Threads.cpp revision f99498ee4de7123e2fd71778c6877be44fbd1506
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 "mediautils/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 128// FIXME This should be based on experimentally observed scheduling jitter 129static const uint32_t kMinNormalSinkBufferSizeMs = 20; 130// maximum normal sink buffer size 131static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 132 133// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 134// FIXME This should be based on experimentally observed scheduling jitter 135static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 136 137// Offloaded output thread standby delay: allows track transition without going to standby 138static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 139 140// Whether to use fast mixer 141static const enum { 142 FastMixer_Never, // never initialize or use: for debugging only 143 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 144 // normal mixer multiplier is 1 145 FastMixer_Static, // initialize if needed, then use all the time if initialized, 146 // multiplier is calculated based on min & max normal mixer buffer size 147 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 148 // multiplier is calculated based on min & max normal mixer buffer size 149 // FIXME for FastMixer_Dynamic: 150 // Supporting this option will require fixing HALs that can't handle large writes. 151 // For example, one HAL implementation returns an error from a large write, 152 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 153 // We could either fix the HAL implementations, or provide a wrapper that breaks 154 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 155} kUseFastMixer = FastMixer_Static; 156 157// Whether to use fast capture 158static const enum { 159 FastCapture_Never, // never initialize or use: for debugging only 160 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 161 FastCapture_Static, // initialize if needed, then use all the time if initialized 162} kUseFastCapture = FastCapture_Static; 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167static const int kPriorityFastCapture = 3; 168 169// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 170// for the track. The client then sub-divides this into smaller buffers for its use. 171// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 172// So for now we just assume that client is double-buffered for fast tracks. 173// FIXME It would be better for client to tell AudioFlinger the value of N, 174// so AudioFlinger could allocate the right amount of memory. 175// See the client's minBufCount and mNotificationFramesAct calculations for details. 176 177// This is the default value, if not specified by property. 178static const int kFastTrackMultiplier = 2; 179 180// The minimum and maximum allowed values 181static const int kFastTrackMultiplierMin = 1; 182static const int kFastTrackMultiplierMax = 2; 183 184// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 185static int sFastTrackMultiplier = kFastTrackMultiplier; 186 187// See Thread::readOnlyHeap(). 188// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 189// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 190// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 191static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 192 193// ---------------------------------------------------------------------------- 194 195static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 196 197static void sFastTrackMultiplierInit() 198{ 199 char value[PROPERTY_VALUE_MAX]; 200 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 201 char *endptr; 202 unsigned long ul = strtoul(value, &endptr, 0); 203 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 204 sFastTrackMultiplier = (int) ul; 205 } 206 } 207} 208 209// ---------------------------------------------------------------------------- 210 211#ifdef ADD_BATTERY_DATA 212// To collect the amplifier usage 213static void addBatteryData(uint32_t params) { 214 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 215 if (service == NULL) { 216 // it already logged 217 return; 218 } 219 220 service->addBatteryData(params); 221} 222#endif 223 224 225// ---------------------------------------------------------------------------- 226// CPU Stats 227// ---------------------------------------------------------------------------- 228 229class CpuStats { 230public: 231 CpuStats(); 232 void sample(const String8 &title); 233#ifdef DEBUG_CPU_USAGE 234private: 235 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 236 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 237 238 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 239 240 int mCpuNum; // thread's current CPU number 241 int mCpukHz; // frequency of thread's current CPU in kHz 242#endif 243}; 244 245CpuStats::CpuStats() 246#ifdef DEBUG_CPU_USAGE 247 : mCpuNum(-1), mCpukHz(-1) 248#endif 249{ 250} 251 252void CpuStats::sample(const