Threads.cpp revision fbdb2aceab7317aa44bc8f301a93eb49e17b2bce
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#define max(a, b) ((a) > (b) ? (a) : (b)) 90 91namespace android { 92 93// retry counts for buffer fill timeout 94// 50 * ~20msecs = 1 second 95static const int8_t kMaxTrackRetries = 50; 96static const int8_t kMaxTrackStartupRetries = 50; 97// allow less retry attempts on direct output thread. 98// direct outputs can be a scarce resource in audio hardware and should 99// be released as quickly as possible. 100static const int8_t kMaxTrackRetriesDirect = 2; 101 102// don't warn about blocked writes or record buffer overflows more often than this 103static const nsecs_t kWarningThrottleNs = seconds(5); 104 105// RecordThread loop sleep time upon application overrun or audio HAL read error 106static const int kRecordThreadSleepUs = 5000; 107 108// maximum time to wait in sendConfigEvent_l() for a status to be received 109static const nsecs_t kConfigEventTimeoutNs = seconds(2); 110 111// minimum sleep time for the mixer thread loop when tracks are active but in underrun 112static const uint32_t kMinThreadSleepTimeUs = 5000; 113// maximum divider applied to the active sleep time in the mixer thread loop 114static const uint32_t kMaxThreadSleepTimeShift = 2; 115 116// minimum normal sink buffer size, expressed in milliseconds rather than frames 117static const uint32_t kMinNormalSinkBufferSizeMs = 20; 118// maximum normal sink buffer size 119static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 120 121// Offloaded output thread standby delay: allows track transition without going to standby 122static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 123 124// Whether to use fast mixer 125static const enum { 126 FastMixer_Never, // never initialize or use: for debugging only 127 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 128 // normal mixer multiplier is 1 129 FastMixer_Static, // initialize if needed, then use all the time if initialized, 130 // multiplier is calculated based on min & max normal mixer buffer size 131 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 132 // multiplier is calculated based on min & max normal mixer buffer size 133 // FIXME for FastMixer_Dynamic: 134 // Supporting this option will require fixing HALs that can't handle large writes. 135 // For example, one HAL implementation returns an error from a large write, 136 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 137 // We could either fix the HAL implementations, or provide a wrapper that breaks 138 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 139} kUseFastMixer = FastMixer_Static; 140 141// Whether to use fast capture 142static const enum { 143 FastCapture_Never, // never initialize or use: for debugging only 144 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 145 FastCapture_Static, // initialize if needed, then use all the time if initialized 146} kUseFastCapture = FastCapture_Static; 147 148// Priorities for requestPriority 149static const int kPriorityAudioApp = 2; 150static const int kPriorityFastMixer = 3; 151static const int kPriorityFastCapture = 3; 152 153// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 154// for the track. The client then sub-divides this into smaller buffers for its use. 155// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 156// So for now we just assume that client is double-buffered for fast tracks. 157// FIXME It would be better for client to tell AudioFlinger the value of N, 158// so AudioFlinger could allocate the right amount of memory. 159// See the client's minBufCount and mNotificationFramesAct calculations for details. 160 161// This is the default value, if not specified by property. 162static const int kFastTrackMultiplier = 2; 163 164// The minimum and maximum allowed values 165static const int kFastTrackMultiplierMin = 1; 166static const int kFastTrackMultiplierMax = 2; 167 168// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 169static int sFastTrackMultiplier = kFastTrackMultiplier; 170 171// See Thread::readOnlyHeap(). 172// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 173// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 174// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 175static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 176 177// ---------------------------------------------------------------------------- 178 179static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 180 181static void sFastTrackMultiplierInit() 182{ 183 char value[PROPERTY_VALUE_MAX]; 184 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 185 char *endptr; 186 unsigned long ul = strtoul(value, &endptr, 0); 187 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 188 sFastTrackMultiplier = (int) ul; 189 } 190 } 191} 192 193// ---------------------------------------------------------------------------- 194 195#ifdef ADD_BATTERY_DATA 196// To collect the amplifier usage 197static void addBatteryData(uint32_t params) { 198 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 199 if (service == NULL) { 200 // it already logged 201 return; 202 } 203 204 service->addBatteryData(params); 205} 206#endif 207 208 209// ---------------------------------------------------------------------------- 210// CPU Stats 211// ---------------------------------------------------------------------------- 212 213class CpuStats { 214public: 215 CpuStats(); 216 void sample(const String8 &title); 217#ifdef DEBUG_CPU_USAGE 218private: 219 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 220 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 221 222 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 223 224 int mCpuNum; // thread's current CPU number 225 int mCpukHz; // frequency of thread's current CPU in kHz 226#endif 227}; 228 229CpuStats::CpuStats() 230#ifdef DEBUG_CPU_USAGE 231 : mCpuNum(-1), mCpukHz(-1) 232#endif 233{ 234} 235 236void CpuStats::sample(const String8 &title 237#ifndef DEBUG_CPU_USAGE 238 __unused 239#endif 240 ) { 241#ifdef DEBUG_CPU_USAGE 242 // get current thread's delta CPU time in wall clock ns 243 double wcNs; 244 bool valid = mCpuUsage.sampleAndEnable(wcNs); 245 246 // record sample for wall clock statistics 247 if (valid) { 248 mWcStats.sample(wcNs); 249 } 250 251 // get the current CPU number 252 int cpuNum = sched_getcpu(); 253 254 // get the current CPU frequency in kHz 255 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 256 257 // check if either CPU number or frequency changed 258 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 259 mCpuNum = cpuNum; 260 mCpukHz = cpukHz; 261 // ignore sample for purposes of cycles 262 valid = false; 263 } 264 265 // if no change in CPU number or frequency, then record sample for cycle statistics 266 if (valid && mCpukHz > 0) { 267 double cycles = wcNs * cpukHz * 0.000001; 268 mHzStats.sample(cycles); 269 } 270 271 unsigned n = mWcStats.n(); 272 // mCpuUsage.elapsed() is expensive, so don't call it every loop 273 if ((n & 127) == 1) { 274 long long elapsed = mCpuUsage.elapsed(); 275 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 276 double perLoop = elapsed / (double) n; 277 double perLoop100 = perLoop * 0.01; 278 double perLoop1k = perLoop * 0.001; 279 double mean = mWcStats.mean(); 280 double stddev = mWcStats.stddev(); 281 double minimum = mWcStats.minimum(); 282 double maximum = mWcStats.maximum(); 283 double meanCycles = mHzStats.mean(); 284 double stddevCycles = mHzStats.stddev(); 285 double minCycles = mHzStats.minimum(); 286 double maxCycles = mHzStats.maximum(); 287 mCpuUsage.resetElapsed(); 288 mWcStats.reset(); 289 mHzStats.reset(); 290 ALOGD("CPU usage for %s over past %.1f secs\n" 291 " (%u mixer loops at %.1f mean ms per loop):\n" 292 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 293 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 294 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 295 title.string(), 296 elapsed * .000000001, n, perLoop * .000001, 297 mean * .001, 298 stddev * .001, 299 minimum * .001, 300 maximum * .001, 301 mean / perLoop100, 302 stddev / perLoop100, 303 minimum / perLoop100, 304 maximum / perLoop100, 305 meanCycles / perLoop1k, 306 stddevCycles / perLoop1k, 307 minCycles / perLoop1k, 308 maxCycles / perLoop1k); 309 310 } 311 } 312#endif 313}; 314 315// ---------------------------------------------------------------------------- 316// ThreadBase 317// ---------------------------------------------------------------------------- 318 319// static 320const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 321{ 322 switch (type) { 323 case MIXER: 324 return "MIXER"; 325 case DIRECT: 326 return "DIRECT"; 327 case DUPLICATING: 328 return "DUPLICATING"; 329 case RECORD: 330 return "RECORD"; 331 case OFFLOAD: 332 return "OFFLOAD"; 333 default: 334 return "unknown"; 335 } 336} 337 338static String8 outputFlagsToString(audio_output_flags_t flags) 339{ 340 static const struct mapping { 341 audio_output_flags_t mFlag; 342 const char * mString; 343 } mappings[] = { 344 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 345 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 346 AUDIO_OUTPUT_FLAG_FAST, "FAST", 347 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 348 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAAD", 349 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 350 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 351 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 352 }; 353 String8 result; 354 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 355 const mapping *entry; 356 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 357 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 358 if (flags & entry->mFlag) { 359 if (!result.isEmpty()) { 360 result.append("|"); 361 } 362 result.append(entry->mString); 363 } 364 } 365 if (flags & ~allFlags) { 366 if (!result.isEmpty()) { 367 result.append("|"); 368 } 369 result.appendFormat("0x%X", flags & ~allFlags); 370 } 371 if (result.isEmpty()) { 372 result.append(entry->mString); 373 } 374 return result; 375} 376 377AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 378 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 379 : Thread(false /*canCallJava*/), 380 mType(type), 381 mAudioFlinger(audioFlinger), 382 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 383 // are set by PlaybackThread::readOutputParameters_l() or 384 // RecordThread::readInputParameters_l() 385 //FIXME: mStandby should be true here. Is this some kind of hack? 386 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 387 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 388 // mName will be set by concrete (non-virtual) subclass 389 mDeathRecipient(new PMDeathRecipient(this)) 390{ 391} 392 393AudioFlinger::ThreadBase::~ThreadBase() 394{ 395 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 396 mConfigEvents.clear(); 397 398 // do not lock the mutex in destructor 399 releaseWakeLock_l(); 400 if (mPowerManager != 0) { 401 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 402 binder->unlinkToDeath(mDeathRecipient); 403 } 404} 405 406status_t AudioFlinger::ThreadBase::readyToRun() 407{ 408 status_t status = initCheck(); 409 if (status == NO_ERROR) { 410 ALOGI("AudioFlinger's thread %p ready to run", this); 411 } else { 412 ALOGE("No working audio driver found."); 413 } 414 return status; 415} 416 417void AudioFlinger::ThreadBase::exit() 418{ 419 ALOGV("ThreadBase::exit"); 420 // do any cleanup required for exit to succeed 421 preExit(); 422 { 423 // This lock prevents the following race in thread (uniprocessor for illustration): 424 // if (!exitPending()) { 425 // // context switch from here to exit() 426 // // exit() calls requestExit(), what exitPending() observes 427 // // exit() calls signal(), which is dropped since no waiters 428 // // context switch back from exit() to here 429 // mWaitWorkCV.wait(...); 430 // // now thread is hung 431 // } 432 AutoMutex lock(mLock); 433 requestExit(); 434 mWaitWorkCV.broadcast(); 435 } 436 // When Thread::requestExitAndWait is made virtual and this method is renamed to 437 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 438 requestExitAndWait(); 439} 440 441status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 442{ 443 status_t status; 444 445 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 446 Mutex::Autolock _l(mLock); 447 448 return sendSetParameterConfigEvent_l(keyValuePairs); 449} 450 451// sendConfigEvent_l() must be called with ThreadBase::mLock held 452// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 453status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 454{ 455 status_t status = NO_ERROR; 456 457 mConfigEvents.add(event); 458 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 459 mWaitWorkCV.signal(); 460 mLock.unlock(); 461 { 462 Mutex::Autolock _l(event->mLock); 463 while (event->mWaitStatus) { 464 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 465 event->mStatus = TIMED_OUT; 466 event->mWaitStatus = false; 467 } 468 } 469 status = event->mStatus; 470 } 471 mLock.lock(); 472 return status; 473} 474 475void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 476{ 477 Mutex::Autolock _l(mLock); 478 sendIoConfigEvent_l(event, param); 479} 480 481// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 482void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 483{ 484 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 485 sendConfigEvent_l(configEvent); 486} 487 488// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 489void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 490{ 491 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 492 sendConfigEvent_l(configEvent); 493} 494 495// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 496status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 497{ 498 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 499 return sendConfigEvent_l(configEvent); 500} 501 502status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 503 const struct audio_patch *patch, 504 audio_patch_handle_t *handle) 505{ 506 Mutex::Autolock _l(mLock); 507 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 508 status_t status = sendConfigEvent_l(configEvent); 509 if (status == NO_ERROR) { 510 CreateAudioPatchConfigEventData *data = 511 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 512 *handle = data->mHandle; 513 } 514 return status; 515} 516 517status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 518 const audio_patch_handle_t handle) 519{ 520 Mutex::Autolock _l(mLock); 521 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 522 return sendConfigEvent_l(configEvent); 523} 524 525 526// post condition: mConfigEvents.isEmpty() 527void AudioFlinger::ThreadBase::processConfigEvents_l() 528{ 529 bool configChanged = false; 530 531 while (!mConfigEvents.isEmpty()) { 532 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 533 sp<ConfigEvent> event = mConfigEvents[0]; 534 mConfigEvents.removeAt(0); 535 switch (event->mType) { 536 case CFG_EVENT_PRIO: { 537 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 538 // FIXME Need to understand why this has to be done asynchronously 539 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 540 true /*asynchronous*/); 541 if (err != 0) { 542 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 543 data->mPrio, data->mPid, data->mTid, err); 544 } 545 } break; 546 case CFG_EVENT_IO: { 547 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 548 audioConfigChanged(data->mEvent, data->mParam); 549 } break; 550 case CFG_EVENT_SET_PARAMETER: { 551 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 552 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 553 configChanged = true; 554 } 555 } break; 556 case CFG_EVENT_CREATE_AUDIO_PATCH: { 557 CreateAudioPatchConfigEventData *data = 558 (CreateAudioPatchConfigEventData *)event->mData.get(); 559 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 560 } break; 561 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 562 ReleaseAudioPatchConfigEventData *data = 563 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 564 event->mStatus = releaseAudioPatch_l(data->mHandle); 565 } break; 566 default: 567 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 568 break; 569 } 570 { 571 Mutex::Autolock _l(event->mLock); 572 if (event->mWaitStatus) { 573 event->mWaitStatus = false; 574 event->mCond.signal(); 575 } 576 } 577 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 578 } 579 580 if (configChanged) { 581 cacheParameters_l(); 582 } 583} 584 585String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 586 String8 s; 587 if (output) { 588 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 589 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 590 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 591 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 592 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 593 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 594 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 595 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 596 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 597 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 598 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 599 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 600 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 601 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 602 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 603 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 604 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 605 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 606 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 607 } else { 608 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 609 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 610 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 611 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 612 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 613 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 614 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 615 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 616 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 617 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 618 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 619 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 620 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 621 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 622 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 623 } 624 int len = s.length(); 625 if (s.length() > 2) { 626 char *str = s.lockBuffer(len); 627 s.unlockBuffer(len - 2); 628 } 629 return s; 630} 631 632void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 633{ 634 const size_t SIZE = 256; 635 char buffer[SIZE]; 636 String8 result; 637 638 bool locked = AudioFlinger::dumpTryLock(mLock); 639 if (!locked) { 640 dprintf(fd, "thread %p may be deadlocked\n", this); 641 } 642 643 dprintf(fd, " I/O handle: %d\n", mId); 644 dprintf(fd, " TID: %d\n", getTid()); 645 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 646 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 647 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 648 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 649 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 650 dprintf(fd, " Channel count: %u\n", mChannelCount); 651 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 652 channelMaskToString(mChannelMask, mType != RECORD).string()); 653 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 654 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 655 dprintf(fd, " Pending config events:"); 656 size_t numConfig = mConfigEvents.size(); 657 if (numConfig) { 658 for (size_t i = 0; i < numConfig; i++) { 659 mConfigEvents[i]->dump(buffer, SIZE); 660 dprintf(fd, "\n %s", buffer); 661 } 662 dprintf(fd, "\n"); 663 } else { 664 dprintf(fd, " none\n"); 665 } 666 667 if (locked) { 668 mLock.unlock(); 669 } 670} 671 672void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 673{ 674 const size_t SIZE = 256; 675 char buffer[SIZE]; 676 String8 result; 677 678 size_t numEffectChains = mEffectChains.size(); 679 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 680 write(fd, buffer, strlen(buffer)); 681 682 for (size_t i = 0; i < numEffectChains; ++i) { 683 sp<EffectChain> chain = mEffectChains[i]; 684 if (chain != 0) { 685 chain->dump(fd, args); 686 } 687 } 688} 689 690void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 691{ 692 Mutex::Autolock _l(mLock); 693 acquireWakeLock_l(uid); 694} 695 696String16 AudioFlinger::ThreadBase::getWakeLockTag() 697{ 698 switch (mType) { 699 case MIXER: 700 return String16("AudioMix"); 701 case DIRECT: 702 return String16("AudioDirectOut"); 703 case DUPLICATING: 704 return String16("AudioDup"); 705 case RECORD: 706 return String16("AudioIn"); 707 case OFFLOAD: 708 return String16("AudioOffload"); 709 default: 710 ALOG_ASSERT(false); 711 return String16("AudioUnknown"); 712 } 713} 714 715void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 716{ 717 getPowerManager_l(); 718 if (mPowerManager != 0) { 719 sp<IBinder> binder = new BBinder(); 720 status_t status; 721 if (uid >= 0) { 722 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 723 binder, 724 getWakeLockTag(), 725 String16("media"), 726 uid, 727 true /* FIXME force oneway contrary to .aidl */); 728 } else { 729 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 730 binder, 731 getWakeLockTag(), 732 String16("media"), 733 true /* FIXME force oneway contrary to .aidl */); 734 } 735 if (status == NO_ERROR) { 736 mWakeLockToken = binder; 737 } 738 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 739 } 740} 741 742void AudioFlinger::ThreadBase::releaseWakeLock() 743{ 744 Mutex::Autolock _l(mLock); 745 releaseWakeLock_l(); 746} 747 748void AudioFlinger::ThreadBase::releaseWakeLock_l() 749{ 750 if (mWakeLockToken != 0) { 751 ALOGV("releaseWakeLock_l() %s", mName); 752 if (mPowerManager != 0) { 753 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 754 true /* FIXME force oneway contrary to .aidl */); 755 } 756 mWakeLockToken.clear(); 757 } 758} 759 760void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 761 Mutex::Autolock _l(mLock); 762 updateWakeLockUids_l(uids); 763} 764 765void AudioFlinger::ThreadBase::getPowerManager_l() { 766 767 if (mPowerManager == 0) { 768 // use checkService() to avoid blocking if power service is not up yet 769 sp<IBinder> binder = 770 defaultServiceManager()->checkService(String16("power")); 771 if (binder == 0) { 772 ALOGW("Thread %s cannot connect to the power manager service", mName); 773 } else { 774 mPowerManager = interface_cast<IPowerManager>(binder); 775 binder->linkToDeath(mDeathRecipient); 776 } 777 } 778} 779 780void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 781 782 getPowerManager_l(); 783 if (mWakeLockToken == NULL) { 784 ALOGE("no wake lock to update!"); 785 return; 786 } 787 if (mPowerManager != 0) { 788 sp<IBinder> binder = new BBinder(); 789 status_t status; 790 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 791 true /* FIXME force oneway contrary to .aidl */); 792 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 793 } 794} 795 796void AudioFlinger::ThreadBase::clearPowerManager() 797{ 798 Mutex::Autolock _l(mLock); 799 releaseWakeLock_l(); 800 mPowerManager.clear(); 801} 802 803void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 804{ 805 sp<ThreadBase> thread = mThread.promote(); 806 if (thread != 0) { 807 thread->clearPowerManager(); 808 } 809 ALOGW("power manager service died !!!"); 810} 811 812void AudioFlinger::ThreadBase::setEffectSuspended( 813 const effect_uuid_t *type, bool suspend, int sessionId) 814{ 815 Mutex::Autolock _l(mLock); 816 setEffectSuspended_l(type, suspend, sessionId); 817} 818 819void AudioFlinger::ThreadBase::setEffectSuspended_l( 820 const effect_uuid_t *type, bool suspend, int sessionId) 821{ 822 sp<EffectChain> chain = getEffectChain_l(sessionId); 823 if (chain != 0) { 824 if (type != NULL) { 825 chain->setEffectSuspended_l(type, suspend); 826 } else { 827 chain->setEffectSuspendedAll_l(suspend); 828 } 829 } 830 831 updateSuspendedSessions_l(type, suspend, sessionId); 832} 833 834void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 835{ 836 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 837 if (index < 0) { 838 return; 839 } 840 841 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 842 mSuspendedSessions.valueAt(index); 843 844 for (size_t i = 0; i < sessionEffects.size(); i++) { 845 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 846 for (int j = 0; j < desc->mRefCount; j++) { 847 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 848 chain->setEffectSuspendedAll_l(true); 849 } else { 850 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 851 desc->mType.timeLow); 852 chain->setEffectSuspended_l(&desc->mType, true); 853 } 854 } 855 } 856} 857 858void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 859 bool suspend, 860 int sessionId) 861{ 862 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 863 864 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 865 866 if (suspend) { 867 if (index >= 0) { 868 sessionEffects = mSuspendedSessions.valueAt(index); 869 } else { 870 mSuspendedSessions.add(sessionId, sessionEffects); 871 } 872 } else { 873 if (index < 0) { 874 return; 875 } 876 sessionEffects = mSuspendedSessions.valueAt(index); 877 } 878 879 880 int key = EffectChain::kKeyForSuspendAll; 881 if (type != NULL) { 882 key = type->timeLow; 883 } 884 index = sessionEffects.indexOfKey(key); 885 886 sp<SuspendedSessionDesc> desc; 887 if (suspend) { 888 if (index >= 0) { 889 desc = sessionEffects.valueAt(index); 890 } else { 891 desc = new SuspendedSessionDesc(); 892 if (type != NULL) { 893 desc->mType = *type; 894 } 895 sessionEffects.add(key, desc); 896 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 897 } 898 desc->mRefCount++; 899 } else { 900 if (index < 0) { 901 return; 902 } 903 desc = sessionEffects.valueAt(index); 904 if (--desc->mRefCount == 0) { 905 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 906 sessionEffects.removeItemsAt(index); 907 if (sessionEffects.isEmpty()) { 908 ALOGV("updateSuspendedSessions_l() restore removing session %d", 909 sessionId); 910 mSuspendedSessions.removeItem(sessionId); 911 } 912 } 913 } 914 if (!sessionEffects.isEmpty()) { 915 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 916 } 917} 918 919void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 920 bool enabled, 921 int sessionId) 922{ 923 Mutex::Autolock _l(mLock); 924 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 925} 926 927void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 928 bool enabled, 929 int sessionId) 930{ 931 if (mType != RECORD) { 932 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 933 // another session. This gives the priority to well behaved effect control panels 934 // and applications not using global effects. 