Threads.cpp revision 0f0631eb55b1f0a7f4b62212b78a3faa0b49919b
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_OUT_IP, "IP", 379 AUDIO_DEVICE_NONE, "NONE", // must be last 380 }, mappingsIn[] = { 381 AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION", 382 AUDIO_DEVICE_IN_AMBIENT, "AMBIENT", 383 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 384 AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 385 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 386 AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL", 387 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 388 AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX", 389 AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC", 390 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 391 AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 392 AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 393 AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY", 394 AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE", 395 AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER", 396 AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER", 397 AUDIO_DEVICE_IN_LINE, "LINE", 398 AUDIO_DEVICE_IN_SPDIF, "SPDIF", 399 AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 400 AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK", 401 AUDIO_DEVICE_IN_IP, "IP", 402 AUDIO_DEVICE_NONE, "NONE", // must be last 403 }; 404 String8 result; 405 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 406 const mapping *entry; 407 if (devices & AUDIO_DEVICE_BIT_IN) { 408 devices &= ~AUDIO_DEVICE_BIT_IN; 409 entry = mappingsIn; 410 } else { 411 entry = mappingsOut; 412 } 413 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 414 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 415 if (devices & entry->mDevices) { 416 if (!result.isEmpty()) { 417 result.append("|"); 418 } 419 result.append(entry->mString); 420 } 421 } 422 if (devices & ~allDevices) { 423 if (!result.isEmpty()) { 424 result.append("|"); 425 } 426 result.appendFormat("0x%X", devices & ~allDevices); 427 } 428 if (result.isEmpty()) { 429 result.append(entry->mString); 430 } 431 return result; 432} 433 434String8 inputFlagsToString(audio_input_flags_t flags) 435{ 436 static const struct mapping { 437 audio_input_flags_t mFlag; 438 const char * mString; 439 } mappings[] = { 440 AUDIO_INPUT_FLAG_FAST, "FAST", 441 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 442 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 443 }; 444 String8 result; 445 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 446 const mapping *entry; 447 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 448 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 449 if (flags & entry->mFlag) { 450 if (!result.isEmpty()) { 451 result.append("|"); 452 } 453 result.append(entry->mString); 454 } 455 } 456 if (flags & ~allFlags) { 457 if (!result.isEmpty()) { 458 result.append("|"); 459 } 460 result.appendFormat("0x%X", flags & ~allFlags); 461 } 462 if (result.isEmpty()) { 463 result.append(entry->mString); 464 } 465 return result; 466} 467 468String8 outputFlagsToString(audio_output_flags_t flags) 469{ 470 static const struct mapping { 471 audio_output_flags_t mFlag; 472 const char * mString; 473 } mappings[] = { 474 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 475 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 476 AUDIO_OUTPUT_FLAG_FAST, "FAST", 477 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 478 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 479 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 480 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 481 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 482 }; 483 String8 result; 484 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 485 const mapping *entry; 486 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 487 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 488 if (flags & entry->mFlag) { 489 if (!result.isEmpty()) { 490 result.append("|"); 491 } 492 result.append(entry->mString); 493 } 494 } 495 if (flags & ~allFlags) { 496 if (!result.isEmpty()) { 497 result.append("|"); 498 } 499 result.appendFormat("0x%X", flags & ~allFlags); 500 } 501 if (result.isEmpty()) { 502 result.append(entry->mString); 503 } 504 return result; 505} 506 507const char *sourceToString(audio_source_t source) 508{ 509 switch (source) { 510 case AUDIO_SOURCE_DEFAULT: return "default"; 511 case AUDIO_SOURCE_MIC: return "mic"; 512 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 513 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 514 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 515 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 516 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 517 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 518 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 519 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 520 case AUDIO_SOURCE_HOTWORD: return "hotword"; 521 default: return "unknown"; 522 } 523} 524 525AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 526 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 527 : Thread(false /*canCallJava*/), 528 mType(type), 529 mAudioFlinger(audioFlinger), 530 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 531 // are set by PlaybackThread::readOutputParameters_l() or 532 // RecordThread::readInputParameters_l() 533 //FIXME: mStandby should be true here. Is this some kind of hack? 534 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 535 mPrevInDevice(AUDIO_DEVICE_NONE), mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 536 // mName will be set by concrete (non-virtual) subclass 537 mDeathRecipient(new PMDeathRecipient(this)), 538 mSystemReady(systemReady) 539{ 540 memset(&mPatch, 0, sizeof(struct audio_patch)); 541} 542 543AudioFlinger::ThreadBase::~ThreadBase() 544{ 545 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 546 mConfigEvents.clear(); 547 548 // do not lock the mutex in destructor 549 releaseWakeLock_l(); 550 if (mPowerManager != 0) { 551 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 552 binder->unlinkToDeath(mDeathRecipient); 553 } 554} 555 556status_t AudioFlinger::ThreadBase::readyToRun() 557{ 558 status_t status = initCheck(); 559 if (status == NO_ERROR) { 560 ALOGI("AudioFlinger's thread %p ready to run", this); 561 } else { 562 ALOGE("No working audio driver found."); 563 } 564 return status; 565} 566 567void AudioFlinger::ThreadBase::exit() 568{ 569 ALOGV("ThreadBase::exit"); 570 // do any cleanup required for exit to succeed 571 preExit(); 572 { 573 // This lock prevents the following race in thread (uniprocessor for illustration): 574 // if (!exitPending()) { 575 // // context switch from here to exit() 576 // // exit() calls requestExit(), what exitPending() observes 577 // // exit() calls signal(), which is dropped since no waiters 578 // // context switch back from exit() to here 579 // mWaitWorkCV.wait(...); 580 // // now thread is hung 581 // } 582 AutoMutex lock(mLock); 583 requestExit(); 584 mWaitWorkCV.broadcast(); 585 } 586 // When Thread::requestExitAndWait is made virtual and this method is renamed to 587 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 588 requestExitAndWait(); 589} 590 591status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 592{ 593 status_t status; 594 595 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 596 Mutex::Autolock _l(mLock); 597 598 return sendSetParameterConfigEvent_l(keyValuePairs); 599} 600 601// sendConfigEvent_l() must be called with ThreadBase::mLock held 602// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 603status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 604{ 605 status_t status = NO_ERROR; 606 607 if (event->mRequiresSystemReady && !mSystemReady) { 608 event->mWaitStatus = false; 609 mPendingConfigEvents.add(event); 610 return status; 611 } 612 mConfigEvents.add(event); 613 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 614 mWaitWorkCV.signal(); 615 mLock.unlock(); 616 { 617 Mutex::Autolock _l(event->mLock); 618 while (event->mWaitStatus) { 619 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 620 event->mStatus = TIMED_OUT; 621 event->mWaitStatus = false; 622 } 623 } 624 status = event->mStatus; 625 } 626 mLock.lock(); 627 return status; 628} 629 630void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event) 631{ 632 Mutex::Autolock _l(mLock); 633 sendIoConfigEvent_l(event); 634} 635 636// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 637void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event) 638{ 639 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event); 640 sendConfigEvent_l(configEvent); 641} 642 643void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 644{ 645 Mutex::Autolock _l(mLock); 646 sendPrioConfigEvent_l(pid, tid, prio); 647} 648 649// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 650void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 651{ 652 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 653 sendConfigEvent_l(configEvent); 654} 655 656// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 657status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 658{ 659 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 660 return sendConfigEvent_l(configEvent); 661} 662 663status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 664 const struct audio_patch *patch, 665 audio_patch_handle_t *handle) 666{ 667 Mutex::Autolock _l(mLock); 668 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 669 status_t status = sendConfigEvent_l(configEvent); 670 if (status == NO_ERROR) { 671 CreateAudioPatchConfigEventData *data = 672 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 673 *handle = data->mHandle; 674 } 675 return status; 676} 677 678status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 679 const audio_patch_handle_t handle) 680{ 681 Mutex::Autolock _l(mLock); 682 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 683 return sendConfigEvent_l(configEvent); 684} 685 686 687// post condition: mConfigEvents.isEmpty() 688void AudioFlinger::ThreadBase::processConfigEvents_l() 689{ 690 bool configChanged = false; 691 692 while (!mConfigEvents.isEmpty()) { 693 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 694 sp<ConfigEvent> event = mConfigEvents[0]; 695 mConfigEvents.removeAt(0); 696 switch (event->mType) { 697 case CFG_EVENT_PRIO: { 698 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 699 // FIXME Need to understand why this has to be done asynchronously 700 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 701 true /*asynchronous*/); 702 if (err != 0) { 703 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 704 data->mPrio, data->mPid, data->mTid, err); 705 } 706 } break; 707 case CFG_EVENT_IO: { 708 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 709 ioConfigChanged(data->mEvent); 710 } break; 711 case CFG_EVENT_SET_PARAMETER: { 712 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 713 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 714 configChanged = true; 715 } 716 } break; 717 case CFG_EVENT_CREATE_AUDIO_PATCH: { 718 CreateAudioPatchConfigEventData *data = 719 (CreateAudioPatchConfigEventData *)event->mData.get(); 720 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 721 } break; 722 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 723 ReleaseAudioPatchConfigEventData *data = 724 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 725 event->mStatus = releaseAudioPatch_l(data->mHandle); 726 } break; 727 default: 728 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 729 break; 730 } 731 { 732 Mutex::Autolock _l(event->mLock); 733 if (event->mWaitStatus) { 734 event->mWaitStatus = false; 735 event->mCond.signal(); 736 } 737 } 738 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 739 } 740 741 if (configChanged) { 742 cacheParameters_l(); 743 } 744} 745 746String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 747 String8 s; 748 const audio_channel_representation_t representation = 749 audio_channel_mask_get_representation(mask); 750 751 switch (representation) { 752 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 753 if (output) { 754 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 755 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 756 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 757 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 758 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 759 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 760 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 761 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 762 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 763 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 764 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 765 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 766 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 767 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 768 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 769 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 770 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 771 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 772 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 773 } else { 774 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 775 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 776 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 777 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 778 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 779 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 780 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 781 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 782 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 783 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 784 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 785 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 786 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 787 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 788 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 789 } 790 const int len = s.length(); 791 if (len > 2) { 792 char *str = s.lockBuffer(len); // needed? 793 s.unlockBuffer(len - 2); // remove trailing ", " 794 } 795 return s; 796 } 797 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 798 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 799 return s; 800 default: 801 s.appendFormat("unknown mask, representation:%d bits:%#x", 802 representation, audio_channel_mask_get_bits(mask)); 803 return s; 804 } 805} 806 807void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 808{ 809 const size_t SIZE = 256; 810 char buffer[SIZE]; 811 String8 result; 812 813 bool locked = AudioFlinger::dumpTryLock(mLock); 814 if (!locked) { 815 dprintf(fd, "thread %p may be deadlocked\n", this); 816 } 817 818 dprintf(fd, " Thread name: %s\n", mThreadName); 819 dprintf(fd, " I/O handle: %d\n", mId); 820 dprintf(fd, " TID: %d\n", getTid()); 821 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 822 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 823 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 824 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 825 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 826 dprintf(fd, " Channel count: %u\n", mChannelCount); 827 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 828 channelMaskToString(mChannelMask, mType != RECORD).string()); 829 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 830 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 831 dprintf(fd, " Pending config events:"); 832 size_t numConfig = mConfigEvents.size(); 833 if (numConfig) { 834 for (size_t i = 0; i < numConfig; i++) { 835 mConfigEvents[i]->dump(buffer, SIZE); 836 dprintf(fd, "\n %s", buffer); 837 } 838 dprintf(fd, "\n"); 839 } else { 840 dprintf(fd, " none\n"); 841 } 842 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 843 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 844 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 845 846 if (locked) { 847 mLock.unlock(); 848 } 849} 850 851void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 852{ 853 const size_t SIZE = 256; 854 char buffer[SIZE]; 855 String8 result; 856 857 size_t numEffectChains = mEffectChains.size(); 858 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 859 write(fd, buffer, strlen(buffer)); 860 861 for (size_t i = 0; i < numEffectChains; ++i) { 862 sp<EffectChain> chain = mEffectChains[i]; 863 if (chain != 0) { 864 chain->dump(fd, args); 865 } 866 } 867} 868 869void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 870{ 871 Mutex::Autolock _l(mLock); 872 acquireWakeLock_l(uid); 873} 874 875String16 AudioFlinger::ThreadBase::getWakeLockTag() 876{ 877 switch (mType) { 878 case MIXER: 879 return String16("AudioMix"); 880 case DIRECT: 881 return String16("AudioDirectOut"); 882 case DUPLICATING: 883 return String16("AudioDup"); 884 case RECORD: 885 return String16("AudioIn"); 886 case OFFLOAD: 887 return String16("AudioOffload"); 888 default: 889 ALOG_ASSERT(false); 890 return String16("AudioUnknown"); 891 } 892} 893 894void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 895{ 896 getPowerManager_l(); 897 if (mPowerManager != 0) { 898 sp<IBinder> binder = new BBinder(); 899 status_t status; 900 if (uid >= 0) { 901 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 902 binder, 903 getWakeLockTag(), 904 String16("media"), 905 uid, 906 true /* FIXME force oneway contrary to .aidl */); 907 } else { 908 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 909 binder, 910 getWakeLockTag(), 911 String16("media"), 912 true /* FIXME force oneway contrary to .aidl */); 913 } 914 if (status == NO_ERROR) { 915 mWakeLockToken = binder; 916 } 917 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 918 } 919} 920 921void AudioFlinger::ThreadBase::releaseWakeLock() 922{ 923 Mutex::Autolock _l(mLock); 924 releaseWakeLock_l(); 925} 926 927void AudioFlinger::ThreadBase::releaseWakeLock_l() 928{ 929 if (mWakeLockToken != 0) { 930 ALOGV("releaseWakeLock_l() %s", mThreadName); 931 if (mPowerManager != 0) { 932 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 933 true /* FIXME force oneway contrary to .aidl */); 934 } 935 mWakeLockToken.clear(); 936 } 937} 938 939void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 940 Mutex::Autolock _l(mLock); 941 updateWakeLockUids_l(uids); 942} 943 944void AudioFlinger::ThreadBase::getPowerManager_l() { 945 if (mSystemReady && mPowerManager == 0) { 946 // use checkService() to avoid blocking if power service is not up yet 947 sp<IBinder> binder = 948 defaultServiceManager()->checkService(String16("power")); 949 if (binder == 0) { 950 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 951 } else { 952 mPowerManager = interface_cast<IPowerManager>(binder); 953 binder->linkToDeath(mDeathRecipient); 954 } 955 } 956} 957 958void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 959 getPowerManager_l(); 960 if (mWakeLockToken == NULL) { 961 ALOGE("no wake lock to update!"); 962 return; 963 } 964 if (mPowerManager != 0) { 965 sp<IBinder> binder = new BBinder(); 966 status_t status; 967 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 968 true /* FIXME force oneway contrary to .aidl */); 969 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 970 } 971} 972 973void AudioFlinger::ThreadBase::clearPowerManager() 974{ 975 Mutex::Autolock _l(mLock); 976 releaseWakeLock_l(); 977 mPowerManager.clear(); 978} 979 980void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 981{ 982 sp<ThreadBase> thread = mThread.promote(); 983 if (thread != 0) { 984 thread->clearPowerManager(); 985 } 986 ALOGW("power manager service died !!!"); 987} 988 989void AudioFlinger::ThreadBase::setEffectSuspended( 990 const effect_uuid_t *type, bool suspend, int sessionId) 991{ 992 Mutex::Autolock _l(mLock); 993 setEffectSuspended_l(type, suspend, sessionId); 994} 995 996void AudioFlinger::ThreadBase::setEffectSuspended_l( 997 const effect_uuid_t *type, bool suspend, int sessionId) 998{ 999 sp<EffectChain> chain = getEffectChain_l(sessionId); 1000 if (chain != 0) { 1001 if (type != NULL) { 1002 chain->setEffectSuspended_l(type, suspend); 1003 } else { 1004 chain->setEffectSuspendedAll_l(suspend); 1005 } 1006 } 1007 1008 updateSuspendedSessions_l(type, suspend, sessionId); 1009} 1010 1011void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1012{ 1013 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1014 if (index < 0) { 1015 return; 1016 } 1017 1018 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1019 mSuspendedSessions.valueAt(index); 1020 1021 for (size_t i = 0; i < sessionEffects.size(); i++) { 1022 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1023 for (int j = 0; j < desc->mRefCount; j++) { 1024 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1025 chain->setEffectSuspendedAll_l(true); 1026 } else { 1027 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1028 desc->mType.timeLow); 1029 chain->setEffectSuspended_l(&desc->mType, true); 1030 } 1031 } 1032 } 1033} 1034 1035void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1036 bool suspend, 1037 int sessionId) 1038{ 1039 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1040 1041 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1042 1043 if (suspend) { 1044 if (index >= 0) { 1045 sessionEffects = mSuspendedSessions.valueAt(index); 1046 } else { 1047 mSuspendedSessions.add(sessionId, sessionEffects); 1048 } 1049 } else { 1050 if (index < 0) { 1051 return; 1052 } 1053 sessionEffects = mSuspendedSessions.valueAt(index); 1054 } 1055 1056 1057 int key = EffectChain::kKeyForSuspendAll; 1058 if (type != NULL) { 1059 key = type->timeLow; 1060 } 1061 index = sessionEffects.indexOfKey(key); 1062 1063 sp<SuspendedSessionDesc> desc; 1064 if (suspend) { 1065 if (index >= 0) { 1066 desc = sessionEffects.valueAt(index); 1067 } else { 1068 desc = new SuspendedSessionDesc(); 1069 if (type != NULL) { 1070 desc->mType = *type; 1071 } 1072 sessionEffects.add(key, desc); 1073 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1074 } 1075 desc->mRefCount++; 1076 } else { 1077 if (index < 0) { 1078 return; 1079 } 1080 desc = sessionEffects.valueAt(index); 1081 if (--desc->mRefCount == 0) { 1082 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1083 sessionEffects.removeItemsAt(index); 1084 if (sessionEffects.isEmpty()) { 1085 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1086 sessionId); 1087 mSuspendedSessions.removeItem(sessionId); 1088 } 1089 } 1090 } 1091 if (!sessionEffects.isEmpty()) { 1092 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1093 } 1094} 1095 1096void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1097 bool enabled, 1098 int sessionId) 1099{ 1100 Mutex::Autolock _l(mLock); 1101 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1102} 1103 1104void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1105 bool enabled, 1106 int sessionId) 1107{ 1108 if (mType != RECORD) { 1109 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1110 // another session. This gives the priority to well behaved effect control panels 1111 // and applications not using global effects. 1112 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1113 // global effects 1114 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1115 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1116 } 1117 } 1118 1119 sp<EffectChain> chain = getEffectChain_l(sessionId); 1120 if (chain != 0) { 1121 chain->checkSuspendOnEffectEnabled(effect, enabled); 1122 } 1123} 1124 1125// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1126sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1127 const sp<AudioFlinger::Client>& client, 1128 const sp<IEffectClient>& effectClient, 1129 int32_t priority, 1130 int sessionId, 1131 effect_descriptor_t *desc, 1132 int *enabled, 1133 status_t *status) 1134{ 1135 sp<EffectModule> effect; 1136 sp<EffectHandle> handle; 1137 status_t lStatus; 1138 sp<EffectChain> chain; 1139 bool chainCreated = false; 1140 bool effectCreated = false; 1141 bool effectRegistered = false; 1142 1143 lStatus = initCheck(); 1144 if (lStatus != NO_ERROR) { 1145 ALOGW("createEffect_l() Audio driver not initialized."); 1146 goto Exit; 1147 } 1148 1149 // Reject any effect on Direct output threads for now, since the format of 1150 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1151 if (mType == DIRECT) { 1152 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1153 desc->name, mThreadName); 1154 lStatus = BAD_VALUE; 1155 goto Exit; 1156 } 1157 1158 // Reject any effect on mixer or duplicating multichannel sinks. 1159 // TODO: fix both format and multichannel issues with effects. 