String8 &title 253#ifndef DEBUG_CPU_USAGE 254 __unused 255#endif 256 ) { 257#ifdef DEBUG_CPU_USAGE 258 // get current thread's delta CPU time in wall clock ns 259 double wcNs; 260 bool valid = mCpuUsage.sampleAndEnable(wcNs); 261 262 // record sample for wall clock statistics 263 if (valid) { 264 mWcStats.sample(wcNs); 265 } 266 267 // get the current CPU number 268 int cpuNum = sched_getcpu(); 269 270 // get the current CPU frequency in kHz 271 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 272 273 // check if either CPU number or frequency changed 274 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 275 mCpuNum = cpuNum; 276 mCpukHz = cpukHz; 277 // ignore sample for purposes of cycles 278 valid = false; 279 } 280 281 // if no change in CPU number or frequency, then record sample for cycle statistics 282 if (valid && mCpukHz > 0) { 283 double cycles = wcNs * cpukHz * 0.000001; 284 mHzStats.sample(cycles); 285 } 286 287 unsigned n = mWcStats.n(); 288 // mCpuUsage.elapsed() is expensive, so don't call it every loop 289 if ((n & 127) == 1) { 290 long long elapsed = mCpuUsage.elapsed(); 291 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 292 double perLoop = elapsed / (double) n; 293 double perLoop100 = perLoop * 0.01; 294 double perLoop1k = perLoop * 0.001; 295 double mean = mWcStats.mean(); 296 double stddev = mWcStats.stddev(); 297 double minimum = mWcStats.minimum(); 298 double maximum = mWcStats.maximum(); 299 double meanCycles = mHzStats.mean(); 300 double stddevCycles = mHzStats.stddev(); 301 double minCycles = mHzStats.minimum(); 302 double maxCycles = mHzStats.maximum(); 303 mCpuUsage.resetElapsed(); 304 mWcStats.reset(); 305 mHzStats.reset(); 306 ALOGD("CPU usage for %s over past %.1f secs\n" 307 " (%u mixer loops at %.1f mean ms per loop):\n" 308 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 309 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 310 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 311 title.string(), 312 elapsed * .000000001, n, perLoop * .000001, 313 mean * .001, 314 stddev * .001, 315 minimum * .001, 316 maximum * .001, 317 mean / perLoop100, 318 stddev / perLoop100, 319 minimum / perLoop100, 320 maximum / perLoop100, 321 meanCycles / perLoop1k, 322 stddevCycles / perLoop1k, 323 minCycles / perLoop1k, 324 maxCycles / perLoop1k); 325 326 } 327 } 328#endif 329}; 330 331// ---------------------------------------------------------------------------- 332// ThreadBase 333// ---------------------------------------------------------------------------- 334 335// static 336const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 337{ 338 switch (type) { 339 case MIXER: 340 return "MIXER"; 341 case DIRECT: 342 return "DIRECT"; 343 case DUPLICATING: 344 return "DUPLICATING"; 345 case RECORD: 346 return "RECORD"; 347 case OFFLOAD: 348 return "OFFLOAD"; 349 default: 350 return "unknown"; 351 } 352} 353 354String8 devicesToString(audio_devices_t devices) 355{ 356 static const struct mapping { 357 audio_devices_t mDevices; 358 const char * mString; 359 } mappingsOut[] = { 360 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 361 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 362 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 363 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 364 AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO", 365 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 366 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT", 367 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 368 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES, "BLUETOOTH_A2DP_HEADPHONES", 369 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER", 370 AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL", 371 AUDIO_DEVICE_OUT_HDMI, "HDMI", 372 AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 373 AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 374 AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY", 375 AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE", 376 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 377 AUDIO_DEVICE_OUT_LINE, "LINE", 378 AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC", 379 AUDIO_DEVICE_OUT_SPDIF, "SPDIF", 380 AUDIO_DEVICE_OUT_FM, "FM", 381 AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE", 382 AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE", 383 AUDIO_DEVICE_OUT_IP, "IP", 384 AUDIO_DEVICE_NONE, "NONE", // must be last 385 }, mappingsIn[] = { 386 AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION", 387 AUDIO_DEVICE_IN_AMBIENT, "AMBIENT", 388 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 389 AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 390 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 391 AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL", 392 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 393 AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX", 394 AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC", 395 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 396 AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 397 AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 398 AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY", 399 AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE", 400 AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER", 401 AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER", 402 AUDIO_DEVICE_IN_LINE, "LINE", 403 AUDIO_DEVICE_IN_SPDIF, "SPDIF", 404 AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 405 AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK", 406 AUDIO_DEVICE_IN_IP, "IP", 407 AUDIO_DEVICE_NONE, "NONE", // must be last 408 }; 409 String8 result; 410 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 411 const mapping *entry; 412 if (devices & AUDIO_DEVICE_BIT_IN) { 413 devices &= ~AUDIO_DEVICE_BIT_IN; 414 entry = mappingsIn; 415 } else { 416 entry = mappingsOut; 417 } 418 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 419 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 420 if (devices & entry->mDevices) { 421 if (!result.isEmpty()) { 422 result.append("|"); 423 } 424 result.append(entry->mString); 425 } 426 } 427 if (devices & ~allDevices) { 428 if (!result.isEmpty()) { 429 result.append("|"); 430 } 431 result.appendFormat("0x%X", devices & ~allDevices); 432 } 433 if (result.isEmpty()) { 434 result.append(entry->mString); 435 } 436 return result; 437} 438 439String8 inputFlagsToString(audio_input_flags_t flags) 440{ 441 static const struct mapping { 442 audio_input_flags_t mFlag; 443 const char * mString; 444 } mappings[] = { 445 AUDIO_INPUT_FLAG_FAST, "FAST", 446 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 447 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 448 }; 449 String8 result; 450 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 451 const mapping *entry; 452 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 453 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 454 if (flags & entry->mFlag) { 455 if (!result.isEmpty()) { 456 result.append("|"); 457 } 458 result.append(entry->mString); 459 } 460 } 461 if (flags & ~allFlags) { 462 if (!result.isEmpty()) { 463 result.append("|"); 464 } 465 result.appendFormat("0x%X", flags & ~allFlags); 466 } 467 if (result.isEmpty()) { 468 result.append(entry->mString); 469 } 470 return result; 471} 472 473String8 outputFlagsToString(audio_output_flags_t flags) 474{ 475 static const struct mapping { 476 audio_output_flags_t mFlag; 477 const char * mString; 478 } mappings[] = { 479 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 480 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 481 AUDIO_OUTPUT_FLAG_FAST, "FAST", 482 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 483 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 484 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 485 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 486 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 487 }; 488 String8 result; 489 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 490 const mapping *entry; 491 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 492 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 493 if (flags & entry->mFlag) { 494 if (!result.isEmpty()) { 495 result.append("|"); 496 } 497 result.append(entry->mString); 498 } 499 } 500 if (flags & ~allFlags) { 501 if (!result.isEmpty()) { 502 result.append("|"); 503 } 504 result.appendFormat("0x%X", flags & ~allFlags); 505 } 506 if (result.isEmpty()) { 507 result.append(entry->mString); 508 } 509 return result; 510} 511 512const char *sourceToString(audio_source_t source) 513{ 514 switch (source) { 515 case AUDIO_SOURCE_DEFAULT: return "default"; 516 case AUDIO_SOURCE_MIC: return "mic"; 517 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 518 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 519 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 520 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 521 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 522 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 523 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 524 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 525 case AUDIO_SOURCE_HOTWORD: return "hotword"; 526 default: return "unknown"; 527 } 528} 529 530AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 531 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 532 : Thread(false /*canCallJava*/), 533 mType(type), 534 mAudioFlinger(audioFlinger), 535 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 536 // are set by PlaybackThread::readOutputParameters_l() or 537 // RecordThread::readInputParameters_l() 538 //FIXME: mStandby should be true here. Is this some kind of hack? 539 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 540 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 541 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 