Threads.cpp revision 40eb1a1f8871909c272e72afaf7d5af84fea2412
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/primitives.h> 40#include <audio_utils/format.h> 41#include <audio_utils/minifloat.h> 42 43// NBAIO implementations 44#include <media/nbaio/AudioStreamInSource.h> 45#include <media/nbaio/AudioStreamOutSink.h> 46#include <media/nbaio/MonoPipe.h> 47#include <media/nbaio/MonoPipeReader.h> 48#include <media/nbaio/Pipe.h> 49#include <media/nbaio/PipeReader.h> 50#include <media/nbaio/SourceAudioBufferProvider.h> 51 52#include <powermanager/PowerManager.h> 53 54#include <common_time/cc_helper.h> 55#include <common_time/local_clock.h> 56 57#include "AudioFlinger.h" 58#include "AudioMixer.h" 59#include "BufferProviders.h" 60#include "FastMixer.h" 61#include "FastCapture.h" 62#include "ServiceUtilities.h" 63#include "SchedulingPolicyService.h" 64 65#ifdef ADD_BATTERY_DATA 66#include <media/IMediaPlayerService.h> 67#include <media/IMediaDeathNotifier.h> 68#endif 69 70#ifdef DEBUG_CPU_USAGE 71#include <cpustats/CentralTendencyStatistics.h> 72#include <cpustats/ThreadCpuUsage.h> 73#endif 74 75// ---------------------------------------------------------------------------- 76 77// Note: the following macro is used for extremely verbose logging message. In 78// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 79// 0; but one side effect of this is to turn all LOGV's as well. Some messages 80// are so verbose that we want to suppress them even when we have ALOG_ASSERT 81// turned on. Do not uncomment the #def below unless you really know what you 82// are doing and want to see all of the extremely verbose messages. 83//#define VERY_VERY_VERBOSE_LOGGING 84#ifdef VERY_VERY_VERBOSE_LOGGING 85#define ALOGVV ALOGV 86#else 87#define ALOGVV(a...) do { } while(0) 88#endif 89 90// TODO: Move these macro/inlines to a header file. 91#define max(a, b) ((a) > (b) ? (a) : (b)) 92template <typename T> 93static inline T min(const T& a, const T& b) 94{ 95 return a < b ? a : b; 96} 97 98#ifndef ARRAY_SIZE 99#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 100#endif 101 102namespace android { 103 104// retry counts for buffer fill timeout 105// 50 * ~20msecs = 1 second 106static const int8_t kMaxTrackRetries = 50; 107static const int8_t kMaxTrackStartupRetries = 50; 108// allow less retry attempts on direct output thread. 109// direct outputs can be a scarce resource in audio hardware and should 110// be released as quickly as possible. 111static const int8_t kMaxTrackRetriesDirect = 2; 112 113// don't warn about blocked writes or record buffer overflows more often than this 114static const nsecs_t kWarningThrottleNs = seconds(5); 115 116// RecordThread loop sleep time upon application overrun or audio HAL read error 117static const int kRecordThreadSleepUs = 5000; 118 119// maximum time to wait in sendConfigEvent_l() for a status to be received 120static const nsecs_t kConfigEventTimeoutNs = seconds(2); 121 122// minimum sleep time for the mixer thread loop when tracks are active but in underrun 123static const uint32_t kMinThreadSleepTimeUs = 5000; 124// maximum divider applied to the active sleep time in the mixer thread loop 125static const uint32_t kMaxThreadSleepTimeShift = 2; 126 127// minimum normal sink buffer size, expressed in milliseconds rather than frames 128static const uint32_t kMinNormalSinkBufferSizeMs = 20; 129// maximum normal sink buffer size 130static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 131 132// Offloaded output thread standby delay: allows track transition without going to standby 133static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 134 135// Whether to use fast mixer 136static const enum { 137 FastMixer_Never, // never initialize or use: for debugging only 138 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 139 // normal mixer multiplier is 1 140 FastMixer_Static, // initialize if needed, then use all the time if initialized, 141 // multiplier is calculated based on min & max normal mixer buffer size 142 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 143 // multiplier is calculated based on min & max normal mixer buffer size 144 // FIXME for FastMixer_Dynamic: 145 // Supporting this option will require fixing HALs that can't handle large writes. 146 // For example, one HAL implementation returns an error from a large write, 147 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 148 // We could either fix the HAL implementations, or provide a wrapper that breaks 149 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 150} kUseFastMixer = FastMixer_Static; 151 152// Whether to use fast capture 153static const enum { 154 FastCapture_Never, // never initialize or use: for debugging only 155 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 156 FastCapture_Static, // initialize if needed, then use all the time if initialized 157} kUseFastCapture = FastCapture_Static; 158 159// Priorities for requestPriority 160static const int kPriorityAudioApp = 2; 161static const int kPriorityFastMixer = 3; 162static const int kPriorityFastCapture = 3; 163 164// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 165// for the track. The client then sub-divides this into smaller buffers for its use. 166// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 167// So for now we just assume that client is double-buffered for fast tracks. 168// FIXME It would be better for client to tell AudioFlinger the value of N, 169// so AudioFlinger could allocate the right amount of memory. 170// See the client's minBufCount and mNotificationFramesAct calculations for details. 171 172// This is the default value, if not specified by property. 173static const int kFastTrackMultiplier = 2; 174 175// The minimum and maximum allowed values 176static const int kFastTrackMultiplierMin = 1; 177static const int kFastTrackMultiplierMax = 2; 178 179// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 180static int sFastTrackMultiplier = kFastTrackMultiplier; 181 182// See Thread::readOnlyHeap(). 183// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 184// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 185// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 186static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 187 188// ---------------------------------------------------------------------------- 189 190static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 191 192static void sFastTrackMultiplierInit() 193{ 194 char value[PROPERTY_VALUE_MAX]; 195 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 196 char *endptr; 197 unsigned long ul = strtoul(value, &endptr, 0); 198 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 199 sFastTrackMultiplier = (int) ul; 200 } 201 } 202} 203 204// ---------------------------------------------------------------------------- 205 206#ifdef ADD_BATTERY_DATA 207// To collect the amplifier usage 208static void addBatteryData(uint32_t params) { 209 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 210 if (service == NULL) { 211 // it already logged 212 return; 213 } 214 215 service->addBatteryData(params); 216} 217#endif 218 219 220// ---------------------------------------------------------------------------- 221// CPU Stats 222// ---------------------------------------------------------------------------- 223 224class CpuStats { 225public: 226 CpuStats(); 227 void sample(const String8 &title); 228#ifdef DEBUG_CPU_USAGE 229private: 230 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 231 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 232 233 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 234 235 int mCpuNum; // thread's current CPU number 236 int mCpukHz; // frequency of thread's current CPU in kHz 237#endif 238}; 239 240CpuStats::CpuStats() 241#ifdef DEBUG_CPU_USAGE 242 : mCpuNum(-1), mCpukHz(-1) 243#endif 244{ 245} 246 247void CpuStats::sample(const String8 &title 248#ifndef DEBUG_CPU_USAGE 249 __unused 250#endif 251 ) { 252#ifdef DEBUG_CPU_USAGE 253 // get current thread's delta CPU time in wall clock ns 254 double wcNs; 255 bool valid = mCpuUsage.sampleAndEnable(wcNs); 256 257 // record sample for wall clock statistics 258 if (valid) { 259 mWcStats.sample(wcNs); 260 } 261 262 // get the current CPU number 263 int cpuNum = sched_getcpu(); 264 265 // get the current CPU frequency in kHz 266 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 267 268 // check if either CPU number or frequency changed 269 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 270 mCpuNum = cpuNum; 271 mCpukHz = cpukHz; 272 // ignore sample for purposes of cycles 273 valid = false; 274 } 275 276 // if no change in CPU number or frequency, then record sample for cycle statistics 277 if (valid && mCpukHz > 0) { 278 double cycles = wcNs * cpukHz * 0.000001; 279 mHzStats.sample(cycles); 280 } 281 282 unsigned n = mWcStats.n(); 283 // mCpuUsage.elapsed() is expensive, so don't call it every loop 284 if ((n & 127) == 1) { 285 long long elapsed = mCpuUsage.elapsed(); 286 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 287 double perLoop = elapsed / (double) n; 288 double perLoop100 = perLoop * 0.01; 289 double perLoop1k = perLoop * 0.001; 290 double mean = mWcStats.mean(); 291 double stddev = mWcStats.stddev(); 292 double minimum = mWcStats.minimum(); 293 double maximum = mWcStats.maximum(); 294 double meanCycles = mHzStats.mean(); 295 double stddevCycles = mHzStats.stddev(); 296 double minCycles = mHzStats.minimum(); 297 double maxCycles = mHzStats.maximum(); 298 mCpuUsage.resetElapsed(); 299 mWcStats.reset(); 300 mHzStats.reset(); 301 ALOGD("CPU usage for %s over past %.1f secs\n" 302 " (%u mixer loops at %.1f mean ms per loop):\n" 303 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 304 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 305 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 306 title.string(), 307 elapsed * .000000001, n, perLoop * .000001, 308 mean * .001, 309 stddev * .001, 310 minimum * .001, 311 maximum * .001, 312 mean / perLoop100, 313 stddev / perLoop100, 314 minimum / perLoop100, 315 maximum / perLoop100, 316 meanCycles / perLoop1k, 317 stddevCycles / perLoop1k, 318 minCycles / perLoop1k, 319 maxCycles / perLoop1k); 320 321 } 322 } 323#endif 324}; 325 326// ---------------------------------------------------------------------------- 327// ThreadBase 328// ---------------------------------------------------------------------------- 329 330// static 331const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 332{ 333 switch (type) { 334 case MIXER: 335 return "MIXER"; 336 case DIRECT: 337 return "DIRECT"; 338 case DUPLICATING: 339 return "DUPLICATING"; 340 case RECORD: 341 return "RECORD"; 342 case OFFLOAD: 343 return "OFFLOAD"; 344 default: 345 return "unknown"; 346 } 347} 348 349String8 devicesToString(audio_devices_t devices) 350{ 351 static const struct mapping { 352 audio_devices_t mDevices; 353 const char * mString; 354 } mappingsOut[] = { 355 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 356 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 357 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 358 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 359 AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO", 360 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 361 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT", 362 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 363 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES, "BLUETOOTH_A2DP_HEADPHONES", 364 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER", 365 AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL", 366 AUDIO_DEVICE_OUT_HDMI, "HDMI", 367 AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 368 AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 369 AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY", 370 AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE", 371 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 372 AUDIO_DEVICE_OUT_LINE, "LINE", 373 AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC", 374 AUDIO_DEVICE_OUT_SPDIF, "SPDIF", 375 AUDIO_DEVICE_OUT_FM, "FM", 376 AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE", 377 AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE", 378 AUDIO_DEVICE_NONE, "NONE", // must be last 379 }, mappingsIn[] = { 380 AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION", 381 AUDIO_DEVICE_IN_AMBIENT, "AMBIENT", 382 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 383 AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 384 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 385 AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL", 386 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 387 AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX", 388 AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC", 389 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 390 AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 391 AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 392 AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY", 393 AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE", 394 AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER", 395 AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER", 396 AUDIO_DEVICE_IN_LINE, "LINE", 397 AUDIO_DEVICE_IN_SPDIF, "SPDIF", 398 AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 399 AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK", 400 AUDIO_DEVICE_NONE, "NONE", // must be last 401 }; 402 String8 result; 403 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 404 const mapping *entry; 405 if (devices & AUDIO_DEVICE_BIT_IN) { 406 devices &= ~AUDIO_DEVICE_BIT_IN; 407 entry = mappingsIn; 408 } else { 409 entry = mappingsOut; 410 } 411 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 412 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 413 if (devices & entry->mDevices) { 414 if (!result.isEmpty()) { 415 result.append("|"); 416 } 417 result.append(entry->mString); 418 } 419 } 420 if (devices & ~allDevices) { 421 if (!result.isEmpty()) { 422 result.append("|"); 423 } 424 result.appendFormat("0x%X", devices & ~allDevices); 425 } 426 if (result.isEmpty()) { 427 result.append(entry->mString); 428 } 429 return result; 430} 431 432String8 inputFlagsToString(audio_input_flags_t flags) 433{ 434 static const struct mapping { 435 audio_input_flags_t mFlag; 436 const char * mString; 437 } mappings[] = { 438 AUDIO_INPUT_FLAG_FAST, "FAST", 439 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 440 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 441 }; 442 String8 result; 443 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 444 const mapping *entry; 445 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 446 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 447 if (flags & entry->mFlag) { 448 if (!result.isEmpty()) { 449 result.append("|"); 450 } 451 result.append(entry->mString); 452 } 453 } 454 if (flags & ~allFlags) { 455 if (!result.isEmpty()) { 456 result.append("|"); 457 } 458 result.appendFormat("0x%X", flags & ~allFlags); 459 } 460 if (result.isEmpty()) { 461 result.append(entry->mString); 462 } 463 return result; 464} 465 466String8 outputFlagsToString(audio_output_flags_t flags) 467{ 468 static const struct mapping { 469 audio_output_flags_t mFlag; 470 const char * mString; 471 } mappings[] = { 472 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 473 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 474 AUDIO_OUTPUT_FLAG_FAST, "FAST", 475 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 476 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 477 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 478 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 479 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 480 }; 481 String8 result; 482 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 483 const mapping *entry; 484 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 485 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 486 if (flags & entry->mFlag) { 487 if (!result.isEmpty()) { 488 result.append("|"); 489 } 490 result.append(entry->mString); 491 } 492 } 493 if (flags & ~allFlags) { 494 if (!result.isEmpty()) { 495 result.append("|"); 496 } 497 result.appendFormat("0x%X", flags & ~allFlags); 498 } 499 if (result.isEmpty()) { 500 result.append(entry->mString); 501 } 502 return result; 503} 504 505const char *sourceToString(audio_source_t source) 506{ 507 switch (source) { 508 case AUDIO_SOURCE_DEFAULT: return "default"; 509 case AUDIO_SOURCE_MIC: return "mic"; 510 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 511 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 512 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 513 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 514 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 515 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 516 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 517 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 518 case AUDIO_SOURCE_HOTWORD: return "hotword"; 519 default: return "unknown"; 520 } 521} 522 523AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 524 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 525 : Thread(false /*canCallJava*/), 526 mType(type), 527 mAudioFlinger(audioFlinger), 528 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 529 // are set by PlaybackThread::readOutputParameters_l() or 530 // RecordThread::readInputParameters_l() 531 //FIXME: mStandby should be true here. Is this some kind of hack? 532 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 533 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 534 // mName will be set by concrete (non-virtual) subclass 535 mDeathRecipient(new PMDeathRecipient(this)), 536 mSystemReady(systemReady) 537{ 538 memset(&mPatch, 0, sizeof(struct audio_patch)); 539} 540 541AudioFlinger::ThreadBase::~ThreadBase() 542{ 543 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 544 mConfigEvents.clear(); 545 546 // do not lock the mutex in destructor 547 releaseWakeLock_l(); 548 if (mPowerManager != 0) { 549 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 550 binder->unlinkToDeath(mDeathRecipient); 551 } 552} 553 554status_t AudioFlinger::ThreadBase::readyToRun() 555{ 556 status_t status = initCheck(); 557 if (status == NO_ERROR) { 558 ALOGI("AudioFlinger's thread %p ready to run", this); 559 } else { 560 ALOGE("No working audio driver found."); 561 } 562 return status; 563} 564 565void AudioFlinger::ThreadBase::exit() 566{ 567 ALOGV("ThreadBase::exit"); 568 // do any cleanup required for exit to succeed 569 preExit(); 570 { 571 // This lock prevents the following race in thread (uniprocessor for illustration): 572 // if (!exitPending()) { 573 // // context switch from here to exit() 574 // // exit() calls requestExit(), what exitPending() observes 575 // // exit() calls signal(), which is dropped since no waiters 576 // // context switch back from exit() to here 577 // mWaitWorkCV.wait(...); 578 // // now thread is hung 579 // } 580 AutoMutex lock(mLock); 581 requestExit(); 582 mWaitWorkCV.broadcast(); 583 } 584 // When Thread::requestExitAndWait is made virtual and this method is renamed to 585 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 586 requestExitAndWait(); 587} 588 589status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 590{ 591 status_t status; 592 593 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 594 Mutex::Autolock _l(mLock); 595 596 return sendSetParameterConfigEvent_l(keyValuePairs); 597} 598 599// sendConfigEvent_l() must be called with ThreadBase::mLock held 600// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 601status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 602{ 603 status_t status = NO_ERROR; 604 605 if (event->mRequiresSystemReady && !mSystemReady) { 606 event->mWaitStatus = false; 607 mPendingConfigEvents.add(event); 608 return status; 609 } 610 mConfigEvents.add(event); 611 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 612 mWaitWorkCV.signal(); 613 mLock.unlock(); 614 { 615 Mutex::Autolock _l(event->mLock); 616 while (event->mWaitStatus) { 617 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 618 event->mStatus = TIMED_OUT; 619 event->mWaitStatus = false; 620 } 621 } 622 status = event->mStatus; 623 } 624 mLock.lock(); 625 return status; 626} 627 628void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event) 629{ 630 Mutex::Autolock _l(mLock); 631 sendIoConfigEvent_l(event); 632} 633 634// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 635void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event) 636{ 637 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event); 638 sendConfigEvent_l(configEvent); 639} 640 641void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 642{ 643 Mutex::Autolock _l(mLock); 644 sendPrioConfigEvent_l(pid, tid, prio); 645} 646 647// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 648void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 649{ 650 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 651 sendConfigEvent_l(configEvent); 652} 653 654// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 655status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 656{ 657 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 658 return sendConfigEvent_l(configEvent); 659} 660 661status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 662 const struct audio_patch *patch, 663 audio_patch_handle_t *handle) 664{ 665 Mutex::Autolock _l(mLock); 666 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 667 status_t status = sendConfigEvent_l(configEvent); 668 if (status == NO_ERROR) { 669 CreateAudioPatchConfigEventData *data = 670 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 671 *handle = data->mHandle; 672 } 673 return status; 674} 675 676status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 677 const audio_patch_handle_t handle) 678{ 679 Mutex::Autolock _l(mLock); 680 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 681 return sendConfigEvent_l(configEvent); 682} 683 684 685// post condition: mConfigEvents.isEmpty() 686void AudioFlinger::ThreadBase::processConfigEvents_l() 687{ 688 bool configChanged = false; 689 690 while (!mConfigEvents.isEmpty()) { 691 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 692 sp<ConfigEvent> event = mConfigEvents[0]; 693 mConfigEvents.removeAt(0); 694 switch (event->mType) { 695 case CFG_EVENT_PRIO: { 696 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 697 // FIXME Need to understand why this has to be done asynchronously 698 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 699 true /*asynchronous*/); 700 if (err != 0) { 701 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 702 data->mPrio, data->mPid, data->mTid, err); 703 } 704 } break; 705 case CFG_EVENT_IO: { 706 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 707 ioConfigChanged(data->mEvent); 708 } break; 709 case CFG_EVENT_SET_PARAMETER: { 710 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 711 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 712 configChanged = true; 713 } 714 } break; 715 case CFG_EVENT_CREATE_AUDIO_PATCH: { 716 CreateAudioPatchConfigEventData *data = 717 (CreateAudioPatchConfigEventData *)event->mData.get(); 718 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 719 } break; 720 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 721 ReleaseAudioPatchConfigEventData *data = 722 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 723 event->mStatus = releaseAudioPatch_l(data->mHandle); 724 } break; 725 default: 726 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 727 break; 728 } 729 { 730 Mutex::Autolock _l(event->mLock); 731 if (event->mWaitStatus) { 732 event->mWaitStatus = false; 733 event->mCond.signal(); 734 } 735 } 736 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 737 } 738 739 if (configChanged) { 740 cacheParameters_l(); 741 } 742} 743 744String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 745 String8 s; 746 const audio_channel_representation_t representation = 747 audio_channel_mask_get_representation(mask); 748 749 switch (representation) { 750 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 751 if (output) { 752 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 753 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 754 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 755 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 756 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 757 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 758 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 759 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 760 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 761 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 762 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 763 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 764 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 765 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 766 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 767 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 768 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 769 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 770 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 771 } else { 772 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 773 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 774 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 775 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 776 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 777 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 778 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 779 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 780 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 781 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 782 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 783 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 784 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 785 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 786 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 787 } 788 const int len = s.length(); 789 if (len > 2) { 790 char *str = s.lockBuffer(len); // needed? 