1160 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1161 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1162 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1163 lStatus = BAD_VALUE; 1164 goto Exit; 1165 } 1166 1167 // Allow global effects only on offloaded and mixer threads 1168 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1169 switch (mType) { 1170 case MIXER: 1171 case OFFLOAD: 1172 break; 1173 case DIRECT: 1174 case DUPLICATING: 1175 case RECORD: 1176 default: 1177 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1178 desc->name, mThreadName); 1179 lStatus = BAD_VALUE; 1180 goto Exit; 1181 } 1182 } 1183 1184 // Only Pre processor effects are allowed on input threads and only on input threads 1185 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1186 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1187 desc->name, desc->flags, mType); 1188 lStatus = BAD_VALUE; 1189 goto Exit; 1190 } 1191 1192 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1193 1194 { // scope for mLock 1195 Mutex::Autolock _l(mLock); 1196 1197 // check for existing effect chain with the requested audio session 1198 chain = getEffectChain_l(sessionId); 1199 if (chain == 0) { 1200 // create a new chain for this session 1201 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1202 chain = new EffectChain(this, sessionId); 1203 addEffectChain_l(chain); 1204 chain->setStrategy(getStrategyForSession_l(sessionId)); 1205 chainCreated = true; 1206 } else { 1207 effect = chain->getEffectFromDesc_l(desc); 1208 } 1209 1210 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1211 1212 if (effect == 0) { 1213 int id = mAudioFlinger->nextUniqueId(); 1214 // Check CPU and memory usage 1215 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1216 if (lStatus != NO_ERROR) { 1217 goto Exit; 1218 } 1219 effectRegistered = true; 1220 // create a new effect module if none present in the chain 1221 effect = new EffectModule(this, chain, desc, id, sessionId); 1222 lStatus = effect->status(); 1223 if (lStatus != NO_ERROR) { 1224 goto Exit; 1225 } 1226 effect->setOffloaded(mType == OFFLOAD, mId); 1227 1228 lStatus = chain->addEffect_l(effect); 1229 if (lStatus != NO_ERROR) { 1230 goto Exit; 1231 } 1232 effectCreated = true; 1233 1234 effect->setDevice(mOutDevice); 1235 effect->setDevice(mInDevice); 1236 effect->setMode(mAudioFlinger->getMode()); 1237 effect->setAudioSource(mAudioSource); 1238 } 1239 // create effect handle and connect it to effect module 1240 handle = new EffectHandle(effect, client, effectClient, priority); 1241 lStatus = handle->initCheck(); 1242 if (lStatus == OK) { 1243 lStatus = effect->addHandle(handle.get()); 1244 } 1245 if (enabled != NULL) { 1246 *enabled = (int)effect->isEnabled(); 1247 } 1248 } 1249 1250Exit: 1251 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1252 Mutex::Autolock _l(mLock); 1253 if (effectCreated) { 1254 chain->removeEffect_l(effect); 1255 } 1256 if (effectRegistered) { 1257 AudioSystem::unregisterEffect(effect->id()); 1258 } 1259 if (chainCreated) { 1260 removeEffectChain_l(chain); 1261 } 1262 handle.clear(); 1263 } 1264 1265 *status = lStatus; 1266 return handle; 1267} 1268 1269sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1270{ 1271 Mutex::Autolock _l(mLock); 1272 return getEffect_l(sessionId, effectId); 1273} 1274 1275sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1276{ 1277 sp<EffectChain> chain = getEffectChain_l(sessionId); 1278 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1279} 1280 1281// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1282// PlaybackThread::mLock held 1283status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1284{ 1285 // check for existing effect chain with the requested audio session 1286 int sessionId = effect->sessionId(); 1287 sp<EffectChain> chain = getEffectChain_l(sessionId); 1288 bool chainCreated = false; 1289 1290 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1291 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1292 this, effect->desc().name, effect->desc().flags); 1293 1294 if (chain == 0) { 1295 // create a new chain for this session 1296 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1297 chain = new EffectChain(this, sessionId); 1298 addEffectChain_l(chain); 1299 chain->setStrategy(getStrategyForSession_l(sessionId)); 1300 chainCreated = true; 1301 } 1302 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1303 1304 if (chain->getEffectFromId_l(effect->id()) != 0) { 1305 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1306 this, effect->desc().name, chain.get()); 1307 return BAD_VALUE; 1308 } 1309 1310 effect->setOffloaded(mType == OFFLOAD, mId); 1311 1312 status_t status = chain->addEffect_l(effect); 1313 if (status != NO_ERROR) { 1314 if (chainCreated) { 1315 removeEffectChain_l(chain); 1316 } 1317 return status; 1318 } 1319 1320 effect->setDevice(mOutDevice); 1321 effect->setDevice(mInDevice); 1322 effect->setMode(mAudioFlinger->getMode()); 1323 effect->setAudioSource(mAudioSource); 1324 return NO_ERROR; 1325} 1326 1327void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1328 1329 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1330 effect_descriptor_t desc = effect->desc(); 1331 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1332 detachAuxEffect_l(effect->id()); 1333 } 1334 1335 sp<EffectChain> chain = effect->chain().promote(); 1336 if (chain != 0) { 1337 // remove effect chain if removing last effect 1338 if (chain->removeEffect_l(effect) == 0) { 1339 removeEffectChain_l(chain); 1340 } 1341 } else { 1342 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1343 } 1344} 1345 1346void AudioFlinger::ThreadBase::lockEffectChains_l( 1347 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1348{ 1349 effectChains = mEffectChains; 1350 for (size_t i = 0; i < mEffectChains.size(); i++) { 1351 mEffectChains[i]->lock(); 1352 } 1353} 1354 1355void AudioFlinger::ThreadBase::unlockEffectChains( 1356 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1357{ 1358 for (size_t i = 0; i < effectChains.size(); i++) { 1359 effectChains[i]->unlock(); 1360 } 1361} 1362 1363sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1364{ 1365 Mutex::Autolock _l(mLock); 1366 return getEffectChain_l(sessionId); 1367} 1368 1369sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1370{ 1371 size_t size = mEffectChains.size(); 1372 for (size_t i = 0; i < size; i++) { 1373 if (mEffectChains[i]->sessionId() == sessionId) { 1374 return mEffectChains[i]; 1375 } 1376 } 1377 return 0; 1378} 1379 1380void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1381{ 1382 Mutex::Autolock _l(mLock); 1383 size_t size = mEffectChains.size(); 1384 for (size_t i = 0; i < size; i++) { 1385 mEffectChains[i]->setMode_l(mode); 1386 } 1387} 1388 1389void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1390{ 1391 config->type = AUDIO_PORT_TYPE_MIX; 1392 config->ext.mix.handle = mId; 1393 config->sample_rate = mSampleRate; 1394 config->format = mFormat; 1395 config->channel_mask = mChannelMask; 1396 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1397 AUDIO_PORT_CONFIG_FORMAT; 1398} 1399 1400void AudioFlinger::ThreadBase::systemReady() 1401{ 1402 Mutex::Autolock _l(mLock); 1403 if (mSystemReady) { 1404 return; 1405 } 1406 mSystemReady = true; 1407 1408 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1409 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1410 } 1411 mPendingConfigEvents.clear(); 1412} 1413 1414 1415// ---------------------------------------------------------------------------- 1416// Playback 1417// ---------------------------------------------------------------------------- 1418 1419AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1420 AudioStreamOut* output, 1421 audio_io_handle_t id, 1422 audio_devices_t device, 1423 type_t type, 1424 bool systemReady) 1425 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1426 mNormalFrameCount(0), mSinkBuffer(NULL), 1427 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1428 mMixerBuffer(NULL), 1429 mMixerBufferSize(0), 1430 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1431 mMixerBufferValid(false), 1432 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1433 mEffectBuffer(NULL), 1434 mEffectBufferSize(0), 1435 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1436 mEffectBufferValid(false), 1437 mSuspended(0), mBytesWritten(0), 1438 mActiveTracksGeneration(0), 1439 // mStreamTypes[] initialized in constructor body 1440 mOutput(output), 1441 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1442 mMixerStatus(MIXER_IDLE), 1443 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1444 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1445 mBytesRemaining(0), 1446 mCurrentWriteLength(0), 1447 mUseAsyncWrite(false), 1448 mWriteAckSequence(0), 1449 mDrainSequence(0), 1450 mSignalPending(false), 1451 mScreenState(AudioFlinger::mScreenState), 1452 // index 0 is reserved for normal mixer's submix 1453 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1454 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1455 // mLatchD, mLatchQ, 1456 mLatchDValid(false), mLatchQValid(false) 1457{ 1458 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1459 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1460 1461 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1462 // it would be safer to explicitly pass initial masterVolume/masterMute as 1463 // parameter. 1464 // 1465 // If the HAL we are using has support for master volume or master mute, 1466 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1467 // and the mute set to false). 1468 mMasterVolume = audioFlinger->masterVolume_l(); 1469 mMasterMute = audioFlinger->masterMute_l(); 1470 if (mOutput && mOutput->audioHwDev) { 1471 if (mOutput->audioHwDev->canSetMasterVolume()) { 1472 mMasterVolume = 1.0; 1473 } 1474 1475 if (mOutput->audioHwDev->canSetMasterMute()) { 1476 mMasterMute = false; 1477 } 1478 } 1479 1480 readOutputParameters_l(); 1481 1482 // ++ operator does not compile 1483 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1484 stream = (audio_stream_type_t) (stream + 1)) { 1485 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1486 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1487 } 1488} 1489 1490AudioFlinger::PlaybackThread::~PlaybackThread() 1491{ 1492 mAudioFlinger->unregisterWriter(mNBLogWriter); 1493 free(mSinkBuffer); 1494 free(mMixerBuffer); 1495 free(mEffectBuffer); 1496} 1497 1498void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1499{ 1500 dumpInternals(fd, args); 1501 dumpTracks(fd, args); 1502 dumpEffectChains(fd, args); 1503} 1504 1505void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1506{ 1507 const size_t SIZE = 256; 1508 char buffer[SIZE]; 1509 String8 result; 1510 1511 result.appendFormat(" Stream volumes in dB: "); 1512 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1513 const stream_type_t *st = &mStreamTypes[i]; 1514 if (i > 0) { 1515 result.appendFormat(", "); 1516 } 1517 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1518 if (st->mute) { 1519 result.append("M"); 1520 } 1521 } 1522 result.append("\n"); 1523 write(fd, result.string(), result.length()); 1524 result.clear(); 1525 1526 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1527 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1528 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1529 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1530 1531 size_t numtracks = mTracks.size(); 1532 size_t numactive = mActiveTracks.size(); 1533 dprintf(fd, " %d Tracks", numtracks); 1534 size_t numactiveseen = 0; 1535 if (numtracks) { 1536 dprintf(fd, " of which %d are active\n", numactive); 1537 Track::appendDumpHeader(result); 1538 for (size_t i = 0; i < numtracks; ++i) { 1539 sp<Track> track = mTracks[i]; 1540 if (track != 0) { 1541 bool active = mActiveTracks.indexOf(track) >= 0; 1542 if (active) { 1543 numactiveseen++; 1544 } 1545 track->dump(buffer, SIZE, active); 1546 result.append(buffer); 1547 } 1548 } 1549 } else { 1550 result.append("\n"); 1551 } 1552 if (numactiveseen != numactive) { 1553 // some tracks in the active list were not in the tracks list 1554 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1555 " not in the track list\n"); 1556 result.append(buffer); 1557 Track::appendDumpHeader(result); 1558 for (size_t i = 0; i < numactive; ++i) { 1559 sp<Track> track = mActiveTracks[i].promote(); 1560 if (track != 0 && mTracks.indexOf(track) < 0) { 1561 track->dump(buffer, SIZE, true); 1562 result.append(buffer); 1563 } 1564 } 1565 } 1566 1567 write(fd, result.string(), result.size()); 1568} 1569 1570void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1571{ 1572 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1573 1574 dumpBase(fd, args); 1575 1576 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1577 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1578 dprintf(fd, " Total writes: %d\n", mNumWrites); 1579 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1580 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1581 dprintf(fd, " Suspend count: %d\n", mSuspended); 1582 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1583 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1584 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1585 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1586 AudioStreamOut *output = mOutput; 1587 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1588 String8 flagsAsString = outputFlagsToString(flags); 1589 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1590} 1591 1592// Thread virtuals 1593 1594void AudioFlinger::PlaybackThread::onFirstRef() 1595{ 1596 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1597} 1598 1599// ThreadBase virtuals 1600void AudioFlinger::PlaybackThread::preExit() 1601{ 1602 ALOGV(" preExit()"); 1603 // FIXME this is using hard-coded strings but in the future, this functionality will be 1604 // converted to use audio HAL extensions required to support tunneling 1605 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1606} 1607 1608// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1609sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1610 const sp<AudioFlinger::Client>& client, 1611 audio_stream_type_t streamType, 1612 uint32_t sampleRate, 1613 audio_format_t format, 1614 audio_channel_mask_t channelMask, 1615 size_t *pFrameCount, 1616 const sp<IMemory>& sharedBuffer, 1617 int sessionId, 1618 IAudioFlinger::track_flags_t *flags, 1619 pid_t tid, 1620 int uid, 1621 status_t *status) 1622{ 1623 size_t frameCount = *pFrameCount; 1624 sp<Track> track; 1625 status_t lStatus; 1626 1627 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1628 1629 // client expresses a preference for FAST, but we get the final say 1630 if (*flags & IAudioFlinger::TRACK_FAST) { 1631 if ( 1632 // not timed 1633 (!isTimed) && 1634 // either of these use cases: 1635 ( 1636 // use case 1: shared buffer with any frame count 1637 ( 1638 (sharedBuffer != 0) 1639 ) || 1640 // use case 2: frame count is default or at least as large as HAL 1641 ( 1642 // we formerly checked for a callback handler (non-0 tid), 1643 // but that is no longer required for TRANSFER_OBTAIN mode 1644 ((frameCount == 0) || 1645 (frameCount >= mFrameCount)) 1646 ) 1647 ) && 1648 // PCM data 1649 audio_is_linear_pcm(format) && 1650 // TODO: extract as a data library function that checks that a computationally 1651 // expensive downmixer is not required: isFastOutputChannelConversion() 1652 (channelMask == mChannelMask || 1653 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1654 (channelMask == AUDIO_CHANNEL_OUT_MONO 1655 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1656 // hardware sample rate 1657 (sampleRate == mSampleRate) && 1658 // normal mixer has an associated fast mixer 1659 hasFastMixer() && 1660 // there are sufficient fast track slots available 1661 (mFastTrackAvailMask != 0) 1662 // FIXME test that MixerThread for this fast track has a capable output HAL 1663 // FIXME add a permission test also? 1664 ) { 1665 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1666 if (frameCount == 0) { 1667 // read the fast track multiplier property the first time it is needed 1668 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1669 if (ok != 0) { 1670 ALOGE("%s pthread_once failed: %d", __func__, ok); 1671 } 1672 frameCount = mFrameCount * sFastTrackMultiplier; 1673 } 1674 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1675 frameCount, mFrameCount); 1676 } else { 1677 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1678 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1679 "sampleRate=%u mSampleRate=%u " 1680 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1681 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1682 audio_is_linear_pcm(format), 1683 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1684 *flags &= ~IAudioFlinger::TRACK_FAST; 1685 } 1686 } 1687 // For normal PCM streaming tracks, update minimum frame count. 1688 // For compatibility with AudioTrack calculation, buffer depth is forced 1689 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1690 // This is probably too conservative, but legacy application code may depend on it. 1691 // If you change this calculation, also review the start threshold which is related. 1692 if (!(*flags & IAudioFlinger::TRACK_FAST) 1693 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1694 // this must match AudioTrack.cpp calculateMinFrameCount(). 1695 // TODO: Move to a common library 1696 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1697 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1698 if (minBufCount < 2) { 1699 minBufCount = 2; 1700 } 1701 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1702 // or the client should compute and pass in a larger buffer request. 1703 size_t minFrameCount = 1704 minBufCount * sourceFramesNeededWithTimestretch( 1705 sampleRate, mNormalFrameCount, 1706 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1707 if (frameCount < minFrameCount) { // including frameCount == 0 1708 frameCount = minFrameCount; 1709 } 1710 } 1711 *pFrameCount = frameCount; 1712 1713 switch (mType) { 1714 1715 case DIRECT: 1716 if (audio_is_linear_pcm(format)) { 1717 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1718 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1719 "for output %p with format %#x", 1720 sampleRate, format, channelMask, mOutput, mFormat); 1721 lStatus = BAD_VALUE; 1722 goto Exit; 1723 } 1724 } 1725 break; 1726 1727 case OFFLOAD: 1728 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1729 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1730 "for output %p with format %#x", 1731 sampleRate, format, channelMask, mOutput, mFormat); 1732 lStatus = BAD_VALUE; 1733 goto Exit; 1734 } 1735 break; 1736 1737 default: 1738 if (!audio_is_linear_pcm(format)) { 1739 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1740 "for output %p with format %#x", 1741 format, mOutput, mFormat); 1742 lStatus = BAD_VALUE; 1743 goto Exit; 1744 } 1745 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1746 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1747 lStatus = BAD_VALUE; 1748 goto Exit; 1749 } 1750 break; 1751 1752 } 1753 1754 lStatus = initCheck(); 1755 if (lStatus != NO_ERROR) { 1756 ALOGE("createTrack_l() audio driver not initialized"); 1757 goto Exit; 1758 } 1759 1760 { // scope for mLock 1761 Mutex::Autolock _l(mLock); 1762 1763 // all tracks in same audio session must share the same routing strategy otherwise 1764 // conflicts will happen when tracks are moved from one output to another by audio policy 1765 // manager 1766 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1767 for (size_t i = 0; i < mTracks.size(); ++i) { 1768 sp<Track> t = mTracks[i]; 1769 if (t != 0 && t->isExternalTrack()) { 1770 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1771 if (sessionId == t->sessionId() && strategy != actual) { 1772 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1773 strategy, actual); 1774 lStatus = BAD_VALUE; 1775 goto Exit; 1776 } 1777 } 1778 } 1779 1780 if (!isTimed) { 1781 track = new Track(this, client, streamType, sampleRate, format, 1782 channelMask, frameCount, NULL, sharedBuffer, 1783 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1784 } else { 1785 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1786 channelMask, frameCount, sharedBuffer, sessionId, uid); 1787 } 1788 1789 // new Track always returns non-NULL, 1790 // but TimedTrack::create() is a factory that could fail by returning NULL 1791 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1792 if (lStatus != NO_ERROR) { 1793 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1794 // track must be cleared from the caller as the caller has the AF lock 1795 goto Exit; 1796 } 1797 mTracks.add(track); 1798 1799 sp<EffectChain> chain = getEffectChain_l(sessionId); 1800 if (chain != 0) { 1801 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1802 track->setMainBuffer(chain->inBuffer()); 1803 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1804 chain->incTrackCnt(); 1805 } 1806 1807 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1808 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1809 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1810 // so ask activity manager to do this on our behalf 1811 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1812 } 1813 } 1814 1815 lStatus = NO_ERROR; 1816 1817Exit: 1818 *status = lStatus; 1819 return track; 1820} 1821 1822uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1823{ 1824 return latency; 1825} 1826 1827uint32_t AudioFlinger::PlaybackThread::latency() const 1828{ 1829 Mutex::Autolock _l(mLock); 1830 return latency_l(); 1831} 1832uint32_t AudioFlinger::PlaybackThread::latency_l() const 1833{ 1834 if (initCheck() == NO_ERROR) { 1835 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1836 } else { 1837 return 0; 1838 } 1839} 1840 1841void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1842{ 1843 Mutex::Autolock _l(mLock); 1844 // Don't apply master volume in SW if our HAL can do it for us. 1845 if (mOutput && mOutput->audioHwDev && 1846 mOutput->audioHwDev->canSetMasterVolume()) { 1847 mMasterVolume = 1.0; 1848 } else { 1849 mMasterVolume = value; 1850 } 1851} 1852 1853void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1854{ 1855 Mutex::Autolock _l(mLock); 1856 // Don't apply master mute in SW if our HAL can do it for us. 1857 if (mOutput && mOutput->audioHwDev && 1858 mOutput->audioHwDev->canSetMasterMute()) { 1859 mMasterMute = false; 1860 } else { 1861 mMasterMute = muted; 1862 } 1863} 1864 1865void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1866{ 1867 Mutex::Autolock _l(mLock); 1868 mStreamTypes[stream].volume = value; 1869 broadcast_l(); 1870} 1871 1872void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1873{ 1874 Mutex::Autolock _l(mLock); 1875 mStreamTypes[stream].mute = muted; 1876 broadcast_l(); 1877} 1878 1879float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1880{ 1881 Mutex::Autolock _l(mLock); 1882 return mStreamTypes[stream].volume; 1883} 1884 1885// addTrack_l() must be called with ThreadBase::mLock held 1886status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1887{ 1888 status_t status = ALREADY_EXISTS; 1889 1890 // set retry count for buffer fill 1891 track->mRetryCount = kMaxTrackStartupRetries; 1892 if (mActiveTracks.indexOf(track) < 0) { 1893 // the track is newly added, make sure it fills up all its 1894 // buffers before playing. This is to ensure the client will 1895 // effectively get the latency it requested. 1896 if (track->isExternalTrack()) { 1897 TrackBase::track_state state = track->mState; 1898 mLock.unlock(); 1899 status = AudioSystem::startOutput(mId, track->streamType(), 1900 (audio_session_t)track->sessionId()); 1901 mLock.lock(); 1902 // abort track was stopped/paused while we released the lock 1903 if (state != track->mState) { 1904 if (status == NO_ERROR) { 1905 mLock.unlock(); 1906 AudioSystem::stopOutput(mId, track->streamType(), 1907 (audio_session_t)track->sessionId()); 1908 mLock.lock(); 1909 } 1910 return INVALID_OPERATION; 1911 } 1912 // abort if start is rejected by audio policy manager 1913 if (status != NO_ERROR) { 1914 return PERMISSION_DENIED; 1915 } 1916#ifdef ADD_BATTERY_DATA 1917 // to track the speaker usage 1918 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1919#endif 1920 } 1921 1922 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1923 track->mResetDone = false; 1924 track->mPresentationCompleteFrames = 0; 1925 mActiveTracks.add(track); 1926 mWakeLockUids.add(track->uid()); 1927 mActiveTracksGeneration++; 1928 mLatestActiveTrack = track; 1929 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1930 if (chain != 0) { 1931 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1932 track->sessionId()); 1933 chain->incActiveTrackCnt(); 1934 } 1935 1936 status = NO_ERROR; 1937 } 1938 1939 onAddNewTrack_l(); 1940 return status; 1941} 1942 1943bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1944{ 1945 track->terminate(); 1946 // active tracks are removed by threadLoop() 1947 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1948 track->mState = TrackBase::STOPPED; 1949 if (!trackActive) { 1950 removeTrack_l(track); 1951 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1952 track->mState = TrackBase::STOPPING_1; 1953 } 1954 1955 return trackActive; 1956} 1957 1958void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1959{ 1960 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1961 mTracks.remove(track); 1962 deleteTrackName_l(track->name()); 1963 // redundant as track is about to be destroyed, for dumpsys only 1964 track->mName = -1; 1965 if (track->isFastTrack()) { 1966 int index = track->mFastIndex; 1967 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1968 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1969 mFastTrackAvailMask |= 1 << index; 1970 // redundant as track is about to be destroyed, for dumpsys only 1971 track->mFastIndex = -1; 1972 } 1973 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1974 if (chain != 0) { 1975 chain->decTrackCnt(); 1976 } 1977} 1978 1979void AudioFlinger::PlaybackThread::broadcast_l() 1980{ 1981 // Thread could be blocked waiting for async 1982 // so signal it to handle state changes immediately 1983 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1984 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1985 mSignalPending = true; 1986 mWaitWorkCV.broadcast(); 1987} 1988 1989String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1990{ 1991 Mutex::Autolock _l(mLock); 1992 if (initCheck() != NO_ERROR) { 1993 return String8(); 1994 } 1995 1996 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1997 const String8 out_s8(s); 1998 free(s); 1999 return out_s8; 2000} 2001 2002void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event) { 2003 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2004 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2005 2006 desc->mIoHandle = mId; 2007 2008 switch (event) { 2009 case AUDIO_OUTPUT_OPENED: 2010 case AUDIO_OUTPUT_CONFIG_CHANGED: 2011 desc->mPatch = mPatch; 2012 desc->mChannelMask = mChannelMask; 2013 desc->mSamplingRate = mSampleRate; 2014 desc->mFormat = mFormat; 2015 desc->mFrameCount = mNormalFrameCount; // FIXME see 2016 // AudioFlinger::frameCount(audio_io_handle_t) 2017 desc->mLatency = latency_l(); 2018 break; 2019 2020 case AUDIO_OUTPUT_CLOSED: 2021 default: 2022 break; 2023 } 2024 mAudioFlinger->ioConfigChanged(event, desc); 2025} 2026 2027void AudioFlinger::PlaybackThread::writeCallback() 2028{ 2029 ALOG_ASSERT(mCallbackThread != 0); 2030 mCallbackThread->resetWriteBlocked(); 2031} 2032 2033void AudioFlinger::PlaybackThread::drainCallback() 2034{ 2035 ALOG_ASSERT(mCallbackThread != 0); 2036 mCallbackThread->resetDraining(); 2037} 2038 2039void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2040{ 2041 Mutex::Autolock _l(mLock); 2042 // reject out of sequence requests 2043 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2044 mWriteAckSequence &= ~1; 2045 mWaitWorkCV.signal(); 2046 } 2047} 2048 2049void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2050{ 2051 Mutex::Autolock _l(mLock); 2052 // reject out of sequence requests 2053 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2054 mDrainSequence &= ~1; 2055 mWaitWorkCV.signal(); 2056 } 2057} 2058 2059// static 2060int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2061 void *param __unused, 2062 void *cookie) 2063{ 2064 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2065 ALOGV("asyncCallback() event %d", event); 2066 switch (event) { 2067 case STREAM_CBK_EVENT_WRITE_READY: 2068 me->writeCallback(); 2069 break; 2070 case STREAM_CBK_EVENT_DRAIN_READY: 2071 me->drainCallback(); 2072 break; 2073 default: 2074 ALOGW("asyncCallback() unknown event %d", event); 2075 break; 2076 } 2077 return 0; 2078} 2079 2080void AudioFlinger::PlaybackThread::readOutputParameters_l() 2081{ 2082 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2083 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2084 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2085 if (!audio_is_output_channel(mChannelMask)) { 2086 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2087 } 2088 if ((mType == MIXER || mType == DUPLICATING) 2089 && !isValidPcmSinkChannelMask(mChannelMask)) { 2090 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2091 mChannelMask); 2092 } 2093 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2094 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2095 mFormat = mHALFormat; 2096 if (!audio_is_valid_format(mFormat)) { 2097 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2098 } 2099 if ((mType == MIXER || mType == DUPLICATING) 2100 && !isValidPcmSinkFormat(mFormat)) { 2101 LOG_FATAL("HAL format %#x not supported for mixed output", 2102 mFormat); 2103 } 2104 mFrameSize = mOutput->getFrameSize(); 2105 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2106 mFrameCount = mBufferSize / mFrameSize; 2107 if (mFrameCount & 15) { 2108 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2109 mFrameCount); 2110 } 2111 2112 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2113 (mOutput->stream->set_callback != NULL)) { 2114 if (mOutput->stream->set_callback(mOutput->stream, 2115 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2116 mUseAsyncWrite = true; 2117 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2118 } 2119 } 2120 2121 mHwSupportsPause = false; 2122 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2123 if (mOutput->stream->pause != NULL) { 2124 if (mOutput->stream->resume != NULL) { 2125 mHwSupportsPause = true; 2126 } else { 2127 ALOGW("direct output implements pause but not resume"); 2128 } 2129 } else if (mOutput->stream->resume != NULL) { 2130 ALOGW("direct output implements resume but not pause"); 2131 } 2132 } 2133 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2134 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2135 } 2136 2137 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2138 // For best precision, we use float instead of the associated output 2139 // device format (typically PCM 16 bit). 