Threads.cpp revision ad9ef61e770c0751a9983aa5c9844dfeb9ed665b
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 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2198 2199 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2200 // Originally this was int16_t[] array, need to remove legacy implications. 2201 free(mSinkBuffer); 2202 mSinkBuffer = NULL; 2203 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2204 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2205 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2206 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2207 2208 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2209 // drives the output. 2210 free(mMixerBuffer); 2211 mMixerBuffer = NULL; 2212 if (mMixerBufferEnabled) { 2213 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2214 mMixerBufferSize = mNormalFrameCount * mChannelCount 2215 * audio_bytes_per_sample(mMixerBufferFormat); 2216 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2217 } 2218 free(mEffectBuffer); 2219 mEffectBuffer = NULL; 2220 if (mEffectBufferEnabled) { 2221 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2222 mEffectBufferSize = mNormalFrameCount * mChannelCount 2223 * audio_bytes_per_sample(mEffectBufferFormat); 2224 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2225 } 2226 2227 // force reconfiguration of effect chains and engines to take new buffer size and audio 2228 // parameters into account 2229 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2230 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2231 // matter. 2232 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2233 Vector< sp<EffectChain> > effectChains = mEffectChains; 2234 for (size_t i = 0; i < effectChains.size(); i ++) { 2235 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2236 } 2237} 2238 2239 2240status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2241{ 2242 if (halFrames == NULL || dspFrames == NULL) { 2243 return BAD_VALUE; 2244 } 2245 Mutex::Autolock _l(mLock); 2246 if (initCheck() != NO_ERROR) { 2247 return INVALID_OPERATION; 2248 } 2249 size_t framesWritten = mBytesWritten / mFrameSize; 2250 *halFrames = framesWritten; 2251 2252 if (isSuspended()) { 2253 // return an estimation of rendered frames when the output is suspended 2254 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2255 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2256 return NO_ERROR; 2257 } else { 2258 status_t status; 2259 uint32_t frames; 2260 status = mOutput->getRenderPosition(&frames); 2261 *dspFrames = (size_t)frames; 2262 return status; 2263 } 2264} 2265 2266uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2267{ 2268 Mutex::Autolock _l(mLock); 2269 uint32_t result = 0; 2270 if (getEffectChain_l(sessionId) != 0) { 2271 result = EFFECT_SESSION; 2272 } 2273 2274 for (size_t i = 0; i < mTracks.size(); ++i) { 2275 sp<Track> track = mTracks[i]; 2276 if (sessionId == track->sessionId() && !track->isInvalid()) { 2277 result |= TRACK_SESSION; 2278 break; 2279 } 2280 } 2281 2282 return result; 2283} 2284 2285uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2286{ 2287 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2288 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2289 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2290 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2291 } 2292 for (size_t i = 0; i < mTracks.size(); i++) { 2293 sp<Track> track = mTracks[i]; 2294 if (sessionId == track->sessionId() && !track->isInvalid()) { 2295 return AudioSystem::getStrategyForStream(track->streamType()); 2296 } 2297 } 2298 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2299} 2300 2301 2302AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2303{ 2304 Mutex::Autolock _l(mLock); 2305 return mOutput; 2306} 2307 2308AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2309{ 2310 Mutex::Autolock _l(mLock); 2311 AudioStreamOut *output = mOutput; 2312 mOutput = NULL; 2313 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2314 // must push a NULL and wait for ack 2315 mOutputSink.clear(); 2316 mPipeSink.clear(); 2317 mNormalSink.clear(); 2318 return output; 2319} 2320 2321// this method must always be called either with ThreadBase mLock held or inside the thread loop 2322audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2323{ 2324 if (mOutput == NULL) { 2325 return NULL; 2326 } 2327 return &mOutput->stream->common; 2328} 2329 2330uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2331{ 2332 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2333} 2334 2335status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2336{ 2337 if (!isValidSyncEvent(event)) { 2338 return BAD_VALUE; 2339 } 2340 2341 Mutex::Autolock _l(mLock); 2342 2343 for (size_t i = 0; i < mTracks.size(); ++i) { 2344 sp<Track> track = mTracks[i]; 2345 if (event->triggerSession() == track->sessionId()) { 2346 (void) track->setSyncEvent(event); 2347 return NO_ERROR; 2348 } 2349 } 2350 2351 return NAME_NOT_FOUND; 2352} 2353 2354bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2355{ 2356 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2357} 2358 2359void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2360 const Vector< sp<Track> >& tracksToRemove) 2361{ 2362 size_t count = tracksToRemove.size(); 2363 if (count > 0) { 2364 for (size_t i = 0 ; i < count ; i++) { 2365 const sp<Track>& track = tracksToRemove.itemAt(i); 2366 if (track->isExternalTrack()) { 2367 AudioSystem::stopOutput(mId, track->streamType(), 2368 (audio_session_t)track->sessionId()); 2369#ifdef ADD_BATTERY_DATA 2370 // to track the speaker usage 2371 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2372#endif 2373 if (track->isTerminated()) { 2374 AudioSystem::releaseOutput(mId, track->streamType(), 2375 (audio_session_t)track->sessionId()); 2376 } 2377 } 2378 } 2379 } 2380} 2381 2382void AudioFlinger::PlaybackThread::checkSilentMode_l() 2383{ 2384 if (!mMasterMute) { 2385 char value[PROPERTY_VALUE_MAX]; 2386 if (property_get("ro.audio.silent", value, "0") > 0) { 2387 char *endptr; 2388 unsigned long ul = strtoul(value, &endptr, 0); 2389 if (*endptr == '\0' && ul != 0) { 2390 ALOGD("Silence is golden"); 2391 // The setprop command will not allow a property to be changed after 2392 // the first time it is set, so we don't have to worry about un-muting. 2393 setMasterMute_l(true); 2394 } 2395 } 2396 } 2397} 2398 2399// shared by MIXER and DIRECT, overridden by DUPLICATING 2400ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2401{ 2402 // FIXME rewrite to reduce number of system calls 2403 mLastWriteTime = systemTime(); 2404 mInWrite = true; 2405 ssize_t bytesWritten; 2406 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2407 2408 // If an NBAIO sink is present, use it to write the normal mixer's submix 2409 if (mNormalSink != 0) { 2410 2411 const size_t count = mBytesRemaining / mFrameSize; 2412 2413 ATRACE_BEGIN("write"); 2414 // update the setpoint when AudioFlinger::mScreenState changes 2415 uint32_t screenState = AudioFlinger::mScreenState; 2416 if (screenState != mScreenState) { 2417 mScreenState = screenState; 2418 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2419 if (pipe != NULL) { 2420 pipe->setAvgFrames((mScreenState & 1) ? 2421 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2422 } 2423 } 2424 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2425 ATRACE_END(); 2426 if (framesWritten > 0) { 2427 bytesWritten = framesWritten * mFrameSize; 2428 } else { 2429 bytesWritten = framesWritten; 2430 } 2431 mLatchDValid = false; 2432 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2433 if (status == NO_ERROR) { 2434 size_t totalFramesWritten = mNormalSink->framesWritten(); 2435 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2436 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2437 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2438 mLatchDValid = true; 2439 } 2440 } 2441 // otherwise use the HAL / AudioStreamOut directly 2442 } else { 2443 // Direct output and offload threads 2444 2445 if (mUseAsyncWrite) { 2446 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2447 mWriteAckSequence += 2; 2448 mWriteAckSequence |= 1; 2449 ALOG_ASSERT(mCallbackThread != 0); 2450 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2451 } 2452 // FIXME We should have an implementation of timestamps for direct output threads. 2453 // They are used e.g for multichannel PCM playback over HDMI. 2454 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2455 if (mUseAsyncWrite && 2456 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2457 // do not wait for async callback in case of error of full write 2458 mWriteAckSequence &= ~1; 2459 ALOG_ASSERT(mCallbackThread != 0); 2460 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2461 } 2462 } 2463 2464 mNumWrites++; 2465 mInWrite = false; 2466 mStandby = false; 2467 return bytesWritten; 2468} 2469 2470void AudioFlinger::PlaybackThread::threadLoop_drain() 2471{ 2472 if (mOutput->stream->drain) { 2473 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2474 if (mUseAsyncWrite) { 2475 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2476 mDrainSequence |= 1; 2477 ALOG_ASSERT(mCallbackThread != 0); 2478 mCallbackThread->setDraining(mDrainSequence); 2479 } 2480 mOutput->stream->drain(mOutput->stream, 2481 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2482 : AUDIO_DRAIN_ALL); 2483 } 2484} 2485 2486void AudioFlinger::PlaybackThread::threadLoop_exit() 2487{ 2488 { 2489 Mutex::Autolock _l(mLock); 2490 for (size_t i = 0; i < mTracks.size(); i++) { 2491 sp<Track> track = mTracks[i]; 2492 track->invalidate(); 2493 } 2494 } 2495} 2496 2497/* 2498The derived values that are cached: 2499 - mSinkBufferSize from frame count * frame size 2500 - mActiveSleepTimeUs from activeSleepTimeUs() 2501 - mIdleSleepTimeUs from idleSleepTimeUs() 2502 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) 2503 - maxPeriod from frame count and sample rate (MIXER only) 2504 2505The parameters that affect these derived values are: 2506 - frame count 2507 - frame size 2508 - sample rate 2509 - device type: A2DP or not 2510 - device latency 2511 - format: PCM or not 2512 - active sleep time 2513 - idle sleep time 2514*/ 2515 2516void AudioFlinger::PlaybackThread::cacheParameters_l() 2517{ 2518 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2519 mActiveSleepTimeUs = activeSleepTimeUs(); 2520 mIdleSleepTimeUs = idleSleepTimeUs(); 2521} 2522 2523void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2524{ 2525 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2526 this, streamType, mTracks.size()); 2527 Mutex::Autolock _l(mLock); 2528 2529 size_t size = mTracks.size(); 2530 for (size_t i = 0; i < size; i++) { 2531 sp<Track> t = mTracks[i]; 2532 if (t->streamType() == streamType) { 2533 t->invalidate(); 2534 } 2535 } 2536} 2537 2538status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2539{ 2540 int session = chain->sessionId(); 2541 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2542 ? mEffectBuffer : mSinkBuffer); 2543 bool ownsBuffer = false; 2544 2545 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2546 if (session > 0) { 2547 // Only one effect chain can be present in direct output thread and it uses 2548 // the sink buffer as input 2549 if (mType != DIRECT) { 2550 size_t numSamples = mNormalFrameCount * mChannelCount; 2551 buffer = new int16_t[numSamples]; 2552 memset(buffer, 0, numSamples * sizeof(int16_t)); 2553 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2554 ownsBuffer = true; 2555 } 2556 2557 // Attach all tracks with same session ID to this chain. 2558 for (size_t i = 0; i < mTracks.size(); ++i) { 2559 sp<Track> track = mTracks[i]; 2560 if (session == track->sessionId()) { 2561 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2562 buffer); 2563 track->setMainBuffer(buffer); 2564 chain->incTrackCnt(); 2565 } 2566 } 2567 2568 // indicate all active tracks in the chain 2569 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2570 sp<Track> track = mActiveTracks[i].promote(); 2571 if (track == 0) { 2572 continue; 2573 } 2574 if (session == track->sessionId()) { 2575 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2576 chain->incActiveTrackCnt(); 2577 } 2578 } 2579 } 2580 chain->setThread(this); 2581 chain->setInBuffer(buffer, ownsBuffer); 2582 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2583 ? mEffectBuffer : mSinkBuffer)); 2584 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2585 // chains list in order to be processed last as it contains output stage effects 2586 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2587 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2588 // after track specific effects and before output stage 2589 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2590 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2591 // Effect chain for other sessions are inserted at beginning of effect 2592 // chains list to be processed before output mix effects. Relative order between other 2593 // sessions is not important 2594 size_t size = mEffectChains.size(); 2595 size_t i = 0; 2596 for (i = 0; i < size; i++) { 2597 if (mEffectChains[i]->sessionId() < session) { 2598 break; 2599 } 2600 } 2601 mEffectChains.insertAt(chain, i); 2602 checkSuspendOnAddEffectChain_l(chain); 2603 2604 return NO_ERROR; 2605} 2606 2607size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2608{ 2609 int session = chain->sessionId(); 2610 2611 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2612 2613 for (size_t i = 0; i < mEffectChains.size(); i++) { 2614 if (chain == mEffectChains[i]) { 2615 mEffectChains.removeAt(i); 2616 // detach all active tracks from the chain 2617 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2618 sp<Track> track = mActiveTracks[i].promote(); 2619 if (track == 0) { 2620 continue; 2621 } 2622 if (session == track->sessionId()) { 2623 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2624 chain.get(), session); 2625 chain->decActiveTrackCnt(); 2626 } 2627 } 2628 2629 // detach all tracks with same session ID from this chain 2630 for (size_t i = 0; i < mTracks.size(); ++i) { 2631 sp<Track> track = mTracks[i]; 2632 if (session == track->sessionId()) { 2633 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2634 chain->decTrackCnt(); 2635 } 2636 } 2637 break; 2638 } 2639 } 2640 return mEffectChains.size(); 2641} 2642 2643status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2644 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2645{ 2646 Mutex::Autolock _l(mLock); 2647 return attachAuxEffect_l(track, EffectId); 2648} 2649 2650status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2651 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2652{ 2653 status_t status = NO_ERROR; 2654 2655 if (EffectId == 0) { 2656 track->setAuxBuffer(0, NULL); 2657 } else { 2658 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2659 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2660 if (effect != 0) { 2661 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2662 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2663 } else { 2664 status = INVALID_OPERATION; 2665 } 2666 } else { 2667 status = BAD_VALUE; 2668 } 2669 } 2670 return status; 2671} 2672 2673void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2674{ 2675 for (size_t i = 0; i < mTracks.size(); ++i) { 2676 sp<Track> track = mTracks[i]; 2677 if (track->auxEffectId() == effectId) { 2678 attachAuxEffect_l(track, 0); 2679 } 2680 } 2681} 2682 2683bool AudioFlinger::PlaybackThread::threadLoop() 2684{ 2685 Vector< sp<Track> > tracksToRemove; 2686 2687 mStandbyTimeNs = systemTime(); 2688 2689 // MIXER 2690 nsecs_t lastWarning = 0; 2691 2692 // DUPLICATING 2693 // FIXME could this be made local to while loop? 2694 writeFrames = 0; 2695 2696 int lastGeneration = 0; 2697 2698 cacheParameters_l(); 2699 mSleepTimeUs = mIdleSleepTimeUs; 2700 2701 if (mType == MIXER) { 2702 sleepTimeShift = 0; 2703 } 2704 2705 CpuStats cpuStats; 2706 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2707 2708 acquireWakeLock(); 2709 2710 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2711 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2712 // and then that string will be logged at the next convenient opportunity. 2713 const char *logString = NULL; 2714 2715 checkSilentMode_l(); 2716 2717 while (!exitPending()) 2718 { 2719 cpuStats.sample(myName); 2720 2721 Vector< sp<EffectChain> > effectChains; 2722 2723 { // scope for mLock 2724 2725 Mutex::Autolock _l(mLock); 2726 2727 processConfigEvents_l(); 2728 2729 if (logString != NULL) { 2730 mNBLogWriter->logTimestamp(); 2731 mNBLogWriter->log(logString); 2732 logString = NULL; 2733 } 2734 2735 // Gather the framesReleased counters for all active tracks, 2736 // and latch them atomically with the timestamp. 2737 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2738 mLatchD.mFramesReleased.clear(); 2739 size_t size = mActiveTracks.size(); 2740 for (size_t i = 0; i < size; i++) { 2741 sp<Track> t = mActiveTracks[i].promote(); 2742 if (t != 0) { 2743 mLatchD.mFramesReleased.add(t.get(), 2744 t->mAudioTrackServerProxy->framesReleased()); 2745 } 2746 } 2747 if (mLatchDValid) { 2748 mLatchQ = mLatchD; 2749 mLatchDValid = false; 2750 mLatchQValid = true; 2751 } 2752 2753 saveOutputTracks(); 2754 if (mSignalPending) { 2755 // A signal was raised while we were unlocked 2756 mSignalPending = false; 2757 } else if (waitingAsyncCallback_l()) { 2758 if (exitPending()) { 2759 break; 2760 } 2761 bool released = false; 2762 // The following works around a bug in the offload driver. Ideally we would release 2763 // the wake lock every time, but that causes the last offload buffer(s) to be 2764 // dropped while the device is on battery, so we need to hold a wake lock during 2765 // the drain phase. 2766 if (mBytesRemaining && !(mDrainSequence & 1)) { 2767 releaseWakeLock_l(); 2768 released = true; 2769 } 2770 mWakeLockUids.clear(); 2771 mActiveTracksGeneration++; 2772 ALOGV("wait async completion"); 2773 mWaitWorkCV.wait(mLock); 2774 ALOGV("async completion/wake"); 2775 if (released) { 2776 acquireWakeLock_l(); 2777 } 2778 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2779 mSleepTimeUs = 0; 2780 2781 continue; 2782 } 2783 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2784 isSuspended()) { 2785 // put audio hardware into standby after short delay 2786 if (shouldStandby_l()) { 2787 2788 threadLoop_standby(); 2789 2790 mStandby = true; 2791 } 2792 2793 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2794 // we're about to wait, flush the binder command buffer 2795 IPCThreadState::self()->flushCommands(); 2796 2797 clearOutputTracks(); 2798 2799 if (exitPending()) { 2800 break; 2801 } 2802 2803 releaseWakeLock_l(); 2804 mWakeLockUids.clear(); 2805 mActiveTracksGeneration++; 2806 // wait until we have something to do... 2807 ALOGV("%s going to sleep", myName.string()); 2808 mWaitWorkCV.wait(mLock); 2809 ALOGV("%s waking up", myName.string()); 2810 acquireWakeLock_l(); 2811 2812 mMixerStatus = MIXER_IDLE; 2813 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2814 mBytesWritten = 0; 2815 mBytesRemaining = 0; 2816 checkSilentMode_l(); 2817 2818 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2819 mSleepTimeUs = mIdleSleepTimeUs; 2820 if (mType == MIXER) { 2821 sleepTimeShift = 0; 2822 } 2823 2824 continue; 2825 } 2826 } 2827 // mMixerStatusIgnoringFastTracks is also updated internally 2828 mMixerStatus = prepareTracks_l(&tracksToRemove); 2829 2830 // compare with previously applied list 2831 if (lastGeneration != mActiveTracksGeneration) { 2832 // update wakelock 2833 updateWakeLockUids_l(mWakeLockUids); 2834 lastGeneration = mActiveTracksGeneration; 2835 } 2836 2837 // prevent any changes in effect chain list and in each effect chain 2838 // during mixing and effect process as the audio buffers could be deleted 2839 // or modified if an effect is created or deleted 2840 lockEffectChains_l(effectChains); 2841 } // mLock scope ends 2842 2843 if (mBytesRemaining == 0) { 2844 mCurrentWriteLength = 0; 2845 if (mMixerStatus == MIXER_TRACKS_READY) { 2846 // threadLoop_mix() sets mCurrentWriteLength 2847 threadLoop_mix(); 2848 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2849 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2850 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2851 // must be written to HAL 2852 threadLoop_sleepTime(); 2853 if (mSleepTimeUs == 0) { 2854 mCurrentWriteLength = mSinkBufferSize; 2855 } 2856 } 2857 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2858 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2859 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2860 // or mSinkBuffer (if there are no effects). 