791 s.unlockBuffer(len - 2); // remove trailing ", " 792 } 793 return s; 794 } 795 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 796 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 797 return s; 798 default: 799 s.appendFormat("unknown mask, representation:%d bits:%#x", 800 representation, audio_channel_mask_get_bits(mask)); 801 return s; 802 } 803} 804 805void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 806{ 807 const size_t SIZE = 256; 808 char buffer[SIZE]; 809 String8 result; 810 811 bool locked = AudioFlinger::dumpTryLock(mLock); 812 if (!locked) { 813 dprintf(fd, "thread %p may be deadlocked\n", this); 814 } 815 816 dprintf(fd, " Thread name: %s\n", mThreadName); 817 dprintf(fd, " I/O handle: %d\n", mId); 818 dprintf(fd, " TID: %d\n", getTid()); 819 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 820 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 821 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 822 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 823 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 824 dprintf(fd, " Channel count: %u\n", mChannelCount); 825 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 826 channelMaskToString(mChannelMask, mType != RECORD).string()); 827 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 828 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 829 dprintf(fd, " Pending config events:"); 830 size_t numConfig = mConfigEvents.size(); 831 if (numConfig) { 832 for (size_t i = 0; i < numConfig; i++) { 833 mConfigEvents[i]->dump(buffer, SIZE); 834 dprintf(fd, "\n %s", buffer); 835 } 836 dprintf(fd, "\n"); 837 } else { 838 dprintf(fd, " none\n"); 839 } 840 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 841 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 842 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 843 844 if (locked) { 845 mLock.unlock(); 846 } 847} 848 849void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 850{ 851 const size_t SIZE = 256; 852 char buffer[SIZE]; 853 String8 result; 854 855 size_t numEffectChains = mEffectChains.size(); 856 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 857 write(fd, buffer, strlen(buffer)); 858 859 for (size_t i = 0; i < numEffectChains; ++i) { 860 sp<EffectChain> chain = mEffectChains[i]; 861 if (chain != 0) { 862 chain->dump(fd, args); 863 } 864 } 865} 866 867void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 868{ 869 Mutex::Autolock _l(mLock); 870 acquireWakeLock_l(uid); 871} 872 873String16 AudioFlinger::ThreadBase::getWakeLockTag() 874{ 875 switch (mType) { 876 case MIXER: 877 return String16("AudioMix"); 878 case DIRECT: 879 return String16("AudioDirectOut"); 880 case DUPLICATING: 881 return String16("AudioDup"); 882 case RECORD: 883 return String16("AudioIn"); 884 case OFFLOAD: 885 return String16("AudioOffload"); 886 default: 887 ALOG_ASSERT(false); 888 return String16("AudioUnknown"); 889 } 890} 891 892void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 893{ 894 getPowerManager_l(); 895 if (mPowerManager != 0) { 896 sp<IBinder> binder = new BBinder(); 897 status_t status; 898 if (uid >= 0) { 899 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 900 binder, 901 getWakeLockTag(), 902 String16("media"), 903 uid, 904 true /* FIXME force oneway contrary to .aidl */); 905 } else { 906 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 907 binder, 908 getWakeLockTag(), 909 String16("media"), 910 true /* FIXME force oneway contrary to .aidl */); 911 } 912 if (status == NO_ERROR) { 913 mWakeLockToken = binder; 914 } 915 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 916 } 917} 918 919void AudioFlinger::ThreadBase::releaseWakeLock() 920{ 921 Mutex::Autolock _l(mLock); 922 releaseWakeLock_l(); 923} 924 925void AudioFlinger::ThreadBase::releaseWakeLock_l() 926{ 927 if (mWakeLockToken != 0) { 928 ALOGV("releaseWakeLock_l() %s", mThreadName); 929 if (mPowerManager != 0) { 930 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 931 true /* FIXME force oneway contrary to .aidl */); 932 } 933 mWakeLockToken.clear(); 934 } 935} 936 937void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 938 Mutex::Autolock _l(mLock); 939 updateWakeLockUids_l(uids); 940} 941 942void AudioFlinger::ThreadBase::getPowerManager_l() { 943 if (mSystemReady && mPowerManager == 0) { 944 // use checkService() to avoid blocking if power service is not up yet 945 sp<IBinder> binder = 946 defaultServiceManager()->checkService(String16("power")); 947 if (binder == 0) { 948 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 949 } else { 950 mPowerManager = interface_cast<IPowerManager>(binder); 951 binder->linkToDeath(mDeathRecipient); 952 } 953 } 954} 955 956void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 957 getPowerManager_l(); 958 if (mWakeLockToken == NULL) { 959 ALOGE("no wake lock to update!"); 960 return; 961 } 962 if (mPowerManager != 0) { 963 sp<IBinder> binder = new BBinder(); 964 status_t status; 965 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 966 true /* FIXME force oneway contrary to .aidl */); 967 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 968 } 969} 970 971void AudioFlinger::ThreadBase::clearPowerManager() 972{ 973 Mutex::Autolock _l(mLock); 974 releaseWakeLock_l(); 975 mPowerManager.clear(); 976} 977 978void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 979{ 980 sp<ThreadBase> thread = mThread.promote(); 981 if (thread != 0) { 982 thread->clearPowerManager(); 983 } 984 ALOGW("power manager service died !!!"); 985} 986 987void AudioFlinger::ThreadBase::setEffectSuspended( 988 const effect_uuid_t *type, bool suspend, int sessionId) 989{ 990 Mutex::Autolock _l(mLock); 991 setEffectSuspended_l(type, suspend, sessionId); 992} 993 994void AudioFlinger::ThreadBase::setEffectSuspended_l( 995 const effect_uuid_t *type, bool suspend, int sessionId) 996{ 997 sp<EffectChain> chain = getEffectChain_l(sessionId); 998 if (chain != 0) { 999 if (type != NULL) { 1000 chain->setEffectSuspended_l(type, suspend); 1001 } else { 1002 chain->setEffectSuspendedAll_l(suspend); 1003 } 1004 } 1005 1006 updateSuspendedSessions_l(type, suspend, sessionId); 1007} 1008 1009void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1010{ 1011 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1012 if (index < 0) { 1013 return; 1014 } 1015 1016 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1017 mSuspendedSessions.valueAt(index); 1018 1019 for (size_t i = 0; i < sessionEffects.size(); i++) { 1020 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1021 for (int j = 0; j < desc->mRefCount; j++) { 1022 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1023 chain->setEffectSuspendedAll_l(true); 1024 } else { 1025 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1026 desc->mType.timeLow); 1027 chain->setEffectSuspended_l(&desc->mType, true); 1028 } 1029 } 1030 } 1031} 1032 1033void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1034 bool suspend, 1035 int sessionId) 1036{ 1037 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1038 1039 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1040 1041 if (suspend) { 1042 if (index >= 0) { 1043 sessionEffects = mSuspendedSessions.valueAt(index); 1044 } else { 1045 mSuspendedSessions.add(sessionId, sessionEffects); 1046 } 1047 } else { 1048 if (index < 0) { 1049 return; 1050 } 1051 sessionEffects = mSuspendedSessions.valueAt(index); 1052 } 1053 1054 1055 int key = EffectChain::kKeyForSuspendAll; 1056 if (type != NULL) { 1057 key = type->timeLow; 1058 } 1059 index = sessionEffects.indexOfKey(key); 1060 1061 sp<SuspendedSessionDesc> desc; 1062 if (suspend) { 1063 if (index >= 0) { 1064 desc = sessionEffects.valueAt(index); 1065 } else { 1066 desc = new SuspendedSessionDesc(); 1067 if (type != NULL) { 1068 desc->mType = *type; 1069 } 1070 sessionEffects.add(key, desc); 1071 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1072 } 1073 desc->mRefCount++; 1074 } else { 1075 if (index < 0) { 1076 return; 1077 } 1078 desc = sessionEffects.valueAt(index); 1079 if (--desc->mRefCount == 0) { 1080 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1081 sessionEffects.removeItemsAt(index); 1082 if (sessionEffects.isEmpty()) { 1083 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1084 sessionId); 1085 mSuspendedSessions.removeItem(sessionId); 1086 } 1087 } 1088 } 1089 if (!sessionEffects.isEmpty()) { 1090 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1091 } 1092} 1093 1094void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1095 bool enabled, 1096 int sessionId) 1097{ 1098 Mutex::Autolock _l(mLock); 1099 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1100} 1101 1102void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1103 bool enabled, 1104 int sessionId) 1105{ 1106 if (mType != RECORD) { 1107 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1108 // another session. This gives the priority to well behaved effect control panels 1109 // and applications not using global effects. 1110 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1111 // global effects 1112 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1113 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1114 } 1115 } 1116 1117 sp<EffectChain> chain = getEffectChain_l(sessionId); 1118 if (chain != 0) { 1119 chain->checkSuspendOnEffectEnabled(effect, enabled); 1120 } 1121} 1122 1123// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1124sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1125 const sp<AudioFlinger::Client>& client, 1126 const sp<IEffectClient>& effectClient, 1127 int32_t priority, 1128 int sessionId, 1129 effect_descriptor_t *desc, 1130 int *enabled, 1131 status_t *status) 1132{ 1133 sp<EffectModule> effect; 1134 sp<EffectHandle> handle; 1135 status_t lStatus; 1136 sp<EffectChain> chain; 1137 bool chainCreated = false; 1138 bool effectCreated = false; 1139 bool effectRegistered = false; 1140 1141 lStatus = initCheck(); 1142 if (lStatus != NO_ERROR) { 1143 ALOGW("createEffect_l() Audio driver not initialized."); 1144 goto Exit; 1145 } 1146 1147 // Reject any effect on Direct output threads for now, since the format of 1148 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1149 if (mType == DIRECT) { 1150 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1151 desc->name, mThreadName); 1152 lStatus = BAD_VALUE; 1153 goto Exit; 1154 } 1155 1156 // Reject any effect on mixer or duplicating multichannel sinks. 1157 // TODO: fix both format and multichannel issues with effects. 1158 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1159 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1160 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1161 lStatus = BAD_VALUE; 1162 goto Exit; 1163 } 1164 1165 // Allow global effects only on offloaded and mixer threads 1166 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1167 switch (mType) { 1168 case MIXER: 1169 case OFFLOAD: 1170 break; 1171 case DIRECT: 1172 case DUPLICATING: 1173 case RECORD: 1174 default: 1175 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1176 desc->name, mThreadName); 1177 lStatus = BAD_VALUE; 1178 goto Exit; 1179 } 1180 } 1181 1182 // Only Pre processor effects are allowed on input threads and only on input threads 1183 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1184 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1185 desc->name, desc->flags, mType); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 1190 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1191 1192 { // scope for mLock 1193 Mutex::Autolock _l(mLock); 1194 1195 // check for existing effect chain with the requested audio session 1196 chain = getEffectChain_l(sessionId); 1197 if (chain == 0) { 1198 // create a new chain for this session 1199 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1200 chain = new EffectChain(this, sessionId); 1201 addEffectChain_l(chain); 1202 chain->setStrategy(getStrategyForSession_l(sessionId)); 1203 chainCreated = true; 1204 } else { 1205 effect = chain->getEffectFromDesc_l(desc); 1206 } 1207 1208 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1209 1210 if (effect == 0) { 1211 int id = mAudioFlinger->nextUniqueId(); 1212 // Check CPU and memory usage 1213 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1214 if (lStatus != NO_ERROR) { 1215 goto Exit; 1216 } 1217 effectRegistered = true; 1218 // create a new effect module if none present in the chain 1219 effect = new EffectModule(this, chain, desc, id, sessionId); 1220 lStatus = effect->status(); 1221 if (lStatus != NO_ERROR) { 1222 goto Exit; 1223 } 1224 effect->setOffloaded(mType == OFFLOAD, mId); 1225 1226 lStatus = chain->addEffect_l(effect); 1227 if (lStatus != NO_ERROR) { 1228 goto Exit; 1229 } 1230 effectCreated = true; 1231 1232 effect->setDevice(mOutDevice); 1233 effect->setDevice(mInDevice); 1234 effect->setMode(mAudioFlinger->getMode()); 1235 effect->setAudioSource(mAudioSource); 1236 } 1237 // create effect handle and connect it to effect module 1238 handle = new EffectHandle(effect, client, effectClient, priority); 1239 lStatus = handle->initCheck(); 1240 if (lStatus == OK) { 1241 lStatus = effect->addHandle(handle.get()); 1242 } 1243 if (enabled != NULL) { 1244 *enabled = (int)effect->isEnabled(); 1245 } 1246 } 1247 1248Exit: 1249 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1250 Mutex::Autolock _l(mLock); 1251 if (effectCreated) { 1252 chain->removeEffect_l(effect); 1253 } 1254 if (effectRegistered) { 1255 AudioSystem::unregisterEffect(effect->id()); 1256 } 1257 if (chainCreated) { 1258 removeEffectChain_l(chain); 1259 } 1260 handle.clear(); 1261 } 1262 1263 *status = lStatus; 1264 return handle; 1265} 1266 1267sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1268{ 1269 Mutex::Autolock _l(mLock); 1270 return getEffect_l(sessionId, effectId); 1271} 1272 1273sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1274{ 1275 sp<EffectChain> chain = getEffectChain_l(sessionId); 1276 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1277} 1278 1279// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1280// PlaybackThread::mLock held 1281status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1282{ 1283 // check for existing effect chain with the requested audio session 1284 int sessionId = effect->sessionId(); 1285 sp<EffectChain> chain = getEffectChain_l(sessionId); 1286 bool chainCreated = false; 1287 1288 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1289 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1290 this, effect->desc().name, effect->desc().flags); 1291 1292 if (chain == 0) { 1293 // create a new chain for this session 1294 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1295 chain = new EffectChain(this, sessionId); 1296 addEffectChain_l(chain); 1297 chain->setStrategy(getStrategyForSession_l(sessionId)); 1298 chainCreated = true; 1299 } 1300 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1301 1302 if (chain->getEffectFromId_l(effect->id()) != 0) { 1303 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1304 this, effect->desc().name, chain.get()); 1305 return BAD_VALUE; 1306 } 1307 1308 effect->setOffloaded(mType == OFFLOAD, mId); 1309 1310 status_t status = chain->addEffect_l(effect); 1311 if (status != NO_ERROR) { 1312 if (chainCreated) { 1313 removeEffectChain_l(chain); 1314 } 1315 return status; 1316 } 1317 1318 effect->setDevice(mOutDevice); 1319 effect->setDevice(mInDevice); 1320 effect->setMode(mAudioFlinger->getMode()); 1321 effect->setAudioSource(mAudioSource); 1322 return NO_ERROR; 1323} 1324 1325void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1326 1327 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1328 effect_descriptor_t desc = effect->desc(); 1329 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1330 detachAuxEffect_l(effect->id()); 1331 } 1332 1333 sp<EffectChain> chain = effect->chain().promote(); 1334 if (chain != 0) { 1335 // remove effect chain if removing last effect 1336 if (chain->removeEffect_l(effect) == 0) { 1337 removeEffectChain_l(chain); 1338 } 1339 } else { 1340 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1341 } 1342} 1343 1344void AudioFlinger::ThreadBase::lockEffectChains_l( 1345 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1346{ 1347 effectChains = mEffectChains; 1348 for (size_t i = 0; i < mEffectChains.size(); i++) { 1349 mEffectChains[i]->lock(); 1350 } 1351} 1352 1353void AudioFlinger::ThreadBase::unlockEffectChains( 1354 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1355{ 1356 for (size_t i = 0; i < effectChains.size(); i++) { 1357 effectChains[i]->unlock(); 1358 } 1359} 1360 1361sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1362{ 1363 Mutex::Autolock _l(mLock); 1364 return getEffectChain_l(sessionId); 1365} 1366 1367sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1368{ 1369 size_t size = mEffectChains.size(); 1370 for (size_t i = 0; i < size; i++) { 1371 if (mEffectChains[i]->sessionId() == sessionId) { 1372 return mEffectChains[i]; 1373 } 1374 } 1375 return 0; 1376} 1377 1378void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1379{ 1380 Mutex::Autolock _l(mLock); 1381 size_t size = mEffectChains.size(); 1382 for (size_t i = 0; i < size; i++) { 1383 mEffectChains[i]->setMode_l(mode); 1384 } 1385} 1386 1387void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1388{ 1389 config->type = AUDIO_PORT_TYPE_MIX; 1390 config->ext.mix.handle = mId; 1391 config->sample_rate = mSampleRate; 1392 config->format = mFormat; 1393 config->channel_mask = mChannelMask; 1394 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1395 AUDIO_PORT_CONFIG_FORMAT; 1396} 1397 1398void AudioFlinger::ThreadBase::systemReady() 1399{ 1400 Mutex::Autolock _l(mLock); 1401 if (mSystemReady) { 1402 return; 1403 } 1404 mSystemReady = true; 1405 1406 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1407 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1408 } 1409 mPendingConfigEvents.clear(); 1410} 1411 1412 1413// ---------------------------------------------------------------------------- 1414// Playback 1415// ---------------------------------------------------------------------------- 1416 1417AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1418 AudioStreamOut* output, 1419 audio_io_handle_t id, 1420 audio_devices_t device, 1421 type_t type, 1422 bool systemReady) 1423 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1424 mNormalFrameCount(0), mSinkBuffer(NULL), 1425 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1426 mMixerBuffer(NULL), 1427 mMixerBufferSize(0), 1428 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1429 mMixerBufferValid(false), 1430 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1431 mEffectBuffer(NULL), 1432 mEffectBufferSize(0), 1433 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1434 mEffectBufferValid(false), 1435 mSuspended(0), mBytesWritten(0), 1436 mActiveTracksGeneration(0), 1437 // mStreamTypes[] initialized in constructor body 1438 mOutput(output), 1439 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1440 mMixerStatus(MIXER_IDLE), 1441 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1442 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1443 mBytesRemaining(0), 1444 mCurrentWriteLength(0), 1445 mUseAsyncWrite(false), 1446 mWriteAckSequence(0), 1447 mDrainSequence(0), 1448 mSignalPending(false), 1449 mScreenState(AudioFlinger::mScreenState), 1450 // index 0 is reserved for normal mixer's submix 1451 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1452 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1453 // mLatchD, mLatchQ, 1454 mLatchDValid(false), mLatchQValid(false) 1455{ 1456 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1457 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1458 1459 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1460 // it would be safer to explicitly pass initial masterVolume/masterMute as 1461 // parameter. 1462 // 1463 // If the HAL we are using has support for master volume or master mute, 1464 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1465 // and the mute set to false). 1466 mMasterVolume = audioFlinger->masterVolume_l(); 1467 mMasterMute = audioFlinger->masterMute_l(); 1468 if (mOutput && mOutput->audioHwDev) { 1469 if (mOutput->audioHwDev->canSetMasterVolume()) { 1470 mMasterVolume = 1.0; 1471 } 1472 1473 if (mOutput->audioHwDev->canSetMasterMute()) { 1474 mMasterMute = false; 1475 } 1476 } 1477 1478 readOutputParameters_l(); 1479 1480 // ++ operator does not compile 1481 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1482 stream = (audio_stream_type_t) (stream + 1)) { 1483 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1484 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1485 } 1486} 1487 1488AudioFlinger::PlaybackThread::~PlaybackThread() 1489{ 1490 mAudioFlinger->unregisterWriter(mNBLogWriter); 1491 free(mSinkBuffer); 1492 free(mMixerBuffer); 1493 free(mEffectBuffer); 1494} 1495 1496void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1497{ 1498 dumpInternals(fd, args); 1499 dumpTracks(fd, args); 1500 dumpEffectChains(fd, args); 1501} 1502 1503void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1504{ 1505 const size_t SIZE = 256; 1506 char buffer[SIZE]; 1507 String8 result; 1508 1509 result.appendFormat(" Stream volumes in dB: "); 1510 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1511 const stream_type_t *st = &mStreamTypes[i]; 1512 if (i > 0) { 1513 result.appendFormat(", "); 1514 } 1515 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1516 if (st->mute) { 1517 result.append("M"); 1518 } 1519 } 1520 result.append("\n"); 1521 write(fd, result.string(), result.length()); 1522 result.clear(); 1523 1524 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1525 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1526 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1527 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1528 1529 size_t numtracks = mTracks.size(); 1530 size_t numactive = mActiveTracks.size(); 1531 dprintf(fd, " %d Tracks", numtracks); 1532 size_t numactiveseen = 0; 1533 if (numtracks) { 1534 dprintf(fd, " of which %d are active\n", numactive); 1535 Track::appendDumpHeader(result); 1536 for (size_t i = 0; i < numtracks; ++i) { 1537 sp<Track> track = mTracks[i]; 1538 if (track != 0) { 1539 bool active = mActiveTracks.indexOf(track) >= 0; 1540 if (active) { 1541 numactiveseen++; 1542 } 1543 track->dump(buffer, SIZE, active); 1544 result.append(buffer); 1545 } 1546 } 1547 } else { 1548 result.append("\n"); 1549 } 1550 if (numactiveseen != numactive) { 1551 // some tracks in the active list were not in the tracks list 1552 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1553 " not in the track list\n"); 1554 result.append(buffer); 1555 Track::appendDumpHeader(result); 1556 for (size_t i = 0; i < numactive; ++i) { 1557 sp<Track> track = mActiveTracks[i].promote(); 1558 if (track != 0 && mTracks.indexOf(track) < 0) { 1559 track->dump(buffer, SIZE, true); 1560 result.append(buffer); 1561 } 1562 } 1563 } 1564 1565 write(fd, result.string(), result.size()); 1566} 1567 1568void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1569{ 1570 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1571 1572 dumpBase(fd, args); 1573 1574 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1575 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1576 dprintf(fd, " Total writes: %d\n", mNumWrites); 1577 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1578 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1579 dprintf(fd, " Suspend count: %d\n", mSuspended); 1580 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1581 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1582 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1583 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1584 AudioStreamOut *output = mOutput; 1585 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1586 String8 flagsAsString = outputFlagsToString(flags); 1587 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1588} 1589 1590// Thread virtuals 1591 1592void AudioFlinger::PlaybackThread::onFirstRef() 1593{ 1594 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1595} 1596 1597// ThreadBase virtuals 1598void AudioFlinger::PlaybackThread::preExit() 1599{ 1600 ALOGV(" preExit()"); 1601 // FIXME this is using hard-coded strings but in the future, this functionality will be 1602 // converted to use audio HAL extensions required to support tunneling 1603 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1604} 1605 1606// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1607sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1608 const sp<AudioFlinger::Client>& client, 1609 audio_stream_type_t streamType, 1610 uint32_t sampleRate, 1611 audio_format_t format, 1612 audio_channel_mask_t channelMask, 1613 size_t *pFrameCount, 1614 const sp<IMemory>& sharedBuffer, 1615 int sessionId, 1616 IAudioFlinger::track_flags_t *flags, 1617 pid_t tid, 1618 int uid, 1619 status_t *status) 1620{ 1621 size_t frameCount = *pFrameCount; 1622 sp<Track> track; 1623 status_t lStatus; 1624 1625 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1626 1627 // client expresses a preference for FAST, but we get the final say 1628 if (*flags & IAudioFlinger::TRACK_FAST) { 1629 if ( 1630 // not timed 1631 (!isTimed) && 1632 // either of these use cases: 1633 ( 1634 // use case 1: shared buffer with any frame count 1635 ( 1636 (sharedBuffer != 0) 1637 ) || 1638 // use case 2: frame count is default or at least as large as HAL 1639 ( 1640 // we formerly checked for a callback handler (non-0 tid), 1641 // but that is no longer required for TRANSFER_OBTAIN mode 1642 ((frameCount == 0) || 1643 (frameCount >= mFrameCount)) 1644 ) 1645 ) && 1646 // PCM data 1647 audio_is_linear_pcm(format) && 1648 // TODO: extract as a data library function that checks that a computationally 1649 // expensive downmixer is not required: isFastOutputChannelConversion() 1650 (channelMask == mChannelMask || 1651 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1652 (channelMask == AUDIO_CHANNEL_OUT_MONO 1653 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1654 // hardware sample rate 1655 (sampleRate == mSampleRate) && 1656 // normal mixer has an associated fast mixer 1657 hasFastMixer() && 1658 // there are sufficient fast track slots available 1659 (mFastTrackAvailMask != 0) 1660 // FIXME test that MixerThread for this fast track has a capable output HAL 1661 // FIXME add a permission test also? 1662 ) { 1663 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1664 if (frameCount == 0) { 1665 // read the fast track multiplier property the first time it is needed 1666 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1667 if (ok != 0) { 1668 ALOGE("%s pthread_once failed: %d", __func__, ok); 1669 } 1670 frameCount = mFrameCount * sFastTrackMultiplier; 1671 } 1672 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1673 frameCount, mFrameCount); 1674 } else { 1675 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1676 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1677 "sampleRate=%u mSampleRate=%u " 1678 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1679 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1680 audio_is_linear_pcm(format), 1681 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1682 *flags &= ~IAudioFlinger::TRACK_FAST; 1683 } 1684 } 1685 // For normal PCM streaming tracks, update minimum frame count. 1686 // For compatibility with AudioTrack calculation, buffer depth is forced 1687 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1688 // This is probably too conservative, but legacy application code may depend on it. 1689 // If you change this calculation, also review the start threshold which is related. 1690 if (!(*flags & IAudioFlinger::TRACK_FAST) 1691 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1692 // this must match AudioTrack.cpp calculateMinFrameCount(). 1693 // TODO: Move to a common library 1694 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1695 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1696 if (minBufCount < 2) { 1697 minBufCount = 2; 1698 } 1699 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1700 // or the client should compute and pass in a larger buffer request. 1701 size_t minFrameCount = 1702 minBufCount * sourceFramesNeededWithTimestretch( 1703 sampleRate, mNormalFrameCount, 1704 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1705 if (frameCount < minFrameCount) { // including frameCount == 0 1706 frameCount = minFrameCount; 1707 } 1708 } 1709 *pFrameCount = frameCount; 1710 1711 switch (mType) { 1712 1713 case DIRECT: 1714 if (audio_is_linear_pcm(format)) { 1715 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1716 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1717 "for output %p with format %#x", 1718 sampleRate, format, channelMask, mOutput, mFormat); 1719 lStatus = BAD_VALUE; 1720 goto Exit; 1721 } 1722 } 1723 break; 1724 1725 case OFFLOAD: 1726 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1727 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1728 "for output %p with format %#x", 1729 sampleRate, format, channelMask, mOutput, mFormat); 1730 lStatus = BAD_VALUE; 1731 goto Exit; 1732 } 1733 break; 1734 1735 default: 1736 if (!audio_is_linear_pcm(format)) { 1737 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1738 "for output %p with format %#x", 1739 format, mOutput, mFormat); 1740 lStatus = BAD_VALUE; 1741 goto Exit; 1742 } 1743 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1744 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1745 lStatus = BAD_VALUE; 1746 goto Exit; 1747 } 1748 break; 1749 1750 } 1751 1752 lStatus = initCheck(); 1753 if (lStatus != NO_ERROR) { 1754 ALOGE("createTrack_l() audio driver not initialized"); 1755 goto Exit; 1756 } 1757 1758 { // scope for mLock 1759 Mutex::Autolock _l(mLock); 1760 1761 // all tracks in same audio session must share the same routing strategy otherwise 1762 // conflicts will happen when tracks are moved from one output to another by audio policy 1763 // manager 1764 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1765 for (size_t i = 0; i < mTracks.size(); ++i) { 1766 sp<Track> t = mTracks[i]; 1767 if (t != 0 && t->isExternalTrack()) { 1768 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1769 if (sessionId == t->sessionId() && strategy != actual) { 1770 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1771 strategy, actual); 1772 lStatus = BAD_VALUE; 1773 goto Exit; 1774 } 1775 } 1776 } 1777 1778 if (!isTimed) { 1779 track = new Track(this, client, streamType, sampleRate, format, 1780 channelMask, frameCount, NULL, sharedBuffer, 1781 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1782 } else { 1783 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1784 channelMask, frameCount, sharedBuffer, sessionId, uid); 1785 } 1786 1787 // new Track always returns non-NULL, 1788 // but TimedTrack::create() is a factory that could fail by returning NULL 1789 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1790 if (lStatus != NO_ERROR) { 1791 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1792 // track must be cleared from the caller as the caller has the AF lock 1793 goto Exit; 1794 } 1795 mTracks.add(track); 1796 1797 sp<EffectChain> chain = getEffectChain_l(sessionId); 1798 if (chain != 0) { 1799 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1800 track->setMainBuffer(chain->inBuffer()); 1801 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1802 chain->incTrackCnt(); 1803 } 1804 1805 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1806 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1807 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1808 // so ask activity manager to do this on our behalf 1809 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1810 } 1811 } 1812 1813 lStatus = NO_ERROR; 1814 1815Exit: 1816 *status = lStatus; 1817 return track; 1818} 1819 1820uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1821{ 1822 return latency; 1823} 1824 1825uint32_t AudioFlinger::PlaybackThread::latency() const 1826{ 1827 Mutex::Autolock _l(mLock); 1828 return latency_l(); 1829} 1830uint32_t AudioFlinger::PlaybackThread::latency_l() const 1831{ 1832 if (initCheck() == NO_ERROR) { 1833 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1834 } else { 1835 return 0; 1836 } 1837} 1838 1839void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1840{ 1841 Mutex::Autolock _l(mLock); 1842 // Don't apply master volume in SW if our HAL can do it for us. 1843 if (mOutput && mOutput->audioHwDev && 1844 mOutput->audioHwDev->canSetMasterVolume()) { 1845 mMasterVolume = 1.0; 1846 } else { 1847 mMasterVolume = value; 1848 } 1849} 1850 1851void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1852{ 1853 Mutex::Autolock _l(mLock); 1854 // Don't apply master mute in SW if our HAL can do it for us. 1855 if (mOutput && mOutput->audioHwDev && 1856 mOutput->audioHwDev->canSetMasterMute()) { 1857 mMasterMute = false; 1858 } else { 1859 mMasterMute = muted; 1860 } 1861} 1862 1863void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1864{ 1865 Mutex::Autolock _l(mLock); 1866 mStreamTypes[stream].volume = value; 1867 broadcast_l(); 1868} 1869 1870void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1871{ 1872 Mutex::Autolock _l(mLock); 1873 mStreamTypes[stream].mute = muted; 1874 broadcast_l(); 1875} 1876 1877float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1878{ 1879 Mutex::Autolock _l(mLock); 1880 return mStreamTypes[stream].volume; 1881} 1882 1883// addTrack_l() must be called with ThreadBase::mLock held 1884status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1885{ 1886 status_t status = ALREADY_EXISTS; 1887 1888 // set retry count for buffer fill 1889 track->mRetryCount = kMaxTrackStartupRetries; 1890 if (mActiveTracks.indexOf(track) < 0) { 1891 // the track is newly added, make sure it fills up all its 1892 // buffers before playing. This is to ensure the client will 1893 // effectively get the latency it requested. 