2140 2141 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2142 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2143 mBufferSize = mFrameSize * mFrameCount; 2144 2145 // TODO: We currently use the associated output device channel mask and sample rate. 2146 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2147 // (if a valid mask) to avoid premature downmix. 2148 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2149 // instead of the output device sample rate to avoid loss of high frequency information. 2150 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2151 } 2152 2153 // Calculate size of normal sink buffer relative to the HAL output buffer size 2154 double multiplier = 1.0; 2155 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2156 kUseFastMixer == FastMixer_Dynamic)) { 2157 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2158 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2159 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2160 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2161 maxNormalFrameCount = maxNormalFrameCount & ~15; 2162 if (maxNormalFrameCount < minNormalFrameCount) { 2163 maxNormalFrameCount = minNormalFrameCount; 2164 } 2165 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2166 if (multiplier <= 1.0) { 2167 multiplier = 1.0; 2168 } else if (multiplier <= 2.0) { 2169 if (2 * mFrameCount <= maxNormalFrameCount) { 2170 multiplier = 2.0; 2171 } else { 2172 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2173 } 2174 } else { 2175 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2176 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2177 // track, but we sometimes have to do this to satisfy the maximum frame count 2178 // constraint) 2179 // FIXME this rounding up should not be done if no HAL SRC 2180 uint32_t truncMult = (uint32_t) multiplier; 2181 if ((truncMult & 1)) { 2182 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2183 ++truncMult; 2184 } 2185 } 2186 multiplier = (double) truncMult; 2187 } 2188 } 2189 mNormalFrameCount = multiplier * mFrameCount; 2190 // round up to nearest 16 frames to satisfy AudioMixer 2191 if (mType == MIXER || mType == DUPLICATING) { 2192 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2193 } 2194 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2195 mNormalFrameCount); 2196 2197 // Check if we want to throttle the processing to no more than 2x normal rate 2198 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2199 mThreadThrottleTimeMs = 0; 2200 mThreadThrottleEndMs = 0; 2201 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2202 2203 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2204 // Originally this was int16_t[] array, need to remove legacy implications. 2205 free(mSinkBuffer); 2206 mSinkBuffer = NULL; 2207 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2208 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2209 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2210 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2211 2212 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2213 // drives the output. 2214 free(mMixerBuffer); 2215 mMixerBuffer = NULL; 2216 if (mMixerBufferEnabled) { 2217 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2218 mMixerBufferSize = mNormalFrameCount * mChannelCount 2219 * audio_bytes_per_sample(mMixerBufferFormat); 2220 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2221 } 2222 free(mEffectBuffer); 2223 mEffectBuffer = NULL; 2224 if (mEffectBufferEnabled) { 2225 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2226 mEffectBufferSize = mNormalFrameCount * mChannelCount 2227 * audio_bytes_per_sample(mEffectBufferFormat); 2228 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2229 } 2230 2231 // force reconfiguration of effect chains and engines to take new buffer size and audio 2232 // parameters into account 2233 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2234 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2235 // matter. 2236 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2237 Vector< sp<EffectChain> > effectChains = mEffectChains; 2238 for (size_t i = 0; i < effectChains.size(); i ++) { 2239 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2240 } 2241} 2242 2243 2244status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2245{ 2246 if (halFrames == NULL || dspFrames == NULL) { 2247 return BAD_VALUE; 2248 } 2249 Mutex::Autolock _l(mLock); 2250 if (initCheck() != NO_ERROR) { 2251 return INVALID_OPERATION; 2252 } 2253 size_t framesWritten = mBytesWritten / mFrameSize; 2254 *halFrames = framesWritten; 2255 2256 if (isSuspended()) { 2257 // return an estimation of rendered frames when the output is suspended 2258 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2259 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2260 return NO_ERROR; 2261 } else { 2262 status_t status; 2263 uint32_t frames; 2264 status = mOutput->getRenderPosition(&frames); 2265 *dspFrames = (size_t)frames; 2266 return status; 2267 } 2268} 2269 2270uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2271{ 2272 Mutex::Autolock _l(mLock); 2273 uint32_t result = 0; 2274 if (getEffectChain_l(sessionId) != 0) { 2275 result = EFFECT_SESSION; 2276 } 2277 2278 for (size_t i = 0; i < mTracks.size(); ++i) { 2279 sp<Track> track = mTracks[i]; 2280 if (sessionId == track->sessionId() && !track->isInvalid()) { 2281 result |= TRACK_SESSION; 2282 break; 2283 } 2284 } 2285 2286 return result; 2287} 2288 2289uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2290{ 2291 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2292 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2293 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2294 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2295 } 2296 for (size_t i = 0; i < mTracks.size(); i++) { 2297 sp<Track> track = mTracks[i]; 2298 if (sessionId == track->sessionId() && !track->isInvalid()) { 2299 return AudioSystem::getStrategyForStream(track->streamType()); 2300 } 2301 } 2302 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2303} 2304 2305 2306AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2307{ 2308 Mutex::Autolock _l(mLock); 2309 return mOutput; 2310} 2311 2312AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2313{ 2314 Mutex::Autolock _l(mLock); 2315 AudioStreamOut *output = mOutput; 2316 mOutput = NULL; 2317 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2318 // must push a NULL and wait for ack 2319 mOutputSink.clear(); 2320 mPipeSink.clear(); 2321 mNormalSink.clear(); 2322 return output; 2323} 2324 2325// this method must always be called either with ThreadBase mLock held or inside the thread loop 2326audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2327{ 2328 if (mOutput == NULL) { 2329 return NULL; 2330 } 2331 return &mOutput->stream->common; 2332} 2333 2334uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2335{ 2336 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2337} 2338 2339status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2340{ 2341 if (!isValidSyncEvent(event)) { 2342 return BAD_VALUE; 2343 } 2344 2345 Mutex::Autolock _l(mLock); 2346 2347 for (size_t i = 0; i < mTracks.size(); ++i) { 2348 sp<Track> track = mTracks[i]; 2349 if (event->triggerSession() == track->sessionId()) { 2350 (void) track->setSyncEvent(event); 2351 return NO_ERROR; 2352 } 2353 } 2354 2355 return NAME_NOT_FOUND; 2356} 2357 2358bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2359{ 2360 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2361} 2362 2363void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2364 const Vector< sp<Track> >& tracksToRemove) 2365{ 2366 size_t count = tracksToRemove.size(); 2367 if (count > 0) { 2368 for (size_t i = 0 ; i < count ; i++) { 2369 const sp<Track>& track = tracksToRemove.itemAt(i); 2370 if (track->isExternalTrack()) { 2371 AudioSystem::stopOutput(mId, track->streamType(), 2372 (audio_session_t)track->sessionId()); 2373#ifdef ADD_BATTERY_DATA 2374 // to track the speaker usage 2375 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2376#endif 2377 if (track->isTerminated()) { 2378 AudioSystem::releaseOutput(mId, track->streamType(), 2379 (audio_session_t)track->sessionId()); 2380 } 2381 } 2382 } 2383 } 2384} 2385 2386void AudioFlinger::PlaybackThread::checkSilentMode_l() 2387{ 2388 if (!mMasterMute) { 2389 char value[PROPERTY_VALUE_MAX]; 2390 if (property_get("ro.audio.silent", value, "0") > 0) { 2391 char *endptr; 2392 unsigned long ul = strtoul(value, &endptr, 0); 2393 if (*endptr == '\0' && ul != 0) { 2394 ALOGD("Silence is golden"); 2395 // The setprop command will not allow a property to be changed after 2396 // the first time it is set, so we don't have to worry about un-muting. 2397 setMasterMute_l(true); 2398 } 2399 } 2400 } 2401} 2402 2403// shared by MIXER and DIRECT, overridden by DUPLICATING 2404ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2405{ 2406 // FIXME rewrite to reduce number of system calls 2407 mLastWriteTime = systemTime(); 2408 mInWrite = true; 2409 ssize_t bytesWritten; 2410 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2411 2412 // If an NBAIO sink is present, use it to write the normal mixer's submix 2413 if (mNormalSink != 0) { 2414 2415 const size_t count = mBytesRemaining / mFrameSize; 2416 2417 ATRACE_BEGIN("write"); 2418 // update the setpoint when AudioFlinger::mScreenState changes 2419 uint32_t screenState = AudioFlinger::mScreenState; 2420 if (screenState != mScreenState) { 2421 mScreenState = screenState; 2422 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2423 if (pipe != NULL) { 2424 pipe->setAvgFrames((mScreenState & 1) ? 2425 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2426 } 2427 } 2428 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2429 ATRACE_END(); 2430 if (framesWritten > 0) { 2431 bytesWritten = framesWritten * mFrameSize; 2432 } else { 2433 bytesWritten = framesWritten; 2434 } 2435 mLatchDValid = false; 2436 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2437 if (status == NO_ERROR) { 2438 size_t totalFramesWritten = mNormalSink->framesWritten(); 2439 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2440 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2441 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2442 mLatchDValid = true; 2443 } 2444 } 2445 // otherwise use the HAL / AudioStreamOut directly 2446 } else { 2447 // Direct output and offload threads 2448 2449 if (mUseAsyncWrite) { 2450 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2451 mWriteAckSequence += 2; 2452 mWriteAckSequence |= 1; 2453 ALOG_ASSERT(mCallbackThread != 0); 2454 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2455 } 2456 // FIXME We should have an implementation of timestamps for direct output threads. 2457 // They are used e.g for multichannel PCM playback over HDMI. 2458 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2459 if (mUseAsyncWrite && 2460 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2461 // do not wait for async callback in case of error of full write 2462 mWriteAckSequence &= ~1; 2463 ALOG_ASSERT(mCallbackThread != 0); 2464 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2465 } 2466 } 2467 2468 mNumWrites++; 2469 mInWrite = false; 2470 mStandby = false; 2471 return bytesWritten; 2472} 2473 2474void AudioFlinger::PlaybackThread::threadLoop_drain() 2475{ 2476 if (mOutput->stream->drain) { 2477 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2478 if (mUseAsyncWrite) { 2479 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2480 mDrainSequence |= 1; 2481 ALOG_ASSERT(mCallbackThread != 0); 2482 mCallbackThread->setDraining(mDrainSequence); 2483 } 2484 mOutput->stream->drain(mOutput->stream, 2485 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2486 : AUDIO_DRAIN_ALL); 2487 } 2488} 2489 2490void AudioFlinger::PlaybackThread::threadLoop_exit() 2491{ 2492 { 2493 Mutex::Autolock _l(mLock); 2494 for (size_t i = 0; i < mTracks.size(); i++) { 2495 sp<Track> track = mTracks[i]; 2496 track->invalidate(); 2497 } 2498 } 2499} 2500 2501/* 2502The derived values that are cached: 2503 - mSinkBufferSize from frame count * frame size 2504 - mActiveSleepTimeUs from activeSleepTimeUs() 2505 - mIdleSleepTimeUs from idleSleepTimeUs() 2506 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) 2507 - maxPeriod from frame count and sample rate (MIXER only) 2508 2509The parameters that affect these derived values are: 2510 - frame count 2511 - frame size 2512 - sample rate 2513 - device type: A2DP or not 2514 - device latency 2515 - format: PCM or not 2516 - active sleep time 2517 - idle sleep time 2518*/ 2519 2520void AudioFlinger::PlaybackThread::cacheParameters_l() 2521{ 2522 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2523 mActiveSleepTimeUs = activeSleepTimeUs(); 2524 mIdleSleepTimeUs = idleSleepTimeUs(); 2525} 2526 2527void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2528{ 2529 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2530 this, streamType, mTracks.size()); 2531 Mutex::Autolock _l(mLock); 2532 2533 size_t size = mTracks.size(); 2534 for (size_t i = 0; i < size; i++) { 2535 sp<Track> t = mTracks[i]; 2536 if (t->streamType() == streamType) { 2537 t->invalidate(); 2538 } 2539 } 2540} 2541 2542status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2543{ 2544 int session = chain->sessionId(); 2545 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2546 ? mEffectBuffer : mSinkBuffer); 2547 bool ownsBuffer = false; 2548 2549 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2550 if (session > 0) { 2551 // Only one effect chain can be present in direct output thread and it uses 2552 // the sink buffer as input 2553 if (mType != DIRECT) { 2554 size_t numSamples = mNormalFrameCount * mChannelCount; 2555 buffer = new int16_t[numSamples]; 2556 memset(buffer, 0, numSamples * sizeof(int16_t)); 2557 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2558 ownsBuffer = true; 2559 } 2560 2561 // Attach all tracks with same session ID to this chain. 2562 for (size_t i = 0; i < mTracks.size(); ++i) { 2563 sp<Track> track = mTracks[i]; 2564 if (session == track->sessionId()) { 2565 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2566 buffer); 2567 track->setMainBuffer(buffer); 2568 chain->incTrackCnt(); 2569 } 2570 } 2571 2572 // indicate all active tracks in the chain 2573 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2574 sp<Track> track = mActiveTracks[i].promote(); 2575 if (track == 0) { 2576 continue; 2577 } 2578 if (session == track->sessionId()) { 2579 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2580 chain->incActiveTrackCnt(); 2581 } 2582 } 2583 } 2584 chain->setThread(this); 2585 chain->setInBuffer(buffer, ownsBuffer); 2586 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2587 ? mEffectBuffer : mSinkBuffer)); 2588 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2589 // chains list in order to be processed last as it contains output stage effects 2590 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2591 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2592 // after track specific effects and before output stage 2593 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2594 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2595 // Effect chain for other sessions are inserted at beginning of effect 2596 // chains list to be processed before output mix effects. Relative order between other 2597 // sessions is not important 2598 size_t size = mEffectChains.size(); 2599 size_t i = 0; 2600 for (i = 0; i < size; i++) { 2601 if (mEffectChains[i]->sessionId() < session) { 2602 break; 2603 } 2604 } 2605 mEffectChains.insertAt(chain, i); 2606 checkSuspendOnAddEffectChain_l(chain); 2607 2608 return NO_ERROR; 2609} 2610 2611size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2612{ 2613 int session = chain->sessionId(); 2614 2615 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2616 2617 for (size_t i = 0; i < mEffectChains.size(); i++) { 2618 if (chain == mEffectChains[i]) { 2619 mEffectChains.removeAt(i); 2620 // detach all active tracks from the chain 2621 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2622 sp<Track> track = mActiveTracks[i].promote(); 2623 if (track == 0) { 2624 continue; 2625 } 2626 if (session == track->sessionId()) { 2627 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2628 chain.get(), session); 2629 chain->decActiveTrackCnt(); 2630 } 2631 } 2632 2633 // detach all tracks with same session ID from this chain 2634 for (size_t i = 0; i < mTracks.size(); ++i) { 2635 sp<Track> track = mTracks[i]; 2636 if (session == track->sessionId()) { 2637 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2638 chain->decTrackCnt(); 2639 } 2640 } 2641 break; 2642 } 2643 } 2644 return mEffectChains.size(); 2645} 2646 2647status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2648 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2649{ 2650 Mutex::Autolock _l(mLock); 2651 return attachAuxEffect_l(track, EffectId); 2652} 2653 2654status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2655 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2656{ 2657 status_t status = NO_ERROR; 2658 2659 if (EffectId == 0) { 2660 track->setAuxBuffer(0, NULL); 2661 } else { 2662 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2663 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2664 if (effect != 0) { 2665 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2666 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2667 } else { 2668 status = INVALID_OPERATION; 2669 } 2670 } else { 2671 status = BAD_VALUE; 2672 } 2673 } 2674 return status; 2675} 2676 2677void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2678{ 2679 for (size_t i = 0; i < mTracks.size(); ++i) { 2680 sp<Track> track = mTracks[i]; 2681 if (track->auxEffectId() == effectId) { 2682 attachAuxEffect_l(track, 0); 2683 } 2684 } 2685} 2686 2687bool AudioFlinger::PlaybackThread::threadLoop() 2688{ 2689 Vector< sp<Track> > tracksToRemove; 2690 2691 mStandbyTimeNs = systemTime(); 2692 2693 // MIXER 2694 nsecs_t lastWarning = 0; 2695 2696 // DUPLICATING 2697 // FIXME could this be made local to while loop? 2698 writeFrames = 0; 2699 2700 int lastGeneration = 0; 2701 2702 cacheParameters_l(); 2703 mSleepTimeUs = mIdleSleepTimeUs; 2704 2705 if (mType == MIXER) { 2706 sleepTimeShift = 0; 2707 } 2708 2709 CpuStats cpuStats; 2710 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2711 2712 acquireWakeLock(); 2713 2714 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2715 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2716 // and then that string will be logged at the next convenient opportunity. 2717 const char *logString = NULL; 2718 2719 checkSilentMode_l(); 2720 2721 while (!exitPending()) 2722 { 2723 cpuStats.sample(myName); 2724 2725 Vector< sp<EffectChain> > effectChains; 2726 2727 { // scope for mLock 2728 2729 Mutex::Autolock _l(mLock); 2730 2731 processConfigEvents_l(); 2732 2733 if (logString != NULL) { 2734 mNBLogWriter->logTimestamp(); 2735 mNBLogWriter->log(logString); 2736 logString = NULL; 2737 } 2738 2739 // Gather the framesReleased counters for all active tracks, 2740 // and latch them atomically with the timestamp. 2741 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2742 mLatchD.mFramesReleased.clear(); 2743 size_t size = mActiveTracks.size(); 2744 for (size_t i = 0; i < size; i++) { 2745 sp<Track> t = mActiveTracks[i].promote(); 2746 if (t != 0) { 2747 mLatchD.mFramesReleased.add(t.get(), 2748 t->mAudioTrackServerProxy->framesReleased()); 2749 } 2750 } 2751 if (mLatchDValid) { 2752 mLatchQ = mLatchD; 2753 mLatchDValid = false; 2754 mLatchQValid = true; 2755 } 2756 2757 saveOutputTracks(); 2758 if (mSignalPending) { 2759 // A signal was raised while we were unlocked 2760 mSignalPending = false; 2761 } else if (waitingAsyncCallback_l()) { 2762 if (exitPending()) { 2763 break; 2764 } 2765 bool released = false; 2766 // The following works around a bug in the offload driver. Ideally we would release 2767 // the wake lock every time, but that causes the last offload buffer(s) to be 2768 // dropped while the device is on battery, so we need to hold a wake lock during 2769 // the drain phase. 2770 if (mBytesRemaining && !(mDrainSequence & 1)) { 2771 releaseWakeLock_l(); 2772 released = true; 2773 } 2774 mWakeLockUids.clear(); 2775 mActiveTracksGeneration++; 2776 ALOGV("wait async completion"); 2777 mWaitWorkCV.wait(mLock); 2778 ALOGV("async completion/wake"); 2779 if (released) { 2780 acquireWakeLock_l(); 2781 } 2782 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2783 mSleepTimeUs = 0; 2784 2785 continue; 2786 } 2787 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2788 isSuspended()) { 2789 // put audio hardware into standby after short delay 2790 if (shouldStandby_l()) { 2791 2792 threadLoop_standby(); 2793 2794 mStandby = true; 2795 } 2796 2797 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2798 // we're about to wait, flush the binder command buffer 2799 IPCThreadState::self()->flushCommands(); 2800 2801 clearOutputTracks(); 2802 2803 if (exitPending()) { 2804 break; 2805 } 2806 2807 releaseWakeLock_l(); 2808 mWakeLockUids.clear(); 2809 mActiveTracksGeneration++; 2810 // wait until we have something to do... 2811 ALOGV("%s going to sleep", myName.string()); 2812 mWaitWorkCV.wait(mLock); 2813 ALOGV("%s waking up", myName.string()); 2814 acquireWakeLock_l(); 2815 2816 mMixerStatus = MIXER_IDLE; 2817 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2818 mBytesWritten = 0; 2819 mBytesRemaining = 0; 2820 checkSilentMode_l(); 2821 2822 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2823 mSleepTimeUs = mIdleSleepTimeUs; 2824 if (mType == MIXER) { 2825 sleepTimeShift = 0; 2826 } 2827 2828 continue; 2829 } 2830 } 2831 // mMixerStatusIgnoringFastTracks is also updated internally 2832 mMixerStatus = prepareTracks_l(&tracksToRemove); 2833 2834 // compare with previously applied list 2835 if (lastGeneration != mActiveTracksGeneration) { 2836 // update wakelock 2837 updateWakeLockUids_l(mWakeLockUids); 2838 lastGeneration = mActiveTracksGeneration; 2839 } 2840 2841 // prevent any changes in effect chain list and in each effect chain 2842 // during mixing and effect process as the audio buffers could be deleted 2843 // or modified if an effect is created or deleted 2844 lockEffectChains_l(effectChains); 2845 } // mLock scope ends 2846 2847 if (mBytesRemaining == 0) { 2848 mCurrentWriteLength = 0; 2849 if (mMixerStatus == MIXER_TRACKS_READY) { 2850 // threadLoop_mix() sets mCurrentWriteLength 2851 threadLoop_mix(); 2852 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2853 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2854 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2855 // must be written to HAL 2856 threadLoop_sleepTime(); 2857 if (mSleepTimeUs == 0) { 2858 mCurrentWriteLength = mSinkBufferSize; 2859 } 2860 } 2861 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2862 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2863 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2864 // or mSinkBuffer (if there are no effects). 2865 // 2866 // This is done pre-effects computation; if effects change to 2867 // support higher precision, this needs to move. 2868 // 2869 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2870 // TODO use mSleepTimeUs == 0 as an additional condition. 2871 if (mMixerBufferValid) { 2872 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2873 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2874 2875 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2876 mNormalFrameCount * mChannelCount); 2877 } 2878 2879 mBytesRemaining = mCurrentWriteLength; 2880 if (isSuspended()) { 2881 mSleepTimeUs = suspendSleepTimeUs(); 2882 // simulate write to HAL when suspended 2883 mBytesWritten += mSinkBufferSize; 2884 mBytesRemaining = 0; 2885 } 2886 2887 // only process effects if we're going to write 2888 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 2889 for (size_t i = 0; i < effectChains.size(); i ++) { 2890 effectChains[i]->process_l(); 2891 } 2892 } 2893 } 2894 // Process effect chains for offloaded thread even if no audio 2895 // was read from audio track: process only updates effect state 2896 // and thus does have to be synchronized with audio writes but may have 2897 // to be called while waiting for async write callback 2898 if (mType == OFFLOAD) { 2899 for (size_t i = 0; i < effectChains.size(); i ++) { 2900 effectChains[i]->process_l(); 2901 } 2902 } 2903 2904 // Only if the Effects buffer is enabled and there is data in the 2905 // Effects buffer (buffer valid), we need to 2906 // copy into the sink buffer. 2907 // TODO use mSleepTimeUs == 0 as an additional condition. 2908 if (mEffectBufferValid) { 2909 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2910 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2911 mNormalFrameCount * mChannelCount); 2912 } 2913 2914 // enable changes in effect chain 2915 unlockEffectChains(effectChains); 2916 2917 if (!waitingAsyncCallback()) { 2918 // mSleepTimeUs == 0 means we must write to audio hardware 2919 if (mSleepTimeUs == 0) { 2920 ssize_t ret = 0; 2921 if (mBytesRemaining) { 2922 ret = threadLoop_write(); 2923 if (ret < 0) { 2924 mBytesRemaining = 0; 2925 } else { 2926 mBytesWritten += ret; 2927 mBytesRemaining -= ret; 2928 } 2929 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2930 (mMixerStatus == MIXER_DRAIN_ALL)) { 2931 threadLoop_drain(); 2932 } 2933 if (mType == MIXER && !mStandby) { 2934 // write blocked detection 2935 nsecs_t now = systemTime(); 2936 nsecs_t delta = now - mLastWriteTime; 2937 if (delta > maxPeriod) { 2938 mNumDelayedWrites++; 2939 if ((now - lastWarning) > kWarningThrottleNs) { 2940 ATRACE_NAME("underrun"); 2941 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2942 ns2ms(delta), mNumDelayedWrites, this); 2943 lastWarning = now; 2944 } 2945 } 2946 2947 if (mThreadThrottle 2948 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 2949 && ret > 0) { // we wrote something 2950 // Limit MixerThread data processing to no more than twice the 2951 // expected processing rate. 2952 // 2953 // This helps prevent underruns with NuPlayer and other applications 2954 // which may set up buffers that are close to the minimum size, or use 2955 // deep buffers, and rely on a double-buffering sleep strategy to fill. 2956 // 2957 // The throttle smooths out sudden large data drains from the device, 2958 // e.g. when it comes out of standby, which often causes problems with 2959 // (1) mixer threads without a fast mixer (which has its own warm-up) 2960 // (2) minimum buffer sized tracks (even if the track is full, 2961 // the app won't fill fast enough to handle the sudden draw). 