2861 // 2862 // This is done pre-effects computation; if effects change to 2863 // support higher precision, this needs to move. 2864 // 2865 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2866 // TODO use mSleepTimeUs == 0 as an additional condition. 2867 if (mMixerBufferValid) { 2868 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2869 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2870 2871 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2872 mNormalFrameCount * mChannelCount); 2873 } 2874 2875 mBytesRemaining = mCurrentWriteLength; 2876 if (isSuspended()) { 2877 mSleepTimeUs = suspendSleepTimeUs(); 2878 // simulate write to HAL when suspended 2879 mBytesWritten += mSinkBufferSize; 2880 mBytesRemaining = 0; 2881 } 2882 2883 // only process effects if we're going to write 2884 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 2885 for (size_t i = 0; i < effectChains.size(); i ++) { 2886 effectChains[i]->process_l(); 2887 } 2888 } 2889 } 2890 // Process effect chains for offloaded thread even if no audio 2891 // was read from audio track: process only updates effect state 2892 // and thus does have to be synchronized with audio writes but may have 2893 // to be called while waiting for async write callback 2894 if (mType == OFFLOAD) { 2895 for (size_t i = 0; i < effectChains.size(); i ++) { 2896 effectChains[i]->process_l(); 2897 } 2898 } 2899 2900 // Only if the Effects buffer is enabled and there is data in the 2901 // Effects buffer (buffer valid), we need to 2902 // copy into the sink buffer. 2903 // TODO use mSleepTimeUs == 0 as an additional condition. 2904 if (mEffectBufferValid) { 2905 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2906 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2907 mNormalFrameCount * mChannelCount); 2908 } 2909 2910 // enable changes in effect chain 2911 unlockEffectChains(effectChains); 2912 2913 if (!waitingAsyncCallback()) { 2914 // mSleepTimeUs == 0 means we must write to audio hardware 2915 if (mSleepTimeUs == 0) { 2916 ssize_t ret = 0; 2917 if (mBytesRemaining) { 2918 ret = threadLoop_write(); 2919 if (ret < 0) { 2920 mBytesRemaining = 0; 2921 } else { 2922 mBytesWritten += ret; 2923 mBytesRemaining -= ret; 2924 } 2925 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2926 (mMixerStatus == MIXER_DRAIN_ALL)) { 2927 threadLoop_drain(); 2928 } 2929 if (mType == MIXER && !mStandby) { 2930 // write blocked detection 2931 nsecs_t now = systemTime(); 2932 nsecs_t delta = now - mLastWriteTime; 2933 if (delta > maxPeriod) { 2934 mNumDelayedWrites++; 2935 if ((now - lastWarning) > kWarningThrottleNs) { 2936 ATRACE_NAME("underrun"); 2937 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2938 ns2ms(delta), mNumDelayedWrites, this); 2939 lastWarning = now; 2940 } 2941 } 2942 2943 if (mThreadThrottle 2944 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 2945 && ret > 0) { // we wrote something 2946 // Limit MixerThread data processing to no more than twice the 2947 // expected processing rate. 2948 // 2949 // This helps prevent underruns with NuPlayer and other applications 2950 // which may set up buffers that are close to the minimum size, or use 2951 // deep buffers, and rely on a double-buffering sleep strategy to fill. 2952 // 2953 // The throttle smooths out sudden large data drains from the device, 2954 // e.g. when it comes out of standby, which often causes problems with 2955 // (1) mixer threads without a fast mixer (which has its own warm-up) 2956 // (2) minimum buffer sized tracks (even if the track is full, 2957 // the app won't fill fast enough to handle the sudden draw). 2958 2959 const int32_t deltaMs = delta / 1000000; 2960 const int32_t throttleMs = mHalfBufferMs - deltaMs; 2961 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 2962 usleep(throttleMs * 1000); 2963 ALOGD("mixer(%p) throttle: ret(%zd) deltaMs(%d) requires sleep %d ms", 2964 this, ret, deltaMs, throttleMs); 2965 } 2966 } 2967 } 2968 2969 } else { 2970 ATRACE_BEGIN("sleep"); 2971 usleep(mSleepTimeUs); 2972 ATRACE_END(); 2973 } 2974 } 2975 2976 // Finally let go of removed track(s), without the lock held 2977 // since we can't guarantee the destructors won't acquire that 2978 // same lock. This will also mutate and push a new fast mixer state. 2979 threadLoop_removeTracks(tracksToRemove); 2980 tracksToRemove.clear(); 2981 2982 // FIXME I don't understand the need for this here; 2983 // it was in the original code but maybe the 2984 // assignment in saveOutputTracks() makes this unnecessary? 2985 clearOutputTracks(); 2986 2987 // Effect chains will be actually deleted here if they were removed from 2988 // mEffectChains list during mixing or effects processing 2989 effectChains.clear(); 2990 2991 // FIXME Note that the above .clear() is no longer necessary since effectChains 2992 // is now local to this block, but will keep it for now (at least until merge done). 2993 } 2994 2995 threadLoop_exit(); 2996 2997 if (!mStandby) { 2998 threadLoop_standby(); 2999 mStandby = true; 3000 } 3001 3002 releaseWakeLock(); 3003 mWakeLockUids.clear(); 3004 mActiveTracksGeneration++; 3005 3006 ALOGV("Thread %p type %d exiting", this, mType); 3007 return false; 3008} 3009 3010// removeTracks_l() must be called with ThreadBase::mLock held 3011void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3012{ 3013 size_t count = tracksToRemove.size(); 3014 if (count > 0) { 3015 for (size_t i=0 ; i<count ; i++) { 3016 const sp<Track>& track = tracksToRemove.itemAt(i); 3017 mActiveTracks.remove(track); 3018 mWakeLockUids.remove(track->uid()); 3019 mActiveTracksGeneration++; 3020 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3021 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3022 if (chain != 0) { 3023 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3024 track->sessionId()); 3025 chain->decActiveTrackCnt(); 3026 } 3027 if (track->isTerminated()) { 3028 removeTrack_l(track); 3029 } 3030 } 3031 } 3032 3033} 3034 3035status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3036{ 3037 if (mNormalSink != 0) { 3038 return mNormalSink->getTimestamp(timestamp); 3039 } 3040 if ((mType == OFFLOAD || mType == DIRECT) 3041 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3042 uint64_t position64; 3043 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3044 if (ret == 0) { 3045 timestamp.mPosition = (uint32_t)position64; 3046 return NO_ERROR; 3047 } 3048 } 3049 return INVALID_OPERATION; 3050} 3051 3052status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3053 audio_patch_handle_t *handle) 3054{ 3055 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3056 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3057 if (mFastMixer != 0) { 3058 FastMixerStateQueue *sq = mFastMixer->sq(); 3059 FastMixerState *state = sq->begin(); 3060 if (!(state->mCommand & FastMixerState::IDLE)) { 3061 previousCommand = state->mCommand; 3062 state->mCommand = FastMixerState::HOT_IDLE; 3063 sq->end(); 3064 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3065 } else { 3066 sq->end(false /*didModify*/); 3067 } 3068 } 3069 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3070 3071 if (!(previousCommand & FastMixerState::IDLE)) { 3072 ALOG_ASSERT(mFastMixer != 0); 3073 FastMixerStateQueue *sq = mFastMixer->sq(); 3074 FastMixerState *state = sq->begin(); 3075 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3076 state->mCommand = previousCommand; 3077 sq->end(); 3078 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3079 } 3080 3081 return status; 3082} 3083 3084status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3085 audio_patch_handle_t *handle) 3086{ 3087 status_t status = NO_ERROR; 3088 3089 // store new device and send to effects 3090 audio_devices_t type = AUDIO_DEVICE_NONE; 3091 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3092 type |= patch->sinks[i].ext.device.type; 3093 } 3094 3095#ifdef ADD_BATTERY_DATA 3096 // when changing the audio output device, call addBatteryData to notify 3097 // the change 3098 if (mOutDevice != type) { 3099 uint32_t params = 0; 3100 // check whether speaker is on 3101 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3102 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3103 } 3104 3105 audio_devices_t deviceWithoutSpeaker 3106 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3107 // check if any other device (except speaker) is on 3108 if (type & deviceWithoutSpeaker) { 3109 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3110 } 3111 3112 if (params != 0) { 3113 addBatteryData(params); 3114 } 3115 } 3116#endif 3117 3118 for (size_t i = 0; i < mEffectChains.size(); i++) { 3119 mEffectChains[i]->setDevice_l(type); 3120 } 3121 mOutDevice = type; 3122 mPatch = *patch; 3123 3124 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3125 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3126 status = hwDevice->create_audio_patch(hwDevice, 3127 patch->num_sources, 3128 patch->sources, 3129 patch->num_sinks, 3130 patch->sinks, 3131 handle); 3132 } else { 3133 char *address; 3134 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3135 //FIXME: we only support address on first sink with HAL version < 3.0 3136 address = audio_device_address_to_parameter( 3137 patch->sinks[0].ext.device.type, 3138 patch->sinks[0].ext.device.address); 3139 } else { 3140 address = (char *)calloc(1, 1); 3141 } 3142 AudioParameter param = AudioParameter(String8(address)); 3143 free(address); 3144 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3145 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3146 param.toString().string()); 3147 *handle = AUDIO_PATCH_HANDLE_NONE; 3148 } 3149 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3150 return status; 3151} 3152 3153status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3154{ 3155 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3156 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3157 if (mFastMixer != 0) { 3158 FastMixerStateQueue *sq = mFastMixer->sq(); 3159 FastMixerState *state = sq->begin(); 3160 if (!(state->mCommand & FastMixerState::IDLE)) { 3161 previousCommand = state->mCommand; 3162 state->mCommand = FastMixerState::HOT_IDLE; 3163 sq->end(); 3164 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3165 } else { 3166 sq->end(false /*didModify*/); 3167 } 3168 } 3169 3170 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3171 3172 if (!(previousCommand & FastMixerState::IDLE)) { 3173 ALOG_ASSERT(mFastMixer != 0); 3174 FastMixerStateQueue *sq = mFastMixer->sq(); 3175 FastMixerState *state = sq->begin(); 3176 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3177 state->mCommand = previousCommand; 3178 sq->end(); 3179 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3180 } 3181 3182 return status; 3183} 3184 3185status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3186{ 3187 status_t status = NO_ERROR; 3188 3189 mOutDevice = AUDIO_DEVICE_NONE; 3190 3191 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3192 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3193 status = hwDevice->release_audio_patch(hwDevice, handle); 3194 } else { 3195 AudioParameter param; 3196 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3197 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3198 param.toString().string()); 3199 } 3200 return status; 3201} 3202 3203void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3204{ 3205 Mutex::Autolock _l(mLock); 3206 mTracks.add(track); 3207} 3208 3209void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3210{ 3211 Mutex::Autolock _l(mLock); 3212 destroyTrack_l(track); 3213} 3214 3215void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3216{ 3217 ThreadBase::getAudioPortConfig(config); 3218 config->role = AUDIO_PORT_ROLE_SOURCE; 3219 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3220 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3221} 3222 3223// ---------------------------------------------------------------------------- 3224 3225AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3226 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3227 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3228 // mAudioMixer below 3229 // mFastMixer below 3230 mFastMixerFutex(0) 3231 // mOutputSink below 3232 // mPipeSink below 3233 // mNormalSink below 3234{ 3235 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3236 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3237 "mFrameCount=%d, mNormalFrameCount=%d", 3238 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3239 mNormalFrameCount); 3240 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3241 3242 if (type == DUPLICATING) { 3243 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3244 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3245 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3246 return; 3247 } 3248 // create an NBAIO sink for the HAL output stream, and negotiate 3249 mOutputSink = new AudioStreamOutSink(output->stream); 3250 size_t numCounterOffers = 0; 3251 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3252 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3253 ALOG_ASSERT(index == 0); 3254 3255 // initialize fast mixer depending on configuration 3256 bool initFastMixer; 3257 switch (kUseFastMixer) { 3258 case FastMixer_Never: 3259 initFastMixer = false; 3260 break; 3261 case FastMixer_Always: 3262 initFastMixer = true; 3263 break; 3264 case FastMixer_Static: 3265 case FastMixer_Dynamic: 3266 initFastMixer = mFrameCount < mNormalFrameCount; 3267 break; 3268 } 3269 if (initFastMixer) { 3270 audio_format_t fastMixerFormat; 3271 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3272 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3273 } else { 3274 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3275 } 3276 if (mFormat != fastMixerFormat) { 3277 // change our Sink format to accept our intermediate precision 3278 mFormat = fastMixerFormat; 3279 free(mSinkBuffer); 3280 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3281 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3282 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3283 } 3284 3285 // create a MonoPipe to connect our submix to FastMixer 3286 NBAIO_Format format = mOutputSink->format(); 3287 NBAIO_Format origformat = format; 3288 // adjust format to match that of the Fast Mixer 3289 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3290 format.mFormat = fastMixerFormat; 3291 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3292 3293 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3294 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3295 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3296 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3297 const NBAIO_Format offers[1] = {format}; 3298 size_t numCounterOffers = 0; 3299 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3300 ALOG_ASSERT(index == 0); 3301 monoPipe->setAvgFrames((mScreenState & 1) ? 3302 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3303 mPipeSink = monoPipe; 3304 3305#ifdef TEE_SINK 3306 if (mTeeSinkOutputEnabled) { 3307 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3308 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3309 const NBAIO_Format offers2[1] = {origformat}; 3310 numCounterOffers = 0; 3311 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3312 ALOG_ASSERT(index == 0); 3313 mTeeSink = teeSink; 3314 PipeReader *teeSource = new PipeReader(*teeSink); 3315 numCounterOffers = 0; 3316 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3317 ALOG_ASSERT(index == 0); 3318 mTeeSource = teeSource; 3319 } 3320#endif 3321 3322 // create fast mixer and configure it initially with just one fast track for our submix 3323 mFastMixer = new FastMixer(); 3324 FastMixerStateQueue *sq = mFastMixer->sq(); 3325#ifdef STATE_QUEUE_DUMP 3326 sq->setObserverDump(&mStateQueueObserverDump); 3327 sq->setMutatorDump(&mStateQueueMutatorDump); 3328#endif 3329 FastMixerState *state = sq->begin(); 3330 FastTrack *fastTrack = &state->mFastTracks[0]; 3331 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3332 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3333 fastTrack->mVolumeProvider = NULL; 3334 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3335 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3336 fastTrack->mGeneration++; 3337 state->mFastTracksGen++; 3338 state->mTrackMask = 1; 3339 // fast mixer will use the HAL output sink 3340 state->mOutputSink = mOutputSink.get(); 3341 state->mOutputSinkGen++; 3342 state->mFrameCount = mFrameCount; 3343 state->mCommand = FastMixerState::COLD_IDLE; 3344 // already done in constructor initialization list 3345 //mFastMixerFutex = 0; 3346 state->mColdFutexAddr = &mFastMixerFutex; 3347 state->mColdGen++; 3348 state->mDumpState = &mFastMixerDumpState; 3349#ifdef TEE_SINK 3350 state->mTeeSink = mTeeSink.get(); 3351#endif 3352 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3353 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3354 sq->end(); 3355 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3356 3357 // start the fast mixer 3358 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3359 pid_t tid = mFastMixer->getTid(); 3360 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3361 3362#ifdef AUDIO_WATCHDOG 3363 // create and start the watchdog 3364 mAudioWatchdog = new AudioWatchdog(); 3365 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3366 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3367 tid = mAudioWatchdog->getTid(); 3368 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3369#endif 3370 3371 } 3372 3373 switch (kUseFastMixer) { 3374 case FastMixer_Never: 3375 case FastMixer_Dynamic: 3376 mNormalSink = mOutputSink; 3377 break; 3378 case FastMixer_Always: 3379 mNormalSink = mPipeSink; 3380 break; 3381 case FastMixer_Static: 3382 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3383 break; 3384 } 3385} 3386 3387AudioFlinger::MixerThread::~MixerThread() 3388{ 3389 if (mFastMixer != 0) { 3390 FastMixerStateQueue *sq = mFastMixer->sq(); 3391 FastMixerState *state = sq->begin(); 3392 if (state->mCommand == FastMixerState::COLD_IDLE) { 3393 int32_t old = android_atomic_inc(&mFastMixerFutex); 3394 if (old == -1) { 3395 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3396 } 3397 } 3398 state->mCommand = FastMixerState::EXIT; 3399 sq->end(); 3400 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3401 mFastMixer->join(); 3402 // Though the fast mixer thread has exited, it's state queue is still valid. 