542 // mName will be set by concrete (non-virtual) subclass 543 mDeathRecipient(new PMDeathRecipient(this)), 544 mSystemReady(systemReady) 545{ 546 memset(&mPatch, 0, sizeof(struct audio_patch)); 547} 548 549AudioFlinger::ThreadBase::~ThreadBase() 550{ 551 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 552 mConfigEvents.clear(); 553 554 // do not lock the mutex in destructor 555 releaseWakeLock_l(); 556 if (mPowerManager != 0) { 557 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 558 binder->unlinkToDeath(mDeathRecipient); 559 } 560} 561 562status_t AudioFlinger::ThreadBase::readyToRun() 563{ 564 status_t status = initCheck(); 565 if (status == NO_ERROR) { 566 ALOGI("AudioFlinger's thread %p ready to run", this); 567 } else { 568 ALOGE("No working audio driver found."); 569 } 570 return status; 571} 572 573void AudioFlinger::ThreadBase::exit() 574{ 575 ALOGV("ThreadBase::exit"); 576 // do any cleanup required for exit to succeed 577 preExit(); 578 { 579 // This lock prevents the following race in thread (uniprocessor for illustration): 580 // if (!exitPending()) { 581 // // context switch from here to exit() 582 // // exit() calls requestExit(), what exitPending() observes 583 // // exit() calls signal(), which is dropped since no waiters 584 // // context switch back from exit() to here 585 // mWaitWorkCV.wait(...); 586 // // now thread is hung 587 // } 588 AutoMutex lock(mLock); 589 requestExit(); 590 mWaitWorkCV.broadcast(); 591 } 592 // When Thread::requestExitAndWait is made virtual and this method is renamed to 593 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 594 requestExitAndWait(); 595} 596 597status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 598{ 599 status_t status; 600 601 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 602 Mutex::Autolock _l(mLock); 603 604 return sendSetParameterConfigEvent_l(keyValuePairs); 605} 606 607// sendConfigEvent_l() must be called with ThreadBase::mLock held 608// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 609status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 610{ 611 status_t status = NO_ERROR; 612 613 if (event->mRequiresSystemReady && !mSystemReady) { 614 event->mWaitStatus = false; 615 mPendingConfigEvents.add(event); 616 return status; 617 } 618 mConfigEvents.add(event); 619 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 620 mWaitWorkCV.signal(); 621 mLock.unlock(); 622 { 623 Mutex::Autolock _l(event->mLock); 624 while (event->mWaitStatus) { 625 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 626 event->mStatus = TIMED_OUT; 627 event->mWaitStatus = false; 628 } 629 } 630 status = event->mStatus; 631 } 632 mLock.lock(); 633 return status; 634} 635 636void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 637{ 638 Mutex::Autolock _l(mLock); 639 sendIoConfigEvent_l(event, pid); 640} 641 642// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 643void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 644{ 645 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 646 sendConfigEvent_l(configEvent); 647} 648 649void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 650{ 651 Mutex::Autolock _l(mLock); 652 sendPrioConfigEvent_l(pid, tid, prio); 653} 654 655// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 656void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 657{ 658 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 659 sendConfigEvent_l(configEvent); 660} 661 662// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 663status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 664{ 665 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 666 return sendConfigEvent_l(configEvent); 667} 668 669status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 670 const struct audio_patch *patch, 671 audio_patch_handle_t *handle) 672{ 673 Mutex::Autolock _l(mLock); 674 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 675 status_t status = sendConfigEvent_l(configEvent); 676 if (status == NO_ERROR) { 677 CreateAudioPatchConfigEventData *data = 678 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 679 *handle = data->mHandle; 680 } 681 return status; 682} 683 684status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 685 const audio_patch_handle_t handle) 686{ 687 Mutex::Autolock _l(mLock); 688 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 689 return sendConfigEvent_l(configEvent); 690} 691 692 693// post condition: mConfigEvents.isEmpty() 694void AudioFlinger::ThreadBase::processConfigEvents_l() 695{ 696 bool configChanged = false; 697 698 while (!mConfigEvents.isEmpty()) { 699 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 700 sp<ConfigEvent> event = mConfigEvents[0]; 701 mConfigEvents.removeAt(0); 702 switch (event->mType) { 703 case CFG_EVENT_PRIO: { 704 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 705 // FIXME Need to understand why this has to be done asynchronously 706 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 