcpy(dst, src, 6377 frames * mDstChannelCount * audio_bytes_per_sample(mDstFormat)); 6378 return; 6379 } 6380 // reallocate buffer if needed 6381 if (mBufFrameSize != 0 && mBufFrames < frames) { 6382 free(mBuf); 6383 mBufFrames = frames; 6384 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6385 } 6386 // do processing 6387 if (mResampler != NULL) { 6388 // src channel count is always >= 2. 6389 void *dstBuf = mBuf != NULL ? mBuf : dst; 6390 // ditherAndClamp() works as long as all buffers returned by 6391 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 6392 if (mDstChannelCount == 1) { 6393 // the resampler always outputs stereo samples. 6394 // FIXME: this rewrites back into src 6395 ditherAndClamp((int32_t *)src, (const int32_t *)src, frames); 6396 downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf, 6397 (const int16_t *)src, frames); 6398 } else { 6399 ditherAndClamp((int32_t *)dstBuf, (const int32_t *)src, frames); 6400 } 6401 } else if (mSrcChannelCount != mDstChannelCount) { 6402 void *dstBuf = mBuf != NULL ? mBuf : dst; 6403 if (mSrcChannelCount == 1) { 6404 upmix_to_stereo_i16_from_mono_i16((int16_t *)dstBuf, (const int16_t *)src, 6405 frames); 6406 } else { 6407 downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf, 6408 (const int16_t *)src, frames); 6409 } 6410 } 6411 if (mSrcFormat != mDstFormat) { 6412 void *srcBuf = mBuf != NULL ? mBuf : src; 6413 memcpy_by_audio_format(dst, mDstFormat, srcBuf, mSrcFormat, 6414 frames * mDstChannelCount); 6415 } 6416} 6417 6418bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6419 status_t& status) 6420{ 6421 bool reconfig = false; 6422 6423 status = NO_ERROR; 6424 6425 audio_format_t reqFormat = mFormat; 6426 uint32_t samplingRate = mSampleRate; 6427 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6428 6429 AudioParameter param = AudioParameter(keyValuePair); 6430 int value; 6431 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6432 // channel count change can be requested. Do we mandate the first client defines the 6433 // HAL sampling rate and channel count or do we allow changes on the fly? 6434 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6435 samplingRate = value; 6436 reconfig = true; 6437 } 6438 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6439 if (!audio_is_linear_pcm((audio_format_t) value)) { 6440 status = BAD_VALUE; 6441 } else { 6442 reqFormat = (audio_format_t) value; 6443 reconfig = true; 6444 } 6445 } 6446 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6447 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6448 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6449 status = BAD_VALUE; 6450 } else { 6451 channelMask = mask; 6452 reconfig = true; 6453 } 6454 } 6455 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6456 // do not accept frame count changes if tracks are open as the track buffer 6457 // size depends on frame count and correct behavior would not be guaranteed 6458 // if frame count is changed after track creation 6459 if (mActiveTracks.size() > 0) { 6460 status = INVALID_OPERATION; 6461 } else { 6462 reconfig = true; 6463 } 6464 } 6465 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6466 // forward device change to effects that have requested to be 6467 // aware of attached audio device. 6468 for (size_t i = 0; i < mEffectChains.size(); i++) { 6469 mEffectChains[i]->setDevice_l(value); 6470 } 6471 6472 // store input device and output device but do not forward output device to audio HAL. 6473 // Note that status is ignored by the caller for output device 6474 // (see AudioFlinger::setParameters() 6475 if (audio_is_output_devices(value)) { 6476 mOutDevice = value; 6477 status = BAD_VALUE; 6478 } else { 6479 mInDevice = value; 6480 // disable AEC and NS if the device is a BT SCO headset supporting those 6481 // pre processings 6482 if (mTracks.size() > 0) { 6483 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6484 mAudioFlinger->btNrecIsOff(); 6485 for (size_t i = 0; i < mTracks.size(); i++) { 6486 sp<RecordTrack> track = mTracks[i]; 6487 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6488 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6489 } 6490 } 6491 } 6492 } 6493 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6494 mAudioSource != (audio_source_t)value) { 6495 // forward device change to effects that have requested to be 6496 // aware of attached audio device. 