2962 2963 const int32_t deltaMs = delta / 1000000; 2964 const int32_t throttleMs = mHalfBufferMs - deltaMs; 2965 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 2966 usleep(throttleMs * 1000); 2967 // notify of throttle start on verbose log 2968 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 2969 "mixer(%p) throttle begin:" 2970 " ret(%zd) deltaMs(%d) requires sleep %d ms", 2971 this, ret, deltaMs, throttleMs); 2972 mThreadThrottleTimeMs += throttleMs; 2973 } else { 2974 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 2975 if (diff > 0) { 2976 // notify of throttle end on debug log 2977 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 2978 mThreadThrottleEndMs = mThreadThrottleTimeMs; 2979 } 2980 } 2981 } 2982 } 2983 2984 } else { 2985 ATRACE_BEGIN("sleep"); 2986 usleep(mSleepTimeUs); 2987 ATRACE_END(); 2988 } 2989 } 2990 2991 // Finally let go of removed track(s), without the lock held 2992 // since we can't guarantee the destructors won't acquire that 2993 // same lock. This will also mutate and push a new fast mixer state. 2994 threadLoop_removeTracks(tracksToRemove); 2995 tracksToRemove.clear(); 2996 2997 // FIXME I don't understand the need for this here; 2998 // it was in the original code but maybe the 2999 // assignment in saveOutputTracks() makes this unnecessary? 3000 clearOutputTracks(); 3001 3002 // Effect chains will be actually deleted here if they were removed from 3003 // mEffectChains list during mixing or effects processing 3004 effectChains.clear(); 3005 3006 // FIXME Note that the above .clear() is no longer necessary since effectChains 3007 // is now local to this block, but will keep it for now (at least until merge done). 3008 } 3009 3010 threadLoop_exit(); 3011 3012 if (!mStandby) { 3013 threadLoop_standby(); 3014 mStandby = true; 3015 } 3016 3017 releaseWakeLock(); 3018 mWakeLockUids.clear(); 3019 mActiveTracksGeneration++; 3020 3021 ALOGV("Thread %p type %d exiting", this, mType); 3022 return false; 3023} 3024 3025// removeTracks_l() must be called with ThreadBase::mLock held 3026void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3027{ 3028 size_t count = tracksToRemove.size(); 3029 if (count > 0) { 3030 for (size_t i=0 ; i<count ; i++) { 3031 const sp<Track>& track = tracksToRemove.itemAt(i); 3032 mActiveTracks.remove(track); 3033 mWakeLockUids.remove(track->uid()); 3034 mActiveTracksGeneration++; 3035 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3036 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3037 if (chain != 0) { 3038 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3039 track->sessionId()); 3040 chain->decActiveTrackCnt(); 3041 } 3042 if (track->isTerminated()) { 3043 removeTrack_l(track); 3044 } 3045 } 3046 } 3047 3048} 3049 3050status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3051{ 3052 if (mNormalSink != 0) { 3053 return mNormalSink->getTimestamp(timestamp); 3054 } 3055 if ((mType == OFFLOAD || mType == DIRECT) 3056 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3057 uint64_t position64; 3058 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3059 if (ret == 0) { 3060 timestamp.mPosition = (uint32_t)position64; 3061 return NO_ERROR; 3062 } 3063 } 3064 return INVALID_OPERATION; 3065} 3066 3067status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3068 audio_patch_handle_t *handle) 3069{ 3070 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3071 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3072 if (mFastMixer != 0) { 3073 FastMixerStateQueue *sq = mFastMixer->sq(); 3074 FastMixerState *state = sq->begin(); 3075 if (!(state->mCommand & FastMixerState::IDLE)) { 3076 previousCommand = state->mCommand; 3077 state->mCommand = FastMixerState::HOT_IDLE; 3078 sq->end(); 3079 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3080 } else { 3081 sq->end(false /*didModify*/); 3082 } 3083 } 3084 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3085 3086 if (!(previousCommand & FastMixerState::IDLE)) { 3087 ALOG_ASSERT(mFastMixer != 0); 3088 FastMixerStateQueue *sq = mFastMixer->sq(); 3089 FastMixerState *state = sq->begin(); 3090 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3091 state->mCommand = previousCommand; 3092 sq->end(); 3093 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3094 } 3095 3096 return status; 3097} 3098 3099status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3100 audio_patch_handle_t *handle) 3101{ 3102 status_t status = NO_ERROR; 3103 3104 // store new device and send to effects 3105 audio_devices_t type = AUDIO_DEVICE_NONE; 3106 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3107 type |= patch->sinks[i].ext.device.type; 3108 } 3109 3110#ifdef ADD_BATTERY_DATA 3111 // when changing the audio output device, call addBatteryData to notify 3112 // the change 3113 if (mOutDevice != type) { 3114 uint32_t params = 0; 3115 // check whether speaker is on 3116 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3117 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3118 } 3119 3120 audio_devices_t deviceWithoutSpeaker 3121 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3122 // check if any other device (except speaker) is on 3123 if (type & deviceWithoutSpeaker) { 3124 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3125 } 3126 3127 if (params != 0) { 3128 addBatteryData(params); 3129 } 3130 } 3131#endif 3132 3133 for (size_t i = 0; i < mEffectChains.size(); i++) { 3134 mEffectChains[i]->setDevice_l(type); 3135 } 3136 bool configChanged = mOutDevice != type; 3137 mOutDevice = type; 3138 mPatch = *patch; 3139 3140 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3141 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3142 status = hwDevice->create_audio_patch(hwDevice, 3143 patch->num_sources, 3144 patch->sources, 3145 patch->num_sinks, 3146 patch->sinks, 3147 handle); 3148 } else { 3149 char *address; 3150 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3151 //FIXME: we only support address on first sink with HAL version < 3.0 3152 address = audio_device_address_to_parameter( 3153 patch->sinks[0].ext.device.type, 3154 patch->sinks[0].ext.device.address); 3155 } else { 3156 address = (char *)calloc(1, 1); 3157 } 3158 AudioParameter param = AudioParameter(String8(address)); 3159 free(address); 3160 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3161 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3162 param.toString().string()); 3163 *handle = AUDIO_PATCH_HANDLE_NONE; 3164 } 3165 if (configChanged) { 3166 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3167 } 3168 return status; 3169} 3170 3171status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3172{ 3173 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3174 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3175 if (mFastMixer != 0) { 3176 FastMixerStateQueue *sq = mFastMixer->sq(); 3177 FastMixerState *state = sq->begin(); 3178 if (!(state->mCommand & FastMixerState::IDLE)) { 3179 previousCommand = state->mCommand; 3180 state->mCommand = FastMixerState::HOT_IDLE; 3181 sq->end(); 3182 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3183 } else { 3184 sq->end(false /*didModify*/); 3185 } 3186 } 3187 3188 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3189 3190 if (!(previousCommand & FastMixerState::IDLE)) { 3191 ALOG_ASSERT(mFastMixer != 0); 3192 FastMixerStateQueue *sq = mFastMixer->sq(); 3193 FastMixerState *state = sq->begin(); 3194 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3195 state->mCommand = previousCommand; 3196 sq->end(); 3197 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3198 } 3199 3200 return status; 3201} 3202 3203status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3204{ 3205 status_t status = NO_ERROR; 3206 3207 mOutDevice = AUDIO_DEVICE_NONE; 3208 3209 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3210 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3211 status = hwDevice->release_audio_patch(hwDevice, handle); 3212 } else { 3213 AudioParameter param; 3214 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3215 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3216 param.toString().string()); 3217 } 3218 return status; 3219} 3220 3221void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3222{ 3223 Mutex::Autolock _l(mLock); 3224 mTracks.add(track); 3225} 3226 3227void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3228{ 3229 Mutex::Autolock _l(mLock); 3230 destroyTrack_l(track); 3231} 3232 3233void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3234{ 3235 ThreadBase::getAudioPortConfig(config); 3236 config->role = AUDIO_PORT_ROLE_SOURCE; 3237 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3238 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3239} 3240 3241// ---------------------------------------------------------------------------- 3242 3243AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3244 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3245 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3246 // mAudioMixer below 3247 // mFastMixer below 3248 mFastMixerFutex(0) 3249 // mOutputSink below 3250 // mPipeSink below 3251 // mNormalSink below 3252{ 3253 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3254 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3255 "mFrameCount=%d, mNormalFrameCount=%d", 3256 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3257 mNormalFrameCount); 3258 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3259 3260 if (type == DUPLICATING) { 3261 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3262 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3263 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3264 return; 3265 } 3266 // create an NBAIO sink for the HAL output stream, and negotiate 3267 mOutputSink = new AudioStreamOutSink(output->stream); 3268 size_t numCounterOffers = 0; 3269 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3270 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3271 ALOG_ASSERT(index == 0); 3272 3273 // initialize fast mixer depending on configuration 3274 bool initFastMixer; 3275 switch (kUseFastMixer) { 3276 case FastMixer_Never: 3277 initFastMixer = false; 3278 break; 3279 case FastMixer_Always: 3280 initFastMixer = true; 3281 break; 3282 case FastMixer_Static: 3283 case FastMixer_Dynamic: 3284 initFastMixer = mFrameCount < mNormalFrameCount; 3285 break; 3286 } 3287 if (initFastMixer) { 3288 audio_format_t fastMixerFormat; 3289 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3290 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3291 } else { 3292 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3293 } 3294 if (mFormat != fastMixerFormat) { 3295 // change our Sink format to accept our intermediate precision 3296 mFormat = fastMixerFormat; 3297 free(mSinkBuffer); 3298 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3299 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3300 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3301 } 3302 3303 // create a MonoPipe to connect our submix to FastMixer 3304 NBAIO_Format format = mOutputSink->format(); 3305 NBAIO_Format origformat = format; 3306 // adjust format to match that of the Fast Mixer 3307 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3308 format.mFormat = fastMixerFormat; 3309 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3310 3311 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3312 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3313 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3314 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3315 const NBAIO_Format offers[1] = {format}; 3316 size_t numCounterOffers = 0; 3317 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3318 ALOG_ASSERT(index == 0); 3319 monoPipe->setAvgFrames((mScreenState & 1) ? 3320 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3321 mPipeSink = monoPipe; 3322 3323#ifdef TEE_SINK 3324 if (mTeeSinkOutputEnabled) { 3325 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3326 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3327 const NBAIO_Format offers2[1] = {origformat}; 3328 numCounterOffers = 0; 3329 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3330 ALOG_ASSERT(index == 0); 3331 mTeeSink = teeSink; 3332 PipeReader *teeSource = new PipeReader(*teeSink); 3333 numCounterOffers = 0; 3334 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3335 ALOG_ASSERT(index == 0); 3336 mTeeSource = teeSource; 3337 } 3338#endif 3339 3340 // create fast mixer and configure it initially with just one fast track for our submix 3341 mFastMixer = new FastMixer(); 3342 FastMixerStateQueue *sq = mFastMixer->sq(); 3343#ifdef STATE_QUEUE_DUMP 3344 sq->setObserverDump(&mStateQueueObserverDump); 3345 sq->setMutatorDump(&mStateQueueMutatorDump); 3346#endif 3347 FastMixerState *state = sq->begin(); 3348 FastTrack *fastTrack = &state->mFastTracks[0]; 3349 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3350 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3351 fastTrack->mVolumeProvider = NULL; 3352 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3353 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3354 fastTrack->mGeneration++; 3355 state->mFastTracksGen++; 3356 state->mTrackMask = 1; 3357 // fast mixer will use the HAL output sink 3358 state->mOutputSink = mOutputSink.get(); 3359 state->mOutputSinkGen++; 3360 state->mFrameCount = mFrameCount; 3361 state->mCommand = FastMixerState::COLD_IDLE; 3362 // already done in constructor initialization list 3363 //mFastMixerFutex = 0; 3364 state->mColdFutexAddr = &mFastMixerFutex; 3365 state->mColdGen++; 3366 state->mDumpState = &mFastMixerDumpState; 3367#ifdef TEE_SINK 3368 state->mTeeSink = mTeeSink.get(); 3369#endif 3370 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3371 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3372 sq->end(); 3373 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3374 3375 // start the fast mixer 3376 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3377 pid_t tid = mFastMixer->getTid(); 3378 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3379 3380#ifdef AUDIO_WATCHDOG 3381 // create and start the watchdog 3382 mAudioWatchdog = new AudioWatchdog(); 3383 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3384 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3385 tid = mAudioWatchdog->getTid(); 3386 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3387#endif 3388 3389 } 3390 3391 switch (kUseFastMixer) { 3392 case FastMixer_Never: 3393 case FastMixer_Dynamic: 3394 mNormalSink = mOutputSink; 3395 break; 3396 case FastMixer_Always: 3397 mNormalSink = mPipeSink; 3398 break; 3399 case FastMixer_Static: 3400 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3401 break; 3402 } 3403} 3404 3405AudioFlinger::MixerThread::~MixerThread() 3406{ 3407 if (mFastMixer != 0) { 3408 FastMixerStateQueue *sq = mFastMixer->sq(); 3409 FastMixerState *state = sq->begin(); 3410 if (state->mCommand == FastMixerState::COLD_IDLE) { 3411 int32_t old = android_atomic_inc(&mFastMixerFutex); 3412 if (old == -1) { 3413 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3414 } 3415 } 3416 state->mCommand = FastMixerState::EXIT; 3417 sq->end(); 3418 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3419 mFastMixer->join(); 3420 // Though the fast mixer thread has exited, it's state queue is still valid. 3421 // We'll use that extract the final state which contains one remaining fast track 3422 // corresponding to our sub-mix. 3423 state = sq->begin(); 3424 ALOG_ASSERT(state->mTrackMask == 1); 3425 FastTrack *fastTrack = &state->mFastTracks[0]; 3426 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3427 delete fastTrack->mBufferProvider; 3428 sq->end(false /*didModify*/); 3429 mFastMixer.clear(); 3430#ifdef AUDIO_WATCHDOG 3431 if (mAudioWatchdog != 0) { 3432 mAudioWatchdog->requestExit(); 3433 mAudioWatchdog->requestExitAndWait(); 3434 mAudioWatchdog.clear(); 3435 } 3436#endif 3437 } 3438 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3439 delete mAudioMixer; 3440} 3441 3442 3443uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3444{ 3445 if (mFastMixer != 0) { 3446 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3447 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3448 } 3449 return latency; 3450} 3451 3452 3453void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3454{ 3455 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3456} 3457 3458ssize_t AudioFlinger::MixerThread::threadLoop_write() 3459{ 3460 // FIXME we should only do one push per cycle; confirm this is true 3461 // Start the fast mixer if it's not already running 3462 if (mFastMixer != 0) { 3463 FastMixerStateQueue *sq = mFastMixer->sq(); 3464 FastMixerState *state = sq->begin(); 3465 if (state->mCommand != FastMixerState::MIX_WRITE && 3466 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3467 if (state->mCommand == FastMixerState::COLD_IDLE) { 3468 int32_t old = android_atomic_inc(&mFastMixerFutex); 3469 if (old == -1) { 3470 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3471 } 3472#ifdef AUDIO_WATCHDOG 3473 if (mAudioWatchdog != 0) { 3474 mAudioWatchdog->resume(); 3475 } 3476#endif 3477 } 3478 state->mCommand = FastMixerState::MIX_WRITE; 3479#ifdef FAST_THREAD_STATISTICS 3480 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3481 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3482#endif 3483 sq->end(); 3484 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3485 if (kUseFastMixer == FastMixer_Dynamic) { 3486 mNormalSink = mPipeSink; 3487 } 3488 } else { 3489 sq->end(false /*didModify*/); 3490 } 3491 } 3492 return PlaybackThread::threadLoop_write(); 3493} 3494 3495void AudioFlinger::MixerThread::threadLoop_standby() 3496{ 3497 // Idle the fast mixer if it's currently running 3498 if (mFastMixer != 0) { 3499 FastMixerStateQueue *sq = mFastMixer->sq(); 3500 FastMixerState *state = sq->begin(); 3501 if (!(state->mCommand & FastMixerState::IDLE)) { 3502 state->mCommand = FastMixerState::COLD_IDLE; 3503 state->mColdFutexAddr = &mFastMixerFutex; 3504 state->mColdGen++; 3505 mFastMixerFutex = 0; 3506 sq->end(); 3507 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3508 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3509 if (kUseFastMixer == FastMixer_Dynamic) { 3510 mNormalSink = mOutputSink; 3511 } 3512#ifdef AUDIO_WATCHDOG 3513 if (mAudioWatchdog != 0) { 3514 mAudioWatchdog->pause(); 3515 } 3516#endif 3517 } else { 3518 sq->end(false /*didModify*/); 3519 } 3520 } 3521 PlaybackThread::threadLoop_standby(); 3522} 3523 3524bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3525{ 3526 return false; 3527} 3528 3529bool AudioFlinger::PlaybackThread::shouldStandby_l() 3530{ 3531 return !mStandby; 3532} 3533 3534bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3535{ 3536 Mutex::Autolock _l(mLock); 3537 return waitingAsyncCallback_l(); 3538} 3539 3540// shared by MIXER and DIRECT, overridden by DUPLICATING 3541void AudioFlinger::PlaybackThread::threadLoop_standby() 3542{ 3543 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3544 mOutput->standby(); 3545 if (mUseAsyncWrite != 0) { 3546 // discard any pending drain or write ack by incrementing sequence 3547 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3548 mDrainSequence = (mDrainSequence + 2) & ~1; 3549 ALOG_ASSERT(mCallbackThread != 0); 3550 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3551 mCallbackThread->setDraining(mDrainSequence); 3552 } 3553 mHwPaused = false; 3554} 3555 3556void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3557{ 3558 ALOGV("signal playback thread"); 3559 broadcast_l(); 3560} 3561 3562void AudioFlinger::MixerThread::threadLoop_mix() 3563{ 3564 // obtain the presentation timestamp of the next output buffer 3565 int64_t pts; 3566 status_t status = INVALID_OPERATION; 3567 3568 if (mNormalSink != 0) { 3569 status = mNormalSink->getNextWriteTimestamp(&pts); 3570 } else { 3571 status = mOutputSink->getNextWriteTimestamp(&pts); 3572 } 3573 3574 if (status != NO_ERROR) { 3575 pts = AudioBufferProvider::kInvalidPTS; 3576 } 3577 3578 // mix buffers... 3579 mAudioMixer->process(pts); 3580 mCurrentWriteLength = mSinkBufferSize; 3581 // increase sleep time progressively when application underrun condition clears. 3582 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3583 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3584 // such that we would underrun the audio HAL. 3585 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3586 sleepTimeShift--; 3587 } 3588 mSleepTimeUs = 0; 3589 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3590 //TODO: delay standby when effects have a tail 3591 3592} 3593 3594void AudioFlinger::MixerThread::threadLoop_sleepTime() 3595{ 3596 // If no tracks are ready, sleep once for the duration of an output 3597 // buffer size, then write 0s to the output 3598 if (mSleepTimeUs == 0) { 3599 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3600 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3601 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3602 mSleepTimeUs = kMinThreadSleepTimeUs; 3603 } 3604 // reduce sleep time in case of consecutive application underruns to avoid 3605 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3606 // duration we would end up writing less data than needed by the audio HAL if 3607 // the condition persists. 3608 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3609 sleepTimeShift++; 3610 } 3611 } else { 3612 mSleepTimeUs = mIdleSleepTimeUs; 3613 } 3614 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3615 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3616 // before effects processing or output. 3617 if (mMixerBufferValid) { 3618 memset(mMixerBuffer, 0, mMixerBufferSize); 3619 } else { 3620 memset(mSinkBuffer, 0, mSinkBufferSize); 3621 } 3622 mSleepTimeUs = 0; 3623 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3624 "anticipated start"); 3625 } 3626 // TODO add standby time extension fct of effect tail 3627} 3628 3629// prepareTracks_l() must be called with ThreadBase::mLock held 3630AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3631 Vector< sp<Track> > *tracksToRemove) 3632{ 3633 3634 mixer_state mixerStatus = MIXER_IDLE; 3635 // find out which tracks need to be processed 3636 size_t count = mActiveTracks.size(); 3637 size_t mixedTracks = 0; 3638 size_t tracksWithEffect = 0; 3639 // counts only _active_ fast tracks 3640 size_t fastTracks = 0; 3641 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3642 3643 float masterVolume = mMasterVolume; 3644 bool masterMute = mMasterMute; 3645 3646 if (masterMute) { 3647 masterVolume = 0; 3648 } 3649 // Delegate master volume control to effect in output mix effect chain if needed 3650 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3651 if (chain != 0) { 3652 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3653 chain->setVolume_l(&v, &v); 3654 masterVolume = (float)((v + (1 << 23)) >> 24); 3655 chain.clear(); 3656 } 3657 3658 // prepare a new state to push 3659 FastMixerStateQueue *sq = NULL; 3660 FastMixerState *state = NULL; 3661 bool didModify = false; 3662 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3663 if (mFastMixer != 0) { 3664 sq = mFastMixer->sq(); 3665 state = sq->begin(); 3666 } 3667 3668 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3669 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3670 3671 for (size_t i=0 ; i<count ; i++) { 3672 const sp<Track> t = mActiveTracks[i].promote(); 3673 if (t == 0) { 3674 continue; 3675 } 3676 3677 // this const just means the local variable doesn't change 3678 Track* const track = t.get(); 3679 3680 // process fast tracks 3681 if (track->isFastTrack()) { 3682 3683 // It's theoretically possible (though unlikely) for a fast track to be created 3684 // and then removed within the same normal mix cycle. This is not a problem, as 3685 // the track never becomes active so it's fast mixer slot is never touched. 3686 // The converse, of removing an (active) track and then creating a new track 3687 // at the identical fast mixer slot within the same normal mix cycle, 3688 // is impossible because the slot isn't marked available until the end of each cycle. 3689 int j = track->mFastIndex; 3690 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3691 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3692 FastTrack *fastTrack = &state->mFastTracks[j]; 3693 3694 // Determine whether the track is currently in underrun condition, 3695 // and whether it had a recent underrun. 3696 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3697 FastTrackUnderruns underruns = ftDump->mUnderruns; 3698 uint32_t recentFull = (underruns.mBitFields.mFull - 3699 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3700 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3701 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3702 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3703 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3704 uint32_t recentUnderruns = recentPartial + recentEmpty; 3705 track->mObservedUnderruns = underruns; 3706 // don't count underruns that occur while stopping or pausing 3707 // or stopped which can occur when flush() is called while active 3708 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3709 recentUnderruns > 0) { 3710 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3711 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3712 } 3713 3714 // This is similar to the state machine for normal tracks, 3715 // with a few modifications for fast tracks. 3716 bool isActive = true; 3717 switch (track->mState) { 3718 case TrackBase::STOPPING_1: 3719 // track stays active in STOPPING_1 state until first underrun 3720 if (recentUnderruns > 0 || track->isTerminated()) { 3721 track->mState = TrackBase::STOPPING_2; 3722 } 3723 break; 3724 case TrackBase::PAUSING: 3725 // ramp down is not yet implemented 3726 track->setPaused(); 3727 break; 3728 case TrackBase::RESUMING: 3729 // ramp up is not yet implemented 3730 track->mState = TrackBase::ACTIVE; 3731 break; 3732 case TrackBase::ACTIVE: 3733 if (recentFull > 0 || recentPartial > 0) { 3734 // track has provided at least some frames recently: reset retry count 3735 track->mRetryCount = kMaxTrackRetries; 3736 } 3737 if (recentUnderruns == 0) { 3738 // no recent underruns: stay active 3739 break; 3740 } 3741 // there has recently been an underrun of some kind 3742 if (track->sharedBuffer() == 0) { 3743 // were any of the recent underruns "empty" (no frames available)? 3744 if (recentEmpty == 0) { 3745 // no, then ignore the partial underruns as they are allowed indefinitely 3746 break; 3747 } 3748 // there has recently been an "empty" underrun: decrement the retry counter 3749 if (--(track->mRetryCount) > 0) { 3750 break; 3751 } 3752 // indicate to client process that the track was disabled because of underrun; 3753 // it will then automatically call start() when data is available 3754 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3755 // remove from active list, but state remains ACTIVE [confusing but true] 3756 isActive = false; 3757 break; 3758 } 3759 // fall through 3760 case TrackBase::STOPPING_2: 3761 case TrackBase::PAUSED: 3762 case TrackBase::STOPPED: 3763 case TrackBase::FLUSHED: // flush() while active 3764 // Check for presentation complete if track is inactive 3765 // We have consumed all the buffers of this track. 