707 true /*asynchronous*/); 708 if (err != 0) { 709 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 710 data->mPrio, data->mPid, data->mTid, err); 711 } 712 } break; 713 case CFG_EVENT_IO: { 714 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 715 ioConfigChanged(data->mEvent, data->mPid); 716 } break; 717 case CFG_EVENT_SET_PARAMETER: { 718 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 719 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 720 configChanged = true; 721 } 722 } break; 723 case CFG_EVENT_CREATE_AUDIO_PATCH: { 724 CreateAudioPatchConfigEventData *data = 725 (CreateAudioPatchConfigEventData *)event->mData.get(); 726 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 727 } break; 728 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 729 ReleaseAudioPatchConfigEventData *data = 730 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 731 event->mStatus = releaseAudioPatch_l(data->mHandle); 732 } break; 733 default: 734 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 735 break; 736 } 737 { 738 Mutex::Autolock _l(event->mLock); 739 if (event->mWaitStatus) { 740 event->mWaitStatus = false; 741 event->mCond.signal(); 742 } 743 } 744 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 745 } 746 747 if (configChanged) { 748 cacheParameters_l(); 749 } 750} 751 752String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 753 String8 s; 754 const audio_channel_representation_t representation = 755 audio_channel_mask_get_representation(mask); 756 757 switch (representation) { 758 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 759 if (output) { 760 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 761 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 762 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 763 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 764 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 765 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 766 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 767 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 768 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 769 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 770 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 771 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 772 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 773 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 774 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 775 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 776 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 777 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 778 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 779 } else { 780 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 781 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 782 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 783 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 784 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 785 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 786 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 787 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 788 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 789 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 790 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 791 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 792 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 793 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 794 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 795 } 796 const int len = s.length(); 797 if (len > 2) { 798 char *str = s.lockBuffer(len); // needed? 799 s.unlockBuffer(len - 2); // remove trailing ", " 800 } 801 return s; 802 } 803 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 804 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 805 return s; 806 default: 807 s.appendFormat("unknown mask, representation:%d bits:%#x", 808 representation, audio_channel_mask_get_bits(mask)); 809 return s; 810 } 811} 812 813void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 814{ 815 const size_t SIZE = 256; 816 char buffer[SIZE]; 817 String8 result; 818 819 bool locked = AudioFlinger::dumpTryLock(mLock); 820 if (!locked) { 821 dprintf(fd, "thread %p may be deadlocked\n", this); 822 } 823 824 dprintf(fd, " Thread name: %s\n", mThreadName); 825 dprintf(fd, " I/O handle: %d\n", mId); 826 dprintf(fd, " TID: %d\n", getTid()); 827 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 828 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 829 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 830 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 831 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 832 dprintf(fd, " Channel count: %u\n", mChannelCount); 833 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 834 channelMaskToString(mChannelMask, mType != RECORD).string()); 