6497 for (size_t i = 0; i < mEffectChains.size(); i++) { 6498 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6499 } 6500 mAudioSource = (audio_source_t)value; 6501 } 6502 6503 if (status == NO_ERROR) { 6504 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6505 keyValuePair.string()); 6506 if (status == INVALID_OPERATION) { 6507 inputStandBy(); 6508 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6509 keyValuePair.string()); 6510 } 6511 if (reconfig) { 6512 if (status == BAD_VALUE && 6513 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6514 audio_is_linear_pcm(reqFormat) && 6515 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6516 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6517 audio_channel_count_from_in_mask( 6518 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6519 (channelMask == AUDIO_CHANNEL_IN_MONO || 6520 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6521 status = NO_ERROR; 6522 } 6523 if (status == NO_ERROR) { 6524 readInputParameters_l(); 6525 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6526 } 6527 } 6528 } 6529 6530 return reconfig; 6531} 6532 6533String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6534{ 6535 Mutex::Autolock _l(mLock); 6536 if (initCheck() != NO_ERROR) { 6537 return String8(); 6538 } 6539 6540 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6541 const String8 out_s8(s); 6542 free(s); 6543 return out_s8; 6544} 6545 6546void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6547 AudioSystem::OutputDescriptor desc; 6548 const void *param2 = NULL; 6549 6550 switch (event) { 6551 case AudioSystem::INPUT_OPENED: 6552 case AudioSystem::INPUT_CONFIG_CHANGED: 6553 desc.channelMask = mChannelMask; 6554 desc.samplingRate = mSampleRate; 6555 desc.format = mFormat; 6556 desc.frameCount = mFrameCount; 6557 desc.latency = 0; 6558 param2 = &desc; 6559 break; 6560 6561 case AudioSystem::INPUT_CLOSED: 6562 default: 6563 break; 6564 } 6565 mAudioFlinger->audioConfigChanged(event, mId, param2); 6566} 6567 6568void AudioFlinger::RecordThread::readInputParameters_l() 6569{ 6570 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6571 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6572 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6573 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6574 mFormat = mHALFormat; 6575 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6576 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6577 } 6578 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6579 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6580 mFrameCount = mBufferSize / mFrameSize; 6581 // This is the formula for calculating the temporary buffer size. 6582 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6583 // 1 full output buffer, regardless of the alignment of the available input. 6584 // The value is somewhat arbitrary, and could probably be even larger. 6585 // A larger value should allow more old data to be read after a track calls start(), 6586 // without increasing latency. 6587 // 6588 // Note this is independent of the maximum downsampling ratio permitted for capture. 6589 mRsmpInFrames = mFrameCount * 7; 6590 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6591 delete[] mRsmpInBuffer; 6592 6593 // TODO optimize audio capture buffer sizes ... 6594 // Here we calculate the size of the sliding buffer used as a source 6595 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6596 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6597 // be better to have it derived from the pipe depth in the long term. 6598 // The current value is higher than necessary. However it should not add to latency. 6599 6600 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6601 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6602 6603 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6604 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6605} 6606 6607uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6608{ 6609 Mutex::Autolock _l(mLock); 6610 if (initCheck() != NO_ERROR) { 6611 return 0; 6612 } 6613 6614 return mInput->stream->get_input_frames_lost(mInput->stream); 6615} 6616 6617uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6618{ 6619 Mutex::Autolock _l(mLock); 6620 uint32_t result = 0; 6621 if (getEffectChain_l(sessionId) != 0) { 6622 result = EFFECT_SESSION; 6623 } 6624 6625 for (size_t i = 0; i < mTracks.size(); ++i) { 6626 if (sessionId == mTracks[i]->sessionId()) { 6627 