3766 // This would be incomplete if we auto-paused on underrun 3767 { 3768 size_t audioHALFrames = 3769 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3770 size_t framesWritten = mBytesWritten / mFrameSize; 3771 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3772 // track stays in active list until presentation is complete 3773 break; 3774 } 3775 } 3776 if (track->isStopping_2()) { 3777 track->mState = TrackBase::STOPPED; 3778 } 3779 if (track->isStopped()) { 3780 // Can't reset directly, as fast mixer is still polling this track 3781 // track->reset(); 3782 // So instead mark this track as needing to be reset after push with ack 3783 resetMask |= 1 << i; 3784 } 3785 isActive = false; 3786 break; 3787 case TrackBase::IDLE: 3788 default: 3789 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3790 } 3791 3792 if (isActive) { 3793 // was it previously inactive? 3794 if (!(state->mTrackMask & (1 << j))) { 3795 ExtendedAudioBufferProvider *eabp = track; 3796 VolumeProvider *vp = track; 3797 fastTrack->mBufferProvider = eabp; 3798 fastTrack->mVolumeProvider = vp; 3799 fastTrack->mChannelMask = track->mChannelMask; 3800 fastTrack->mFormat = track->mFormat; 3801 fastTrack->mGeneration++; 3802 state->mTrackMask |= 1 << j; 3803 didModify = true; 3804 // no acknowledgement required for newly active tracks 3805 } 3806 // cache the combined master volume and stream type volume for fast mixer; this 3807 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3808 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3809 ++fastTracks; 3810 } else { 3811 // was it previously active? 3812 if (state->mTrackMask & (1 << j)) { 3813 fastTrack->mBufferProvider = NULL; 3814 fastTrack->mGeneration++; 3815 state->mTrackMask &= ~(1 << j); 3816 didModify = true; 3817 // If any fast tracks were removed, we must wait for acknowledgement 3818 // because we're about to decrement the last sp<> on those tracks. 3819 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3820 } else { 3821 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3822 } 3823 tracksToRemove->add(track); 3824 // Avoids a misleading display in dumpsys 3825 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3826 } 3827 continue; 3828 } 3829 3830 { // local variable scope to avoid goto warning 3831 3832 audio_track_cblk_t* cblk = track->cblk(); 3833 3834 // The first time a track is added we wait 3835 // for all its buffers to be filled before processing it 3836 int name = track->name(); 3837 // make sure that we have enough frames to mix one full buffer. 3838 // enforce this condition only once to enable draining the buffer in case the client 3839 // app does not call stop() and relies on underrun to stop: 3840 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3841 // during last round 3842 size_t desiredFrames; 3843 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3844 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3845 3846 desiredFrames = sourceFramesNeededWithTimestretch( 3847 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3848 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3849 // add frames already consumed but not yet released by the resampler 3850 // because mAudioTrackServerProxy->framesReady() will include these frames 3851 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3852 3853 uint32_t minFrames = 1; 3854 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3855 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3856 minFrames = desiredFrames; 3857 } 3858 3859 size_t framesReady = track->framesReady(); 3860 if (ATRACE_ENABLED()) { 3861 // I wish we had formatted trace names 3862 char traceName[16]; 3863 strcpy(traceName, "nRdy"); 3864 int name = track->name(); 3865 if (AudioMixer::TRACK0 <= name && 3866 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3867 name -= AudioMixer::TRACK0; 3868 traceName[4] = (name / 10) + '0'; 3869 traceName[5] = (name % 10) + '0'; 3870 } else { 3871 traceName[4] = '?'; 3872 traceName[5] = '?'; 3873 } 3874 traceName[6] = '\0'; 3875 ATRACE_INT(traceName, framesReady); 3876 } 3877 if ((framesReady >= minFrames) && track->isReady() && 3878 !track->isPaused() && !track->isTerminated()) 3879 { 3880 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3881 3882 mixedTracks++; 3883 3884 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3885 // there is an effect chain connected to the track 3886 chain.clear(); 3887 if (track->mainBuffer() != mSinkBuffer && 3888 track->mainBuffer() != mMixerBuffer) { 3889 if (mEffectBufferEnabled) { 3890 mEffectBufferValid = true; // Later can set directly. 3891 } 3892 chain = getEffectChain_l(track->sessionId()); 3893 // Delegate volume control to effect in track effect chain if needed 3894 if (chain != 0) { 3895 tracksWithEffect++; 3896 } else { 3897 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3898 "session %d", 3899 name, track->sessionId()); 3900 } 3901 } 3902 3903 3904 int param = AudioMixer::VOLUME; 3905 if (track->mFillingUpStatus == Track::FS_FILLED) { 3906 // no ramp for the first volume setting 3907 track->mFillingUpStatus = Track::FS_ACTIVE; 3908 if (track->mState == TrackBase::RESUMING) { 3909 track->mState = TrackBase::ACTIVE; 3910 param = AudioMixer::RAMP_VOLUME; 3911 } 3912 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3913 // FIXME should not make a decision based on mServer 3914 } else if (cblk->mServer != 0) { 3915 // If the track is stopped before the first frame was mixed, 3916 // do not apply ramp 3917 param = AudioMixer::RAMP_VOLUME; 3918 } 3919 3920 // compute volume for this track 3921 uint32_t vl, vr; // in U8.24 integer format 3922 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3923 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3924 vl = vr = 0; 3925 vlf = vrf = vaf = 0.; 3926 if (track->isPausing()) { 3927 track->setPaused(); 3928 } 3929 } else { 3930 3931 // read original volumes with volume control 3932 float typeVolume = mStreamTypes[track->streamType()].volume; 3933 float v = masterVolume * typeVolume; 3934 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3935 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3936 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3937 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3938 // track volumes come from shared memory, so can't be trusted and must be clamped 3939 if (vlf > GAIN_FLOAT_UNITY) { 3940 ALOGV("Track left volume out of range: %.3g", vlf); 3941 vlf = GAIN_FLOAT_UNITY; 3942 } 3943 if (vrf > GAIN_FLOAT_UNITY) { 3944 ALOGV("Track right volume out of range: %.3g", vrf); 3945 vrf = GAIN_FLOAT_UNITY; 3946 } 3947 // now apply the master volume and stream type volume 3948 vlf *= v; 3949 vrf *= v; 3950 // assuming master volume and stream type volume each go up to 1.0, 3951 // then derive vl and vr as U8.24 versions for the effect chain 3952 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3953 vl = (uint32_t) (scaleto8_24 * vlf); 3954 vr = (uint32_t) (scaleto8_24 * vrf); 3955 // vl and vr are now in U8.24 format 3956 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3957 // send level comes from shared memory and so may be corrupt 3958 if (sendLevel > MAX_GAIN_INT) { 3959 ALOGV("Track send level out of range: %04X", sendLevel); 3960 sendLevel = MAX_GAIN_INT; 3961 } 3962 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3963 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3964 } 3965 3966 // Delegate volume control to effect in track effect chain if needed 3967 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3968 // Do not ramp volume if volume is controlled by effect 3969 param = AudioMixer::VOLUME; 3970 // Update remaining floating point volume levels 3971 vlf = (float)vl / (1 << 24); 3972 vrf = (float)vr / (1 << 24); 3973 track->mHasVolumeController = true; 3974 } else { 3975 // force no volume ramp when volume controller was just disabled or removed 3976 // from effect chain to avoid volume spike 3977 if (track->mHasVolumeController) { 3978 param = AudioMixer::VOLUME; 3979 } 3980 track->mHasVolumeController = false; 3981 } 3982 3983 // XXX: these things DON'T need to be done each time 3984 mAudioMixer->setBufferProvider(name, track); 3985 mAudioMixer->enable(name); 3986 3987 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3988 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3989 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3990 mAudioMixer->setParameter( 3991 name, 3992 AudioMixer::TRACK, 3993 AudioMixer::FORMAT, (void *)track->format()); 3994 mAudioMixer->setParameter( 3995 name, 3996 AudioMixer::TRACK, 3997 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3998 mAudioMixer->setParameter( 3999 name, 4000 AudioMixer::TRACK, 4001 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4002 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4003 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4004 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4005 if (reqSampleRate == 0) { 4006 reqSampleRate = mSampleRate; 4007 } else if (reqSampleRate > maxSampleRate) { 4008 reqSampleRate = maxSampleRate; 4009 } 4010 mAudioMixer->setParameter( 4011 name, 4012 AudioMixer::RESAMPLE, 4013 AudioMixer::SAMPLE_RATE, 4014 (void *)(uintptr_t)reqSampleRate); 4015 4016 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4017 mAudioMixer->setParameter( 4018 name, 4019 AudioMixer::TIMESTRETCH, 4020 AudioMixer::PLAYBACK_RATE, 4021 &playbackRate); 4022 4023 /* 4024 * Select the appropriate output buffer for the track. 4025 * 4026 * Tracks with effects go into their own effects chain buffer 4027 * and from there into either mEffectBuffer or mSinkBuffer. 4028 * 4029 * Other tracks can use mMixerBuffer for higher precision 4030 * channel accumulation. If this buffer is enabled 4031 * (mMixerBufferEnabled true), then selected tracks will accumulate 4032 * into it. 4033 * 4034 */ 4035 if (mMixerBufferEnabled 4036 && (track->mainBuffer() == mSinkBuffer 4037 || track->mainBuffer() == mMixerBuffer)) { 4038 mAudioMixer->setParameter( 4039 name, 4040 AudioMixer::TRACK, 4041 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4042 mAudioMixer->setParameter( 4043 name, 4044 AudioMixer::TRACK, 4045 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4046 // TODO: override track->mainBuffer()? 4047 mMixerBufferValid = true; 4048 } else { 4049 mAudioMixer->setParameter( 4050 name, 4051 AudioMixer::TRACK, 4052 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4053 mAudioMixer->setParameter( 4054 name, 4055 AudioMixer::TRACK, 4056 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4057 } 4058 mAudioMixer->setParameter( 4059 name, 4060 AudioMixer::TRACK, 4061 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4062 4063 // reset retry count 4064 track->mRetryCount = kMaxTrackRetries; 4065 4066 // If one track is ready, set the mixer ready if: 4067 // - the mixer was not ready during previous round OR 4068 // - no other track is not ready 4069 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4070 mixerStatus != MIXER_TRACKS_ENABLED) { 4071 mixerStatus = MIXER_TRACKS_READY; 4072 } 4073 } else { 4074 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4075 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4076 track, framesReady, desiredFrames); 4077 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4078 } 4079 // clear effect chain input buffer if an active track underruns to avoid sending 4080 // previous audio buffer again to effects 4081 chain = getEffectChain_l(track->sessionId()); 4082 if (chain != 0) { 4083 chain->clearInputBuffer(); 4084 } 4085 4086 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4087 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4088 track->isStopped() || track->isPaused()) { 4089 // We have consumed all the buffers of this track. 4090 // Remove it from the list of active tracks. 4091 // TODO: use actual buffer filling status instead of latency when available from 4092 // audio HAL 4093 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4094 size_t framesWritten = mBytesWritten / mFrameSize; 4095 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4096 if (track->isStopped()) { 4097 track->reset(); 4098 } 4099 tracksToRemove->add(track); 4100 } 4101 } else { 4102 // No buffers for this track. Give it a few chances to 4103 // fill a buffer, then remove it from active list. 4104 if (--(track->mRetryCount) <= 0) { 4105 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4106 tracksToRemove->add(track); 4107 // indicate to client process that the track was disabled because of underrun; 4108 // it will then automatically call start() when data is available 4109 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4110 // If one track is not ready, mark the mixer also not ready if: 4111 // - the mixer was ready during previous round OR 4112 // - no other track is ready 4113 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4114 mixerStatus != MIXER_TRACKS_READY) { 4115 mixerStatus = MIXER_TRACKS_ENABLED; 4116 } 4117 } 4118 mAudioMixer->disable(name); 4119 } 4120 4121 } // local variable scope to avoid goto warning 4122track_is_ready: ; 4123 4124 } 4125 4126 // Push the new FastMixer state if necessary 4127 bool pauseAudioWatchdog = false; 4128 if (didModify) { 4129 state->mFastTracksGen++; 4130 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4131 if (kUseFastMixer == FastMixer_Dynamic && 4132 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4133 state->mCommand = FastMixerState::COLD_IDLE; 4134 state->mColdFutexAddr = &mFastMixerFutex; 4135 state->mColdGen++; 4136 mFastMixerFutex = 0; 4137 if (kUseFastMixer == FastMixer_Dynamic) { 4138 mNormalSink = mOutputSink; 4139 } 4140 // If we go into cold idle, need to wait for acknowledgement 4141 // so that fast mixer stops doing I/O. 4142 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4143 pauseAudioWatchdog = true; 4144 } 4145 } 4146 if (sq != NULL) { 4147 sq->end(didModify); 4148 sq->push(block); 4149 } 4150#ifdef AUDIO_WATCHDOG 4151 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4152 mAudioWatchdog->pause(); 4153 } 4154#endif 4155 4156 // Now perform the deferred reset on fast tracks that have stopped 4157 while (resetMask != 0) { 4158 size_t i = __builtin_ctz(resetMask); 4159 ALOG_ASSERT(i < count); 4160 resetMask &= ~(1 << i); 4161 sp<Track> t = mActiveTracks[i].promote(); 4162 if (t == 0) { 4163 continue; 4164 } 4165 Track* track = t.get(); 4166 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4167 track->reset(); 4168 } 4169 4170 // remove all the tracks that need to be... 4171 removeTracks_l(*tracksToRemove); 4172 4173 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4174 mEffectBufferValid = true; 4175 } 4176 4177 if (mEffectBufferValid) { 4178 // as long as there are effects we should clear the effects buffer, to avoid 4179 // passing a non-clean buffer to the effect chain 4180 memset(mEffectBuffer, 0, mEffectBufferSize); 4181 } 4182 // sink or mix buffer must be cleared if all tracks are connected to an 4183 // effect chain as in this case the mixer will not write to the sink or mix buffer 4184 // and track effects will accumulate into it 4185 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4186 (mixedTracks == 0 && fastTracks > 0))) { 4187 // FIXME as a performance optimization, should remember previous zero status 4188 if (mMixerBufferValid) { 4189 memset(mMixerBuffer, 0, mMixerBufferSize); 4190 // TODO: In testing, mSinkBuffer below need not be cleared because 4191 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4192 // after mixing. 4193 // 4194 // To enforce this guarantee: 4195 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4196 // (mixedTracks == 0 && fastTracks > 0)) 4197 // must imply MIXER_TRACKS_READY. 4198 // Later, we may clear buffers regardless, and skip much of this logic. 4199 } 4200 // FIXME as a performance optimization, should remember previous zero status 4201 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4202 } 4203 4204 // if any fast tracks, then status is ready 4205 mMixerStatusIgnoringFastTracks = mixerStatus; 4206 if (fastTracks > 0) { 4207 mixerStatus = MIXER_TRACKS_READY; 4208 } 4209 return mixerStatus; 4210} 4211 4212// getTrackName_l() must be called with ThreadBase::mLock held 4213int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4214 audio_format_t format, int sessionId) 4215{ 4216 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4217} 4218 4219// deleteTrackName_l() must be called with ThreadBase::mLock held 4220void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4221{ 4222 ALOGV("remove track (%d) and delete from mixer", name); 4223 mAudioMixer->deleteTrackName(name); 4224} 4225 4226// checkForNewParameter_l() must be called with ThreadBase::mLock held 4227bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4228 status_t& status) 4229{ 4230 bool reconfig = false; 4231 4232 status = NO_ERROR; 4233 4234 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4235 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4236 if (mFastMixer != 0) { 4237 FastMixerStateQueue *sq = mFastMixer->sq(); 4238 FastMixerState *state = sq->begin(); 4239 if (!(state->mCommand & FastMixerState::IDLE)) { 4240 previousCommand = state->mCommand; 4241 state->mCommand = FastMixerState::HOT_IDLE; 4242 sq->end(); 4243 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4244 } else { 4245 sq->end(false /*didModify*/); 4246 } 4247 } 4248 4249 AudioParameter param = AudioParameter(keyValuePair); 4250 int value; 4251 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4252 reconfig = true; 4253 } 4254 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4255 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4256 status = BAD_VALUE; 4257 } else { 4258 // no need to save value, since it's constant 4259 reconfig = true; 4260 } 4261 } 4262 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4263 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4264 status = BAD_VALUE; 4265 } else { 4266 // no need to save value, since it's constant 4267 reconfig = true; 4268 } 4269 } 4270 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4271 // do not accept frame count changes if tracks are open as the track buffer 4272 // size depends on frame count and correct behavior would not be guaranteed 4273 // if frame count is changed after track creation 4274 if (!mTracks.isEmpty()) { 4275 status = INVALID_OPERATION; 4276 } else { 4277 reconfig = true; 4278 } 4279 } 4280 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4281#ifdef ADD_BATTERY_DATA 4282 // when changing the audio output device, call addBatteryData to notify 4283 // the change 4284 if (mOutDevice != value) { 4285 uint32_t params = 0; 4286 // check whether speaker is on 4287 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4288 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4289 } 4290 4291 audio_devices_t deviceWithoutSpeaker 4292 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4293 // check if any other device (except speaker) is on 4294 if (value & deviceWithoutSpeaker) { 4295 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4296 } 4297 4298 if (params != 0) { 4299 addBatteryData(params); 4300 } 4301 } 4302#endif 4303 4304 // forward device change to effects that have requested to be 4305 // aware of attached audio device. 4306 if (value != AUDIO_DEVICE_NONE) { 4307 mOutDevice = value; 4308 for (size_t i = 0; i < mEffectChains.size(); i++) { 4309 mEffectChains[i]->setDevice_l(mOutDevice); 4310 } 4311 } 4312 } 4313 4314 if (status == NO_ERROR) { 4315 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4316 keyValuePair.string()); 4317 if (!mStandby && status == INVALID_OPERATION) { 4318 mOutput->standby(); 4319 mStandby = true; 4320 mBytesWritten = 0; 4321 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4322 keyValuePair.string()); 4323 } 4324 if (status == NO_ERROR && reconfig) { 4325 readOutputParameters_l(); 4326 delete mAudioMixer; 4327 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4328 for (size_t i = 0; i < mTracks.size() ; i++) { 4329 int name = getTrackName_l(mTracks[i]->mChannelMask, 4330 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4331 if (name < 0) { 4332 break; 4333 } 4334 mTracks[i]->mName = name; 4335 } 4336 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4337 } 4338 } 4339 4340 if (!(previousCommand & FastMixerState::IDLE)) { 4341 ALOG_ASSERT(mFastMixer != 0); 4342 FastMixerStateQueue *sq = mFastMixer->sq(); 4343 FastMixerState *state = sq->begin(); 4344 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4345 state->mCommand = previousCommand; 4346 sq->end(); 4347 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4348 } 4349 4350 return reconfig; 4351} 4352 4353 4354void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4355{ 4356 const size_t SIZE = 256; 4357 char buffer[SIZE]; 4358 String8 result; 4359 4360 PlaybackThread::dumpInternals(fd, args); 4361 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4362 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4363 4364 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4365 const FastMixerDumpState copy(mFastMixerDumpState); 4366 copy.dump(fd); 4367 4368#ifdef STATE_QUEUE_DUMP 4369 // Similar for state queue 4370 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4371 observerCopy.dump(fd); 4372 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4373 mutatorCopy.dump(fd); 4374#endif 4375 4376#ifdef TEE_SINK 4377 // Write the tee output to a .wav file 4378 dumpTee(fd, mTeeSource, mId); 4379#endif 4380 4381#ifdef AUDIO_WATCHDOG 4382 if (mAudioWatchdog != 0) { 4383 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4384 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4385 wdCopy.dump(fd); 4386 } 4387#endif 4388} 4389 4390uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4391{ 4392 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4393} 4394 4395uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4396{ 4397 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4398} 4399 4400void AudioFlinger::MixerThread::cacheParameters_l() 4401{ 4402 PlaybackThread::cacheParameters_l(); 4403 4404 // FIXME: Relaxed timing because of a certain device that can't meet latency 4405 // Should be reduced to 2x after the vendor fixes the driver issue 4406 // increase threshold again due to low power audio mode. The way this warning 4407 // threshold is calculated and its usefulness should be reconsidered anyway. 4408 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4409} 4410 4411// ---------------------------------------------------------------------------- 4412 4413AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4414 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4415 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4416 // mLeftVolFloat, mRightVolFloat 4417{ 4418} 4419 4420AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4421 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4422 ThreadBase::type_t type, bool systemReady) 4423 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4424 // mLeftVolFloat, mRightVolFloat 4425{ 4426} 4427 4428AudioFlinger::DirectOutputThread::~DirectOutputThread() 4429{ 4430} 4431 4432void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4433{ 4434 audio_track_cblk_t* cblk = track->cblk(); 4435 float left, right; 4436 4437 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4438 left = right = 0; 4439 } else { 4440 float typeVolume = mStreamTypes[track->streamType()].volume; 4441 float v = mMasterVolume * typeVolume; 4442 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4443 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4444 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4445 if (left > GAIN_FLOAT_UNITY) { 4446 left = GAIN_FLOAT_UNITY; 4447 } 4448 left *= v; 4449 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4450 if (right > GAIN_FLOAT_UNITY) { 4451 right = GAIN_FLOAT_UNITY; 4452 } 4453 right *= v; 4454 } 4455 4456 if (lastTrack) { 4457 if (left != mLeftVolFloat || right != mRightVolFloat) { 4458 mLeftVolFloat = left; 4459 mRightVolFloat = right; 4460 4461 // Convert volumes from float to 8.24 4462 uint32_t vl = (uint32_t)(left * (1 << 24)); 4463 uint32_t vr = (uint32_t)(right * (1 << 24)); 4464 4465 // Delegate volume control to effect in track effect chain if needed 4466 // only one effect chain can be present on DirectOutputThread, so if 4467 // there is one, the track is connected to it 4468 if (!mEffectChains.isEmpty()) { 4469 mEffectChains[0]->setVolume_l(&vl, &vr); 4470 left = (float)vl / (1 << 24); 4471 right = (float)vr / (1 << 24); 4472 } 4473 if (mOutput->stream->set_volume) { 4474 mOutput->stream->set_volume(mOutput->stream, left, right); 4475 } 4476 } 4477 } 4478} 4479 4480void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4481{ 4482 sp<Track> previousTrack = mPreviousTrack.promote(); 4483 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4484 4485 if (previousTrack != 0 && latestTrack != 0) { 4486 if (mType == DIRECT) { 4487 if (previousTrack.get() != latestTrack.get()) { 4488 mFlushPending = true; 4489 } 4490 } else /* mType == OFFLOAD */ { 4491 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4492 mFlushPending = true; 4493 } 4494 } 4495 } 4496 PlaybackThread::onAddNewTrack_l(); 4497} 4498 4499AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4500 Vector< sp<Track> > *tracksToRemove 4501) 4502{ 4503 size_t count = mActiveTracks.size(); 4504 mixer_state mixerStatus = MIXER_IDLE; 4505 bool doHwPause = false; 4506 bool doHwResume = false; 4507 4508 // find out which tracks need to be processed 4509 for (size_t i = 0; i < count; i++) { 4510 sp<Track> t = mActiveTracks[i].promote(); 4511 // The track died recently 4512 if (t == 0) { 4513 continue; 4514 } 4515 4516 if (t->isInvalid()) { 4517 ALOGW("An invalidated track shouldn't be in active list"); 4518 tracksToRemove->add(t); 4519 continue; 4520 } 4521 4522 Track* const track = t.get(); 4523 audio_track_cblk_t* cblk = track->cblk(); 4524 // Only consider last track started for volume and mixer state control. 4525 // In theory an older track could underrun and restart after the new one starts 4526 // but as we only care about the transition phase between two tracks on a 4527 // direct output, it is not a problem to ignore the underrun case. 4528 sp<Track> l = mLatestActiveTrack.promote(); 4529 bool last = l.get() == track; 4530 4531 if (track->isPausing()) { 4532 track->setPaused(); 4533 if (mHwSupportsPause && last && !mHwPaused) { 4534 doHwPause = true; 4535 mHwPaused = true; 4536 } 4537 tracksToRemove->add(track); 4538 } else if (track->isFlushPending()) { 4539 track->flushAck(); 4540 if (last) { 4541 mFlushPending = true; 4542 } 4543 } else if (track->isResumePending()) { 4544 track->resumeAck(); 4545 if (last && mHwPaused) { 4546 doHwResume = true; 4547 mHwPaused = false; 4548 } 4549 } 4550 4551 // The first time a track is added we wait 4552 // for all its buffers to be filled before processing it. 4553 // Allow draining the buffer in case the client 4554 // app does not call stop() and relies on underrun to stop: 4555 // hence the test on (track->mRetryCount > 1). 4556 // If retryCount<=1 then track is about to underrun and be removed. 