935 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 936 // global effects 937 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 938 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 939 } 940 } 941 942 sp<EffectChain> chain = getEffectChain_l(sessionId); 943 if (chain != 0) { 944 chain->checkSuspendOnEffectEnabled(effect, enabled); 945 } 946} 947 948// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 949sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 950 const sp<AudioFlinger::Client>& client, 951 const sp<IEffectClient>& effectClient, 952 int32_t priority, 953 int sessionId, 954 effect_descriptor_t *desc, 955 int *enabled, 956 status_t *status) 957{ 958 sp<EffectModule> effect; 959 sp<EffectHandle> handle; 960 status_t lStatus; 961 sp<EffectChain> chain; 962 bool chainCreated = false; 963 bool effectCreated = false; 964 bool effectRegistered = false; 965 966 lStatus = initCheck(); 967 if (lStatus != NO_ERROR) { 968 ALOGW("createEffect_l() Audio driver not initialized."); 969 goto Exit; 970 } 971 972 // Reject any effect on Direct output threads for now, since the format of 973 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 974 if (mType == DIRECT) { 975 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 976 desc->name, mName); 977 lStatus = BAD_VALUE; 978 goto Exit; 979 } 980 981 // Reject any effect on mixer or duplicating multichannel sinks. 982 // TODO: fix both format and multichannel issues with effects. 983 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 984 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 985 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 986 lStatus = BAD_VALUE; 987 goto Exit; 988 } 989 990 // Allow global effects only on offloaded and mixer threads 991 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 992 switch (mType) { 993 case MIXER: 994 case OFFLOAD: 995 break; 996 case DIRECT: 997 case DUPLICATING: 998 case RECORD: 999 default: 1000 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 1001 lStatus = BAD_VALUE; 1002 goto Exit; 1003 } 1004 } 1005 1006 // Only Pre processor effects are allowed on input threads and only on input threads 1007 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1008 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1009 desc->name, desc->flags, mType); 1010 lStatus = BAD_VALUE; 1011 goto Exit; 1012 } 1013 1014 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1015 1016 { // scope for mLock 1017 Mutex::Autolock _l(mLock); 1018 1019 // check for existing effect chain with the requested audio session 1020 chain = getEffectChain_l(sessionId); 1021 if (chain == 0) { 1022 // create a new chain for this session 1023 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1024 chain = new EffectChain(this, sessionId); 1025 addEffectChain_l(chain); 1026 chain->setStrategy(getStrategyForSession_l(sessionId)); 1027 chainCreated = true; 1028 } else { 1029 effect = chain->getEffectFromDesc_l(desc); 1030 } 1031 1032 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1033 1034 if (effect == 0) { 1035 int id = mAudioFlinger->nextUniqueId(); 1036 // Check CPU and memory usage 1037 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1038 if (lStatus != NO_ERROR) { 1039 goto Exit; 1040 } 1041 effectRegistered = true; 1042 // create a new effect module if none present in the chain 1043 effect = new EffectModule(this, chain, desc, id, sessionId); 1044 lStatus = effect->status(); 1045 if (lStatus != NO_ERROR) { 1046 goto Exit; 1047 } 1048 effect->setOffloaded(mType == OFFLOAD, mId); 1049 1050 lStatus = chain->addEffect_l(effect); 1051 if (lStatus != NO_ERROR) { 1052 goto Exit; 1053 } 1054 effectCreated = true; 1055 1056 effect->setDevice(mOutDevice); 1057 effect->setDevice(mInDevice); 1058 effect->setMode(mAudioFlinger->getMode()); 1059 effect->setAudioSource(mAudioSource); 1060 } 1061 // create effect handle and connect it to effect module 1062 handle = new EffectHandle(effect, client, effectClient, priority); 1063 lStatus = handle->initCheck(); 1064 if (lStatus == OK) { 1065 lStatus = effect->addHandle(handle.get()); 1066 } 1067 if (enabled != NULL) { 1068 *enabled = (int)effect->isEnabled(); 1069 } 1070 } 1071 1072Exit: 1073 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1074 Mutex::Autolock _l(mLock); 1075 if (effectCreated) { 1076 chain->removeEffect_l(effect); 1077 } 1078 if (effectRegistered) { 1079 AudioSystem::unregisterEffect(effect->id()); 1080 } 1081 if (chainCreated) { 1082 removeEffectChain_l(chain); 1083 } 1084 handle.clear(); 1085 } 1086 1087 *status = lStatus; 1088 return handle; 1089} 1090 1091sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1092{ 1093 Mutex::Autolock _l(mLock); 1094 return getEffect_l(sessionId, effectId); 1095} 1096 1097sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1098{ 1099 sp<EffectChain> chain = getEffectChain_l(sessionId); 1100 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1101} 1102 1103// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1104// PlaybackThread::mLock held 1105status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1106{ 1107 // check for existing effect chain with the requested audio session 1108 int sessionId = effect->sessionId(); 1109 sp<EffectChain> chain = getEffectChain_l(sessionId); 1110 bool chainCreated = false; 1111 1112 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1113 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1114 this, effect->desc().name, effect->desc().flags); 1115 1116 if (chain == 0) { 1117 // create a new chain for this session 1118 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1119 chain = new EffectChain(this, sessionId); 1120 addEffectChain_l(chain); 1121 chain->setStrategy(getStrategyForSession_l(sessionId)); 1122 chainCreated = true; 1123 } 1124 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1125 1126 if (chain->getEffectFromId_l(effect->id()) != 0) { 1127 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1128 this, effect->desc().name, chain.get()); 1129 return BAD_VALUE; 1130 } 1131 1132 effect->setOffloaded(mType == OFFLOAD, mId); 1133 1134 status_t status = chain->addEffect_l(effect); 1135 if (status != NO_ERROR) { 1136 if (chainCreated) { 1137 removeEffectChain_l(chain); 1138 } 1139 return status; 1140 } 1141 1142 effect->setDevice(mOutDevice); 1143 effect->setDevice(mInDevice); 1144 effect->setMode(mAudioFlinger->getMode()); 1145 effect->setAudioSource(mAudioSource); 1146 return NO_ERROR; 1147} 1148 1149void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1150 1151 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1152 effect_descriptor_t desc = effect->desc(); 1153 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1154 detachAuxEffect_l(effect->id()); 1155 } 1156 1157 sp<EffectChain> chain = effect->chain().promote(); 1158 if (chain != 0) { 1159 // remove effect chain if removing last effect 1160 if (chain->removeEffect_l(effect) == 0) { 1161 removeEffectChain_l(chain); 1162 } 1163 } else { 1164 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1165 } 1166} 1167 1168void AudioFlinger::ThreadBase::lockEffectChains_l( 1169 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1170{ 1171 effectChains = mEffectChains; 1172 for (size_t i = 0; i < mEffectChains.size(); i++) { 1173 mEffectChains[i]->lock(); 1174 } 1175} 1176 1177void AudioFlinger::ThreadBase::unlockEffectChains( 1178 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1179{ 1180 for (size_t i = 0; i < effectChains.size(); i++) { 1181 effectChains[i]->unlock(); 1182 } 1183} 1184 1185sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1186{ 1187 Mutex::Autolock _l(mLock); 1188 return getEffectChain_l(sessionId); 1189} 1190 1191sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1192{ 1193 size_t size = mEffectChains.size(); 1194 for (size_t i = 0; i < size; i++) { 1195 if (mEffectChains[i]->sessionId() == sessionId) { 1196 return mEffectChains[i]; 1197 } 1198 } 1199 return 0; 1200} 1201 1202void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1203{ 1204 Mutex::Autolock _l(mLock); 1205 size_t size = mEffectChains.size(); 1206 for (size_t i = 0; i < size; i++) { 1207 mEffectChains[i]->setMode_l(mode); 1208 } 1209} 1210 1211void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1212{ 1213 config->type = AUDIO_PORT_TYPE_MIX; 1214 config->ext.mix.handle = mId; 1215 config->sample_rate = mSampleRate; 1216 config->format = mFormat; 1217 config->channel_mask = mChannelMask; 1218 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1219 AUDIO_PORT_CONFIG_FORMAT; 1220} 1221 1222 1223// ---------------------------------------------------------------------------- 1224// Playback 1225// ---------------------------------------------------------------------------- 1226 1227AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1228 AudioStreamOut* output, 1229 audio_io_handle_t id, 1230 audio_devices_t device, 1231 type_t type) 1232 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1233 mNormalFrameCount(0), mSinkBuffer(NULL), 1234 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1235 mMixerBuffer(NULL), 1236 mMixerBufferSize(0), 1237 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1238 mMixerBufferValid(false), 1239 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1240 mEffectBuffer(NULL), 1241 mEffectBufferSize(0), 1242 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1243 mEffectBufferValid(false), 1244 mSuspended(0), mBytesWritten(0), 1245 mActiveTracksGeneration(0), 1246 // mStreamTypes[] initialized in constructor body 1247 mOutput(output), 1248 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1249 mMixerStatus(MIXER_IDLE), 1250 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1251 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1252 mBytesRemaining(0), 1253 mCurrentWriteLength(0), 1254 mUseAsyncWrite(false), 1255 mWriteAckSequence(0), 1256 mDrainSequence(0), 1257 mSignalPending(false), 1258 mScreenState(AudioFlinger::mScreenState), 1259 // index 0 is reserved for normal mixer's submix 1260 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1261 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1262 // mLatchD, mLatchQ, 1263 mLatchDValid(false), mLatchQValid(false) 1264{ 1265 snprintf(mName, kNameLength, "AudioOut_%X", id); 1266 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1267 1268 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1269 // it would be safer to explicitly pass initial masterVolume/masterMute as 1270 // parameter. 1271 // 1272 // If the HAL we are using has support for master volume or master mute, 1273 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1274 // and the mute set to false). 1275 mMasterVolume = audioFlinger->masterVolume_l(); 1276 mMasterMute = audioFlinger->masterMute_l(); 1277 if (mOutput && mOutput->audioHwDev) { 1278 if (mOutput->audioHwDev->canSetMasterVolume()) { 1279 mMasterVolume = 1.0; 1280 } 1281 1282 if (mOutput->audioHwDev->canSetMasterMute()) { 1283 mMasterMute = false; 1284 } 1285 } 1286 1287 readOutputParameters_l(); 1288 1289 // ++ operator does not compile 1290 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1291 stream = (audio_stream_type_t) (stream + 1)) { 1292 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1293 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1294 } 1295} 1296 1297AudioFlinger::PlaybackThread::~PlaybackThread() 1298{ 1299 mAudioFlinger->unregisterWriter(mNBLogWriter); 1300 free(mSinkBuffer); 1301 free(mMixerBuffer); 1302 free(mEffectBuffer); 1303} 1304 1305void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1306{ 1307 dumpInternals(fd, args); 1308 dumpTracks(fd, args); 1309 dumpEffectChains(fd, args); 1310} 1311 1312void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1313{ 1314 const size_t SIZE = 256; 1315 char buffer[SIZE]; 1316 String8 result; 1317 1318 result.appendFormat(" Stream volumes in dB: "); 1319 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1320 const stream_type_t *st = &mStreamTypes[i]; 1321 if (i > 0) { 1322 result.appendFormat(", "); 1323 } 1324 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1325 if (st->mute) { 1326 result.append("M"); 1327 } 1328 } 1329 result.append("\n"); 1330 write(fd, result.string(), result.length()); 1331 result.clear(); 1332 1333 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1334 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1335 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1336 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1337 1338 size_t numtracks = mTracks.size(); 1339 size_t numactive = mActiveTracks.size(); 1340 dprintf(fd, " %d Tracks", numtracks); 1341 size_t numactiveseen = 0; 1342 if (numtracks) { 1343 dprintf(fd, " of which %d are active\n", numactive); 1344 Track::appendDumpHeader(result); 1345 for (size_t i = 0; i < numtracks; ++i) { 1346 sp<Track> track = mTracks[i]; 1347 if (track != 0) { 1348 bool active = mActiveTracks.indexOf(track) >= 0; 1349 if (active) { 1350 numactiveseen++; 1351 } 1352 track->dump(buffer, SIZE, active); 1353 result.append(buffer); 1354 } 1355 } 1356 } else { 1357 result.append("\n"); 1358 } 1359 if (numactiveseen != numactive) { 1360 // some tracks in the active list were not in the tracks list 1361 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1362 " not in the track list\n"); 1363 result.append(buffer); 1364 Track::appendDumpHeader(result); 1365 for (size_t i = 0; i < numactive; ++i) { 1366 sp<Track> track = mActiveTracks[i].promote(); 1367 if (track != 0 && mTracks.indexOf(track) < 0) { 1368 track->dump(buffer, SIZE, true); 1369 result.append(buffer); 1370 } 1371 } 1372 } 1373 1374 write(fd, result.string(), result.size()); 1375} 1376 1377void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1378{ 1379 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1380 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1381 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1382 dprintf(fd, " Total writes: %d\n", mNumWrites); 1383 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1384 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1385 dprintf(fd, " Suspend count: %d\n", mSuspended); 1386 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1387 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1388 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1389 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1390 AudioStreamOut *output = mOutput; 1391 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1392 String8 flagsAsString = outputFlagsToString(flags); 1393 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1394 1395 dumpBase(fd, args); 1396} 1397 1398// Thread virtuals 1399 1400void AudioFlinger::PlaybackThread::onFirstRef() 1401{ 1402 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1403} 1404 1405// ThreadBase virtuals 1406void AudioFlinger::PlaybackThread::preExit() 1407{ 1408 ALOGV(" preExit()"); 1409 // FIXME this is using hard-coded strings but in the future, this functionality will be 1410 // converted to use audio HAL extensions required to support tunneling 1411 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1412} 1413 1414// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1415sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1416 const sp<AudioFlinger::Client>& client, 1417 audio_stream_type_t streamType, 1418 uint32_t sampleRate, 1419 audio_format_t format, 1420 audio_channel_mask_t channelMask, 1421 size_t *pFrameCount, 1422 const sp<IMemory>& sharedBuffer, 1423 int sessionId, 1424 IAudioFlinger::track_flags_t *flags, 1425 pid_t tid, 1426 int uid, 1427 status_t *status) 1428{ 1429 size_t frameCount = *pFrameCount; 1430 sp<Track> track; 1431 status_t lStatus; 1432 1433 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1434 1435 // client expresses a preference for FAST, but we get the final say 1436 if (*flags & IAudioFlinger::TRACK_FAST) { 1437 if ( 1438 // not timed 1439 (!isTimed) && 1440 // either of these use cases: 1441 ( 1442 // use case 1: shared buffer with any frame count 1443 ( 1444 (sharedBuffer != 0) 1445 ) || 1446 // use case 2: callback handler and frame count is default or at least as large as HAL 1447 ( 1448 (tid != -1) && 1449 ((frameCount == 0) || 1450 (frameCount >= mFrameCount)) 1451 ) 1452 ) && 1453 // PCM data 1454 audio_is_linear_pcm(format) && 1455 // identical channel mask to sink, or mono in and stereo sink 1456 (channelMask == mChannelMask || 1457 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1458 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1459 // hardware sample rate 1460 (sampleRate == mSampleRate) && 1461 // normal mixer has an associated fast mixer 1462 hasFastMixer() && 1463 // there are sufficient fast track slots available 1464 (mFastTrackAvailMask != 0) 1465 // FIXME test that MixerThread for this fast track has a capable output HAL 1466 // FIXME add a permission test also? 1467 ) { 1468 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1469 if (frameCount == 0) { 1470 // read the fast track multiplier property the first time it is needed 1471 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1472 if (ok != 0) { 1473 ALOGE("%s pthread_once failed: %d", __func__, ok); 1474 } 1475 frameCount = mFrameCount * sFastTrackMultiplier; 1476 } 1477 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1478 frameCount, mFrameCount); 1479 } else { 1480 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1481 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1482 "sampleRate=%u mSampleRate=%u " 1483 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1484 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1485 audio_is_linear_pcm(format), 1486 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1487 *flags &= ~IAudioFlinger::TRACK_FAST; 1488 } 1489 } 1490 // For normal PCM streaming tracks, update minimum frame count. 1491 // For compatibility with AudioTrack calculation, buffer depth is forced 1492 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1493 // This is probably too conservative, but legacy application code may depend on it. 1494 // If you change this calculation, also review the start threshold which is related. 1495 if (!(*flags & IAudioFlinger::TRACK_FAST) 1496 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1497 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1498 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1499 if (minBufCount < 2) { 1500 minBufCount = 2; 1501 } 1502 size_t minFrameCount = 1503 minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate); 1504 if (frameCount < minFrameCount) { // including frameCount == 0 1505 frameCount = minFrameCount; 1506 } 1507 } 1508 *pFrameCount = frameCount; 1509 1510 switch (mType) { 1511 1512 case DIRECT: 1513 if (audio_is_linear_pcm(format)) { 1514 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1515 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1516 "for output %p with format %#x", 1517 sampleRate, format, channelMask, mOutput, mFormat); 1518 lStatus = BAD_VALUE; 1519 goto Exit; 1520 } 1521 } 1522 break; 1523 1524 case OFFLOAD: 1525 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1526 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1527 "for output %p with format %#x", 1528 sampleRate, format, channelMask, mOutput, mFormat); 1529 lStatus = BAD_VALUE; 1530 goto Exit; 1531 } 1532 break; 1533 1534 default: 1535 if (!audio_is_linear_pcm(format)) { 1536 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1537 "for output %p with format %#x", 1538 format, mOutput, mFormat); 1539 lStatus = BAD_VALUE; 1540 goto Exit; 1541 } 1542 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1543 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1544 lStatus = BAD_VALUE; 1545 goto Exit; 1546 } 1547 break; 1548 1549 } 1550 1551 lStatus = initCheck(); 1552 if (lStatus != NO_ERROR) { 1553 ALOGE("createTrack_l() audio driver not initialized"); 1554 goto Exit; 1555 } 1556 1557 { // scope for mLock 1558 Mutex::Autolock _l(mLock); 1559 1560 // all tracks in same audio session must share the same routing strategy otherwise 1561 // conflicts will happen when tracks are moved from one output to another by audio policy 1562 // manager 1563 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1564 for (size_t i = 0; i < mTracks.size(); ++i) { 1565 sp<Track> t = mTracks[i]; 1566 if (t != 0 && t->isExternalTrack()) { 1567 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1568 if (sessionId == t->sessionId() && strategy != actual) { 1569 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1570 strategy, actual); 1571 lStatus = BAD_VALUE; 1572 goto Exit; 1573 } 1574 } 1575 } 1576 1577 if (!isTimed) { 1578 track = new Track(this, client, streamType, sampleRate, format, 1579 channelMask, frameCount, NULL, sharedBuffer, 1580 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1581 } else { 1582 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1583 channelMask, frameCount, sharedBuffer, sessionId, uid); 1584 } 1585 1586 // new Track always returns non-NULL, 1587 // but TimedTrack::create() is a factory that could fail by returning NULL 1588 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1589 if (lStatus != NO_ERROR) { 1590 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1591 // track must be cleared from the caller as the caller has the AF lock 1592 goto Exit; 1593 } 1594 mTracks.add(track); 1595 1596 sp<EffectChain> chain = getEffectChain_l(sessionId); 1597 if (chain != 0) { 1598 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1599 track->setMainBuffer(chain->inBuffer()); 1600 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1601 chain->incTrackCnt(); 1602 } 1603 1604 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1605 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1606 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1607 // so ask activity manager to do this on our behalf 1608 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1609 } 1610 } 1611 1612 lStatus = NO_ERROR; 1613 1614Exit: 1615 *status = lStatus; 1616 return track; 1617} 1618 1619uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1620{ 1621 return latency; 1622} 1623 1624uint32_t AudioFlinger::PlaybackThread::latency() const 1625{ 1626 Mutex::Autolock _l(mLock); 1627 return latency_l(); 1628} 1629uint32_t AudioFlinger::PlaybackThread::latency_l() const 1630{ 1631 if (initCheck() == NO_ERROR) { 1632 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1633 } else { 1634 return 0; 1635 } 1636} 1637 1638void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1639{ 1640 Mutex::Autolock _l(mLock); 1641 // Don't apply master volume in SW if our HAL can do it for us. 1642 if (mOutput && mOutput->audioHwDev && 1643 mOutput->audioHwDev->canSetMasterVolume()) { 1644 mMasterVolume = 1.0; 1645 } else { 1646 mMasterVolume = value; 1647 } 1648} 1649 1650void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1651{ 1652 Mutex::Autolock _l(mLock); 1653 // Don't apply master mute in SW if our HAL can do it for us. 1654 if (mOutput && mOutput->audioHwDev && 1655 mOutput->audioHwDev->canSetMasterMute()) { 1656 mMasterMute = false; 1657 } else { 1658 mMasterMute = muted; 1659 } 1660} 1661 1662void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1663{ 1664 Mutex::Autolock _l(mLock); 1665 mStreamTypes[stream].volume = value; 1666 broadcast_l(); 1667} 1668 1669void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1670{ 1671 Mutex::Autolock _l(mLock); 1672 mStreamTypes[stream].mute = muted; 1673 broadcast_l(); 1674} 1675 1676float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1677{ 1678 Mutex::Autolock _l(mLock); 1679 return mStreamTypes[stream].volume; 1680} 1681 1682// addTrack_l() must be called with ThreadBase::mLock held 1683status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1684{ 1685 status_t status = ALREADY_EXISTS; 1686 1687 // set retry count for buffer fill 1688 track->mRetryCount = kMaxTrackStartupRetries; 1689 if (mActiveTracks.indexOf(track) < 0) { 1690 // the track is newly added, make sure it fills up all its 1691 // buffers before playing. This is to ensure the client will 1692 // effectively get the latency it requested. 