835 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 836 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 837 dprintf(fd, " Pending config events:"); 838 size_t numConfig = mConfigEvents.size(); 839 if (numConfig) { 840 for (size_t i = 0; i < numConfig; i++) { 841 mConfigEvents[i]->dump(buffer, SIZE); 842 dprintf(fd, "\n %s", buffer); 843 } 844 dprintf(fd, "\n"); 845 } else { 846 dprintf(fd, " none\n"); 847 } 848 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 849 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 850 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 851 852 if (locked) { 853 mLock.unlock(); 854 } 855} 856 857void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 858{ 859 const size_t SIZE = 256; 860 char buffer[SIZE]; 861 String8 result; 862 863 size_t numEffectChains = mEffectChains.size(); 864 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 865 write(fd, buffer, strlen(buffer)); 866 867 for (size_t i = 0; i < numEffectChains; ++i) { 868 sp<EffectChain> chain = mEffectChains[i]; 869 if (chain != 0) { 870 chain->dump(fd, args); 871 } 872 } 873} 874 875void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 876{ 877 Mutex::Autolock _l(mLock); 878 acquireWakeLock_l(uid); 879} 880 881String16 AudioFlinger::ThreadBase::getWakeLockTag() 882{ 883 switch (mType) { 884 case MIXER: 885 return String16("AudioMix"); 886 case DIRECT: 887 return String16("AudioDirectOut"); 888 case DUPLICATING: 889 return String16("AudioDup"); 890 case RECORD: 891 return String16("AudioIn"); 892 case OFFLOAD: 893 return String16("AudioOffload"); 894 default: 895 ALOG_ASSERT(false); 896 return String16("AudioUnknown"); 897 } 898} 899 900void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 901{ 902 getPowerManager_l(); 903 if (mPowerManager != 0) { 904 sp<IBinder> binder = new BBinder(); 905 status_t status; 906 if (uid >= 0) { 907 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 908 binder, 909 getWakeLockTag(), 910 String16("media"), 911 uid, 912 true /* FIXME force oneway contrary to .aidl */); 913 } else { 914 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 915 binder, 916 getWakeLockTag(), 917 String16("media"), 918 true /* FIXME force oneway contrary to .aidl */); 919 } 920 if (status == NO_ERROR) { 921 mWakeLockToken = binder; 922 } 923 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 924 } 925} 926 927void AudioFlinger::ThreadBase::releaseWakeLock() 928{ 929 Mutex::Autolock _l(mLock); 930 releaseWakeLock_l(); 931} 932 933void AudioFlinger::ThreadBase::releaseWakeLock_l() 934{ 935 if (mWakeLockToken != 0) { 936 ALOGV("releaseWakeLock_l() %s", mThreadName); 937 if (mPowerManager != 0) { 938 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 939 true /* FIXME force oneway contrary to .aidl */); 940 } 941 mWakeLockToken.clear(); 942 } 943} 944 945void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 946 Mutex::Autolock _l(mLock); 947 updateWakeLockUids_l(uids); 948} 949 950void AudioFlinger::ThreadBase::getPowerManager_l() { 951 if (mSystemReady && mPowerManager == 0) { 952 // use checkService() to avoid blocking if power service is not up yet 953 sp<IBinder> binder = 954 defaultServiceManager()->checkService(String16("power")); 955 if (binder == 0) { 956 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 957 } else { 958 mPowerManager = interface_cast<IPowerManager>(binder); 959 binder->linkToDeath(mDeathRecipient); 960 } 961 } 962} 963 964void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 965 getPowerManager_l(); 966 if (mWakeLockToken == NULL) { 967 ALOGE("no wake lock to update!"); 968 return; 969 } 970 if (mPowerManager != 0) { 971 sp<IBinder> binder = new BBinder(); 972 status_t status; 973 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 974 true /* FIXME force oneway contrary to .aidl */); 975 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 976 } 977} 978 979void AudioFlinger::ThreadBase::clearPowerManager() 980{ 981 Mutex::Autolock _l(mLock); 982 releaseWakeLock_l(); 983 mPowerManager.clear(); 984} 985 986void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 987{ 988 sp<ThreadBase> thread = mThread.promote(); 989 if (thread != 0) { 990 thread->clearPowerManager(); 991 } 992 ALOGW("power manager service died !!!"); 993} 994 995void AudioFlinger::ThreadBase::setEffectSuspended( 996 const effect_uuid_t *type, bool suspend, int sessionId) 997{ 998 Mutex::Autolock _l(mLock); 999 setEffectSuspended_l(type, suspend, sessionId); 1000} 1001 1002void AudioFlinger::ThreadBase::setEffectSuspended_l( 1003 const effect_uuid_t *type, bool suspend, int sessionId) 1004{ 1005 sp<EffectChain> chain = getEffectChain_l(sessionId); 1006 if (chain != 0) { 1007 if (type != NULL) { 1008 chain->setEffectSuspended_l(type, suspend); 1009 } else { 1010 chain->setEffectSuspendedAll_l(suspend); 1011 } 1012 } 1013 1014 updateSuspendedSessions_l(type, suspend, sessionId); 1015} 1016 1017void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1018{ 