4557 uint32_t minFrames; 4558 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4559 && (track->mRetryCount > 1)) { 4560 minFrames = mNormalFrameCount; 4561 } else { 4562 minFrames = 1; 4563 } 4564 4565 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4566 !track->isStopping_2() && !track->isStopped()) 4567 { 4568 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4569 4570 if (track->mFillingUpStatus == Track::FS_FILLED) { 4571 track->mFillingUpStatus = Track::FS_ACTIVE; 4572 // make sure processVolume_l() will apply new volume even if 0 4573 mLeftVolFloat = mRightVolFloat = -1.0; 4574 if (!mHwSupportsPause) { 4575 track->resumeAck(); 4576 } 4577 } 4578 4579 // compute volume for this track 4580 processVolume_l(track, last); 4581 if (last) { 4582 sp<Track> previousTrack = mPreviousTrack.promote(); 4583 if (previousTrack != 0) { 4584 if (track != previousTrack.get()) { 4585 // Flush any data still being written from last track 4586 mBytesRemaining = 0; 4587 // Invalidate previous track to force a seek when resuming. 4588 previousTrack->invalidate(); 4589 } 4590 } 4591 mPreviousTrack = track; 4592 4593 // reset retry count 4594 track->mRetryCount = kMaxTrackRetriesDirect; 4595 mActiveTrack = t; 4596 mixerStatus = MIXER_TRACKS_READY; 4597 if (mHwPaused) { 4598 doHwResume = true; 4599 mHwPaused = false; 4600 } 4601 } 4602 } else { 4603 // clear effect chain input buffer if the last active track started underruns 4604 // to avoid sending previous audio buffer again to effects 4605 if (!mEffectChains.isEmpty() && last) { 4606 mEffectChains[0]->clearInputBuffer(); 4607 } 4608 if (track->isStopping_1()) { 4609 track->mState = TrackBase::STOPPING_2; 4610 if (last && mHwPaused) { 4611 doHwResume = true; 4612 mHwPaused = false; 4613 } 4614 } 4615 if ((track->sharedBuffer() != 0) || track->isStopped() || 4616 track->isStopping_2() || track->isPaused()) { 4617 // We have consumed all the buffers of this track. 4618 // Remove it from the list of active tracks. 4619 size_t audioHALFrames; 4620 if (audio_is_linear_pcm(mFormat)) { 4621 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4622 } else { 4623 audioHALFrames = 0; 4624 } 4625 4626 size_t framesWritten = mBytesWritten / mFrameSize; 4627 if (mStandby || !last || 4628 track->presentationComplete(framesWritten, audioHALFrames)) { 4629 if (track->isStopping_2()) { 4630 track->mState = TrackBase::STOPPED; 4631 } 4632 if (track->isStopped()) { 4633 track->reset(); 4634 } 4635 tracksToRemove->add(track); 4636 } 4637 } else { 4638 // No buffers for this track. Give it a few chances to 4639 // fill a buffer, then remove it from active list. 4640 // Only consider last track started for mixer state control 4641 if (--(track->mRetryCount) <= 0) { 4642 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4643 tracksToRemove->add(track); 4644 // indicate to client process that the track was disabled because of underrun; 4645 // it will then automatically call start() when data is available 4646 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4647 } else if (last) { 4648 mixerStatus = MIXER_TRACKS_ENABLED; 4649 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4650 doHwPause = true; 4651 mHwPaused = true; 4652 } 4653 } 4654 } 4655 } 4656 } 4657 4658 // if an active track did not command a flush, check for pending flush on stopped tracks 4659 if (!mFlushPending) { 4660 for (size_t i = 0; i < mTracks.size(); i++) { 4661 if (mTracks[i]->isFlushPending()) { 4662 mTracks[i]->flushAck(); 4663 mFlushPending = true; 4664 } 4665 } 4666 } 4667 4668 // make sure the pause/flush/resume sequence is executed in the right order. 4669 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4670 // before flush and then resume HW. This can happen in case of pause/flush/resume 4671 // if resume is received before pause is executed. 4672 if (mHwSupportsPause && !mStandby && 4673 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4674 mOutput->stream->pause(mOutput->stream); 4675 } 4676 if (mFlushPending) { 4677 flushHw_l(); 4678 } 4679 if (mHwSupportsPause && !mStandby && doHwResume) { 4680 mOutput->stream->resume(mOutput->stream); 4681 } 4682 // remove all the tracks that need to be... 4683 removeTracks_l(*tracksToRemove); 4684 4685 return mixerStatus; 4686} 4687 4688void AudioFlinger::DirectOutputThread::threadLoop_mix() 4689{ 4690 size_t frameCount = mFrameCount; 4691 int8_t *curBuf = (int8_t *)mSinkBuffer; 4692 // output audio to hardware 4693 while (frameCount) { 4694 AudioBufferProvider::Buffer buffer; 4695 buffer.frameCount = frameCount; 4696 status_t status = mActiveTrack->getNextBuffer(&buffer); 4697 if (status != NO_ERROR || buffer.raw == NULL) { 4698 memset(curBuf, 0, frameCount * mFrameSize); 4699 break; 4700 } 4701 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4702 frameCount -= buffer.frameCount; 4703 curBuf += buffer.frameCount * mFrameSize; 4704 mActiveTrack->releaseBuffer(&buffer); 4705 } 4706 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4707 mSleepTimeUs = 0; 4708 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4709 mActiveTrack.clear(); 4710} 4711 4712void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4713{ 4714 // do not write to HAL when paused 4715 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4716 mSleepTimeUs = mIdleSleepTimeUs; 4717 return; 4718 } 4719 if (mSleepTimeUs == 0) { 4720 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4721 mSleepTimeUs = mActiveSleepTimeUs; 4722 } else { 4723 mSleepTimeUs = mIdleSleepTimeUs; 4724 } 4725 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4726 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4727 mSleepTimeUs = 0; 4728 } 4729} 4730 4731void AudioFlinger::DirectOutputThread::threadLoop_exit() 4732{ 4733 { 4734 Mutex::Autolock _l(mLock); 4735 for (size_t i = 0; i < mTracks.size(); i++) { 4736 if (mTracks[i]->isFlushPending()) { 4737 mTracks[i]->flushAck(); 4738 mFlushPending = true; 4739 } 4740 } 4741 if (mFlushPending) { 4742 flushHw_l(); 4743 } 4744 } 4745 PlaybackThread::threadLoop_exit(); 4746} 4747 4748// must be called with thread mutex locked 4749bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4750{ 4751 bool trackPaused = false; 4752 bool trackStopped = false; 4753 4754 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4755 // after a timeout and we will enter standby then. 4756 if (mTracks.size() > 0) { 4757 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4758 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4759 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4760 } 4761 4762 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4763} 4764 4765// getTrackName_l() must be called with ThreadBase::mLock held 4766int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4767 audio_format_t format __unused, int sessionId __unused) 4768{ 4769 return 0; 4770} 4771 4772// deleteTrackName_l() must be called with ThreadBase::mLock held 4773void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4774{ 4775} 4776 4777// checkForNewParameter_l() must be called with ThreadBase::mLock held 4778bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4779 status_t& status) 4780{ 4781 bool reconfig = false; 4782 4783 status = NO_ERROR; 4784 4785 AudioParameter param = AudioParameter(keyValuePair); 4786 int value; 4787 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4788 // forward device change to effects that have requested to be 4789 // aware of attached audio device. 4790 if (value != AUDIO_DEVICE_NONE) { 4791 mOutDevice = value; 4792 for (size_t i = 0; i < mEffectChains.size(); i++) { 4793 mEffectChains[i]->setDevice_l(mOutDevice); 4794 } 4795 } 4796 } 4797 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4798 // do not accept frame count changes if tracks are open as the track buffer 4799 // size depends on frame count and correct behavior would not be garantied 4800 // if frame count is changed after track creation 4801 if (!mTracks.isEmpty()) { 4802 status = INVALID_OPERATION; 4803 } else { 4804 reconfig = true; 4805 } 4806 } 4807 if (status == NO_ERROR) { 4808 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4809 keyValuePair.string()); 4810 if (!mStandby && status == INVALID_OPERATION) { 4811 mOutput->standby(); 4812 mStandby = true; 4813 mBytesWritten = 0; 4814 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4815 keyValuePair.string()); 4816 } 4817 if (status == NO_ERROR && reconfig) { 4818 readOutputParameters_l(); 4819 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4820 } 4821 } 4822 4823 return reconfig; 4824} 4825 4826uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4827{ 4828 uint32_t time; 4829 if (audio_is_linear_pcm(mFormat)) { 4830 time = PlaybackThread::activeSleepTimeUs(); 4831 } else { 4832 time = 10000; 4833 } 4834 return time; 4835} 4836 4837uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4838{ 4839 uint32_t time; 4840 if (audio_is_linear_pcm(mFormat)) { 4841 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4842 } else { 4843 time = 10000; 4844 } 4845 return time; 4846} 4847 4848uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4849{ 4850 uint32_t time; 4851 if (audio_is_linear_pcm(mFormat)) { 4852 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4853 } else { 4854 time = 10000; 4855 } 4856 return time; 4857} 4858 4859void AudioFlinger::DirectOutputThread::cacheParameters_l() 4860{ 4861 PlaybackThread::cacheParameters_l(); 4862 4863 // use shorter standby delay as on normal output to release 4864 // hardware resources as soon as possible 4865 // no delay on outputs with HW A/V sync 4866 if (usesHwAvSync()) { 4867 mStandbyDelayNs = 0; 4868 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 4869 mStandbyDelayNs = kOffloadStandbyDelayNs; 4870 } else { 4871 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 4872 } 4873} 4874 4875void AudioFlinger::DirectOutputThread::flushHw_l() 4876{ 4877 mOutput->flush(); 4878 mHwPaused = false; 4879 mFlushPending = false; 4880} 4881 4882// ---------------------------------------------------------------------------- 4883 4884AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4885 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4886 : Thread(false /*canCallJava*/), 4887 mPlaybackThread(playbackThread), 4888 mWriteAckSequence(0), 4889 mDrainSequence(0) 4890{ 4891} 4892 4893AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4894{ 4895} 4896 4897void AudioFlinger::AsyncCallbackThread::onFirstRef() 4898{ 4899 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4900} 4901 4902bool AudioFlinger::AsyncCallbackThread::threadLoop() 4903{ 4904 while (!exitPending()) { 4905 uint32_t writeAckSequence; 4906 uint32_t drainSequence; 4907 4908 { 4909 Mutex::Autolock _l(mLock); 4910 while (!((mWriteAckSequence & 1) || 4911 (mDrainSequence & 1) || 4912 exitPending())) { 4913 mWaitWorkCV.wait(mLock); 4914 } 4915 4916 if (exitPending()) { 4917 break; 4918 } 4919 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4920 mWriteAckSequence, mDrainSequence); 4921 writeAckSequence = mWriteAckSequence; 4922 mWriteAckSequence &= ~1; 4923 drainSequence = mDrainSequence; 4924 mDrainSequence &= ~1; 4925 } 4926 { 4927 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4928 if (playbackThread != 0) { 4929 if (writeAckSequence & 1) { 4930 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4931 } 4932 if (drainSequence & 1) { 4933 playbackThread->resetDraining(drainSequence >> 1); 4934 } 4935 } 4936 } 4937 } 4938 return false; 4939} 4940 4941void AudioFlinger::AsyncCallbackThread::exit() 4942{ 4943 ALOGV("AsyncCallbackThread::exit"); 4944 Mutex::Autolock _l(mLock); 4945 requestExit(); 4946 mWaitWorkCV.broadcast(); 4947} 4948 4949void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4950{ 4951 Mutex::Autolock _l(mLock); 4952 // bit 0 is cleared 4953 mWriteAckSequence = sequence << 1; 4954} 4955 4956void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4957{ 4958 Mutex::Autolock _l(mLock); 4959 // ignore unexpected callbacks 4960 if (mWriteAckSequence & 2) { 4961 mWriteAckSequence |= 1; 4962 mWaitWorkCV.signal(); 4963 } 4964} 4965 4966void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4967{ 4968 Mutex::Autolock _l(mLock); 4969 // bit 0 is cleared 4970 mDrainSequence = sequence << 1; 4971} 4972 4973void AudioFlinger::AsyncCallbackThread::resetDraining() 4974{ 4975 Mutex::Autolock _l(mLock); 4976 // ignore unexpected callbacks 4977 if (mDrainSequence & 2) { 4978 mDrainSequence |= 1; 4979 mWaitWorkCV.signal(); 4980 } 4981} 4982 4983 4984// ---------------------------------------------------------------------------- 4985AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4986 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 4987 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 4988 mPausedBytesRemaining(0) 4989{ 4990 //FIXME: mStandby should be set to true by ThreadBase constructor 4991 mStandby = true; 4992} 4993 4994void AudioFlinger::OffloadThread::threadLoop_exit() 4995{ 4996 if (mFlushPending || mHwPaused) { 4997 // If a flush is pending or track was paused, just discard buffered data 4998 flushHw_l(); 4999 } else { 5000 mMixerStatus = MIXER_DRAIN_ALL; 5001 threadLoop_drain(); 5002 } 5003 if (mUseAsyncWrite) { 5004 ALOG_ASSERT(mCallbackThread != 0); 5005 mCallbackThread->exit(); 5006 } 5007 PlaybackThread::threadLoop_exit(); 5008} 5009 5010AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5011 Vector< sp<Track> > *tracksToRemove 5012) 5013{ 5014 size_t count = mActiveTracks.size(); 5015 5016 mixer_state mixerStatus = MIXER_IDLE; 5017 bool doHwPause = false; 5018 bool doHwResume = false; 5019 5020 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5021 5022 // find out which tracks need to be processed 5023 for (size_t i = 0; i < count; i++) { 5024 sp<Track> t = mActiveTracks[i].promote(); 5025 // The track died recently 5026 if (t == 0) { 5027 continue; 5028 } 5029 Track* const track = t.get(); 5030 audio_track_cblk_t* cblk = track->cblk(); 5031 // Only consider last track started for volume and mixer state control. 5032 // In theory an older track could underrun and restart after the new one starts 5033 // but as we only care about the transition phase between two tracks on a 5034 // direct output, it is not a problem to ignore the underrun case. 5035 sp<Track> l = mLatestActiveTrack.promote(); 5036 bool last = l.get() == track; 5037 5038 if (track->isInvalid()) { 5039 ALOGW("An invalidated track shouldn't be in active list"); 5040 tracksToRemove->add(track); 5041 continue; 5042 } 5043 5044 if (track->mState == TrackBase::IDLE) { 5045 ALOGW("An idle track shouldn't be in active list"); 5046 continue; 5047 } 5048 5049 if (track->isPausing()) { 5050 track->setPaused(); 5051 if (last) { 5052 if (mHwSupportsPause && !mHwPaused) { 5053 doHwPause = true; 5054 mHwPaused = true; 5055 } 5056 // If we were part way through writing the mixbuffer to 5057 // the HAL we must save this until we resume 5058 // BUG - this will be wrong if a different track is made active, 5059 // in that case we want to discard the pending data in the 5060 // mixbuffer and tell the client to present it again when the 5061 // track is resumed 5062 mPausedWriteLength = mCurrentWriteLength; 5063 mPausedBytesRemaining = mBytesRemaining; 5064 mBytesRemaining = 0; // stop writing 5065 } 5066 tracksToRemove->add(track); 5067 } else if (track->isFlushPending()) { 5068 track->flushAck(); 5069 if (last) { 5070 mFlushPending = true; 5071 } 5072 } else if (track->isResumePending()){ 5073 track->resumeAck(); 5074 if (last) { 5075 if (mPausedBytesRemaining) { 5076 // Need to continue write that was interrupted 5077 mCurrentWriteLength = mPausedWriteLength; 5078 mBytesRemaining = mPausedBytesRemaining; 5079 mPausedBytesRemaining = 0; 5080 } 5081 if (mHwPaused) { 5082 doHwResume = true; 5083 mHwPaused = false; 5084 // threadLoop_mix() will handle the case that we need to 5085 // resume an interrupted write 5086 } 5087 // enable write to audio HAL 5088 mSleepTimeUs = 0; 5089 5090 // Do not handle new data in this iteration even if track->framesReady() 5091 mixerStatus = MIXER_TRACKS_ENABLED; 5092 } 5093 } else if (track->framesReady() && track->isReady() && 5094 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5095 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5096 if (track->mFillingUpStatus == Track::FS_FILLED) { 5097 track->mFillingUpStatus = Track::FS_ACTIVE; 5098 // make sure processVolume_l() will apply new volume even if 0 5099 mLeftVolFloat = mRightVolFloat = -1.0; 5100 } 5101 5102 if (last) { 5103 sp<Track> previousTrack = mPreviousTrack.promote(); 5104 if (previousTrack != 0) { 5105 if (track != previousTrack.get()) { 5106 // Flush any data still being written from last track 5107 mBytesRemaining = 0; 5108 if (mPausedBytesRemaining) { 5109 // Last track was paused so we also need to flush saved 5110 // mixbuffer state and invalidate track so that it will 5111 // re-submit that unwritten data when it is next resumed 5112 mPausedBytesRemaining = 0; 5113 // Invalidate is a bit drastic - would be more efficient 5114 // to have a flag to tell client that some of the 5115 // previously written data was lost 5116 previousTrack->invalidate(); 5117 } 5118 // flush data already sent to the DSP if changing audio session as audio 5119 // comes from a different source. Also invalidate previous track to force a 5120 // seek when resuming. 5121 if (previousTrack->sessionId() != track->sessionId()) { 5122 previousTrack->invalidate(); 5123 } 5124 } 5125 } 5126 mPreviousTrack = track; 5127 // reset retry count 5128 track->mRetryCount = kMaxTrackRetriesOffload; 5129 mActiveTrack = t; 5130 mixerStatus = MIXER_TRACKS_READY; 5131 } 5132 } else { 5133 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5134 if (track->isStopping_1()) { 5135 // Hardware buffer can hold a large amount of audio so we must 5136 // wait for all current track's data to drain before we say 5137 // that the track is stopped. 5138 if (mBytesRemaining == 0) { 5139 // Only start draining when all data in mixbuffer 5140 // has been written 5141 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5142 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5143 // do not drain if no data was ever sent to HAL (mStandby == true) 5144 if (last && !mStandby) { 5145 // do not modify drain sequence if we are already draining. This happens 5146 // when resuming from pause after drain. 5147 if ((mDrainSequence & 1) == 0) { 5148 mSleepTimeUs = 0; 5149 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5150 mixerStatus = MIXER_DRAIN_TRACK; 5151 mDrainSequence += 2; 5152 } 5153 if (mHwPaused) { 5154 // It is possible to move from PAUSED to STOPPING_1 without 5155 // a resume so we must ensure hardware is running 5156 doHwResume = true; 5157 mHwPaused = false; 5158 } 5159 } 5160 } 5161 } else if (track->isStopping_2()) { 5162 // Drain has completed or we are in standby, signal presentation complete 5163 if (!(mDrainSequence & 1) || !last || mStandby) { 5164 track->mState = TrackBase::STOPPED; 5165 size_t audioHALFrames = 5166 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5167 size_t framesWritten = 5168 mBytesWritten / mOutput->getFrameSize(); 5169 track->presentationComplete(framesWritten, audioHALFrames); 5170 track->reset(); 5171 tracksToRemove->add(track); 5172 } 5173 } else { 5174 // No buffers for this track. Give it a few chances to 5175 // fill a buffer, then remove it from active list. 5176 if (--(track->mRetryCount) <= 0) { 5177 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5178 track->name()); 5179 tracksToRemove->add(track); 5180 // indicate to client process that the track was disabled because of underrun; 5181 // it will then automatically call start() when data is available 5182 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5183 } else if (last){ 5184 mixerStatus = MIXER_TRACKS_ENABLED; 5185 } 5186 } 5187 } 5188 // compute volume for this track 5189 processVolume_l(track, last); 5190 } 5191 5192 // make sure the pause/flush/resume sequence is executed in the right order. 5193 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5194 // before flush and then resume HW. This can happen in case of pause/flush/resume 5195 // if resume is received before pause is executed. 5196 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5197 mOutput->stream->pause(mOutput->stream); 5198 } 5199 if (mFlushPending) { 5200 flushHw_l(); 5201 } 5202 if (!mStandby && doHwResume) { 5203 mOutput->stream->resume(mOutput->stream); 5204 } 5205 5206 // remove all the tracks that need to be... 5207 removeTracks_l(*tracksToRemove); 5208 5209 return mixerStatus; 5210} 5211 5212// must be called with thread mutex locked 5213bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5214{ 5215 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5216 mWriteAckSequence, mDrainSequence); 5217 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5218 return true; 5219 } 5220 return false; 5221} 5222 5223bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5224{ 5225 Mutex::Autolock _l(mLock); 5226 return waitingAsyncCallback_l(); 5227} 5228 5229void AudioFlinger::OffloadThread::flushHw_l() 5230{ 5231 DirectOutputThread::flushHw_l(); 5232 // Flush anything still waiting in the mixbuffer 5233 mCurrentWriteLength = 0; 5234 mBytesRemaining = 0; 5235 mPausedWriteLength = 0; 5236 mPausedBytesRemaining = 0; 5237 5238 if (mUseAsyncWrite) { 5239 // discard any pending drain or write ack by incrementing sequence 5240 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5241 mDrainSequence = (mDrainSequence + 2) & ~1; 5242 ALOG_ASSERT(mCallbackThread != 0); 5243 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5244 mCallbackThread->setDraining(mDrainSequence); 5245 } 5246} 5247 5248// ---------------------------------------------------------------------------- 5249 5250AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5251 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5252 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5253 systemReady, DUPLICATING), 5254 mWaitTimeMs(UINT_MAX) 5255{ 5256 addOutputTrack(mainThread); 5257} 5258 5259AudioFlinger::DuplicatingThread::~DuplicatingThread() 5260{ 5261 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5262 mOutputTracks[i]->destroy(); 5263 } 5264} 5265 5266void AudioFlinger::DuplicatingThread::threadLoop_mix() 5267{ 5268 // mix buffers... 5269 if (outputsReady(outputTracks)) { 5270 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5271 } else { 5272 if (mMixerBufferValid) { 5273 memset(mMixerBuffer, 0, mMixerBufferSize); 5274 } else { 5275 memset(mSinkBuffer, 0, mSinkBufferSize); 5276 } 5277 } 5278 mSleepTimeUs = 0; 5279 writeFrames = mNormalFrameCount; 5280 mCurrentWriteLength = mSinkBufferSize; 5281 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5282} 5283 5284void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5285{ 5286 if (mSleepTimeUs == 0) { 5287 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5288 mSleepTimeUs = mActiveSleepTimeUs; 5289 } else { 5290 mSleepTimeUs = mIdleSleepTimeUs; 5291 } 5292 } else if (mBytesWritten != 0) { 5293 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5294 writeFrames = mNormalFrameCount; 5295 memset(mSinkBuffer, 0, mSinkBufferSize); 5296 } else { 5297 // flush remaining overflow buffers in output tracks 5298 writeFrames = 0; 5299 } 5300 mSleepTimeUs = 0; 5301 } 5302} 5303 5304ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5305{ 5306 for (size_t i = 0; i < outputTracks.size(); i++) { 5307 outputTracks[i]->write(mSinkBuffer, writeFrames); 5308 } 5309 mStandby = false; 5310 return (ssize_t)mSinkBufferSize; 5311} 5312 5313void AudioFlinger::DuplicatingThread::threadLoop_standby() 5314{ 5315 // DuplicatingThread implements standby by stopping all tracks 5316 for (size_t i = 0; i < outputTracks.size(); i++) { 5317 outputTracks[i]->stop(); 5318 } 5319} 5320 5321void AudioFlinger::DuplicatingThread::saveOutputTracks() 5322{ 5323 outputTracks = mOutputTracks; 5324} 5325 5326void AudioFlinger::DuplicatingThread::clearOutputTracks() 5327{ 5328 outputTracks.clear(); 5329} 5330 5331void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5332{ 5333 Mutex::Autolock _l(mLock); 5334 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5335 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5336 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5337 const size_t frameCount = 5338 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5339 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5340 // from different OutputTracks and their associated MixerThreads (e.g. one may 5341 // nearly empty and the other may be dropping data). 