3403 // We'll use that extract the final state which contains one remaining fast track 3404 // corresponding to our sub-mix. 3405 state = sq->begin(); 3406 ALOG_ASSERT(state->mTrackMask == 1); 3407 FastTrack *fastTrack = &state->mFastTracks[0]; 3408 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3409 delete fastTrack->mBufferProvider; 3410 sq->end(false /*didModify*/); 3411 mFastMixer.clear(); 3412#ifdef AUDIO_WATCHDOG 3413 if (mAudioWatchdog != 0) { 3414 mAudioWatchdog->requestExit(); 3415 mAudioWatchdog->requestExitAndWait(); 3416 mAudioWatchdog.clear(); 3417 } 3418#endif 3419 } 3420 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3421 delete mAudioMixer; 3422} 3423 3424 3425uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3426{ 3427 if (mFastMixer != 0) { 3428 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3429 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3430 } 3431 return latency; 3432} 3433 3434 3435void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3436{ 3437 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3438} 3439 3440ssize_t AudioFlinger::MixerThread::threadLoop_write() 3441{ 3442 // FIXME we should only do one push per cycle; confirm this is true 3443 // Start the fast mixer if it's not already running 3444 if (mFastMixer != 0) { 3445 FastMixerStateQueue *sq = mFastMixer->sq(); 3446 FastMixerState *state = sq->begin(); 3447 if (state->mCommand != FastMixerState::MIX_WRITE && 3448 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3449 if (state->mCommand == FastMixerState::COLD_IDLE) { 3450 int32_t old = android_atomic_inc(&mFastMixerFutex); 3451 if (old == -1) { 3452 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3453 } 3454#ifdef AUDIO_WATCHDOG 3455 if (mAudioWatchdog != 0) { 3456 mAudioWatchdog->resume(); 3457 } 3458#endif 3459 } 3460 state->mCommand = FastMixerState::MIX_WRITE; 3461#ifdef FAST_THREAD_STATISTICS 3462 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3463 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3464#endif 3465 sq->end(); 3466 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3467 if (kUseFastMixer == FastMixer_Dynamic) { 3468 mNormalSink = mPipeSink; 3469 } 3470 } else { 3471 sq->end(false /*didModify*/); 3472 } 3473 } 3474 return PlaybackThread::threadLoop_write(); 3475} 3476 3477void AudioFlinger::MixerThread::threadLoop_standby() 3478{ 3479 // Idle the fast mixer if it's currently running 3480 if (mFastMixer != 0) { 3481 FastMixerStateQueue *sq = mFastMixer->sq(); 3482 FastMixerState *state = sq->begin(); 3483 if (!(state->mCommand & FastMixerState::IDLE)) { 3484 state->mCommand = FastMixerState::COLD_IDLE; 3485 state->mColdFutexAddr = &mFastMixerFutex; 3486 state->mColdGen++; 3487 mFastMixerFutex = 0; 3488 sq->end(); 3489 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3490 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3491 if (kUseFastMixer == FastMixer_Dynamic) { 3492 mNormalSink = mOutputSink; 3493 } 3494#ifdef AUDIO_WATCHDOG 3495 if (mAudioWatchdog != 0) { 3496 mAudioWatchdog->pause(); 3497 } 3498#endif 3499 } else { 3500 sq->end(false /*didModify*/); 3501 } 3502 } 3503 PlaybackThread::threadLoop_standby(); 3504} 3505 3506bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3507{ 3508 return false; 3509} 3510 3511bool AudioFlinger::PlaybackThread::shouldStandby_l() 3512{ 3513 return !mStandby; 3514} 3515 3516bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3517{ 3518 Mutex::Autolock _l(mLock); 3519 return waitingAsyncCallback_l(); 3520} 3521 3522// shared by MIXER and DIRECT, overridden by DUPLICATING 3523void AudioFlinger::PlaybackThread::threadLoop_standby() 3524{ 3525 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3526 mOutput->standby(); 3527 if (mUseAsyncWrite != 0) { 3528 // discard any pending drain or write ack by incrementing sequence 3529 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3530 mDrainSequence = (mDrainSequence + 2) & ~1; 3531 ALOG_ASSERT(mCallbackThread != 0); 3532 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3533 mCallbackThread->setDraining(mDrainSequence); 3534 } 3535 mHwPaused = false; 3536} 3537 3538void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3539{ 3540 ALOGV("signal playback thread"); 3541 broadcast_l(); 3542} 3543 3544void AudioFlinger::MixerThread::threadLoop_mix() 3545{ 3546 // obtain the presentation timestamp of the next output buffer 3547 int64_t pts; 3548 status_t status = INVALID_OPERATION; 3549 3550 if (mNormalSink != 0) { 3551 status = mNormalSink->getNextWriteTimestamp(&pts); 3552 } else { 3553 status = mOutputSink->getNextWriteTimestamp(&pts); 3554 } 3555 3556 if (status != NO_ERROR) { 3557 pts = AudioBufferProvider::kInvalidPTS; 3558 } 3559 3560 // mix buffers... 3561 mAudioMixer->process(pts); 3562 mCurrentWriteLength = mSinkBufferSize; 3563 // increase sleep time progressively when application underrun condition clears. 3564 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3565 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3566 // such that we would underrun the audio HAL. 3567 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3568 sleepTimeShift--; 3569 } 3570 mSleepTimeUs = 0; 3571 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3572 //TODO: delay standby when effects have a tail 3573 3574} 3575 3576void AudioFlinger::MixerThread::threadLoop_sleepTime() 3577{ 3578 // If no tracks are ready, sleep once for the duration of an output 3579 // buffer size, then write 0s to the output 3580 if (mSleepTimeUs == 0) { 3581 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3582 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3583 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3584 mSleepTimeUs = kMinThreadSleepTimeUs; 3585 } 3586 // reduce sleep time in case of consecutive application underruns to avoid 3587 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3588 // duration we would end up writing less data than needed by the audio HAL if 3589 // the condition persists. 3590 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3591 sleepTimeShift++; 3592 } 3593 } else { 3594 mSleepTimeUs = mIdleSleepTimeUs; 3595 } 3596 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3597 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3598 // before effects processing or output. 3599 if (mMixerBufferValid) { 3600 memset(mMixerBuffer, 0, mMixerBufferSize); 3601 } else { 3602 memset(mSinkBuffer, 0, mSinkBufferSize); 3603 } 3604 mSleepTimeUs = 0; 3605 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3606 "anticipated start"); 3607 } 3608 // TODO add standby time extension fct of effect tail 3609} 3610 3611// prepareTracks_l() must be called with ThreadBase::mLock held 3612AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3613 Vector< sp<Track> > *tracksToRemove) 3614{ 3615 3616 mixer_state mixerStatus = MIXER_IDLE; 3617 // find out which tracks need to be processed 3618 size_t count = mActiveTracks.size(); 3619 size_t mixedTracks = 0; 3620 size_t tracksWithEffect = 0; 3621 // counts only _active_ fast tracks 3622 size_t fastTracks = 0; 3623 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3624 3625 float masterVolume = mMasterVolume; 3626 bool masterMute = mMasterMute; 3627 3628 if (masterMute) { 3629 masterVolume = 0; 3630 } 3631 // Delegate master volume control to effect in output mix effect chain if needed 3632 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3633 if (chain != 0) { 3634 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3635 chain->setVolume_l(&v, &v); 3636 masterVolume = (float)((v + (1 << 23)) >> 24); 3637 chain.clear(); 3638 } 3639 3640 // prepare a new state to push 3641 FastMixerStateQueue *sq = NULL; 3642 FastMixerState *state = NULL; 3643 bool didModify = false; 3644 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3645 if (mFastMixer != 0) { 3646 sq = mFastMixer->sq(); 3647 state = sq->begin(); 3648 } 3649 3650 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3651 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3652 3653 for (size_t i=0 ; i<count ; i++) { 3654 const sp<Track> t = mActiveTracks[i].promote(); 3655 if (t == 0) { 3656 continue; 3657 } 3658 3659 // this const just means the local variable doesn't change 3660 Track* const track = t.get(); 3661 3662 // process fast tracks 3663 if (track->isFastTrack()) { 3664 3665 // It's theoretically possible (though unlikely) for a fast track to be created 3666 // and then removed within the same normal mix cycle. This is not a problem, as 3667 // the track never becomes active so it's fast mixer slot is never touched. 3668 // The converse, of removing an (active) track and then creating a new track 3669 // at the identical fast mixer slot within the same normal mix cycle, 3670 // is impossible because the slot isn't marked available until the end of each cycle. 3671 int j = track->mFastIndex; 3672 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3673 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3674 FastTrack *fastTrack = &state->mFastTracks[j]; 3675 3676 // Determine whether the track is currently in underrun condition, 3677 // and whether it had a recent underrun. 3678 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3679 FastTrackUnderruns underruns = ftDump->mUnderruns; 3680 uint32_t recentFull = (underruns.mBitFields.mFull - 3681 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3682 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3683 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3684 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3685 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3686 uint32_t recentUnderruns = recentPartial + recentEmpty; 3687 track->mObservedUnderruns = underruns; 3688 // don't count underruns that occur while stopping or pausing 3689 // or stopped which can occur when flush() is called while active 3690 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3691 recentUnderruns > 0) { 3692 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3693 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3694 } 3695 3696 // This is similar to the state machine for normal tracks, 3697 // with a few modifications for fast tracks. 3698 bool isActive = true; 3699 switch (track->mState) { 3700 case TrackBase::STOPPING_1: 3701 // track stays active in STOPPING_1 state until first underrun 3702 if (recentUnderruns > 0 || track->isTerminated()) { 3703 track->mState = TrackBase::STOPPING_2; 3704 } 3705 break; 3706 case TrackBase::PAUSING: 3707 // ramp down is not yet implemented 3708 track->setPaused(); 3709 break; 3710 case TrackBase::RESUMING: 3711 // ramp up is not yet implemented 3712 track->mState = TrackBase::ACTIVE; 3713 break; 3714 case TrackBase::ACTIVE: 3715 if (recentFull > 0 || recentPartial > 0) { 3716 // track has provided at least some frames recently: reset retry count 3717 track->mRetryCount = kMaxTrackRetries; 3718 } 3719 if (recentUnderruns == 0) { 3720 // no recent underruns: stay active 3721 break; 3722 } 3723 // there has recently been an underrun of some kind 3724 if (track->sharedBuffer() == 0) { 3725 // were any of the recent underruns "empty" (no frames available)? 3726 if (recentEmpty == 0) { 3727 // no, then ignore the partial underruns as they are allowed indefinitely 3728 break; 3729 } 3730 // there has recently been an "empty" underrun: decrement the retry counter 3731 if (--(track->mRetryCount) > 0) { 3732 break; 3733 } 3734 // indicate to client process that the track was disabled because of underrun; 3735 // it will then automatically call start() when data is available 3736 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3737 // remove from active list, but state remains ACTIVE [confusing but true] 3738 isActive = false; 3739 break; 3740 } 3741 // fall through 3742 case TrackBase::STOPPING_2: 3743 case TrackBase::PAUSED: 3744 case TrackBase::STOPPED: 3745 case TrackBase::FLUSHED: // flush() while active 3746 // Check for presentation complete if track is inactive 3747 // We have consumed all the buffers of this track. 3748 // This would be incomplete if we auto-paused on underrun 3749 { 3750 size_t audioHALFrames = 3751 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3752 size_t framesWritten = mBytesWritten / mFrameSize; 3753 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3754 // track stays in active list until presentation is complete 3755 break; 3756 } 3757 } 3758 if (track->isStopping_2()) { 3759 track->mState = TrackBase::STOPPED; 3760 } 3761 if (track->isStopped()) { 3762 // Can't reset directly, as fast mixer is still polling this track 3763 // track->reset(); 3764 // So instead mark this track as needing to be reset after push with ack 3765 resetMask |= 1 << i; 3766 } 3767 isActive = false; 3768 break; 3769 case TrackBase::IDLE: 3770 default: 3771 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3772 } 3773 3774 if (isActive) { 3775 // was it previously inactive? 3776 if (!(state->mTrackMask & (1 << j))) { 3777 ExtendedAudioBufferProvider *eabp = track; 3778 VolumeProvider *vp = track; 3779 fastTrack->mBufferProvider = eabp; 3780 fastTrack->mVolumeProvider = vp; 3781 fastTrack->mChannelMask = track->mChannelMask; 3782 fastTrack->mFormat = track->mFormat; 3783 fastTrack->mGeneration++; 3784 state->mTrackMask |= 1 << j; 3785 didModify = true; 3786 // no acknowledgement required for newly active tracks 3787 } 3788 // cache the combined master volume and stream type volume for fast mixer; this 3789 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3790 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3791 ++fastTracks; 3792 } else { 3793 // was it previously active? 3794 if (state->mTrackMask & (1 << j)) { 3795 fastTrack->mBufferProvider = NULL; 3796 fastTrack->mGeneration++; 3797 state->mTrackMask &= ~(1 << j); 3798 didModify = true; 3799 // If any fast tracks were removed, we must wait for acknowledgement 3800 // because we're about to decrement the last sp<> on those tracks. 3801 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3802 } else { 3803 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3804 } 3805 tracksToRemove->add(track); 3806 // Avoids a misleading display in dumpsys 3807 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3808 } 3809 continue; 3810 } 3811 3812 { // local variable scope to avoid goto warning 3813 3814 audio_track_cblk_t* cblk = track->cblk(); 3815 3816 // The first time a track is added we wait 3817 // for all its buffers to be filled before processing it 3818 int name = track->name(); 3819 // make sure that we have enough frames to mix one full buffer. 3820 // enforce this condition only once to enable draining the buffer in case the client 3821 // app does not call stop() and relies on underrun to stop: 3822 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3823 // during last round 3824 size_t desiredFrames; 3825 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3826 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3827 3828 desiredFrames = sourceFramesNeededWithTimestretch( 3829 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3830 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3831 // add frames already consumed but not yet released by the resampler 3832 // because mAudioTrackServerProxy->framesReady() will include these frames 3833 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3834 3835 uint32_t minFrames = 1; 3836 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3837 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3838 minFrames = desiredFrames; 3839 } 3840 3841 size_t framesReady = track->framesReady(); 3842 if (ATRACE_ENABLED()) { 3843 // I wish we had formatted trace names 3844 char traceName[16]; 3845 strcpy(traceName, "nRdy"); 3846 int name = track->name(); 3847 if (AudioMixer::TRACK0 <= name && 3848 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3849 name -= AudioMixer::TRACK0; 3850 traceName[4] = (name / 10) + '0'; 3851 traceName[5] = (name % 10) + '0'; 3852 } else { 3853 traceName[4] = '?'; 3854 traceName[5] = '?'; 3855 } 3856 traceName[6] = '\0'; 3857 ATRACE_INT(traceName, framesReady); 3858 } 3859 if ((framesReady >= minFrames) && track->isReady() && 3860 !track->isPaused() && !track->isTerminated()) 3861 { 3862 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3863 3864 mixedTracks++; 3865 3866 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3867 // there is an effect chain connected to the track 3868 chain.clear(); 3869 if (track->mainBuffer() != mSinkBuffer && 3870 track->mainBuffer() != mMixerBuffer) { 3871 if (mEffectBufferEnabled) { 3872 mEffectBufferValid = true; // Later can set directly. 3873 } 3874 chain = getEffectChain_l(track->sessionId()); 3875 // Delegate volume control to effect in track effect chain if needed 3876 if (chain != 0) { 3877 tracksWithEffect++; 3878 } else { 3879 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3880 "session %d", 3881 name, track->sessionId()); 3882 } 3883 } 3884 3885 3886 int param = AudioMixer::VOLUME; 3887 if (track->mFillingUpStatus == Track::FS_FILLED) { 3888 // no ramp for the first volume setting 3889 track->mFillingUpStatus = Track::FS_ACTIVE; 3890 if (track->mState == TrackBase::RESUMING) { 3891 track->mState = TrackBase::ACTIVE; 3892 param = AudioMixer::RAMP_VOLUME; 3893 } 3894 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3895 // FIXME should not make a decision based on mServer 3896 } else if (cblk->mServer != 0) { 3897 // If the track is stopped before the first frame was mixed, 3898 // do not apply ramp 3899 param = AudioMixer::RAMP_VOLUME; 3900 } 3901 3902 // compute volume for this track 3903 uint32_t vl, vr; // in U8.24 integer format 3904 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3905 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3906 vl = vr = 0; 3907 vlf = vrf = vaf = 0.; 3908 if (track->isPausing()) { 3909 track->setPaused(); 3910 } 3911 } else { 3912 3913 // read original volumes with volume control 3914 float typeVolume = mStreamTypes[track->streamType()].volume; 3915 float v = masterVolume * typeVolume; 3916 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3917 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3918 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3919 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3920 // track volumes come from shared memory, so can't be trusted and must be clamped 3921 if (vlf > GAIN_FLOAT_UNITY) { 3922 ALOGV("Track left volume out of range: %.3g", vlf); 3923 vlf = GAIN_FLOAT_UNITY; 3924 } 3925 if (vrf > GAIN_FLOAT_UNITY) { 3926 ALOGV("Track right volume out of range: %.3g", vrf); 3927 vrf = GAIN_FLOAT_UNITY; 3928 } 3929 // now apply the master volume and stream type volume 3930 vlf *= v; 3931 vrf *= v; 3932 // assuming master volume and stream type volume each go up to 1.0, 3933 // then derive vl and vr as U8.24 versions for the effect chain 3934 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3935 vl = (uint32_t) (scaleto8_24 * vlf); 3936 vr = (uint32_t) (scaleto8_24 * vrf); 3937 // vl and vr are now in U8.24 format 3938 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3939 // send level comes from shared memory and so may be corrupt 3940 if (sendLevel > MAX_GAIN_INT) { 3941 ALOGV("Track send level out of range: %04X", sendLevel); 3942 sendLevel = MAX_GAIN_INT; 3943 } 3944 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3945 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3946 } 3947 3948 // Delegate volume control to effect in track effect chain if needed 3949 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3950 // Do not ramp volume if volume is controlled by effect 3951 param = AudioMixer::VOLUME; 3952 // Update remaining floating point volume levels 3953 vlf = (float)vl / (1 << 24); 3954 vrf = (float)vr / (1 << 24); 3955 track->mHasVolumeController = true; 3956 } else { 3957 // force no volume ramp when volume controller was just disabled or removed 3958 // from effect chain to avoid volume spike 3959 if (track->mHasVolumeController) { 3960 param = AudioMixer::VOLUME; 3961 } 3962 track->mHasVolumeController = false; 3963 } 3964 3965 // XXX: these things DON'T need to be done each time 3966 mAudioMixer->setBufferProvider(name, track); 3967 mAudioMixer->enable(name); 3968 3969 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3970 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3971 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3972 mAudioMixer->setParameter( 3973 name, 3974 AudioMixer::TRACK, 3975 AudioMixer::FORMAT, (void *)track->format()); 3976 mAudioMixer->setParameter( 3977 name, 3978 AudioMixer::TRACK, 3979 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3980 mAudioMixer->setParameter( 3981 name, 3982 AudioMixer::TRACK, 3983 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3984 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3985 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3986 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3987 if (reqSampleRate == 0) { 3988 reqSampleRate = mSampleRate; 3989 } else if (reqSampleRate > maxSampleRate) { 3990 reqSampleRate = maxSampleRate; 3991 } 3992 mAudioMixer->setParameter( 3993 name, 3994 AudioMixer::RESAMPLE, 3995 AudioMixer::SAMPLE_RATE, 3996 (void *)(uintptr_t)reqSampleRate); 3997 3998 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3999 mAudioMixer->setParameter( 4000 name, 4001 AudioMixer::TIMESTRETCH, 4002 AudioMixer::PLAYBACK_RATE, 4003 &playbackRate); 4004 4005 /* 4006 * Select the appropriate output buffer for the track. 