1894 if (track->isExternalTrack()) { 1895 TrackBase::track_state state = track->mState; 1896 mLock.unlock(); 1897 status = AudioSystem::startOutput(mId, track->streamType(), 1898 (audio_session_t)track->sessionId()); 1899 mLock.lock(); 1900 // abort track was stopped/paused while we released the lock 1901 if (state != track->mState) { 1902 if (status == NO_ERROR) { 1903 mLock.unlock(); 1904 AudioSystem::stopOutput(mId, track->streamType(), 1905 (audio_session_t)track->sessionId()); 1906 mLock.lock(); 1907 } 1908 return INVALID_OPERATION; 1909 } 1910 // abort if start is rejected by audio policy manager 1911 if (status != NO_ERROR) { 1912 return PERMISSION_DENIED; 1913 } 1914#ifdef ADD_BATTERY_DATA 1915 // to track the speaker usage 1916 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1917#endif 1918 } 1919 1920 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1921 track->mResetDone = false; 1922 track->mPresentationCompleteFrames = 0; 1923 mActiveTracks.add(track); 1924 mWakeLockUids.add(track->uid()); 1925 mActiveTracksGeneration++; 1926 mLatestActiveTrack = track; 1927 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1928 if (chain != 0) { 1929 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1930 track->sessionId()); 1931 chain->incActiveTrackCnt(); 1932 } 1933 1934 status = NO_ERROR; 1935 } 1936 1937 onAddNewTrack_l(); 1938 return status; 1939} 1940 1941bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1942{ 1943 track->terminate(); 1944 // active tracks are removed by threadLoop() 1945 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1946 track->mState = TrackBase::STOPPED; 1947 if (!trackActive) { 1948 removeTrack_l(track); 1949 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1950 track->mState = TrackBase::STOPPING_1; 1951 } 1952 1953 return trackActive; 1954} 1955 1956void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1957{ 1958 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1959 mTracks.remove(track); 1960 deleteTrackName_l(track->name()); 1961 // redundant as track is about to be destroyed, for dumpsys only 1962 track->mName = -1; 1963 if (track->isFastTrack()) { 1964 int index = track->mFastIndex; 1965 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1966 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1967 mFastTrackAvailMask |= 1 << index; 1968 // redundant as track is about to be destroyed, for dumpsys only 1969 track->mFastIndex = -1; 1970 } 1971 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1972 if (chain != 0) { 1973 chain->decTrackCnt(); 1974 } 1975} 1976 1977void AudioFlinger::PlaybackThread::broadcast_l() 1978{ 1979 // Thread could be blocked waiting for async 1980 // so signal it to handle state changes immediately 1981 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1982 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1983 mSignalPending = true; 1984 mWaitWorkCV.broadcast(); 1985} 1986 1987String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1988{ 1989 Mutex::Autolock _l(mLock); 1990 if (initCheck() != NO_ERROR) { 1991 return String8(); 1992 } 1993 1994 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1995 const String8 out_s8(s); 1996 free(s); 1997 return out_s8; 1998} 1999 2000void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event) { 2001 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2002 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2003 2004 desc->mIoHandle = mId; 2005 2006 switch (event) { 2007 case AUDIO_OUTPUT_OPENED: 2008 case AUDIO_OUTPUT_CONFIG_CHANGED: 2009 desc->mPatch = mPatch; 2010 desc->mChannelMask = mChannelMask; 2011 desc->mSamplingRate = mSampleRate; 2012 desc->mFormat = mFormat; 2013 desc->mFrameCount = mNormalFrameCount; // FIXME see 2014 // AudioFlinger::frameCount(audio_io_handle_t) 2015 desc->mLatency = latency_l(); 2016 break; 2017 2018 case AUDIO_OUTPUT_CLOSED: 2019 default: 2020 break; 2021 } 2022 mAudioFlinger->ioConfigChanged(event, desc); 2023} 2024 2025void AudioFlinger::PlaybackThread::writeCallback() 2026{ 2027 ALOG_ASSERT(mCallbackThread != 0); 2028 mCallbackThread->resetWriteBlocked(); 2029} 2030 2031void AudioFlinger::PlaybackThread::drainCallback() 2032{ 2033 ALOG_ASSERT(mCallbackThread != 0); 2034 mCallbackThread->resetDraining(); 2035} 2036 2037void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2038{ 2039 Mutex::Autolock _l(mLock); 2040 // reject out of sequence requests 2041 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2042 mWriteAckSequence &= ~1; 2043 mWaitWorkCV.signal(); 2044 } 2045} 2046 2047void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2048{ 2049 Mutex::Autolock _l(mLock); 2050 // reject out of sequence requests 2051 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2052 mDrainSequence &= ~1; 2053 mWaitWorkCV.signal(); 2054 } 2055} 2056 2057// static 2058int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2059 void *param __unused, 2060 void *cookie) 2061{ 2062 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2063 ALOGV("asyncCallback() event %d", event); 2064 switch (event) { 2065 case STREAM_CBK_EVENT_WRITE_READY: 2066 me->writeCallback(); 2067 break; 2068 case STREAM_CBK_EVENT_DRAIN_READY: 2069 me->drainCallback(); 2070 break; 2071 default: 2072 ALOGW("asyncCallback() unknown event %d", event); 2073 break; 2074 } 2075 return 0; 2076} 2077 2078void AudioFlinger::PlaybackThread::readOutputParameters_l() 2079{ 2080 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2081 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2082 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2083 if (!audio_is_output_channel(mChannelMask)) { 2084 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2085 } 2086 if ((mType == MIXER || mType == DUPLICATING) 2087 && !isValidPcmSinkChannelMask(mChannelMask)) { 2088 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2089 mChannelMask); 2090 } 2091 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2092 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2093 mFormat = mHALFormat; 2094 if (!audio_is_valid_format(mFormat)) { 2095 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2096 } 2097 if ((mType == MIXER || mType == DUPLICATING) 2098 && !isValidPcmSinkFormat(mFormat)) { 2099 LOG_FATAL("HAL format %#x not supported for mixed output", 2100 mFormat); 2101 } 2102 mFrameSize = mOutput->getFrameSize(); 2103 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2104 mFrameCount = mBufferSize / mFrameSize; 2105 if (mFrameCount & 15) { 2106 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2107 mFrameCount); 2108 } 2109 2110 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2111 (mOutput->stream->set_callback != NULL)) { 2112 if (mOutput->stream->set_callback(mOutput->stream, 2113 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2114 mUseAsyncWrite = true; 2115 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2116 } 2117 } 2118 2119 mHwSupportsPause = false; 2120 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2121 if (mOutput->stream->pause != NULL) { 2122 if (mOutput->stream->resume != NULL) { 2123 mHwSupportsPause = true; 2124 } else { 2125 ALOGW("direct output implements pause but not resume"); 2126 } 2127 } else if (mOutput->stream->resume != NULL) { 2128 ALOGW("direct output implements resume but not pause"); 2129 } 2130 } 2131 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2132 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2133 } 2134 2135 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2136 // For best precision, we use float instead of the associated output 2137 // device format (typically PCM 16 bit). 2138 2139 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2140 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2141 mBufferSize = mFrameSize * mFrameCount; 2142 2143 // TODO: We currently use the associated output device channel mask and sample rate. 2144 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2145 // (if a valid mask) to avoid premature downmix. 2146 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2147 // instead of the output device sample rate to avoid loss of high frequency information. 2148 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2149 } 2150 2151 // Calculate size of normal sink buffer relative to the HAL output buffer size 2152 double multiplier = 1.0; 2153 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2154 kUseFastMixer == FastMixer_Dynamic)) { 2155 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2156 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2157 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2158 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2159 maxNormalFrameCount = maxNormalFrameCount & ~15; 2160 if (maxNormalFrameCount < minNormalFrameCount) { 2161 maxNormalFrameCount = minNormalFrameCount; 2162 } 2163 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2164 if (multiplier <= 1.0) { 2165 multiplier = 1.0; 2166 } else if (multiplier <= 2.0) { 2167 if (2 * mFrameCount <= maxNormalFrameCount) { 2168 multiplier = 2.0; 2169 } else { 2170 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2171 } 2172 } else { 2173 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2174 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2175 // track, but we sometimes have to do this to satisfy the maximum frame count 2176 // constraint) 2177 // FIXME this rounding up should not be done if no HAL SRC 2178 uint32_t truncMult = (uint32_t) multiplier; 2179 if ((truncMult & 1)) { 2180 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2181 ++truncMult; 2182 } 2183 } 2184 multiplier = (double) truncMult; 2185 } 2186 } 2187 mNormalFrameCount = multiplier * mFrameCount; 2188 // round up to nearest 16 frames to satisfy AudioMixer 2189 if (mType == MIXER || mType == DUPLICATING) { 2190 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2191 } 2192 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2193 mNormalFrameCount); 2194 2195 // Check if we want to throttle the processing to no more than 2x normal rate 2196 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2197 mThreadThrottleTimeMs = 0; 2198 mThreadThrottleEndMs = 0; 2199 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2200 2201 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2202 // Originally this was int16_t[] array, need to remove legacy implications. 2203 free(mSinkBuffer); 2204 mSinkBuffer = NULL; 2205 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2206 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2207 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2208 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2209 2210 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2211 // drives the output. 2212 free(mMixerBuffer); 2213 mMixerBuffer = NULL; 2214 if (mMixerBufferEnabled) { 2215 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2216 mMixerBufferSize = mNormalFrameCount * mChannelCount 2217 * audio_bytes_per_sample(mMixerBufferFormat); 2218 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2219 } 2220 free(mEffectBuffer); 2221 mEffectBuffer = NULL; 2222 if (mEffectBufferEnabled) { 2223 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2224 mEffectBufferSize = mNormalFrameCount * mChannelCount 2225 * audio_bytes_per_sample(mEffectBufferFormat); 2226 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2227 } 2228 2229 // force reconfiguration of effect chains and engines to take new buffer size and audio 2230 // parameters into account 2231 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2232 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2233 // matter. 2234 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2235 Vector< sp<EffectChain> > effectChains = mEffectChains; 2236 for (size_t i = 0; i < effectChains.size(); i ++) { 2237 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2238 } 2239} 2240 2241 2242status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2243{ 2244 if (halFrames == NULL || dspFrames == NULL) { 2245 return BAD_VALUE; 2246 } 2247 Mutex::Autolock _l(mLock); 2248 if (initCheck() != NO_ERROR) { 2249 return INVALID_OPERATION; 2250 } 2251 size_t framesWritten = mBytesWritten / mFrameSize; 2252 *halFrames = framesWritten; 2253 2254 if (isSuspended()) { 2255 // return an estimation of rendered frames when the output is suspended 2256 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2257 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2258 return NO_ERROR; 2259 } else { 2260 status_t status; 2261 uint32_t frames; 2262 status = mOutput->getRenderPosition(&frames); 2263 *dspFrames = (size_t)frames; 2264 return status; 2265 } 2266} 2267 2268uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2269{ 2270 Mutex::Autolock _l(mLock); 2271 uint32_t result = 0; 2272 if (getEffectChain_l(sessionId) != 0) { 2273 result = EFFECT_SESSION; 2274 } 2275 2276 for (size_t i = 0; i < mTracks.size(); ++i) { 2277 sp<Track> track = mTracks[i]; 2278 if (sessionId == track->sessionId() && !track->isInvalid()) { 2279 result |= TRACK_SESSION; 2280 break; 2281 } 2282 } 2283 2284 return result; 2285} 2286 2287uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2288{ 2289 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2290 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2291 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2292 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2293 } 2294 for (size_t i = 0; i < mTracks.size(); i++) { 2295 sp<Track> track = mTracks[i]; 2296 if (sessionId == track->sessionId() && !track->isInvalid()) { 2297 return AudioSystem::getStrategyForStream(track->streamType()); 2298 } 2299 } 2300 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2301} 2302 2303 2304AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2305{ 2306 Mutex::Autolock _l(mLock); 2307 return mOutput; 2308} 2309 2310AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2311{ 2312 Mutex::Autolock _l(mLock); 2313 AudioStreamOut *output = mOutput; 2314 mOutput = NULL; 2315 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2316 // must push a NULL and wait for ack 2317 mOutputSink.clear(); 2318 mPipeSink.clear(); 2319 mNormalSink.clear(); 2320 return output; 2321} 2322 2323// this method must always be called either with ThreadBase mLock held or inside the thread loop 2324audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2325{ 2326 if (mOutput == NULL) { 2327 return NULL; 2328 } 2329 return &mOutput->stream->common; 2330} 2331 2332uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2333{ 2334 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2335} 2336 2337status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2338{ 2339 if (!isValidSyncEvent(event)) { 2340 return BAD_VALUE; 2341 } 2342 2343 Mutex::Autolock _l(mLock); 2344 2345 for (size_t i = 0; i < mTracks.size(); ++i) { 2346 sp<Track> track = mTracks[i]; 2347 if (event->triggerSession() == track->sessionId()) { 2348 (void) track->setSyncEvent(event); 2349 return NO_ERROR; 2350 } 2351 } 2352 2353 return NAME_NOT_FOUND; 2354} 2355 2356bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2357{ 2358 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2359} 2360 2361void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2362 const Vector< sp<Track> >& tracksToRemove) 2363{ 2364 size_t count = tracksToRemove.size(); 2365 if (count > 0) { 2366 for (size_t i = 0 ; i < count ; i++) { 2367 const sp<Track>& track = tracksToRemove.itemAt(i); 2368 if (track->isExternalTrack()) { 2369 AudioSystem::stopOutput(mId, track->streamType(), 2370 (audio_session_t)track->sessionId()); 2371#ifdef ADD_BATTERY_DATA 2372 // to track the speaker usage 2373 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2374#endif 2375 if (track->isTerminated()) { 2376 AudioSystem::releaseOutput(mId, track->streamType(), 2377 (audio_session_t)track->sessionId()); 2378 } 2379 } 2380 } 2381 } 2382} 2383 2384void AudioFlinger::PlaybackThread::checkSilentMode_l() 2385{ 2386 if (!mMasterMute) { 2387 char value[PROPERTY_VALUE_MAX]; 2388 if (property_get("ro.audio.silent", value, "0") > 0) { 2389 char *endptr; 2390 unsigned long ul = strtoul(value, &endptr, 0); 2391 if (*endptr == '\0' && ul != 0) { 2392 ALOGD("Silence is golden"); 2393 // The setprop command will not allow a property to be changed after 2394 // the first time it is set, so we don't have to worry about un-muting. 2395 setMasterMute_l(true); 2396 } 2397 } 2398 } 2399} 2400 2401// shared by MIXER and DIRECT, overridden by DUPLICATING 2402ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2403{ 2404 // FIXME rewrite to reduce number of system calls 2405 mLastWriteTime = systemTime(); 2406 mInWrite = true; 2407 ssize_t bytesWritten; 2408 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2409 2410 // If an NBAIO sink is present, use it to write the normal mixer's submix 2411 if (mNormalSink != 0) { 2412 2413 const size_t count = mBytesRemaining / mFrameSize; 2414 2415 ATRACE_BEGIN("write"); 2416 // update the setpoint when AudioFlinger::mScreenState changes 2417 uint32_t screenState = AudioFlinger::mScreenState; 2418 if (screenState != mScreenState) { 2419 mScreenState = screenState; 2420 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2421 if (pipe != NULL) { 2422 pipe->setAvgFrames((mScreenState & 1) ? 2423 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2424 } 2425 } 2426 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2427 ATRACE_END(); 2428 if (framesWritten > 0) { 2429 bytesWritten = framesWritten * mFrameSize; 2430 } else { 2431 bytesWritten = framesWritten; 2432 } 2433 mLatchDValid = false; 2434 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2435 if (status == NO_ERROR) { 2436 size_t totalFramesWritten = mNormalSink->framesWritten(); 2437 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2438 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2439 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2440 mLatchDValid = true; 2441 } 2442 } 2443 // otherwise use the HAL / AudioStreamOut directly 2444 } else { 2445 // Direct output and offload threads 2446 2447 if (mUseAsyncWrite) { 2448 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2449 mWriteAckSequence += 2; 2450 mWriteAckSequence |= 1; 2451 ALOG_ASSERT(mCallbackThread != 0); 2452 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2453 } 2454 // FIXME We should have an implementation of timestamps for direct output threads. 2455 // They are used e.g for multichannel PCM playback over HDMI. 2456 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2457 if (mUseAsyncWrite && 2458 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2459 // do not wait for async callback in case of error of full write 2460 mWriteAckSequence &= ~1; 2461 ALOG_ASSERT(mCallbackThread != 0); 2462 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2463 } 2464 } 2465 2466 mNumWrites++; 2467 mInWrite = false; 2468 mStandby = false; 2469 return bytesWritten; 2470} 2471 2472void AudioFlinger::PlaybackThread::threadLoop_drain() 2473{ 2474 if (mOutput->stream->drain) { 2475 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2476 if (mUseAsyncWrite) { 2477 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2478 mDrainSequence |= 1; 2479 ALOG_ASSERT(mCallbackThread != 0); 2480 mCallbackThread->setDraining(mDrainSequence); 2481 } 2482 mOutput->stream->drain(mOutput->stream, 2483 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2484 : AUDIO_DRAIN_ALL); 2485 } 2486} 2487 2488void AudioFlinger::PlaybackThread::threadLoop_exit() 2489{ 2490 { 2491 Mutex::Autolock _l(mLock); 2492 for (size_t i = 0; i < mTracks.size(); i++) { 2493 sp<Track> track = mTracks[i]; 2494 track->invalidate(); 2495 } 2496 } 2497} 2498 2499/* 2500The derived values that are cached: 2501 - mSinkBufferSize from frame count * frame size 2502 - mActiveSleepTimeUs from activeSleepTimeUs() 2503 - mIdleSleepTimeUs from idleSleepTimeUs() 2504 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) 2505 - maxPeriod from frame count and sample rate (MIXER only) 2506 2507The parameters that affect these derived values are: 2508 - frame count 2509 - frame size 2510 - sample rate 2511 - device type: A2DP or not 2512 - device latency 2513 - format: PCM or not 2514 - active sleep time 2515 - idle sleep time 2516*/ 2517 2518void AudioFlinger::PlaybackThread::cacheParameters_l() 2519{ 2520 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2521 mActiveSleepTimeUs = activeSleepTimeUs(); 2522 mIdleSleepTimeUs = idleSleepTimeUs(); 2523} 2524 2525void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2526{ 2527 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2528 this, streamType, mTracks.size()); 2529 Mutex::Autolock _l(mLock); 2530 2531 size_t size = mTracks.size(); 2532 for (size_t i = 0; i < size; i++) { 2533 sp<Track> t = mTracks[i]; 2534 if (t->streamType() == streamType) { 2535 t->invalidate(); 2536 } 2537 } 2538} 2539 2540status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2541{ 2542 int session = chain->sessionId(); 2543 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2544 ? mEffectBuffer : mSinkBuffer); 2545 bool ownsBuffer = false; 2546 2547 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2548 if (session > 0) { 2549 // Only one effect chain can be present in direct output thread and it uses 2550 // the sink buffer as input 2551 if (mType != DIRECT) { 2552 size_t numSamples = mNormalFrameCount * mChannelCount; 2553 buffer = new int16_t[numSamples]; 2554 memset(buffer, 0, numSamples * sizeof(int16_t)); 2555 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2556 ownsBuffer = true; 2557 } 2558 2559 // Attach all tracks with same session ID to this chain. 2560 for (size_t i = 0; i < mTracks.size(); ++i) { 2561 sp<Track> track = mTracks[i]; 2562 if (session == track->sessionId()) { 2563 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2564 buffer); 2565 track->setMainBuffer(buffer); 2566 chain->incTrackCnt(); 2567 } 2568 } 2569 2570 // indicate all active tracks in the chain 2571 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2572 sp<Track> track = mActiveTracks[i].promote(); 2573 if (track == 0) { 2574 continue; 2575 } 2576 if (session == track->sessionId()) { 2577 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2578 chain->incActiveTrackCnt(); 2579 } 2580 } 2581 } 2582 chain->setThread(this); 2583 chain->setInBuffer(buffer, ownsBuffer); 2584 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2585 ? mEffectBuffer : mSinkBuffer)); 2586 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2587 // chains list in order to be processed last as it contains output stage effects 2588 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2589 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2590 // after track specific effects and before output stage 2591 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2592 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2593 // Effect chain for other sessions are inserted at beginning of effect 2594 // chains list to be processed before output mix effects. Relative order between other 2595 // sessions is not important 2596 size_t size = mEffectChains.size(); 2597 size_t i = 0; 2598 for (i = 0; i < size; i++) { 2599 if (mEffectChains[i]->sessionId() < session) { 2600 break; 2601 } 2602 } 2603 mEffectChains.insertAt(chain, i); 2604 checkSuspendOnAddEffectChain_l(chain); 2605 2606 return NO_ERROR; 2607} 2608 2609size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2610{ 2611 int session = chain->sessionId(); 2612 2613 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2614 2615 for (size_t i = 0; i < mEffectChains.size(); i++) { 2616 if (chain == mEffectChains[i]) { 2617 mEffectChains.removeAt(i); 2618 // detach all active tracks from the chain 2619 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2620 sp<Track> track = mActiveTracks[i].promote(); 2621 if (track == 0) { 2622 continue; 2623 } 2624 if (session == track->sessionId()) { 2625 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2626 chain.get(), session); 2627 chain->decActiveTrackCnt(); 2628 } 2629 } 2630 2631 // detach all tracks with same session ID from this chain 2632 for (size_t i = 0; i < mTracks.size(); ++i) { 2633 sp<Track> track = mTracks[i]; 2634 if (session == track->sessionId()) { 2635 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2636 chain->decTrackCnt(); 2637 } 2638 } 2639 break; 2640 } 2641 } 2642 return mEffectChains.size(); 2643} 2644 2645status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2646 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2647{ 2648 Mutex::Autolock _l(mLock); 2649 return attachAuxEffect_l(track, EffectId); 2650} 2651 2652status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2653 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2654{ 2655 status_t status = NO_ERROR; 2656 2657 if (EffectId == 0) { 2658 track->setAuxBuffer(0, NULL); 2659 } else { 2660 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2661 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2662 if (effect != 0) { 2663 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2664 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2665 } else { 2666 status = INVALID_OPERATION; 2667 } 2668 } else { 2669 status = BAD_VALUE; 2670 } 2671 } 2672 return status; 2673} 2674 2675void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2676{ 2677 for (size_t i = 0; i < mTracks.size(); ++i) { 2678 sp<Track> track = mTracks[i]; 2679 if (track->auxEffectId() == effectId) { 2680 attachAuxEffect_l(track, 0); 2681 } 2682 } 2683} 2684 2685bool AudioFlinger::PlaybackThread::threadLoop() 2686{ 2687 Vector< sp<Track> > tracksToRemove; 2688 2689 mStandbyTimeNs = systemTime(); 2690 2691 // MIXER 2692 nsecs_t lastWarning = 0; 2693 2694 // DUPLICATING 2695 // FIXME could this be made local to while loop? 2696 writeFrames = 0; 2697 2698 int lastGeneration = 0; 2699 2700 cacheParameters_l(); 2701 mSleepTimeUs = mIdleSleepTimeUs; 2702 2703 if (mType == MIXER) { 2704 sleepTimeShift = 0; 2705 } 2706 2707 CpuStats cpuStats; 2708 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2709 2710 acquireWakeLock(); 2711 2712 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2713 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2714 // and then that string will be logged at the next convenient opportunity. 