5342 5343 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5344 this, 5345 mSampleRate, 5346 mFormat, 5347 mChannelMask, 5348 frameCount, 5349 IPCThreadState::self()->getCallingUid()); 5350 if (outputTrack->cblk() != NULL) { 5351 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5352 mOutputTracks.add(outputTrack); 5353 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5354 updateWaitTime_l(); 5355 } 5356} 5357 5358void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5359{ 5360 Mutex::Autolock _l(mLock); 5361 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5362 if (mOutputTracks[i]->thread() == thread) { 5363 mOutputTracks[i]->destroy(); 5364 mOutputTracks.removeAt(i); 5365 updateWaitTime_l(); 5366 if (thread->getOutput() == mOutput) { 5367 mOutput = NULL; 5368 } 5369 return; 5370 } 5371 } 5372 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5373} 5374 5375// caller must hold mLock 5376void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5377{ 5378 mWaitTimeMs = UINT_MAX; 5379 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5380 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5381 if (strong != 0) { 5382 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5383 if (waitTimeMs < mWaitTimeMs) { 5384 mWaitTimeMs = waitTimeMs; 5385 } 5386 } 5387 } 5388} 5389 5390 5391bool AudioFlinger::DuplicatingThread::outputsReady( 5392 const SortedVector< sp<OutputTrack> > &outputTracks) 5393{ 5394 for (size_t i = 0; i < outputTracks.size(); i++) { 5395 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5396 if (thread == 0) { 5397 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5398 outputTracks[i].get()); 5399 return false; 5400 } 5401 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5402 // see note at standby() declaration 5403 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5404 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5405 thread.get()); 5406 return false; 5407 } 5408 } 5409 return true; 5410} 5411 5412uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5413{ 5414 return (mWaitTimeMs * 1000) / 2; 5415} 5416 5417void AudioFlinger::DuplicatingThread::cacheParameters_l() 5418{ 5419 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5420 updateWaitTime_l(); 5421 5422 MixerThread::cacheParameters_l(); 5423} 5424 5425// ---------------------------------------------------------------------------- 5426// Record 5427// ---------------------------------------------------------------------------- 5428 5429AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5430 AudioStreamIn *input, 5431 audio_io_handle_t id, 5432 audio_devices_t outDevice, 5433 audio_devices_t inDevice, 5434 bool systemReady 5435#ifdef TEE_SINK 5436 , const sp<NBAIO_Sink>& teeSink 5437#endif 5438 ) : 5439 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5440 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5441 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5442 mRsmpInRear(0) 5443#ifdef TEE_SINK 5444 , mTeeSink(teeSink) 5445#endif 5446 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5447 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5448 // mFastCapture below 5449 , mFastCaptureFutex(0) 5450 // mInputSource 5451 // mPipeSink 5452 // mPipeSource 5453 , mPipeFramesP2(0) 5454 // mPipeMemory 5455 // mFastCaptureNBLogWriter 5456 , mFastTrackAvail(false) 5457{ 5458 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5459 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5460 5461 readInputParameters_l(); 5462 5463 // create an NBAIO source for the HAL input stream, and negotiate 5464 mInputSource = new AudioStreamInSource(input->stream); 5465 size_t numCounterOffers = 0; 5466 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5467 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5468 ALOG_ASSERT(index == 0); 5469 5470 // initialize fast capture depending on configuration 5471 bool initFastCapture; 5472 switch (kUseFastCapture) { 5473 case FastCapture_Never: 5474 initFastCapture = false; 5475 break; 5476 case FastCapture_Always: 5477 initFastCapture = true; 5478 break; 5479 case FastCapture_Static: 5480 uint32_t primaryOutputSampleRate; 5481 { 5482 AutoMutex _l(audioFlinger->mHardwareLock); 5483 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5484 } 5485 initFastCapture = 5486 // either capture sample rate is same as (a reasonable) primary output sample rate 5487 ((isMusicRate(primaryOutputSampleRate) && 5488 (mSampleRate == primaryOutputSampleRate)) || 5489 // or primary output sample rate is unknown, and capture sample rate is reasonable 5490 ((primaryOutputSampleRate == 0) && 5491 isMusicRate(mSampleRate))) && 5492 // and the buffer size is < 12 ms 5493 (mFrameCount * 1000) / mSampleRate < 12; 5494 break; 5495 // case FastCapture_Dynamic: 5496 } 5497 5498 if (initFastCapture) { 5499 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5500 NBAIO_Format format = mInputSource->format(); 5501 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5502 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5503 void *pipeBuffer; 5504 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5505 sp<IMemory> pipeMemory; 5506 if ((roHeap == 0) || 5507 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5508 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5509 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5510 goto failed; 5511 } 5512 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5513 memset(pipeBuffer, 0, pipeSize); 5514 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5515 const NBAIO_Format offers[1] = {format}; 5516 size_t numCounterOffers = 0; 5517 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5518 ALOG_ASSERT(index == 0); 5519 mPipeSink = pipe; 5520 PipeReader *pipeReader = new PipeReader(*pipe); 5521 numCounterOffers = 0; 5522 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5523 ALOG_ASSERT(index == 0); 5524 mPipeSource = pipeReader; 5525 mPipeFramesP2 = pipeFramesP2; 5526 mPipeMemory = pipeMemory; 5527 5528 // create fast capture 5529 mFastCapture = new FastCapture(); 5530 FastCaptureStateQueue *sq = mFastCapture->sq(); 5531#ifdef STATE_QUEUE_DUMP 5532 // FIXME 5533#endif 5534 FastCaptureState *state = sq->begin(); 5535 state->mCblk = NULL; 5536 state->mInputSource = mInputSource.get(); 5537 state->mInputSourceGen++; 5538 state->mPipeSink = pipe; 5539 state->mPipeSinkGen++; 5540 state->mFrameCount = mFrameCount; 5541 state->mCommand = FastCaptureState::COLD_IDLE; 5542 // already done in constructor initialization list 5543 //mFastCaptureFutex = 0; 5544 state->mColdFutexAddr = &mFastCaptureFutex; 5545 state->mColdGen++; 5546 state->mDumpState = &mFastCaptureDumpState; 5547#ifdef TEE_SINK 5548 // FIXME 5549#endif 5550 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5551 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5552 sq->end(); 5553 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5554 5555 // start the fast capture 5556 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5557 pid_t tid = mFastCapture->getTid(); 5558 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5559#ifdef AUDIO_WATCHDOG 5560 // FIXME 5561#endif 5562 5563 mFastTrackAvail = true; 5564 } 5565failed: ; 5566 5567 // FIXME mNormalSource 5568} 5569 5570AudioFlinger::RecordThread::~RecordThread() 5571{ 5572 if (mFastCapture != 0) { 5573 FastCaptureStateQueue *sq = mFastCapture->sq(); 5574 FastCaptureState *state = sq->begin(); 5575 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5576 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5577 if (old == -1) { 5578 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5579 } 5580 } 5581 state->mCommand = FastCaptureState::EXIT; 5582 sq->end(); 5583 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5584 mFastCapture->join(); 5585 mFastCapture.clear(); 5586 } 5587 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5588 mAudioFlinger->unregisterWriter(mNBLogWriter); 5589 free(mRsmpInBuffer); 5590} 5591 5592void AudioFlinger::RecordThread::onFirstRef() 5593{ 5594 run(mThreadName, PRIORITY_URGENT_AUDIO); 5595} 5596 5597bool AudioFlinger::RecordThread::threadLoop() 5598{ 5599 nsecs_t lastWarning = 0; 5600 5601 inputStandBy(); 5602 5603reacquire_wakelock: 5604 sp<RecordTrack> activeTrack; 5605 int activeTracksGen; 5606 { 5607 Mutex::Autolock _l(mLock); 5608 size_t size = mActiveTracks.size(); 5609 activeTracksGen = mActiveTracksGen; 5610 if (size > 0) { 5611 // FIXME an arbitrary choice 5612 activeTrack = mActiveTracks[0]; 5613 acquireWakeLock_l(activeTrack->uid()); 5614 if (size > 1) { 5615 SortedVector<int> tmp; 5616 for (size_t i = 0; i < size; i++) { 5617 tmp.add(mActiveTracks[i]->uid()); 5618 } 5619 updateWakeLockUids_l(tmp); 5620 } 5621 } else { 5622 acquireWakeLock_l(-1); 5623 } 5624 } 5625 5626 // used to request a deferred sleep, to be executed later while mutex is unlocked 5627 uint32_t sleepUs = 0; 5628 5629 // loop while there is work to do 5630 for (;;) { 5631 Vector< sp<EffectChain> > effectChains; 5632 5633 // sleep with mutex unlocked 5634 if (sleepUs > 0) { 5635 ATRACE_BEGIN("sleep"); 5636 usleep(sleepUs); 5637 ATRACE_END(); 5638 sleepUs = 0; 5639 } 5640 5641 // activeTracks accumulates a copy of a subset of mActiveTracks 5642 Vector< sp<RecordTrack> > activeTracks; 5643 5644 // reference to the (first and only) active fast track 5645 sp<RecordTrack> fastTrack; 5646 5647 // reference to a fast track which is about to be removed 5648 sp<RecordTrack> fastTrackToRemove; 5649 5650 { // scope for mLock 5651 Mutex::Autolock _l(mLock); 5652 5653 processConfigEvents_l(); 5654 5655 // check exitPending here because checkForNewParameters_l() and 5656 // checkForNewParameters_l() can temporarily release mLock 5657 if (exitPending()) { 5658 break; 5659 } 5660 5661 // if no active track(s), then standby and release wakelock 5662 size_t size = mActiveTracks.size(); 5663 if (size == 0) { 5664 standbyIfNotAlreadyInStandby(); 5665 // exitPending() can't become true here 5666 releaseWakeLock_l(); 5667 ALOGV("RecordThread: loop stopping"); 5668 // go to sleep 5669 mWaitWorkCV.wait(mLock); 5670 ALOGV("RecordThread: loop starting"); 5671 goto reacquire_wakelock; 5672 } 5673 5674 if (mActiveTracksGen != activeTracksGen) { 5675 activeTracksGen = mActiveTracksGen; 5676 SortedVector<int> tmp; 5677 for (size_t i = 0; i < size; i++) { 5678 tmp.add(mActiveTracks[i]->uid()); 5679 } 5680 updateWakeLockUids_l(tmp); 5681 } 5682 5683 bool doBroadcast = false; 5684 for (size_t i = 0; i < size; ) { 5685 5686 activeTrack = mActiveTracks[i]; 5687 if (activeTrack->isTerminated()) { 5688 if (activeTrack->isFastTrack()) { 5689 ALOG_ASSERT(fastTrackToRemove == 0); 5690 fastTrackToRemove = activeTrack; 5691 } 5692 removeTrack_l(activeTrack); 5693 mActiveTracks.remove(activeTrack); 5694 mActiveTracksGen++; 5695 size--; 5696 continue; 5697 } 5698 5699 TrackBase::track_state activeTrackState = activeTrack->mState; 5700 switch (activeTrackState) { 5701 5702 case TrackBase::PAUSING: 5703 mActiveTracks.remove(activeTrack); 5704 mActiveTracksGen++; 5705 doBroadcast = true; 5706 size--; 5707 continue; 5708 5709 case TrackBase::STARTING_1: 5710 sleepUs = 10000; 5711 i++; 5712 continue; 5713 5714 case TrackBase::STARTING_2: 5715 doBroadcast = true; 5716 mStandby = false; 5717 activeTrack->mState = TrackBase::ACTIVE; 5718 break; 5719 5720 case TrackBase::ACTIVE: 5721 break; 5722 5723 case TrackBase::IDLE: 5724 i++; 5725 continue; 5726 5727 default: 5728 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5729 } 5730 5731 activeTracks.add(activeTrack); 5732 i++; 5733 5734 if (activeTrack->isFastTrack()) { 5735 ALOG_ASSERT(!mFastTrackAvail); 5736 ALOG_ASSERT(fastTrack == 0); 5737 fastTrack = activeTrack; 5738 } 5739 } 5740 if (doBroadcast) { 5741 mStartStopCond.broadcast(); 5742 } 5743 5744 // sleep if there are no active tracks to process 5745 if (activeTracks.size() == 0) { 5746 if (sleepUs == 0) { 5747 sleepUs = kRecordThreadSleepUs; 5748 } 5749 continue; 5750 } 5751 sleepUs = 0; 5752 5753 lockEffectChains_l(effectChains); 5754 } 5755 5756 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5757 5758 size_t size = effectChains.size(); 5759 for (size_t i = 0; i < size; i++) { 5760 // thread mutex is not locked, but effect chain is locked 5761 effectChains[i]->process_l(); 5762 } 5763 5764 // Push a new fast capture state if fast capture is not already running, or cblk change 5765 if (mFastCapture != 0) { 5766 FastCaptureStateQueue *sq = mFastCapture->sq(); 5767 FastCaptureState *state = sq->begin(); 5768 bool didModify = false; 5769 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5770 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5771 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5772 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5773 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5774 if (old == -1) { 5775 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5776 } 5777 } 5778 state->mCommand = FastCaptureState::READ_WRITE; 5779#if 0 // FIXME 5780 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5781 FastThreadDumpState::kSamplingNforLowRamDevice : 5782 FastThreadDumpState::kSamplingN); 5783#endif 5784 didModify = true; 5785 } 5786 audio_track_cblk_t *cblkOld = state->mCblk; 5787 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5788 if (cblkNew != cblkOld) { 5789 state->mCblk = cblkNew; 5790 // block until acked if removing a fast track 5791 if (cblkOld != NULL) { 5792 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5793 } 5794 didModify = true; 5795 } 5796 sq->end(didModify); 5797 if (didModify) { 5798 sq->push(block); 5799#if 0 5800 if (kUseFastCapture == FastCapture_Dynamic) { 5801 mNormalSource = mPipeSource; 5802 } 5803#endif 5804 } 5805 } 5806 5807 // now run the fast track destructor with thread mutex unlocked 5808 fastTrackToRemove.clear(); 5809 5810 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5811 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5812 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5813 // If destination is non-contiguous, first read past the nominal end of buffer, then 5814 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5815 5816 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5817 ssize_t framesRead; 5818 5819 // If an NBAIO source is present, use it to read the normal capture's data 5820 if (mPipeSource != 0) { 5821 size_t framesToRead = mBufferSize / mFrameSize; 5822 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5823 framesToRead, AudioBufferProvider::kInvalidPTS); 5824 if (framesRead == 0) { 5825 // since pipe is non-blocking, simulate blocking input 5826 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5827 } 5828 // otherwise use the HAL / AudioStreamIn directly 5829 } else { 5830 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5831 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5832 if (bytesRead < 0) { 5833 framesRead = bytesRead; 5834 } else { 5835 framesRead = bytesRead / mFrameSize; 5836 } 5837 } 5838 5839 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5840 ALOGE("read failed: framesRead=%d", framesRead); 5841 // Force input into standby so that it tries to recover at next read attempt 5842 inputStandBy(); 5843 sleepUs = kRecordThreadSleepUs; 5844 } 5845 if (framesRead <= 0) { 5846 goto unlock; 5847 } 5848 ALOG_ASSERT(framesRead > 0); 5849 5850 if (mTeeSink != 0) { 5851 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5852 } 5853 // If destination is non-contiguous, we now correct for reading past end of buffer. 5854 { 5855 size_t part1 = mRsmpInFramesP2 - rear; 5856 if ((size_t) framesRead > part1) { 5857 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5858 (framesRead - part1) * mFrameSize); 5859 } 5860 } 5861 rear = mRsmpInRear += framesRead; 5862 5863 size = activeTracks.size(); 5864 // loop over each active track 5865 for (size_t i = 0; i < size; i++) { 5866 activeTrack = activeTracks[i]; 5867 5868 // skip fast tracks, as those are handled directly by FastCapture 5869 if (activeTrack->isFastTrack()) { 5870 continue; 5871 } 5872 5873 // TODO: This code probably should be moved to RecordTrack. 5874 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5875 5876 enum { 5877 OVERRUN_UNKNOWN, 5878 OVERRUN_TRUE, 5879 OVERRUN_FALSE 5880 } overrun = OVERRUN_UNKNOWN; 5881 5882 // loop over getNextBuffer to handle circular sink 5883 for (;;) { 5884 5885 activeTrack->mSink.frameCount = ~0; 5886 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5887 size_t framesOut = activeTrack->mSink.frameCount; 5888 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5889 5890 // check available frames and handle overrun conditions 5891 // if the record track isn't draining fast enough. 5892 bool hasOverrun; 5893 size_t framesIn; 5894 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5895 if (hasOverrun) { 5896 overrun = OVERRUN_TRUE; 5897 } 5898 if (framesOut == 0 || framesIn == 0) { 5899 break; 5900 } 5901 5902 // Don't allow framesOut to be larger than what is possible with resampling 5903 // from framesIn. 5904 // This isn't strictly necessary but helps limit buffer resizing in 5905 // RecordBufferConverter. TODO: remove when no longer needed. 5906 framesOut = min(framesOut, 5907 destinationFramesPossible( 5908 framesIn, mSampleRate, activeTrack->mSampleRate)); 5909 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5910 framesOut = activeTrack->mRecordBufferConverter->convert( 5911 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5912 5913 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5914 overrun = OVERRUN_FALSE; 5915 } 5916 5917 if (activeTrack->mFramesToDrop == 0) { 5918 if (framesOut > 0) { 5919 activeTrack->mSink.frameCount = framesOut; 5920 activeTrack->releaseBuffer(&activeTrack->mSink); 5921 } 5922 } else { 5923 // FIXME could do a partial drop of framesOut 5924 if (activeTrack->mFramesToDrop > 0) { 5925 activeTrack->mFramesToDrop -= framesOut; 5926 if (activeTrack->mFramesToDrop <= 0) { 5927 activeTrack->clearSyncStartEvent(); 5928 } 5929 } else { 5930 activeTrack->mFramesToDrop += framesOut; 5931 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5932 activeTrack->mSyncStartEvent->isCancelled()) { 5933 ALOGW("Synced record %s, session %d, trigger session %d", 5934 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5935 activeTrack->sessionId(), 5936 (activeTrack->mSyncStartEvent != 0) ? 5937 activeTrack->mSyncStartEvent->triggerSession() : 0); 5938 activeTrack->clearSyncStartEvent(); 5939 } 5940 } 5941 } 5942 5943 if (framesOut == 0) { 5944 break; 5945 } 5946 } 5947 5948 switch (overrun) { 5949 case OVERRUN_TRUE: 5950 // client isn't retrieving buffers fast enough 5951 if (!activeTrack->setOverflow()) { 5952 nsecs_t now = systemTime(); 5953 // FIXME should lastWarning per track? 5954 if ((now - lastWarning) > kWarningThrottleNs) { 5955 ALOGW("RecordThread: buffer overflow"); 5956 lastWarning = now; 5957 } 5958 } 5959 break; 5960 case OVERRUN_FALSE: 5961 activeTrack->clearOverflow(); 5962 break; 5963 case OVERRUN_UNKNOWN: 5964 break; 5965 } 5966 5967 } 5968 5969unlock: 5970 // enable changes in effect chain 5971 unlockEffectChains(effectChains); 5972 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5973 } 5974 5975 standbyIfNotAlreadyInStandby(); 5976 5977 { 5978 Mutex::Autolock _l(mLock); 5979 for (size_t i = 0; i < mTracks.size(); i++) { 5980 sp<RecordTrack> track = mTracks[i]; 5981 track->invalidate(); 5982 } 5983 mActiveTracks.clear(); 5984 mActiveTracksGen++; 5985 mStartStopCond.broadcast(); 5986 } 5987 5988 releaseWakeLock(); 5989 5990 ALOGV("RecordThread %p exiting", this); 5991 return false; 5992} 5993 5994void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5995{ 5996 if (!mStandby) { 5997 inputStandBy(); 5998 mStandby = true; 5999 } 6000} 6001 6002void AudioFlinger::RecordThread::inputStandBy() 6003{ 6004 // Idle the fast capture if it's currently running 6005 if (mFastCapture != 0) { 6006 FastCaptureStateQueue *sq = mFastCapture->sq(); 6007 FastCaptureState *state = sq->begin(); 6008 if (!(state->mCommand & FastCaptureState::IDLE)) { 6009 state->mCommand = FastCaptureState::COLD_IDLE; 6010 state->mColdFutexAddr = &mFastCaptureFutex; 6011 state->mColdGen++; 6012 mFastCaptureFutex = 0; 6013 sq->end(); 6014 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6015 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6016#if 0 6017 if (kUseFastCapture == FastCapture_Dynamic) { 6018 // FIXME 6019 } 6020#endif 6021#ifdef AUDIO_WATCHDOG 6022 // FIXME 6023#endif 6024 } else { 6025 sq->end(false /*didModify*/); 6026 } 6027 } 6028 mInput->stream->common.standby(&mInput->stream->common); 6029} 6030 6031// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6032sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6033 const sp<AudioFlinger::Client>& client, 6034 uint32_t sampleRate, 6035 audio_format_t format, 6036 audio_channel_mask_t channelMask, 6037 size_t *pFrameCount, 6038 int sessionId, 6039 size_t *notificationFrames, 6040 int uid, 6041 IAudioFlinger::track_flags_t *flags, 6042 pid_t tid, 6043 status_t *status) 6044{ 6045 size_t frameCount = *pFrameCount; 6046 sp<RecordTrack> track; 6047 status_t lStatus; 6048 6049 // client expresses a preference for FAST, but we get the final say 6050 if (*flags & IAudioFlinger::TRACK_FAST) { 6051 if ( 6052 // we formerly checked for a callback handler (non-0 tid), 6053 // but that is no longer required for TRANSFER_OBTAIN mode 6054 // 6055 // frame count is not specified, or is exactly the pipe depth 6056 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6057 // PCM data 6058 audio_is_linear_pcm(format) && 6059 // native format 6060 (format == mFormat) && 6061 // native channel mask 6062 (channelMask == mChannelMask) && 6063 // native hardware sample rate 6064 (sampleRate == mSampleRate) && 6065 // record thread has an associated fast capture 6066 hasFastCapture() && 6067 // there are sufficient fast track slots available 6068 mFastTrackAvail 6069 ) { 6070 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6071 frameCount, mFrameCount); 6072 } else { 6073 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6074 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6075 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6076 frameCount, mFrameCount, mPipeFramesP2, 6077 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6078 hasFastCapture(), tid, mFastTrackAvail); 6079 *flags &= ~IAudioFlinger::TRACK_FAST; 6080 } 6081 } 6082 6083 // compute track buffer size in frames, and suggest the notification frame count 6084 if (*flags & IAudioFlinger::TRACK_FAST) { 6085 // fast track: frame count is exactly the pipe depth 6086 frameCount = mPipeFramesP2; 6087 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6088 *notificationFrames = mFrameCount; 6089 } else { 6090 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6091 // or 20 ms if there is a fast capture 6092 // TODO This could be a roundupRatio inline, and const 6093 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6094 * sampleRate + mSampleRate - 1) / mSampleRate; 6095 // minimum number of notification periods is at least kMinNotifications, 6096 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6097 static const size_t kMinNotifications = 3; 6098 static const uint32_t kMinMs = 30; 6099 // TODO This could be a roundupRatio inline 6100 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6101 // TODO This could be a roundupRatio inline 6102 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6103 maxNotificationFrames; 6104 const size_t minFrameCount = maxNotificationFrames * 6105 max(kMinNotifications, minNotificationsByMs); 6106 frameCount = max(frameCount, minFrameCount); 6107 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6108 *notificationFrames = maxNotificationFrames; 6109 } 6110 } 6111 *pFrameCount = frameCount; 6112 6113 lStatus = initCheck(); 6114 if (lStatus != NO_ERROR) { 6115 ALOGE("createRecordTrack_l() audio driver not initialized"); 6116 goto Exit; 6117 } 6118 6119 { // scope for mLock 6120 Mutex::Autolock _l(mLock); 6121 6122 track = new RecordTrack(this, client, sampleRate, 6123 format, channelMask, frameCount, NULL, sessionId, uid, 6124 *flags, TrackBase::TYPE_DEFAULT); 6125 6126 lStatus = track->initCheck(); 6127 if (lStatus != NO_ERROR) { 6128 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6129 // track must be cleared from the caller as the caller has the AF lock 6130 goto Exit; 6131 } 6132 mTracks.add(track); 6133 6134 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6135 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6136 mAudioFlinger->btNrecIsOff(); 6137 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6138 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6139 6140 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6141 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6142 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6143 // so ask activity manager to do this on our behalf 6144 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6145 } 6146 } 6147 6148 lStatus = NO_ERROR; 6149 6150Exit: 6151 *status = lStatus; 6152 return track; 6153} 6154 6155status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6156 AudioSystem::sync_event_t event, 6157 int triggerSession) 6158{ 6159 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6160 sp<ThreadBase> strongMe = this; 6161 status_t status = NO_ERROR; 6162 6163 if (event == AudioSystem::SYNC_EVENT_NONE) { 6164 recordTrack->clearSyncStartEvent(); 6165 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6166 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6167 triggerSession, 6168 recordTrack->sessionId(), 6169 syncStartEventCallback, 6170 recordTrack); 6171 // Sync event can be cancelled by the trigger session if the track is not in a 6172 // compatible state in which case we start record immediately 6173 if (recordTrack->mSyncStartEvent->isCancelled()) { 6174 recordTrack->clearSyncStartEvent(); 6175 } else { 6176 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6177 recordTrack->mFramesToDrop = - 6178 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6179 } 6180 } 6181 6182 { 6183 // This section is a rendezvous between binder thread executing start() and RecordThread 6184 AutoMutex lock(mLock); 6185 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6186 if (recordTrack->mState == TrackBase::PAUSING) { 6187 ALOGV("active record track PAUSING -> ACTIVE"); 6188 recordTrack->mState = TrackBase::ACTIVE; 6189 } else { 6190 ALOGV("active record track state %d", recordTrack->mState); 6191 } 6192 return status; 6193 } 6194 6195 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6196 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6197 // or using a separate command thread 6198 recordTrack->mState = TrackBase::STARTING_1; 6199 mActiveTracks.add(recordTrack); 6200 mActiveTracksGen++; 6201 status_t status = NO_ERROR; 6202 if (recordTrack->isExternalTrack()) { 6203 mLock.unlock(); 6204 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6205 mLock.lock(); 6206 // FIXME should verify that recordTrack is still in mActiveTracks 6207 if (status != NO_ERROR) { 6208 mActiveTracks.remove(recordTrack); 6209 mActiveTracksGen++; 6210 recordTrack->clearSyncStartEvent(); 6211 ALOGV("RecordThread::start error %d", status); 6212 return status; 6213 } 6214 } 6215 // Catch up with current buffer indices if thread is already running. 6216 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6217 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6218 // see previously buffered data before it called start(), but with greater risk of overrun. 