1693 if (track->isExternalTrack()) { 1694 TrackBase::track_state state = track->mState; 1695 mLock.unlock(); 1696 status = AudioSystem::startOutput(mId, track->streamType(), 1697 (audio_session_t)track->sessionId()); 1698 mLock.lock(); 1699 // abort track was stopped/paused while we released the lock 1700 if (state != track->mState) { 1701 if (status == NO_ERROR) { 1702 mLock.unlock(); 1703 AudioSystem::stopOutput(mId, track->streamType(), 1704 (audio_session_t)track->sessionId()); 1705 mLock.lock(); 1706 } 1707 return INVALID_OPERATION; 1708 } 1709 // abort if start is rejected by audio policy manager 1710 if (status != NO_ERROR) { 1711 return PERMISSION_DENIED; 1712 } 1713#ifdef ADD_BATTERY_DATA 1714 // to track the speaker usage 1715 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1716#endif 1717 } 1718 1719 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1720 track->mResetDone = false; 1721 track->mPresentationCompleteFrames = 0; 1722 mActiveTracks.add(track); 1723 mWakeLockUids.add(track->uid()); 1724 mActiveTracksGeneration++; 1725 mLatestActiveTrack = track; 1726 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1727 if (chain != 0) { 1728 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1729 track->sessionId()); 1730 chain->incActiveTrackCnt(); 1731 } 1732 1733 status = NO_ERROR; 1734 } 1735 1736 onAddNewTrack_l(); 1737 return status; 1738} 1739 1740bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1741{ 1742 track->terminate(); 1743 // active tracks are removed by threadLoop() 1744 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1745 track->mState = TrackBase::STOPPED; 1746 if (!trackActive) { 1747 removeTrack_l(track); 1748 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1749 track->mState = TrackBase::STOPPING_1; 1750 } 1751 1752 return trackActive; 1753} 1754 1755void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1756{ 1757 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1758 mTracks.remove(track); 1759 deleteTrackName_l(track->name()); 1760 // redundant as track is about to be destroyed, for dumpsys only 1761 track->mName = -1; 1762 if (track->isFastTrack()) { 1763 int index = track->mFastIndex; 1764 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1765 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1766 mFastTrackAvailMask |= 1 << index; 1767 // redundant as track is about to be destroyed, for dumpsys only 1768 track->mFastIndex = -1; 1769 } 1770 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1771 if (chain != 0) { 1772 chain->decTrackCnt(); 1773 } 1774} 1775 1776void AudioFlinger::PlaybackThread::broadcast_l() 1777{ 1778 // Thread could be blocked waiting for async 1779 // so signal it to handle state changes immediately 1780 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1781 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1782 mSignalPending = true; 1783 mWaitWorkCV.broadcast(); 1784} 1785 1786String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1787{ 1788 Mutex::Autolock _l(mLock); 1789 if (initCheck() != NO_ERROR) { 1790 return String8(); 1791 } 1792 1793 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1794 const String8 out_s8(s); 1795 free(s); 1796 return out_s8; 1797} 1798 1799void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1800 AudioSystem::OutputDescriptor desc; 1801 void *param2 = NULL; 1802 1803 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1804 param); 1805 1806 switch (event) { 1807 case AudioSystem::OUTPUT_OPENED: 1808 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1809 desc.channelMask = mChannelMask; 1810 desc.samplingRate = mSampleRate; 1811 desc.format = mFormat; 1812 desc.frameCount = mNormalFrameCount; // FIXME see 1813 // AudioFlinger::frameCount(audio_io_handle_t) 1814 desc.latency = latency_l(); 1815 param2 = &desc; 1816 break; 1817 1818 case AudioSystem::STREAM_CONFIG_CHANGED: 1819 param2 = ¶m; 1820 case AudioSystem::OUTPUT_CLOSED: 1821 default: 1822 break; 1823 } 1824 mAudioFlinger->audioConfigChanged(event, mId, param2); 1825} 1826 1827void AudioFlinger::PlaybackThread::writeCallback() 1828{ 1829 ALOG_ASSERT(mCallbackThread != 0); 1830 mCallbackThread->resetWriteBlocked(); 1831} 1832 1833void AudioFlinger::PlaybackThread::drainCallback() 1834{ 1835 ALOG_ASSERT(mCallbackThread != 0); 1836 mCallbackThread->resetDraining(); 1837} 1838 1839void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1840{ 1841 Mutex::Autolock _l(mLock); 1842 // reject out of sequence requests 1843 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1844 mWriteAckSequence &= ~1; 1845 mWaitWorkCV.signal(); 1846 } 1847} 1848 1849void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 // reject out of sequence requests 1853 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1854 mDrainSequence &= ~1; 1855 mWaitWorkCV.signal(); 1856 } 1857} 1858 1859// static 1860int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1861 void *param __unused, 1862 void *cookie) 1863{ 1864 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1865 ALOGV("asyncCallback() event %d", event); 1866 switch (event) { 1867 case STREAM_CBK_EVENT_WRITE_READY: 1868 me->writeCallback(); 1869 break; 1870 case STREAM_CBK_EVENT_DRAIN_READY: 1871 me->drainCallback(); 1872 break; 1873 default: 1874 ALOGW("asyncCallback() unknown event %d", event); 1875 break; 1876 } 1877 return 0; 1878} 1879 1880void AudioFlinger::PlaybackThread::readOutputParameters_l() 1881{ 1882 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1883 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1884 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1885 if (!audio_is_output_channel(mChannelMask)) { 1886 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1887 } 1888 if ((mType == MIXER || mType == DUPLICATING) 1889 && !isValidPcmSinkChannelMask(mChannelMask)) { 1890 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1891 mChannelMask); 1892 } 1893 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1894 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1895 mFormat = mHALFormat; 1896 if (!audio_is_valid_format(mFormat)) { 1897 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1898 } 1899 if ((mType == MIXER || mType == DUPLICATING) 1900 && !isValidPcmSinkFormat(mFormat)) { 1901 LOG_FATAL("HAL format %#x not supported for mixed output", 1902 mFormat); 1903 } 1904 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1905 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1906 mFrameCount = mBufferSize / mFrameSize; 1907 if (mFrameCount & 15) { 1908 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1909 mFrameCount); 1910 } 1911 1912 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1913 (mOutput->stream->set_callback != NULL)) { 1914 if (mOutput->stream->set_callback(mOutput->stream, 1915 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1916 mUseAsyncWrite = true; 1917 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1918 } 1919 } 1920 1921 mHwSupportsPause = false; 1922 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 1923 if (mOutput->stream->pause != NULL) { 1924 if (mOutput->stream->resume != NULL) { 1925 mHwSupportsPause = true; 1926 } else { 1927 ALOGW("direct output implements pause but not resume"); 1928 } 1929 } else if (mOutput->stream->resume != NULL) { 1930 ALOGW("direct output implements resume but not pause"); 1931 } 1932 } 1933 1934 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 1935 // For best precision, we use float instead of the associated output 1936 // device format (typically PCM 16 bit). 1937 1938 mFormat = AUDIO_FORMAT_PCM_FLOAT; 1939 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 1940 mBufferSize = mFrameSize * mFrameCount; 1941 1942 // TODO: We currently use the associated output device channel mask and sample rate. 1943 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 1944 // (if a valid mask) to avoid premature downmix. 1945 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 1946 // instead of the output device sample rate to avoid loss of high frequency information. 1947 // This may need to be updated as MixerThread/OutputTracks are added and not here. 1948 } 1949 1950 // Calculate size of normal sink buffer relative to the HAL output buffer size 1951 double multiplier = 1.0; 1952 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1953 kUseFastMixer == FastMixer_Dynamic)) { 1954 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1955 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1956 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1957 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1958 maxNormalFrameCount = maxNormalFrameCount & ~15; 1959 if (maxNormalFrameCount < minNormalFrameCount) { 1960 maxNormalFrameCount = minNormalFrameCount; 1961 } 1962 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1963 if (multiplier <= 1.0) { 1964 multiplier = 1.0; 1965 } else if (multiplier <= 2.0) { 1966 if (2 * mFrameCount <= maxNormalFrameCount) { 1967 multiplier = 2.0; 1968 } else { 1969 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1970 } 1971 } else { 1972 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1973 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1974 // track, but we sometimes have to do this to satisfy the maximum frame count 1975 // constraint) 1976 // FIXME this rounding up should not be done if no HAL SRC 1977 uint32_t truncMult = (uint32_t) multiplier; 1978 if ((truncMult & 1)) { 1979 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1980 ++truncMult; 1981 } 1982 } 1983 multiplier = (double) truncMult; 1984 } 1985 } 1986 mNormalFrameCount = multiplier * mFrameCount; 1987 // round up to nearest 16 frames to satisfy AudioMixer 1988 if (mType == MIXER || mType == DUPLICATING) { 1989 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1990 } 1991 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1992 mNormalFrameCount); 1993 1994 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1995 // Originally this was int16_t[] array, need to remove legacy implications. 1996 free(mSinkBuffer); 1997 mSinkBuffer = NULL; 1998 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1999 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2000 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2001 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2002 2003 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2004 // drives the output. 2005 free(mMixerBuffer); 2006 mMixerBuffer = NULL; 2007 if (mMixerBufferEnabled) { 2008 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2009 mMixerBufferSize = mNormalFrameCount * mChannelCount 2010 * audio_bytes_per_sample(mMixerBufferFormat); 2011 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2012 } 2013 free(mEffectBuffer); 2014 mEffectBuffer = NULL; 2015 if (mEffectBufferEnabled) { 2016 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2017 mEffectBufferSize = mNormalFrameCount * mChannelCount 2018 * audio_bytes_per_sample(mEffectBufferFormat); 2019 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2020 } 2021 2022 // force reconfiguration of effect chains and engines to take new buffer size and audio 2023 // parameters into account 2024 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2025 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2026 // matter. 2027 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2028 Vector< sp<EffectChain> > effectChains = mEffectChains; 2029 for (size_t i = 0; i < effectChains.size(); i ++) { 2030 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2031 } 2032} 2033 2034 2035status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2036{ 2037 if (halFrames == NULL || dspFrames == NULL) { 2038 return BAD_VALUE; 2039 } 2040 Mutex::Autolock _l(mLock); 2041 if (initCheck() != NO_ERROR) { 2042 return INVALID_OPERATION; 2043 } 2044 size_t framesWritten = mBytesWritten / mFrameSize; 2045 *halFrames = framesWritten; 2046 2047 if (isSuspended()) { 2048 // return an estimation of rendered frames when the output is suspended 2049 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2050 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2051 return NO_ERROR; 2052 } else { 2053 status_t status; 2054 uint32_t frames; 2055 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 2056 *dspFrames = (size_t)frames; 2057 return status; 2058 } 2059} 2060 2061uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2062{ 2063 Mutex::Autolock _l(mLock); 2064 uint32_t result = 0; 2065 if (getEffectChain_l(sessionId) != 0) { 2066 result = EFFECT_SESSION; 2067 } 2068 2069 for (size_t i = 0; i < mTracks.size(); ++i) { 2070 sp<Track> track = mTracks[i]; 2071 if (sessionId == track->sessionId() && !track->isInvalid()) { 2072 result |= TRACK_SESSION; 2073 break; 2074 } 2075 } 2076 2077 return result; 2078} 2079 2080uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2081{ 2082 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2083 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2084 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2085 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2086 } 2087 for (size_t i = 0; i < mTracks.size(); i++) { 2088 sp<Track> track = mTracks[i]; 2089 if (sessionId == track->sessionId() && !track->isInvalid()) { 2090 return AudioSystem::getStrategyForStream(track->streamType()); 2091 } 2092 } 2093 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2094} 2095 2096 2097AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2098{ 2099 Mutex::Autolock _l(mLock); 2100 return mOutput; 2101} 2102 2103AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2104{ 2105 Mutex::Autolock _l(mLock); 2106 AudioStreamOut *output = mOutput; 2107 mOutput = NULL; 2108 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2109 // must push a NULL and wait for ack 2110 mOutputSink.clear(); 2111 mPipeSink.clear(); 2112 mNormalSink.clear(); 2113 return output; 2114} 2115 2116// this method must always be called either with ThreadBase mLock held or inside the thread loop 2117audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2118{ 2119 if (mOutput == NULL) { 2120 return NULL; 2121 } 2122 return &mOutput->stream->common; 2123} 2124 2125uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2126{ 2127 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2128} 2129 2130status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2131{ 2132 if (!isValidSyncEvent(event)) { 2133 return BAD_VALUE; 2134 } 2135 2136 Mutex::Autolock _l(mLock); 2137 2138 for (size_t i = 0; i < mTracks.size(); ++i) { 2139 sp<Track> track = mTracks[i]; 2140 if (event->triggerSession() == track->sessionId()) { 2141 (void) track->setSyncEvent(event); 2142 return NO_ERROR; 2143 } 2144 } 2145 2146 return NAME_NOT_FOUND; 2147} 2148 2149bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2150{ 2151 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2152} 2153 2154void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2155 const Vector< sp<Track> >& tracksToRemove) 2156{ 2157 size_t count = tracksToRemove.size(); 2158 if (count > 0) { 2159 for (size_t i = 0 ; i < count ; i++) { 2160 const sp<Track>& track = tracksToRemove.itemAt(i); 2161 if (track->isExternalTrack()) { 2162 AudioSystem::stopOutput(mId, track->streamType(), 2163 (audio_session_t)track->sessionId()); 2164#ifdef ADD_BATTERY_DATA 2165 // to track the speaker usage 2166 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2167#endif 2168 if (track->isTerminated()) { 2169 AudioSystem::releaseOutput(mId, track->streamType(), 2170 (audio_session_t)track->sessionId()); 2171 } 2172 } 2173 } 2174 } 2175} 2176 2177void AudioFlinger::PlaybackThread::checkSilentMode_l() 2178{ 2179 if (!mMasterMute) { 2180 char value[PROPERTY_VALUE_MAX]; 2181 if (property_get("ro.audio.silent", value, "0") > 0) { 2182 char *endptr; 2183 unsigned long ul = strtoul(value, &endptr, 0); 2184 if (*endptr == '\0' && ul != 0) { 2185 ALOGD("Silence is golden"); 2186 // The setprop command will not allow a property to be changed after 2187 // the first time it is set, so we don't have to worry about un-muting. 2188 setMasterMute_l(true); 2189 } 2190 } 2191 } 2192} 2193 2194// shared by MIXER and DIRECT, overridden by DUPLICATING 2195ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2196{ 2197 // FIXME rewrite to reduce number of system calls 2198 mLastWriteTime = systemTime(); 2199 mInWrite = true; 2200 ssize_t bytesWritten; 2201 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2202 2203 // If an NBAIO sink is present, use it to write the normal mixer's submix 2204 if (mNormalSink != 0) { 2205 2206 const size_t count = mBytesRemaining / mFrameSize; 2207 2208 ATRACE_BEGIN("write"); 2209 // update the setpoint when AudioFlinger::mScreenState changes 2210 uint32_t screenState = AudioFlinger::mScreenState; 2211 if (screenState != mScreenState) { 2212 mScreenState = screenState; 2213 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2214 if (pipe != NULL) { 2215 pipe->setAvgFrames((mScreenState & 1) ? 2216 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2217 } 2218 } 2219 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2220 ATRACE_END(); 2221 if (framesWritten > 0) { 2222 bytesWritten = framesWritten * mFrameSize; 2223 } else { 2224 bytesWritten = framesWritten; 2225 } 2226 mLatchDValid = false; 2227 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2228 if (status == NO_ERROR) { 2229 size_t totalFramesWritten = mNormalSink->framesWritten(); 2230 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2231 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2232 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2233 mLatchDValid = true; 2234 } 2235 } 2236 // otherwise use the HAL / AudioStreamOut directly 2237 } else { 2238 // Direct output and offload threads 2239 2240 if (mUseAsyncWrite) { 2241 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2242 mWriteAckSequence += 2; 2243 mWriteAckSequence |= 1; 2244 ALOG_ASSERT(mCallbackThread != 0); 2245 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2246 } 2247 // FIXME We should have an implementation of timestamps for direct output threads. 2248 // They are used e.g for multichannel PCM playback over HDMI. 2249 bytesWritten = mOutput->stream->write(mOutput->stream, 2250 (char *)mSinkBuffer + offset, mBytesRemaining); 2251 if (mUseAsyncWrite && 2252 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2253 // do not wait for async callback in case of error of full write 2254 mWriteAckSequence &= ~1; 2255 ALOG_ASSERT(mCallbackThread != 0); 2256 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2257 } 2258 } 2259 2260 mNumWrites++; 2261 mInWrite = false; 2262 mStandby = false; 2263 return bytesWritten; 2264} 2265 2266void AudioFlinger::PlaybackThread::threadLoop_drain() 2267{ 2268 if (mOutput->stream->drain) { 2269 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2270 if (mUseAsyncWrite) { 2271 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2272 mDrainSequence |= 1; 2273 ALOG_ASSERT(mCallbackThread != 0); 2274 mCallbackThread->setDraining(mDrainSequence); 2275 } 2276 mOutput->stream->drain(mOutput->stream, 2277 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2278 : AUDIO_DRAIN_ALL); 2279 } 2280} 2281 2282void AudioFlinger::PlaybackThread::threadLoop_exit() 2283{ 2284 { 2285 Mutex::Autolock _l(mLock); 2286 for (size_t i = 0; i < mTracks.size(); i++) { 2287 sp<Track> track = mTracks[i]; 2288 track->invalidate(); 2289 } 2290 } 2291} 2292 2293/* 2294The derived values that are cached: 2295 - mSinkBufferSize from frame count * frame size 2296 - activeSleepTime from activeSleepTimeUs() 2297 - idleSleepTime from idleSleepTimeUs() 2298 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2299 - maxPeriod from frame count and sample rate (MIXER only) 2300 2301The parameters that affect these derived values are: 2302 - frame count 2303 - frame size 2304 - sample rate 2305 - device type: A2DP or not 2306 - device latency 2307 - format: PCM or not 2308 - active sleep time 2309 - idle sleep time 2310*/ 2311 2312void AudioFlinger::PlaybackThread::cacheParameters_l() 2313{ 2314 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2315 activeSleepTime = activeSleepTimeUs(); 2316 idleSleepTime = idleSleepTimeUs(); 2317} 2318 2319void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2320{ 2321 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2322 this, streamType, mTracks.size()); 2323 Mutex::Autolock _l(mLock); 2324 2325 size_t size = mTracks.size(); 2326 for (size_t i = 0; i < size; i++) { 2327 sp<Track> t = mTracks[i]; 2328 if (t->streamType() == streamType) { 2329 t->invalidate(); 2330 } 2331 } 2332} 2333 2334status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2335{ 2336 int session = chain->sessionId(); 2337 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2338 ? mEffectBuffer : mSinkBuffer); 2339 bool ownsBuffer = false; 2340 2341 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2342 if (session > 0) { 2343 // Only one effect chain can be present in direct output thread and it uses 2344 // the sink buffer as input 2345 if (mType != DIRECT) { 2346 size_t numSamples = mNormalFrameCount * mChannelCount; 2347 buffer = new int16_t[numSamples]; 2348 memset(buffer, 0, numSamples * sizeof(int16_t)); 2349 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2350 ownsBuffer = true; 2351 } 2352 2353 // Attach all tracks with same session ID to this chain. 2354 for (size_t i = 0; i < mTracks.size(); ++i) { 2355 sp<Track> track = mTracks[i]; 2356 if (session == track->sessionId()) { 2357 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2358 buffer); 2359 track->setMainBuffer(buffer); 2360 chain->incTrackCnt(); 2361 } 2362 } 2363 2364 // indicate all active tracks in the chain 2365 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2366 sp<Track> track = mActiveTracks[i].promote(); 2367 if (track == 0) { 2368 continue; 2369 } 2370 if (session == track->sessionId()) { 2371 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2372 chain->incActiveTrackCnt(); 2373 } 2374 } 2375 } 2376 chain->setThread(this); 2377 chain->setInBuffer(buffer, ownsBuffer); 2378 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2379 ? mEffectBuffer : mSinkBuffer)); 2380 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2381 // chains list in order to be processed last as it contains output stage effects 2382 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2383 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2384 // after track specific effects and before output stage 2385 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2386 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2387 // Effect chain for other sessions are inserted at beginning of effect 2388 // chains list to be processed before output mix effects. Relative order between other 2389 // sessions is not important 2390 size_t size = mEffectChains.size(); 2391 size_t i = 0; 2392 for (i = 0; i < size; i++) { 2393 if (mEffectChains[i]->sessionId() < session) { 2394 break; 2395 } 2396 } 2397 mEffectChains.insertAt(chain, i); 2398 checkSuspendOnAddEffectChain_l(chain); 2399 2400 return NO_ERROR; 2401} 2402 2403size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2404{ 2405 int session = chain->sessionId(); 2406 2407 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2408 2409 for (size_t i = 0; i < mEffectChains.size(); i++) { 2410 if (chain == mEffectChains[i]) { 2411 mEffectChains.removeAt(i); 2412 // detach all active tracks from the chain 2413 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2414 sp<Track> track = mActiveTracks[i].promote(); 2415 if (track == 0) { 2416 continue; 2417 } 2418 if (session == track->sessionId()) { 2419 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2420 chain.get(), session); 2421 chain->decActiveTrackCnt(); 2422 } 2423 } 2424 2425 // detach all tracks with same session ID from this chain 2426 for (size_t i = 0; i < mTracks.size(); ++i) { 2427 sp<Track> track = mTracks[i]; 2428 if (session == track->sessionId()) { 2429 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2430 chain->decTrackCnt(); 2431 } 2432 } 2433 break; 2434 } 2435 } 2436 return mEffectChains.size(); 2437} 2438 2439status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2440 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2441{ 2442 Mutex::Autolock _l(mLock); 2443 return attachAuxEffect_l(track, EffectId); 2444} 2445 2446status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2447 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2448{ 2449 status_t status = NO_ERROR; 2450 2451 if (EffectId == 0) { 2452 track->setAuxBuffer(0, NULL); 2453 } else { 2454 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2455 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2456 if (effect != 0) { 2457 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2458 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2459 } else { 2460 status = INVALID_OPERATION; 2461 } 2462 } else { 2463 status = BAD_VALUE; 2464 } 2465 } 2466 return status; 2467} 2468 2469void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2470{ 2471 for (size_t i = 0; i < mTracks.size(); ++i) { 2472 sp<Track> track = mTracks[i]; 2473 if (track->auxEffectId() == effectId) { 2474 attachAuxEffect_l(track, 0); 2475 } 2476 } 2477} 2478 2479bool AudioFlinger::PlaybackThread::threadLoop() 2480{ 2481 Vector< sp<Track> > tracksToRemove; 2482 2483 standbyTime = systemTime(); 2484 2485 // MIXER 2486 nsecs_t lastWarning = 0; 2487 2488 // DUPLICATING 2489 // FIXME could this be made local to while loop? 2490 writeFrames = 0; 2491 2492 int lastGeneration = 0; 2493 2494 cacheParameters_l(); 2495 sleepTime = idleSleepTime; 2496 2497 if (mType == MIXER) { 2498 sleepTimeShift = 0; 2499 } 2500 2501 CpuStats cpuStats; 2502 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2503 2504 acquireWakeLock(); 2505 2506 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2507 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2508 // and then that string will be logged at the next convenient opportunity. 2509 const char *logString = NULL; 2510 2511 checkSilentMode_l(); 2512 2513 while (!exitPending()) 2514 { 2515 cpuStats.sample(myName); 2516 2517 Vector< sp<EffectChain> > effectChains; 2518 2519 { // scope for mLock 2520 2521 Mutex::Autolock _l(mLock); 2522 2523 processConfigEvents_l(); 2524 2525 if (logString != NULL) { 2526 mNBLogWriter->logTimestamp(); 2527 mNBLogWriter->log(logString); 2528 logString = NULL; 2529 } 2530 2531 // Gather the framesReleased counters for all active tracks, 2532 // and latch them atomically with the timestamp. 