4007 * 4008 * Tracks with effects go into their own effects chain buffer 4009 * and from there into either mEffectBuffer or mSinkBuffer. 4010 * 4011 * Other tracks can use mMixerBuffer for higher precision 4012 * channel accumulation. If this buffer is enabled 4013 * (mMixerBufferEnabled true), then selected tracks will accumulate 4014 * into it. 4015 * 4016 */ 4017 if (mMixerBufferEnabled 4018 && (track->mainBuffer() == mSinkBuffer 4019 || track->mainBuffer() == mMixerBuffer)) { 4020 mAudioMixer->setParameter( 4021 name, 4022 AudioMixer::TRACK, 4023 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4024 mAudioMixer->setParameter( 4025 name, 4026 AudioMixer::TRACK, 4027 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4028 // TODO: override track->mainBuffer()? 4029 mMixerBufferValid = true; 4030 } else { 4031 mAudioMixer->setParameter( 4032 name, 4033 AudioMixer::TRACK, 4034 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4035 mAudioMixer->setParameter( 4036 name, 4037 AudioMixer::TRACK, 4038 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4039 } 4040 mAudioMixer->setParameter( 4041 name, 4042 AudioMixer::TRACK, 4043 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4044 4045 // reset retry count 4046 track->mRetryCount = kMaxTrackRetries; 4047 4048 // If one track is ready, set the mixer ready if: 4049 // - the mixer was not ready during previous round OR 4050 // - no other track is not ready 4051 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4052 mixerStatus != MIXER_TRACKS_ENABLED) { 4053 mixerStatus = MIXER_TRACKS_READY; 4054 } 4055 } else { 4056 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4057 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4058 track, framesReady, desiredFrames); 4059 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4060 } 4061 // clear effect chain input buffer if an active track underruns to avoid sending 4062 // previous audio buffer again to effects 4063 chain = getEffectChain_l(track->sessionId()); 4064 if (chain != 0) { 4065 chain->clearInputBuffer(); 4066 } 4067 4068 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4069 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4070 track->isStopped() || track->isPaused()) { 4071 // We have consumed all the buffers of this track. 4072 // Remove it from the list of active tracks. 4073 // TODO: use actual buffer filling status instead of latency when available from 4074 // audio HAL 4075 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4076 size_t framesWritten = mBytesWritten / mFrameSize; 4077 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4078 if (track->isStopped()) { 4079 track->reset(); 4080 } 4081 tracksToRemove->add(track); 4082 } 4083 } else { 4084 // No buffers for this track. Give it a few chances to 4085 // fill a buffer, then remove it from active list. 4086 if (--(track->mRetryCount) <= 0) { 4087 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4088 tracksToRemove->add(track); 4089 // indicate to client process that the track was disabled because of underrun; 4090 // it will then automatically call start() when data is available 4091 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4092 // If one track is not ready, mark the mixer also not ready if: 4093 // - the mixer was ready during previous round OR 4094 // - no other track is ready 4095 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4096 mixerStatus != MIXER_TRACKS_READY) { 4097 mixerStatus = MIXER_TRACKS_ENABLED; 4098 } 4099 } 4100 mAudioMixer->disable(name); 4101 } 4102 4103 } // local variable scope to avoid goto warning 4104track_is_ready: ; 4105 4106 } 4107 4108 // Push the new FastMixer state if necessary 4109 bool pauseAudioWatchdog = false; 4110 if (didModify) { 4111 state->mFastTracksGen++; 4112 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4113 if (kUseFastMixer == FastMixer_Dynamic && 4114 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4115 state->mCommand = FastMixerState::COLD_IDLE; 4116 state->mColdFutexAddr = &mFastMixerFutex; 4117 state->mColdGen++; 4118 mFastMixerFutex = 0; 4119 if (kUseFastMixer == FastMixer_Dynamic) { 4120 mNormalSink = mOutputSink; 4121 } 4122 // If we go into cold idle, need to wait for acknowledgement 4123 // so that fast mixer stops doing I/O. 4124 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4125 pauseAudioWatchdog = true; 4126 } 4127 } 4128 if (sq != NULL) { 4129 sq->end(didModify); 4130 sq->push(block); 4131 } 4132#ifdef AUDIO_WATCHDOG 4133 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4134 mAudioWatchdog->pause(); 4135 } 4136#endif 4137 4138 // Now perform the deferred reset on fast tracks that have stopped 4139 while (resetMask != 0) { 4140 size_t i = __builtin_ctz(resetMask); 4141 ALOG_ASSERT(i < count); 4142 resetMask &= ~(1 << i); 4143 sp<Track> t = mActiveTracks[i].promote(); 4144 if (t == 0) { 4145 continue; 4146 } 4147 Track* track = t.get(); 4148 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4149 track->reset(); 4150 } 4151 4152 // remove all the tracks that need to be... 4153 removeTracks_l(*tracksToRemove); 4154 4155 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4156 mEffectBufferValid = true; 4157 } 4158 4159 if (mEffectBufferValid) { 4160 // as long as there are effects we should clear the effects buffer, to avoid 4161 // passing a non-clean buffer to the effect chain 4162 memset(mEffectBuffer, 0, mEffectBufferSize); 4163 } 4164 // sink or mix buffer must be cleared if all tracks are connected to an 4165 // effect chain as in this case the mixer will not write to the sink or mix buffer 4166 // and track effects will accumulate into it 4167 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4168 (mixedTracks == 0 && fastTracks > 0))) { 4169 // FIXME as a performance optimization, should remember previous zero status 4170 if (mMixerBufferValid) { 4171 memset(mMixerBuffer, 0, mMixerBufferSize); 4172 // TODO: In testing, mSinkBuffer below need not be cleared because 4173 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4174 // after mixing. 4175 // 4176 // To enforce this guarantee: 4177 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4178 // (mixedTracks == 0 && fastTracks > 0)) 4179 // must imply MIXER_TRACKS_READY. 4180 // Later, we may clear buffers regardless, and skip much of this logic. 4181 } 4182 // FIXME as a performance optimization, should remember previous zero status 4183 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4184 } 4185 4186 // if any fast tracks, then status is ready 4187 mMixerStatusIgnoringFastTracks = mixerStatus; 4188 if (fastTracks > 0) { 4189 mixerStatus = MIXER_TRACKS_READY; 4190 } 4191 return mixerStatus; 4192} 4193 4194// getTrackName_l() must be called with ThreadBase::mLock held 4195int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4196 audio_format_t format, int sessionId) 4197{ 4198 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4199} 4200 4201// deleteTrackName_l() must be called with ThreadBase::mLock held 4202void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4203{ 4204 ALOGV("remove track (%d) and delete from mixer", name); 4205 mAudioMixer->deleteTrackName(name); 4206} 4207 4208// checkForNewParameter_l() must be called with ThreadBase::mLock held 4209bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4210 status_t& status) 4211{ 4212 bool reconfig = false; 4213 4214 status = NO_ERROR; 4215 4216 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4217 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4218 if (mFastMixer != 0) { 4219 FastMixerStateQueue *sq = mFastMixer->sq(); 4220 FastMixerState *state = sq->begin(); 4221 if (!(state->mCommand & FastMixerState::IDLE)) { 4222 previousCommand = state->mCommand; 4223 state->mCommand = FastMixerState::HOT_IDLE; 4224 sq->end(); 4225 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4226 } else { 4227 sq->end(false /*didModify*/); 4228 } 4229 } 4230 4231 AudioParameter param = AudioParameter(keyValuePair); 4232 int value; 4233 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4234 reconfig = true; 4235 } 4236 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4237 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4238 status = BAD_VALUE; 4239 } else { 4240 // no need to save value, since it's constant 4241 reconfig = true; 4242 } 4243 } 4244 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4245 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4246 status = BAD_VALUE; 4247 } else { 4248 // no need to save value, since it's constant 4249 reconfig = true; 4250 } 4251 } 4252 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4253 // do not accept frame count changes if tracks are open as the track buffer 4254 // size depends on frame count and correct behavior would not be guaranteed 4255 // if frame count is changed after track creation 4256 if (!mTracks.isEmpty()) { 4257 status = INVALID_OPERATION; 4258 } else { 4259 reconfig = true; 4260 } 4261 } 4262 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4263#ifdef ADD_BATTERY_DATA 4264 // when changing the audio output device, call addBatteryData to notify 4265 // the change 4266 if (mOutDevice != value) { 4267 uint32_t params = 0; 4268 // check whether speaker is on 4269 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4270 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4271 } 4272 4273 audio_devices_t deviceWithoutSpeaker 4274 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4275 // check if any other device (except speaker) is on 4276 if (value & deviceWithoutSpeaker) { 4277 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4278 } 4279 4280 if (params != 0) { 4281 addBatteryData(params); 4282 } 4283 } 4284#endif 4285 4286 // forward device change to effects that have requested to be 4287 // aware of attached audio device. 4288 if (value != AUDIO_DEVICE_NONE) { 4289 mOutDevice = value; 4290 for (size_t i = 0; i < mEffectChains.size(); i++) { 4291 mEffectChains[i]->setDevice_l(mOutDevice); 4292 } 4293 } 4294 } 4295 4296 if (status == NO_ERROR) { 4297 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4298 keyValuePair.string()); 4299 if (!mStandby && status == INVALID_OPERATION) { 4300 mOutput->standby(); 4301 mStandby = true; 4302 mBytesWritten = 0; 4303 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4304 keyValuePair.string()); 4305 } 4306 if (status == NO_ERROR && reconfig) { 4307 readOutputParameters_l(); 4308 delete mAudioMixer; 4309 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4310 for (size_t i = 0; i < mTracks.size() ; i++) { 4311 int name = getTrackName_l(mTracks[i]->mChannelMask, 4312 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4313 if (name < 0) { 4314 break; 4315 } 4316 mTracks[i]->mName = name; 4317 } 4318 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4319 } 4320 } 4321 4322 if (!(previousCommand & FastMixerState::IDLE)) { 4323 ALOG_ASSERT(mFastMixer != 0); 4324 FastMixerStateQueue *sq = mFastMixer->sq(); 4325 FastMixerState *state = sq->begin(); 4326 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4327 state->mCommand = previousCommand; 4328 sq->end(); 4329 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4330 } 4331 4332 return reconfig; 4333} 4334 4335 4336void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4337{ 4338 const size_t SIZE = 256; 4339 char buffer[SIZE]; 4340 String8 result; 4341 4342 PlaybackThread::dumpInternals(fd, args); 4343 4344 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4345 4346 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4347 const FastMixerDumpState copy(mFastMixerDumpState); 4348 copy.dump(fd); 4349 4350#ifdef STATE_QUEUE_DUMP 4351 // Similar for state queue 4352 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4353 observerCopy.dump(fd); 4354 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4355 mutatorCopy.dump(fd); 4356#endif 4357 4358#ifdef TEE_SINK 4359 // Write the tee output to a .wav file 4360 dumpTee(fd, mTeeSource, mId); 4361#endif 4362 4363#ifdef AUDIO_WATCHDOG 4364 if (mAudioWatchdog != 0) { 4365 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4366 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4367 wdCopy.dump(fd); 4368 } 4369#endif 4370} 4371 4372uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4373{ 4374 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4375} 4376 4377uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4378{ 4379 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4380} 4381 4382void AudioFlinger::MixerThread::cacheParameters_l() 4383{ 4384 PlaybackThread::cacheParameters_l(); 4385 4386 // FIXME: Relaxed timing because of a certain device that can't meet latency 4387 // Should be reduced to 2x after the vendor fixes the driver issue 4388 // increase threshold again due to low power audio mode. The way this warning 4389 // threshold is calculated and its usefulness should be reconsidered anyway. 4390 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4391} 4392 4393// ---------------------------------------------------------------------------- 4394 4395AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4396 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4397 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4398 // mLeftVolFloat, mRightVolFloat 4399{ 4400} 4401 4402AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4403 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4404 ThreadBase::type_t type, bool systemReady) 4405 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4406 // mLeftVolFloat, mRightVolFloat 4407{ 4408} 4409 4410AudioFlinger::DirectOutputThread::~DirectOutputThread() 4411{ 4412} 4413 4414void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4415{ 4416 audio_track_cblk_t* cblk = track->cblk(); 4417 float left, right; 4418 4419 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4420 left = right = 0; 4421 } else { 4422 float typeVolume = mStreamTypes[track->streamType()].volume; 4423 float v = mMasterVolume * typeVolume; 4424 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4425 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4426 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4427 if (left > GAIN_FLOAT_UNITY) { 4428 left = GAIN_FLOAT_UNITY; 4429 } 4430 left *= v; 4431 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4432 if (right > GAIN_FLOAT_UNITY) { 4433 right = GAIN_FLOAT_UNITY; 4434 } 4435 right *= v; 4436 } 4437 4438 if (lastTrack) { 4439 if (left != mLeftVolFloat || right != mRightVolFloat) { 4440 mLeftVolFloat = left; 4441 mRightVolFloat = right; 4442 4443 // Convert volumes from float to 8.24 4444 uint32_t vl = (uint32_t)(left * (1 << 24)); 4445 uint32_t vr = (uint32_t)(right * (1 << 24)); 4446 4447 // Delegate volume control to effect in track effect chain if needed 4448 // only one effect chain can be present on DirectOutputThread, so if 4449 // there is one, the track is connected to it 4450 if (!mEffectChains.isEmpty()) { 4451 mEffectChains[0]->setVolume_l(&vl, &vr); 4452 left = (float)vl / (1 << 24); 4453 right = (float)vr / (1 << 24); 4454 } 4455 if (mOutput->stream->set_volume) { 4456 mOutput->stream->set_volume(mOutput->stream, left, right); 4457 } 4458 } 4459 } 4460} 4461 4462void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4463{ 4464 sp<Track> previousTrack = mPreviousTrack.promote(); 4465 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4466 4467 if (previousTrack != 0 && latestTrack != 0 && 4468 (previousTrack->sessionId() != latestTrack->sessionId())) { 4469 mFlushPending = true; 4470 } 4471 PlaybackThread::onAddNewTrack_l(); 4472} 4473 4474AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4475 Vector< sp<Track> > *tracksToRemove 4476) 4477{ 4478 size_t count = mActiveTracks.size(); 4479 mixer_state mixerStatus = MIXER_IDLE; 4480 bool doHwPause = false; 4481 bool doHwResume = false; 4482 4483 // find out which tracks need to be processed 4484 for (size_t i = 0; i < count; i++) { 4485 sp<Track> t = mActiveTracks[i].promote(); 4486 // The track died recently 4487 if (t == 0) { 4488 continue; 4489 } 4490 4491 if (t->isInvalid()) { 4492 ALOGW("An invalidated track shouldn't be in active list"); 4493 tracksToRemove->add(t); 4494 continue; 4495 } 4496 4497 Track* const track = t.get(); 4498 audio_track_cblk_t* cblk = track->cblk(); 4499 // Only consider last track started for volume and mixer state control. 4500 // In theory an older track could underrun and restart after the new one starts 4501 // but as we only care about the transition phase between two tracks on a 4502 // direct output, it is not a problem to ignore the underrun case. 4503 sp<Track> l = mLatestActiveTrack.promote(); 4504 bool last = l.get() == track; 4505 4506 if (track->isPausing()) { 4507 track->setPaused(); 4508 if (mHwSupportsPause && last && !mHwPaused) { 4509 doHwPause = true; 4510 mHwPaused = true; 4511 } 4512 tracksToRemove->add(track); 4513 } else if (track->isFlushPending()) { 4514 track->flushAck(); 4515 if (last) { 4516 mFlushPending = true; 4517 } 4518 } else if (track->isResumePending()) { 4519 track->resumeAck(); 4520 if (last && mHwPaused) { 4521 doHwResume = true; 4522 mHwPaused = false; 4523 } 4524 } 4525 4526 // The first time a track is added we wait 4527 // for all its buffers to be filled before processing it. 4528 // Allow draining the buffer in case the client 4529 // app does not call stop() and relies on underrun to stop: 4530 // hence the test on (track->mRetryCount > 1). 4531 // If retryCount<=1 then track is about to underrun and be removed. 4532 uint32_t minFrames; 4533 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4534 && (track->mRetryCount > 1)) { 4535 minFrames = mNormalFrameCount; 4536 } else { 4537 minFrames = 1; 4538 } 4539 4540 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4541 !track->isStopping_2() && !track->isStopped()) 4542 { 4543 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4544 4545 if (track->mFillingUpStatus == Track::FS_FILLED) { 4546 track->mFillingUpStatus = Track::FS_ACTIVE; 4547 // make sure processVolume_l() will apply new volume even if 0 4548 mLeftVolFloat = mRightVolFloat = -1.0; 4549 if (!mHwSupportsPause) { 4550 track->resumeAck(); 4551 } 4552 } 4553 4554 // compute volume for this track 4555 processVolume_l(track, last); 4556 if (last) { 4557 sp<Track> previousTrack = mPreviousTrack.promote(); 4558 if (previousTrack != 0) { 4559 if (track != previousTrack.get()) { 4560 // Flush any data still being written from last track 4561 mBytesRemaining = 0; 4562 // flush data already sent if changing audio session as audio 4563 // comes from a different source. Also invalidate previous track to force a 4564 // seek when resuming. 4565 if (previousTrack->sessionId() != track->sessionId()) { 4566 previousTrack->invalidate(); 4567 } 4568 } 4569 } 4570 mPreviousTrack = track; 4571 4572 // reset retry count 4573 track->mRetryCount = kMaxTrackRetriesDirect; 4574 mActiveTrack = t; 4575 mixerStatus = MIXER_TRACKS_READY; 4576 if (mHwPaused) { 4577 doHwResume = true; 4578 mHwPaused = false; 4579 } 4580 } 4581 } else { 4582 // clear effect chain input buffer if the last active track started underruns 4583 // to avoid sending previous audio buffer again to effects 4584 if (!mEffectChains.isEmpty() && last) { 4585 mEffectChains[0]->clearInputBuffer(); 4586 } 4587 if (track->isStopping_1()) { 4588 track->mState = TrackBase::STOPPING_2; 4589 if (last && mHwPaused) { 4590 doHwResume = true; 4591 mHwPaused = false; 4592 } 4593 } 4594 if ((track->sharedBuffer() != 0) || track->isStopped() || 4595 track->isStopping_2() || track->isPaused()) { 4596 // We have consumed all the buffers of this track. 