6219 6220 recordTrack->mResamplerBufferProvider->reset(); 6221 // clear any converter state as new data will be discontinuous 6222 recordTrack->mRecordBufferConverter->reset(); 6223 recordTrack->mState = TrackBase::STARTING_2; 6224 // signal thread to start 6225 mWaitWorkCV.broadcast(); 6226 if (mActiveTracks.indexOf(recordTrack) < 0) { 6227 ALOGV("Record failed to start"); 6228 status = BAD_VALUE; 6229 goto startError; 6230 } 6231 return status; 6232 } 6233 6234startError: 6235 if (recordTrack->isExternalTrack()) { 6236 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6237 } 6238 recordTrack->clearSyncStartEvent(); 6239 // FIXME I wonder why we do not reset the state here? 6240 return status; 6241} 6242 6243void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6244{ 6245 sp<SyncEvent> strongEvent = event.promote(); 6246 6247 if (strongEvent != 0) { 6248 sp<RefBase> ptr = strongEvent->cookie().promote(); 6249 if (ptr != 0) { 6250 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6251 recordTrack->handleSyncStartEvent(strongEvent); 6252 } 6253 } 6254} 6255 6256bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6257 ALOGV("RecordThread::stop"); 6258 AutoMutex _l(mLock); 6259 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6260 return false; 6261 } 6262 // note that threadLoop may still be processing the track at this point [without lock] 6263 recordTrack->mState = TrackBase::PAUSING; 6264 // do not wait for mStartStopCond if exiting 6265 if (exitPending()) { 6266 return true; 6267 } 6268 // FIXME incorrect usage of wait: no explicit predicate or loop 6269 mStartStopCond.wait(mLock); 6270 // if we have been restarted, recordTrack is in mActiveTracks here 6271 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6272 ALOGV("Record stopped OK"); 6273 return true; 6274 } 6275 return false; 6276} 6277 6278bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6279{ 6280 return false; 6281} 6282 6283status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6284{ 6285#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6286 if (!isValidSyncEvent(event)) { 6287 return BAD_VALUE; 6288 } 6289 6290 int eventSession = event->triggerSession(); 6291 status_t ret = NAME_NOT_FOUND; 6292 6293 Mutex::Autolock _l(mLock); 6294 6295 for (size_t i = 0; i < mTracks.size(); i++) { 6296 sp<RecordTrack> track = mTracks[i]; 6297 if (eventSession == track->sessionId()) { 6298 (void) track->setSyncEvent(event); 6299 ret = NO_ERROR; 6300 } 6301 } 6302 return ret; 6303#else 6304 return BAD_VALUE; 6305#endif 6306} 6307 6308// destroyTrack_l() must be called with ThreadBase::mLock held 6309void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6310{ 6311 track->terminate(); 6312 track->mState = TrackBase::STOPPED; 6313 // active tracks are removed by threadLoop() 6314 if (mActiveTracks.indexOf(track) < 0) { 6315 removeTrack_l(track); 6316 } 6317} 6318 6319void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6320{ 6321 mTracks.remove(track); 6322 // need anything related to effects here? 6323 if (track->isFastTrack()) { 6324 ALOG_ASSERT(!mFastTrackAvail); 6325 mFastTrackAvail = true; 6326 } 6327} 6328 6329void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6330{ 6331 dumpInternals(fd, args); 6332 dumpTracks(fd, args); 6333 dumpEffectChains(fd, args); 6334} 6335 6336void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6337{ 6338 dprintf(fd, "\nInput thread %p:\n", this); 6339 6340 dumpBase(fd, args); 6341 6342 if (mActiveTracks.size() == 0) { 6343 dprintf(fd, " No active record clients\n"); 6344 } 6345 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6346 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6347 6348 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6349 const FastCaptureDumpState copy(mFastCaptureDumpState); 6350 copy.dump(fd); 6351} 6352 6353void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6354{ 6355 const size_t SIZE = 256; 6356 char buffer[SIZE]; 6357 String8 result; 6358 6359 size_t numtracks = mTracks.size(); 6360 size_t numactive = mActiveTracks.size(); 6361 size_t numactiveseen = 0; 6362 dprintf(fd, " %d Tracks", numtracks); 6363 if (numtracks) { 6364 dprintf(fd, " of which %d are active\n", numactive); 6365 RecordTrack::appendDumpHeader(result); 6366 for (size_t i = 0; i < numtracks ; ++i) { 6367 sp<RecordTrack> track = mTracks[i]; 6368 if (track != 0) { 6369 bool active = mActiveTracks.indexOf(track) >= 0; 6370 if (active) { 6371 numactiveseen++; 6372 } 6373 track->dump(buffer, SIZE, active); 6374 result.append(buffer); 6375 } 6376 } 6377 } else { 6378 dprintf(fd, "\n"); 6379 } 6380 6381 if (numactiveseen != numactive) { 6382 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6383 " not in the track list\n"); 6384 result.append(buffer); 6385 RecordTrack::appendDumpHeader(result); 6386 for (size_t i = 0; i < numactive; ++i) { 6387 sp<RecordTrack> track = mActiveTracks[i]; 6388 if (mTracks.indexOf(track) < 0) { 6389 track->dump(buffer, SIZE, true); 6390 result.append(buffer); 6391 } 6392 } 6393 6394 } 6395 write(fd, result.string(), result.size()); 6396} 6397 6398 6399void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6400{ 6401 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6402 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6403 mRsmpInFront = recordThread->mRsmpInRear; 6404 mRsmpInUnrel = 0; 6405} 6406 6407void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6408 size_t *framesAvailable, bool *hasOverrun) 6409{ 6410 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6411 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6412 const int32_t rear = recordThread->mRsmpInRear; 6413 const int32_t front = mRsmpInFront; 6414 const ssize_t filled = rear - front; 6415 6416 size_t framesIn; 6417 bool overrun = false; 6418 if (filled < 0) { 6419 // should not happen, but treat like a massive overrun and re-sync 6420 framesIn = 0; 6421 mRsmpInFront = rear; 6422 overrun = true; 6423 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6424 framesIn = (size_t) filled; 6425 } else { 6426 // client is not keeping up with server, but give it latest data 6427 framesIn = recordThread->mRsmpInFrames; 6428 mRsmpInFront = /* front = */ rear - framesIn; 6429 overrun = true; 6430 } 6431 if (framesAvailable != NULL) { 6432 *framesAvailable = framesIn; 6433 } 6434 if (hasOverrun != NULL) { 6435 *hasOverrun = overrun; 6436 } 6437} 6438 6439// AudioBufferProvider interface 6440status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6441 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6442{ 6443 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6444 if (threadBase == 0) { 6445 buffer->frameCount = 0; 6446 buffer->raw = NULL; 6447 return NOT_ENOUGH_DATA; 6448 } 6449 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6450 int32_t rear = recordThread->mRsmpInRear; 6451 int32_t front = mRsmpInFront; 6452 ssize_t filled = rear - front; 6453 // FIXME should not be P2 (don't want to increase latency) 6454 // FIXME if client not keeping up, discard 6455 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6456 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6457 front &= recordThread->mRsmpInFramesP2 - 1; 6458 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6459 if (part1 > (size_t) filled) { 6460 part1 = filled; 6461 } 6462 size_t ask = buffer->frameCount; 6463 ALOG_ASSERT(ask > 0); 6464 if (part1 > ask) { 6465 part1 = ask; 6466 } 6467 if (part1 == 0) { 6468 // out of data is fine since the resampler will return a short-count. 6469 buffer->raw = NULL; 6470 buffer->frameCount = 0; 6471 mRsmpInUnrel = 0; 6472 return NOT_ENOUGH_DATA; 6473 } 6474 6475 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6476 buffer->frameCount = part1; 6477 mRsmpInUnrel = part1; 6478 return NO_ERROR; 6479} 6480 6481// AudioBufferProvider interface 6482void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6483 AudioBufferProvider::Buffer* buffer) 6484{ 6485 size_t stepCount = buffer->frameCount; 6486 if (stepCount == 0) { 6487 return; 6488 } 6489 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6490 mRsmpInUnrel -= stepCount; 6491 mRsmpInFront += stepCount; 6492 buffer->raw = NULL; 6493 buffer->frameCount = 0; 6494} 6495 6496AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6497 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6498 uint32_t srcSampleRate, 6499 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6500 uint32_t dstSampleRate) : 6501 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6502 // mSrcFormat 6503 // mSrcSampleRate 6504 // mDstChannelMask 6505 // mDstFormat 6506 // mDstSampleRate 6507 // mSrcChannelCount 6508 // mDstChannelCount 6509 // mDstFrameSize 6510 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6511 mResampler(NULL), 6512 mIsLegacyDownmix(false), 6513 mIsLegacyUpmix(false), 6514 mRequiresFloat(false), 6515 mInputConverterProvider(NULL) 6516{ 6517 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6518 dstChannelMask, dstFormat, dstSampleRate); 6519} 6520 6521AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6522 free(mBuf); 6523 delete mResampler; 6524 delete mInputConverterProvider; 6525} 6526 6527size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6528 AudioBufferProvider *provider, size_t frames) 6529{ 6530 if (mInputConverterProvider != NULL) { 6531 mInputConverterProvider->setBufferProvider(provider); 6532 provider = mInputConverterProvider; 6533 } 6534 6535 if (mResampler == NULL) { 6536 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6537 mSrcSampleRate, mSrcFormat, mDstFormat); 6538 6539 AudioBufferProvider::Buffer buffer; 6540 for (size_t i = frames; i > 0; ) { 6541 buffer.frameCount = i; 6542 status_t status = provider->getNextBuffer(&buffer, 0); 6543 if (status != OK || buffer.frameCount == 0) { 6544 frames -= i; // cannot fill request. 6545 break; 6546 } 6547 // format convert to destination buffer 6548 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6549 6550 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6551 i -= buffer.frameCount; 6552 provider->releaseBuffer(&buffer); 6553 } 6554 } else { 6555 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6556 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6557 6558 // reallocate buffer if needed 6559 if (mBufFrameSize != 0 && mBufFrames < frames) { 6560 free(mBuf); 6561 mBufFrames = frames; 6562 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6563 } 6564 // resampler accumulates, but we only have one source track 6565 memset(mBuf, 0, frames * mBufFrameSize); 6566 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6567 // format convert to destination buffer 6568 convertResampler(dst, mBuf, frames); 6569 } 6570 return frames; 6571} 6572 6573status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6574 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6575 uint32_t srcSampleRate, 6576 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6577 uint32_t dstSampleRate) 6578{ 6579 // quick evaluation if there is any change. 6580 if (mSrcFormat == srcFormat 6581 && mSrcChannelMask == srcChannelMask 6582 && mSrcSampleRate == srcSampleRate 6583 && mDstFormat == dstFormat 6584 && mDstChannelMask == dstChannelMask 6585 && mDstSampleRate == dstSampleRate) { 6586 return NO_ERROR; 6587 } 6588 6589 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6590 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6591 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6592 const bool valid = 6593 audio_is_input_channel(srcChannelMask) 6594 && audio_is_input_channel(dstChannelMask) 6595 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6596 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6597 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6598 ; // no upsampling checks for now 6599 if (!valid) { 6600 return BAD_VALUE; 6601 } 6602 6603 mSrcFormat = srcFormat; 6604 mSrcChannelMask = srcChannelMask; 6605 mSrcSampleRate = srcSampleRate; 6606 mDstFormat = dstFormat; 6607 mDstChannelMask = dstChannelMask; 6608 mDstSampleRate = dstSampleRate; 6609 6610 // compute derived parameters 6611 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6612 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6613 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6614 6615 // do we need to resample? 6616 delete mResampler; 6617 mResampler = NULL; 6618 if (mSrcSampleRate != mDstSampleRate) { 6619 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6620 mSrcChannelCount, mDstSampleRate); 6621 mResampler->setSampleRate(mSrcSampleRate); 6622 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6623 } 6624 6625 // are we running legacy channel conversion modes? 6626 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6627 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6628 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6629 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6630 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6631 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6632 6633 // do we need to process in float? 6634 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6635 6636 // do we need a staging buffer to convert for destination (we can still optimize this)? 6637 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6638 if (mResampler != NULL) { 6639 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6640 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6641 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6642 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6643 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6644 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6645 } else { 6646 mBufFrameSize = 0; 6647 } 6648 mBufFrames = 0; // force the buffer to be resized. 6649 6650 // do we need an input converter buffer provider to give us float? 6651 delete mInputConverterProvider; 6652 mInputConverterProvider = NULL; 6653 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6654 mInputConverterProvider = new ReformatBufferProvider( 6655 audio_channel_count_from_in_mask(mSrcChannelMask), 6656 mSrcFormat, 6657 AUDIO_FORMAT_PCM_FLOAT, 6658 256 /* provider buffer frame count */); 6659 } 6660 6661 // do we need a remixer to do channel mask conversion 6662 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6663 (void) memcpy_by_index_array_initialization_from_channel_mask( 6664 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6665 } 6666 return NO_ERROR; 6667} 6668 6669void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6670 void *dst, const void *src, size_t frames) 6671{ 6672 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6673 if (mBufFrameSize != 0 && mBufFrames < frames) { 6674 free(mBuf); 6675 mBufFrames = frames; 6676 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6677 } 6678 // do we need to do legacy upmix and downmix? 6679 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6680 void *dstBuf = mBuf != NULL ? mBuf : dst; 6681 if (mIsLegacyUpmix) { 6682 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6683 (const float *)src, frames); 6684 } else /*mIsLegacyDownmix */ { 6685 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6686 (const float *)src, frames); 6687 } 6688 if (mBuf != NULL) { 6689 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6690 frames * mDstChannelCount); 6691 } 6692 return; 6693 } 6694 // do we need to do channel mask conversion? 6695 if (mSrcChannelMask != mDstChannelMask) { 6696 void *dstBuf = mBuf != NULL ? mBuf : dst; 6697 memcpy_by_index_array(dstBuf, mDstChannelCount, 6698 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6699 if (dstBuf == dst) { 6700 return; // format is the same 6701 } 6702 } 6703 // convert to destination buffer 6704 const void *convertBuf = mBuf != NULL ? mBuf : src; 6705 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6706 frames * mDstChannelCount); 6707} 6708 6709void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6710 void *dst, /*not-a-const*/ void *src, size_t frames) 6711{ 6712 // src buffer format is ALWAYS float when entering this routine 6713 if (mIsLegacyUpmix) { 6714 ; // mono to stereo already handled by resampler 6715 } else if (mIsLegacyDownmix 6716 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6717 // the resampler outputs stereo for mono input channel (a feature?) 6718 // must convert to mono 6719 downmix_to_mono_float_from_stereo_float((float *)src, 6720 (const float *)src, frames); 6721 } else if (mSrcChannelMask != mDstChannelMask) { 6722 // convert to mono channel again for channel mask conversion (could be skipped 6723 // with further optimization). 6724 if (mSrcChannelCount == 1) { 6725 downmix_to_mono_float_from_stereo_float((float *)src, 6726 (const float *)src, frames); 6727 } 6728 // convert to destination format (in place, OK as float is larger than other types) 6729 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6730 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6731 frames * mSrcChannelCount); 6732 } 6733 // channel convert and save to dst 6734 memcpy_by_index_array(dst, mDstChannelCount, 6735 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6736 return; 6737 } 6738 // convert to destination format and save to dst 6739 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6740 frames * mDstChannelCount); 6741} 6742 6743bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6744 status_t& status) 6745{ 6746 bool reconfig = false; 6747 6748 status = NO_ERROR; 6749 6750 audio_format_t reqFormat = mFormat; 6751 uint32_t samplingRate = mSampleRate; 6752 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6753 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6754 6755 AudioParameter param = AudioParameter(keyValuePair); 6756 int value; 6757 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6758 // channel count change can be requested. Do we mandate the first client defines the 6759 // HAL sampling rate and channel count or do we allow changes on the fly? 6760 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6761 samplingRate = value; 6762 reconfig = true; 6763 } 6764 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6765 if (!audio_is_linear_pcm((audio_format_t) value)) { 6766 status = BAD_VALUE; 6767 } else { 6768 reqFormat = (audio_format_t) value; 6769 reconfig = true; 6770 } 6771 } 6772 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6773 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6774 if (!audio_is_input_channel(mask) || 6775 audio_channel_count_from_in_mask(mask) > FCC_8) { 6776 status = BAD_VALUE; 6777 } else { 6778 channelMask = mask; 6779 reconfig = true; 6780 } 6781 } 6782 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6783 // do not accept frame count changes if tracks are open as the track buffer 6784 // size depends on frame count and correct behavior would not be guaranteed 6785 // if frame count is changed after track creation 6786 if (mActiveTracks.size() > 0) { 6787 status = INVALID_OPERATION; 6788 } else { 6789 reconfig = true; 6790 } 6791 } 6792 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6793 // forward device change to effects that have requested to be 6794 // aware of attached audio device. 6795 for (size_t i = 0; i < mEffectChains.size(); i++) { 6796 mEffectChains[i]->setDevice_l(value); 6797 } 6798 6799 // store input device and output device but do not forward output device to audio HAL. 6800 // Note that status is ignored by the caller for output device 6801 // (see AudioFlinger::setParameters() 6802 if (audio_is_output_devices(value)) { 6803 mOutDevice = value; 6804 status = BAD_VALUE; 6805 } else { 6806 mInDevice = value; 6807 if (value != AUDIO_DEVICE_NONE) { 6808 mPrevInDevice = value; 6809 } 6810 // disable AEC and NS if the device is a BT SCO headset supporting those 6811 // pre processings 6812 if (mTracks.size() > 0) { 6813 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6814 mAudioFlinger->btNrecIsOff(); 6815 for (size_t i = 0; i < mTracks.size(); i++) { 6816 sp<RecordTrack> track = mTracks[i]; 6817 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6818 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6819 } 6820 } 6821 } 6822 } 6823 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6824 mAudioSource != (audio_source_t)value) { 6825 // forward device change to effects that have requested to be 6826 // aware of attached audio device. 6827 for (size_t i = 0; i < mEffectChains.size(); i++) { 6828 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6829 } 6830 mAudioSource = (audio_source_t)value; 6831 } 6832 6833 if (status == NO_ERROR) { 6834 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6835 keyValuePair.string()); 6836 if (status == INVALID_OPERATION) { 6837 inputStandBy(); 6838 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6839 keyValuePair.string()); 6840 } 6841 if (reconfig) { 6842 if (status == BAD_VALUE && 6843 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6844 audio_is_linear_pcm(reqFormat) && 6845 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6846 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6847 audio_channel_count_from_in_mask( 6848 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6849 status = NO_ERROR; 6850 } 6851 if (status == NO_ERROR) { 6852 readInputParameters_l(); 6853 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6854 } 6855 } 6856 } 6857 6858 return reconfig; 6859} 6860 6861String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6862{ 6863 Mutex::Autolock _l(mLock); 6864 if (initCheck() != NO_ERROR) { 6865 return String8(); 6866 } 6867 6868 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6869 const String8 out_s8(s); 6870 free(s); 6871 return out_s8; 6872} 6873 6874void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event) { 6875 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6876 6877 desc->mIoHandle = mId; 6878 6879 switch (event) { 6880 case AUDIO_INPUT_OPENED: 6881 case AUDIO_INPUT_CONFIG_CHANGED: 6882 desc->mPatch = mPatch; 6883 desc->mChannelMask = mChannelMask; 6884 desc->mSamplingRate = mSampleRate; 6885 desc->mFormat = mFormat; 6886 desc->mFrameCount = mFrameCount; 6887 desc->mLatency = 0; 6888 break; 6889 6890 case AUDIO_INPUT_CLOSED: 6891 default: 6892 break; 6893 } 6894 mAudioFlinger->ioConfigChanged(event, desc); 6895} 6896 6897void AudioFlinger::RecordThread::readInputParameters_l() 6898{ 6899 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6900 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6901 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6902 if (mChannelCount > FCC_8) { 6903 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6904 } 6905 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6906 mFormat = mHALFormat; 6907 if (!audio_is_linear_pcm(mFormat)) { 6908 ALOGE("HAL format %#x is not linear pcm", mFormat); 6909 } 6910 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6911 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6912 mFrameCount = mBufferSize / mFrameSize; 6913 // This is the formula for calculating the temporary buffer size. 6914 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6915 // 1 full output buffer, regardless of the alignment of the available input. 6916 // The value is somewhat arbitrary, and could probably be even larger. 6917 // A larger value should allow more old data to be read after a track calls start(), 6918 // without increasing latency. 6919 // 6920 // Note this is independent of the maximum downsampling ratio permitted for capture. 6921 mRsmpInFrames = mFrameCount * 7; 6922 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6923 free(mRsmpInBuffer); 6924 6925 // TODO optimize audio capture buffer sizes ... 6926 // Here we calculate the size of the sliding buffer used as a source 6927 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6928 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6929 // be better to have it derived from the pipe depth in the long term. 6930 // The current value is higher than necessary. However it should not add to latency. 6931 6932 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6933 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6934 6935 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6936 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6937} 6938 6939uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6940{ 6941 Mutex::Autolock _l(mLock); 6942 if (initCheck() != NO_ERROR) { 6943 return 0; 6944 } 6945 6946 return mInput->stream->get_input_frames_lost(mInput->stream); 6947} 6948 6949uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6950{ 6951 Mutex::Autolock _l(mLock); 6952 uint32_t result = 0; 6953 if (getEffectChain_l(sessionId) != 0) { 6954 result = EFFECT_SESSION; 6955 } 6956 6957 for (size_t i = 0; i < mTracks.size(); ++i) { 6958 if (sessionId == mTracks[i]->sessionId()) { 6959 result |= TRACK_SESSION; 6960 break; 6961 } 6962 } 6963 6964 return result; 6965} 6966 6967KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6968{ 6969 KeyedVector<int, bool> ids; 6970 Mutex::Autolock _l(mLock); 6971 for (size_t j = 0; j < mTracks.size(); ++j) { 6972 sp<RecordThread::RecordTrack> track = mTracks[j]; 6973 int sessionId = track->sessionId(); 6974 if (ids.indexOfKey(sessionId) < 0) { 6975 ids.add(sessionId, true); 6976 } 6977 } 6978 return ids; 6979} 6980 6981AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6982{ 6983 Mutex::Autolock _l(mLock); 6984 AudioStreamIn *input = mInput; 6985 mInput = NULL; 6986 return input; 6987} 6988 6989// this method must always be called either with ThreadBase mLock held or inside the thread loop 6990audio_stream_t* AudioFlinger::RecordThread::stream() const 6991{ 6992 if (mInput == NULL) { 6993 return NULL; 6994 } 6995 return &mInput->stream->common; 6996} 6997 6998status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6999{ 7000 // only one chain per input thread 7001 if (mEffectChains.size() != 0) { 7002 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7003 return INVALID_OPERATION; 7004 } 7005 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7006 chain->setThread(this); 7007 chain->setInBuffer(NULL); 7008 chain->setOutBuffer(NULL); 7009 7010 checkSuspendOnAddEffectChain_l(chain); 7011 7012 // make sure enabled pre processing effects state is communicated to the HAL as we 7013 // just moved them to a new input stream. 7014 chain->syncHalEffectsState(); 7015 7016 mEffectChains.add(chain); 7017 7018 return NO_ERROR; 7019} 7020 7021size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7022{ 7023 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7024 ALOGW_IF(mEffectChains.size() != 1, 7025 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7026 chain.get(), mEffectChains.size(), this); 7027 if (mEffectChains.size() == 1) { 7028 mEffectChains.removeAt(0); 7029 } 7030 return 0; 7031} 7032 7033status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7034 audio_patch_handle_t *handle) 7035{ 7036 status_t status = NO_ERROR; 7037 7038 // store new device and send to effects 7039 mInDevice = patch->sources[0].ext.device.type; 7040 mPatch = *patch; 7041 for (size_t i = 0; i < mEffectChains.size(); i++) { 7042 mEffectChains[i]->setDevice_l(mInDevice); 7043 } 7044 7045 // disable AEC and NS if the device is a BT SCO headset supporting those 7046 // pre processings 7047 if (mTracks.size() > 0) { 7048 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7049 mAudioFlinger->btNrecIsOff(); 7050 for (size_t i = 0; i < mTracks.size(); i++) { 7051 sp<RecordTrack> track = mTracks[i]; 7052 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7053 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7054 } 7055 } 7056 7057 // store new source and send to effects 7058 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7059 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7060 for (size_t i = 0; i < mEffectChains.size(); i++) { 7061 mEffectChains[i]->setAudioSource_l(mAudioSource); 7062 } 7063 } 7064 7065 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7066 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7067 status = hwDevice->create_audio_patch(hwDevice, 7068 patch->num_sources, 7069 patch->sources, 7070 patch->num_sinks, 7071 patch->sinks, 7072 handle); 7073 } else { 7074 char *address; 7075 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7076 address = audio_device_address_to_parameter( 7077 patch->sources[0].ext.device.type, 7078 patch->sources[0].ext.device.address); 7079 } else { 7080 address = (char *)calloc(1, 1); 7081 } 7082 AudioParameter param = AudioParameter(String8(address)); 7083 free(address); 7084 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7085 (int)patch->sources[0].ext.device.type); 7086 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7087 (int)patch->sinks[0].ext.mix.usecase.source); 7088 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7089 param.toString().string()); 7090 *handle = AUDIO_PATCH_HANDLE_NONE; 7091 } 7092 7093 if (mInDevice != mPrevInDevice) { 7094 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7095 mPrevInDevice = mInDevice; 7096 } 7097 7098 return status; 7099} 7100 7101status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7102{ 7103 status_t status = NO_ERROR; 7104 7105 mInDevice = AUDIO_DEVICE_NONE; 7106 7107 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7108 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7109 status = hwDevice->release_audio_patch(hwDevice, handle); 7110 } else { 7111 AudioParameter param; 7112 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7113 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7114 param.toString().string()); 7115 } 7116 return status; 7117} 7118 7119void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7120{ 7121 Mutex::Autolock _l(mLock); 7122 mTracks.add(record); 7123} 7124 7125void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7126{ 7127 Mutex::Autolock _l(mLock); 7128 destroyTrack_l(record); 7129} 7130 7131void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7132{ 7133 ThreadBase::getAudioPortConfig(config); 7134 config->role = AUDIO_PORT_ROLE_SINK; 7135 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7136 config->ext.mix.usecase.source = mAudioSource; 7137} 7138 7139} // namespace android 7140