2533 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2534 mLatchD.mFramesReleased.clear(); 2535 size_t size = mActiveTracks.size(); 2536 for (size_t i = 0; i < size; i++) { 2537 sp<Track> t = mActiveTracks[i].promote(); 2538 if (t != 0) { 2539 mLatchD.mFramesReleased.add(t.get(), 2540 t->mAudioTrackServerProxy->framesReleased()); 2541 } 2542 } 2543 if (mLatchDValid) { 2544 mLatchQ = mLatchD; 2545 mLatchDValid = false; 2546 mLatchQValid = true; 2547 } 2548 2549 saveOutputTracks(); 2550 if (mSignalPending) { 2551 // A signal was raised while we were unlocked 2552 mSignalPending = false; 2553 } else if (waitingAsyncCallback_l()) { 2554 if (exitPending()) { 2555 break; 2556 } 2557 releaseWakeLock_l(); 2558 mWakeLockUids.clear(); 2559 mActiveTracksGeneration++; 2560 ALOGV("wait async completion"); 2561 mWaitWorkCV.wait(mLock); 2562 ALOGV("async completion/wake"); 2563 acquireWakeLock_l(); 2564 standbyTime = systemTime() + standbyDelay; 2565 sleepTime = 0; 2566 2567 continue; 2568 } 2569 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2570 isSuspended()) { 2571 // put audio hardware into standby after short delay 2572 if (shouldStandby_l()) { 2573 2574 threadLoop_standby(); 2575 2576 mStandby = true; 2577 } 2578 2579 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2580 // we're about to wait, flush the binder command buffer 2581 IPCThreadState::self()->flushCommands(); 2582 2583 clearOutputTracks(); 2584 2585 if (exitPending()) { 2586 break; 2587 } 2588 2589 releaseWakeLock_l(); 2590 mWakeLockUids.clear(); 2591 mActiveTracksGeneration++; 2592 // wait until we have something to do... 2593 ALOGV("%s going to sleep", myName.string()); 2594 mWaitWorkCV.wait(mLock); 2595 ALOGV("%s waking up", myName.string()); 2596 acquireWakeLock_l(); 2597 2598 mMixerStatus = MIXER_IDLE; 2599 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2600 mBytesWritten = 0; 2601 mBytesRemaining = 0; 2602 checkSilentMode_l(); 2603 2604 standbyTime = systemTime() + standbyDelay; 2605 sleepTime = idleSleepTime; 2606 if (mType == MIXER) { 2607 sleepTimeShift = 0; 2608 } 2609 2610 continue; 2611 } 2612 } 2613 // mMixerStatusIgnoringFastTracks is also updated internally 2614 mMixerStatus = prepareTracks_l(&tracksToRemove); 2615 2616 // compare with previously applied list 2617 if (lastGeneration != mActiveTracksGeneration) { 2618 // update wakelock 2619 updateWakeLockUids_l(mWakeLockUids); 2620 lastGeneration = mActiveTracksGeneration; 2621 } 2622 2623 // prevent any changes in effect chain list and in each effect chain 2624 // during mixing and effect process as the audio buffers could be deleted 2625 // or modified if an effect is created or deleted 2626 lockEffectChains_l(effectChains); 2627 } // mLock scope ends 2628 2629 if (mBytesRemaining == 0) { 2630 mCurrentWriteLength = 0; 2631 if (mMixerStatus == MIXER_TRACKS_READY) { 2632 // threadLoop_mix() sets mCurrentWriteLength 2633 threadLoop_mix(); 2634 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2635 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2636 // threadLoop_sleepTime sets sleepTime to 0 if data 2637 // must be written to HAL 2638 threadLoop_sleepTime(); 2639 if (sleepTime == 0) { 2640 mCurrentWriteLength = mSinkBufferSize; 2641 } 2642 } 2643 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2644 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2645 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2646 // or mSinkBuffer (if there are no effects). 2647 // 2648 // This is done pre-effects computation; if effects change to 2649 // support higher precision, this needs to move. 2650 // 2651 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2652 // TODO use sleepTime == 0 as an additional condition. 2653 if (mMixerBufferValid) { 2654 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2655 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2656 2657 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2658 mNormalFrameCount * mChannelCount); 2659 } 2660 2661 mBytesRemaining = mCurrentWriteLength; 2662 if (isSuspended()) { 2663 sleepTime = suspendSleepTimeUs(); 2664 // simulate write to HAL when suspended 2665 mBytesWritten += mSinkBufferSize; 2666 mBytesRemaining = 0; 2667 } 2668 2669 // only process effects if we're going to write 2670 if (sleepTime == 0 && mType != OFFLOAD) { 2671 for (size_t i = 0; i < effectChains.size(); i ++) { 2672 effectChains[i]->process_l(); 2673 } 2674 } 2675 } 2676 // Process effect chains for offloaded thread even if no audio 2677 // was read from audio track: process only updates effect state 2678 // and thus does have to be synchronized with audio writes but may have 2679 // to be called while waiting for async write callback 2680 if (mType == OFFLOAD) { 2681 for (size_t i = 0; i < effectChains.size(); i ++) { 2682 effectChains[i]->process_l(); 2683 } 2684 } 2685 2686 // Only if the Effects buffer is enabled and there is data in the 2687 // Effects buffer (buffer valid), we need to 2688 // copy into the sink buffer. 2689 // TODO use sleepTime == 0 as an additional condition. 2690 if (mEffectBufferValid) { 2691 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2692 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2693 mNormalFrameCount * mChannelCount); 2694 } 2695 2696 // enable changes in effect chain 2697 unlockEffectChains(effectChains); 2698 2699 if (!waitingAsyncCallback()) { 2700 // sleepTime == 0 means we must write to audio hardware 2701 if (sleepTime == 0) { 2702 if (mBytesRemaining) { 2703 ssize_t ret = threadLoop_write(); 2704 if (ret < 0) { 2705 mBytesRemaining = 0; 2706 } else { 2707 mBytesWritten += ret; 2708 mBytesRemaining -= ret; 2709 } 2710 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2711 (mMixerStatus == MIXER_DRAIN_ALL)) { 2712 threadLoop_drain(); 2713 } 2714 if (mType == MIXER) { 2715 // write blocked detection 2716 nsecs_t now = systemTime(); 2717 nsecs_t delta = now - mLastWriteTime; 2718 if (!mStandby && delta > maxPeriod) { 2719 mNumDelayedWrites++; 2720 if ((now - lastWarning) > kWarningThrottleNs) { 2721 ATRACE_NAME("underrun"); 2722 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2723 ns2ms(delta), mNumDelayedWrites, this); 2724 lastWarning = now; 2725 } 2726 } 2727 } 2728 2729 } else { 2730 ATRACE_BEGIN("sleep"); 2731 usleep(sleepTime); 2732 ATRACE_END(); 2733 } 2734 } 2735 2736 // Finally let go of removed track(s), without the lock held 2737 // since we can't guarantee the destructors won't acquire that 2738 // same lock. This will also mutate and push a new fast mixer state. 2739 threadLoop_removeTracks(tracksToRemove); 2740 tracksToRemove.clear(); 2741 2742 // FIXME I don't understand the need for this here; 2743 // it was in the original code but maybe the 2744 // assignment in saveOutputTracks() makes this unnecessary? 2745 clearOutputTracks(); 2746 2747 // Effect chains will be actually deleted here if they were removed from 2748 // mEffectChains list during mixing or effects processing 2749 effectChains.clear(); 2750 2751 // FIXME Note that the above .clear() is no longer necessary since effectChains 2752 // is now local to this block, but will keep it for now (at least until merge done). 2753 } 2754 2755 threadLoop_exit(); 2756 2757 if (!mStandby) { 2758 threadLoop_standby(); 2759 mStandby = true; 2760 } 2761 2762 releaseWakeLock(); 2763 mWakeLockUids.clear(); 2764 mActiveTracksGeneration++; 2765 2766 ALOGV("Thread %p type %d exiting", this, mType); 2767 return false; 2768} 2769 2770// removeTracks_l() must be called with ThreadBase::mLock held 2771void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2772{ 2773 size_t count = tracksToRemove.size(); 2774 if (count > 0) { 2775 for (size_t i=0 ; i<count ; i++) { 2776 const sp<Track>& track = tracksToRemove.itemAt(i); 2777 mActiveTracks.remove(track); 2778 mWakeLockUids.remove(track->uid()); 2779 mActiveTracksGeneration++; 2780 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2781 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2782 if (chain != 0) { 2783 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2784 track->sessionId()); 2785 chain->decActiveTrackCnt(); 2786 } 2787 if (track->isTerminated()) { 2788 removeTrack_l(track); 2789 } 2790 } 2791 } 2792 2793} 2794 2795status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2796{ 2797 if (mNormalSink != 0) { 2798 return mNormalSink->getTimestamp(timestamp); 2799 } 2800 if ((mType == OFFLOAD || mType == DIRECT) 2801 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2802 uint64_t position64; 2803 int ret = mOutput->stream->get_presentation_position( 2804 mOutput->stream, &position64, ×tamp.mTime); 2805 if (ret == 0) { 2806 timestamp.mPosition = (uint32_t)position64; 2807 return NO_ERROR; 2808 } 2809 } 2810 return INVALID_OPERATION; 2811} 2812 2813status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2814 audio_patch_handle_t *handle) 2815{ 2816 status_t status = NO_ERROR; 2817 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2818 // store new device and send to effects 2819 audio_devices_t type = AUDIO_DEVICE_NONE; 2820 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2821 type |= patch->sinks[i].ext.device.type; 2822 } 2823 mOutDevice = type; 2824 for (size_t i = 0; i < mEffectChains.size(); i++) { 2825 mEffectChains[i]->setDevice_l(mOutDevice); 2826 } 2827 2828 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2829 status = hwDevice->create_audio_patch(hwDevice, 2830 patch->num_sources, 2831 patch->sources, 2832 patch->num_sinks, 2833 patch->sinks, 2834 handle); 2835 } else { 2836 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2837 } 2838 return status; 2839} 2840 2841status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2842{ 2843 status_t status = NO_ERROR; 2844 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2845 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2846 status = hwDevice->release_audio_patch(hwDevice, handle); 2847 } else { 2848 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2849 } 2850 return status; 2851} 2852 2853void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2854{ 2855 Mutex::Autolock _l(mLock); 2856 mTracks.add(track); 2857} 2858 2859void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2860{ 2861 Mutex::Autolock _l(mLock); 2862 destroyTrack_l(track); 2863} 2864 2865void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2866{ 2867 ThreadBase::getAudioPortConfig(config); 2868 config->role = AUDIO_PORT_ROLE_SOURCE; 2869 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2870 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2871} 2872 2873// ---------------------------------------------------------------------------- 2874 2875AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2876 audio_io_handle_t id, audio_devices_t device, type_t type) 2877 : PlaybackThread(audioFlinger, output, id, device, type), 2878 // mAudioMixer below 2879 // mFastMixer below 2880 mFastMixerFutex(0) 2881 // mOutputSink below 2882 // mPipeSink below 2883 // mNormalSink below 2884{ 2885 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2886 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2887 "mFrameCount=%d, mNormalFrameCount=%d", 2888 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2889 mNormalFrameCount); 2890 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2891 2892 if (type == DUPLICATING) { 2893 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 2894 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 2895 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 2896 return; 2897 } 2898 // create an NBAIO sink for the HAL output stream, and negotiate 2899 mOutputSink = new AudioStreamOutSink(output->stream); 2900 size_t numCounterOffers = 0; 2901 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2902 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2903 ALOG_ASSERT(index == 0); 2904 2905 // initialize fast mixer depending on configuration 2906 bool initFastMixer; 2907 switch (kUseFastMixer) { 2908 case FastMixer_Never: 2909 initFastMixer = false; 2910 break; 2911 case FastMixer_Always: 2912 initFastMixer = true; 2913 break; 2914 case FastMixer_Static: 2915 case FastMixer_Dynamic: 2916 initFastMixer = mFrameCount < mNormalFrameCount; 2917 break; 2918 } 2919 if (initFastMixer) { 2920 audio_format_t fastMixerFormat; 2921 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2922 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2923 } else { 2924 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2925 } 2926 if (mFormat != fastMixerFormat) { 2927 // change our Sink format to accept our intermediate precision 2928 mFormat = fastMixerFormat; 2929 free(mSinkBuffer); 2930 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2931 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2932 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2933 } 2934 2935 // create a MonoPipe to connect our submix to FastMixer 2936 NBAIO_Format format = mOutputSink->format(); 2937 NBAIO_Format origformat = format; 2938 // adjust format to match that of the Fast Mixer 2939 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 2940 format.mFormat = fastMixerFormat; 2941 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2942 2943 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2944 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2945 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2946 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2947 const NBAIO_Format offers[1] = {format}; 2948 size_t numCounterOffers = 0; 2949 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2950 ALOG_ASSERT(index == 0); 2951 monoPipe->setAvgFrames((mScreenState & 1) ? 2952 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2953 mPipeSink = monoPipe; 2954 2955#ifdef TEE_SINK 2956 if (mTeeSinkOutputEnabled) { 2957 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2958 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 2959 const NBAIO_Format offers2[1] = {origformat}; 2960 numCounterOffers = 0; 2961 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 2962 ALOG_ASSERT(index == 0); 2963 mTeeSink = teeSink; 2964 PipeReader *teeSource = new PipeReader(*teeSink); 2965 numCounterOffers = 0; 2966 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 2967 ALOG_ASSERT(index == 0); 2968 mTeeSource = teeSource; 2969 } 2970#endif 2971 2972 // create fast mixer and configure it initially with just one fast track for our submix 2973 mFastMixer = new FastMixer(); 2974 FastMixerStateQueue *sq = mFastMixer->sq(); 2975#ifdef STATE_QUEUE_DUMP 2976 sq->setObserverDump(&mStateQueueObserverDump); 2977 sq->setMutatorDump(&mStateQueueMutatorDump); 2978#endif 2979 FastMixerState *state = sq->begin(); 2980 FastTrack *fastTrack = &state->mFastTracks[0]; 2981 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2982 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2983 fastTrack->mVolumeProvider = NULL; 2984 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2985 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2986 fastTrack->mGeneration++; 2987 state->mFastTracksGen++; 2988 state->mTrackMask = 1; 2989 // fast mixer will use the HAL output sink 2990 state->mOutputSink = mOutputSink.get(); 2991 state->mOutputSinkGen++; 2992 state->mFrameCount = mFrameCount; 2993 state->mCommand = FastMixerState::COLD_IDLE; 2994 // already done in constructor initialization list 2995 //mFastMixerFutex = 0; 2996 state->mColdFutexAddr = &mFastMixerFutex; 2997 state->mColdGen++; 2998 state->mDumpState = &mFastMixerDumpState; 2999#ifdef TEE_SINK 3000 state->mTeeSink = mTeeSink.get(); 3001#endif 3002 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3003 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3004 sq->end(); 3005 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3006 3007 // start the fast mixer 3008 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3009 pid_t tid = mFastMixer->getTid(); 3010 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3011 if (err != 0) { 3012 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3013 kPriorityFastMixer, getpid_cached, tid, err); 3014 } 3015 3016#ifdef AUDIO_WATCHDOG 3017 // create and start the watchdog 3018 mAudioWatchdog = new AudioWatchdog(); 3019 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3020 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3021 tid = mAudioWatchdog->getTid(); 3022 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3023 if (err != 0) { 3024 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3025 kPriorityFastMixer, getpid_cached, tid, err); 3026 } 3027#endif 3028 3029 } 3030 3031 switch (kUseFastMixer) { 3032 case FastMixer_Never: 3033 case FastMixer_Dynamic: 3034 mNormalSink = mOutputSink; 3035 break; 3036 case FastMixer_Always: 3037 mNormalSink = mPipeSink; 3038 break; 3039 case FastMixer_Static: 3040 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3041 break; 3042 } 3043} 3044 3045AudioFlinger::MixerThread::~MixerThread() 3046{ 3047 if (mFastMixer != 0) { 3048 FastMixerStateQueue *sq = mFastMixer->sq(); 3049 FastMixerState *state = sq->begin(); 3050 if (state->mCommand == FastMixerState::COLD_IDLE) { 3051 int32_t old = android_atomic_inc(&mFastMixerFutex); 3052 if (old == -1) { 3053 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3054 } 3055 } 3056 state->mCommand = FastMixerState::EXIT; 3057 sq->end(); 3058 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3059 mFastMixer->join(); 3060 // Though the fast mixer thread has exited, it's state queue is still valid. 3061 // We'll use that extract the final state which contains one remaining fast track 3062 // corresponding to our sub-mix. 3063 state = sq->begin(); 3064 ALOG_ASSERT(state->mTrackMask == 1); 3065 FastTrack *fastTrack = &state->mFastTracks[0]; 3066 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3067 delete fastTrack->mBufferProvider; 3068 sq->end(false /*didModify*/); 3069 mFastMixer.clear(); 3070#ifdef AUDIO_WATCHDOG 3071 if (mAudioWatchdog != 0) { 3072 mAudioWatchdog->requestExit(); 3073 mAudioWatchdog->requestExitAndWait(); 3074 mAudioWatchdog.clear(); 3075 } 3076#endif 3077 } 3078 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3079 delete mAudioMixer; 3080} 3081 3082 3083uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3084{ 3085 if (mFastMixer != 0) { 3086 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3087 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3088 } 3089 return latency; 3090} 3091 3092 3093void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3094{ 3095 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3096} 3097 3098ssize_t AudioFlinger::MixerThread::threadLoop_write() 3099{ 3100 // FIXME we should only do one push per cycle; confirm this is true 3101 // Start the fast mixer if it's not already running 3102 if (mFastMixer != 0) { 3103 FastMixerStateQueue *sq = mFastMixer->sq(); 3104 FastMixerState *state = sq->begin(); 3105 if (state->mCommand != FastMixerState::MIX_WRITE && 3106 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3107 if (state->mCommand == FastMixerState::COLD_IDLE) { 3108 int32_t old = android_atomic_inc(&mFastMixerFutex); 3109 if (old == -1) { 3110 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3111 } 3112#ifdef AUDIO_WATCHDOG 3113 if (mAudioWatchdog != 0) { 3114 mAudioWatchdog->resume(); 3115 } 3116#endif 3117 } 3118 state->mCommand = FastMixerState::MIX_WRITE; 3119#ifdef FAST_THREAD_STATISTICS 3120 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3121 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3122#endif 3123 sq->end(); 3124 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3125 if (kUseFastMixer == FastMixer_Dynamic) { 3126 mNormalSink = mPipeSink; 3127 } 3128 } else { 3129 sq->end(false /*didModify*/); 3130 } 3131 } 3132 return PlaybackThread::threadLoop_write(); 3133} 3134 3135void AudioFlinger::MixerThread::threadLoop_standby() 3136{ 3137 // Idle the fast mixer if it's currently running 3138 if (mFastMixer != 0) { 3139 FastMixerStateQueue *sq = mFastMixer->sq(); 3140 FastMixerState *state = sq->begin(); 3141 if (!(state->mCommand & FastMixerState::IDLE)) { 3142 state->mCommand = FastMixerState::COLD_IDLE; 3143 state->mColdFutexAddr = &mFastMixerFutex; 3144 state->mColdGen++; 3145 mFastMixerFutex = 0; 3146 sq->end(); 3147 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3148 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3149 if (kUseFastMixer == FastMixer_Dynamic) { 3150 mNormalSink = mOutputSink; 3151 } 3152#ifdef AUDIO_WATCHDOG 3153 if (mAudioWatchdog != 0) { 3154 mAudioWatchdog->pause(); 3155 } 3156#endif 3157 } else { 3158 sq->end(false /*didModify*/); 3159 } 3160 } 3161 PlaybackThread::threadLoop_standby(); 3162} 3163 3164bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3165{ 3166 return false; 3167} 3168 3169bool AudioFlinger::PlaybackThread::shouldStandby_l() 3170{ 3171 return !mStandby; 3172} 3173 3174bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3175{ 3176 Mutex::Autolock _l(mLock); 3177 return waitingAsyncCallback_l(); 3178} 3179 3180// shared by MIXER and DIRECT, overridden by DUPLICATING 3181void AudioFlinger::PlaybackThread::threadLoop_standby() 3182{ 3183 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3184 mOutput->stream->common.standby(&mOutput->stream->common); 3185 if (mUseAsyncWrite != 0) { 3186 // discard any pending drain or write ack by incrementing sequence 3187 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3188 mDrainSequence = (mDrainSequence + 2) & ~1; 3189 ALOG_ASSERT(mCallbackThread != 0); 3190 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3191 mCallbackThread->setDraining(mDrainSequence); 3192 } 3193 mHwPaused = false; 3194} 3195 3196void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3197{ 3198 ALOGV("signal playback thread"); 3199 broadcast_l(); 3200} 3201 3202void AudioFlinger::MixerThread::threadLoop_mix() 3203{ 3204 // obtain the presentation timestamp of the next output buffer 3205 int64_t pts; 3206 status_t status = INVALID_OPERATION; 3207 3208 if (mNormalSink != 0) { 3209 status = mNormalSink->getNextWriteTimestamp(&pts); 3210 } else { 3211 status = mOutputSink->getNextWriteTimestamp(&pts); 3212 } 3213 3214 if (status != NO_ERROR) { 3215 pts = AudioBufferProvider::kInvalidPTS; 3216 } 3217 3218 // mix buffers... 3219 mAudioMixer->process(pts); 3220 mCurrentWriteLength = mSinkBufferSize; 3221 // increase sleep time progressively when application underrun condition clears. 3222 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3223 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3224 // such that we would underrun the audio HAL. 3225 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3226 sleepTimeShift--; 3227 } 3228 sleepTime = 0; 3229 standbyTime = systemTime() + standbyDelay; 3230 //TODO: delay standby when effects have a tail 3231 3232} 3233 3234void AudioFlinger::MixerThread::threadLoop_sleepTime() 3235{ 3236 // If no tracks are ready, sleep once for the duration of an output 3237 // buffer size, then write 0s to the output 3238 if (sleepTime == 0) { 3239 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3240 sleepTime = activeSleepTime >> sleepTimeShift; 3241 if (sleepTime < kMinThreadSleepTimeUs) { 3242 sleepTime = kMinThreadSleepTimeUs; 3243 } 3244 // reduce sleep time in case of consecutive application underruns to avoid 3245 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3246 // duration we would end up writing less data than needed by the audio HAL if 3247 // the condition persists. 3248 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3249 sleepTimeShift++; 3250 } 3251 } else { 3252 sleepTime = idleSleepTime; 3253 } 3254 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3255 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3256 // before effects processing or output. 3257 if (mMixerBufferValid) { 3258 memset(mMixerBuffer, 0, mMixerBufferSize); 3259 } else { 3260 memset(mSinkBuffer, 0, mSinkBufferSize); 3261 } 3262 sleepTime = 0; 3263 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3264 "anticipated start"); 3265 } 3266 // TODO add standby time extension fct of effect tail 3267} 3268 3269// prepareTracks_l() must be called with ThreadBase::mLock held 3270AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3271 Vector< sp<Track> > *tracksToRemove) 3272{ 3273 3274 mixer_state mixerStatus = MIXER_IDLE; 3275 // find out which tracks need to be processed 3276 size_t count = mActiveTracks.size(); 3277 size_t mixedTracks = 0; 3278 size_t tracksWithEffect = 0; 3279 // counts only _active_ fast tracks 3280 size_t fastTracks = 0; 3281 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3282 3283 float masterVolume = mMasterVolume; 3284 bool masterMute = mMasterMute; 3285 3286 if (masterMute) { 3287 masterVolume = 0; 3288 } 3289 // Delegate master volume control to effect in output mix effect chain if needed 3290 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3291 if (chain != 0) { 3292 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3293 chain->setVolume_l(&v, &v); 3294 masterVolume = (float)((v + (1 << 23)) >> 24); 3295 chain.clear(); 3296 } 3297 3298 // prepare a new state to push 3299 FastMixerStateQueue *sq = NULL; 3300 FastMixerState *state = NULL; 3301 bool didModify = false; 3302 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3303 if (mFastMixer != 0) { 3304 sq = mFastMixer->sq(); 3305 state = sq->begin(); 3306 } 3307 3308 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3309 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3310 3311 for (size_t i=0 ; i<count ; i++) { 3312 const sp<Track> t = mActiveTracks[i].promote(); 3313 if (t == 0) { 3314 continue; 3315 } 3316 3317 // this const just means the local variable doesn't change 3318 Track* const track = t.get(); 3319 3320 // process fast tracks 3321 if (track->isFastTrack()) { 3322 3323 // It's theoretically possible (though unlikely) for a fast track to be created 3324 // and then removed within the same normal mix cycle. This is not a problem, as 3325 // the track never becomes active so it's fast mixer slot is never touched. 3326 // The converse, of removing an (active) track and then creating a new track 3327 // at the identical fast mixer slot within the same normal mix cycle, 3328 // is impossible because the slot isn't marked available until the end of each cycle. 