result |= TRACK_SESSION; 6628 break; 6629 } 6630 } 6631 6632 return result; 6633} 6634 6635KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6636{ 6637 KeyedVector<int, bool> ids; 6638 Mutex::Autolock _l(mLock); 6639 for (size_t j = 0; j < mTracks.size(); ++j) { 6640 sp<RecordThread::RecordTrack> track = mTracks[j]; 6641 int sessionId = track->sessionId(); 6642 if (ids.indexOfKey(sessionId) < 0) { 6643 ids.add(sessionId, true); 6644 } 6645 } 6646 return ids; 6647} 6648 6649AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6650{ 6651 Mutex::Autolock _l(mLock); 6652 AudioStreamIn *input = mInput; 6653 mInput = NULL; 6654 return input; 6655} 6656 6657// this method must always be called either with ThreadBase mLock held or inside the thread loop 6658audio_stream_t* AudioFlinger::RecordThread::stream() const 6659{ 6660 if (mInput == NULL) { 6661 return NULL; 6662 } 6663 return &mInput->stream->common; 6664} 6665 6666status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6667{ 6668 // only one chain per input thread 6669 if (mEffectChains.size() != 0) { 6670 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6671 return INVALID_OPERATION; 6672 } 6673 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6674 chain->setThread(this); 6675 chain->setInBuffer(NULL); 6676 chain->setOutBuffer(NULL); 6677 6678 checkSuspendOnAddEffectChain_l(chain); 6679 6680 // make sure enabled pre processing effects state is communicated to the HAL as we 6681 // just moved them to a new input stream. 6682 chain->syncHalEffectsState(); 6683 6684 mEffectChains.add(chain); 6685 6686 return NO_ERROR; 6687} 6688 6689size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6690{ 6691 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6692 ALOGW_IF(mEffectChains.size() != 1, 6693 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6694 chain.get(), mEffectChains.size(), this); 6695 if (mEffectChains.size() == 1) { 6696 mEffectChains.removeAt(0); 6697 } 6698 return 0; 6699} 6700 6701status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6702 audio_patch_handle_t *handle) 6703{ 6704 status_t status = NO_ERROR; 6705 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6706 // store new device and send to effects 6707 mInDevice = patch->sources[0].ext.device.type; 6708 for (size_t i = 0; i < mEffectChains.size(); i++) { 6709 mEffectChains[i]->setDevice_l(mInDevice); 6710 } 6711 6712 // disable AEC and NS if the device is a BT SCO headset supporting those 6713 // pre processings 6714 if (mTracks.size() > 0) { 6715 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6716 mAudioFlinger->btNrecIsOff(); 6717 for (size_t i = 0; i < mTracks.size(); i++) { 6718 sp<RecordTrack> track = mTracks[i]; 6719 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6720 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6721 } 6722 } 6723 6724 // store new source and send to effects 6725 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6726 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6727 for (size_t i = 0; i < mEffectChains.size(); i++) { 6728 mEffectChains[i]->setAudioSource_l(mAudioSource); 6729 } 6730 } 6731 6732 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6733 status = hwDevice->create_audio_patch(hwDevice, 6734 patch->num_sources, 6735 patch->sources, 6736 patch->num_sinks, 6737 patch->sinks, 6738 handle); 6739 } else { 6740 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6741 } 6742 return status; 6743} 6744 6745status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6746{ 6747 status_t status = NO_ERROR; 6748 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6749 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6750 status = hwDevice->release_audio_patch(hwDevice, handle); 6751 } else { 6752 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6753 } 6754 return status; 6755} 6756 6757void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6758{ 6759 Mutex::Autolock _l(mLock); 6760 mTracks.add(record); 6761} 6762 6763void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6764{ 6765 Mutex::Autolock _l(mLock); 6766 destroyTrack_l(record); 6767} 6768 6769void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6770{ 6771 ThreadBase::getAudioPortConfig(config); 6772 config->role = AUDIO_PORT_ROLE_SINK; 6773 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6774 config->ext.mix.usecase.source = mAudioSource; 6775} 6776 6777} // namespace android 6778