2715 const char *logString = NULL; 2716 2717 checkSilentMode_l(); 2718 2719 while (!exitPending()) 2720 { 2721 cpuStats.sample(myName); 2722 2723 Vector< sp<EffectChain> > effectChains; 2724 2725 { // scope for mLock 2726 2727 Mutex::Autolock _l(mLock); 2728 2729 processConfigEvents_l(); 2730 2731 if (logString != NULL) { 2732 mNBLogWriter->logTimestamp(); 2733 mNBLogWriter->log(logString); 2734 logString = NULL; 2735 } 2736 2737 // Gather the framesReleased counters for all active tracks, 2738 // and latch them atomically with the timestamp. 2739 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2740 mLatchD.mFramesReleased.clear(); 2741 size_t size = mActiveTracks.size(); 2742 for (size_t i = 0; i < size; i++) { 2743 sp<Track> t = mActiveTracks[i].promote(); 2744 if (t != 0) { 2745 mLatchD.mFramesReleased.add(t.get(), 2746 t->mAudioTrackServerProxy->framesReleased()); 2747 } 2748 } 2749 if (mLatchDValid) { 2750 mLatchQ = mLatchD; 2751 mLatchDValid = false; 2752 mLatchQValid = true; 2753 } 2754 2755 saveOutputTracks(); 2756 if (mSignalPending) { 2757 // A signal was raised while we were unlocked 2758 mSignalPending = false; 2759 } else if (waitingAsyncCallback_l()) { 2760 if (exitPending()) { 2761 break; 2762 } 2763 bool released = false; 2764 // The following works around a bug in the offload driver. Ideally we would release 2765 // the wake lock every time, but that causes the last offload buffer(s) to be 2766 // dropped while the device is on battery, so we need to hold a wake lock during 2767 // the drain phase. 2768 if (mBytesRemaining && !(mDrainSequence & 1)) { 2769 releaseWakeLock_l(); 2770 released = true; 2771 } 2772 mWakeLockUids.clear(); 2773 mActiveTracksGeneration++; 2774 ALOGV("wait async completion"); 2775 mWaitWorkCV.wait(mLock); 2776 ALOGV("async completion/wake"); 2777 if (released) { 2778 acquireWakeLock_l(); 2779 } 2780 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2781 mSleepTimeUs = 0; 2782 2783 continue; 2784 } 2785 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2786 isSuspended()) { 2787 // put audio hardware into standby after short delay 2788 if (shouldStandby_l()) { 2789 2790 threadLoop_standby(); 2791 2792 mStandby = true; 2793 } 2794 2795 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2796 // we're about to wait, flush the binder command buffer 2797 IPCThreadState::self()->flushCommands(); 2798 2799 clearOutputTracks(); 2800 2801 if (exitPending()) { 2802 break; 2803 } 2804 2805 releaseWakeLock_l(); 2806 mWakeLockUids.clear(); 2807 mActiveTracksGeneration++; 2808 // wait until we have something to do... 2809 ALOGV("%s going to sleep", myName.string()); 2810 mWaitWorkCV.wait(mLock); 2811 ALOGV("%s waking up", myName.string()); 2812 acquireWakeLock_l(); 2813 2814 mMixerStatus = MIXER_IDLE; 2815 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2816 mBytesWritten = 0; 2817 mBytesRemaining = 0; 2818 checkSilentMode_l(); 2819 2820 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2821 mSleepTimeUs = mIdleSleepTimeUs; 2822 if (mType == MIXER) { 2823 sleepTimeShift = 0; 2824 } 2825 2826 continue; 2827 } 2828 } 2829 // mMixerStatusIgnoringFastTracks is also updated internally 2830 mMixerStatus = prepareTracks_l(&tracksToRemove); 2831 2832 // compare with previously applied list 2833 if (lastGeneration != mActiveTracksGeneration) { 2834 // update wakelock 2835 updateWakeLockUids_l(mWakeLockUids); 2836 lastGeneration = mActiveTracksGeneration; 2837 } 2838 2839 // prevent any changes in effect chain list and in each effect chain 2840 // during mixing and effect process as the audio buffers could be deleted 2841 // or modified if an effect is created or deleted 2842 lockEffectChains_l(effectChains); 2843 } // mLock scope ends 2844 2845 if (mBytesRemaining == 0) { 2846 mCurrentWriteLength = 0; 2847 if (mMixerStatus == MIXER_TRACKS_READY) { 2848 // threadLoop_mix() sets mCurrentWriteLength 2849 threadLoop_mix(); 2850 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2851 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2852 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2853 // must be written to HAL 2854 threadLoop_sleepTime(); 2855 if (mSleepTimeUs == 0) { 2856 mCurrentWriteLength = mSinkBufferSize; 2857 } 2858 } 2859 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2860 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2861 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2862 // or mSinkBuffer (if there are no effects). 2863 // 2864 // This is done pre-effects computation; if effects change to 2865 // support higher precision, this needs to move. 2866 // 2867 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2868 // TODO use mSleepTimeUs == 0 as an additional condition. 2869 if (mMixerBufferValid) { 2870 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2871 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2872 2873 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2874 mNormalFrameCount * mChannelCount); 2875 } 2876 2877 mBytesRemaining = mCurrentWriteLength; 2878 if (isSuspended()) { 2879 mSleepTimeUs = suspendSleepTimeUs(); 2880 // simulate write to HAL when suspended 2881 mBytesWritten += mSinkBufferSize; 2882 mBytesRemaining = 0; 2883 } 2884 2885 // only process effects if we're going to write 2886 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 2887 for (size_t i = 0; i < effectChains.size(); i ++) { 2888 effectChains[i]->process_l(); 2889 } 2890 } 2891 } 2892 // Process effect chains for offloaded thread even if no audio 2893 // was read from audio track: process only updates effect state 2894 // and thus does have to be synchronized with audio writes but may have 2895 // to be called while waiting for async write callback 2896 if (mType == OFFLOAD) { 2897 for (size_t i = 0; i < effectChains.size(); i ++) { 2898 effectChains[i]->process_l(); 2899 } 2900 } 2901 2902 // Only if the Effects buffer is enabled and there is data in the 2903 // Effects buffer (buffer valid), we need to 2904 // copy into the sink buffer. 2905 // TODO use mSleepTimeUs == 0 as an additional condition. 2906 if (mEffectBufferValid) { 2907 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2908 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2909 mNormalFrameCount * mChannelCount); 2910 } 2911 2912 // enable changes in effect chain 2913 unlockEffectChains(effectChains); 2914 2915 if (!waitingAsyncCallback()) { 2916 // mSleepTimeUs == 0 means we must write to audio hardware 2917 if (mSleepTimeUs == 0) { 2918 ssize_t ret = 0; 2919 if (mBytesRemaining) { 2920 ret = threadLoop_write(); 2921 if (ret < 0) { 2922 mBytesRemaining = 0; 2923 } else { 2924 mBytesWritten += ret; 2925 mBytesRemaining -= ret; 2926 } 2927 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2928 (mMixerStatus == MIXER_DRAIN_ALL)) { 2929 threadLoop_drain(); 2930 } 2931 if (mType == MIXER && !mStandby) { 2932 // write blocked detection 2933 nsecs_t now = systemTime(); 2934 nsecs_t delta = now - mLastWriteTime; 2935 if (delta > maxPeriod) { 2936 mNumDelayedWrites++; 2937 if ((now - lastWarning) > kWarningThrottleNs) { 2938 ATRACE_NAME("underrun"); 2939 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2940 ns2ms(delta), mNumDelayedWrites, this); 2941 lastWarning = now; 2942 } 2943 } 2944 2945 if (mThreadThrottle 2946 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 2947 && ret > 0) { // we wrote something 2948 // Limit MixerThread data processing to no more than twice the 2949 // expected processing rate. 2950 // 2951 // This helps prevent underruns with NuPlayer and other applications 2952 // which may set up buffers that are close to the minimum size, or use 2953 // deep buffers, and rely on a double-buffering sleep strategy to fill. 2954 // 2955 // The throttle smooths out sudden large data drains from the device, 2956 // e.g. when it comes out of standby, which often causes problems with 2957 // (1) mixer threads without a fast mixer (which has its own warm-up) 2958 // (2) minimum buffer sized tracks (even if the track is full, 2959 // the app won't fill fast enough to handle the sudden draw). 2960 2961 const int32_t deltaMs = delta / 1000000; 2962 const int32_t throttleMs = mHalfBufferMs - deltaMs; 2963 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 2964 usleep(throttleMs * 1000); 2965 // notify of throttle start on verbose log 2966 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 2967 "mixer(%p) throttle begin:" 2968 " ret(%zd) deltaMs(%d) requires sleep %d ms", 2969 this, ret, deltaMs, throttleMs); 2970 mThreadThrottleTimeMs += throttleMs; 2971 } else { 2972 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 2973 if (diff > 0) { 2974 // notify of throttle end on debug log 2975 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 2976 mThreadThrottleEndMs = mThreadThrottleTimeMs; 2977 } 2978 } 2979 } 2980 } 2981 2982 } else { 2983 ATRACE_BEGIN("sleep"); 2984 usleep(mSleepTimeUs); 2985 ATRACE_END(); 2986 } 2987 } 2988 2989 // Finally let go of removed track(s), without the lock held 2990 // since we can't guarantee the destructors won't acquire that 2991 // same lock. This will also mutate and push a new fast mixer state. 2992 threadLoop_removeTracks(tracksToRemove); 2993 tracksToRemove.clear(); 2994 2995 // FIXME I don't understand the need for this here; 2996 // it was in the original code but maybe the 2997 // assignment in saveOutputTracks() makes this unnecessary? 2998 clearOutputTracks(); 2999 3000 // Effect chains will be actually deleted here if they were removed from 3001 // mEffectChains list during mixing or effects processing 3002 effectChains.clear(); 3003 3004 // FIXME Note that the above .clear() is no longer necessary since effectChains 3005 // is now local to this block, but will keep it for now (at least until merge done). 3006 } 3007 3008 threadLoop_exit(); 3009 3010 if (!mStandby) { 3011 threadLoop_standby(); 3012 mStandby = true; 3013 } 3014 3015 releaseWakeLock(); 3016 mWakeLockUids.clear(); 3017 mActiveTracksGeneration++; 3018 3019 ALOGV("Thread %p type %d exiting", this, mType); 3020 return false; 3021} 3022 3023// removeTracks_l() must be called with ThreadBase::mLock held 3024void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3025{ 3026 size_t count = tracksToRemove.size(); 3027 if (count > 0) { 3028 for (size_t i=0 ; i<count ; i++) { 3029 const sp<Track>& track = tracksToRemove.itemAt(i); 3030 mActiveTracks.remove(track); 3031 mWakeLockUids.remove(track->uid()); 3032 mActiveTracksGeneration++; 3033 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3034 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3035 if (chain != 0) { 3036 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3037 track->sessionId()); 3038 chain->decActiveTrackCnt(); 3039 } 3040 if (track->isTerminated()) { 3041 removeTrack_l(track); 3042 } 3043 } 3044 } 3045 3046} 3047 3048status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3049{ 3050 if (mNormalSink != 0) { 3051 return mNormalSink->getTimestamp(timestamp); 3052 } 3053 if ((mType == OFFLOAD || mType == DIRECT) 3054 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3055 uint64_t position64; 3056 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3057 if (ret == 0) { 3058 timestamp.mPosition = (uint32_t)position64; 3059 return NO_ERROR; 3060 } 3061 } 3062 return INVALID_OPERATION; 3063} 3064 3065status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3066 audio_patch_handle_t *handle) 3067{ 3068 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3069 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3070 if (mFastMixer != 0) { 3071 FastMixerStateQueue *sq = mFastMixer->sq(); 3072 FastMixerState *state = sq->begin(); 3073 if (!(state->mCommand & FastMixerState::IDLE)) { 3074 previousCommand = state->mCommand; 3075 state->mCommand = FastMixerState::HOT_IDLE; 3076 sq->end(); 3077 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3078 } else { 3079 sq->end(false /*didModify*/); 3080 } 3081 } 3082 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3083 3084 if (!(previousCommand & FastMixerState::IDLE)) { 3085 ALOG_ASSERT(mFastMixer != 0); 3086 FastMixerStateQueue *sq = mFastMixer->sq(); 3087 FastMixerState *state = sq->begin(); 3088 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3089 state->mCommand = previousCommand; 3090 sq->end(); 3091 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3092 } 3093 3094 return status; 3095} 3096 3097status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3098 audio_patch_handle_t *handle) 3099{ 3100 status_t status = NO_ERROR; 3101 3102 // store new device and send to effects 3103 audio_devices_t type = AUDIO_DEVICE_NONE; 3104 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3105 type |= patch->sinks[i].ext.device.type; 3106 } 3107 3108#ifdef ADD_BATTERY_DATA 3109 // when changing the audio output device, call addBatteryData to notify 3110 // the change 3111 if (mOutDevice != type) { 3112 uint32_t params = 0; 3113 // check whether speaker is on 3114 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3115 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3116 } 3117 3118 audio_devices_t deviceWithoutSpeaker 3119 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3120 // check if any other device (except speaker) is on 3121 if (type & deviceWithoutSpeaker) { 3122 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3123 } 3124 3125 if (params != 0) { 3126 addBatteryData(params); 3127 } 3128 } 3129#endif 3130 3131 for (size_t i = 0; i < mEffectChains.size(); i++) { 3132 mEffectChains[i]->setDevice_l(type); 3133 } 3134 mOutDevice = type; 3135 mPatch = *patch; 3136 3137 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3138 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3139 status = hwDevice->create_audio_patch(hwDevice, 3140 patch->num_sources, 3141 patch->sources, 3142 patch->num_sinks, 3143 patch->sinks, 3144 handle); 3145 } else { 3146 char *address; 3147 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3148 //FIXME: we only support address on first sink with HAL version < 3.0 3149 address = audio_device_address_to_parameter( 3150 patch->sinks[0].ext.device.type, 3151 patch->sinks[0].ext.device.address); 3152 } else { 3153 address = (char *)calloc(1, 1); 3154 } 3155 AudioParameter param = AudioParameter(String8(address)); 3156 free(address); 3157 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3158 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3159 param.toString().string()); 3160 *handle = AUDIO_PATCH_HANDLE_NONE; 3161 } 3162 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3163 return status; 3164} 3165 3166status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3167{ 3168 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3169 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3170 if (mFastMixer != 0) { 3171 FastMixerStateQueue *sq = mFastMixer->sq(); 3172 FastMixerState *state = sq->begin(); 3173 if (!(state->mCommand & FastMixerState::IDLE)) { 3174 previousCommand = state->mCommand; 3175 state->mCommand = FastMixerState::HOT_IDLE; 3176 sq->end(); 3177 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3178 } else { 3179 sq->end(false /*didModify*/); 3180 } 3181 } 3182 3183 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3184 3185 if (!(previousCommand & FastMixerState::IDLE)) { 3186 ALOG_ASSERT(mFastMixer != 0); 3187 FastMixerStateQueue *sq = mFastMixer->sq(); 3188 FastMixerState *state = sq->begin(); 3189 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3190 state->mCommand = previousCommand; 3191 sq->end(); 3192 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3193 } 3194 3195 return status; 3196} 3197 3198status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3199{ 3200 status_t status = NO_ERROR; 3201 3202 mOutDevice = AUDIO_DEVICE_NONE; 3203 3204 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3205 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3206 status = hwDevice->release_audio_patch(hwDevice, handle); 3207 } else { 3208 AudioParameter param; 3209 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3210 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3211 param.toString().string()); 3212 } 3213 return status; 3214} 3215 3216void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3217{ 3218 Mutex::Autolock _l(mLock); 3219 mTracks.add(track); 3220} 3221 3222void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3223{ 3224 Mutex::Autolock _l(mLock); 3225 destroyTrack_l(track); 3226} 3227 3228void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3229{ 3230 ThreadBase::getAudioPortConfig(config); 3231 config->role = AUDIO_PORT_ROLE_SOURCE; 3232 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3233 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3234} 3235 3236// ---------------------------------------------------------------------------- 3237 3238AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3239 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3240 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3241 // mAudioMixer below 3242 // mFastMixer below 3243 mFastMixerFutex(0) 3244 // mOutputSink below 3245 // mPipeSink below 3246 // mNormalSink below 3247{ 3248 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3249 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3250 "mFrameCount=%d, mNormalFrameCount=%d", 3251 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3252 mNormalFrameCount); 3253 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3254 3255 if (type == DUPLICATING) { 3256 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3257 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3258 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3259 return; 3260 } 3261 // create an NBAIO sink for the HAL output stream, and negotiate 3262 mOutputSink = new AudioStreamOutSink(output->stream); 3263 size_t numCounterOffers = 0; 3264 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3265 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3266 ALOG_ASSERT(index == 0); 3267 3268 // initialize fast mixer depending on configuration 3269 bool initFastMixer; 3270 switch (kUseFastMixer) { 3271 case FastMixer_Never: 3272 initFastMixer = false; 3273 break; 3274 case FastMixer_Always: 3275 initFastMixer = true; 3276 break; 3277 case FastMixer_Static: 3278 case FastMixer_Dynamic: 3279 initFastMixer = mFrameCount < mNormalFrameCount; 3280 break; 3281 } 3282 if (initFastMixer) { 3283 audio_format_t fastMixerFormat; 3284 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3285 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3286 } else { 3287 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3288 } 3289 if (mFormat != fastMixerFormat) { 3290 // change our Sink format to accept our intermediate precision 3291 mFormat = fastMixerFormat; 3292 free(mSinkBuffer); 3293 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3294 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3295 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3296 } 3297 3298 // create a MonoPipe to connect our submix to FastMixer 3299 NBAIO_Format format = mOutputSink->format(); 3300 NBAIO_Format origformat = format; 3301 // adjust format to match that of the Fast Mixer 3302 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3303 format.mFormat = fastMixerFormat; 3304 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3305 3306 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3307 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3308 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3309 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3310 const NBAIO_Format offers[1] = {format}; 3311 size_t numCounterOffers = 0; 3312 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3313 ALOG_ASSERT(index == 0); 3314 monoPipe->setAvgFrames((mScreenState & 1) ? 3315 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3316 mPipeSink = monoPipe; 3317 3318#ifdef TEE_SINK 3319 if (mTeeSinkOutputEnabled) { 3320 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3321 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3322 const NBAIO_Format offers2[1] = {origformat}; 3323 numCounterOffers = 0; 3324 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3325 ALOG_ASSERT(index == 0); 3326 mTeeSink = teeSink; 3327 PipeReader *teeSource = new PipeReader(*teeSink); 3328 numCounterOffers = 0; 3329 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3330 ALOG_ASSERT(index == 0); 3331 mTeeSource = teeSource; 3332 } 3333#endif 3334 3335 // create fast mixer and configure it initially with just one fast track for our submix 3336 mFastMixer = new FastMixer(); 3337 FastMixerStateQueue *sq = mFastMixer->sq(); 3338#ifdef STATE_QUEUE_DUMP 3339 sq->setObserverDump(&mStateQueueObserverDump); 3340 sq->setMutatorDump(&mStateQueueMutatorDump); 3341#endif 3342 FastMixerState *state = sq->begin(); 3343 FastTrack *fastTrack = &state->mFastTracks[0]; 3344 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3345 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3346 fastTrack->mVolumeProvider = NULL; 3347 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3348 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3349 fastTrack->mGeneration++; 3350 state->mFastTracksGen++; 3351 state->mTrackMask = 1; 3352 // fast mixer will use the HAL output sink 3353 state->mOutputSink = mOutputSink.get(); 3354 state->mOutputSinkGen++; 3355 state->mFrameCount = mFrameCount; 3356 state->mCommand = FastMixerState::COLD_IDLE; 3357 // already done in constructor initialization list 3358 //mFastMixerFutex = 0; 3359 state->mColdFutexAddr = &mFastMixerFutex; 3360 state->mColdGen++; 3361 state->mDumpState = &mFastMixerDumpState; 3362#ifdef TEE_SINK 3363 state->mTeeSink = mTeeSink.get(); 3364#endif 3365 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3366 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3367 sq->end(); 3368 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3369 3370 // start the fast mixer 3371 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3372 pid_t tid = mFastMixer->getTid(); 3373 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3374 3375#ifdef AUDIO_WATCHDOG 3376 // create and start the watchdog 3377 mAudioWatchdog = new AudioWatchdog(); 3378 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3379 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3380 tid = mAudioWatchdog->getTid(); 3381 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3382#endif 3383 3384 } 3385 3386 switch (kUseFastMixer) { 3387 case FastMixer_Never: 3388 case FastMixer_Dynamic: 3389 mNormalSink = mOutputSink; 3390 break; 3391 case FastMixer_Always: 3392 mNormalSink = mPipeSink; 3393 break; 3394 case FastMixer_Static: 3395 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3396 break; 3397 } 3398} 3399 3400AudioFlinger::MixerThread::~MixerThread() 3401{ 3402 if (mFastMixer != 0) { 3403 FastMixerStateQueue *sq = mFastMixer->sq(); 3404 FastMixerState *state = sq->begin(); 3405 if (state->mCommand == FastMixerState::COLD_IDLE) { 3406 int32_t old = android_atomic_inc(&mFastMixerFutex); 3407 if (old == -1) { 3408 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3409 } 3410 } 3411 state->mCommand = FastMixerState::EXIT; 3412 sq->end(); 3413 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3414 mFastMixer->join(); 3415 // Though the fast mixer thread has exited, it's state queue is still valid. 3416 // We'll use that extract the final state which contains one remaining fast track 3417 // corresponding to our sub-mix. 3418 state = sq->begin(); 3419 ALOG_ASSERT(state->mTrackMask == 1); 3420 FastTrack *fastTrack = &state->mFastTracks[0]; 3421 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3422 delete fastTrack->mBufferProvider; 3423 sq->end(false /*didModify*/); 3424 mFastMixer.clear(); 3425#ifdef AUDIO_WATCHDOG 3426 if (mAudioWatchdog != 0) { 3427 mAudioWatchdog->requestExit(); 3428 mAudioWatchdog->requestExitAndWait(); 3429 mAudioWatchdog.clear(); 3430 } 3431#endif 3432 } 3433 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3434 delete mAudioMixer; 3435} 3436 3437 3438uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3439{ 3440 if (mFastMixer != 0) { 3441 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3442 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3443 } 3444 return latency; 3445} 3446 3447 3448void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3449{ 3450 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3451} 3452 3453ssize_t AudioFlinger::MixerThread::threadLoop_write() 3454{ 3455 // FIXME we should only do one push per cycle; confirm this is true 3456 // Start the fast mixer if it's not already running 3457 if (mFastMixer != 0) { 3458 FastMixerStateQueue *sq = mFastMixer->sq(); 3459 FastMixerState *state = sq->begin(); 3460 if (state->mCommand != FastMixerState::MIX_WRITE && 3461 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3462 if (state->mCommand == FastMixerState::COLD_IDLE) { 3463 int32_t old = android_atomic_inc(&mFastMixerFutex); 3464 if (old == -1) { 3465 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3466 } 3467#ifdef AUDIO_WATCHDOG 3468 if (mAudioWatchdog != 0) { 3469 mAudioWatchdog->resume(); 3470 } 3471#endif 3472 } 3473 state->mCommand = FastMixerState::MIX_WRITE; 3474#ifdef FAST_THREAD_STATISTICS 3475 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3476 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3477#endif 3478 sq->end(); 3479 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3480 if (kUseFastMixer == FastMixer_Dynamic) { 3481 mNormalSink = mPipeSink; 3482 } 3483 } else { 3484 sq->end(false /*didModify*/); 3485 } 3486 } 3487 return PlaybackThread::threadLoop_write(); 3488} 3489 3490void AudioFlinger::MixerThread::threadLoop_standby() 3491{ 3492 // Idle the fast mixer if it's currently running 3493 if (mFastMixer != 0) { 3494 FastMixerStateQueue *sq = mFastMixer->sq(); 3495 FastMixerState *state = sq->begin(); 3496 if (!(state->mCommand & FastMixerState::IDLE)) { 3497 state->mCommand = FastMixerState::COLD_IDLE; 3498 state->mColdFutexAddr = &mFastMixerFutex; 3499 state->mColdGen++; 3500 mFastMixerFutex = 0; 3501 sq->end(); 3502 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3503 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3504 if (kUseFastMixer == FastMixer_Dynamic) { 3505 mNormalSink = mOutputSink; 3506 } 3507#ifdef AUDIO_WATCHDOG 3508 if (mAudioWatchdog != 0) { 3509 mAudioWatchdog->pause(); 3510 } 3511#endif 3512 } else { 3513 sq->end(false /*didModify*/); 3514 } 3515 } 3516 PlaybackThread::threadLoop_standby(); 3517} 3518 3519bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3520{ 3521 return false; 3522} 3523 3524bool AudioFlinger::PlaybackThread::shouldStandby_l() 3525{ 3526 return !mStandby; 3527} 3528 3529bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3530{ 3531 Mutex::Autolock _l(mLock); 3532 return waitingAsyncCallback_l(); 3533} 3534 3535// shared by MIXER and DIRECT, overridden by DUPLICATING 3536void AudioFlinger::PlaybackThread::threadLoop_standby() 3537{ 3538 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3539 mOutput->standby(); 3540 if (mUseAsyncWrite != 0) { 3541 // discard any pending drain or write ack by incrementing sequence 3542 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3543 mDrainSequence = (mDrainSequence + 2) & ~1; 3544 ALOG_ASSERT(mCallbackThread != 0); 3545 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3546 mCallbackThread->setDraining(mDrainSequence); 3547 } 3548 mHwPaused = false; 3549} 3550 3551void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3552{ 3553 ALOGV("signal playback thread"); 3554 broadcast_l(); 3555} 3556 3557void AudioFlinger::MixerThread::threadLoop_mix() 3558{ 3559 // obtain the presentation timestamp of the next output buffer 3560 int64_t pts; 3561 status_t status = INVALID_OPERATION; 3562 3563 if (mNormalSink != 0) { 3564 status = mNormalSink->getNextWriteTimestamp(&pts); 3565 } else { 3566 status = mOutputSink->getNextWriteTimestamp(&pts); 3567 } 3568 3569 if (status != NO_ERROR) { 3570 pts = AudioBufferProvider::kInvalidPTS; 3571 } 3572 3573 // mix buffers... 3574 mAudioMixer->process(pts); 3575 mCurrentWriteLength = mSinkBufferSize; 3576 // increase sleep time progressively when application underrun condition clears. 3577 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3578 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3579 // such that we would underrun the audio HAL. 3580 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3581 sleepTimeShift--; 3582 } 3583 mSleepTimeUs = 0; 3584 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3585 //TODO: delay standby when effects have a tail 3586 3587} 3588 3589void AudioFlinger::MixerThread::threadLoop_sleepTime() 3590{ 3591 // If no tracks are ready, sleep once for the duration of an output 3592 // buffer size, then write 0s to the output 3593 if (mSleepTimeUs == 0) { 3594 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3595 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3596 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3597 mSleepTimeUs = kMinThreadSleepTimeUs; 3598 } 3599 // reduce sleep time in case of consecutive application underruns to avoid 3600 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3601 // duration we would end up writing less data than needed by the audio HAL if 3602 // the condition persists. 