3329 int j = track->mFastIndex; 3330 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3331 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3332 FastTrack *fastTrack = &state->mFastTracks[j]; 3333 3334 // Determine whether the track is currently in underrun condition, 3335 // and whether it had a recent underrun. 3336 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3337 FastTrackUnderruns underruns = ftDump->mUnderruns; 3338 uint32_t recentFull = (underruns.mBitFields.mFull - 3339 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3340 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3341 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3342 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3343 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3344 uint32_t recentUnderruns = recentPartial + recentEmpty; 3345 track->mObservedUnderruns = underruns; 3346 // don't count underruns that occur while stopping or pausing 3347 // or stopped which can occur when flush() is called while active 3348 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3349 recentUnderruns > 0) { 3350 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3351 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3352 } 3353 3354 // This is similar to the state machine for normal tracks, 3355 // with a few modifications for fast tracks. 3356 bool isActive = true; 3357 switch (track->mState) { 3358 case TrackBase::STOPPING_1: 3359 // track stays active in STOPPING_1 state until first underrun 3360 if (recentUnderruns > 0 || track->isTerminated()) { 3361 track->mState = TrackBase::STOPPING_2; 3362 } 3363 break; 3364 case TrackBase::PAUSING: 3365 // ramp down is not yet implemented 3366 track->setPaused(); 3367 break; 3368 case TrackBase::RESUMING: 3369 // ramp up is not yet implemented 3370 track->mState = TrackBase::ACTIVE; 3371 break; 3372 case TrackBase::ACTIVE: 3373 if (recentFull > 0 || recentPartial > 0) { 3374 // track has provided at least some frames recently: reset retry count 3375 track->mRetryCount = kMaxTrackRetries; 3376 } 3377 if (recentUnderruns == 0) { 3378 // no recent underruns: stay active 3379 break; 3380 } 3381 // there has recently been an underrun of some kind 3382 if (track->sharedBuffer() == 0) { 3383 // were any of the recent underruns "empty" (no frames available)? 3384 if (recentEmpty == 0) { 3385 // no, then ignore the partial underruns as they are allowed indefinitely 3386 break; 3387 } 3388 // there has recently been an "empty" underrun: decrement the retry counter 3389 if (--(track->mRetryCount) > 0) { 3390 break; 3391 } 3392 // indicate to client process that the track was disabled because of underrun; 3393 // it will then automatically call start() when data is available 3394 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3395 // remove from active list, but state remains ACTIVE [confusing but true] 3396 isActive = false; 3397 break; 3398 } 3399 // fall through 3400 case TrackBase::STOPPING_2: 3401 case TrackBase::PAUSED: 3402 case TrackBase::STOPPED: 3403 case TrackBase::FLUSHED: // flush() while active 3404 // Check for presentation complete if track is inactive 3405 // We have consumed all the buffers of this track. 3406 // This would be incomplete if we auto-paused on underrun 3407 { 3408 size_t audioHALFrames = 3409 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3410 size_t framesWritten = mBytesWritten / mFrameSize; 3411 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3412 // track stays in active list until presentation is complete 3413 break; 3414 } 3415 } 3416 if (track->isStopping_2()) { 3417 track->mState = TrackBase::STOPPED; 3418 } 3419 if (track->isStopped()) { 3420 // Can't reset directly, as fast mixer is still polling this track 3421 // track->reset(); 3422 // So instead mark this track as needing to be reset after push with ack 3423 resetMask |= 1 << i; 3424 } 3425 isActive = false; 3426 break; 3427 case TrackBase::IDLE: 3428 default: 3429 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3430 } 3431 3432 if (isActive) { 3433 // was it previously inactive? 3434 if (!(state->mTrackMask & (1 << j))) { 3435 ExtendedAudioBufferProvider *eabp = track; 3436 VolumeProvider *vp = track; 3437 fastTrack->mBufferProvider = eabp; 3438 fastTrack->mVolumeProvider = vp; 3439 fastTrack->mChannelMask = track->mChannelMask; 3440 fastTrack->mFormat = track->mFormat; 3441 fastTrack->mGeneration++; 3442 state->mTrackMask |= 1 << j; 3443 didModify = true; 3444 // no acknowledgement required for newly active tracks 3445 } 3446 // cache the combined master volume and stream type volume for fast mixer; this 3447 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3448 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3449 ++fastTracks; 3450 } else { 3451 // was it previously active? 3452 if (state->mTrackMask & (1 << j)) { 3453 fastTrack->mBufferProvider = NULL; 3454 fastTrack->mGeneration++; 3455 state->mTrackMask &= ~(1 << j); 3456 didModify = true; 3457 // If any fast tracks were removed, we must wait for acknowledgement 3458 // because we're about to decrement the last sp<> on those tracks. 3459 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3460 } else { 3461 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3462 } 3463 tracksToRemove->add(track); 3464 // Avoids a misleading display in dumpsys 3465 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3466 } 3467 continue; 3468 } 3469 3470 { // local variable scope to avoid goto warning 3471 3472 audio_track_cblk_t* cblk = track->cblk(); 3473 3474 // The first time a track is added we wait 3475 // for all its buffers to be filled before processing it 3476 int name = track->name(); 3477 // make sure that we have enough frames to mix one full buffer. 3478 // enforce this condition only once to enable draining the buffer in case the client 3479 // app does not call stop() and relies on underrun to stop: 3480 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3481 // during last round 3482 size_t desiredFrames; 3483 uint32_t sr = track->sampleRate(); 3484 if (sr == mSampleRate) { 3485 desiredFrames = mNormalFrameCount; 3486 } else { 3487 desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); 3488 // add frames already consumed but not yet released by the resampler 3489 // because mAudioTrackServerProxy->framesReady() will include these frames 3490 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3491#if 0 3492 // the minimum track buffer size is normally twice the number of frames necessary 3493 // to fill one buffer and the resampler should not leave more than one buffer worth 3494 // of unreleased frames after each pass, but just in case... 3495 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3496#endif 3497 } 3498 uint32_t minFrames = 1; 3499 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3500 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3501 minFrames = desiredFrames; 3502 } 3503 3504 size_t framesReady = track->framesReady(); 3505 if (ATRACE_ENABLED()) { 3506 // I wish we had formatted trace names 3507 char traceName[16]; 3508 strcpy(traceName, "nRdy"); 3509 int name = track->name(); 3510 if (AudioMixer::TRACK0 <= name && 3511 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3512 name -= AudioMixer::TRACK0; 3513 traceName[4] = (name / 10) + '0'; 3514 traceName[5] = (name % 10) + '0'; 3515 } else { 3516 traceName[4] = '?'; 3517 traceName[5] = '?'; 3518 } 3519 traceName[6] = '\0'; 3520 ATRACE_INT(traceName, framesReady); 3521 } 3522 if ((framesReady >= minFrames) && track->isReady() && 3523 !track->isPaused() && !track->isTerminated()) 3524 { 3525 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3526 3527 mixedTracks++; 3528 3529 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3530 // there is an effect chain connected to the track 3531 chain.clear(); 3532 if (track->mainBuffer() != mSinkBuffer && 3533 track->mainBuffer() != mMixerBuffer) { 3534 if (mEffectBufferEnabled) { 3535 mEffectBufferValid = true; // Later can set directly. 3536 } 3537 chain = getEffectChain_l(track->sessionId()); 3538 // Delegate volume control to effect in track effect chain if needed 3539 if (chain != 0) { 3540 tracksWithEffect++; 3541 } else { 3542 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3543 "session %d", 3544 name, track->sessionId()); 3545 } 3546 } 3547 3548 3549 int param = AudioMixer::VOLUME; 3550 if (track->mFillingUpStatus == Track::FS_FILLED) { 3551 // no ramp for the first volume setting 3552 track->mFillingUpStatus = Track::FS_ACTIVE; 3553 if (track->mState == TrackBase::RESUMING) { 3554 track->mState = TrackBase::ACTIVE; 3555 param = AudioMixer::RAMP_VOLUME; 3556 } 3557 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3558 // FIXME should not make a decision based on mServer 3559 } else if (cblk->mServer != 0) { 3560 // If the track is stopped before the first frame was mixed, 3561 // do not apply ramp 3562 param = AudioMixer::RAMP_VOLUME; 3563 } 3564 3565 // compute volume for this track 3566 uint32_t vl, vr; // in U8.24 integer format 3567 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3568 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3569 vl = vr = 0; 3570 vlf = vrf = vaf = 0.; 3571 if (track->isPausing()) { 3572 track->setPaused(); 3573 } 3574 } else { 3575 3576 // read original volumes with volume control 3577 float typeVolume = mStreamTypes[track->streamType()].volume; 3578 float v = masterVolume * typeVolume; 3579 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3580 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3581 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3582 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3583 // track volumes come from shared memory, so can't be trusted and must be clamped 3584 if (vlf > GAIN_FLOAT_UNITY) { 3585 ALOGV("Track left volume out of range: %.3g", vlf); 3586 vlf = GAIN_FLOAT_UNITY; 3587 } 3588 if (vrf > GAIN_FLOAT_UNITY) { 3589 ALOGV("Track right volume out of range: %.3g", vrf); 3590 vrf = GAIN_FLOAT_UNITY; 3591 } 3592 // now apply the master volume and stream type volume 3593 vlf *= v; 3594 vrf *= v; 3595 // assuming master volume and stream type volume each go up to 1.0, 3596 // then derive vl and vr as U8.24 versions for the effect chain 3597 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3598 vl = (uint32_t) (scaleto8_24 * vlf); 3599 vr = (uint32_t) (scaleto8_24 * vrf); 3600 // vl and vr are now in U8.24 format 3601 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3602 // send level comes from shared memory and so may be corrupt 3603 if (sendLevel > MAX_GAIN_INT) { 3604 ALOGV("Track send level out of range: %04X", sendLevel); 3605 sendLevel = MAX_GAIN_INT; 3606 } 3607 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3608 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3609 } 3610 3611 // Delegate volume control to effect in track effect chain if needed 3612 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3613 // Do not ramp volume if volume is controlled by effect 3614 param = AudioMixer::VOLUME; 3615 // Update remaining floating point volume levels 3616 vlf = (float)vl / (1 << 24); 3617 vrf = (float)vr / (1 << 24); 3618 track->mHasVolumeController = true; 3619 } else { 3620 // force no volume ramp when volume controller was just disabled or removed 3621 // from effect chain to avoid volume spike 3622 if (track->mHasVolumeController) { 3623 param = AudioMixer::VOLUME; 3624 } 3625 track->mHasVolumeController = false; 3626 } 3627 3628 // XXX: these things DON'T need to be done each time 3629 mAudioMixer->setBufferProvider(name, track); 3630 mAudioMixer->enable(name); 3631 3632 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3633 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3634 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3635 mAudioMixer->setParameter( 3636 name, 3637 AudioMixer::TRACK, 3638 AudioMixer::FORMAT, (void *)track->format()); 3639 mAudioMixer->setParameter( 3640 name, 3641 AudioMixer::TRACK, 3642 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3643 mAudioMixer->setParameter( 3644 name, 3645 AudioMixer::TRACK, 3646 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3647 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3648 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3649 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3650 if (reqSampleRate == 0) { 3651 reqSampleRate = mSampleRate; 3652 } else if (reqSampleRate > maxSampleRate) { 3653 reqSampleRate = maxSampleRate; 3654 } 3655 mAudioMixer->setParameter( 3656 name, 3657 AudioMixer::RESAMPLE, 3658 AudioMixer::SAMPLE_RATE, 3659 (void *)(uintptr_t)reqSampleRate); 3660 /* 3661 * Select the appropriate output buffer for the track. 3662 * 3663 * Tracks with effects go into their own effects chain buffer 3664 * and from there into either mEffectBuffer or mSinkBuffer. 3665 * 3666 * Other tracks can use mMixerBuffer for higher precision 3667 * channel accumulation. If this buffer is enabled 3668 * (mMixerBufferEnabled true), then selected tracks will accumulate 3669 * into it. 3670 * 3671 */ 3672 if (mMixerBufferEnabled 3673 && (track->mainBuffer() == mSinkBuffer 3674 || track->mainBuffer() == mMixerBuffer)) { 3675 mAudioMixer->setParameter( 3676 name, 3677 AudioMixer::TRACK, 3678 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3679 mAudioMixer->setParameter( 3680 name, 3681 AudioMixer::TRACK, 3682 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3683 // TODO: override track->mainBuffer()? 3684 mMixerBufferValid = true; 3685 } else { 3686 mAudioMixer->setParameter( 3687 name, 3688 AudioMixer::TRACK, 3689 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3690 mAudioMixer->setParameter( 3691 name, 3692 AudioMixer::TRACK, 3693 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3694 } 3695 mAudioMixer->setParameter( 3696 name, 3697 AudioMixer::TRACK, 3698 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3699 3700 // reset retry count 3701 track->mRetryCount = kMaxTrackRetries; 3702 3703 // If one track is ready, set the mixer ready if: 3704 // - the mixer was not ready during previous round OR 3705 // - no other track is not ready 3706 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3707 mixerStatus != MIXER_TRACKS_ENABLED) { 3708 mixerStatus = MIXER_TRACKS_READY; 3709 } 3710 } else { 3711 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3712 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3713 } 3714 // clear effect chain input buffer if an active track underruns to avoid sending 3715 // previous audio buffer again to effects 3716 chain = getEffectChain_l(track->sessionId()); 3717 if (chain != 0) { 3718 chain->clearInputBuffer(); 3719 } 3720 3721 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3722 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3723 track->isStopped() || track->isPaused()) { 3724 // We have consumed all the buffers of this track. 3725 // Remove it from the list of active tracks. 3726 // TODO: use actual buffer filling status instead of latency when available from 3727 // audio HAL 3728 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3729 size_t framesWritten = mBytesWritten / mFrameSize; 3730 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3731 if (track->isStopped()) { 3732 track->reset(); 3733 } 3734 tracksToRemove->add(track); 3735 } 3736 } else { 3737 // No buffers for this track. Give it a few chances to 3738 // fill a buffer, then remove it from active list. 3739 if (--(track->mRetryCount) <= 0) { 3740 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3741 tracksToRemove->add(track); 3742 // indicate to client process that the track was disabled because of underrun; 3743 // it will then automatically call start() when data is available 3744 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3745 // If one track is not ready, mark the mixer also not ready if: 3746 // - the mixer was ready during previous round OR 3747 // - no other track is ready 3748 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3749 mixerStatus != MIXER_TRACKS_READY) { 3750 mixerStatus = MIXER_TRACKS_ENABLED; 3751 } 3752 } 3753 mAudioMixer->disable(name); 3754 } 3755 3756 } // local variable scope to avoid goto warning 3757track_is_ready: ; 3758 3759 } 3760 3761 // Push the new FastMixer state if necessary 3762 bool pauseAudioWatchdog = false; 3763 if (didModify) { 3764 state->mFastTracksGen++; 3765 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3766 if (kUseFastMixer == FastMixer_Dynamic && 3767 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3768 state->mCommand = FastMixerState::COLD_IDLE; 3769 state->mColdFutexAddr = &mFastMixerFutex; 3770 state->mColdGen++; 3771 mFastMixerFutex = 0; 3772 if (kUseFastMixer == FastMixer_Dynamic) { 3773 mNormalSink = mOutputSink; 3774 } 3775 // If we go into cold idle, need to wait for acknowledgement 3776 // so that fast mixer stops doing I/O. 3777 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3778 pauseAudioWatchdog = true; 3779 } 3780 } 3781 if (sq != NULL) { 3782 sq->end(didModify); 3783 sq->push(block); 3784 } 3785#ifdef AUDIO_WATCHDOG 3786 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3787 mAudioWatchdog->pause(); 3788 } 3789#endif 3790 3791 // Now perform the deferred reset on fast tracks that have stopped 3792 while (resetMask != 0) { 3793 size_t i = __builtin_ctz(resetMask); 3794 ALOG_ASSERT(i < count); 3795 resetMask &= ~(1 << i); 3796 sp<Track> t = mActiveTracks[i].promote(); 3797 if (t == 0) { 3798 continue; 3799 } 3800 Track* track = t.get(); 3801 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3802 track->reset(); 3803 } 3804 3805 // remove all the tracks that need to be... 3806 removeTracks_l(*tracksToRemove); 3807 3808 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3809 mEffectBufferValid = true; 3810 } 3811 3812 if (mEffectBufferValid) { 3813 // as long as there are effects we should clear the effects buffer, to avoid 3814 // passing a non-clean buffer to the effect chain 3815 memset(mEffectBuffer, 0, mEffectBufferSize); 3816 } 3817 // sink or mix buffer must be cleared if all tracks are connected to an 3818 // effect chain as in this case the mixer will not write to the sink or mix buffer 3819 // and track effects will accumulate into it 3820 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3821 (mixedTracks == 0 && fastTracks > 0))) { 3822 // FIXME as a performance optimization, should remember previous zero status 3823 if (mMixerBufferValid) { 3824 memset(mMixerBuffer, 0, mMixerBufferSize); 3825 // TODO: In testing, mSinkBuffer below need not be cleared because 3826 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3827 // after mixing. 3828 // 3829 // To enforce this guarantee: 3830 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3831 // (mixedTracks == 0 && fastTracks > 0)) 3832 // must imply MIXER_TRACKS_READY. 3833 // Later, we may clear buffers regardless, and skip much of this logic. 3834 } 3835 // FIXME as a performance optimization, should remember previous zero status 3836 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3837 } 3838 3839 // if any fast tracks, then status is ready 3840 mMixerStatusIgnoringFastTracks = mixerStatus; 3841 if (fastTracks > 0) { 3842 mixerStatus = MIXER_TRACKS_READY; 3843 } 3844 return mixerStatus; 3845} 3846 3847// getTrackName_l() must be called with ThreadBase::mLock held 3848int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3849 audio_format_t format, int sessionId) 3850{ 3851 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3852} 3853 3854// deleteTrackName_l() must be called with ThreadBase::mLock held 3855void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3856{ 3857 ALOGV("remove track (%d) and delete from mixer", name); 3858 mAudioMixer->deleteTrackName(name); 3859} 3860 3861// checkForNewParameter_l() must be called with ThreadBase::mLock held 3862bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3863 status_t& status) 3864{ 3865 bool reconfig = false; 3866 3867 status = NO_ERROR; 3868 3869 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3870 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3871 if (mFastMixer != 0) { 3872 FastMixerStateQueue *sq = mFastMixer->sq(); 3873 FastMixerState *state = sq->begin(); 3874 if (!(state->mCommand & FastMixerState::IDLE)) { 3875 previousCommand = state->mCommand; 3876 state->mCommand = FastMixerState::HOT_IDLE; 3877 sq->end(); 3878 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3879 } else { 3880 sq->end(false /*didModify*/); 3881 } 3882 } 3883 3884 AudioParameter param = AudioParameter(keyValuePair); 3885 int value; 3886 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3887 reconfig = true; 3888 } 3889 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3890 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3891 status = BAD_VALUE; 3892 } else { 3893 // no need to save value, since it's constant 3894 reconfig = true; 3895 } 3896 } 3897 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3898 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3899 status = BAD_VALUE; 3900 } else { 3901 // no need to save value, since it's constant 3902 reconfig = true; 3903 } 3904 } 3905 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3906 // do not accept frame count changes if tracks are open as the track buffer 3907 // size depends on frame count and correct behavior would not be guaranteed 3908 // if frame count is changed after track creation 3909 if (!mTracks.isEmpty()) { 3910 status = INVALID_OPERATION; 3911 } else { 3912 reconfig = true; 3913 } 3914 } 3915 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3916#ifdef ADD_BATTERY_DATA 3917 // when changing the audio output device, call addBatteryData to notify 3918 // the change 3919 if (mOutDevice != value) { 3920 uint32_t params = 0; 3921 // check whether speaker is on 3922 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3923 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3924 } 3925 3926 audio_devices_t deviceWithoutSpeaker 3927 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3928 // check if any other device (except speaker) is on 3929 if (value & deviceWithoutSpeaker ) { 3930 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3931 } 3932 3933 if (params != 0) { 3934 addBatteryData(params); 3935 } 3936 } 3937#endif 3938 3939 // forward device change to effects that have requested to be 3940 // aware of attached audio device. 3941 if (value != AUDIO_DEVICE_NONE) { 3942 mOutDevice = value; 3943 for (size_t i = 0; i < mEffectChains.size(); i++) { 3944 mEffectChains[i]->setDevice_l(mOutDevice); 3945 } 3946 } 3947 } 3948 3949 if (status == NO_ERROR) { 3950 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3951 keyValuePair.string()); 3952 if (!mStandby && status == INVALID_OPERATION) { 3953 mOutput->stream->common.standby(&mOutput->stream->common); 3954 mStandby = true; 3955 mBytesWritten = 0; 3956 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3957 keyValuePair.string()); 3958 } 3959 if (status == NO_ERROR && reconfig) { 3960 readOutputParameters_l(); 3961 delete mAudioMixer; 3962 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3963 for (size_t i = 0; i < mTracks.size() ; i++) { 3964 int name = getTrackName_l(mTracks[i]->mChannelMask, 3965 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3966 if (name < 0) { 3967 break; 3968 } 3969 mTracks[i]->mName = name; 3970 } 3971 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3972 } 3973 } 3974 3975 if (!(previousCommand & FastMixerState::IDLE)) { 3976 ALOG_ASSERT(mFastMixer != 0); 3977 FastMixerStateQueue *sq = mFastMixer->sq(); 3978 FastMixerState *state = sq->begin(); 3979 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3980 state->mCommand = previousCommand; 3981 sq->end(); 3982 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3983 } 3984 3985 return reconfig; 3986} 3987 3988 3989void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3990{ 3991 const size_t SIZE = 256; 3992 char buffer[SIZE]; 3993 String8 result; 3994 3995 PlaybackThread::dumpInternals(fd, args); 3996 3997 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3998 3999 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4000 const FastMixerDumpState copy(mFastMixerDumpState); 4001 copy.dump(fd); 4002 4003#ifdef STATE_QUEUE_DUMP 4004 // Similar for state queue 4005 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4006 observerCopy.dump(fd); 4007 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4008 mutatorCopy.dump(fd); 4009#endif 4010 4011#ifdef TEE_SINK 4012 // Write the tee output to a .wav file 4013 dumpTee(fd, mTeeSource, mId); 4014#endif 4015 4016#ifdef AUDIO_WATCHDOG 4017 if (mAudioWatchdog != 0) { 4018 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4019 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4020 wdCopy.dump(fd); 4021 } 4022#endif 4023} 4024 4025uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4026{ 4027 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4028} 4029 4030uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4031{ 4032 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4033} 4034 4035void AudioFlinger::MixerThread::cacheParameters_l() 4036{ 4037 PlaybackThread::cacheParameters_l(); 4038 4039 // FIXME: Relaxed timing because of a certain device that can't meet latency 4040 // Should be reduced to 2x after the vendor fixes the driver issue 4041 // increase threshold again due to low power audio mode. The way this warning 4042 // threshold is calculated and its usefulness should be reconsidered anyway. 