4597 // Remove it from the list of active tracks. 4598 size_t audioHALFrames; 4599 if (audio_is_linear_pcm(mFormat)) { 4600 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4601 } else { 4602 audioHALFrames = 0; 4603 } 4604 4605 size_t framesWritten = mBytesWritten / mFrameSize; 4606 if (mStandby || !last || 4607 track->presentationComplete(framesWritten, audioHALFrames)) { 4608 if (track->isStopping_2()) { 4609 track->mState = TrackBase::STOPPED; 4610 } 4611 if (track->isStopped()) { 4612 track->reset(); 4613 } 4614 tracksToRemove->add(track); 4615 } 4616 } else { 4617 // No buffers for this track. Give it a few chances to 4618 // fill a buffer, then remove it from active list. 4619 // Only consider last track started for mixer state control 4620 if (--(track->mRetryCount) <= 0) { 4621 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4622 tracksToRemove->add(track); 4623 // indicate to client process that the track was disabled because of underrun; 4624 // it will then automatically call start() when data is available 4625 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4626 } else if (last) { 4627 mixerStatus = MIXER_TRACKS_ENABLED; 4628 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4629 doHwPause = true; 4630 mHwPaused = true; 4631 } 4632 } 4633 } 4634 } 4635 } 4636 4637 // if an active track did not command a flush, check for pending flush on stopped tracks 4638 if (!mFlushPending) { 4639 for (size_t i = 0; i < mTracks.size(); i++) { 4640 if (mTracks[i]->isFlushPending()) { 4641 mTracks[i]->flushAck(); 4642 mFlushPending = true; 4643 } 4644 } 4645 } 4646 4647 // make sure the pause/flush/resume sequence is executed in the right order. 4648 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4649 // before flush and then resume HW. This can happen in case of pause/flush/resume 4650 // if resume is received before pause is executed. 4651 if (mHwSupportsPause && !mStandby && 4652 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4653 mOutput->stream->pause(mOutput->stream); 4654 } 4655 if (mFlushPending) { 4656 flushHw_l(); 4657 } 4658 if (mHwSupportsPause && !mStandby && doHwResume) { 4659 mOutput->stream->resume(mOutput->stream); 4660 } 4661 // remove all the tracks that need to be... 4662 removeTracks_l(*tracksToRemove); 4663 4664 return mixerStatus; 4665} 4666 4667void AudioFlinger::DirectOutputThread::threadLoop_mix() 4668{ 4669 size_t frameCount = mFrameCount; 4670 int8_t *curBuf = (int8_t *)mSinkBuffer; 4671 // output audio to hardware 4672 while (frameCount) { 4673 AudioBufferProvider::Buffer buffer; 4674 buffer.frameCount = frameCount; 4675 status_t status = mActiveTrack->getNextBuffer(&buffer); 4676 if (status != NO_ERROR || buffer.raw == NULL) { 4677 memset(curBuf, 0, frameCount * mFrameSize); 4678 break; 4679 } 4680 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4681 frameCount -= buffer.frameCount; 4682 curBuf += buffer.frameCount * mFrameSize; 4683 mActiveTrack->releaseBuffer(&buffer); 4684 } 4685 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4686 mSleepTimeUs = 0; 4687 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4688 mActiveTrack.clear(); 4689} 4690 4691void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4692{ 4693 // do not write to HAL when paused 4694 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4695 mSleepTimeUs = mIdleSleepTimeUs; 4696 return; 4697 } 4698 if (mSleepTimeUs == 0) { 4699 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4700 mSleepTimeUs = mActiveSleepTimeUs; 4701 } else { 4702 mSleepTimeUs = mIdleSleepTimeUs; 4703 } 4704 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4705 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4706 mSleepTimeUs = 0; 4707 } 4708} 4709 4710void AudioFlinger::DirectOutputThread::threadLoop_exit() 4711{ 4712 { 4713 Mutex::Autolock _l(mLock); 4714 for (size_t i = 0; i < mTracks.size(); i++) { 4715 if (mTracks[i]->isFlushPending()) { 4716 mTracks[i]->flushAck(); 4717 mFlushPending = true; 4718 } 4719 } 4720 if (mFlushPending) { 4721 flushHw_l(); 4722 } 4723 } 4724 PlaybackThread::threadLoop_exit(); 4725} 4726 4727// must be called with thread mutex locked 4728bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4729{ 4730 bool trackPaused = false; 4731 bool trackStopped = false; 4732 4733 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4734 // after a timeout and we will enter standby then. 4735 if (mTracks.size() > 0) { 4736 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4737 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4738 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4739 } 4740 4741 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4742} 4743 4744// getTrackName_l() must be called with ThreadBase::mLock held 4745int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4746 audio_format_t format __unused, int sessionId __unused) 4747{ 4748 return 0; 4749} 4750 4751// deleteTrackName_l() must be called with ThreadBase::mLock held 4752void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4753{ 4754} 4755 4756// checkForNewParameter_l() must be called with ThreadBase::mLock held 4757bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4758 status_t& status) 4759{ 4760 bool reconfig = false; 4761 4762 status = NO_ERROR; 4763 4764 AudioParameter param = AudioParameter(keyValuePair); 4765 int value; 4766 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4767 // forward device change to effects that have requested to be 4768 // aware of attached audio device. 4769 if (value != AUDIO_DEVICE_NONE) { 4770 mOutDevice = value; 4771 for (size_t i = 0; i < mEffectChains.size(); i++) { 4772 mEffectChains[i]->setDevice_l(mOutDevice); 4773 } 4774 } 4775 } 4776 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4777 // do not accept frame count changes if tracks are open as the track buffer 4778 // size depends on frame count and correct behavior would not be garantied 4779 // if frame count is changed after track creation 4780 if (!mTracks.isEmpty()) { 4781 status = INVALID_OPERATION; 4782 } else { 4783 reconfig = true; 4784 } 4785 } 4786 if (status == NO_ERROR) { 4787 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4788 keyValuePair.string()); 4789 if (!mStandby && status == INVALID_OPERATION) { 4790 mOutput->standby(); 4791 mStandby = true; 4792 mBytesWritten = 0; 4793 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4794 keyValuePair.string()); 4795 } 4796 if (status == NO_ERROR && reconfig) { 4797 readOutputParameters_l(); 4798 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4799 } 4800 } 4801 4802 return reconfig; 4803} 4804 4805uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4806{ 4807 uint32_t time; 4808 if (audio_is_linear_pcm(mFormat)) { 4809 time = PlaybackThread::activeSleepTimeUs(); 4810 } else { 4811 time = 10000; 4812 } 4813 return time; 4814} 4815 4816uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4817{ 4818 uint32_t time; 4819 if (audio_is_linear_pcm(mFormat)) { 4820 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4821 } else { 4822 time = 10000; 4823 } 4824 return time; 4825} 4826 4827uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4828{ 4829 uint32_t time; 4830 if (audio_is_linear_pcm(mFormat)) { 4831 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4832 } else { 4833 time = 10000; 4834 } 4835 return time; 4836} 4837 4838void AudioFlinger::DirectOutputThread::cacheParameters_l() 4839{ 4840 PlaybackThread::cacheParameters_l(); 4841 4842 // use shorter standby delay as on normal output to release 4843 // hardware resources as soon as possible 4844 // no delay on outputs with HW A/V sync 4845 if (usesHwAvSync()) { 4846 mStandbyDelayNs = 0; 4847 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 4848 mStandbyDelayNs = kOffloadStandbyDelayNs; 4849 } else { 4850 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 4851 } 4852} 4853 4854void AudioFlinger::DirectOutputThread::flushHw_l() 4855{ 4856 mOutput->flush(); 4857 mHwPaused = false; 4858 mFlushPending = false; 4859} 4860 4861// ---------------------------------------------------------------------------- 4862 4863AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4864 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4865 : Thread(false /*canCallJava*/), 4866 mPlaybackThread(playbackThread), 4867 mWriteAckSequence(0), 4868 mDrainSequence(0) 4869{ 4870} 4871 4872AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4873{ 4874} 4875 4876void AudioFlinger::AsyncCallbackThread::onFirstRef() 4877{ 4878 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4879} 4880 4881bool AudioFlinger::AsyncCallbackThread::threadLoop() 4882{ 4883 while (!exitPending()) { 4884 uint32_t writeAckSequence; 4885 uint32_t drainSequence; 4886 4887 { 4888 Mutex::Autolock _l(mLock); 4889 while (!((mWriteAckSequence & 1) || 4890 (mDrainSequence & 1) || 4891 exitPending())) { 4892 mWaitWorkCV.wait(mLock); 4893 } 4894 4895 if (exitPending()) { 4896 break; 4897 } 4898 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4899 mWriteAckSequence, mDrainSequence); 4900 writeAckSequence = mWriteAckSequence; 4901 mWriteAckSequence &= ~1; 4902 drainSequence = mDrainSequence; 4903 mDrainSequence &= ~1; 4904 } 4905 { 4906 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4907 if (playbackThread != 0) { 4908 if (writeAckSequence & 1) { 4909 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4910 } 4911 if (drainSequence & 1) { 4912 playbackThread->resetDraining(drainSequence >> 1); 4913 } 4914 } 4915 } 4916 } 4917 return false; 4918} 4919 4920void AudioFlinger::AsyncCallbackThread::exit() 4921{ 4922 ALOGV("AsyncCallbackThread::exit"); 4923 Mutex::Autolock _l(mLock); 4924 requestExit(); 4925 mWaitWorkCV.broadcast(); 4926} 4927 4928void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4929{ 4930 Mutex::Autolock _l(mLock); 4931 // bit 0 is cleared 4932 mWriteAckSequence = sequence << 1; 4933} 4934 4935void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4936{ 4937 Mutex::Autolock _l(mLock); 4938 // ignore unexpected callbacks 4939 if (mWriteAckSequence & 2) { 4940 mWriteAckSequence |= 1; 4941 mWaitWorkCV.signal(); 4942 } 4943} 4944 4945void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4946{ 4947 Mutex::Autolock _l(mLock); 4948 // bit 0 is cleared 4949 mDrainSequence = sequence << 1; 4950} 4951 4952void AudioFlinger::AsyncCallbackThread::resetDraining() 4953{ 4954 Mutex::Autolock _l(mLock); 4955 // ignore unexpected callbacks 4956 if (mDrainSequence & 2) { 4957 mDrainSequence |= 1; 4958 mWaitWorkCV.signal(); 4959 } 4960} 4961 4962 4963// ---------------------------------------------------------------------------- 4964AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4965 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 4966 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 4967 mPausedBytesRemaining(0) 4968{ 4969 //FIXME: mStandby should be set to true by ThreadBase constructor 4970 mStandby = true; 4971} 4972 4973void AudioFlinger::OffloadThread::threadLoop_exit() 4974{ 4975 if (mFlushPending || mHwPaused) { 4976 // If a flush is pending or track was paused, just discard buffered data 4977 flushHw_l(); 4978 } else { 4979 mMixerStatus = MIXER_DRAIN_ALL; 4980 threadLoop_drain(); 4981 } 4982 if (mUseAsyncWrite) { 4983 ALOG_ASSERT(mCallbackThread != 0); 4984 mCallbackThread->exit(); 4985 } 4986 PlaybackThread::threadLoop_exit(); 4987} 4988 4989AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4990 Vector< sp<Track> > *tracksToRemove 4991) 4992{ 4993 size_t count = mActiveTracks.size(); 4994 4995 mixer_state mixerStatus = MIXER_IDLE; 4996 bool doHwPause = false; 4997 bool doHwResume = false; 4998 4999 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5000 5001 // find out which tracks need to be processed 5002 for (size_t i = 0; i < count; i++) { 5003 sp<Track> t = mActiveTracks[i].promote(); 5004 // The track died recently 5005 if (t == 0) { 5006 continue; 5007 } 5008 Track* const track = t.get(); 5009 audio_track_cblk_t* cblk = track->cblk(); 5010 // Only consider last track started for volume and mixer state control. 5011 // In theory an older track could underrun and restart after the new one starts 5012 // but as we only care about the transition phase between two tracks on a 5013 // direct output, it is not a problem to ignore the underrun case. 5014 sp<Track> l = mLatestActiveTrack.promote(); 5015 bool last = l.get() == track; 5016 5017 if (track->isInvalid()) { 5018 ALOGW("An invalidated track shouldn't be in active list"); 5019 tracksToRemove->add(track); 5020 continue; 5021 } 5022 5023 if (track->mState == TrackBase::IDLE) { 5024 ALOGW("An idle track shouldn't be in active list"); 5025 continue; 5026 } 5027 5028 if (track->isPausing()) { 5029 track->setPaused(); 5030 if (last) { 5031 if (mHwSupportsPause && !mHwPaused) { 5032 doHwPause = true; 5033 mHwPaused = true; 5034 } 5035 // If we were part way through writing the mixbuffer to 5036 // the HAL we must save this until we resume 5037 // BUG - this will be wrong if a different track is made active, 5038 // in that case we want to discard the pending data in the 5039 // mixbuffer and tell the client to present it again when the 5040 // track is resumed 5041 mPausedWriteLength = mCurrentWriteLength; 5042 mPausedBytesRemaining = mBytesRemaining; 5043 mBytesRemaining = 0; // stop writing 5044 } 5045 tracksToRemove->add(track); 5046 } else if (track->isFlushPending()) { 5047 track->flushAck(); 5048 if (last) { 5049 mFlushPending = true; 5050 } 5051 } else if (track->isResumePending()){ 5052 track->resumeAck(); 5053 if (last) { 5054 if (mPausedBytesRemaining) { 5055 // Need to continue write that was interrupted 5056 mCurrentWriteLength = mPausedWriteLength; 5057 mBytesRemaining = mPausedBytesRemaining; 5058 mPausedBytesRemaining = 0; 5059 } 5060 if (mHwPaused) { 5061 doHwResume = true; 5062 mHwPaused = false; 5063 // threadLoop_mix() will handle the case that we need to 5064 // resume an interrupted write 5065 } 5066 // enable write to audio HAL 5067 mSleepTimeUs = 0; 5068 5069 // Do not handle new data in this iteration even if track->framesReady() 5070 mixerStatus = MIXER_TRACKS_ENABLED; 5071 } 5072 } else if (track->framesReady() && track->isReady() && 5073 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5074 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5075 if (track->mFillingUpStatus == Track::FS_FILLED) { 5076 track->mFillingUpStatus = Track::FS_ACTIVE; 5077 // make sure processVolume_l() will apply new volume even if 0 5078 mLeftVolFloat = mRightVolFloat = -1.0; 5079 } 5080 5081 if (last) { 5082 sp<Track> previousTrack = mPreviousTrack.promote(); 5083 if (previousTrack != 0) { 5084 if (track != previousTrack.get()) { 5085 // Flush any data still being written from last track 5086 mBytesRemaining = 0; 5087 if (mPausedBytesRemaining) { 5088 // Last track was paused so we also need to flush saved 5089 // mixbuffer state and invalidate track so that it will 5090 // re-submit that unwritten data when it is next resumed 5091 mPausedBytesRemaining = 0; 5092 // Invalidate is a bit drastic - would be more efficient 5093 // to have a flag to tell client that some of the 5094 // previously written data was lost 5095 previousTrack->invalidate(); 5096 } 5097 // flush data already sent to the DSP if changing audio session as audio 5098 // comes from a different source. Also invalidate previous track to force a 5099 // seek when resuming. 5100 if (previousTrack->sessionId() != track->sessionId()) { 5101 previousTrack->invalidate(); 5102 } 5103 } 5104 } 5105 mPreviousTrack = track; 5106 // reset retry count 5107 track->mRetryCount = kMaxTrackRetriesOffload; 5108 mActiveTrack = t; 5109 mixerStatus = MIXER_TRACKS_READY; 5110 } 5111 } else { 5112 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5113 if (track->isStopping_1()) { 5114 // Hardware buffer can hold a large amount of audio so we must 5115 // wait for all current track's data to drain before we say 5116 // that the track is stopped. 5117 if (mBytesRemaining == 0) { 5118 // Only start draining when all data in mixbuffer 5119 // has been written 5120 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5121 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5122 // do not drain if no data was ever sent to HAL (mStandby == true) 5123 if (last && !mStandby) { 5124 // do not modify drain sequence if we are already draining. This happens 5125 // when resuming from pause after drain. 5126 if ((mDrainSequence & 1) == 0) { 5127 mSleepTimeUs = 0; 5128 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5129 mixerStatus = MIXER_DRAIN_TRACK; 5130 mDrainSequence += 2; 5131 } 5132 if (mHwPaused) { 5133 // It is possible to move from PAUSED to STOPPING_1 without 5134 // a resume so we must ensure hardware is running 5135 doHwResume = true; 5136 mHwPaused = false; 5137 } 5138 } 5139 } 5140 } else if (track->isStopping_2()) { 5141 // Drain has completed or we are in standby, signal presentation complete 5142 if (!(mDrainSequence & 1) || !last || mStandby) { 5143 track->mState = TrackBase::STOPPED; 5144 size_t audioHALFrames = 5145 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5146 size_t framesWritten = 5147 mBytesWritten / mOutput->getFrameSize(); 5148 track->presentationComplete(framesWritten, audioHALFrames); 5149 track->reset(); 5150 tracksToRemove->add(track); 5151 } 5152 } else { 5153 // No buffers for this track. Give it a few chances to 5154 // fill a buffer, then remove it from active list. 5155 if (--(track->mRetryCount) <= 0) { 5156 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5157 track->name()); 5158 tracksToRemove->add(track); 5159 // indicate to client process that the track was disabled because of underrun; 5160 // it will then automatically call start() when data is available 5161 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5162 } else if (last){ 5163 mixerStatus = MIXER_TRACKS_ENABLED; 5164 } 5165 } 5166 } 5167 // compute volume for this track 5168 processVolume_l(track, last); 5169 } 5170 5171 // make sure the pause/flush/resume sequence is executed in the right order. 5172 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5173 // before flush and then resume HW. This can happen in case of pause/flush/resume 5174 // if resume is received before pause is executed. 5175 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5176 mOutput->stream->pause(mOutput->stream); 5177 } 5178 if (mFlushPending) { 5179 flushHw_l(); 5180 } 5181 if (!mStandby && doHwResume) { 5182 mOutput->stream->resume(mOutput->stream); 5183 } 5184 5185 // remove all the tracks that need to be... 5186 removeTracks_l(*tracksToRemove); 5187 5188 return mixerStatus; 5189} 5190 5191// must be called with thread mutex locked 5192bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5193{ 5194 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5195 mWriteAckSequence, mDrainSequence); 5196 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5197 return true; 5198 } 5199 return false; 5200} 5201 5202bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5203{ 5204 Mutex::Autolock _l(mLock); 5205 return waitingAsyncCallback_l(); 5206} 5207 5208void AudioFlinger::OffloadThread::flushHw_l() 5209{ 5210 DirectOutputThread::flushHw_l(); 5211 // Flush anything still waiting in the mixbuffer 5212 mCurrentWriteLength = 0; 5213 mBytesRemaining = 0; 5214 mPausedWriteLength = 0; 5215 mPausedBytesRemaining = 0; 5216 5217 if (mUseAsyncWrite) { 5218 // discard any pending drain or write ack by incrementing sequence 5219 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5220 mDrainSequence = (mDrainSequence + 2) & ~1; 5221 ALOG_ASSERT(mCallbackThread != 0); 5222 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5223 mCallbackThread->setDraining(mDrainSequence); 5224 } 5225} 5226 5227// ---------------------------------------------------------------------------- 5228 5229AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5230 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5231 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5232 systemReady, DUPLICATING), 5233 mWaitTimeMs(UINT_MAX) 5234{ 5235 addOutputTrack(mainThread); 5236} 5237 5238AudioFlinger::DuplicatingThread::~DuplicatingThread() 5239{ 5240 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5241 mOutputTracks[i]->destroy(); 5242 } 5243} 5244 5245void AudioFlinger::DuplicatingThread::threadLoop_mix() 5246{ 5247 // mix buffers... 