3603 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3604 sleepTimeShift++; 3605 } 3606 } else { 3607 mSleepTimeUs = mIdleSleepTimeUs; 3608 } 3609 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3610 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3611 // before effects processing or output. 3612 if (mMixerBufferValid) { 3613 memset(mMixerBuffer, 0, mMixerBufferSize); 3614 } else { 3615 memset(mSinkBuffer, 0, mSinkBufferSize); 3616 } 3617 mSleepTimeUs = 0; 3618 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3619 "anticipated start"); 3620 } 3621 // TODO add standby time extension fct of effect tail 3622} 3623 3624// prepareTracks_l() must be called with ThreadBase::mLock held 3625AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3626 Vector< sp<Track> > *tracksToRemove) 3627{ 3628 3629 mixer_state mixerStatus = MIXER_IDLE; 3630 // find out which tracks need to be processed 3631 size_t count = mActiveTracks.size(); 3632 size_t mixedTracks = 0; 3633 size_t tracksWithEffect = 0; 3634 // counts only _active_ fast tracks 3635 size_t fastTracks = 0; 3636 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3637 3638 float masterVolume = mMasterVolume; 3639 bool masterMute = mMasterMute; 3640 3641 if (masterMute) { 3642 masterVolume = 0; 3643 } 3644 // Delegate master volume control to effect in output mix effect chain if needed 3645 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3646 if (chain != 0) { 3647 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3648 chain->setVolume_l(&v, &v); 3649 masterVolume = (float)((v + (1 << 23)) >> 24); 3650 chain.clear(); 3651 } 3652 3653 // prepare a new state to push 3654 FastMixerStateQueue *sq = NULL; 3655 FastMixerState *state = NULL; 3656 bool didModify = false; 3657 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3658 if (mFastMixer != 0) { 3659 sq = mFastMixer->sq(); 3660 state = sq->begin(); 3661 } 3662 3663 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3664 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3665 3666 for (size_t i=0 ; i<count ; i++) { 3667 const sp<Track> t = mActiveTracks[i].promote(); 3668 if (t == 0) { 3669 continue; 3670 } 3671 3672 // this const just means the local variable doesn't change 3673 Track* const track = t.get(); 3674 3675 // process fast tracks 3676 if (track->isFastTrack()) { 3677 3678 // It's theoretically possible (though unlikely) for a fast track to be created 3679 // and then removed within the same normal mix cycle. This is not a problem, as 3680 // the track never becomes active so it's fast mixer slot is never touched. 3681 // The converse, of removing an (active) track and then creating a new track 3682 // at the identical fast mixer slot within the same normal mix cycle, 3683 // is impossible because the slot isn't marked available until the end of each cycle. 3684 int j = track->mFastIndex; 3685 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3686 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3687 FastTrack *fastTrack = &state->mFastTracks[j]; 3688 3689 // Determine whether the track is currently in underrun condition, 3690 // and whether it had a recent underrun. 3691 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3692 FastTrackUnderruns underruns = ftDump->mUnderruns; 3693 uint32_t recentFull = (underruns.mBitFields.mFull - 3694 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3695 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3696 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3697 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3698 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3699 uint32_t recentUnderruns = recentPartial + recentEmpty; 3700 track->mObservedUnderruns = underruns; 3701 // don't count underruns that occur while stopping or pausing 3702 // or stopped which can occur when flush() is called while active 3703 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3704 recentUnderruns > 0) { 3705 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3706 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3707 } 3708 3709 // This is similar to the state machine for normal tracks, 3710 // with a few modifications for fast tracks. 3711 bool isActive = true; 3712 switch (track->mState) { 3713 case TrackBase::STOPPING_1: 3714 // track stays active in STOPPING_1 state until first underrun 3715 if (recentUnderruns > 0 || track->isTerminated()) { 3716 track->mState = TrackBase::STOPPING_2; 3717 } 3718 break; 3719 case TrackBase::PAUSING: 3720 // ramp down is not yet implemented 3721 track->setPaused(); 3722 break; 3723 case TrackBase::RESUMING: 3724 // ramp up is not yet implemented 3725 track->mState = TrackBase::ACTIVE; 3726 break; 3727 case TrackBase::ACTIVE: 3728 if (recentFull > 0 || recentPartial > 0) { 3729 // track has provided at least some frames recently: reset retry count 3730 track->mRetryCount = kMaxTrackRetries; 3731 } 3732 if (recentUnderruns == 0) { 3733 // no recent underruns: stay active 3734 break; 3735 } 3736 // there has recently been an underrun of some kind 3737 if (track->sharedBuffer() == 0) { 3738 // were any of the recent underruns "empty" (no frames available)? 3739 if (recentEmpty == 0) { 3740 // no, then ignore the partial underruns as they are allowed indefinitely 3741 break; 3742 } 3743 // there has recently been an "empty" underrun: decrement the retry counter 3744 if (--(track->mRetryCount) > 0) { 3745 break; 3746 } 3747 // indicate to client process that the track was disabled because of underrun; 3748 // it will then automatically call start() when data is available 3749 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3750 // remove from active list, but state remains ACTIVE [confusing but true] 3751 isActive = false; 3752 break; 3753 } 3754 // fall through 3755 case TrackBase::STOPPING_2: 3756 case TrackBase::PAUSED: 3757 case TrackBase::STOPPED: 3758 case TrackBase::FLUSHED: // flush() while active 3759 // Check for presentation complete if track is inactive 3760 // We have consumed all the buffers of this track. 3761 // This would be incomplete if we auto-paused on underrun 3762 { 3763 size_t audioHALFrames = 3764 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3765 size_t framesWritten = mBytesWritten / mFrameSize; 3766 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3767 // track stays in active list until presentation is complete 3768 break; 3769 } 3770 } 3771 if (track->isStopping_2()) { 3772 track->mState = TrackBase::STOPPED; 3773 } 3774 if (track->isStopped()) { 3775 // Can't reset directly, as fast mixer is still polling this track 3776 // track->reset(); 3777 // So instead mark this track as needing to be reset after push with ack 3778 resetMask |= 1 << i; 3779 } 3780 isActive = false; 3781 break; 3782 case TrackBase::IDLE: 3783 default: 3784 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3785 } 3786 3787 if (isActive) { 3788 // was it previously inactive? 3789 if (!(state->mTrackMask & (1 << j))) { 3790 ExtendedAudioBufferProvider *eabp = track; 3791 VolumeProvider *vp = track; 3792 fastTrack->mBufferProvider = eabp; 3793 fastTrack->mVolumeProvider = vp; 3794 fastTrack->mChannelMask = track->mChannelMask; 3795 fastTrack->mFormat = track->mFormat; 3796 fastTrack->mGeneration++; 3797 state->mTrackMask |= 1 << j; 3798 didModify = true; 3799 // no acknowledgement required for newly active tracks 3800 } 3801 // cache the combined master volume and stream type volume for fast mixer; this 3802 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3803 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3804 ++fastTracks; 3805 } else { 3806 // was it previously active? 3807 if (state->mTrackMask & (1 << j)) { 3808 fastTrack->mBufferProvider = NULL; 3809 fastTrack->mGeneration++; 3810 state->mTrackMask &= ~(1 << j); 3811 didModify = true; 3812 // If any fast tracks were removed, we must wait for acknowledgement 3813 // because we're about to decrement the last sp<> on those tracks. 3814 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3815 } else { 3816 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3817 } 3818 tracksToRemove->add(track); 3819 // Avoids a misleading display in dumpsys 3820 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3821 } 3822 continue; 3823 } 3824 3825 { // local variable scope to avoid goto warning 3826 3827 audio_track_cblk_t* cblk = track->cblk(); 3828 3829 // The first time a track is added we wait 3830 // for all its buffers to be filled before processing it 3831 int name = track->name(); 3832 // make sure that we have enough frames to mix one full buffer. 3833 // enforce this condition only once to enable draining the buffer in case the client 3834 // app does not call stop() and relies on underrun to stop: 3835 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3836 // during last round 3837 size_t desiredFrames; 3838 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3839 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3840 3841 desiredFrames = sourceFramesNeededWithTimestretch( 3842 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3843 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3844 // add frames already consumed but not yet released by the resampler 3845 // because mAudioTrackServerProxy->framesReady() will include these frames 3846 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3847 3848 uint32_t minFrames = 1; 3849 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3850 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3851 minFrames = desiredFrames; 3852 } 3853 3854 size_t framesReady = track->framesReady(); 3855 if (ATRACE_ENABLED()) { 3856 // I wish we had formatted trace names 3857 char traceName[16]; 3858 strcpy(traceName, "nRdy"); 3859 int name = track->name(); 3860 if (AudioMixer::TRACK0 <= name && 3861 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3862 name -= AudioMixer::TRACK0; 3863 traceName[4] = (name / 10) + '0'; 3864 traceName[5] = (name % 10) + '0'; 3865 } else { 3866 traceName[4] = '?'; 3867 traceName[5] = '?'; 3868 } 3869 traceName[6] = '\0'; 3870 ATRACE_INT(traceName, framesReady); 3871 } 3872 if ((framesReady >= minFrames) && track->isReady() && 3873 !track->isPaused() && !track->isTerminated()) 3874 { 3875 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3876 3877 mixedTracks++; 3878 3879 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3880 // there is an effect chain connected to the track 3881 chain.clear(); 3882 if (track->mainBuffer() != mSinkBuffer && 3883 track->mainBuffer() != mMixerBuffer) { 3884 if (mEffectBufferEnabled) { 3885 mEffectBufferValid = true; // Later can set directly. 3886 } 3887 chain = getEffectChain_l(track->sessionId()); 3888 // Delegate volume control to effect in track effect chain if needed 3889 if (chain != 0) { 3890 tracksWithEffect++; 3891 } else { 3892 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3893 "session %d", 3894 name, track->sessionId()); 3895 } 3896 } 3897 3898 3899 int param = AudioMixer::VOLUME; 3900 if (track->mFillingUpStatus == Track::FS_FILLED) { 3901 // no ramp for the first volume setting 3902 track->mFillingUpStatus = Track::FS_ACTIVE; 3903 if (track->mState == TrackBase::RESUMING) { 3904 track->mState = TrackBase::ACTIVE; 3905 param = AudioMixer::RAMP_VOLUME; 3906 } 3907 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3908 // FIXME should not make a decision based on mServer 3909 } else if (cblk->mServer != 0) { 3910 // If the track is stopped before the first frame was mixed, 3911 // do not apply ramp 3912 param = AudioMixer::RAMP_VOLUME; 3913 } 3914 3915 // compute volume for this track 3916 uint32_t vl, vr; // in U8.24 integer format 3917 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3918 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3919 vl = vr = 0; 3920 vlf = vrf = vaf = 0.; 3921 if (track->isPausing()) { 3922 track->setPaused(); 3923 } 3924 } else { 3925 3926 // read original volumes with volume control 3927 float typeVolume = mStreamTypes[track->streamType()].volume; 3928 float v = masterVolume * typeVolume; 3929 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3930 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3931 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3932 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3933 // track volumes come from shared memory, so can't be trusted and must be clamped 3934 if (vlf > GAIN_FLOAT_UNITY) { 3935 ALOGV("Track left volume out of range: %.3g", vlf); 3936 vlf = GAIN_FLOAT_UNITY; 3937 } 3938 if (vrf > GAIN_FLOAT_UNITY) { 3939 ALOGV("Track right volume out of range: %.3g", vrf); 3940 vrf = GAIN_FLOAT_UNITY; 3941 } 3942 // now apply the master volume and stream type volume 3943 vlf *= v; 3944 vrf *= v; 3945 // assuming master volume and stream type volume each go up to 1.0, 3946 // then derive vl and vr as U8.24 versions for the effect chain 3947 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3948 vl = (uint32_t) (scaleto8_24 * vlf); 3949 vr = (uint32_t) (scaleto8_24 * vrf); 3950 // vl and vr are now in U8.24 format 3951 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3952 // send level comes from shared memory and so may be corrupt 3953 if (sendLevel > MAX_GAIN_INT) { 3954 ALOGV("Track send level out of range: %04X", sendLevel); 3955 sendLevel = MAX_GAIN_INT; 3956 } 3957 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3958 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3959 } 3960 3961 // Delegate volume control to effect in track effect chain if needed 3962 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3963 // Do not ramp volume if volume is controlled by effect 3964 param = AudioMixer::VOLUME; 3965 // Update remaining floating point volume levels 3966 vlf = (float)vl / (1 << 24); 3967 vrf = (float)vr / (1 << 24); 3968 track->mHasVolumeController = true; 3969 } else { 3970 // force no volume ramp when volume controller was just disabled or removed 3971 // from effect chain to avoid volume spike 3972 if (track->mHasVolumeController) { 3973 param = AudioMixer::VOLUME; 3974 } 3975 track->mHasVolumeController = false; 3976 } 3977 3978 // XXX: these things DON'T need to be done each time 3979 mAudioMixer->setBufferProvider(name, track); 3980 mAudioMixer->enable(name); 3981 3982 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3983 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3984 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3985 mAudioMixer->setParameter( 3986 name, 3987 AudioMixer::TRACK, 3988 AudioMixer::FORMAT, (void *)track->format()); 3989 mAudioMixer->setParameter( 3990 name, 3991 AudioMixer::TRACK, 3992 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3993 mAudioMixer->setParameter( 3994 name, 3995 AudioMixer::TRACK, 3996 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3997 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3998 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3999 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4000 if (reqSampleRate == 0) { 4001 reqSampleRate = mSampleRate; 4002 } else if (reqSampleRate > maxSampleRate) { 4003 reqSampleRate = maxSampleRate; 4004 } 4005 mAudioMixer->setParameter( 4006 name, 4007 AudioMixer::RESAMPLE, 4008 AudioMixer::SAMPLE_RATE, 4009 (void *)(uintptr_t)reqSampleRate); 4010 4011 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4012 mAudioMixer->setParameter( 4013 name, 4014 AudioMixer::TIMESTRETCH, 4015 AudioMixer::PLAYBACK_RATE, 4016 &playbackRate); 4017 4018 /* 4019 * Select the appropriate output buffer for the track. 4020 * 4021 * Tracks with effects go into their own effects chain buffer 4022 * and from there into either mEffectBuffer or mSinkBuffer. 4023 * 4024 * Other tracks can use mMixerBuffer for higher precision 4025 * channel accumulation. If this buffer is enabled 4026 * (mMixerBufferEnabled true), then selected tracks will accumulate 4027 * into it. 4028 * 4029 */ 4030 if (mMixerBufferEnabled 4031 && (track->mainBuffer() == mSinkBuffer 4032 || track->mainBuffer() == mMixerBuffer)) { 4033 mAudioMixer->setParameter( 4034 name, 4035 AudioMixer::TRACK, 4036 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4037 mAudioMixer->setParameter( 4038 name, 4039 AudioMixer::TRACK, 4040 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4041 // TODO: override track->mainBuffer()? 4042 mMixerBufferValid = true; 4043 } else { 4044 mAudioMixer->setParameter( 4045 name, 4046 AudioMixer::TRACK, 4047 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4048 mAudioMixer->setParameter( 4049 name, 4050 AudioMixer::TRACK, 4051 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4052 } 4053 mAudioMixer->setParameter( 4054 name, 4055 AudioMixer::TRACK, 4056 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4057 4058 // reset retry count 4059 track->mRetryCount = kMaxTrackRetries; 4060 4061 // If one track is ready, set the mixer ready if: 4062 // - the mixer was not ready during previous round OR 4063 // - no other track is not ready 4064 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4065 mixerStatus != MIXER_TRACKS_ENABLED) { 4066 mixerStatus = MIXER_TRACKS_READY; 4067 } 4068 } else { 4069 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4070 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4071 track, framesReady, desiredFrames); 4072 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4073 } 4074 // clear effect chain input buffer if an active track underruns to avoid sending 4075 // previous audio buffer again to effects 4076 chain = getEffectChain_l(track->sessionId()); 4077 if (chain != 0) { 4078 chain->clearInputBuffer(); 4079 } 4080 4081 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4082 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4083 track->isStopped() || track->isPaused()) { 4084 // We have consumed all the buffers of this track. 4085 // Remove it from the list of active tracks. 4086 // TODO: use actual buffer filling status instead of latency when available from 4087 // audio HAL 4088 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4089 size_t framesWritten = mBytesWritten / mFrameSize; 4090 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4091 if (track->isStopped()) { 4092 track->reset(); 4093 } 4094 tracksToRemove->add(track); 4095 } 4096 } else { 4097 // No buffers for this track. Give it a few chances to 4098 // fill a buffer, then remove it from active list. 4099 if (--(track->mRetryCount) <= 0) { 4100 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4101 tracksToRemove->add(track); 4102 // indicate to client process that the track was disabled because of underrun; 4103 // it will then automatically call start() when data is available 4104 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4105 // If one track is not ready, mark the mixer also not ready if: 4106 // - the mixer was ready during previous round OR 4107 // - no other track is ready 4108 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4109 mixerStatus != MIXER_TRACKS_READY) { 4110 mixerStatus = MIXER_TRACKS_ENABLED; 4111 } 4112 } 4113 mAudioMixer->disable(name); 4114 } 4115 4116 } // local variable scope to avoid goto warning 4117track_is_ready: ; 4118 4119 } 4120 4121 // Push the new FastMixer state if necessary 4122 bool pauseAudioWatchdog = false; 4123 if (didModify) { 4124 state->mFastTracksGen++; 4125 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4126 if (kUseFastMixer == FastMixer_Dynamic && 4127 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4128 state->mCommand = FastMixerState::COLD_IDLE; 4129 state->mColdFutexAddr = &mFastMixerFutex; 4130 state->mColdGen++; 4131 mFastMixerFutex = 0; 4132 if (kUseFastMixer == FastMixer_Dynamic) { 4133 mNormalSink = mOutputSink; 4134 } 4135 // If we go into cold idle, need to wait for acknowledgement 4136 // so that fast mixer stops doing I/O. 4137 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4138 pauseAudioWatchdog = true; 4139 } 4140 } 4141 if (sq != NULL) { 4142 sq->end(didModify); 4143 sq->push(block); 4144 } 4145#ifdef AUDIO_WATCHDOG 4146 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4147 mAudioWatchdog->pause(); 4148 } 4149#endif 4150 4151 // Now perform the deferred reset on fast tracks that have stopped 4152 while (resetMask != 0) { 4153 size_t i = __builtin_ctz(resetMask); 4154 ALOG_ASSERT(i < count); 4155 resetMask &= ~(1 << i); 4156 sp<Track> t = mActiveTracks[i].promote(); 4157 if (t == 0) { 4158 continue; 4159 } 4160 Track* track = t.get(); 4161 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4162 track->reset(); 4163 } 4164 4165 // remove all the tracks that need to be... 4166 removeTracks_l(*tracksToRemove); 4167 4168 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4169 mEffectBufferValid = true; 4170 } 4171 4172 if (mEffectBufferValid) { 4173 // as long as there are effects we should clear the effects buffer, to avoid 4174 // passing a non-clean buffer to the effect chain 4175 memset(mEffectBuffer, 0, mEffectBufferSize); 4176 } 4177 // sink or mix buffer must be cleared if all tracks are connected to an 4178 // effect chain as in this case the mixer will not write to the sink or mix buffer 4179 // and track effects will accumulate into it 4180 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4181 (mixedTracks == 0 && fastTracks > 0))) { 4182 // FIXME as a performance optimization, should remember previous zero status 4183 if (mMixerBufferValid) { 4184 memset(mMixerBuffer, 0, mMixerBufferSize); 4185 // TODO: In testing, mSinkBuffer below need not be cleared because 4186 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4187 // after mixing. 4188 // 4189 // To enforce this guarantee: 4190 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4191 // (mixedTracks == 0 && fastTracks > 0)) 4192 // must imply MIXER_TRACKS_READY. 4193 // Later, we may clear buffers regardless, and skip much of this logic. 4194 } 4195 // FIXME as a performance optimization, should remember previous zero status 4196 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4197 } 4198 4199 // if any fast tracks, then status is ready 4200 mMixerStatusIgnoringFastTracks = mixerStatus; 4201 if (fastTracks > 0) { 4202 mixerStatus = MIXER_TRACKS_READY; 4203 } 4204 return mixerStatus; 4205} 4206 4207// getTrackName_l() must be called with ThreadBase::mLock held 4208int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4209 audio_format_t format, int sessionId) 4210{ 4211 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4212} 4213 4214// deleteTrackName_l() must be called with ThreadBase::mLock held 4215void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4216{ 4217 ALOGV("remove track (%d) and delete from mixer", name); 4218 mAudioMixer->deleteTrackName(name); 4219} 4220 4221// checkForNewParameter_l() must be called with ThreadBase::mLock held 4222bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4223 status_t& status) 4224{ 4225 bool reconfig = false; 4226 4227 status = NO_ERROR; 4228 4229 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4230 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4231 if (mFastMixer != 0) { 4232 FastMixerStateQueue *sq = mFastMixer->sq(); 4233 FastMixerState *state = sq->begin(); 4234 if (!(state->mCommand & FastMixerState::IDLE)) { 4235 previousCommand = state->mCommand; 4236 state->mCommand = FastMixerState::HOT_IDLE; 4237 sq->end(); 4238 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4239 } else { 4240 sq->end(false /*didModify*/); 4241 } 4242 } 4243 4244 AudioParameter param = AudioParameter(keyValuePair); 4245 int value; 4246 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4247 reconfig = true; 4248 } 4249 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4250 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4251 status = BAD_VALUE; 4252 } else { 4253 // no need to save value, since it's constant 4254 reconfig = true; 4255 } 4256 } 4257 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4258 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4259 status = BAD_VALUE; 4260 } else { 4261 // no need to save value, since it's constant 4262 reconfig = true; 4263 } 4264 } 4265 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4266 // do not accept frame count changes if tracks are open as the track buffer 4267 // size depends on frame count and correct behavior would not be guaranteed 4268 // if frame count is changed after track creation 4269 if (!mTracks.isEmpty()) { 4270 status = INVALID_OPERATION; 4271 } else { 4272 reconfig = true; 4273 } 4274 } 4275 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4276#ifdef ADD_BATTERY_DATA 4277 // when changing the audio output device, call addBatteryData to notify 4278 // the change 4279 if (mOutDevice != value) { 4280 uint32_t params = 0; 4281 // check whether speaker is on 4282 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4283 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4284 } 4285 4286 audio_devices_t deviceWithoutSpeaker 4287 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4288 // check if any other device (except speaker) is on 4289 if (value & deviceWithoutSpeaker) { 4290 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4291 } 4292 4293 if (params != 0) { 4294 addBatteryData(params); 4295 } 4296 } 4297#endif 4298 4299 // forward device change to effects that have requested to be 4300 // aware of attached audio device. 4301 if (value != AUDIO_DEVICE_NONE) { 4302 mOutDevice = value; 4303 for (size_t i = 0; i < mEffectChains.size(); i++) { 4304 mEffectChains[i]->setDevice_l(mOutDevice); 4305 } 4306 } 4307 } 4308 4309 if (status == NO_ERROR) { 4310 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4311 keyValuePair.string()); 4312 if (!mStandby && status == INVALID_OPERATION) { 4313 mOutput->standby(); 4314 mStandby = true; 4315 mBytesWritten = 0; 4316 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4317 keyValuePair.string()); 4318 } 4319 if (status == NO_ERROR && reconfig) { 4320 readOutputParameters_l(); 4321 delete mAudioMixer; 4322 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4323 for (size_t i = 0; i < mTracks.size() ; i++) { 4324 int name = getTrackName_l(mTracks[i]->mChannelMask, 4325 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4326 if (name < 0) { 4327 break; 4328 } 4329 mTracks[i]->mName = name; 4330 } 4331 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4332 } 4333 } 4334 4335 if (!(previousCommand & FastMixerState::IDLE)) { 4336 ALOG_ASSERT(mFastMixer != 0); 4337 FastMixerStateQueue *sq = mFastMixer->sq(); 4338 FastMixerState *state = sq->begin(); 4339 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4340 state->mCommand = previousCommand; 4341 sq->end(); 4342 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4343 } 4344 4345 return reconfig; 4346} 4347 4348 4349void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4350{ 4351 const size_t SIZE = 256; 4352 char buffer[SIZE]; 4353 String8 result; 4354 4355 PlaybackThread::dumpInternals(fd, args); 4356 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4357 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4358 4359 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4360 const FastMixerDumpState copy(mFastMixerDumpState); 4361 copy.dump(fd); 4362 4363#ifdef STATE_QUEUE_DUMP 4364 // Similar for state queue 4365 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4366 observerCopy.dump(fd); 4367 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4368 mutatorCopy.dump(fd); 4369#endif 4370 4371#ifdef TEE_SINK 4372 // Write the tee output to a .wav file 4373 dumpTee(fd, mTeeSource, mId); 4374#endif 4375 4376#ifdef AUDIO_WATCHDOG 4377 if (mAudioWatchdog != 0) { 4378 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4379 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4380 wdCopy.dump(fd); 4381 } 4382#endif 4383} 4384 4385uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4386{ 4387 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4388} 4389 4390uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4391{ 4392 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4393} 4394 4395void AudioFlinger::MixerThread::cacheParameters_l() 4396{ 4397 PlaybackThread::cacheParameters_l(); 4398 4399 // FIXME: Relaxed timing because of a certain device that can't meet latency 4400 // Should be reduced to 2x after the vendor fixes the driver issue 4401 // increase threshold again due to low power audio mode. The way this warning 4402 // threshold is calculated and its usefulness should be reconsidered anyway. 