4043 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4044} 4045 4046// ---------------------------------------------------------------------------- 4047 4048AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4049 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4050 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4051 // mLeftVolFloat, mRightVolFloat 4052{ 4053} 4054 4055AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4056 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4057 ThreadBase::type_t type) 4058 : PlaybackThread(audioFlinger, output, id, device, type) 4059 // mLeftVolFloat, mRightVolFloat 4060{ 4061} 4062 4063AudioFlinger::DirectOutputThread::~DirectOutputThread() 4064{ 4065} 4066 4067void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4068{ 4069 audio_track_cblk_t* cblk = track->cblk(); 4070 float left, right; 4071 4072 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4073 left = right = 0; 4074 } else { 4075 float typeVolume = mStreamTypes[track->streamType()].volume; 4076 float v = mMasterVolume * typeVolume; 4077 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4078 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4079 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4080 if (left > GAIN_FLOAT_UNITY) { 4081 left = GAIN_FLOAT_UNITY; 4082 } 4083 left *= v; 4084 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4085 if (right > GAIN_FLOAT_UNITY) { 4086 right = GAIN_FLOAT_UNITY; 4087 } 4088 right *= v; 4089 } 4090 4091 if (lastTrack) { 4092 if (left != mLeftVolFloat || right != mRightVolFloat) { 4093 mLeftVolFloat = left; 4094 mRightVolFloat = right; 4095 4096 // Convert volumes from float to 8.24 4097 uint32_t vl = (uint32_t)(left * (1 << 24)); 4098 uint32_t vr = (uint32_t)(right * (1 << 24)); 4099 4100 // Delegate volume control to effect in track effect chain if needed 4101 // only one effect chain can be present on DirectOutputThread, so if 4102 // there is one, the track is connected to it 4103 if (!mEffectChains.isEmpty()) { 4104 mEffectChains[0]->setVolume_l(&vl, &vr); 4105 left = (float)vl / (1 << 24); 4106 right = (float)vr / (1 << 24); 4107 } 4108 if (mOutput->stream->set_volume) { 4109 mOutput->stream->set_volume(mOutput->stream, left, right); 4110 } 4111 } 4112 } 4113} 4114 4115 4116AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4117 Vector< sp<Track> > *tracksToRemove 4118) 4119{ 4120 size_t count = mActiveTracks.size(); 4121 mixer_state mixerStatus = MIXER_IDLE; 4122 bool doHwPause = false; 4123 bool doHwResume = false; 4124 bool flushPending = false; 4125 4126 // find out which tracks need to be processed 4127 for (size_t i = 0; i < count; i++) { 4128 sp<Track> t = mActiveTracks[i].promote(); 4129 // The track died recently 4130 if (t == 0) { 4131 continue; 4132 } 4133 4134 Track* const track = t.get(); 4135 audio_track_cblk_t* cblk = track->cblk(); 4136 // Only consider last track started for volume and mixer state control. 4137 // In theory an older track could underrun and restart after the new one starts 4138 // but as we only care about the transition phase between two tracks on a 4139 // direct output, it is not a problem to ignore the underrun case. 4140 sp<Track> l = mLatestActiveTrack.promote(); 4141 bool last = l.get() == track; 4142 4143 if (mHwSupportsPause && track->isPausing()) { 4144 track->setPaused(); 4145 if (last && !mHwPaused) { 4146 doHwPause = true; 4147 mHwPaused = true; 4148 } 4149 tracksToRemove->add(track); 4150 } else if (track->isFlushPending()) { 4151 track->flushAck(); 4152 if (last) { 4153 flushPending = true; 4154 } 4155 } else if (mHwSupportsPause && track->isResumePending()){ 4156 track->resumeAck(); 4157 if (last) { 4158 if (mHwPaused) { 4159 doHwResume = true; 4160 mHwPaused = false; 4161 } 4162 } 4163 } 4164 4165 // The first time a track is added we wait 4166 // for all its buffers to be filled before processing it. 4167 // Allow draining the buffer in case the client 4168 // app does not call stop() and relies on underrun to stop: 4169 // hence the test on (track->mRetryCount > 1). 4170 // If retryCount<=1 then track is about to underrun and be removed. 4171 uint32_t minFrames; 4172 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4173 && (track->mRetryCount > 1)) { 4174 minFrames = mNormalFrameCount; 4175 } else { 4176 minFrames = 1; 4177 } 4178 4179 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4180 !track->isStopping_2() && !track->isStopped()) 4181 { 4182 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4183 4184 if (track->mFillingUpStatus == Track::FS_FILLED) { 4185 track->mFillingUpStatus = Track::FS_ACTIVE; 4186 // make sure processVolume_l() will apply new volume even if 0 4187 mLeftVolFloat = mRightVolFloat = -1.0; 4188 if (!mHwSupportsPause) { 4189 track->resumeAck(); 4190 } 4191 } 4192 4193 // compute volume for this track 4194 processVolume_l(track, last); 4195 if (last) { 4196 // reset retry count 4197 track->mRetryCount = kMaxTrackRetriesDirect; 4198 mActiveTrack = t; 4199 mixerStatus = MIXER_TRACKS_READY; 4200 if (usesHwAvSync() && mHwPaused) { 4201 doHwResume = true; 4202 mHwPaused = false; 4203 } 4204 } 4205 } else { 4206 // clear effect chain input buffer if the last active track started underruns 4207 // to avoid sending previous audio buffer again to effects 4208 if (!mEffectChains.isEmpty() && last) { 4209 mEffectChains[0]->clearInputBuffer(); 4210 } 4211 if (track->isStopping_1()) { 4212 track->mState = TrackBase::STOPPING_2; 4213 } 4214 if ((track->sharedBuffer() != 0) || track->isStopped() || 4215 track->isStopping_2() || track->isPaused()) { 4216 // We have consumed all the buffers of this track. 4217 // Remove it from the list of active tracks. 4218 size_t audioHALFrames; 4219 if (audio_is_linear_pcm(mFormat)) { 4220 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4221 } else { 4222 audioHALFrames = 0; 4223 } 4224 4225 size_t framesWritten = mBytesWritten / mFrameSize; 4226 if (mStandby || !last || 4227 track->presentationComplete(framesWritten, audioHALFrames)) { 4228 if (track->isStopping_2()) { 4229 track->mState = TrackBase::STOPPED; 4230 } 4231 if (track->isStopped()) { 4232 track->reset(); 4233 } 4234 tracksToRemove->add(track); 4235 } 4236 } else { 4237 // No buffers for this track. Give it a few chances to 4238 // fill a buffer, then remove it from active list. 4239 // Only consider last track started for mixer state control 4240 if (--(track->mRetryCount) <= 0) { 4241 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4242 tracksToRemove->add(track); 4243 // indicate to client process that the track was disabled because of underrun; 4244 // it will then automatically call start() when data is available 4245 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4246 } else if (last) { 4247 mixerStatus = MIXER_TRACKS_ENABLED; 4248 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4249 doHwPause = true; 4250 mHwPaused = true; 4251 } 4252 } 4253 } 4254 } 4255 } 4256 4257 // if an active track did not command a flush, check for pending flush on stopped tracks 4258 if (!flushPending) { 4259 for (size_t i = 0; i < mTracks.size(); i++) { 4260 if (mTracks[i]->isFlushPending()) { 4261 mTracks[i]->flushAck(); 4262 flushPending = true; 4263 } 4264 } 4265 } 4266 4267 // make sure the pause/flush/resume sequence is executed in the right order. 4268 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4269 // before flush and then resume HW. This can happen in case of pause/flush/resume 4270 // if resume is received before pause is executed. 4271 if (mHwSupportsPause && !mStandby && 4272 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4273 mOutput->stream->pause(mOutput->stream); 4274 } 4275 if (flushPending) { 4276 flushHw_l(); 4277 } 4278 if (mHwSupportsPause && !mStandby && doHwResume) { 4279 mOutput->stream->resume(mOutput->stream); 4280 } 4281 // remove all the tracks that need to be... 4282 removeTracks_l(*tracksToRemove); 4283 4284 return mixerStatus; 4285} 4286 4287void AudioFlinger::DirectOutputThread::threadLoop_mix() 4288{ 4289 size_t frameCount = mFrameCount; 4290 int8_t *curBuf = (int8_t *)mSinkBuffer; 4291 // output audio to hardware 4292 while (frameCount) { 4293 AudioBufferProvider::Buffer buffer; 4294 buffer.frameCount = frameCount; 4295 mActiveTrack->getNextBuffer(&buffer); 4296 if (buffer.raw == NULL) { 4297 memset(curBuf, 0, frameCount * mFrameSize); 4298 break; 4299 } 4300 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4301 frameCount -= buffer.frameCount; 4302 curBuf += buffer.frameCount * mFrameSize; 4303 mActiveTrack->releaseBuffer(&buffer); 4304 } 4305 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4306 sleepTime = 0; 4307 standbyTime = systemTime() + standbyDelay; 4308 mActiveTrack.clear(); 4309} 4310 4311void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4312{ 4313 // do not write to HAL when paused 4314 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4315 sleepTime = idleSleepTime; 4316 return; 4317 } 4318 if (sleepTime == 0) { 4319 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4320 sleepTime = activeSleepTime; 4321 } else { 4322 sleepTime = idleSleepTime; 4323 } 4324 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4325 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4326 sleepTime = 0; 4327 } 4328} 4329 4330void AudioFlinger::DirectOutputThread::threadLoop_exit() 4331{ 4332 { 4333 Mutex::Autolock _l(mLock); 4334 bool flushPending = false; 4335 for (size_t i = 0; i < mTracks.size(); i++) { 4336 if (mTracks[i]->isFlushPending()) { 4337 mTracks[i]->flushAck(); 4338 flushPending = true; 4339 } 4340 } 4341 if (flushPending) { 4342 flushHw_l(); 4343 } 4344 } 4345 PlaybackThread::threadLoop_exit(); 4346} 4347 4348// must be called with thread mutex locked 4349bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4350{ 4351 bool trackPaused = false; 4352 4353 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4354 // after a timeout and we will enter standby then. 4355 if (mTracks.size() > 0) { 4356 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4357 } 4358 4359 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused)); 4360} 4361 4362// getTrackName_l() must be called with ThreadBase::mLock held 4363int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4364 audio_format_t format __unused, int sessionId __unused) 4365{ 4366 return 0; 4367} 4368 4369// deleteTrackName_l() must be called with ThreadBase::mLock held 4370void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4371{ 4372} 4373 4374// checkForNewParameter_l() must be called with ThreadBase::mLock held 4375bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4376 status_t& status) 4377{ 4378 bool reconfig = false; 4379 4380 status = NO_ERROR; 4381 4382 AudioParameter param = AudioParameter(keyValuePair); 4383 int value; 4384 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4385 // forward device change to effects that have requested to be 4386 // aware of attached audio device. 4387 if (value != AUDIO_DEVICE_NONE) { 4388 mOutDevice = value; 4389 for (size_t i = 0; i < mEffectChains.size(); i++) { 4390 mEffectChains[i]->setDevice_l(mOutDevice); 4391 } 4392 } 4393 } 4394 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4395 // do not accept frame count changes if tracks are open as the track buffer 4396 // size depends on frame count and correct behavior would not be garantied 4397 // if frame count is changed after track creation 4398 if (!mTracks.isEmpty()) { 4399 status = INVALID_OPERATION; 4400 } else { 4401 reconfig = true; 4402 } 4403 } 4404 if (status == NO_ERROR) { 4405 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4406 keyValuePair.string()); 4407 if (!mStandby && status == INVALID_OPERATION) { 4408 mOutput->stream->common.standby(&mOutput->stream->common); 4409 mStandby = true; 4410 mBytesWritten = 0; 4411 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4412 keyValuePair.string()); 4413 } 4414 if (status == NO_ERROR && reconfig) { 4415 readOutputParameters_l(); 4416 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4417 } 4418 } 4419 4420 return reconfig; 4421} 4422 4423uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4424{ 4425 uint32_t time; 4426 if (audio_is_linear_pcm(mFormat)) { 4427 time = PlaybackThread::activeSleepTimeUs(); 4428 } else { 4429 time = 10000; 4430 } 4431 return time; 4432} 4433 4434uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4435{ 4436 uint32_t time; 4437 if (audio_is_linear_pcm(mFormat)) { 4438 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4439 } else { 4440 time = 10000; 4441 } 4442 return time; 4443} 4444 4445uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4446{ 4447 uint32_t time; 4448 if (audio_is_linear_pcm(mFormat)) { 4449 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4450 } else { 4451 time = 10000; 4452 } 4453 return time; 4454} 4455 4456void AudioFlinger::DirectOutputThread::cacheParameters_l() 4457{ 4458 PlaybackThread::cacheParameters_l(); 4459 4460 // use shorter standby delay as on normal output to release 4461 // hardware resources as soon as possible 4462 if (audio_is_linear_pcm(mFormat)) { 4463 standbyDelay = microseconds(activeSleepTime*2); 4464 } else { 4465 standbyDelay = kOffloadStandbyDelayNs; 4466 } 4467} 4468 4469void AudioFlinger::DirectOutputThread::flushHw_l() 4470{ 4471 if (mOutput->stream->flush != NULL) { 4472 mOutput->stream->flush(mOutput->stream); 4473 } 4474 mHwPaused = false; 4475} 4476 4477// ---------------------------------------------------------------------------- 4478 4479AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4480 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4481 : Thread(false /*canCallJava*/), 4482 mPlaybackThread(playbackThread), 4483 mWriteAckSequence(0), 4484 mDrainSequence(0) 4485{ 4486} 4487 4488AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4489{ 4490} 4491 4492void AudioFlinger::AsyncCallbackThread::onFirstRef() 4493{ 4494 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4495} 4496 4497bool AudioFlinger::AsyncCallbackThread::threadLoop() 4498{ 4499 while (!exitPending()) { 4500 uint32_t writeAckSequence; 4501 uint32_t drainSequence; 4502 4503 { 4504 Mutex::Autolock _l(mLock); 4505 while (!((mWriteAckSequence & 1) || 4506 (mDrainSequence & 1) || 4507 exitPending())) { 4508 mWaitWorkCV.wait(mLock); 4509 } 4510 4511 if (exitPending()) { 4512 break; 4513 } 4514 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4515 mWriteAckSequence, mDrainSequence); 4516 writeAckSequence = mWriteAckSequence; 4517 mWriteAckSequence &= ~1; 4518 drainSequence = mDrainSequence; 4519 mDrainSequence &= ~1; 4520 } 4521 { 4522 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4523 if (playbackThread != 0) { 4524 if (writeAckSequence & 1) { 4525 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4526 } 4527 if (drainSequence & 1) { 4528 playbackThread->resetDraining(drainSequence >> 1); 4529 } 4530 } 4531 } 4532 } 4533 return false; 4534} 4535 4536void AudioFlinger::AsyncCallbackThread::exit() 4537{ 4538 ALOGV("AsyncCallbackThread::exit"); 4539 Mutex::Autolock _l(mLock); 4540 requestExit(); 4541 mWaitWorkCV.broadcast(); 4542} 4543 4544void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4545{ 4546 Mutex::Autolock _l(mLock); 4547 // bit 0 is cleared 4548 mWriteAckSequence = sequence << 1; 4549} 4550 4551void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4552{ 4553 Mutex::Autolock _l(mLock); 4554 // ignore unexpected callbacks 4555 if (mWriteAckSequence & 2) { 4556 mWriteAckSequence |= 1; 4557 mWaitWorkCV.signal(); 4558 } 4559} 4560 4561void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4562{ 4563 Mutex::Autolock _l(mLock); 4564 // bit 0 is cleared 4565 mDrainSequence = sequence << 1; 4566} 4567 4568void AudioFlinger::AsyncCallbackThread::resetDraining() 4569{ 4570 Mutex::Autolock _l(mLock); 4571 // ignore unexpected callbacks 4572 if (mDrainSequence & 2) { 4573 mDrainSequence |= 1; 4574 mWaitWorkCV.signal(); 4575 } 4576} 4577 4578 4579// ---------------------------------------------------------------------------- 4580AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4581 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4582 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4583 mPausedBytesRemaining(0) 4584{ 4585 //FIXME: mStandby should be set to true by ThreadBase constructor 4586 mStandby = true; 4587} 4588 4589void AudioFlinger::OffloadThread::threadLoop_exit() 4590{ 4591 if (mFlushPending || mHwPaused) { 4592 // If a flush is pending or track was paused, just discard buffered data 4593 flushHw_l(); 4594 } else { 4595 mMixerStatus = MIXER_DRAIN_ALL; 4596 threadLoop_drain(); 4597 } 4598 if (mUseAsyncWrite) { 4599 ALOG_ASSERT(mCallbackThread != 0); 4600 mCallbackThread->exit(); 4601 } 4602 PlaybackThread::threadLoop_exit(); 4603} 4604 4605AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4606 Vector< sp<Track> > *tracksToRemove 4607) 4608{ 4609 size_t count = mActiveTracks.size(); 4610 4611 mixer_state mixerStatus = MIXER_IDLE; 4612 bool doHwPause = false; 4613 bool doHwResume = false; 4614 4615 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4616 4617 // find out which tracks need to be processed 4618 for (size_t i = 0; i < count; i++) { 4619 sp<Track> t = mActiveTracks[i].promote(); 4620 // The track died recently 4621 if (t == 0) { 4622 continue; 4623 } 4624 Track* const track = t.get(); 4625 audio_track_cblk_t* cblk = track->cblk(); 4626 // Only consider last track started for volume and mixer state control. 4627 // In theory an older track could underrun and restart after the new one starts 4628 // but as we only care about the transition phase between two tracks on a 4629 // direct output, it is not a problem to ignore the underrun case. 4630 sp<Track> l = mLatestActiveTrack.promote(); 4631 bool last = l.get() == track; 4632 4633 if (track->isInvalid()) { 4634 ALOGW("An invalidated track shouldn't be in active list"); 4635 tracksToRemove->add(track); 4636 continue; 4637 } 4638 4639 if (track->mState == TrackBase::IDLE) { 4640 ALOGW("An idle track shouldn't be in active list"); 4641 continue; 4642 } 4643 4644 if (track->isPausing()) { 4645 track->setPaused(); 4646 if (last) { 4647 if (!mHwPaused) { 4648 doHwPause = true; 4649 mHwPaused = true; 4650 } 4651 // If we were part way through writing the mixbuffer to 4652 // the HAL we must save this until we resume 4653 // BUG - this will be wrong if a different track is made active, 4654 // in that case we want to discard the pending data in the 4655 // mixbuffer and tell the client to present it again when the 4656 // track is resumed 4657 mPausedWriteLength = mCurrentWriteLength; 4658 mPausedBytesRemaining = mBytesRemaining; 4659 mBytesRemaining = 0; // stop writing 4660 } 4661 tracksToRemove->add(track); 4662 } else if (track->isFlushPending()) { 4663 track->flushAck(); 4664 if (last) { 4665 mFlushPending = true; 4666 } 4667 } else if (track->isResumePending()){ 4668 track->resumeAck(); 4669 if (last) { 4670 if (mPausedBytesRemaining) { 4671 // Need to continue write that was interrupted 4672 mCurrentWriteLength = mPausedWriteLength; 4673 mBytesRemaining = mPausedBytesRemaining; 4674 mPausedBytesRemaining = 0; 4675 } 4676 if (mHwPaused) { 4677 doHwResume = true; 4678 mHwPaused = false; 4679 // threadLoop_mix() will handle the case that we need to 4680 // resume an interrupted write 4681 } 4682 // enable write to audio HAL 4683 sleepTime = 0; 4684 4685 // Do not handle new data in this iteration even if track->framesReady() 4686 mixerStatus = MIXER_TRACKS_ENABLED; 4687 } 4688 } else if (track->framesReady() && track->isReady() && 4689 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4690 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4691 if (track->mFillingUpStatus == Track::FS_FILLED) { 4692 track->mFillingUpStatus = Track::FS_ACTIVE; 4693 // make sure processVolume_l() will apply new volume even if 0 4694 mLeftVolFloat = mRightVolFloat = -1.0; 4695 } 4696 4697 if (last) { 4698 sp<Track> previousTrack = mPreviousTrack.promote(); 4699 if (previousTrack != 0) { 4700 if (track != previousTrack.get()) { 4701 // Flush any data still being written from last track 4702 mBytesRemaining = 0; 4703 if (mPausedBytesRemaining) { 4704 // Last track was paused so we also need to flush saved 4705 // mixbuffer state and invalidate track so that it will 4706 // re-submit that unwritten data when it is next resumed 4707 mPausedBytesRemaining = 0; 4708 // Invalidate is a bit drastic - would be more efficient 4709 // to have a flag to tell client that some of the 4710 // previously written data was lost 4711 previousTrack->invalidate(); 4712 } 4713 // flush data already sent to the DSP if changing audio session as audio 4714 // comes from a different source. Also invalidate previous track to force a 4715 // seek when resuming. 4716 if (previousTrack->sessionId() != track->sessionId()) { 4717 previousTrack->invalidate(); 4718 } 4719 } 4720 } 4721 mPreviousTrack = track; 4722 // reset retry count 4723 track->mRetryCount = kMaxTrackRetriesOffload; 4724 mActiveTrack = t; 4725 mixerStatus = MIXER_TRACKS_READY; 4726 } 4727 } else { 4728 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4729 if (track->isStopping_1()) { 4730 // Hardware buffer can hold a large amount of audio so we must 4731 // wait for all current track's data to drain before we say 4732 // that the track is stopped. 4733 if (mBytesRemaining == 0) { 4734 // Only start draining when all data in mixbuffer 4735 // has been written 4736 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4737 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4738 // do not drain if no data was ever sent to HAL (mStandby == true) 4739 if (last && !mStandby) { 4740 // do not modify drain sequence if we are already draining. This happens 4741 // when resuming from pause after drain. 4742 if ((mDrainSequence & 1) == 0) { 4743 sleepTime = 0; 4744 standbyTime = systemTime() + standbyDelay; 4745 mixerStatus = MIXER_DRAIN_TRACK; 4746 mDrainSequence += 2; 4747 } 4748 if (mHwPaused) { 4749 // It is possible to move from PAUSED to STOPPING_1 without 4750 // a resume so we must ensure hardware is running 4751 doHwResume = true; 4752 mHwPaused = false; 4753 } 4754 } 4755 } 4756 } else if (track->isStopping_2()) { 4757 // Drain has completed or we are in standby, signal presentation complete 4758 if (!(mDrainSequence & 1) || !last || mStandby) { 4759 track->mState = TrackBase::STOPPED; 4760 size_t audioHALFrames = 4761 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4762 size_t framesWritten = 4763 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4764 track->presentationComplete(framesWritten, audioHALFrames); 4765 track->reset(); 4766 tracksToRemove->add(track); 4767 } 4768 } else { 4769 // No buffers for this track. Give it a few chances to 4770 // fill a buffer, then remove it from active list. 4771 if (--(track->mRetryCount) <= 0) { 4772 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4773 track->name()); 4774 tracksToRemove->add(track); 4775 // indicate to client process that the track was disabled because of underrun; 4776 // it will then automatically call start() when data is available 4777 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4778 } else if (last){ 4779 mixerStatus = MIXER_TRACKS_ENABLED; 4780 } 4781 } 4782 } 4783 // compute volume for this track 4784 processVolume_l(track, last); 4785 } 4786 4787 // make sure the pause/flush/resume sequence is executed in the right order. 4788 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4789 // before flush and then resume HW. This can happen in case of pause/flush/resume 4790 // if resume is received before pause is executed. 4791 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4792 mOutput->stream->pause(mOutput->stream); 4793 } 4794 if (mFlushPending) { 4795 flushHw_l(); 4796 mFlushPending = false; 4797 } 4798 if (!mStandby && doHwResume) { 4799 mOutput->stream->resume(mOutput->stream); 4800 } 4801 4802 // remove all the tracks that need to be... 4803 removeTracks_l(*tracksToRemove); 4804 4805 return mixerStatus; 4806} 4807 4808// must be called with thread mutex locked 4809bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4810{ 4811 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4812 mWriteAckSequence, mDrainSequence); 4813 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4814 return true; 4815 } 4816 return false; 4817} 4818 4819bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4820{ 4821 Mutex::Autolock _l(mLock); 4822 return waitingAsyncCallback_l(); 4823} 4824 4825void AudioFlinger::OffloadThread::flushHw_l() 4826{ 4827 DirectOutputThread::flushHw_l(); 4828 // Flush anything still waiting in the mixbuffer 4829 mCurrentWriteLength = 0; 4830 mBytesRemaining = 0; 4831 mPausedWriteLength = 0; 4832 mPausedBytesRemaining = 0; 4833 4834 if (mUseAsyncWrite) { 4835 // discard any pending drain or write ack by incrementing sequence 4836 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4837 mDrainSequence = (mDrainSequence + 2) & ~1; 4838 ALOG_ASSERT(mCallbackThread != 0); 4839 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4840 mCallbackThread->setDraining(mDrainSequence); 4841 } 4842} 4843 4844void AudioFlinger::OffloadThread::onAddNewTrack_l() 4845{ 4846 sp<Track> previousTrack = mPreviousTrack.promote(); 4847 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4848 4849 if (previousTrack != 0 && latestTrack != 0 && 4850 (previousTrack->sessionId() != latestTrack->sessionId())) { 4851 mFlushPending = true; 4852 } 4853 PlaybackThread::onAddNewTrack_l(); 4854} 4855 4856// ---------------------------------------------------------------------------- 4857 4858AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4859 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4860 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4861 DUPLICATING), 4862 mWaitTimeMs(UINT_MAX) 4863{ 4864 addOutputTrack(mainThread); 4865} 4866 4867AudioFlinger::DuplicatingThread::~DuplicatingThread() 4868{ 4869 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4870 mOutputTracks[i]->destroy(); 4871 } 4872} 4873 4874void AudioFlinger::DuplicatingThread::threadLoop_mix() 4875{ 4876 // mix buffers... 