1019 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1020 if (index < 0) { 1021 return; 1022 } 1023 1024 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1025 mSuspendedSessions.valueAt(index); 1026 1027 for (size_t i = 0; i < sessionEffects.size(); i++) { 1028 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1029 for (int j = 0; j < desc->mRefCount; j++) { 1030 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1031 chain->setEffectSuspendedAll_l(true); 1032 } else { 1033 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1034 desc->mType.timeLow); 1035 chain->setEffectSuspended_l(&desc->mType, true); 1036 } 1037 } 1038 } 1039} 1040 1041void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1042 bool suspend, 1043 int sessionId) 1044{ 1045 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1046 1047 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1048 1049 if (suspend) { 1050 if (index >= 0) { 1051 sessionEffects = mSuspendedSessions.valueAt(index); 1052 } else { 1053 mSuspendedSessions.add(sessionId, sessionEffects); 1054 } 1055 } else { 1056 if (index < 0) { 1057 return; 1058 } 1059 sessionEffects = mSuspendedSessions.valueAt(index); 1060 } 1061 1062 1063 int key = EffectChain::kKeyForSuspendAll; 1064 if (type != NULL) { 1065 key = type->timeLow; 1066 } 1067 index = sessionEffects.indexOfKey(key); 1068 1069 sp<SuspendedSessionDesc> desc; 1070 if (suspend) { 1071 if (index >= 0) { 1072 desc = sessionEffects.valueAt(index); 1073 } else { 1074 desc = new SuspendedSessionDesc(); 1075 if (type != NULL) { 1076 desc->mType = *type; 1077 } 1078 sessionEffects.add(key, desc); 1079 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1080 } 1081 desc->mRefCount++; 1082 } else { 1083 if (index < 0) { 1084 return; 1085 } 1086 desc = sessionEffects.valueAt(index); 1087 if (--desc->mRefCount == 0) { 1088 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1089 sessionEffects.removeItemsAt(index); 1090 if (sessionEffects.isEmpty()) { 1091 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1092 sessionId); 1093 mSuspendedSessions.removeItem(sessionId); 1094 } 1095 } 1096 } 1097 if (!sessionEffects.isEmpty()) { 1098 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1099 } 1100} 1101 1102void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1103 bool enabled, 1104 int sessionId) 1105{ 1106 Mutex::Autolock _l(mLock); 1107 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1108} 1109 1110void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1111 bool enabled, 1112 int sessionId) 1113{ 1114 if (mType != RECORD) { 1115 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1116 // another session. This gives the priority to well behaved effect control panels 1117 // and applications not using global effects. 1118 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1119 // global effects 1120 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1121 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1122 } 1123 } 1124 1125 sp<EffectChain> chain = getEffectChain_l(sessionId); 1126 if (chain != 0) { 1127 chain->checkSuspendOnEffectEnabled(effect, enabled); 1128 } 1129} 1130 1131// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1132sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1133 const sp<AudioFlinger::Client>& client, 1134 const sp<IEffectClient>& effectClient, 1135 int32_t priority, 1136 int sessionId, 1137 effect_descriptor_t *desc, 1138 int *enabled, 1139 status_t *status) 1140{ 1141 sp<EffectModule> effect; 1142 sp<EffectHandle> handle; 1143 status_t lStatus; 1144 sp<EffectChain> chain; 1145 bool chainCreated = false; 1146 bool effectCreated = false; 1147 bool effectRegistered = false; 1148 1149 lStatus = initCheck(); 1150 if (lStatus != NO_ERROR) { 1151 ALOGW("createEffect_l() Audio driver not initialized."); 1152 goto Exit; 1153 } 1154 1155 // Reject any effect on Direct output threads for now, since the format of 1156 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1157 if (mType == DIRECT) { 1158 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1159 desc->name, mThreadName); 1160 lStatus = BAD_VALUE; 1161 goto Exit; 1162 } 1163 1164 // Reject any effect on mixer or duplicating multichannel sinks. 1165 // TODO: fix both format and multichannel issues with effects. 