4403 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4404} 4405 4406// ---------------------------------------------------------------------------- 4407 4408AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4409 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4410 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4411 // mLeftVolFloat, mRightVolFloat 4412{ 4413} 4414 4415AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4416 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4417 ThreadBase::type_t type, bool systemReady) 4418 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4419 // mLeftVolFloat, mRightVolFloat 4420{ 4421} 4422 4423AudioFlinger::DirectOutputThread::~DirectOutputThread() 4424{ 4425} 4426 4427void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4428{ 4429 audio_track_cblk_t* cblk = track->cblk(); 4430 float left, right; 4431 4432 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4433 left = right = 0; 4434 } else { 4435 float typeVolume = mStreamTypes[track->streamType()].volume; 4436 float v = mMasterVolume * typeVolume; 4437 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4438 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4439 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4440 if (left > GAIN_FLOAT_UNITY) { 4441 left = GAIN_FLOAT_UNITY; 4442 } 4443 left *= v; 4444 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4445 if (right > GAIN_FLOAT_UNITY) { 4446 right = GAIN_FLOAT_UNITY; 4447 } 4448 right *= v; 4449 } 4450 4451 if (lastTrack) { 4452 if (left != mLeftVolFloat || right != mRightVolFloat) { 4453 mLeftVolFloat = left; 4454 mRightVolFloat = right; 4455 4456 // Convert volumes from float to 8.24 4457 uint32_t vl = (uint32_t)(left * (1 << 24)); 4458 uint32_t vr = (uint32_t)(right * (1 << 24)); 4459 4460 // Delegate volume control to effect in track effect chain if needed 4461 // only one effect chain can be present on DirectOutputThread, so if 4462 // there is one, the track is connected to it 4463 if (!mEffectChains.isEmpty()) { 4464 mEffectChains[0]->setVolume_l(&vl, &vr); 4465 left = (float)vl / (1 << 24); 4466 right = (float)vr / (1 << 24); 4467 } 4468 if (mOutput->stream->set_volume) { 4469 mOutput->stream->set_volume(mOutput->stream, left, right); 4470 } 4471 } 4472 } 4473} 4474 4475void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4476{ 4477 sp<Track> previousTrack = mPreviousTrack.promote(); 4478 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4479 4480 if (previousTrack != 0 && latestTrack != 0 && 4481 (previousTrack->sessionId() != latestTrack->sessionId())) { 4482 mFlushPending = true; 4483 } 4484 PlaybackThread::onAddNewTrack_l(); 4485} 4486 4487AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4488 Vector< sp<Track> > *tracksToRemove 4489) 4490{ 4491 size_t count = mActiveTracks.size(); 4492 mixer_state mixerStatus = MIXER_IDLE; 4493 bool doHwPause = false; 4494 bool doHwResume = false; 4495 4496 // find out which tracks need to be processed 4497 for (size_t i = 0; i < count; i++) { 4498 sp<Track> t = mActiveTracks[i].promote(); 4499 // The track died recently 4500 if (t == 0) { 4501 continue; 4502 } 4503 4504 if (t->isInvalid()) { 4505 ALOGW("An invalidated track shouldn't be in active list"); 4506 tracksToRemove->add(t); 4507 continue; 4508 } 4509 4510 Track* const track = t.get(); 4511 audio_track_cblk_t* cblk = track->cblk(); 4512 // Only consider last track started for volume and mixer state control. 4513 // In theory an older track could underrun and restart after the new one starts 4514 // but as we only care about the transition phase between two tracks on a 4515 // direct output, it is not a problem to ignore the underrun case. 4516 sp<Track> l = mLatestActiveTrack.promote(); 4517 bool last = l.get() == track; 4518 4519 if (track->isPausing()) { 4520 track->setPaused(); 4521 if (mHwSupportsPause && last && !mHwPaused) { 4522 doHwPause = true; 4523 mHwPaused = true; 4524 } 4525 tracksToRemove->add(track); 4526 } else if (track->isFlushPending()) { 4527 track->flushAck(); 4528 if (last) { 4529 mFlushPending = true; 4530 } 4531 } else if (track->isResumePending()) { 4532 track->resumeAck(); 4533 if (last && mHwPaused) { 4534 doHwResume = true; 4535 mHwPaused = false; 4536 } 4537 } 4538 4539 // The first time a track is added we wait 4540 // for all its buffers to be filled before processing it. 4541 // Allow draining the buffer in case the client 4542 // app does not call stop() and relies on underrun to stop: 4543 // hence the test on (track->mRetryCount > 1). 4544 // If retryCount<=1 then track is about to underrun and be removed. 4545 uint32_t minFrames; 4546 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4547 && (track->mRetryCount > 1)) { 4548 minFrames = mNormalFrameCount; 4549 } else { 4550 minFrames = 1; 4551 } 4552 4553 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4554 !track->isStopping_2() && !track->isStopped()) 4555 { 4556 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4557 4558 if (track->mFillingUpStatus == Track::FS_FILLED) { 4559 track->mFillingUpStatus = Track::FS_ACTIVE; 4560 // make sure processVolume_l() will apply new volume even if 0 4561 mLeftVolFloat = mRightVolFloat = -1.0; 4562 if (!mHwSupportsPause) { 4563 track->resumeAck(); 4564 } 4565 } 4566 4567 // compute volume for this track 4568 processVolume_l(track, last); 4569 if (last) { 4570 sp<Track> previousTrack = mPreviousTrack.promote(); 4571 if (previousTrack != 0) { 4572 if (track != previousTrack.get()) { 4573 // Flush any data still being written from last track 4574 mBytesRemaining = 0; 4575 // flush data already sent if changing audio session as audio 4576 // comes from a different source. Also invalidate previous track to force a 4577 // seek when resuming. 4578 if (previousTrack->sessionId() != track->sessionId()) { 4579 previousTrack->invalidate(); 4580 } 4581 } 4582 } 4583 mPreviousTrack = track; 4584 4585 // reset retry count 4586 track->mRetryCount = kMaxTrackRetriesDirect; 4587 mActiveTrack = t; 4588 mixerStatus = MIXER_TRACKS_READY; 4589 if (mHwPaused) { 4590 doHwResume = true; 4591 mHwPaused = false; 4592 } 4593 } 4594 } else { 4595 // clear effect chain input buffer if the last active track started underruns 4596 // to avoid sending previous audio buffer again to effects 4597 if (!mEffectChains.isEmpty() && last) { 4598 mEffectChains[0]->clearInputBuffer(); 4599 } 4600 if (track->isStopping_1()) { 4601 track->mState = TrackBase::STOPPING_2; 4602 if (last && mHwPaused) { 4603 doHwResume = true; 4604 mHwPaused = false; 4605 } 4606 } 4607 if ((track->sharedBuffer() != 0) || track->isStopped() || 4608 track->isStopping_2() || track->isPaused()) { 4609 // We have consumed all the buffers of this track. 4610 // Remove it from the list of active tracks. 4611 size_t audioHALFrames; 4612 if (audio_is_linear_pcm(mFormat)) { 4613 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4614 } else { 4615 audioHALFrames = 0; 4616 } 4617 4618 size_t framesWritten = mBytesWritten / mFrameSize; 4619 if (mStandby || !last || 4620 track->presentationComplete(framesWritten, audioHALFrames)) { 4621 if (track->isStopping_2()) { 4622 track->mState = TrackBase::STOPPED; 4623 } 4624 if (track->isStopped()) { 4625 track->reset(); 4626 } 4627 tracksToRemove->add(track); 4628 } 4629 } else { 4630 // No buffers for this track. Give it a few chances to 4631 // fill a buffer, then remove it from active list. 4632 // Only consider last track started for mixer state control 4633 if (--(track->mRetryCount) <= 0) { 4634 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4635 tracksToRemove->add(track); 4636 // indicate to client process that the track was disabled because of underrun; 4637 // it will then automatically call start() when data is available 4638 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4639 } else if (last) { 4640 mixerStatus = MIXER_TRACKS_ENABLED; 4641 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4642 doHwPause = true; 4643 mHwPaused = true; 4644 } 4645 } 4646 } 4647 } 4648 } 4649 4650 // if an active track did not command a flush, check for pending flush on stopped tracks 4651 if (!mFlushPending) { 4652 for (size_t i = 0; i < mTracks.size(); i++) { 4653 if (mTracks[i]->isFlushPending()) { 4654 mTracks[i]->flushAck(); 4655 mFlushPending = true; 4656 } 4657 } 4658 } 4659 4660 // make sure the pause/flush/resume sequence is executed in the right order. 4661 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4662 // before flush and then resume HW. This can happen in case of pause/flush/resume 4663 // if resume is received before pause is executed. 4664 if (mHwSupportsPause && !mStandby && 4665 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4666 mOutput->stream->pause(mOutput->stream); 4667 } 4668 if (mFlushPending) { 4669 flushHw_l(); 4670 } 4671 if (mHwSupportsPause && !mStandby && doHwResume) { 4672 mOutput->stream->resume(mOutput->stream); 4673 } 4674 // remove all the tracks that need to be... 4675 removeTracks_l(*tracksToRemove); 4676 4677 return mixerStatus; 4678} 4679 4680void AudioFlinger::DirectOutputThread::threadLoop_mix() 4681{ 4682 size_t frameCount = mFrameCount; 4683 int8_t *curBuf = (int8_t *)mSinkBuffer; 4684 // output audio to hardware 4685 while (frameCount) { 4686 AudioBufferProvider::Buffer buffer; 4687 buffer.frameCount = frameCount; 4688 status_t status = mActiveTrack->getNextBuffer(&buffer); 4689 if (status != NO_ERROR || buffer.raw == NULL) { 4690 memset(curBuf, 0, frameCount * mFrameSize); 4691 break; 4692 } 4693 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4694 frameCount -= buffer.frameCount; 4695 curBuf += buffer.frameCount * mFrameSize; 4696 mActiveTrack->releaseBuffer(&buffer); 4697 } 4698 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4699 mSleepTimeUs = 0; 4700 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4701 mActiveTrack.clear(); 4702} 4703 4704void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4705{ 4706 // do not write to HAL when paused 4707 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4708 mSleepTimeUs = mIdleSleepTimeUs; 4709 return; 4710 } 4711 if (mSleepTimeUs == 0) { 4712 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4713 mSleepTimeUs = mActiveSleepTimeUs; 4714 } else { 4715 mSleepTimeUs = mIdleSleepTimeUs; 4716 } 4717 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4718 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4719 mSleepTimeUs = 0; 4720 } 4721} 4722 4723void AudioFlinger::DirectOutputThread::threadLoop_exit() 4724{ 4725 { 4726 Mutex::Autolock _l(mLock); 4727 for (size_t i = 0; i < mTracks.size(); i++) { 4728 if (mTracks[i]->isFlushPending()) { 4729 mTracks[i]->flushAck(); 4730 mFlushPending = true; 4731 } 4732 } 4733 if (mFlushPending) { 4734 flushHw_l(); 4735 } 4736 } 4737 PlaybackThread::threadLoop_exit(); 4738} 4739 4740// must be called with thread mutex locked 4741bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4742{ 4743 bool trackPaused = false; 4744 bool trackStopped = false; 4745 4746 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4747 // after a timeout and we will enter standby then. 4748 if (mTracks.size() > 0) { 4749 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4750 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4751 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4752 } 4753 4754 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4755} 4756 4757// getTrackName_l() must be called with ThreadBase::mLock held 4758int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4759 audio_format_t format __unused, int sessionId __unused) 4760{ 4761 return 0; 4762} 4763 4764// deleteTrackName_l() must be called with ThreadBase::mLock held 4765void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4766{ 4767} 4768 4769// checkForNewParameter_l() must be called with ThreadBase::mLock held 4770bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4771 status_t& status) 4772{ 4773 bool reconfig = false; 4774 4775 status = NO_ERROR; 4776 4777 AudioParameter param = AudioParameter(keyValuePair); 4778 int value; 4779 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4780 // forward device change to effects that have requested to be 4781 // aware of attached audio device. 4782 if (value != AUDIO_DEVICE_NONE) { 4783 mOutDevice = value; 4784 for (size_t i = 0; i < mEffectChains.size(); i++) { 4785 mEffectChains[i]->setDevice_l(mOutDevice); 4786 } 4787 } 4788 } 4789 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4790 // do not accept frame count changes if tracks are open as the track buffer 4791 // size depends on frame count and correct behavior would not be garantied 4792 // if frame count is changed after track creation 4793 if (!mTracks.isEmpty()) { 4794 status = INVALID_OPERATION; 4795 } else { 4796 reconfig = true; 4797 } 4798 } 4799 if (status == NO_ERROR) { 4800 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4801 keyValuePair.string()); 4802 if (!mStandby && status == INVALID_OPERATION) { 4803 mOutput->standby(); 4804 mStandby = true; 4805 mBytesWritten = 0; 4806 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4807 keyValuePair.string()); 4808 } 4809 if (status == NO_ERROR && reconfig) { 4810 readOutputParameters_l(); 4811 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4812 } 4813 } 4814 4815 return reconfig; 4816} 4817 4818uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4819{ 4820 uint32_t time; 4821 if (audio_is_linear_pcm(mFormat)) { 4822 time = PlaybackThread::activeSleepTimeUs(); 4823 } else { 4824 time = 10000; 4825 } 4826 return time; 4827} 4828 4829uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4830{ 4831 uint32_t time; 4832 if (audio_is_linear_pcm(mFormat)) { 4833 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4834 } else { 4835 time = 10000; 4836 } 4837 return time; 4838} 4839 4840uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4841{ 4842 uint32_t time; 4843 if (audio_is_linear_pcm(mFormat)) { 4844 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4845 } else { 4846 time = 10000; 4847 } 4848 return time; 4849} 4850 4851void AudioFlinger::DirectOutputThread::cacheParameters_l() 4852{ 4853 PlaybackThread::cacheParameters_l(); 4854 4855 // use shorter standby delay as on normal output to release 4856 // hardware resources as soon as possible 4857 // no delay on outputs with HW A/V sync 4858 if (usesHwAvSync()) { 4859 mStandbyDelayNs = 0; 4860 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 4861 mStandbyDelayNs = kOffloadStandbyDelayNs; 4862 } else { 4863 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 4864 } 4865} 4866 4867void AudioFlinger::DirectOutputThread::flushHw_l() 4868{ 4869 mOutput->flush(); 4870 mHwPaused = false; 4871 mFlushPending = false; 4872} 4873 4874// ---------------------------------------------------------------------------- 4875 4876AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4877 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4878 : Thread(false /*canCallJava*/), 4879 mPlaybackThread(playbackThread), 4880 mWriteAckSequence(0), 4881 mDrainSequence(0) 4882{ 4883} 4884 4885AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4886{ 4887} 4888 4889void AudioFlinger::AsyncCallbackThread::onFirstRef() 4890{ 4891 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4892} 4893 4894bool AudioFlinger::AsyncCallbackThread::threadLoop() 4895{ 4896 while (!exitPending()) { 4897 uint32_t writeAckSequence; 4898 uint32_t drainSequence; 4899 4900 { 4901 Mutex::Autolock _l(mLock); 4902 while (!((mWriteAckSequence & 1) || 4903 (mDrainSequence & 1) || 4904 exitPending())) { 4905 mWaitWorkCV.wait(mLock); 4906 } 4907 4908 if (exitPending()) { 4909 break; 4910 } 4911 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4912 mWriteAckSequence, mDrainSequence); 4913 writeAckSequence = mWriteAckSequence; 4914 mWriteAckSequence &= ~1; 4915 drainSequence = mDrainSequence; 4916 mDrainSequence &= ~1; 4917 } 4918 { 4919 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4920 if (playbackThread != 0) { 4921 if (writeAckSequence & 1) { 4922 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4923 } 4924 if (drainSequence & 1) { 4925 playbackThread->resetDraining(drainSequence >> 1); 4926 } 4927 } 4928 } 4929 } 4930 return false; 4931} 4932 4933void AudioFlinger::AsyncCallbackThread::exit() 4934{ 4935 ALOGV("AsyncCallbackThread::exit"); 4936 Mutex::Autolock _l(mLock); 4937 requestExit(); 4938 mWaitWorkCV.broadcast(); 4939} 4940 4941void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4942{ 4943 Mutex::Autolock _l(mLock); 4944 // bit 0 is cleared 4945 mWriteAckSequence = sequence << 1; 4946} 4947 4948void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4949{ 4950 Mutex::Autolock _l(mLock); 4951 // ignore unexpected callbacks 4952 if (mWriteAckSequence & 2) { 4953 mWriteAckSequence |= 1; 4954 mWaitWorkCV.signal(); 4955 } 4956} 4957 4958void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4959{ 4960 Mutex::Autolock _l(mLock); 4961 // bit 0 is cleared 4962 mDrainSequence = sequence << 1; 4963} 4964 4965void AudioFlinger::AsyncCallbackThread::resetDraining() 4966{ 4967 Mutex::Autolock _l(mLock); 4968 // ignore unexpected callbacks 4969 if (mDrainSequence & 2) { 4970 mDrainSequence |= 1; 4971 mWaitWorkCV.signal(); 4972 } 4973} 4974 4975 4976// ---------------------------------------------------------------------------- 4977AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4978 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 4979 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 4980 mPausedBytesRemaining(0) 4981{ 4982 //FIXME: mStandby should be set to true by ThreadBase constructor 4983 mStandby = true; 4984} 4985 4986void AudioFlinger::OffloadThread::threadLoop_exit() 4987{ 4988 if (mFlushPending || mHwPaused) { 4989 // If a flush is pending or track was paused, just discard buffered data 4990 flushHw_l(); 4991 } else { 4992 mMixerStatus = MIXER_DRAIN_ALL; 4993 threadLoop_drain(); 4994 } 4995 if (mUseAsyncWrite) { 4996 ALOG_ASSERT(mCallbackThread != 0); 4997 mCallbackThread->exit(); 4998 } 4999 PlaybackThread::threadLoop_exit(); 5000} 5001 5002AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5003 Vector< sp<Track> > *tracksToRemove 5004) 5005{ 5006 size_t count = mActiveTracks.size(); 5007 5008 mixer_state mixerStatus = MIXER_IDLE; 5009 bool doHwPause = false; 5010 bool doHwResume = false; 5011 5012 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5013 5014 // find out which tracks need to be processed 5015 for (size_t i = 0; i < count; i++) { 5016 sp<Track> t = mActiveTracks[i].promote(); 5017 // The track died recently 5018 if (t == 0) { 5019 continue; 5020 } 5021 Track* const track = t.get(); 5022 audio_track_cblk_t* cblk = track->cblk(); 5023 // Only consider last track started for volume and mixer state control. 5024 // In theory an older track could underrun and restart after the new one starts 5025 // but as we only care about the transition phase between two tracks on a 5026 // direct output, it is not a problem to ignore the underrun case. 5027 sp<Track> l = mLatestActiveTrack.promote(); 5028 bool last = l.get() == track; 5029 5030 if (track->isInvalid()) { 5031 ALOGW("An invalidated track shouldn't be in active list"); 5032 tracksToRemove->add(track); 5033 continue; 5034 } 5035 5036 if (track->mState == TrackBase::IDLE) { 5037 ALOGW("An idle track shouldn't be in active list"); 5038 continue; 5039 } 5040 5041 if (track->isPausing()) { 5042 track->setPaused(); 5043 if (last) { 5044 if (mHwSupportsPause && !mHwPaused) { 5045 doHwPause = true; 5046 mHwPaused = true; 5047 } 5048 // If we were part way through writing the mixbuffer to 5049 // the HAL we must save this until we resume 5050 // BUG - this will be wrong if a different track is made active, 5051 // in that case we want to discard the pending data in the 5052 // mixbuffer and tell the client to present it again when the 5053 // track is resumed 5054 mPausedWriteLength = mCurrentWriteLength; 5055 mPausedBytesRemaining = mBytesRemaining; 5056 mBytesRemaining = 0; // stop writing 5057 } 5058 tracksToRemove->add(track); 5059 } else if (track->isFlushPending()) { 5060 track->flushAck(); 5061 if (last) { 5062 mFlushPending = true; 5063 } 5064 } else if (track->isResumePending()){ 5065 track->resumeAck(); 5066 if (last) { 5067 if (mPausedBytesRemaining) { 5068 // Need to continue write that was interrupted 5069 mCurrentWriteLength = mPausedWriteLength; 5070 mBytesRemaining = mPausedBytesRemaining; 5071 mPausedBytesRemaining = 0; 5072 } 5073 if (mHwPaused) { 5074 doHwResume = true; 5075 mHwPaused = false; 5076 // threadLoop_mix() will handle the case that we need to 5077 // resume an interrupted write 5078 } 5079 // enable write to audio HAL 5080 mSleepTimeUs = 0; 5081 5082 // Do not handle new data in this iteration even if track->framesReady() 5083 mixerStatus = MIXER_TRACKS_ENABLED; 5084 } 5085 } else if (track->framesReady() && track->isReady() && 5086 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5087 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5088 if (track->mFillingUpStatus == Track::FS_FILLED) { 5089 track->mFillingUpStatus = Track::FS_ACTIVE; 5090 // make sure processVolume_l() will apply new volume even if 0 5091 mLeftVolFloat = mRightVolFloat = -1.0; 5092 } 5093 5094 if (last) { 5095 sp<Track> previousTrack = mPreviousTrack.promote(); 5096 if (previousTrack != 0) { 5097 if (track != previousTrack.get()) { 5098 // Flush any data still being written from last track 5099 mBytesRemaining = 0; 5100 if (mPausedBytesRemaining) { 5101 // Last track was paused so we also need to flush saved 5102 // mixbuffer state and invalidate track so that it will 5103 // re-submit that unwritten data when it is next resumed 5104 mPausedBytesRemaining = 0; 5105 // Invalidate is a bit drastic - would be more efficient 5106 // to have a flag to tell client that some of the 5107 // previously written data was lost 5108 previousTrack->invalidate(); 5109 } 5110 // flush data already sent to the DSP if changing audio session as audio 5111 // comes from a different source. Also invalidate previous track to force a 5112 // seek when resuming. 5113 if (previousTrack->sessionId() != track->sessionId()) { 5114 previousTrack->invalidate(); 5115 } 5116 } 5117 } 5118 mPreviousTrack = track; 5119 // reset retry count 5120 track->mRetryCount = kMaxTrackRetriesOffload; 5121 mActiveTrack = t; 5122 mixerStatus = MIXER_TRACKS_READY; 5123 } 5124 } else { 5125 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5126 if (track->isStopping_1()) { 5127 // Hardware buffer can hold a large amount of audio so we must 5128 // wait for all current track's data to drain before we say 5129 // that the track is stopped. 5130 if (mBytesRemaining == 0) { 5131 // Only start draining when all data in mixbuffer 5132 // has been written 5133 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5134 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5135 // do not drain if no data was ever sent to HAL (mStandby == true) 5136 if (last && !mStandby) { 5137 // do not modify drain sequence if we are already draining. This happens 5138 // when resuming from pause after drain. 5139 if ((mDrainSequence & 1) == 0) { 5140 mSleepTimeUs = 0; 5141 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5142 mixerStatus = MIXER_DRAIN_TRACK; 5143 mDrainSequence += 2; 5144 } 5145 if (mHwPaused) { 5146 // It is possible to move from PAUSED to STOPPING_1 without 5147 // a resume so we must ensure hardware is running 5148 doHwResume = true; 5149 mHwPaused = false; 5150 } 5151 } 5152 } 5153 } else if (track->isStopping_2()) { 5154 // Drain has completed or we are in standby, signal presentation complete 5155 if (!(mDrainSequence & 1) || !last || mStandby) { 5156 track->mState = TrackBase::STOPPED; 5157 size_t audioHALFrames = 5158 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5159 size_t framesWritten = 5160 mBytesWritten / mOutput->getFrameSize(); 5161 track->presentationComplete(framesWritten, audioHALFrames); 5162 track->reset(); 5163 tracksToRemove->add(track); 5164 } 5165 } else { 5166 // No buffers for this track. Give it a few chances to 5167 // fill a buffer, then remove it from active list. 5168 if (--(track->mRetryCount) <= 0) { 5169 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5170 track->name()); 5171 tracksToRemove->add(track); 5172 // indicate to client process that the track was disabled because of underrun; 5173 // it will then automatically call start() when data is available 5174 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5175 } else if (last){ 5176 mixerStatus = MIXER_TRACKS_ENABLED; 5177 } 5178 } 5179 } 5180 // compute volume for this track 5181 processVolume_l(track, last); 5182 } 5183 5184 // make sure the pause/flush/resume sequence is executed in the right order. 5185 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5186 // before flush and then resume HW. This can happen in case of pause/flush/resume 5187 // if resume is received before pause is executed. 5188 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5189 mOutput->stream->pause(mOutput->stream); 5190 } 5191 if (mFlushPending) { 5192 flushHw_l(); 5193 } 5194 if (!mStandby && doHwResume) { 5195 mOutput->stream->resume(mOutput->stream); 5196 } 5197 5198 // remove all the tracks that need to be... 5199 removeTracks_l(*tracksToRemove); 5200 5201 return mixerStatus; 5202} 5203 5204// must be called with thread mutex locked 5205bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5206{ 5207 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5208 mWriteAckSequence, mDrainSequence); 5209 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5210 return true; 5211 } 5212 return false; 5213} 5214 5215bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5216{ 5217 Mutex::Autolock _l(mLock); 5218 return waitingAsyncCallback_l(); 5219} 5220 5221void AudioFlinger::OffloadThread::flushHw_l() 5222{ 5223 DirectOutputThread::flushHw_l(); 5224 // Flush anything still waiting in the mixbuffer 5225 mCurrentWriteLength = 0; 5226 mBytesRemaining = 0; 5227 mPausedWriteLength = 0; 5228 mPausedBytesRemaining = 0; 5229 5230 if (mUseAsyncWrite) { 5231 // discard any pending drain or write ack by incrementing sequence 5232 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5233 mDrainSequence = (mDrainSequence + 2) & ~1; 5234 ALOG_ASSERT(mCallbackThread != 0); 5235 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5236 mCallbackThread->setDraining(mDrainSequence); 5237 } 5238} 5239 5240// ---------------------------------------------------------------------------- 5241 5242AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5243 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5244 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5245 systemReady, DUPLICATING), 5246 mWaitTimeMs(UINT_MAX) 5247{ 5248 addOutputTrack(mainThread); 5249} 5250 5251AudioFlinger::DuplicatingThread::~DuplicatingThread() 5252{ 5253 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5254 mOutputTracks[i]->destroy(); 5255 } 5256} 5257 5258void AudioFlinger::DuplicatingThread::threadLoop_mix() 5259{ 5260 // mix buffers... 5261 if (outputsReady(outputTracks)) { 5262 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5263 } else { 5264 if (mMixerBufferValid) { 5265 memset(mMixerBuffer, 0, mMixerBufferSize); 5266 } else { 5267 memset(mSinkBuffer, 0, mSinkBufferSize); 5268 } 5269 } 5270 mSleepTimeUs = 0; 5271 writeFrames = mNormalFrameCount; 5272 mCurrentWriteLength = mSinkBufferSize; 5273 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5274} 5275 5276void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5277{ 5278 if (mSleepTimeUs == 0) { 5279 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5280 mSleepTimeUs = mActiveSleepTimeUs; 5281 } else { 5282 mSleepTimeUs = mIdleSleepTimeUs; 5283 } 5284 } else if (mBytesWritten != 0) { 5285 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5286 writeFrames = mNormalFrameCount; 5287 memset(mSinkBuffer, 0, mSinkBufferSize); 5288 } else { 5289 // flush remaining overflow buffers in output tracks 5290 writeFrames = 0; 5291 } 5292 mSleepTimeUs = 0; 5293 } 5294} 5295 5296ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5297{ 5298 for (size_t i = 0; i < outputTracks.size(); i++) { 5299 outputTracks[i]->write(mSinkBuffer, writeFrames); 5300 } 5301 mStandby = false; 5302 return (ssize_t)mSinkBufferSize; 5303} 5304 5305void AudioFlinger::DuplicatingThread::threadLoop_standby() 5306{ 5307 // DuplicatingThread implements standby by stopping all tracks 5308 for (size_t i = 0; i < outputTracks.size(); i++) { 5309 outputTracks[i]->stop(); 5310 } 5311} 5312 5313void AudioFlinger::DuplicatingThread::saveOutputTracks() 5314{ 5315 outputTracks = mOutputTracks; 5316} 5317 5318void AudioFlinger::DuplicatingThread::clearOutputTracks() 5319{ 5320 outputTracks.clear(); 5321} 5322 5323void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5324{ 5325 Mutex::Autolock _l(mLock); 5326 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5327 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5328 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5329 const size_t frameCount = 5330 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5331 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5332 // from different OutputTracks and their associated MixerThreads (e.g. one may 5333 // nearly empty and the other may be dropping data). 