4877 if (outputsReady(outputTracks)) { 4878 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4879 } else { 4880 if (mMixerBufferValid) { 4881 memset(mMixerBuffer, 0, mMixerBufferSize); 4882 } else { 4883 memset(mSinkBuffer, 0, mSinkBufferSize); 4884 } 4885 } 4886 sleepTime = 0; 4887 writeFrames = mNormalFrameCount; 4888 mCurrentWriteLength = mSinkBufferSize; 4889 standbyTime = systemTime() + standbyDelay; 4890} 4891 4892void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4893{ 4894 if (sleepTime == 0) { 4895 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4896 sleepTime = activeSleepTime; 4897 } else { 4898 sleepTime = idleSleepTime; 4899 } 4900 } else if (mBytesWritten != 0) { 4901 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4902 writeFrames = mNormalFrameCount; 4903 memset(mSinkBuffer, 0, mSinkBufferSize); 4904 } else { 4905 // flush remaining overflow buffers in output tracks 4906 writeFrames = 0; 4907 } 4908 sleepTime = 0; 4909 } 4910} 4911 4912ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4913{ 4914 for (size_t i = 0; i < outputTracks.size(); i++) { 4915 outputTracks[i]->write(mSinkBuffer, writeFrames); 4916 } 4917 mStandby = false; 4918 return (ssize_t)mSinkBufferSize; 4919} 4920 4921void AudioFlinger::DuplicatingThread::threadLoop_standby() 4922{ 4923 // DuplicatingThread implements standby by stopping all tracks 4924 for (size_t i = 0; i < outputTracks.size(); i++) { 4925 outputTracks[i]->stop(); 4926 } 4927} 4928 4929void AudioFlinger::DuplicatingThread::saveOutputTracks() 4930{ 4931 outputTracks = mOutputTracks; 4932} 4933 4934void AudioFlinger::DuplicatingThread::clearOutputTracks() 4935{ 4936 outputTracks.clear(); 4937} 4938 4939void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4940{ 4941 Mutex::Autolock _l(mLock); 4942 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 4943 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 4944 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 4945 const size_t frameCount = 4946 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 4947 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 4948 // from different OutputTracks and their associated MixerThreads (e.g. one may 4949 // nearly empty and the other may be dropping data). 4950 4951 sp<OutputTrack> outputTrack = new OutputTrack(thread, 4952 this, 4953 mSampleRate, 4954 mFormat, 4955 mChannelMask, 4956 frameCount, 4957 IPCThreadState::self()->getCallingUid()); 4958 if (outputTrack->cblk() != NULL) { 4959 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 4960 mOutputTracks.add(outputTrack); 4961 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 4962 updateWaitTime_l(); 4963 } 4964} 4965 4966void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4967{ 4968 Mutex::Autolock _l(mLock); 4969 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4970 if (mOutputTracks[i]->thread() == thread) { 4971 mOutputTracks[i]->destroy(); 4972 mOutputTracks.removeAt(i); 4973 updateWaitTime_l(); 4974 return; 4975 } 4976 } 4977 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4978} 4979 4980// caller must hold mLock 4981void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4982{ 4983 mWaitTimeMs = UINT_MAX; 4984 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4985 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4986 if (strong != 0) { 4987 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4988 if (waitTimeMs < mWaitTimeMs) { 4989 mWaitTimeMs = waitTimeMs; 4990 } 4991 } 4992 } 4993} 4994 4995 4996bool AudioFlinger::DuplicatingThread::outputsReady( 4997 const SortedVector< sp<OutputTrack> > &outputTracks) 4998{ 4999 for (size_t i = 0; i < outputTracks.size(); i++) { 5000 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5001 if (thread == 0) { 5002 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5003 outputTracks[i].get()); 5004 return false; 5005 } 5006 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5007 // see note at standby() declaration 5008 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5009 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5010 thread.get()); 5011 return false; 5012 } 5013 } 5014 return true; 5015} 5016 5017uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5018{ 5019 return (mWaitTimeMs * 1000) / 2; 5020} 5021 5022void AudioFlinger::DuplicatingThread::cacheParameters_l() 5023{ 5024 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5025 updateWaitTime_l(); 5026 5027 MixerThread::cacheParameters_l(); 5028} 5029 5030// ---------------------------------------------------------------------------- 5031// Record 5032// ---------------------------------------------------------------------------- 5033 5034AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5035 AudioStreamIn *input, 5036 audio_io_handle_t id, 5037 audio_devices_t outDevice, 5038 audio_devices_t inDevice 5039#ifdef TEE_SINK 5040 , const sp<NBAIO_Sink>& teeSink 5041#endif 5042 ) : 5043 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5044 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5045 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5046 mRsmpInRear(0) 5047#ifdef TEE_SINK 5048 , mTeeSink(teeSink) 5049#endif 5050 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5051 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5052 // mFastCapture below 5053 , mFastCaptureFutex(0) 5054 // mInputSource 5055 // mPipeSink 5056 // mPipeSource 5057 , mPipeFramesP2(0) 5058 // mPipeMemory 5059 // mFastCaptureNBLogWriter 5060 , mFastTrackAvail(false) 5061{ 5062 snprintf(mName, kNameLength, "AudioIn_%X", id); 5063 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 5064 5065 readInputParameters_l(); 5066 5067 // create an NBAIO source for the HAL input stream, and negotiate 5068 mInputSource = new AudioStreamInSource(input->stream); 5069 size_t numCounterOffers = 0; 5070 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5071 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5072 ALOG_ASSERT(index == 0); 5073 5074 // initialize fast capture depending on configuration 5075 bool initFastCapture; 5076 switch (kUseFastCapture) { 5077 case FastCapture_Never: 5078 initFastCapture = false; 5079 break; 5080 case FastCapture_Always: 5081 initFastCapture = true; 5082 break; 5083 case FastCapture_Static: 5084 uint32_t primaryOutputSampleRate; 5085 { 5086 AutoMutex _l(audioFlinger->mHardwareLock); 5087 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5088 } 5089 initFastCapture = 5090 // either capture sample rate is same as (a reasonable) primary output sample rate 5091 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5092 (mSampleRate == primaryOutputSampleRate)) || 5093 // or primary output sample rate is unknown, and capture sample rate is reasonable 5094 ((primaryOutputSampleRate == 0) && 5095 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5096 // and the buffer size is < 12 ms 5097 (mFrameCount * 1000) / mSampleRate < 12; 5098 break; 5099 // case FastCapture_Dynamic: 5100 } 5101 5102 if (initFastCapture) { 5103 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 5104 NBAIO_Format format = mInputSource->format(); 5105 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5106 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5107 void *pipeBuffer; 5108 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5109 sp<IMemory> pipeMemory; 5110 if ((roHeap == 0) || 5111 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5112 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5113 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5114 goto failed; 5115 } 5116 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5117 memset(pipeBuffer, 0, pipeSize); 5118 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5119 const NBAIO_Format offers[1] = {format}; 5120 size_t numCounterOffers = 0; 5121 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5122 ALOG_ASSERT(index == 0); 5123 mPipeSink = pipe; 5124 PipeReader *pipeReader = new PipeReader(*pipe); 5125 numCounterOffers = 0; 5126 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5127 ALOG_ASSERT(index == 0); 5128 mPipeSource = pipeReader; 5129 mPipeFramesP2 = pipeFramesP2; 5130 mPipeMemory = pipeMemory; 5131 5132 // create fast capture 5133 mFastCapture = new FastCapture(); 5134 FastCaptureStateQueue *sq = mFastCapture->sq(); 5135#ifdef STATE_QUEUE_DUMP 5136 // FIXME 5137#endif 5138 FastCaptureState *state = sq->begin(); 5139 state->mCblk = NULL; 5140 state->mInputSource = mInputSource.get(); 5141 state->mInputSourceGen++; 5142 state->mPipeSink = pipe; 5143 state->mPipeSinkGen++; 5144 state->mFrameCount = mFrameCount; 5145 state->mCommand = FastCaptureState::COLD_IDLE; 5146 // already done in constructor initialization list 5147 //mFastCaptureFutex = 0; 5148 state->mColdFutexAddr = &mFastCaptureFutex; 5149 state->mColdGen++; 5150 state->mDumpState = &mFastCaptureDumpState; 5151#ifdef TEE_SINK 5152 // FIXME 5153#endif 5154 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5155 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5156 sq->end(); 5157 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5158 5159 // start the fast capture 5160 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5161 pid_t tid = mFastCapture->getTid(); 5162 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5163 if (err != 0) { 5164 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5165 kPriorityFastCapture, getpid_cached, tid, err); 5166 } 5167 5168#ifdef AUDIO_WATCHDOG 5169 // FIXME 5170#endif 5171 5172 mFastTrackAvail = true; 5173 } 5174failed: ; 5175 5176 // FIXME mNormalSource 5177} 5178 5179 5180AudioFlinger::RecordThread::~RecordThread() 5181{ 5182 if (mFastCapture != 0) { 5183 FastCaptureStateQueue *sq = mFastCapture->sq(); 5184 FastCaptureState *state = sq->begin(); 5185 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5186 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5187 if (old == -1) { 5188 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5189 } 5190 } 5191 state->mCommand = FastCaptureState::EXIT; 5192 sq->end(); 5193 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5194 mFastCapture->join(); 5195 mFastCapture.clear(); 5196 } 5197 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5198 mAudioFlinger->unregisterWriter(mNBLogWriter); 5199 delete[] mRsmpInBuffer; 5200} 5201 5202void AudioFlinger::RecordThread::onFirstRef() 5203{ 5204 run(mName, PRIORITY_URGENT_AUDIO); 5205} 5206 5207bool AudioFlinger::RecordThread::threadLoop() 5208{ 5209 nsecs_t lastWarning = 0; 5210 5211 inputStandBy(); 5212 5213reacquire_wakelock: 5214 sp<RecordTrack> activeTrack; 5215 int activeTracksGen; 5216 { 5217 Mutex::Autolock _l(mLock); 5218 size_t size = mActiveTracks.size(); 5219 activeTracksGen = mActiveTracksGen; 5220 if (size > 0) { 5221 // FIXME an arbitrary choice 5222 activeTrack = mActiveTracks[0]; 5223 acquireWakeLock_l(activeTrack->uid()); 5224 if (size > 1) { 5225 SortedVector<int> tmp; 5226 for (size_t i = 0; i < size; i++) { 5227 tmp.add(mActiveTracks[i]->uid()); 5228 } 5229 updateWakeLockUids_l(tmp); 5230 } 5231 } else { 5232 acquireWakeLock_l(-1); 5233 } 5234 } 5235 5236 // used to request a deferred sleep, to be executed later while mutex is unlocked 5237 uint32_t sleepUs = 0; 5238 5239 // loop while there is work to do 5240 for (;;) { 5241 Vector< sp<EffectChain> > effectChains; 5242 5243 // sleep with mutex unlocked 5244 if (sleepUs > 0) { 5245 ATRACE_BEGIN("sleep"); 5246 usleep(sleepUs); 5247 ATRACE_END(); 5248 sleepUs = 0; 5249 } 5250 5251 // activeTracks accumulates a copy of a subset of mActiveTracks 5252 Vector< sp<RecordTrack> > activeTracks; 5253 5254 // reference to the (first and only) active fast track 5255 sp<RecordTrack> fastTrack; 5256 5257 // reference to a fast track which is about to be removed 5258 sp<RecordTrack> fastTrackToRemove; 5259 5260 { // scope for mLock 5261 Mutex::Autolock _l(mLock); 5262 5263 processConfigEvents_l(); 5264 5265 // check exitPending here because checkForNewParameters_l() and 5266 // checkForNewParameters_l() can temporarily release mLock 5267 if (exitPending()) { 5268 break; 5269 } 5270 5271 // if no active track(s), then standby and release wakelock 5272 size_t size = mActiveTracks.size(); 5273 if (size == 0) { 5274 standbyIfNotAlreadyInStandby(); 5275 // exitPending() can't become true here 5276 releaseWakeLock_l(); 5277 ALOGV("RecordThread: loop stopping"); 5278 // go to sleep 5279 mWaitWorkCV.wait(mLock); 5280 ALOGV("RecordThread: loop starting"); 5281 goto reacquire_wakelock; 5282 } 5283 5284 if (mActiveTracksGen != activeTracksGen) { 5285 activeTracksGen = mActiveTracksGen; 5286 SortedVector<int> tmp; 5287 for (size_t i = 0; i < size; i++) { 5288 tmp.add(mActiveTracks[i]->uid()); 5289 } 5290 updateWakeLockUids_l(tmp); 5291 } 5292 5293 bool doBroadcast = false; 5294 for (size_t i = 0; i < size; ) { 5295 5296 activeTrack = mActiveTracks[i]; 5297 if (activeTrack->isTerminated()) { 5298 if (activeTrack->isFastTrack()) { 5299 ALOG_ASSERT(fastTrackToRemove == 0); 5300 fastTrackToRemove = activeTrack; 5301 } 5302 removeTrack_l(activeTrack); 5303 mActiveTracks.remove(activeTrack); 5304 mActiveTracksGen++; 5305 size--; 5306 continue; 5307 } 5308 5309 TrackBase::track_state activeTrackState = activeTrack->mState; 5310 switch (activeTrackState) { 5311 5312 case TrackBase::PAUSING: 5313 mActiveTracks.remove(activeTrack); 5314 mActiveTracksGen++; 5315 doBroadcast = true; 5316 size--; 5317 continue; 5318 5319 case TrackBase::STARTING_1: 5320 sleepUs = 10000; 5321 i++; 5322 continue; 5323 5324 case TrackBase::STARTING_2: 5325 doBroadcast = true; 5326 mStandby = false; 5327 activeTrack->mState = TrackBase::ACTIVE; 5328 break; 5329 5330 case TrackBase::ACTIVE: 5331 break; 5332 5333 case TrackBase::IDLE: 5334 i++; 5335 continue; 5336 5337 default: 5338 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5339 } 5340 5341 activeTracks.add(activeTrack); 5342 i++; 5343 5344 if (activeTrack->isFastTrack()) { 5345 ALOG_ASSERT(!mFastTrackAvail); 5346 ALOG_ASSERT(fastTrack == 0); 5347 fastTrack = activeTrack; 5348 } 5349 } 5350 if (doBroadcast) { 5351 mStartStopCond.broadcast(); 5352 } 5353 5354 // sleep if there are no active tracks to process 5355 if (activeTracks.size() == 0) { 5356 if (sleepUs == 0) { 5357 sleepUs = kRecordThreadSleepUs; 5358 } 5359 continue; 5360 } 5361 sleepUs = 0; 5362 5363 lockEffectChains_l(effectChains); 5364 } 5365 5366 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5367 5368 size_t size = effectChains.size(); 5369 for (size_t i = 0; i < size; i++) { 5370 // thread mutex is not locked, but effect chain is locked 5371 effectChains[i]->process_l(); 5372 } 5373 5374 // Push a new fast capture state if fast capture is not already running, or cblk change 5375 if (mFastCapture != 0) { 5376 FastCaptureStateQueue *sq = mFastCapture->sq(); 5377 FastCaptureState *state = sq->begin(); 5378 bool didModify = false; 5379 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5380 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5381 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5382 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5383 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5384 if (old == -1) { 5385 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5386 } 5387 } 5388 state->mCommand = FastCaptureState::READ_WRITE; 5389#if 0 // FIXME 5390 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5391 FastThreadDumpState::kSamplingNforLowRamDevice : 5392 FastThreadDumpState::kSamplingN); 5393#endif 5394 didModify = true; 5395 } 5396 audio_track_cblk_t *cblkOld = state->mCblk; 5397 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5398 if (cblkNew != cblkOld) { 5399 state->mCblk = cblkNew; 5400 // block until acked if removing a fast track 5401 if (cblkOld != NULL) { 5402 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5403 } 5404 didModify = true; 5405 } 5406 sq->end(didModify); 5407 if (didModify) { 5408 sq->push(block); 5409#if 0 5410 if (kUseFastCapture == FastCapture_Dynamic) { 5411 mNormalSource = mPipeSource; 5412 } 5413#endif 5414 } 5415 } 5416 5417 // now run the fast track destructor with thread mutex unlocked 5418 fastTrackToRemove.clear(); 5419 5420 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5421 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5422 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5423 // If destination is non-contiguous, first read past the nominal end of buffer, then 5424 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5425 5426 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5427 ssize_t framesRead; 5428 5429 // If an NBAIO source is present, use it to read the normal capture's data 5430 if (mPipeSource != 0) { 5431 size_t framesToRead = mBufferSize / mFrameSize; 5432 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5433 framesToRead, AudioBufferProvider::kInvalidPTS); 5434 if (framesRead == 0) { 5435 // since pipe is non-blocking, simulate blocking input 5436 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5437 } 5438 // otherwise use the HAL / AudioStreamIn directly 5439 } else { 5440 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5441 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5442 if (bytesRead < 0) { 5443 framesRead = bytesRead; 5444 } else { 5445 framesRead = bytesRead / mFrameSize; 5446 } 5447 } 5448 5449 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5450 ALOGE("read failed: framesRead=%d", framesRead); 5451 // Force input into standby so that it tries to recover at next read attempt 5452 inputStandBy(); 5453 sleepUs = kRecordThreadSleepUs; 5454 } 5455 if (framesRead <= 0) { 5456 goto unlock; 5457 } 5458 ALOG_ASSERT(framesRead > 0); 5459 5460 if (mTeeSink != 0) { 5461 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5462 } 5463 // If destination is non-contiguous, we now correct for reading past end of buffer. 5464 { 5465 size_t part1 = mRsmpInFramesP2 - rear; 5466 if ((size_t) framesRead > part1) { 5467 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5468 (framesRead - part1) * mFrameSize); 5469 } 5470 } 5471 rear = mRsmpInRear += framesRead; 5472 5473 size = activeTracks.size(); 5474 // loop over each active track 5475 for (size_t i = 0; i < size; i++) { 5476 activeTrack = activeTracks[i]; 5477 5478 // skip fast tracks, as those are handled directly by FastCapture 5479 if (activeTrack->isFastTrack()) { 5480 continue; 5481 } 5482 5483 enum { 5484 OVERRUN_UNKNOWN, 5485 OVERRUN_TRUE, 5486 OVERRUN_FALSE 5487 } overrun = OVERRUN_UNKNOWN; 5488 5489 // loop over getNextBuffer to handle circular sink 5490 for (;;) { 5491 5492 activeTrack->mSink.frameCount = ~0; 5493 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5494 size_t framesOut = activeTrack->mSink.frameCount; 5495 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5496 5497 int32_t front = activeTrack->mRsmpInFront; 5498 ssize_t filled = rear - front; 5499 size_t framesIn; 5500 5501 if (filled < 0) { 5502 // should not happen, but treat like a massive overrun and re-sync 5503 framesIn = 0; 5504 activeTrack->mRsmpInFront = rear; 5505 overrun = OVERRUN_TRUE; 5506 } else if ((size_t) filled <= mRsmpInFrames) { 5507 framesIn = (size_t) filled; 5508 } else { 5509 // client is not keeping up with server, but give it latest data 5510 framesIn = mRsmpInFrames; 5511 activeTrack->mRsmpInFront = front = rear - framesIn; 5512 overrun = OVERRUN_TRUE; 5513 } 5514 5515 if (framesOut == 0 || framesIn == 0) { 5516 break; 5517 } 5518 5519 if (activeTrack->mResampler == NULL) { 5520 // no resampling 5521 if (framesIn > framesOut) { 5522 framesIn = framesOut; 5523 } else { 5524 framesOut = framesIn; 5525 } 5526 int8_t *dst = activeTrack->mSink.i8; 5527 while (framesIn > 0) { 5528 front &= mRsmpInFramesP2 - 1; 5529 size_t part1 = mRsmpInFramesP2 - front; 5530 if (part1 > framesIn) { 5531 part1 = framesIn; 5532 } 5533 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5534 if (mChannelCount == activeTrack->mChannelCount) { 5535 memcpy(dst, src, part1 * mFrameSize); 5536 } else if (mChannelCount == 1) { 5537 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5538 part1); 5539 } else { 5540 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 5541 (const int16_t *)src, part1); 5542 } 5543 dst += part1 * activeTrack->mFrameSize; 5544 front += part1; 5545 framesIn -= part1; 5546 } 5547 activeTrack->mRsmpInFront += framesOut; 5548 5549 } else { 5550 // resampling 5551 // FIXME framesInNeeded should really be part of resampler API, and should 5552 // depend on the SRC ratio 5553 // to keep mRsmpInBuffer full so resampler always has sufficient input 5554 size_t framesInNeeded; 5555 // FIXME only re-calculate when it changes, and optimize for common ratios 5556 // Do not precompute in/out because floating point is not associative 5557 // e.g. a*b/c != a*(b/c). 5558 const double in(mSampleRate); 5559 const double out(activeTrack->mSampleRate); 5560 framesInNeeded = ceil(framesOut * in / out) + 1; 5561 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5562 framesInNeeded, framesOut, in / out); 5563 // Although we theoretically have framesIn in circular buffer, some of those are 5564 // unreleased frames, and thus must be discounted for purpose of budgeting. 5565 size_t unreleased = activeTrack->mRsmpInUnrel; 5566 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5567 if (framesIn < framesInNeeded) { 5568 ALOGV("not enough to resample: have %u frames in but need %u in to " 5569 "produce %u out given in/out ratio of %.4g", 5570 framesIn, framesInNeeded, framesOut, in / out); 5571 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5572 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5573 if (newFramesOut == 0) { 5574 break; 5575 } 5576 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5577 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5578 framesInNeeded, newFramesOut, out / in); 5579 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5580 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5581 "given in/out ratio of %.4g", 5582 framesIn, framesInNeeded, newFramesOut, in / out); 5583 framesOut = newFramesOut; 5584 } else { 5585 ALOGV("success 1: have %u in and need %u in to produce %u out " 5586 "given in/out ratio of %.4g", 5587 framesIn, framesInNeeded, framesOut, in / out); 5588 } 5589 5590 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5591 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5592 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5593 delete[] activeTrack->mRsmpOutBuffer; 5594 // resampler always outputs stereo 5595 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5596 activeTrack->mRsmpOutFrameCount = framesOut; 5597 } 5598 5599 // resampler accumulates, but we only have one source track 5600 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5601 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5602 // FIXME how about having activeTrack implement this interface itself? 5603 activeTrack->mResamplerBufferProvider 5604 /*this*/ /* AudioBufferProvider* */); 5605 // ditherAndClamp() works as long as all buffers returned by 5606 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5607 if (activeTrack->mChannelCount == 1) { 5608 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5609 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5610 framesOut); 5611 // the resampler always outputs stereo samples: 5612 // do post stereo to mono conversion 5613 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5614 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5615 } else { 5616 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5617 activeTrack->mRsmpOutBuffer, framesOut); 5618 } 5619 // now done with mRsmpOutBuffer 5620 5621 } 5622 5623 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5624 overrun = OVERRUN_FALSE; 5625 } 5626 5627 if (activeTrack->mFramesToDrop == 0) { 5628 if (framesOut > 0) { 5629 activeTrack->mSink.frameCount = framesOut; 5630 activeTrack->releaseBuffer(&activeTrack->mSink); 5631 } 5632 } else { 5633 // FIXME could do a partial drop of framesOut 5634 if (activeTrack->mFramesToDrop > 0) { 5635 activeTrack->mFramesToDrop -= framesOut; 5636 if (activeTrack->mFramesToDrop <= 0) { 5637 activeTrack->clearSyncStartEvent(); 5638 } 5639 } else { 5640 activeTrack->mFramesToDrop += framesOut; 5641 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5642 activeTrack->mSyncStartEvent->isCancelled()) { 5643 ALOGW("Synced record %s, session %d, trigger session %d", 5644 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5645 activeTrack->sessionId(), 5646 (activeTrack->mSyncStartEvent != 0) ? 5647 activeTrack->mSyncStartEvent->triggerSession() : 0); 5648 activeTrack->clearSyncStartEvent(); 5649 } 5650 } 5651 } 5652 5653 if (framesOut == 0) { 5654 break; 5655 } 5656 } 5657 5658 switch (overrun) { 5659 case OVERRUN_TRUE: 5660 // client isn't retrieving buffers fast enough 5661 if (!activeTrack->setOverflow()) { 5662 nsecs_t now = systemTime(); 5663 // FIXME should lastWarning per track? 5664 if ((now - lastWarning) > kWarningThrottleNs) { 5665 ALOGW("RecordThread: buffer overflow"); 5666 lastWarning = now; 5667 } 5668 } 5669 break; 5670 case OVERRUN_FALSE: 5671 activeTrack->clearOverflow(); 5672 break; 5673 case OVERRUN_UNKNOWN: 5674 break; 5675 } 5676 5677 } 5678 5679unlock: 5680 // enable changes in effect chain 5681 unlockEffectChains(effectChains); 5682 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5683 } 5684 5685 standbyIfNotAlreadyInStandby(); 5686 5687 { 5688 Mutex::Autolock _l(mLock); 5689 for (size_t i = 0; i < mTracks.size(); i++) { 5690 sp<RecordTrack> track = mTracks[i]; 5691 track->invalidate(); 5692 } 5693 mActiveTracks.clear(); 5694 mActiveTracksGen++; 5695 mStartStopCond.broadcast(); 5696 } 5697 5698 releaseWakeLock(); 5699 5700 ALOGV("RecordThread %p exiting", this); 5701 return false; 5702} 5703 5704void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5705{ 5706 if (!mStandby) { 5707 inputStandBy(); 5708 mStandby = true; 5709 } 5710} 5711 5712void AudioFlinger::RecordThread::inputStandBy() 5713{ 5714 // Idle the fast capture if it's currently running 5715 if (mFastCapture != 0) { 5716 FastCaptureStateQueue *sq = mFastCapture->sq(); 5717 FastCaptureState *state = sq->begin(); 5718 if (!