5248 if (outputsReady(outputTracks)) { 5249 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5250 } else { 5251 if (mMixerBufferValid) { 5252 memset(mMixerBuffer, 0, mMixerBufferSize); 5253 } else { 5254 memset(mSinkBuffer, 0, mSinkBufferSize); 5255 } 5256 } 5257 mSleepTimeUs = 0; 5258 writeFrames = mNormalFrameCount; 5259 mCurrentWriteLength = mSinkBufferSize; 5260 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5261} 5262 5263void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5264{ 5265 if (mSleepTimeUs == 0) { 5266 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5267 mSleepTimeUs = mActiveSleepTimeUs; 5268 } else { 5269 mSleepTimeUs = mIdleSleepTimeUs; 5270 } 5271 } else if (mBytesWritten != 0) { 5272 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5273 writeFrames = mNormalFrameCount; 5274 memset(mSinkBuffer, 0, mSinkBufferSize); 5275 } else { 5276 // flush remaining overflow buffers in output tracks 5277 writeFrames = 0; 5278 } 5279 mSleepTimeUs = 0; 5280 } 5281} 5282 5283ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5284{ 5285 for (size_t i = 0; i < outputTracks.size(); i++) { 5286 outputTracks[i]->write(mSinkBuffer, writeFrames); 5287 } 5288 mStandby = false; 5289 return (ssize_t)mSinkBufferSize; 5290} 5291 5292void AudioFlinger::DuplicatingThread::threadLoop_standby() 5293{ 5294 // DuplicatingThread implements standby by stopping all tracks 5295 for (size_t i = 0; i < outputTracks.size(); i++) { 5296 outputTracks[i]->stop(); 5297 } 5298} 5299 5300void AudioFlinger::DuplicatingThread::saveOutputTracks() 5301{ 5302 outputTracks = mOutputTracks; 5303} 5304 5305void AudioFlinger::DuplicatingThread::clearOutputTracks() 5306{ 5307 outputTracks.clear(); 5308} 5309 5310void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5311{ 5312 Mutex::Autolock _l(mLock); 5313 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5314 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5315 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5316 const size_t frameCount = 5317 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5318 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5319 // from different OutputTracks and their associated MixerThreads (e.g. one may 5320 // nearly empty and the other may be dropping data). 5321 5322 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5323 this, 5324 mSampleRate, 5325 mFormat, 5326 mChannelMask, 5327 frameCount, 5328 IPCThreadState::self()->getCallingUid()); 5329 if (outputTrack->cblk() != NULL) { 5330 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5331 mOutputTracks.add(outputTrack); 5332 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5333 updateWaitTime_l(); 5334 } 5335} 5336 5337void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5338{ 5339 Mutex::Autolock _l(mLock); 5340 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5341 if (mOutputTracks[i]->thread() == thread) { 5342 mOutputTracks[i]->destroy(); 5343 mOutputTracks.removeAt(i); 5344 updateWaitTime_l(); 5345 if (thread->getOutput() == mOutput) { 5346 mOutput = NULL; 5347 } 5348 return; 5349 } 5350 } 5351 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5352} 5353 5354// caller must hold mLock 5355void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5356{ 5357 mWaitTimeMs = UINT_MAX; 5358 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5359 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5360 if (strong != 0) { 5361 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5362 if (waitTimeMs < mWaitTimeMs) { 5363 mWaitTimeMs = waitTimeMs; 5364 } 5365 } 5366 } 5367} 5368 5369 5370bool AudioFlinger::DuplicatingThread::outputsReady( 5371 const SortedVector< sp<OutputTrack> > &outputTracks) 5372{ 5373 for (size_t i = 0; i < outputTracks.size(); i++) { 5374 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5375 if (thread == 0) { 5376 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5377 outputTracks[i].get()); 5378 return false; 5379 } 5380 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5381 // see note at standby() declaration 5382 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5383 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5384 thread.get()); 5385 return false; 5386 } 5387 } 5388 return true; 5389} 5390 5391uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5392{ 5393 return (mWaitTimeMs * 1000) / 2; 5394} 5395 5396void AudioFlinger::DuplicatingThread::cacheParameters_l() 5397{ 5398 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5399 updateWaitTime_l(); 5400 5401 MixerThread::cacheParameters_l(); 5402} 5403 5404// ---------------------------------------------------------------------------- 5405// Record 5406// ---------------------------------------------------------------------------- 5407 5408AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5409 AudioStreamIn *input, 5410 audio_io_handle_t id, 5411 audio_devices_t outDevice, 5412 audio_devices_t inDevice, 5413 bool systemReady 5414#ifdef TEE_SINK 5415 , const sp<NBAIO_Sink>& teeSink 5416#endif 5417 ) : 5418 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5419 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5420 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5421 mRsmpInRear(0) 5422#ifdef TEE_SINK 5423 , mTeeSink(teeSink) 5424#endif 5425 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5426 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5427 // mFastCapture below 5428 , mFastCaptureFutex(0) 5429 // mInputSource 5430 // mPipeSink 5431 // mPipeSource 5432 , mPipeFramesP2(0) 5433 // mPipeMemory 5434 // mFastCaptureNBLogWriter 5435 , mFastTrackAvail(false) 5436{ 5437 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5438 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5439 5440 readInputParameters_l(); 5441 5442 // create an NBAIO source for the HAL input stream, and negotiate 5443 mInputSource = new AudioStreamInSource(input->stream); 5444 size_t numCounterOffers = 0; 5445 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5446 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5447 ALOG_ASSERT(index == 0); 5448 5449 // initialize fast capture depending on configuration 5450 bool initFastCapture; 5451 switch (kUseFastCapture) { 5452 case FastCapture_Never: 5453 initFastCapture = false; 5454 break; 5455 case FastCapture_Always: 5456 initFastCapture = true; 5457 break; 5458 case FastCapture_Static: 5459 uint32_t primaryOutputSampleRate; 5460 { 5461 AutoMutex _l(audioFlinger->mHardwareLock); 5462 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5463 } 5464 initFastCapture = 5465 // either capture sample rate is same as (a reasonable) primary output sample rate 5466 ((isMusicRate(primaryOutputSampleRate) && 5467 (mSampleRate == primaryOutputSampleRate)) || 5468 // or primary output sample rate is unknown, and capture sample rate is reasonable 5469 ((primaryOutputSampleRate == 0) && 5470 isMusicRate(mSampleRate))) && 5471 // and the buffer size is < 12 ms 5472 (mFrameCount * 1000) / mSampleRate < 12; 5473 break; 5474 // case FastCapture_Dynamic: 5475 } 5476 5477 if (initFastCapture) { 5478 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5479 NBAIO_Format format = mInputSource->format(); 5480 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5481 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5482 void *pipeBuffer; 5483 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5484 sp<IMemory> pipeMemory; 5485 if ((roHeap == 0) || 5486 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5487 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5488 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5489 goto failed; 5490 } 5491 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5492 memset(pipeBuffer, 0, pipeSize); 5493 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5494 const NBAIO_Format offers[1] = {format}; 5495 size_t numCounterOffers = 0; 5496 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5497 ALOG_ASSERT(index == 0); 5498 mPipeSink = pipe; 5499 PipeReader *pipeReader = new PipeReader(*pipe); 5500 numCounterOffers = 0; 5501 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5502 ALOG_ASSERT(index == 0); 5503 mPipeSource = pipeReader; 5504 mPipeFramesP2 = pipeFramesP2; 5505 mPipeMemory = pipeMemory; 5506 5507 // create fast capture 5508 mFastCapture = new FastCapture(); 5509 FastCaptureStateQueue *sq = mFastCapture->sq(); 5510#ifdef STATE_QUEUE_DUMP 5511 // FIXME 5512#endif 5513 FastCaptureState *state = sq->begin(); 5514 state->mCblk = NULL; 5515 state->mInputSource = mInputSource.get(); 5516 state->mInputSourceGen++; 5517 state->mPipeSink = pipe; 5518 state->mPipeSinkGen++; 5519 state->mFrameCount = mFrameCount; 5520 state->mCommand = FastCaptureState::COLD_IDLE; 5521 // already done in constructor initialization list 5522 //mFastCaptureFutex = 0; 5523 state->mColdFutexAddr = &mFastCaptureFutex; 5524 state->mColdGen++; 5525 state->mDumpState = &mFastCaptureDumpState; 5526#ifdef TEE_SINK 5527 // FIXME 5528#endif 5529 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5530 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5531 sq->end(); 5532 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5533 5534 // start the fast capture 5535 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5536 pid_t tid = mFastCapture->getTid(); 5537 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5538#ifdef AUDIO_WATCHDOG 5539 // FIXME 5540#endif 5541 5542 mFastTrackAvail = true; 5543 } 5544failed: ; 5545 5546 // FIXME mNormalSource 5547} 5548 5549AudioFlinger::RecordThread::~RecordThread() 5550{ 5551 if (mFastCapture != 0) { 5552 FastCaptureStateQueue *sq = mFastCapture->sq(); 5553 FastCaptureState *state = sq->begin(); 5554 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5555 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5556 if (old == -1) { 5557 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5558 } 5559 } 5560 state->mCommand = FastCaptureState::EXIT; 5561 sq->end(); 5562 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5563 mFastCapture->join(); 5564 mFastCapture.clear(); 5565 } 5566 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5567 mAudioFlinger->unregisterWriter(mNBLogWriter); 5568 free(mRsmpInBuffer); 5569} 5570 5571void AudioFlinger::RecordThread::onFirstRef() 5572{ 5573 run(mThreadName, PRIORITY_URGENT_AUDIO); 5574} 5575 5576bool AudioFlinger::RecordThread::threadLoop() 5577{ 5578 nsecs_t lastWarning = 0; 5579 5580 inputStandBy(); 5581 5582reacquire_wakelock: 5583 sp<RecordTrack> activeTrack; 5584 int activeTracksGen; 5585 { 5586 Mutex::Autolock _l(mLock); 5587 size_t size = mActiveTracks.size(); 5588 activeTracksGen = mActiveTracksGen; 5589 if (size > 0) { 5590 // FIXME an arbitrary choice 5591 activeTrack = mActiveTracks[0]; 5592 acquireWakeLock_l(activeTrack->uid()); 5593 if (size > 1) { 5594 SortedVector<int> tmp; 5595 for (size_t i = 0; i < size; i++) { 5596 tmp.add(mActiveTracks[i]->uid()); 5597 } 5598 updateWakeLockUids_l(tmp); 5599 } 5600 } else { 5601 acquireWakeLock_l(-1); 5602 } 5603 } 5604 5605 // used to request a deferred sleep, to be executed later while mutex is unlocked 5606 uint32_t sleepUs = 0; 5607 5608 // loop while there is work to do 5609 for (;;) { 5610 Vector< sp<EffectChain> > effectChains; 5611 5612 // sleep with mutex unlocked 5613 if (sleepUs > 0) { 5614 ATRACE_BEGIN("sleep"); 5615 usleep(sleepUs); 5616 ATRACE_END(); 5617 sleepUs = 0; 5618 } 5619 5620 // activeTracks accumulates a copy of a subset of mActiveTracks 5621 Vector< sp<RecordTrack> > activeTracks; 5622 5623 // reference to the (first and only) active fast track 5624 sp<RecordTrack> fastTrack; 5625 5626 // reference to a fast track which is about to be removed 5627 sp<RecordTrack> fastTrackToRemove; 5628 5629 { // scope for mLock 5630 Mutex::Autolock _l(mLock); 5631 5632 processConfigEvents_l(); 5633 5634 // check exitPending here because checkForNewParameters_l() and 5635 // checkForNewParameters_l() can temporarily release mLock 5636 if (exitPending()) { 5637 break; 5638 } 5639 5640 // if no active track(s), then standby and release wakelock 5641 size_t size = mActiveTracks.size(); 5642 if (size == 0) { 5643 standbyIfNotAlreadyInStandby(); 5644 // exitPending() can't become true here 5645 releaseWakeLock_l(); 5646 ALOGV("RecordThread: loop stopping"); 5647 // go to sleep 5648 mWaitWorkCV.wait(mLock); 5649 ALOGV("RecordThread: loop starting"); 5650 goto reacquire_wakelock; 5651 } 5652 5653 if (mActiveTracksGen != activeTracksGen) { 5654 activeTracksGen = mActiveTracksGen; 5655 SortedVector<int> tmp; 5656 for (size_t i = 0; i < size; i++) { 5657 tmp.add(mActiveTracks[i]->uid()); 5658 } 5659 updateWakeLockUids_l(tmp); 5660 } 5661 5662 bool doBroadcast = false; 5663 for (size_t i = 0; i < size; ) { 5664 5665 activeTrack = mActiveTracks[i]; 5666 if (activeTrack->isTerminated()) { 5667 if (activeTrack->isFastTrack()) { 5668 ALOG_ASSERT(fastTrackToRemove == 0); 5669 fastTrackToRemove = activeTrack; 5670 } 5671 removeTrack_l(activeTrack); 5672 mActiveTracks.remove(activeTrack); 5673 mActiveTracksGen++; 5674 size--; 5675 continue; 5676 } 5677 5678 TrackBase::track_state activeTrackState = activeTrack->mState; 5679 switch (activeTrackState) { 5680 5681 case TrackBase::PAUSING: 5682 mActiveTracks.remove(activeTrack); 5683 mActiveTracksGen++; 5684 doBroadcast = true; 5685 size--; 5686 continue; 5687 5688 case TrackBase::STARTING_1: 5689 sleepUs = 10000; 5690 i++; 5691 continue; 5692 5693 case TrackBase::STARTING_2: 5694 doBroadcast = true; 5695 mStandby = false; 5696 activeTrack->mState = TrackBase::ACTIVE; 5697 break; 5698 5699 case TrackBase::ACTIVE: 5700 break; 5701 5702 case TrackBase::IDLE: 5703 i++; 5704 continue; 5705 5706 default: 5707 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5708 } 5709 5710 activeTracks.add(activeTrack); 5711 i++; 5712 5713 if (activeTrack->isFastTrack()) { 5714 ALOG_ASSERT(!mFastTrackAvail); 5715 ALOG_ASSERT(fastTrack == 0); 5716 fastTrack = activeTrack; 5717 } 5718 } 5719 if (doBroadcast) { 5720 mStartStopCond.broadcast(); 5721 } 5722 5723 // sleep if there are no active tracks to process 5724 if (activeTracks.size() == 0) { 5725 if (sleepUs == 0) { 5726 sleepUs = kRecordThreadSleepUs; 5727 } 5728 continue; 5729 } 5730 sleepUs = 0; 5731 5732 lockEffectChains_l(effectChains); 5733 } 5734 5735 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5736 5737 size_t size = effectChains.size(); 5738 for (size_t i = 0; i < size; i++) { 5739 // thread mutex is not locked, but effect chain is locked 5740 effectChains[i]->process_l(); 5741 } 5742 5743 // Push a new fast capture state if fast capture is not already running, or cblk change 5744 if (mFastCapture != 0) { 5745 FastCaptureStateQueue *sq = mFastCapture->sq(); 5746 FastCaptureState *state = sq->begin(); 5747 bool didModify = false; 5748 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5749 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5750 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5751 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5752 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5753 if (old == -1) { 5754 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5755 } 5756 } 5757 state->mCommand = FastCaptureState::READ_WRITE; 5758#if 0 // FIXME 5759 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5760 FastThreadDumpState::kSamplingNforLowRamDevice : 5761 FastThreadDumpState::kSamplingN); 5762#endif 5763 didModify = true; 5764 } 5765 audio_track_cblk_t *cblkOld = state->mCblk; 5766 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5767 if (cblkNew != cblkOld) { 5768 state->mCblk = cblkNew; 5769 // block until acked if removing a fast track 5770 if (cblkOld != NULL) { 5771 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5772 } 5773 didModify = true; 5774 } 5775 sq->end(didModify); 5776 if (didModify) { 5777 sq->push(block); 5778#if 0 5779 if (kUseFastCapture == FastCapture_Dynamic) { 5780 mNormalSource = mPipeSource; 5781 } 5782#endif 5783 } 5784 } 5785 5786 // now run the fast track destructor with thread mutex unlocked 5787 fastTrackToRemove.clear(); 5788 5789 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5790 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5791 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5792 // If destination is non-contiguous, first read past the nominal end of buffer, then 5793 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5794 5795 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5796 ssize_t framesRead; 5797 5798 // If an NBAIO source is present, use it to read the normal capture's data 5799 if (mPipeSource != 0) { 5800 size_t framesToRead = mBufferSize / mFrameSize; 5801 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5802 framesToRead, AudioBufferProvider::kInvalidPTS); 5803 if (framesRead == 0) { 5804 // since pipe is non-blocking, simulate blocking input 5805 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5806 } 5807 // otherwise use the HAL / AudioStreamIn directly 5808 } else { 5809 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5810 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5811 if (bytesRead < 0) { 5812 framesRead = bytesRead; 5813 } else { 5814 framesRead = bytesRead / mFrameSize; 5815 } 5816 } 5817 5818 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5819 ALOGE("read failed: framesRead=%d", framesRead); 5820 // Force input into standby so that it tries to recover at next read attempt 5821 inputStandBy(); 5822 sleepUs = kRecordThreadSleepUs; 5823 } 5824 if (framesRead <= 0) { 5825 goto unlock; 5826 } 5827 ALOG_ASSERT(framesRead > 0); 5828 5829 if (mTeeSink != 0) { 5830 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5831 } 5832 // If destination is non-contiguous, we now correct for reading past end of buffer. 5833 { 5834 size_t part1 = mRsmpInFramesP2 - rear; 5835 if ((size_t) framesRead > part1) { 5836 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5837 (framesRead - part1) * mFrameSize); 5838 } 5839 } 5840 rear = mRsmpInRear += framesRead; 5841 5842 size = activeTracks.size(); 5843 // loop over each active track 5844 for (size_t i = 0; i < size; i++) { 5845 activeTrack = activeTracks[i]; 5846 5847 // skip fast tracks, as those are handled directly by FastCapture 5848 if (activeTrack->isFastTrack()) { 5849 continue; 5850 } 5851 5852 // TODO: This code probably should be moved to RecordTrack. 5853 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5854 5855 enum { 5856 OVERRUN_UNKNOWN, 5857 OVERRUN_TRUE, 5858 OVERRUN_FALSE 5859 } overrun = OVERRUN_UNKNOWN; 5860 5861 // loop over getNextBuffer to handle circular sink 5862 for (;;) { 5863 5864 activeTrack->mSink.frameCount = ~0; 5865 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5866 size_t framesOut = activeTrack->mSink.frameCount; 5867 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5868 5869 // check available frames and handle overrun conditions 5870 // if the record track isn't draining fast enough. 5871 bool hasOverrun; 5872 size_t framesIn; 5873 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5874 if (hasOverrun) { 5875 overrun = OVERRUN_TRUE; 5876 } 5877 if (framesOut == 0 || framesIn == 0) { 5878 break; 5879 } 5880 5881 // Don't allow framesOut to be larger than what is possible with resampling 5882 // from framesIn. 5883 // This isn't strictly necessary but helps limit buffer resizing in 5884 // RecordBufferConverter. TODO: remove when no longer needed. 