5334 5335 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5336 this, 5337 mSampleRate, 5338 mFormat, 5339 mChannelMask, 5340 frameCount, 5341 IPCThreadState::self()->getCallingUid()); 5342 if (outputTrack->cblk() != NULL) { 5343 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5344 mOutputTracks.add(outputTrack); 5345 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5346 updateWaitTime_l(); 5347 } 5348} 5349 5350void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5351{ 5352 Mutex::Autolock _l(mLock); 5353 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5354 if (mOutputTracks[i]->thread() == thread) { 5355 mOutputTracks[i]->destroy(); 5356 mOutputTracks.removeAt(i); 5357 updateWaitTime_l(); 5358 if (thread->getOutput() == mOutput) { 5359 mOutput = NULL; 5360 } 5361 return; 5362 } 5363 } 5364 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5365} 5366 5367// caller must hold mLock 5368void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5369{ 5370 mWaitTimeMs = UINT_MAX; 5371 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5372 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5373 if (strong != 0) { 5374 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5375 if (waitTimeMs < mWaitTimeMs) { 5376 mWaitTimeMs = waitTimeMs; 5377 } 5378 } 5379 } 5380} 5381 5382 5383bool AudioFlinger::DuplicatingThread::outputsReady( 5384 const SortedVector< sp<OutputTrack> > &outputTracks) 5385{ 5386 for (size_t i = 0; i < outputTracks.size(); i++) { 5387 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5388 if (thread == 0) { 5389 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5390 outputTracks[i].get()); 5391 return false; 5392 } 5393 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5394 // see note at standby() declaration 5395 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5396 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5397 thread.get()); 5398 return false; 5399 } 5400 } 5401 return true; 5402} 5403 5404uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5405{ 5406 return (mWaitTimeMs * 1000) / 2; 5407} 5408 5409void AudioFlinger::DuplicatingThread::cacheParameters_l() 5410{ 5411 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5412 updateWaitTime_l(); 5413 5414 MixerThread::cacheParameters_l(); 5415} 5416 5417// ---------------------------------------------------------------------------- 5418// Record 5419// ---------------------------------------------------------------------------- 5420 5421AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5422 AudioStreamIn *input, 5423 audio_io_handle_t id, 5424 audio_devices_t outDevice, 5425 audio_devices_t inDevice, 5426 bool systemReady 5427#ifdef TEE_SINK 5428 , const sp<NBAIO_Sink>& teeSink 5429#endif 5430 ) : 5431 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5432 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5433 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5434 mRsmpInRear(0) 5435#ifdef TEE_SINK 5436 , mTeeSink(teeSink) 5437#endif 5438 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5439 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5440 // mFastCapture below 5441 , mFastCaptureFutex(0) 5442 // mInputSource 5443 // mPipeSink 5444 // mPipeSource 5445 , mPipeFramesP2(0) 5446 // mPipeMemory 5447 // mFastCaptureNBLogWriter 5448 , mFastTrackAvail(false) 5449{ 5450 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5451 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5452 5453 readInputParameters_l(); 5454 5455 // create an NBAIO source for the HAL input stream, and negotiate 5456 mInputSource = new AudioStreamInSource(input->stream); 5457 size_t numCounterOffers = 0; 5458 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5459 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5460 ALOG_ASSERT(index == 0); 5461 5462 // initialize fast capture depending on configuration 5463 bool initFastCapture; 5464 switch (kUseFastCapture) { 5465 case FastCapture_Never: 5466 initFastCapture = false; 5467 break; 5468 case FastCapture_Always: 5469 initFastCapture = true; 5470 break; 5471 case FastCapture_Static: 5472 uint32_t primaryOutputSampleRate; 5473 { 5474 AutoMutex _l(audioFlinger->mHardwareLock); 5475 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5476 } 5477 initFastCapture = 5478 // either capture sample rate is same as (a reasonable) primary output sample rate 5479 ((isMusicRate(primaryOutputSampleRate) && 5480 (mSampleRate == primaryOutputSampleRate)) || 5481 // or primary output sample rate is unknown, and capture sample rate is reasonable 5482 ((primaryOutputSampleRate == 0) && 5483 isMusicRate(mSampleRate))) && 5484 // and the buffer size is < 12 ms 5485 (mFrameCount * 1000) / mSampleRate < 12; 5486 break; 5487 // case FastCapture_Dynamic: 5488 } 5489 5490 if (initFastCapture) { 5491 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5492 NBAIO_Format format = mInputSource->format(); 5493 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5494 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5495 void *pipeBuffer; 5496 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5497 sp<IMemory> pipeMemory; 5498 if ((roHeap == 0) || 5499 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5500 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5501 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5502 goto failed; 5503 } 5504 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5505 memset(pipeBuffer, 0, pipeSize); 5506 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5507 const NBAIO_Format offers[1] = {format}; 5508 size_t numCounterOffers = 0; 5509 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5510 ALOG_ASSERT(index == 0); 5511 mPipeSink = pipe; 5512 PipeReader *pipeReader = new PipeReader(*pipe); 5513 numCounterOffers = 0; 5514 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5515 ALOG_ASSERT(index == 0); 5516 mPipeSource = pipeReader; 5517 mPipeFramesP2 = pipeFramesP2; 5518 mPipeMemory = pipeMemory; 5519 5520 // create fast capture 5521 mFastCapture = new FastCapture(); 5522 FastCaptureStateQueue *sq = mFastCapture->sq(); 5523#ifdef STATE_QUEUE_DUMP 5524 // FIXME 5525#endif 5526 FastCaptureState *state = sq->begin(); 5527 state->mCblk = NULL; 5528 state->mInputSource = mInputSource.get(); 5529 state->mInputSourceGen++; 5530 state->mPipeSink = pipe; 5531 state->mPipeSinkGen++; 5532 state->mFrameCount = mFrameCount; 5533 state->mCommand = FastCaptureState::COLD_IDLE; 5534 // already done in constructor initialization list 5535 //mFastCaptureFutex = 0; 5536 state->mColdFutexAddr = &mFastCaptureFutex; 5537 state->mColdGen++; 5538 state->mDumpState = &mFastCaptureDumpState; 5539#ifdef TEE_SINK 5540 // FIXME 5541#endif 5542 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5543 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5544 sq->end(); 5545 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5546 5547 // start the fast capture 5548 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5549 pid_t tid = mFastCapture->getTid(); 5550 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5551#ifdef AUDIO_WATCHDOG 5552 // FIXME 5553#endif 5554 5555 mFastTrackAvail = true; 5556 } 5557failed: ; 5558 5559 // FIXME mNormalSource 5560} 5561 5562AudioFlinger::RecordThread::~RecordThread() 5563{ 5564 if (mFastCapture != 0) { 5565 FastCaptureStateQueue *sq = mFastCapture->sq(); 5566 FastCaptureState *state = sq->begin(); 5567 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5568 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5569 if (old == -1) { 5570 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5571 } 5572 } 5573 state->mCommand = FastCaptureState::EXIT; 5574 sq->end(); 5575 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5576 mFastCapture->join(); 5577 mFastCapture.clear(); 5578 } 5579 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5580 mAudioFlinger->unregisterWriter(mNBLogWriter); 5581 free(mRsmpInBuffer); 5582} 5583 5584void AudioFlinger::RecordThread::onFirstRef() 5585{ 5586 run(mThreadName, PRIORITY_URGENT_AUDIO); 5587} 5588 5589bool AudioFlinger::RecordThread::threadLoop() 5590{ 5591 nsecs_t lastWarning = 0; 5592 5593 inputStandBy(); 5594 5595reacquire_wakelock: 5596 sp<RecordTrack> activeTrack; 5597 int activeTracksGen; 5598 { 5599 Mutex::Autolock _l(mLock); 5600 size_t size = mActiveTracks.size(); 5601 activeTracksGen = mActiveTracksGen; 5602 if (size > 0) { 5603 // FIXME an arbitrary choice 5604 activeTrack = mActiveTracks[0]; 5605 acquireWakeLock_l(activeTrack->uid()); 5606 if (size > 1) { 5607 SortedVector<int> tmp; 5608 for (size_t i = 0; i < size; i++) { 5609 tmp.add(mActiveTracks[i]->uid()); 5610 } 5611 updateWakeLockUids_l(tmp); 5612 } 5613 } else { 5614 acquireWakeLock_l(-1); 5615 } 5616 } 5617 5618 // used to request a deferred sleep, to be executed later while mutex is unlocked 5619 uint32_t sleepUs = 0; 5620 5621 // loop while there is work to do 5622 for (;;) { 5623 Vector< sp<EffectChain> > effectChains; 5624 5625 // sleep with mutex unlocked 5626 if (sleepUs > 0) { 5627 ATRACE_BEGIN("sleep"); 5628 usleep(sleepUs); 5629 ATRACE_END(); 5630 sleepUs = 0; 5631 } 5632 5633 // activeTracks accumulates a copy of a subset of mActiveTracks 5634 Vector< sp<RecordTrack> > activeTracks; 5635 5636 // reference to the (first and only) active fast track 5637 sp<RecordTrack> fastTrack; 5638 5639 // reference to a fast track which is about to be removed 5640 sp<RecordTrack> fastTrackToRemove; 5641 5642 { // scope for mLock 5643 Mutex::Autolock _l(mLock); 5644 5645 processConfigEvents_l(); 5646 5647 // check exitPending here because checkForNewParameters_l() and 5648 // checkForNewParameters_l() can temporarily release mLock 5649 if (exitPending()) { 5650 break; 5651 } 5652 5653 // if no active track(s), then standby and release wakelock 5654 size_t size = mActiveTracks.size(); 5655 if (size == 0) { 5656 standbyIfNotAlreadyInStandby(); 5657 // exitPending() can't become true here 5658 releaseWakeLock_l(); 5659 ALOGV("RecordThread: loop stopping"); 5660 // go to sleep 5661 mWaitWorkCV.wait(mLock); 5662 ALOGV("RecordThread: loop starting"); 5663 goto reacquire_wakelock; 5664 } 5665 5666 if (mActiveTracksGen != activeTracksGen) { 5667 activeTracksGen = mActiveTracksGen; 5668 SortedVector<int> tmp; 5669 for (size_t i = 0; i < size; i++) { 5670 tmp.add(mActiveTracks[i]->uid()); 5671 } 5672 updateWakeLockUids_l(tmp); 5673 } 5674 5675 bool doBroadcast = false; 5676 for (size_t i = 0; i < size; ) { 5677 5678 activeTrack = mActiveTracks[i]; 5679 if (activeTrack->isTerminated()) { 5680 if (activeTrack->isFastTrack()) { 5681 ALOG_ASSERT(fastTrackToRemove == 0); 5682 fastTrackToRemove = activeTrack; 5683 } 5684 removeTrack_l(activeTrack); 5685 mActiveTracks.remove(activeTrack); 5686 mActiveTracksGen++; 5687 size--; 5688 continue; 5689 } 5690 5691 TrackBase::track_state activeTrackState = activeTrack->mState; 5692 switch (activeTrackState) { 5693 5694 case TrackBase::PAUSING: 5695 mActiveTracks.remove(activeTrack); 5696 mActiveTracksGen++; 5697 doBroadcast = true; 5698 size--; 5699 continue; 5700 5701 case TrackBase::STARTING_1: 5702 sleepUs = 10000; 5703 i++; 5704 continue; 5705 5706 case TrackBase::STARTING_2: 5707 doBroadcast = true; 5708 mStandby = false; 5709 activeTrack->mState = TrackBase::ACTIVE; 5710 break; 5711 5712 case TrackBase::ACTIVE: 5713 break; 5714 5715 case TrackBase::IDLE: 5716 i++; 5717 continue; 5718 5719 default: 5720 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5721 } 5722 5723 activeTracks.add(activeTrack); 5724 i++; 5725 5726 if (activeTrack->isFastTrack()) { 5727 ALOG_ASSERT(!mFastTrackAvail); 5728 ALOG_ASSERT(fastTrack == 0); 5729 fastTrack = activeTrack; 5730 } 5731 } 5732 if (doBroadcast) { 5733 mStartStopCond.broadcast(); 5734 } 5735 5736 // sleep if there are no active tracks to process 5737 if (activeTracks.size() == 0) { 5738 if (sleepUs == 0) { 5739 sleepUs = kRecordThreadSleepUs; 5740 } 5741 continue; 5742 } 5743 sleepUs = 0; 5744 5745 lockEffectChains_l(effectChains); 5746 } 5747 5748 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5749 5750 size_t size = effectChains.size(); 5751 for (size_t i = 0; i < size; i++) { 5752 // thread mutex is not locked, but effect chain is locked 5753 effectChains[i]->process_l(); 5754 } 5755 5756 // Push a new fast capture state if fast capture is not already running, or cblk change 5757 if (mFastCapture != 0) { 5758 FastCaptureStateQueue *sq = mFastCapture->sq(); 5759 FastCaptureState *state = sq->begin(); 5760 bool didModify = false; 5761 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5762 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5763 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5764 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5765 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5766 if (old == -1) { 5767 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5768 } 5769 } 5770 state->mCommand = FastCaptureState::READ_WRITE; 5771#if 0 // FIXME 5772 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5773 FastThreadDumpState::kSamplingNforLowRamDevice : 5774 FastThreadDumpState::kSamplingN); 5775#endif 5776 didModify = true; 5777 } 5778 audio_track_cblk_t *cblkOld = state->mCblk; 5779 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5780 if (cblkNew != cblkOld) { 5781 state->mCblk = cblkNew; 5782 // block until acked if removing a fast track 5783 if (cblkOld != NULL) { 5784 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5785 } 5786 didModify = true; 5787 } 5788 sq->end(didModify); 5789 if (didModify) { 5790 sq->push(block); 5791#if 0 5792 if (kUseFastCapture == FastCapture_Dynamic) { 5793 mNormalSource = mPipeSource; 5794 } 5795#endif 5796 } 5797 } 5798 5799 // now run the fast track destructor with thread mutex unlocked 5800 fastTrackToRemove.clear(); 5801 5802 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5803 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5804 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5805 // If destination is non-contiguous, first read past the nominal end of buffer, then 5806 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5807 5808 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5809 ssize_t framesRead; 5810 5811 // If an NBAIO source is present, use it to read the normal capture's data 5812 if (mPipeSource != 0) { 5813 size_t framesToRead = mBufferSize / mFrameSize; 5814 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5815 framesToRead, AudioBufferProvider::kInvalidPTS); 5816 if (framesRead == 0) { 5817 // since pipe is non-blocking, simulate blocking input 5818 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5819 } 5820 // otherwise use the HAL / AudioStreamIn directly 5821 } else { 5822 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5823 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5824 if (bytesRead < 0) { 5825 framesRead = bytesRead; 5826 } else { 5827 framesRead = bytesRead / mFrameSize; 5828 } 5829 } 5830 5831 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5832 ALOGE("read failed: framesRead=%d", framesRead); 5833 // Force input into standby so that it tries to recover at next read attempt 5834 inputStandBy(); 5835 sleepUs = kRecordThreadSleepUs; 5836 } 5837 if (framesRead <= 0) { 5838 goto unlock; 5839 } 5840 ALOG_ASSERT(framesRead > 0); 5841 5842 if (mTeeSink != 0) { 5843 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5844 } 5845 // If destination is non-contiguous, we now correct for reading past end of buffer. 5846 { 5847 size_t part1 = mRsmpInFramesP2 - rear; 5848 if ((size_t) framesRead > part1) { 5849 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5850 (framesRead - part1) * mFrameSize); 5851 } 5852 } 5853 rear = mRsmpInRear += framesRead; 5854 5855 size = activeTracks.size(); 5856 // loop over each active track 5857 for (size_t i = 0; i < size; i++) { 5858 activeTrack = activeTracks[i]; 5859 5860 // skip fast tracks, as those are handled directly by FastCapture 5861 if (activeTrack->isFastTrack()) { 5862 continue; 5863 } 5864 5865 // TODO: This code probably should be moved to RecordTrack. 5866 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5867 5868 enum { 5869 OVERRUN_UNKNOWN, 5870 OVERRUN_TRUE, 5871 OVERRUN_FALSE 5872 } overrun = OVERRUN_UNKNOWN; 5873 5874 // loop over getNextBuffer to handle circular sink 5875 for (;;) { 5876 5877 activeTrack->mSink.frameCount = ~0; 5878 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5879 size_t framesOut = activeTrack->mSink.frameCount; 5880 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5881 5882 // check available frames and handle overrun conditions 5883 // if the record track isn't draining fast enough. 5884 bool hasOverrun; 5885 size_t framesIn; 5886 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5887 if (hasOverrun) { 5888 overrun = OVERRUN_TRUE; 5889 } 5890 if (framesOut == 0 || framesIn == 0) { 5891 break; 5892 } 5893 5894 // Don't allow framesOut to be larger than what is possible with resampling 5895 // from framesIn. 5896 // This isn't strictly necessary but helps limit buffer resizing in 5897 // RecordBufferConverter. TODO: remove when no longer needed. 5898 framesOut = min(framesOut, 5899 destinationFramesPossible( 5900 framesIn, mSampleRate, activeTrack->mSampleRate)); 5901 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5902 framesOut = activeTrack->mRecordBufferConverter->convert( 5903 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5904 5905 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5906 overrun = OVERRUN_FALSE; 5907 } 5908 5909 if (activeTrack->mFramesToDrop == 0) { 5910 if (framesOut > 0) { 5911 activeTrack->mSink.frameCount = framesOut; 5912 activeTrack->releaseBuffer(&activeTrack->mSink); 5913 } 5914 } else { 5915 // FIXME could do a partial drop of framesOut 5916 if (activeTrack->mFramesToDrop > 0) { 5917 activeTrack->mFramesToDrop -= framesOut; 5918 if (activeTrack->mFramesToDrop <= 0) { 5919 activeTrack->clearSyncStartEvent(); 5920 } 5921 } else { 5922 activeTrack->mFramesToDrop += framesOut; 5923 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5924 activeTrack->mSyncStartEvent->isCancelled()) { 5925 ALOGW("Synced record %s, session %d, trigger session %d", 5926 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5927 activeTrack->sessionId(), 5928 (activeTrack->mSyncStartEvent != 0) ? 5929 activeTrack->mSyncStartEvent->triggerSession() : 0); 5930 activeTrack->clearSyncStartEvent(); 5931 } 5932 } 5933 } 5934 5935 if (framesOut == 0) { 5936 break; 5937 } 5938 } 5939 5940 switch (overrun) { 5941 case OVERRUN_TRUE: 5942 // client isn't retrieving buffers fast enough 5943 if (!activeTrack->setOverflow()) { 5944 nsecs_t now = systemTime(); 5945 // FIXME should lastWarning per track? 5946 if ((now - lastWarning) > kWarningThrottleNs) { 5947 ALOGW("RecordThread: buffer overflow"); 5948 lastWarning = now; 5949 } 5950 } 5951 break; 5952 case OVERRUN_FALSE: 5953 activeTrack->clearOverflow(); 5954 break; 5955 case OVERRUN_UNKNOWN: 5956 break; 5957 } 5958 5959 } 5960 5961unlock: 5962 // enable changes in effect chain 5963 unlockEffectChains(effectChains); 5964 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5965 } 5966 5967 standbyIfNotAlreadyInStandby(); 5968 5969 { 5970 Mutex::Autolock _l(mLock); 5971 for (size_t i = 0; i < mTracks.size(); i++) { 5972 sp<RecordTrack> track = mTracks[i]; 5973 track->invalidate(); 5974 } 5975 mActiveTracks.clear(); 5976 mActiveTracksGen++; 5977 mStartStopCond.broadcast(); 5978 } 5979 5980 releaseWakeLock(); 5981 5982 ALOGV("RecordThread %p exiting", this); 5983 return false; 5984} 5985 5986void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5987{ 5988 if (!mStandby) { 5989 inputStandBy(); 5990 mStandby = true; 5991 } 5992} 5993 5994void AudioFlinger::RecordThread::inputStandBy() 5995{ 5996 // Idle the fast capture if it's currently running 5997 if (mFastCapture != 0) { 5998 FastCaptureStateQueue *sq = mFastCapture->sq(); 5999 FastCaptureState *state = sq->begin(); 6000 if (!(state->mCommand & FastCaptureState::IDLE)) { 6001 state->mCommand = FastCaptureState::COLD_IDLE; 6002 state->mColdFutexAddr = &mFastCaptureFutex; 6003 state->mColdGen++; 6004 mFastCaptureFutex = 0; 6005 sq->end(); 6006 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6007 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6008#if 0 6009 if (kUseFastCapture == FastCapture_Dynamic) { 6010 // FIXME 6011 } 6012#endif 6013#ifdef AUDIO_WATCHDOG 6014 // FIXME 6015#endif 6016 } else { 6017 sq->end(false /*didModify*/); 6018 } 6019 } 6020 mInput->stream->common.standby(&mInput->stream->common); 6021} 6022 6023// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6024sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6025 const sp<AudioFlinger::Client>& client, 6026 uint32_t sampleRate, 6027 audio_format_t format, 6028 audio_channel_mask_t channelMask, 6029 size_t *pFrameCount, 6030 int sessionId, 6031 size_t *notificationFrames, 6032 int uid, 6033 IAudioFlinger::track_flags_t *flags, 6034 pid_t tid, 6035 status_t *status) 6036{ 6037 size_t frameCount = *pFrameCount; 6038 sp<RecordTrack> track; 6039 status_t lStatus; 6040 6041 // client expresses a preference for FAST, but we get the final say 6042 if (*flags & IAudioFlinger::TRACK_FAST) { 6043 if ( 6044 // we formerly checked for a callback handler (non-0 tid), 6045 // but that is no longer required for TRANSFER_OBTAIN mode 6046 // 6047 // frame count is not specified, or is exactly the pipe depth 6048 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6049 // PCM data 6050 audio_is_linear_pcm(format) && 6051 // native format 6052 (format == mFormat) && 6053 // native channel mask 6054 (channelMask == mChannelMask) && 6055 // native hardware sample rate 6056 (sampleRate == mSampleRate) && 6057 // record thread has an associated fast capture 6058 hasFastCapture() && 6059 // there are sufficient fast track slots available 6060 mFastTrackAvail 6061 ) { 6062 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6063 frameCount, mFrameCount); 6064 } else { 6065 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6066 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6067 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6068 frameCount, mFrameCount, mPipeFramesP2, 6069 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6070 hasFastCapture(), tid, mFastTrackAvail); 6071 *flags &= ~IAudioFlinger::TRACK_FAST; 6072 } 6073 } 6074 6075 // compute track buffer size in frames, and suggest the notification frame count 6076 if (*flags & IAudioFlinger::TRACK_FAST) { 6077 // fast track: frame count is exactly the pipe depth 6078 frameCount = mPipeFramesP2; 6079 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6080 *notificationFrames = mFrameCount; 6081 } else { 6082 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6083 // or 20 ms if there is a fast capture 6084 // TODO This could be a roundupRatio inline, and const 6085 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6086 * sampleRate + mSampleRate - 1) / mSampleRate; 6087 // minimum number of notification periods is at least kMinNotifications, 6088 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6089 static const size_t kMinNotifications = 3; 6090 static const uint32_t kMinMs = 30; 6091 // TODO This could be a roundupRatio inline 6092 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6093 // TODO This could be a roundupRatio inline 6094 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6095 maxNotificationFrames; 6096 const size_t minFrameCount = maxNotificationFrames * 6097 max(kMinNotifications, minNotificationsByMs); 6098 frameCount = max(frameCount, minFrameCount); 6099 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6100 *notificationFrames = maxNotificationFrames; 6101 } 6102 } 6103 *pFrameCount = frameCount; 6104 6105 lStatus = initCheck(); 6106 if (lStatus != NO_ERROR) { 6107 ALOGE("createRecordTrack_l() audio driver not initialized"); 6108 goto Exit; 6109 } 6110 6111 { // scope for mLock 6112 Mutex::Autolock _l(mLock); 6113 6114 track = new RecordTrack(this, client, sampleRate, 6115 format, channelMask, frameCount, NULL, sessionId, uid, 6116 *flags, TrackBase::TYPE_DEFAULT); 6117 6118 lStatus = track->initCheck(); 6119 if (lStatus != NO_ERROR) { 6120 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6121 // track must be cleared from the caller as the caller has the AF lock 6122 goto Exit; 6123 } 6124 mTracks.add(track); 6125 6126 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6127 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6128 mAudioFlinger->btNrecIsOff(); 6129 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6130 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6131 6132 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6133 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6134 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6135 // so ask activity manager to do this on our behalf 6136 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6137 } 6138 } 6139 6140 lStatus = NO_ERROR; 6141 6142Exit: 6143 *status = lStatus; 6144 return track; 6145} 6146 6147status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6148 AudioSystem::sync_event_t event, 6149 int triggerSession) 6150{ 6151 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6152 sp<ThreadBase> strongMe = this; 6153 status_t status = NO_ERROR; 6154 6155 if (event == AudioSystem::SYNC_EVENT_NONE) { 6156 recordTrack->clearSyncStartEvent(); 6157 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6158 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6159 triggerSession, 6160 recordTrack->sessionId(), 6161 syncStartEventCallback, 6162 recordTrack); 6163 // Sync event can be cancelled by the trigger session if the track is not in a 6164 // compatible state in which case we start record immediately 6165 if (recordTrack->mSyncStartEvent->isCancelled()) { 6166 recordTrack->clearSyncStartEvent(); 6167 } else { 6168 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6169 recordTrack->mFramesToDrop = - 6170 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6171 } 6172 } 6173 6174 { 6175 // This section is a rendezvous between binder thread executing start() and RecordThread 6176 AutoMutex lock(mLock); 6177 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6178 if (recordTrack->mState == TrackBase::PAUSING) { 6179 ALOGV("active record track PAUSING -> ACTIVE"); 6180 recordTrack->mState = TrackBase::ACTIVE; 6181 } else { 6182 ALOGV("active record track state %d", recordTrack->mState); 6183 } 6184 return status; 6185 } 6186 6187 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6188 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6189 // or using a separate command thread 6190 recordTrack->mState = TrackBase::STARTING_1; 6191 mActiveTracks.add(recordTrack); 6192 mActiveTracksGen++; 6193 status_t status = NO_ERROR; 6194 if (recordTrack->isExternalTrack()) { 6195 mLock.unlock(); 6196 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6197 mLock.lock(); 6198 // FIXME should verify that recordTrack is still in mActiveTracks 6199 if (status != NO_ERROR) { 6200 mActiveTracks.remove(recordTrack); 6201 mActiveTracksGen++; 6202 recordTrack->clearSyncStartEvent(); 6203 ALOGV("RecordThread::start error %d", status); 6204 return status; 6205 } 6206 } 6207 // Catch up with current buffer indices if thread is already running. 6208 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6209 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6210 // see previously buffered data before it called start(), but with greater risk of overrun. 