(state->mCommand & FastCaptureState::IDLE)) { 5719 state->mCommand = FastCaptureState::COLD_IDLE; 5720 state->mColdFutexAddr = &mFastCaptureFutex; 5721 state->mColdGen++; 5722 mFastCaptureFutex = 0; 5723 sq->end(); 5724 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5725 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5726#if 0 5727 if (kUseFastCapture == FastCapture_Dynamic) { 5728 // FIXME 5729 } 5730#endif 5731#ifdef AUDIO_WATCHDOG 5732 // FIXME 5733#endif 5734 } else { 5735 sq->end(false /*didModify*/); 5736 } 5737 } 5738 mInput->stream->common.standby(&mInput->stream->common); 5739} 5740 5741// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5742sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5743 const sp<AudioFlinger::Client>& client, 5744 uint32_t sampleRate, 5745 audio_format_t format, 5746 audio_channel_mask_t channelMask, 5747 size_t *pFrameCount, 5748 int sessionId, 5749 size_t *notificationFrames, 5750 int uid, 5751 IAudioFlinger::track_flags_t *flags, 5752 pid_t tid, 5753 status_t *status) 5754{ 5755 size_t frameCount = *pFrameCount; 5756 sp<RecordTrack> track; 5757 status_t lStatus; 5758 5759 // client expresses a preference for FAST, but we get the final say 5760 if (*flags & IAudioFlinger::TRACK_FAST) { 5761 if ( 5762 // use case: callback handler 5763 (tid != -1) && 5764 // frame count is not specified, or is exactly the pipe depth 5765 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5766 // PCM data 5767 audio_is_linear_pcm(format) && 5768 // native format 5769 (format == mFormat) && 5770 // native channel mask 5771 (channelMask == mChannelMask) && 5772 // native hardware sample rate 5773 (sampleRate == mSampleRate) && 5774 // record thread has an associated fast capture 5775 hasFastCapture() && 5776 // there are sufficient fast track slots available 5777 mFastTrackAvail 5778 ) { 5779 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5780 frameCount, mFrameCount); 5781 } else { 5782 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5783 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5784 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5785 frameCount, mFrameCount, mPipeFramesP2, 5786 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5787 hasFastCapture(), tid, mFastTrackAvail); 5788 *flags &= ~IAudioFlinger::TRACK_FAST; 5789 } 5790 } 5791 5792 // compute track buffer size in frames, and suggest the notification frame count 5793 if (*flags & IAudioFlinger::TRACK_FAST) { 5794 // fast track: frame count is exactly the pipe depth 5795 frameCount = mPipeFramesP2; 5796 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5797 *notificationFrames = mFrameCount; 5798 } else { 5799 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5800 // or 20 ms if there is a fast capture 5801 // TODO This could be a roundupRatio inline, and const 5802 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5803 * sampleRate + mSampleRate - 1) / mSampleRate; 5804 // minimum number of notification periods is at least kMinNotifications, 5805 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5806 static const size_t kMinNotifications = 3; 5807 static const uint32_t kMinMs = 30; 5808 // TODO This could be a roundupRatio inline 5809 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5810 // TODO This could be a roundupRatio inline 5811 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5812 maxNotificationFrames; 5813 const size_t minFrameCount = maxNotificationFrames * 5814 max(kMinNotifications, minNotificationsByMs); 5815 frameCount = max(frameCount, minFrameCount); 5816 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5817 *notificationFrames = maxNotificationFrames; 5818 } 5819 } 5820 *pFrameCount = frameCount; 5821 5822 lStatus = initCheck(); 5823 if (lStatus != NO_ERROR) { 5824 ALOGE("createRecordTrack_l() audio driver not initialized"); 5825 goto Exit; 5826 } 5827 5828 { // scope for mLock 5829 Mutex::Autolock _l(mLock); 5830 5831 track = new RecordTrack(this, client, sampleRate, 5832 format, channelMask, frameCount, NULL, sessionId, uid, 5833 *flags, TrackBase::TYPE_DEFAULT); 5834 5835 lStatus = track->initCheck(); 5836 if (lStatus != NO_ERROR) { 5837 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5838 // track must be cleared from the caller as the caller has the AF lock 5839 goto Exit; 5840 } 5841 mTracks.add(track); 5842 5843 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5844 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5845 mAudioFlinger->btNrecIsOff(); 5846 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5847 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5848 5849 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5850 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5851 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5852 // so ask activity manager to do this on our behalf 5853 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5854 } 5855 } 5856 5857 lStatus = NO_ERROR; 5858 5859Exit: 5860 *status = lStatus; 5861 return track; 5862} 5863 5864status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5865 AudioSystem::sync_event_t event, 5866 int triggerSession) 5867{ 5868 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5869 sp<ThreadBase> strongMe = this; 5870 status_t status = NO_ERROR; 5871 5872 if (event == AudioSystem::SYNC_EVENT_NONE) { 5873 recordTrack->clearSyncStartEvent(); 5874 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5875 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5876 triggerSession, 5877 recordTrack->sessionId(), 5878 syncStartEventCallback, 5879 recordTrack); 5880 // Sync event can be cancelled by the trigger session if the track is not in a 5881 // compatible state in which case we start record immediately 5882 if (recordTrack->mSyncStartEvent->isCancelled()) { 5883 recordTrack->clearSyncStartEvent(); 5884 } else { 5885 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5886 recordTrack->mFramesToDrop = - 5887 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5888 } 5889 } 5890 5891 { 5892 // This section is a rendezvous between binder thread executing start() and RecordThread 5893 AutoMutex lock(mLock); 5894 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5895 if (recordTrack->mState == TrackBase::PAUSING) { 5896 ALOGV("active record track PAUSING -> ACTIVE"); 5897 recordTrack->mState = TrackBase::ACTIVE; 5898 } else { 5899 ALOGV("active record track state %d", recordTrack->mState); 5900 } 5901 return status; 5902 } 5903 5904 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5905 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5906 // or using a separate command thread 5907 recordTrack->mState = TrackBase::STARTING_1; 5908 mActiveTracks.add(recordTrack); 5909 mActiveTracksGen++; 5910 status_t status = NO_ERROR; 5911 if (recordTrack->isExternalTrack()) { 5912 mLock.unlock(); 5913 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5914 mLock.lock(); 5915 // FIXME should verify that recordTrack is still in mActiveTracks 5916 if (status != NO_ERROR) { 5917 mActiveTracks.remove(recordTrack); 5918 mActiveTracksGen++; 5919 recordTrack->clearSyncStartEvent(); 5920 ALOGV("RecordThread::start error %d", status); 5921 return status; 5922 } 5923 } 5924 // Catch up with current buffer indices if thread is already running. 5925 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5926 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5927 // see previously buffered data before it called start(), but with greater risk of overrun. 5928 5929 recordTrack->mRsmpInFront = mRsmpInRear; 5930 recordTrack->mRsmpInUnrel = 0; 5931 // FIXME why reset? 5932 if (recordTrack->mResampler != NULL) { 5933 recordTrack->mResampler->reset(); 5934 } 5935 recordTrack->mState = TrackBase::STARTING_2; 5936 // signal thread to start 5937 mWaitWorkCV.broadcast(); 5938 if (mActiveTracks.indexOf(recordTrack) < 0) { 5939 ALOGV("Record failed to start"); 5940 status = BAD_VALUE; 5941 goto startError; 5942 } 5943 return status; 5944 } 5945 5946startError: 5947 if (recordTrack->isExternalTrack()) { 5948 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5949 } 5950 recordTrack->clearSyncStartEvent(); 5951 // FIXME I wonder why we do not reset the state here? 5952 return status; 5953} 5954 5955void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5956{ 5957 sp<SyncEvent> strongEvent = event.promote(); 5958 5959 if (strongEvent != 0) { 5960 sp<RefBase> ptr = strongEvent->cookie().promote(); 5961 if (ptr != 0) { 5962 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5963 recordTrack->handleSyncStartEvent(strongEvent); 5964 } 5965 } 5966} 5967 5968bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5969 ALOGV("RecordThread::stop"); 5970 AutoMutex _l(mLock); 5971 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5972 return false; 5973 } 5974 // note that threadLoop may still be processing the track at this point [without lock] 5975 recordTrack->mState = TrackBase::PAUSING; 5976 // do not wait for mStartStopCond if exiting 5977 if (exitPending()) { 5978 return true; 5979 } 5980 // FIXME incorrect usage of wait: no explicit predicate or loop 5981 mStartStopCond.wait(mLock); 5982 // if we have been restarted, recordTrack is in mActiveTracks here 5983 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5984 ALOGV("Record stopped OK"); 5985 return true; 5986 } 5987 return false; 5988} 5989 5990bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5991{ 5992 return false; 5993} 5994 5995status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5996{ 5997#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5998 if (!isValidSyncEvent(event)) { 5999 return BAD_VALUE; 6000 } 6001 6002 int eventSession = event->triggerSession(); 6003 status_t ret = NAME_NOT_FOUND; 6004 6005 Mutex::Autolock _l(mLock); 6006 6007 for (size_t i = 0; i < mTracks.size(); i++) { 6008 sp<RecordTrack> track = mTracks[i]; 6009 if (eventSession == track->sessionId()) { 6010 (void) track->setSyncEvent(event); 6011 ret = NO_ERROR; 6012 } 6013 } 6014 return ret; 6015#else 6016 return BAD_VALUE; 6017#endif 6018} 6019 6020// destroyTrack_l() must be called with ThreadBase::mLock held 6021void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6022{ 6023 track->terminate(); 6024 track->mState = TrackBase::STOPPED; 6025 // active tracks are removed by threadLoop() 6026 if (mActiveTracks.indexOf(track) < 0) { 6027 removeTrack_l(track); 6028 } 6029} 6030 6031void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6032{ 6033 mTracks.remove(track); 6034 // need anything related to effects here? 6035 if (track->isFastTrack()) { 6036 ALOG_ASSERT(!mFastTrackAvail); 6037 mFastTrackAvail = true; 6038 } 6039} 6040 6041void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6042{ 6043 dumpInternals(fd, args); 6044 dumpTracks(fd, args); 6045 dumpEffectChains(fd, args); 6046} 6047 6048void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6049{ 6050 dprintf(fd, "\nInput thread %p:\n", this); 6051 6052 if (mActiveTracks.size() > 0) { 6053 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 6054 } else { 6055 dprintf(fd, " No active record clients\n"); 6056 } 6057 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6058 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6059 6060 dumpBase(fd, args); 6061} 6062 6063void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6064{ 6065 const size_t SIZE = 256; 6066 char buffer[SIZE]; 6067 String8 result; 6068 6069 size_t numtracks = mTracks.size(); 6070 size_t numactive = mActiveTracks.size(); 6071 size_t numactiveseen = 0; 6072 dprintf(fd, " %d Tracks", numtracks); 6073 if (numtracks) { 6074 dprintf(fd, " of which %d are active\n", numactive); 6075 RecordTrack::appendDumpHeader(result); 6076 for (size_t i = 0; i < numtracks ; ++i) { 6077 sp<RecordTrack> track = mTracks[i]; 6078 if (track != 0) { 6079 bool active = mActiveTracks.indexOf(track) >= 0; 6080 if (active) { 6081 numactiveseen++; 6082 } 6083 track->dump(buffer, SIZE, active); 6084 result.append(buffer); 6085 } 6086 } 6087 } else { 6088 dprintf(fd, "\n"); 6089 } 6090 6091 if (numactiveseen != numactive) { 6092 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6093 " not in the track list\n"); 6094 result.append(buffer); 6095 RecordTrack::appendDumpHeader(result); 6096 for (size_t i = 0; i < numactive; ++i) { 6097 sp<RecordTrack> track = mActiveTracks[i]; 6098 if (mTracks.indexOf(track) < 0) { 6099 track->dump(buffer, SIZE, true); 6100 result.append(buffer); 6101 } 6102 } 6103 6104 } 6105 write(fd, result.string(), result.size()); 6106} 6107 6108// AudioBufferProvider interface 6109status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6110 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6111{ 6112 RecordTrack *activeTrack = mRecordTrack; 6113 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 6114 if (threadBase == 0) { 6115 buffer->frameCount = 0; 6116 buffer->raw = NULL; 6117 return NOT_ENOUGH_DATA; 6118 } 6119 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6120 int32_t rear = recordThread->mRsmpInRear; 6121 int32_t front = activeTrack->mRsmpInFront; 6122 ssize_t filled = rear - front; 6123 // FIXME should not be P2 (don't want to increase latency) 6124 // FIXME if client not keeping up, discard 6125 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6126 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6127 front &= recordThread->mRsmpInFramesP2 - 1; 6128 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6129 if (part1 > (size_t) filled) { 6130 part1 = filled; 6131 } 6132 size_t ask = buffer->frameCount; 6133 ALOG_ASSERT(ask > 0); 6134 if (part1 > ask) { 6135 part1 = ask; 6136 } 6137 if (part1 == 0) { 6138 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 6139 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 6140 buffer->raw = NULL; 6141 buffer->frameCount = 0; 6142 activeTrack->mRsmpInUnrel = 0; 6143 return NOT_ENOUGH_DATA; 6144 } 6145 6146 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6147 buffer->frameCount = part1; 6148 activeTrack->mRsmpInUnrel = part1; 6149 return NO_ERROR; 6150} 6151 6152// AudioBufferProvider interface 6153void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6154 AudioBufferProvider::Buffer* buffer) 6155{ 6156 RecordTrack *activeTrack = mRecordTrack; 6157 size_t stepCount = buffer->frameCount; 6158 if (stepCount == 0) { 6159 return; 6160 } 6161 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 6162 activeTrack->mRsmpInUnrel -= stepCount; 6163 activeTrack->mRsmpInFront += stepCount; 6164 buffer->raw = NULL; 6165 buffer->frameCount = 0; 6166} 6167 6168bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6169 status_t& status) 6170{ 6171 bool reconfig = false; 6172 6173 status = NO_ERROR; 6174 6175 audio_format_t reqFormat = mFormat; 6176 uint32_t samplingRate = mSampleRate; 6177 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6178 6179 AudioParameter param = AudioParameter(keyValuePair); 6180 int value; 6181 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6182 // channel count change can be requested. Do we mandate the first client defines the 6183 // HAL sampling rate and channel count or do we allow changes on the fly? 6184 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6185 samplingRate = value; 6186 reconfig = true; 6187 } 6188 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6189 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 6190 status = BAD_VALUE; 6191 } else { 6192 reqFormat = (audio_format_t) value; 6193 reconfig = true; 6194 } 6195 } 6196 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6197 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6198 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6199 status = BAD_VALUE; 6200 } else { 6201 channelMask = mask; 6202 reconfig = true; 6203 } 6204 } 6205 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6206 // do not accept frame count changes if tracks are open as the track buffer 6207 // size depends on frame count and correct behavior would not be guaranteed 6208 // if frame count is changed after track creation 6209 if (mActiveTracks.size() > 0) { 6210 status = INVALID_OPERATION; 6211 } else { 6212 reconfig = true; 6213 } 6214 } 6215 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6216 // forward device change to effects that have requested to be 6217 // aware of attached audio device. 6218 for (size_t i = 0; i < mEffectChains.size(); i++) { 6219 mEffectChains[i]->setDevice_l(value); 6220 } 6221 6222 // store input device and output device but do not forward output device to audio HAL. 6223 // Note that status is ignored by the caller for output device 6224 // (see AudioFlinger::setParameters() 6225 if (audio_is_output_devices(value)) { 6226 mOutDevice = value; 6227 status = BAD_VALUE; 6228 } else { 6229 mInDevice = value; 6230 // disable AEC and NS if the device is a BT SCO headset supporting those 6231 // pre processings 6232 if (mTracks.size() > 0) { 6233 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6234 mAudioFlinger->btNrecIsOff(); 6235 for (size_t i = 0; i < mTracks.size(); i++) { 6236 sp<RecordTrack> track = mTracks[i]; 6237 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6238 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6239 } 6240 } 6241 } 6242 } 6243 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6244 mAudioSource != (audio_source_t)value) { 6245 // forward device change to effects that have requested to be 6246 // aware of attached audio device. 6247 for (size_t i = 0; i < mEffectChains.size(); i++) { 6248 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6249 } 6250 mAudioSource = (audio_source_t)value; 6251 } 6252 6253 if (status == NO_ERROR) { 6254 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6255 keyValuePair.string()); 6256 if (status == INVALID_OPERATION) { 6257 inputStandBy(); 6258 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6259 keyValuePair.string()); 6260 } 6261 if (reconfig) { 6262 if (status == BAD_VALUE && 6263 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6264 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6265 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6266 <= (2 * samplingRate)) && 6267 audio_channel_count_from_in_mask( 6268 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6269 (channelMask == AUDIO_CHANNEL_IN_MONO || 6270 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6271 status = NO_ERROR; 6272 } 6273 if (status == NO_ERROR) { 6274 readInputParameters_l(); 6275 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6276 } 6277 } 6278 } 6279 6280 return reconfig; 6281} 6282 6283String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6284{ 6285 Mutex::Autolock _l(mLock); 6286 if (initCheck() != NO_ERROR) { 6287 return String8(); 6288 } 6289 6290 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6291 const String8 out_s8(s); 6292 free(s); 6293 return out_s8; 6294} 6295 6296void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6297 AudioSystem::OutputDescriptor desc; 6298 const void *param2 = NULL; 6299 6300 switch (event) { 6301 case AudioSystem::INPUT_OPENED: 6302 case AudioSystem::INPUT_CONFIG_CHANGED: 6303 desc.channelMask = mChannelMask; 6304 desc.samplingRate = mSampleRate; 6305 desc.format = mFormat; 6306 desc.frameCount = mFrameCount; 6307 desc.latency = 0; 6308 param2 = &desc; 6309 break; 6310 6311 case AudioSystem::INPUT_CLOSED: 6312 default: 6313 break; 6314 } 6315 mAudioFlinger->audioConfigChanged(event, mId, param2); 6316} 6317 6318void AudioFlinger::RecordThread::readInputParameters_l() 6319{ 6320 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6321 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6322 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6323 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6324 mFormat = mHALFormat; 6325 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6326 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6327 } 6328 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6329 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6330 mFrameCount = mBufferSize / mFrameSize; 6331 // This is the formula for calculating the temporary buffer size. 6332 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6333 // 1 full output buffer, regardless of the alignment of the available input. 6334 // The value is somewhat arbitrary, and could probably be even larger. 6335 // A larger value should allow more old data to be read after a track calls start(), 6336 // without increasing latency. 6337 mRsmpInFrames = mFrameCount * 7; 6338 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6339 delete[] mRsmpInBuffer; 6340 6341 // TODO optimize audio capture buffer sizes ... 6342 // Here we calculate the size of the sliding buffer used as a source 6343 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6344 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6345 // be better to have it derived from the pipe depth in the long term. 6346 // The current value is higher than necessary. However it should not add to latency. 6347 6348 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6349 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6350 6351 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6352 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6353} 6354 6355uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6356{ 6357 Mutex::Autolock _l(mLock); 6358 if (initCheck() != NO_ERROR) { 6359 return 0; 6360 } 6361 6362 return mInput->stream->get_input_frames_lost(mInput->stream); 6363} 6364 6365uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6366{ 6367 Mutex::Autolock _l(mLock); 6368 uint32_t result = 0; 6369 if (getEffectChain_l(sessionId) != 0) { 6370 result = EFFECT_SESSION; 6371 } 6372 6373 for (size_t i = 0; i < mTracks.size(); ++i) { 6374 if (sessionId == mTracks[i]->sessionId()) { 6375 result |= TRACK_SESSION; 6376 break; 6377 } 6378 } 6379 6380 return result; 6381} 6382 6383KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6384{ 6385 KeyedVector<int, bool> ids; 6386 Mutex::Autolock _l(mLock); 6387 for (size_t j = 0; j < mTracks.size(); ++j) { 6388 sp<RecordThread::RecordTrack> track = mTracks[j]; 6389 int sessionId = track->sessionId(); 6390 if (ids.indexOfKey(sessionId) < 0) { 6391 ids.add(sessionId, true); 6392 } 6393 } 6394 return ids; 6395} 6396 6397AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6398{ 6399 Mutex::Autolock _l(mLock); 6400 AudioStreamIn *input = mInput; 6401 mInput = NULL; 6402 return input; 6403} 6404 6405// this method must always be called either with ThreadBase mLock held or inside the thread loop 6406audio_stream_t* AudioFlinger::RecordThread::stream() const 6407{ 6408 if (mInput == NULL) { 6409 return NULL; 6410 } 6411 return &mInput->stream->common; 6412} 6413 6414status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6415{ 6416 // only one chain per input thread 6417 if (mEffectChains.size() != 0) { 6418 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6419 return INVALID_OPERATION; 6420 } 6421 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6422 chain->setThread(this); 6423 chain->setInBuffer(NULL); 6424 chain->setOutBuffer(NULL); 6425 6426 checkSuspendOnAddEffectChain_l(chain); 6427 6428 // make sure enabled pre processing effects state is communicated to the HAL as we 6429 // just moved them to a new input stream. 6430 chain->syncHalEffectsState(); 6431 6432 mEffectChains.add(chain); 6433 6434 return NO_ERROR; 6435} 6436 6437size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6438{ 6439 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6440 ALOGW_IF(mEffectChains.size() != 1, 6441 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6442 chain.get(), mEffectChains.size(), this); 6443 if (mEffectChains.size() == 1) { 6444 mEffectChains.removeAt(0); 6445 } 6446 return 0; 6447} 6448 6449status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6450 audio_patch_handle_t *handle) 6451{ 6452 status_t status = NO_ERROR; 6453 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6454 // store new device and send to effects 6455 mInDevice = patch->sources[0].ext.device.type; 6456 for (size_t i = 0; i < mEffectChains.size(); i++) { 6457 mEffectChains[i]->setDevice_l(mInDevice); 6458 } 6459 6460 // disable AEC and NS if the device is a BT SCO headset supporting those 6461 // pre processings 6462 if (mTracks.size() > 0) { 6463 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6464 mAudioFlinger->btNrecIsOff(); 6465 for (size_t i = 0; i < mTracks.size(); i++) { 6466 sp<RecordTrack> track = mTracks[i]; 6467 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6468 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6469 } 6470 } 6471 6472 // store new source and send to effects 6473 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6474 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6475 for (size_t i = 0; i < mEffectChains.size(); i++) { 6476 mEffectChains[i]->setAudioSource_l(mAudioSource); 6477 } 6478 } 6479 6480 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6481 status = hwDevice->create_audio_patch(hwDevice, 6482 patch->num_sources, 6483 patch->sources, 6484 patch->num_sinks, 6485 patch->sinks, 6486 handle); 6487 } else { 6488 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6489 } 6490 return status; 6491} 6492 6493status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6494{ 6495 status_t status = NO_ERROR; 6496 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6497 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6498 status = hwDevice->release_audio_patch(hwDevice, handle); 6499 } else { 6500 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6501 } 6502 return status; 6503} 6504 6505void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6506{ 6507 Mutex::Autolock _l(mLock); 6508 mTracks.add(record); 6509} 6510 6511void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6512{ 6513 Mutex::Autolock _l(mLock); 6514 destroyTrack_l(record); 6515} 6516 6517void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6518{ 6519 ThreadBase::getAudioPortConfig(config); 6520 config->role = AUDIO_PORT_ROLE_SINK; 6521 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6522 config->ext.mix.usecase.source = mAudioSource; 6523} 6524 6525} // namespace android 6526