5885 framesOut = min(framesOut, 5886 destinationFramesPossible( 5887 framesIn, mSampleRate, activeTrack->mSampleRate)); 5888 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5889 framesOut = activeTrack->mRecordBufferConverter->convert( 5890 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5891 5892 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5893 overrun = OVERRUN_FALSE; 5894 } 5895 5896 if (activeTrack->mFramesToDrop == 0) { 5897 if (framesOut > 0) { 5898 activeTrack->mSink.frameCount = framesOut; 5899 activeTrack->releaseBuffer(&activeTrack->mSink); 5900 } 5901 } else { 5902 // FIXME could do a partial drop of framesOut 5903 if (activeTrack->mFramesToDrop > 0) { 5904 activeTrack->mFramesToDrop -= framesOut; 5905 if (activeTrack->mFramesToDrop <= 0) { 5906 activeTrack->clearSyncStartEvent(); 5907 } 5908 } else { 5909 activeTrack->mFramesToDrop += framesOut; 5910 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5911 activeTrack->mSyncStartEvent->isCancelled()) { 5912 ALOGW("Synced record %s, session %d, trigger session %d", 5913 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5914 activeTrack->sessionId(), 5915 (activeTrack->mSyncStartEvent != 0) ? 5916 activeTrack->mSyncStartEvent->triggerSession() : 0); 5917 activeTrack->clearSyncStartEvent(); 5918 } 5919 } 5920 } 5921 5922 if (framesOut == 0) { 5923 break; 5924 } 5925 } 5926 5927 switch (overrun) { 5928 case OVERRUN_TRUE: 5929 // client isn't retrieving buffers fast enough 5930 if (!activeTrack->setOverflow()) { 5931 nsecs_t now = systemTime(); 5932 // FIXME should lastWarning per track? 5933 if ((now - lastWarning) > kWarningThrottleNs) { 5934 ALOGW("RecordThread: buffer overflow"); 5935 lastWarning = now; 5936 } 5937 } 5938 break; 5939 case OVERRUN_FALSE: 5940 activeTrack->clearOverflow(); 5941 break; 5942 case OVERRUN_UNKNOWN: 5943 break; 5944 } 5945 5946 } 5947 5948unlock: 5949 // enable changes in effect chain 5950 unlockEffectChains(effectChains); 5951 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5952 } 5953 5954 standbyIfNotAlreadyInStandby(); 5955 5956 { 5957 Mutex::Autolock _l(mLock); 5958 for (size_t i = 0; i < mTracks.size(); i++) { 5959 sp<RecordTrack> track = mTracks[i]; 5960 track->invalidate(); 5961 } 5962 mActiveTracks.clear(); 5963 mActiveTracksGen++; 5964 mStartStopCond.broadcast(); 5965 } 5966 5967 releaseWakeLock(); 5968 5969 ALOGV("RecordThread %p exiting", this); 5970 return false; 5971} 5972 5973void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5974{ 5975 if (!mStandby) { 5976 inputStandBy(); 5977 mStandby = true; 5978 } 5979} 5980 5981void AudioFlinger::RecordThread::inputStandBy() 5982{ 5983 // Idle the fast capture if it's currently running 5984 if (mFastCapture != 0) { 5985 FastCaptureStateQueue *sq = mFastCapture->sq(); 5986 FastCaptureState *state = sq->begin(); 5987 if (!(state->mCommand & FastCaptureState::IDLE)) { 5988 state->mCommand = FastCaptureState::COLD_IDLE; 5989 state->mColdFutexAddr = &mFastCaptureFutex; 5990 state->mColdGen++; 5991 mFastCaptureFutex = 0; 5992 sq->end(); 5993 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5994 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5995#if 0 5996 if (kUseFastCapture == FastCapture_Dynamic) { 5997 // FIXME 5998 } 5999#endif 6000#ifdef AUDIO_WATCHDOG 6001 // FIXME 6002#endif 6003 } else { 6004 sq->end(false /*didModify*/); 6005 } 6006 } 6007 mInput->stream->common.standby(&mInput->stream->common); 6008} 6009 6010// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6011sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6012 const sp<AudioFlinger::Client>& client, 6013 uint32_t sampleRate, 6014 audio_format_t format, 6015 audio_channel_mask_t channelMask, 6016 size_t *pFrameCount, 6017 int sessionId, 6018 size_t *notificationFrames, 6019 int uid, 6020 IAudioFlinger::track_flags_t *flags, 6021 pid_t tid, 6022 status_t *status) 6023{ 6024 size_t frameCount = *pFrameCount; 6025 sp<RecordTrack> track; 6026 status_t lStatus; 6027 6028 // client expresses a preference for FAST, but we get the final say 6029 if (*flags & IAudioFlinger::TRACK_FAST) { 6030 if ( 6031 // we formerly checked for a callback handler (non-0 tid), 6032 // but that is no longer required for TRANSFER_OBTAIN mode 6033 // 6034 // frame count is not specified, or is exactly the pipe depth 6035 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6036 // PCM data 6037 audio_is_linear_pcm(format) && 6038 // native format 6039 (format == mFormat) && 6040 // native channel mask 6041 (channelMask == mChannelMask) && 6042 // native hardware sample rate 6043 (sampleRate == mSampleRate) && 6044 // record thread has an associated fast capture 6045 hasFastCapture() && 6046 // there are sufficient fast track slots available 6047 mFastTrackAvail 6048 ) { 6049 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6050 frameCount, mFrameCount); 6051 } else { 6052 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6053 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6054 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6055 frameCount, mFrameCount, mPipeFramesP2, 6056 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6057 hasFastCapture(), tid, mFastTrackAvail); 6058 *flags &= ~IAudioFlinger::TRACK_FAST; 6059 } 6060 } 6061 6062 // compute track buffer size in frames, and suggest the notification frame count 6063 if (*flags & IAudioFlinger::TRACK_FAST) { 6064 // fast track: frame count is exactly the pipe depth 6065 frameCount = mPipeFramesP2; 6066 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6067 *notificationFrames = mFrameCount; 6068 } else { 6069 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6070 // or 20 ms if there is a fast capture 6071 // TODO This could be a roundupRatio inline, and const 6072 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6073 * sampleRate + mSampleRate - 1) / mSampleRate; 6074 // minimum number of notification periods is at least kMinNotifications, 6075 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6076 static const size_t kMinNotifications = 3; 6077 static const uint32_t kMinMs = 30; 6078 // TODO This could be a roundupRatio inline 6079 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6080 // TODO This could be a roundupRatio inline 6081 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6082 maxNotificationFrames; 6083 const size_t minFrameCount = maxNotificationFrames * 6084 max(kMinNotifications, minNotificationsByMs); 6085 frameCount = max(frameCount, minFrameCount); 6086 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6087 *notificationFrames = maxNotificationFrames; 6088 } 6089 } 6090 *pFrameCount = frameCount; 6091 6092 lStatus = initCheck(); 6093 if (lStatus != NO_ERROR) { 6094 ALOGE("createRecordTrack_l() audio driver not initialized"); 6095 goto Exit; 6096 } 6097 6098 { // scope for mLock 6099 Mutex::Autolock _l(mLock); 6100 6101 track = new RecordTrack(this, client, sampleRate, 6102 format, channelMask, frameCount, NULL, sessionId, uid, 6103 *flags, TrackBase::TYPE_DEFAULT); 6104 6105 lStatus = track->initCheck(); 6106 if (lStatus != NO_ERROR) { 6107 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6108 // track must be cleared from the caller as the caller has the AF lock 6109 goto Exit; 6110 } 6111 mTracks.add(track); 6112 6113 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6114 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6115 mAudioFlinger->btNrecIsOff(); 6116 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6117 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6118 6119 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6120 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6121 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6122 // so ask activity manager to do this on our behalf 6123 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6124 } 6125 } 6126 6127 lStatus = NO_ERROR; 6128 6129Exit: 6130 *status = lStatus; 6131 return track; 6132} 6133 6134status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6135 AudioSystem::sync_event_t event, 6136 int triggerSession) 6137{ 6138 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6139 sp<ThreadBase> strongMe = this; 6140 status_t status = NO_ERROR; 6141 6142 if (event == AudioSystem::SYNC_EVENT_NONE) { 6143 recordTrack->clearSyncStartEvent(); 6144 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6145 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6146 triggerSession, 6147 recordTrack->sessionId(), 6148 syncStartEventCallback, 6149 recordTrack); 6150 // Sync event can be cancelled by the trigger session if the track is not in a 6151 // compatible state in which case we start record immediately 6152 if (recordTrack->mSyncStartEvent->isCancelled()) { 6153 recordTrack->clearSyncStartEvent(); 6154 } else { 6155 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6156 recordTrack->mFramesToDrop = - 6157 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6158 } 6159 } 6160 6161 { 6162 // This section is a rendezvous between binder thread executing start() and RecordThread 6163 AutoMutex lock(mLock); 6164 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6165 if (recordTrack->mState == TrackBase::PAUSING) { 6166 ALOGV("active record track PAUSING -> ACTIVE"); 6167 recordTrack->mState = TrackBase::ACTIVE; 6168 } else { 6169 ALOGV("active record track state %d", recordTrack->mState); 6170 } 6171 return status; 6172 } 6173 6174 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6175 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6176 // or using a separate command thread 6177 recordTrack->mState = TrackBase::STARTING_1; 6178 mActiveTracks.add(recordTrack); 6179 mActiveTracksGen++; 6180 status_t status = NO_ERROR; 6181 if (recordTrack->isExternalTrack()) { 6182 mLock.unlock(); 6183 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6184 mLock.lock(); 6185 // FIXME should verify that recordTrack is still in mActiveTracks 6186 if (status != NO_ERROR) { 6187 mActiveTracks.remove(recordTrack); 6188 mActiveTracksGen++; 6189 recordTrack->clearSyncStartEvent(); 6190 ALOGV("RecordThread::start error %d", status); 6191 return status; 6192 } 6193 } 6194 // Catch up with current buffer indices if thread is already running. 6195 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6196 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6197 // see previously buffered data before it called start(), but with greater risk of overrun. 6198 6199 recordTrack->mResamplerBufferProvider->reset(); 6200 // clear any converter state as new data will be discontinuous 6201 recordTrack->mRecordBufferConverter->reset(); 6202 recordTrack->mState = TrackBase::STARTING_2; 6203 // signal thread to start 6204 mWaitWorkCV.broadcast(); 6205 if (mActiveTracks.indexOf(recordTrack) < 0) { 6206 ALOGV("Record failed to start"); 6207 status = BAD_VALUE; 6208 goto startError; 6209 } 6210 return status; 6211 } 6212 6213startError: 6214 if (recordTrack->isExternalTrack()) { 6215 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6216 } 6217 recordTrack->clearSyncStartEvent(); 6218 // FIXME I wonder why we do not reset the state here? 6219 return status; 6220} 6221 6222void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6223{ 6224 sp<SyncEvent> strongEvent = event.promote(); 6225 6226 if (strongEvent != 0) { 6227 sp<RefBase> ptr = strongEvent->cookie().promote(); 6228 if (ptr != 0) { 6229 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6230 recordTrack->handleSyncStartEvent(strongEvent); 6231 } 6232 } 6233} 6234 6235bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6236 ALOGV("RecordThread::stop"); 6237 AutoMutex _l(mLock); 6238 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6239 return false; 6240 } 6241 // note that threadLoop may still be processing the track at this point [without lock] 6242 recordTrack->mState = TrackBase::PAUSING; 6243 // do not wait for mStartStopCond if exiting 6244 if (exitPending()) { 6245 return true; 6246 } 6247 // FIXME incorrect usage of wait: no explicit predicate or loop 6248 mStartStopCond.wait(mLock); 6249 // if we have been restarted, recordTrack is in mActiveTracks here 6250 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6251 ALOGV("Record stopped OK"); 6252 return true; 6253 } 6254 return false; 6255} 6256 6257bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6258{ 6259 return false; 6260} 6261 6262status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6263{ 6264#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6265 if (!isValidSyncEvent(event)) { 6266 return BAD_VALUE; 6267 } 6268 6269 int eventSession = event->triggerSession(); 6270 status_t ret = NAME_NOT_FOUND; 6271 6272 Mutex::Autolock _l(mLock); 6273 6274 for (size_t i = 0; i < mTracks.size(); i++) { 6275 sp<RecordTrack> track = mTracks[i]; 6276 if (eventSession == track->sessionId()) { 6277 (void) track->setSyncEvent(event); 6278 ret = NO_ERROR; 6279 } 6280 } 6281 return ret; 6282#else 6283 return BAD_VALUE; 6284#endif 6285} 6286 6287// destroyTrack_l() must be called with ThreadBase::mLock held 6288void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6289{ 6290 track->terminate(); 6291 track->mState = TrackBase::STOPPED; 6292 // active tracks are removed by threadLoop() 6293 if (mActiveTracks.indexOf(track) < 0) { 6294 removeTrack_l(track); 6295 } 6296} 6297 6298void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6299{ 6300 mTracks.remove(track); 6301 // need anything related to effects here? 6302 if (track->isFastTrack()) { 6303 ALOG_ASSERT(!mFastTrackAvail); 6304 mFastTrackAvail = true; 6305 } 6306} 6307 6308void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6309{ 6310 dumpInternals(fd, args); 6311 dumpTracks(fd, args); 6312 dumpEffectChains(fd, args); 6313} 6314 6315void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6316{ 6317 dprintf(fd, "\nInput thread %p:\n", this); 6318 6319 dumpBase(fd, args); 6320 6321 if (mActiveTracks.size() == 0) { 6322 dprintf(fd, " No active record clients\n"); 6323 } 6324 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6325 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6326 6327 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6328 const FastCaptureDumpState copy(mFastCaptureDumpState); 6329 copy.dump(fd); 6330} 6331 6332void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6333{ 6334 const size_t SIZE = 256; 6335 char buffer[SIZE]; 6336 String8 result; 6337 6338 size_t numtracks = mTracks.size(); 6339 size_t numactive = mActiveTracks.size(); 6340 size_t numactiveseen = 0; 6341 dprintf(fd, " %d Tracks", numtracks); 6342 if (numtracks) { 6343 dprintf(fd, " of which %d are active\n", numactive); 6344 RecordTrack::appendDumpHeader(result); 6345 for (size_t i = 0; i < numtracks ; ++i) { 6346 sp<RecordTrack> track = mTracks[i]; 6347 if (track != 0) { 6348 bool active = mActiveTracks.indexOf(track) >= 0; 6349 if (active) { 6350 numactiveseen++; 6351 } 6352 track->dump(buffer, SIZE, active); 6353 result.append(buffer); 6354 } 6355 } 6356 } else { 6357 dprintf(fd, "\n"); 6358 } 6359 6360 if (numactiveseen != numactive) { 6361 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6362 " not in the track list\n"); 6363 result.append(buffer); 6364 RecordTrack::appendDumpHeader(result); 6365 for (size_t i = 0; i < numactive; ++i) { 6366 sp<RecordTrack> track = mActiveTracks[i]; 6367 if (mTracks.indexOf(track) < 0) { 6368 track->dump(buffer, SIZE, true); 6369 result.append(buffer); 6370 } 6371 } 6372 6373 } 6374 write(fd, result.string(), result.size()); 6375} 6376 6377 6378void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6379{ 6380 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6381 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6382 mRsmpInFront = recordThread->mRsmpInRear; 6383 mRsmpInUnrel = 0; 6384} 6385 6386void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6387 size_t *framesAvailable, bool *hasOverrun) 6388{ 6389 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6390 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6391 const int32_t rear = recordThread->mRsmpInRear; 6392 const int32_t front = mRsmpInFront; 6393 const ssize_t filled = rear - front; 6394 6395 size_t framesIn; 6396 bool overrun = false; 6397 if (filled < 0) { 6398 // should not happen, but treat like a massive overrun and re-sync 6399 framesIn = 0; 6400 mRsmpInFront = rear; 6401 overrun = true; 6402 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6403 framesIn = (size_t) filled; 6404 } else { 6405 // client is not keeping up with server, but give it latest data 6406 framesIn = recordThread->mRsmpInFrames; 6407 mRsmpInFront = /* front = */ rear - framesIn; 6408 overrun = true; 6409 } 6410 if (framesAvailable != NULL) { 6411 *framesAvailable = framesIn; 6412 } 6413 if (hasOverrun != NULL) { 6414 *hasOverrun = overrun; 6415 } 6416} 6417 6418// AudioBufferProvider interface 6419status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6420 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6421{ 6422 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6423 if (threadBase == 0) { 6424 buffer->frameCount = 0; 6425 buffer->raw = NULL; 6426 return NOT_ENOUGH_DATA; 6427 } 6428 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6429 int32_t rear = recordThread->mRsmpInRear; 6430 int32_t front = mRsmpInFront; 6431 ssize_t filled = rear - front; 6432 // FIXME should not be P2 (don't want to increase latency) 6433 // FIXME if client not keeping up, discard 6434 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6435 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6436 front &= recordThread->mRsmpInFramesP2 - 1; 6437 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6438 if (part1 > (size_t) filled) { 6439 part1 = filled; 6440 } 6441 size_t ask = buffer->frameCount; 6442 ALOG_ASSERT(ask > 0); 6443 if (part1 > ask) { 6444 part1 = ask; 6445 } 6446 if (part1 == 0) { 6447 // out of data is fine since the resampler will return a short-count. 6448 buffer->raw = NULL; 6449 buffer->frameCount = 0; 6450 mRsmpInUnrel = 0; 6451 return NOT_ENOUGH_DATA; 6452 } 6453 6454 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6455 buffer->frameCount = part1; 6456 mRsmpInUnrel = part1; 6457 return NO_ERROR; 6458} 6459 6460// AudioBufferProvider interface 6461void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6462 AudioBufferProvider::Buffer* buffer) 6463{ 6464 size_t stepCount = buffer->frameCount; 6465 if (stepCount == 0) { 6466 return; 6467 } 6468 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6469 mRsmpInUnrel -= stepCount; 6470 mRsmpInFront += stepCount; 6471 buffer->raw = NULL; 6472 buffer->frameCount = 0; 6473} 6474 6475AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6476 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6477 uint32_t srcSampleRate, 6478 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6479 uint32_t dstSampleRate) : 6480 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6481 // mSrcFormat 6482 // mSrcSampleRate 6483 // mDstChannelMask 6484 // mDstFormat 6485 // mDstSampleRate 6486 // mSrcChannelCount 6487 // mDstChannelCount 6488 // mDstFrameSize 6489 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6490 mResampler(NULL), 6491 mIsLegacyDownmix(false), 6492 mIsLegacyUpmix(false), 6493 mRequiresFloat(false), 6494 mInputConverterProvider(NULL) 6495{ 6496 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6497 dstChannelMask, dstFormat, dstSampleRate); 6498} 6499 6500AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6501 free(mBuf); 6502 delete mResampler; 6503 delete mInputConverterProvider; 6504} 6505 6506size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6507 AudioBufferProvider *provider, size_t frames) 6508{ 6509 if (mInputConverterProvider != NULL) { 6510 mInputConverterProvider->setBufferProvider(provider); 6511 provider = mInputConverterProvider; 6512 } 6513 6514 if (mResampler == NULL) { 6515 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6516 mSrcSampleRate, mSrcFormat, mDstFormat); 6517 6518 AudioBufferProvider::Buffer buffer; 6519 for (size_t i = frames; i > 0; ) { 6520 buffer.frameCount = i; 6521 status_t status = provider->getNextBuffer(&buffer, 0); 6522 if (status != OK || buffer.frameCount == 0) { 6523 frames -= i; // cannot fill request. 