6211 6212 recordTrack->mResamplerBufferProvider->reset(); 6213 // clear any converter state as new data will be discontinuous 6214 recordTrack->mRecordBufferConverter->reset(); 6215 recordTrack->mState = TrackBase::STARTING_2; 6216 // signal thread to start 6217 mWaitWorkCV.broadcast(); 6218 if (mActiveTracks.indexOf(recordTrack) < 0) { 6219 ALOGV("Record failed to start"); 6220 status = BAD_VALUE; 6221 goto startError; 6222 } 6223 return status; 6224 } 6225 6226startError: 6227 if (recordTrack->isExternalTrack()) { 6228 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6229 } 6230 recordTrack->clearSyncStartEvent(); 6231 // FIXME I wonder why we do not reset the state here? 6232 return status; 6233} 6234 6235void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6236{ 6237 sp<SyncEvent> strongEvent = event.promote(); 6238 6239 if (strongEvent != 0) { 6240 sp<RefBase> ptr = strongEvent->cookie().promote(); 6241 if (ptr != 0) { 6242 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6243 recordTrack->handleSyncStartEvent(strongEvent); 6244 } 6245 } 6246} 6247 6248bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6249 ALOGV("RecordThread::stop"); 6250 AutoMutex _l(mLock); 6251 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6252 return false; 6253 } 6254 // note that threadLoop may still be processing the track at this point [without lock] 6255 recordTrack->mState = TrackBase::PAUSING; 6256 // do not wait for mStartStopCond if exiting 6257 if (exitPending()) { 6258 return true; 6259 } 6260 // FIXME incorrect usage of wait: no explicit predicate or loop 6261 mStartStopCond.wait(mLock); 6262 // if we have been restarted, recordTrack is in mActiveTracks here 6263 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6264 ALOGV("Record stopped OK"); 6265 return true; 6266 } 6267 return false; 6268} 6269 6270bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6271{ 6272 return false; 6273} 6274 6275status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6276{ 6277#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6278 if (!isValidSyncEvent(event)) { 6279 return BAD_VALUE; 6280 } 6281 6282 int eventSession = event->triggerSession(); 6283 status_t ret = NAME_NOT_FOUND; 6284 6285 Mutex::Autolock _l(mLock); 6286 6287 for (size_t i = 0; i < mTracks.size(); i++) { 6288 sp<RecordTrack> track = mTracks[i]; 6289 if (eventSession == track->sessionId()) { 6290 (void) track->setSyncEvent(event); 6291 ret = NO_ERROR; 6292 } 6293 } 6294 return ret; 6295#else 6296 return BAD_VALUE; 6297#endif 6298} 6299 6300// destroyTrack_l() must be called with ThreadBase::mLock held 6301void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6302{ 6303 track->terminate(); 6304 track->mState = TrackBase::STOPPED; 6305 // active tracks are removed by threadLoop() 6306 if (mActiveTracks.indexOf(track) < 0) { 6307 removeTrack_l(track); 6308 } 6309} 6310 6311void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6312{ 6313 mTracks.remove(track); 6314 // need anything related to effects here? 6315 if (track->isFastTrack()) { 6316 ALOG_ASSERT(!mFastTrackAvail); 6317 mFastTrackAvail = true; 6318 } 6319} 6320 6321void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6322{ 6323 dumpInternals(fd, args); 6324 dumpTracks(fd, args); 6325 dumpEffectChains(fd, args); 6326} 6327 6328void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6329{ 6330 dprintf(fd, "\nInput thread %p:\n", this); 6331 6332 dumpBase(fd, args); 6333 6334 if (mActiveTracks.size() == 0) { 6335 dprintf(fd, " No active record clients\n"); 6336 } 6337 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6338 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6339 6340 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6341 const FastCaptureDumpState copy(mFastCaptureDumpState); 6342 copy.dump(fd); 6343} 6344 6345void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6346{ 6347 const size_t SIZE = 256; 6348 char buffer[SIZE]; 6349 String8 result; 6350 6351 size_t numtracks = mTracks.size(); 6352 size_t numactive = mActiveTracks.size(); 6353 size_t numactiveseen = 0; 6354 dprintf(fd, " %d Tracks", numtracks); 6355 if (numtracks) { 6356 dprintf(fd, " of which %d are active\n", numactive); 6357 RecordTrack::appendDumpHeader(result); 6358 for (size_t i = 0; i < numtracks ; ++i) { 6359 sp<RecordTrack> track = mTracks[i]; 6360 if (track != 0) { 6361 bool active = mActiveTracks.indexOf(track) >= 0; 6362 if (active) { 6363 numactiveseen++; 6364 } 6365 track->dump(buffer, SIZE, active); 6366 result.append(buffer); 6367 } 6368 } 6369 } else { 6370 dprintf(fd, "\n"); 6371 } 6372 6373 if (numactiveseen != numactive) { 6374 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6375 " not in the track list\n"); 6376 result.append(buffer); 6377 RecordTrack::appendDumpHeader(result); 6378 for (size_t i = 0; i < numactive; ++i) { 6379 sp<RecordTrack> track = mActiveTracks[i]; 6380 if (mTracks.indexOf(track) < 0) { 6381 track->dump(buffer, SIZE, true); 6382 result.append(buffer); 6383 } 6384 } 6385 6386 } 6387 write(fd, result.string(), result.size()); 6388} 6389 6390 6391void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6392{ 6393 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6394 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6395 mRsmpInFront = recordThread->mRsmpInRear; 6396 mRsmpInUnrel = 0; 6397} 6398 6399void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6400 size_t *framesAvailable, bool *hasOverrun) 6401{ 6402 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6403 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6404 const int32_t rear = recordThread->mRsmpInRear; 6405 const int32_t front = mRsmpInFront; 6406 const ssize_t filled = rear - front; 6407 6408 size_t framesIn; 6409 bool overrun = false; 6410 if (filled < 0) { 6411 // should not happen, but treat like a massive overrun and re-sync 6412 framesIn = 0; 6413 mRsmpInFront = rear; 6414 overrun = true; 6415 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6416 framesIn = (size_t) filled; 6417 } else { 6418 // client is not keeping up with server, but give it latest data 6419 framesIn = recordThread->mRsmpInFrames; 6420 mRsmpInFront = /* front = */ rear - framesIn; 6421 overrun = true; 6422 } 6423 if (framesAvailable != NULL) { 6424 *framesAvailable = framesIn; 6425 } 6426 if (hasOverrun != NULL) { 6427 *hasOverrun = overrun; 6428 } 6429} 6430 6431// AudioBufferProvider interface 6432status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6433 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6434{ 6435 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6436 if (threadBase == 0) { 6437 buffer->frameCount = 0; 6438 buffer->raw = NULL; 6439 return NOT_ENOUGH_DATA; 6440 } 6441 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6442 int32_t rear = recordThread->mRsmpInRear; 6443 int32_t front = mRsmpInFront; 6444 ssize_t filled = rear - front; 6445 // FIXME should not be P2 (don't want to increase latency) 6446 // FIXME if client not keeping up, discard 6447 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6448 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6449 front &= recordThread->mRsmpInFramesP2 - 1; 6450 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6451 if (part1 > (size_t) filled) { 6452 part1 = filled; 6453 } 6454 size_t ask = buffer->frameCount; 6455 ALOG_ASSERT(ask > 0); 6456 if (part1 > ask) { 6457 part1 = ask; 6458 } 6459 if (part1 == 0) { 6460 // out of data is fine since the resampler will return a short-count. 6461 buffer->raw = NULL; 6462 buffer->frameCount = 0; 6463 mRsmpInUnrel = 0; 6464 return NOT_ENOUGH_DATA; 6465 } 6466 6467 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6468 buffer->frameCount = part1; 6469 mRsmpInUnrel = part1; 6470 return NO_ERROR; 6471} 6472 6473// AudioBufferProvider interface 6474void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6475 AudioBufferProvider::Buffer* buffer) 6476{ 6477 size_t stepCount = buffer->frameCount; 6478 if (stepCount == 0) { 6479 return; 6480 } 6481 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6482 mRsmpInUnrel -= stepCount; 6483 mRsmpInFront += stepCount; 6484 buffer->raw = NULL; 6485 buffer->frameCount = 0; 6486} 6487 6488AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6489 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6490 uint32_t srcSampleRate, 6491 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6492 uint32_t dstSampleRate) : 6493 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6494 // mSrcFormat 6495 // mSrcSampleRate 6496 // mDstChannelMask 6497 // mDstFormat 6498 // mDstSampleRate 6499 // mSrcChannelCount 6500 // mDstChannelCount 6501 // mDstFrameSize 6502 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6503 mResampler(NULL), 6504 mIsLegacyDownmix(false), 6505 mIsLegacyUpmix(false), 6506 mRequiresFloat(false), 6507 mInputConverterProvider(NULL) 6508{ 6509 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6510 dstChannelMask, dstFormat, dstSampleRate); 6511} 6512 6513AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6514 free(mBuf); 6515 delete mResampler; 6516 delete mInputConverterProvider; 6517} 6518 6519size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6520 AudioBufferProvider *provider, size_t frames) 6521{ 6522 if (mInputConverterProvider != NULL) { 6523 mInputConverterProvider->setBufferProvider(provider); 6524 provider = mInputConverterProvider; 6525 } 6526 6527 if (mResampler == NULL) { 6528 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6529 mSrcSampleRate, mSrcFormat, mDstFormat); 6530 6531 AudioBufferProvider::Buffer buffer; 6532 for (size_t i = frames; i > 0; ) { 6533 buffer.frameCount = i; 6534 status_t status = provider->getNextBuffer(&buffer, 0); 6535 if (status != OK || buffer.frameCount == 0) { 6536 frames -= i; // cannot fill request. 6537 break; 6538 } 6539 // format convert to destination buffer 6540 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6541 6542 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6543 i -= buffer.frameCount; 6544 provider->releaseBuffer(&buffer); 6545 } 6546 } else { 6547 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6548 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6549 6550 // reallocate buffer if needed 6551 if (mBufFrameSize != 0 && mBufFrames < frames) { 6552 free(mBuf); 6553 mBufFrames = frames; 6554 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6555 } 6556 // resampler accumulates, but we only have one source track 6557 memset(mBuf, 0, frames * mBufFrameSize); 6558 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6559 // format convert to destination buffer 6560 convertResampler(dst, mBuf, frames); 6561 } 6562 return frames; 6563} 6564 6565status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6566 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6567 uint32_t srcSampleRate, 6568 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6569 uint32_t dstSampleRate) 6570{ 6571 // quick evaluation if there is any change. 6572 if (mSrcFormat == srcFormat 6573 && mSrcChannelMask == srcChannelMask 6574 && mSrcSampleRate == srcSampleRate 6575 && mDstFormat == dstFormat 6576 && mDstChannelMask == dstChannelMask 6577 && mDstSampleRate == dstSampleRate) { 6578 return NO_ERROR; 6579 } 6580 6581 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6582 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6583 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6584 const bool valid = 6585 audio_is_input_channel(srcChannelMask) 6586 && audio_is_input_channel(dstChannelMask) 6587 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6588 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6589 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6590 ; // no upsampling checks for now 6591 if (!valid) { 6592 return BAD_VALUE; 6593 } 6594 6595 mSrcFormat = srcFormat; 6596 mSrcChannelMask = srcChannelMask; 6597 mSrcSampleRate = srcSampleRate; 6598 mDstFormat = dstFormat; 6599 mDstChannelMask = dstChannelMask; 6600 mDstSampleRate = dstSampleRate; 6601 6602 // compute derived parameters 6603 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6604 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6605 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6606 6607 // do we need to resample? 6608 delete mResampler; 6609 mResampler = NULL; 6610 if (mSrcSampleRate != mDstSampleRate) { 6611 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6612 mSrcChannelCount, mDstSampleRate); 6613 mResampler->setSampleRate(mSrcSampleRate); 6614 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6615 } 6616 6617 // are we running legacy channel conversion modes? 6618 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6619 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6620 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6621 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6622 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6623 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6624 6625 // do we need to process in float? 6626 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6627 6628 // do we need a staging buffer to convert for destination (we can still optimize this)? 6629 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6630 if (mResampler != NULL) { 6631 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6632 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6633 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6634 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6635 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6636 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6637 } else { 6638 mBufFrameSize = 0; 6639 } 6640 mBufFrames = 0; // force the buffer to be resized. 6641 6642 // do we need an input converter buffer provider to give us float? 6643 delete mInputConverterProvider; 6644 mInputConverterProvider = NULL; 6645 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6646 mInputConverterProvider = new ReformatBufferProvider( 6647 audio_channel_count_from_in_mask(mSrcChannelMask), 6648 mSrcFormat, 6649 AUDIO_FORMAT_PCM_FLOAT, 6650 256 /* provider buffer frame count */); 6651 } 6652 6653 // do we need a remixer to do channel mask conversion 6654 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6655 (void) memcpy_by_index_array_initialization_from_channel_mask( 6656 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6657 } 6658 return NO_ERROR; 6659} 6660 6661void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6662 void *dst, const void *src, size_t frames) 6663{ 6664 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6665 if (mBufFrameSize != 0 && mBufFrames < frames) { 6666 free(mBuf); 6667 mBufFrames = frames; 6668 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6669 } 6670 // do we need to do legacy upmix and downmix? 6671 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6672 void *dstBuf = mBuf != NULL ? mBuf : dst; 6673 if (mIsLegacyUpmix) { 6674 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6675 (const float *)src, frames); 6676 } else /*mIsLegacyDownmix */ { 6677 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6678 (const float *)src, frames); 6679 } 6680 if (mBuf != NULL) { 6681 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6682 frames * mDstChannelCount); 6683 } 6684 return; 6685 } 6686 // do we need to do channel mask conversion? 6687 if (mSrcChannelMask != mDstChannelMask) { 6688 void *dstBuf = mBuf != NULL ? mBuf : dst; 6689 memcpy_by_index_array(dstBuf, mDstChannelCount, 6690 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6691 if (dstBuf == dst) { 6692 return; // format is the same 6693 } 6694 } 6695 // convert to destination buffer 6696 const void *convertBuf = mBuf != NULL ? mBuf : src; 6697 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6698 frames * mDstChannelCount); 6699} 6700 6701void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6702 void *dst, /*not-a-const*/ void *src, size_t frames) 6703{ 6704 // src buffer format is ALWAYS float when entering this routine 6705 if (mIsLegacyUpmix) { 6706 ; // mono to stereo already handled by resampler 6707 } else if (mIsLegacyDownmix 6708 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6709 // the resampler outputs stereo for mono input channel (a feature?) 6710 // must convert to mono 6711 downmix_to_mono_float_from_stereo_float((float *)src, 6712 (const float *)src, frames); 6713 } else if (mSrcChannelMask != mDstChannelMask) { 6714 // convert to mono channel again for channel mask conversion (could be skipped 6715 // with further optimization). 6716 if (mSrcChannelCount == 1) { 6717 downmix_to_mono_float_from_stereo_float((float *)src, 6718 (const float *)src, frames); 6719 } 6720 // convert to destination format (in place, OK as float is larger than other types) 6721 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6722 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6723 frames * mSrcChannelCount); 6724 } 6725 // channel convert and save to dst 6726 memcpy_by_index_array(dst, mDstChannelCount, 6727 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6728 return; 6729 } 6730 // convert to destination format and save to dst 6731 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6732 frames * mDstChannelCount); 6733} 6734 6735bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6736 status_t& status) 6737{ 6738 bool reconfig = false; 6739 6740 status = NO_ERROR; 6741 6742 audio_format_t reqFormat = mFormat; 6743 uint32_t samplingRate = mSampleRate; 6744 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6745 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6746 6747 AudioParameter param = AudioParameter(keyValuePair); 6748 int value; 6749 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6750 // channel count change can be requested. Do we mandate the first client defines the 6751 // HAL sampling rate and channel count or do we allow changes on the fly? 6752 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6753 samplingRate = value; 6754 reconfig = true; 6755 } 6756 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6757 if (!audio_is_linear_pcm((audio_format_t) value)) { 6758 status = BAD_VALUE; 6759 } else { 6760 reqFormat = (audio_format_t) value; 6761 reconfig = true; 6762 } 6763 } 6764 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6765 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6766 if (!audio_is_input_channel(mask) || 6767 audio_channel_count_from_in_mask(mask) > FCC_8) { 6768 status = BAD_VALUE; 6769 } else { 6770 channelMask = mask; 6771 reconfig = true; 6772 } 6773 } 6774 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6775 // do not accept frame count changes if tracks are open as the track buffer 6776 // size depends on frame count and correct behavior would not be guaranteed 6777 // if frame count is changed after track creation 6778 if (mActiveTracks.size() > 0) { 6779 status = INVALID_OPERATION; 6780 } else { 6781 reconfig = true; 6782 } 6783 } 6784 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6785 // forward device change to effects that have requested to be 6786 // aware of attached audio device. 6787 for (size_t i = 0; i < mEffectChains.size(); i++) { 6788 mEffectChains[i]->setDevice_l(value); 6789 } 6790 6791 // store input device and output device but do not forward output device to audio HAL. 6792 // Note that status is ignored by the caller for output device 6793 // (see AudioFlinger::setParameters() 6794 if (audio_is_output_devices(value)) { 6795 mOutDevice = value; 6796 status = BAD_VALUE; 6797 } else { 6798 mInDevice = value; 6799 // disable AEC and NS if the device is a BT SCO headset supporting those 6800 // pre processings 6801 if (mTracks.size() > 0) { 6802 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6803 mAudioFlinger->btNrecIsOff(); 6804 for (size_t i = 0; i < mTracks.size(); i++) { 6805 sp<RecordTrack> track = mTracks[i]; 6806 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6807 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6808 } 6809 } 6810 } 6811 } 6812 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6813 mAudioSource != (audio_source_t)value) { 6814 // forward device change to effects that have requested to be 6815 // aware of attached audio device. 6816 for (size_t i = 0; i < mEffectChains.size(); i++) { 6817 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6818 } 6819 mAudioSource = (audio_source_t)value; 6820 } 6821 6822 if (status == NO_ERROR) { 6823 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6824 keyValuePair.string()); 6825 if (status == INVALID_OPERATION) { 6826 inputStandBy(); 6827 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6828 keyValuePair.string()); 6829 } 6830 if (reconfig) { 6831 if (status == BAD_VALUE && 6832 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6833 audio_is_linear_pcm(reqFormat) && 6834 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6835 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6836 audio_channel_count_from_in_mask( 6837 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6838 status = NO_ERROR; 6839 } 6840 if (status == NO_ERROR) { 6841 readInputParameters_l(); 6842 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6843 } 6844 } 6845 } 6846 6847 return reconfig; 6848} 6849 6850String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6851{ 6852 Mutex::Autolock _l(mLock); 6853 if (initCheck() != NO_ERROR) { 6854 return String8(); 6855 } 6856 6857 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6858 const String8 out_s8(s); 6859 free(s); 6860 return out_s8; 6861} 6862 6863void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event) { 6864 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6865 6866 desc->mIoHandle = mId; 6867 6868 switch (event) { 6869 case AUDIO_INPUT_OPENED: 6870 case AUDIO_INPUT_CONFIG_CHANGED: 6871 desc->mPatch = mPatch; 6872 desc->mChannelMask = mChannelMask; 6873 desc->mSamplingRate = mSampleRate; 6874 desc->mFormat = mFormat; 6875 desc->mFrameCount = mFrameCount; 6876 desc->mLatency = 0; 6877 break; 6878 6879 case AUDIO_INPUT_CLOSED: 6880 default: 6881 break; 6882 } 6883 mAudioFlinger->ioConfigChanged(event, desc); 6884} 6885 6886void AudioFlinger::RecordThread::readInputParameters_l() 6887{ 6888 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6889 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6890 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6891 if (mChannelCount > FCC_8) { 6892 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6893 } 6894 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6895 mFormat = mHALFormat; 6896 if (!audio_is_linear_pcm(mFormat)) { 6897 ALOGE("HAL format %#x is not linear pcm", mFormat); 6898 } 6899 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6900 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6901 mFrameCount = mBufferSize / mFrameSize; 6902 // This is the formula for calculating the temporary buffer size. 6903 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6904 // 1 full output buffer, regardless of the alignment of the available input. 6905 // The value is somewhat arbitrary, and could probably be even larger. 6906 // A larger value should allow more old data to be read after a track calls start(), 6907 // without increasing latency. 6908 // 6909 // Note this is independent of the maximum downsampling ratio permitted for capture. 6910 mRsmpInFrames = mFrameCount * 7; 6911 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6912 free(mRsmpInBuffer); 6913 6914 // TODO optimize audio capture buffer sizes ... 6915 // Here we calculate the size of the sliding buffer used as a source 6916 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6917 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6918 // be better to have it derived from the pipe depth in the long term. 6919 // The current value is higher than necessary. However it should not add to latency. 6920 6921 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6922 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6923 6924 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6925 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6926} 6927 6928uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6929{ 6930 Mutex::Autolock _l(mLock); 6931 if (initCheck() != NO_ERROR) { 6932 return 0; 6933 } 6934 6935 return mInput->stream->get_input_frames_lost(mInput->stream); 6936} 6937 6938uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6939{ 6940 Mutex::Autolock _l(mLock); 6941 uint32_t result = 0; 6942 if (getEffectChain_l(sessionId) != 0) { 6943 result = EFFECT_SESSION; 6944 } 6945 6946 for (size_t i = 0; i < mTracks.size(); ++i) { 6947 if (sessionId == mTracks[i]->sessionId()) { 6948 result |= TRACK_SESSION; 6949 break; 6950 } 6951 } 6952 6953 return result; 6954} 6955 6956KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6957{ 6958 KeyedVector<int, bool> ids; 6959 Mutex::Autolock _l(mLock); 6960 for (size_t j = 0; j < mTracks.size(); ++j) { 6961 sp<RecordThread::RecordTrack> track = mTracks[j]; 6962 int sessionId = track->sessionId(); 6963 if (ids.indexOfKey(sessionId) < 0) { 6964 ids.add(sessionId, true); 6965 } 6966 } 6967 return ids; 6968} 6969 6970AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6971{ 6972 Mutex::Autolock _l(mLock); 6973 AudioStreamIn *input = mInput; 6974 mInput = NULL; 6975 return input; 6976} 6977 6978// this method must always be called either with ThreadBase mLock held or inside the thread loop 6979audio_stream_t* AudioFlinger::RecordThread::stream() const 6980{ 6981 if (mInput == NULL) { 6982 return NULL; 6983 } 6984 return &mInput->stream->common; 6985} 6986 6987status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6988{ 6989 // only one chain per input thread 6990 if (mEffectChains.size() != 0) { 6991 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6992 return INVALID_OPERATION; 6993 } 6994 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6995 chain->setThread(this); 6996 chain->setInBuffer(NULL); 6997 chain->setOutBuffer(NULL); 6998 6999 checkSuspendOnAddEffectChain_l(chain); 7000 7001 // make sure enabled pre processing effects state is communicated to the HAL as we 7002 // just moved them to a new input stream. 7003 chain->syncHalEffectsState(); 7004 7005 mEffectChains.add(chain); 7006 7007 return NO_ERROR; 7008} 7009 7010size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7011{ 7012 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7013 ALOGW_IF(mEffectChains.size() != 1, 7014 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7015 chain.get(), mEffectChains.size(), this); 7016 if (mEffectChains.size() == 1) { 7017 mEffectChains.removeAt(0); 7018 } 7019 return 0; 7020} 7021 7022status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7023 audio_patch_handle_t *handle) 7024{ 7025 status_t status = NO_ERROR; 7026 7027 // store new device and send to effects 7028 mInDevice = patch->sources[0].ext.device.type; 7029 mPatch = *patch; 7030 for (size_t i = 0; i < mEffectChains.size(); i++) { 7031 mEffectChains[i]->setDevice_l(mInDevice); 7032 } 7033 7034 // disable AEC and NS if the device is a BT SCO headset supporting those 7035 // pre processings 7036 if (mTracks.size() > 0) { 7037 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7038 mAudioFlinger->btNrecIsOff(); 7039 for (size_t i = 0; i < mTracks.size(); i++) { 7040 sp<RecordTrack> track = mTracks[i]; 7041 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7042 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7043 } 7044 } 7045 7046 // store new source and send to effects 7047 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7048 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7049 for (size_t i = 0; i < mEffectChains.size(); i++) { 7050 mEffectChains[i]->setAudioSource_l(mAudioSource); 7051 } 7052 } 7053 7054 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7055 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7056 status = hwDevice->create_audio_patch(hwDevice, 7057 patch->num_sources, 7058 patch->sources, 7059 patch->num_sinks, 7060 patch->sinks, 7061 handle); 7062 } else { 7063 char *address; 7064 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7065 address = audio_device_address_to_parameter( 7066 patch->sources[0].ext.device.type, 7067 patch->sources[0].ext.device.address); 7068 } else { 7069 address = (char *)calloc(1, 1); 7070 } 7071 AudioParameter param = AudioParameter(String8(address)); 7072 free(address); 7073 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7074 (int)patch->sources[0].ext.device.type); 7075 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7076 (int)patch->sinks[0].ext.mix.usecase.source); 7077 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7078 param.toString().string()); 7079 *handle = AUDIO_PATCH_HANDLE_NONE; 7080 } 7081 7082 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7083 7084 return status; 7085} 7086 7087status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7088{ 7089 status_t status = NO_ERROR; 7090 7091 mInDevice = AUDIO_DEVICE_NONE; 7092 7093 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7094 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7095 status = hwDevice->release_audio_patch(hwDevice, handle); 7096 } else { 7097 AudioParameter param; 7098 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7099 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7100 param.toString().string()); 7101 } 7102 return status; 7103} 7104 7105void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7106{ 7107 Mutex::Autolock _l(mLock); 7108 mTracks.add(record); 7109} 7110 7111void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7112{ 7113 Mutex::Autolock _l(mLock); 7114 destroyTrack_l(record); 7115} 7116 7117void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7118{ 7119 ThreadBase::getAudioPortConfig(config); 7120 config->role = AUDIO_PORT_ROLE_SINK; 7121 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7122 config->ext.mix.usecase.source = mAudioSource; 7123} 7124 7125} // namespace android 7126