6524 break; 6525 } 6526 // format convert to destination buffer 6527 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6528 6529 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6530 i -= buffer.frameCount; 6531 provider->releaseBuffer(&buffer); 6532 } 6533 } else { 6534 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6535 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6536 6537 // reallocate buffer if needed 6538 if (mBufFrameSize != 0 && mBufFrames < frames) { 6539 free(mBuf); 6540 mBufFrames = frames; 6541 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6542 } 6543 // resampler accumulates, but we only have one source track 6544 memset(mBuf, 0, frames * mBufFrameSize); 6545 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6546 // format convert to destination buffer 6547 convertResampler(dst, mBuf, frames); 6548 } 6549 return frames; 6550} 6551 6552status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6553 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6554 uint32_t srcSampleRate, 6555 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6556 uint32_t dstSampleRate) 6557{ 6558 // quick evaluation if there is any change. 6559 if (mSrcFormat == srcFormat 6560 && mSrcChannelMask == srcChannelMask 6561 && mSrcSampleRate == srcSampleRate 6562 && mDstFormat == dstFormat 6563 && mDstChannelMask == dstChannelMask 6564 && mDstSampleRate == dstSampleRate) { 6565 return NO_ERROR; 6566 } 6567 6568 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6569 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6570 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6571 const bool valid = 6572 audio_is_input_channel(srcChannelMask) 6573 && audio_is_input_channel(dstChannelMask) 6574 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6575 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6576 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6577 ; // no upsampling checks for now 6578 if (!valid) { 6579 return BAD_VALUE; 6580 } 6581 6582 mSrcFormat = srcFormat; 6583 mSrcChannelMask = srcChannelMask; 6584 mSrcSampleRate = srcSampleRate; 6585 mDstFormat = dstFormat; 6586 mDstChannelMask = dstChannelMask; 6587 mDstSampleRate = dstSampleRate; 6588 6589 // compute derived parameters 6590 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6591 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6592 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6593 6594 // do we need to resample? 6595 delete mResampler; 6596 mResampler = NULL; 6597 if (mSrcSampleRate != mDstSampleRate) { 6598 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6599 mSrcChannelCount, mDstSampleRate); 6600 mResampler->setSampleRate(mSrcSampleRate); 6601 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6602 } 6603 6604 // are we running legacy channel conversion modes? 6605 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6606 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6607 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6608 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6609 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6610 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6611 6612 // do we need to process in float? 6613 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6614 6615 // do we need a staging buffer to convert for destination (we can still optimize this)? 6616 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6617 if (mResampler != NULL) { 6618 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6619 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6620 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6621 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6622 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6623 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6624 } else { 6625 mBufFrameSize = 0; 6626 } 6627 mBufFrames = 0; // force the buffer to be resized. 6628 6629 // do we need an input converter buffer provider to give us float? 6630 delete mInputConverterProvider; 6631 mInputConverterProvider = NULL; 6632 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6633 mInputConverterProvider = new ReformatBufferProvider( 6634 audio_channel_count_from_in_mask(mSrcChannelMask), 6635 mSrcFormat, 6636 AUDIO_FORMAT_PCM_FLOAT, 6637 256 /* provider buffer frame count */); 6638 } 6639 6640 // do we need a remixer to do channel mask conversion 6641 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6642 (void) memcpy_by_index_array_initialization_from_channel_mask( 6643 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6644 } 6645 return NO_ERROR; 6646} 6647 6648void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6649 void *dst, const void *src, size_t frames) 6650{ 6651 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6652 if (mBufFrameSize != 0 && mBufFrames < frames) { 6653 free(mBuf); 6654 mBufFrames = frames; 6655 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6656 } 6657 // do we need to do legacy upmix and downmix? 6658 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6659 void *dstBuf = mBuf != NULL ? mBuf : dst; 6660 if (mIsLegacyUpmix) { 6661 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6662 (const float *)src, frames); 6663 } else /*mIsLegacyDownmix */ { 6664 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6665 (const float *)src, frames); 6666 } 6667 if (mBuf != NULL) { 6668 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6669 frames * mDstChannelCount); 6670 } 6671 return; 6672 } 6673 // do we need to do channel mask conversion? 6674 if (mSrcChannelMask != mDstChannelMask) { 6675 void *dstBuf = mBuf != NULL ? mBuf : dst; 6676 memcpy_by_index_array(dstBuf, mDstChannelCount, 6677 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6678 if (dstBuf == dst) { 6679 return; // format is the same 6680 } 6681 } 6682 // convert to destination buffer 6683 const void *convertBuf = mBuf != NULL ? mBuf : src; 6684 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6685 frames * mDstChannelCount); 6686} 6687 6688void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6689 void *dst, /*not-a-const*/ void *src, size_t frames) 6690{ 6691 // src buffer format is ALWAYS float when entering this routine 6692 if (mIsLegacyUpmix) { 6693 ; // mono to stereo already handled by resampler 6694 } else if (mIsLegacyDownmix 6695 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6696 // the resampler outputs stereo for mono input channel (a feature?) 6697 // must convert to mono 6698 downmix_to_mono_float_from_stereo_float((float *)src, 6699 (const float *)src, frames); 6700 } else if (mSrcChannelMask != mDstChannelMask) { 6701 // convert to mono channel again for channel mask conversion (could be skipped 6702 // with further optimization). 6703 if (mSrcChannelCount == 1) { 6704 downmix_to_mono_float_from_stereo_float((float *)src, 6705 (const float *)src, frames); 6706 } 6707 // convert to destination format (in place, OK as float is larger than other types) 6708 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6709 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6710 frames * mSrcChannelCount); 6711 } 6712 // channel convert and save to dst 6713 memcpy_by_index_array(dst, mDstChannelCount, 6714 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6715 return; 6716 } 6717 // convert to destination format and save to dst 6718 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6719 frames * mDstChannelCount); 6720} 6721 6722bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6723 status_t& status) 6724{ 6725 bool reconfig = false; 6726 6727 status = NO_ERROR; 6728 6729 audio_format_t reqFormat = mFormat; 6730 uint32_t samplingRate = mSampleRate; 6731 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6732 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6733 6734 AudioParameter param = AudioParameter(keyValuePair); 6735 int value; 6736 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6737 // channel count change can be requested. Do we mandate the first client defines the 6738 // HAL sampling rate and channel count or do we allow changes on the fly? 6739 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6740 samplingRate = value; 6741 reconfig = true; 6742 } 6743 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6744 if (!audio_is_linear_pcm((audio_format_t) value)) { 6745 status = BAD_VALUE; 6746 } else { 6747 reqFormat = (audio_format_t) value; 6748 reconfig = true; 6749 } 6750 } 6751 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6752 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6753 if (!audio_is_input_channel(mask) || 6754 audio_channel_count_from_in_mask(mask) > FCC_8) { 6755 status = BAD_VALUE; 6756 } else { 6757 channelMask = mask; 6758 reconfig = true; 6759 } 6760 } 6761 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6762 // do not accept frame count changes if tracks are open as the track buffer 6763 // size depends on frame count and correct behavior would not be guaranteed 6764 // if frame count is changed after track creation 6765 if (mActiveTracks.size() > 0) { 6766 status = INVALID_OPERATION; 6767 } else { 6768 reconfig = true; 6769 } 6770 } 6771 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6772 // forward device change to effects that have requested to be 6773 // aware of attached audio device. 6774 for (size_t i = 0; i < mEffectChains.size(); i++) { 6775 mEffectChains[i]->setDevice_l(value); 6776 } 6777 6778 // store input device and output device but do not forward output device to audio HAL. 6779 // Note that status is ignored by the caller for output device 6780 // (see AudioFlinger::setParameters() 6781 if (audio_is_output_devices(value)) { 6782 mOutDevice = value; 6783 status = BAD_VALUE; 6784 } else { 6785 mInDevice = value; 6786 // disable AEC and NS if the device is a BT SCO headset supporting those 6787 // pre processings 6788 if (mTracks.size() > 0) { 6789 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6790 mAudioFlinger->btNrecIsOff(); 6791 for (size_t i = 0; i < mTracks.size(); i++) { 6792 sp<RecordTrack> track = mTracks[i]; 6793 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6794 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6795 } 6796 } 6797 } 6798 } 6799 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6800 mAudioSource != (audio_source_t)value) { 6801 // forward device change to effects that have requested to be 6802 // aware of attached audio device. 6803 for (size_t i = 0; i < mEffectChains.size(); i++) { 6804 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6805 } 6806 mAudioSource = (audio_source_t)value; 6807 } 6808 6809 if (status == NO_ERROR) { 6810 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6811 keyValuePair.string()); 6812 if (status == INVALID_OPERATION) { 6813 inputStandBy(); 6814 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6815 keyValuePair.string()); 6816 } 6817 if (reconfig) { 6818 if (status == BAD_VALUE && 6819 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6820 audio_is_linear_pcm(reqFormat) && 6821 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6822 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6823 audio_channel_count_from_in_mask( 6824 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6825 status = NO_ERROR; 6826 } 6827 if (status == NO_ERROR) { 6828 readInputParameters_l(); 6829 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6830 } 6831 } 6832 } 6833 6834 return reconfig; 6835} 6836 6837String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6838{ 6839 Mutex::Autolock _l(mLock); 6840 if (initCheck() != NO_ERROR) { 6841 return String8(); 6842 } 6843 6844 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6845 const String8 out_s8(s); 6846 free(s); 6847 return out_s8; 6848} 6849 6850void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event) { 6851 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6852 6853 desc->mIoHandle = mId; 6854 6855 switch (event) { 6856 case AUDIO_INPUT_OPENED: 6857 case AUDIO_INPUT_CONFIG_CHANGED: 6858 desc->mPatch = mPatch; 6859 desc->mChannelMask = mChannelMask; 6860 desc->mSamplingRate = mSampleRate; 6861 desc->mFormat = mFormat; 6862 desc->mFrameCount = mFrameCount; 6863 desc->mLatency = 0; 6864 break; 6865 6866 case AUDIO_INPUT_CLOSED: 6867 default: 6868 break; 6869 } 6870 mAudioFlinger->ioConfigChanged(event, desc); 6871} 6872 6873void AudioFlinger::RecordThread::readInputParameters_l() 6874{ 6875 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6876 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6877 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6878 if (mChannelCount > FCC_8) { 6879 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6880 } 6881 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6882 mFormat = mHALFormat; 6883 if (!audio_is_linear_pcm(mFormat)) { 6884 ALOGE("HAL format %#x is not linear pcm", mFormat); 6885 } 6886 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6887 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6888 mFrameCount = mBufferSize / mFrameSize; 6889 // This is the formula for calculating the temporary buffer size. 6890 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6891 // 1 full output buffer, regardless of the alignment of the available input. 6892 // The value is somewhat arbitrary, and could probably be even larger. 6893 // A larger value should allow more old data to be read after a track calls start(), 6894 // without increasing latency. 6895 // 6896 // Note this is independent of the maximum downsampling ratio permitted for capture. 6897 mRsmpInFrames = mFrameCount * 7; 6898 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6899 free(mRsmpInBuffer); 6900 6901 // TODO optimize audio capture buffer sizes ... 6902 // Here we calculate the size of the sliding buffer used as a source 6903 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6904 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6905 // be better to have it derived from the pipe depth in the long term. 6906 // The current value is higher than necessary. However it should not add to latency. 6907 6908 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6909 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6910 6911 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6912 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6913} 6914 6915uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6916{ 6917 Mutex::Autolock _l(mLock); 6918 if (initCheck() != NO_ERROR) { 6919 return 0; 6920 } 6921 6922 return mInput->stream->get_input_frames_lost(mInput->stream); 6923} 6924 6925uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6926{ 6927 Mutex::Autolock _l(mLock); 6928 uint32_t result = 0; 6929 if (getEffectChain_l(sessionId) != 0) { 6930 result = EFFECT_SESSION; 6931 } 6932 6933 for (size_t i = 0; i < mTracks.size(); ++i) { 6934 if (sessionId == mTracks[i]->sessionId()) { 6935 result |= TRACK_SESSION; 6936 break; 6937 } 6938 } 6939 6940 return result; 6941} 6942 6943KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6944{ 6945 KeyedVector<int, bool> ids; 6946 Mutex::Autolock _l(mLock); 6947 for (size_t j = 0; j < mTracks.size(); ++j) { 6948 sp<RecordThread::RecordTrack> track = mTracks[j]; 6949 int sessionId = track->sessionId(); 6950 if (ids.indexOfKey(sessionId) < 0) { 6951 ids.add(sessionId, true); 6952 } 6953 } 6954 return ids; 6955} 6956 6957AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6958{ 6959 Mutex::Autolock _l(mLock); 6960 AudioStreamIn *input = mInput; 6961 mInput = NULL; 6962 return input; 6963} 6964 6965// this method must always be called either with ThreadBase mLock held or inside the thread loop 6966audio_stream_t* AudioFlinger::RecordThread::stream() const 6967{ 6968 if (mInput == NULL) { 6969 return NULL; 6970 } 6971 return &mInput->stream->common; 6972} 6973 6974status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6975{ 6976 // only one chain per input thread 6977 if (mEffectChains.size() != 0) { 6978 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6979 return INVALID_OPERATION; 6980 } 6981 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6982 chain->setThread(this); 6983 chain->setInBuffer(NULL); 6984 chain->setOutBuffer(NULL); 6985 6986 checkSuspendOnAddEffectChain_l(chain); 6987 6988 // make sure enabled pre processing effects state is communicated to the HAL as we 6989 // just moved them to a new input stream. 6990 chain->syncHalEffectsState(); 6991 6992 mEffectChains.add(chain); 6993 6994 return NO_ERROR; 6995} 6996 6997size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6998{ 6999 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7000 ALOGW_IF(mEffectChains.size() != 1, 7001 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7002 chain.get(), mEffectChains.size(), this); 7003 if (mEffectChains.size() == 1) { 7004 mEffectChains.removeAt(0); 7005 } 7006 return 0; 7007} 7008 7009status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7010 audio_patch_handle_t *handle) 7011{ 7012 status_t status = NO_ERROR; 7013 7014 // store new device and send to effects 7015 mInDevice = patch->sources[0].ext.device.type; 7016 mPatch = *patch; 7017 for (size_t i = 0; i < mEffectChains.size(); i++) { 7018 mEffectChains[i]->setDevice_l(mInDevice); 7019 } 7020 7021 // disable AEC and NS if the device is a BT SCO headset supporting those 7022 // pre processings 7023 if (mTracks.size() > 0) { 7024 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7025 mAudioFlinger->btNrecIsOff(); 7026 for (size_t i = 0; i < mTracks.size(); i++) { 7027 sp<RecordTrack> track = mTracks[i]; 7028 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7029 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7030 } 7031 } 7032 7033 // store new source and send to effects 7034 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7035 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7036 for (size_t i = 0; i < mEffectChains.size(); i++) { 7037 mEffectChains[i]->setAudioSource_l(mAudioSource); 7038 } 7039 } 7040 7041 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7042 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7043 status = hwDevice->create_audio_patch(hwDevice, 7044 patch->num_sources, 7045 patch->sources, 7046 patch->num_sinks, 7047 patch->sinks, 7048 handle); 7049 } else { 7050 char *address; 7051 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7052 address = audio_device_address_to_parameter( 7053 patch->sources[0].ext.device.type, 7054 patch->sources[0].ext.device.address); 7055 } else { 7056 address = (char *)calloc(1, 1); 7057 } 7058 AudioParameter param = AudioParameter(String8(address)); 7059 free(address); 7060 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7061 (int)patch->sources[0].ext.device.type); 7062 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7063 (int)patch->sinks[0].ext.mix.usecase.source); 7064 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7065 param.toString().string()); 7066 *handle = AUDIO_PATCH_HANDLE_NONE; 7067 } 7068 7069 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7070 7071 return status; 7072} 7073 7074status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7075{ 7076 status_t status = NO_ERROR; 7077 7078 mInDevice = AUDIO_DEVICE_NONE; 7079 7080 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7081 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7082 status = hwDevice->release_audio_patch(hwDevice, handle); 7083 } else { 7084 AudioParameter param; 7085 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7086 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7087 param.toString().string()); 7088 } 7089 return status; 7090} 7091 7092void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7093{ 7094 Mutex::Autolock _l(mLock); 7095 mTracks.add(record); 7096} 7097 7098void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7099{ 7100 Mutex::Autolock _l(mLock); 7101 destroyTrack_l(record); 7102} 7103 7104void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7105{ 7106 ThreadBase::getAudioPortConfig(config); 7107 config->role = AUDIO_PORT_ROLE_SINK; 7108 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7109 config->ext.mix.usecase.source = mAudioSource; 7110} 7111 7112} // namespace android 7113