1166 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1167 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1168 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1169 lStatus = BAD_VALUE; 1170 goto Exit; 1171 } 1172 1173 // Allow global effects only on offloaded and mixer threads 1174 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1175 switch (mType) { 1176 case MIXER: 1177 case OFFLOAD: 1178 break; 1179 case DIRECT: 1180 case DUPLICATING: 1181 case RECORD: 1182 default: 1183 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1184 desc->name, mThreadName); 1185 lStatus = BAD_VALUE; 1186 goto Exit; 1187 } 1188 } 1189 1190 // Only Pre processor effects are allowed on input threads and only on input threads 1191 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1192 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1193 desc->name, desc->flags, mType); 1194 lStatus = BAD_VALUE; 1195 goto Exit; 1196 } 1197 1198 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1199 1200 { // scope for mLock 1201 Mutex::Autolock _l(mLock); 1202 1203 // check for existing effect chain with the requested audio session 1204 chain = getEffectChain_l(sessionId); 1205 if (chain == 0) { 1206 // create a new chain for this session 1207 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1208 chain = new EffectChain(this, sessionId); 1209 addEffectChain_l(chain); 1210 chain->setStrategy(getStrategyForSession_l(sessionId)); 1211 chainCreated = true; 1212 } else { 1213 effect = chain->getEffectFromDesc_l(desc); 1214 } 1215 1216 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1217 1218 if (effect == 0) { 1219 int id = mAudioFlinger->nextUniqueId(); 1220 // Check CPU and memory usage 1221 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1222 if (lStatus != NO_ERROR) { 1223 goto Exit; 1224 } 1225 effectRegistered = true; 1226 // create a new effect module if none present in the chain 1227 effect = new EffectModule(this, chain, desc, id, sessionId); 1228 lStatus = effect->status(); 1229 if (lStatus != NO_ERROR) { 1230 goto Exit; 1231 } 1232 effect->setOffloaded(mType == OFFLOAD, mId); 1233 1234 lStatus = chain->addEffect_l(effect); 1235 if (lStatus != NO_ERROR) { 1236 goto Exit; 1237 } 1238 effectCreated = true; 1239 1240 effect->setDevice(mOutDevice); 1241 effect->setDevice(mInDevice); 1242 effect->setMode(mAudioFlinger->getMode()); 1243 effect->setAudioSource(mAudioSource); 1244 } 1245 // create effect handle and connect it to effect module 1246 handle = new EffectHandle(effect, client, effectClient, priority); 1247 lStatus = handle->initCheck(); 1248 if (lStatus == OK) { 1249 lStatus = effect->addHandle(handle.get()); 1250 } 1251 if (enabled != NULL) { 1252 *enabled = (int)effect->isEnabled(); 1253 } 1254 } 1255 1256Exit: 1257 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1258 Mutex::Autolock _l(mLock); 1259 if (effectCreated) { 1260 chain->removeEffect_l(effect); 1261 } 1262 if (effectRegistered) { 1263 AudioSystem::unregisterEffect(effect->id()); 1264 } 1265 if (chainCreated) { 1266 removeEffectChain_l(chain); 1267 } 1268 handle.clear(); 1269 } 1270 1271 *status = lStatus; 1272 return handle; 1273} 1274 1275sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1276{ 1277 Mutex::Autolock _l(mLock); 1278 return getEffect_l(sessionId, effectId); 1279} 1280 1281sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1282{ 1283 sp<EffectChain> chain = getEffectChain_l(sessionId); 1284 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1285} 1286 1287// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1288// PlaybackThread::mLock held 1289status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1290{ 1291 // check for existing effect chain with the requested audio session 1292 int sessionId = effect->sessionId(); 1293 sp<EffectChain> chain = getEffectChain_l(sessionId); 1294 bool chainCreated = false; 1295 1296 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1297 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1298 this, effect->desc().name, effect->desc().flags); 1299 1300 if (chain == 0) { 1301 // create a new chain for this session 1302 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1303 chain = new EffectChain(this, sessionId); 1304 addEffectChain_l(chain); 1305 chain->setStrategy(getStrategyForSession_l(sessionId)); 1306 chainCreated = true; 1307 } 1308 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1309 1310 if (chain->getEffectFromId_l(effect->id()) != 0) { 1311 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1312 this, effect->desc().name, chain.get()); 1313 return BAD_VALUE; 1314 } 1315 1316 effect->setOffloaded(mType == OFFLOAD, mId); 1317 1318 status_t status = chain->addEffect_l(effect); 1319 if (status != NO_ERROR) { 1320 if (chainCreated) { 1321 removeEffectChain_l(chain); 1322 } 1323 return status; 1324 } 1325 1326 effect->setDevice(mOutDevice); 1327 effect->setDevice(mInDevice); 1328 effect->setMode(mAudioFlinger->getMode()); 1329 effect->setAudioSource(mAudioSource); 1330 return NO_ERROR; 1331} 1332 1333void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1334 1335 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1336 effect_descriptor_t desc = effect->desc(); 1337 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1338 detachAuxEffect_l(effect->id()); 1339 } 1340 1341 sp<EffectChain> chain = effect->chain().promote(); 1342 if (chain != 0) { 1343 // remove effect chain if removing last effect 1344 if (chain->removeEffect_l(effect) == 0) { 1345 removeEffectChain_l(chain); 1346 } 1347 } else { 1348 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1349 } 1350} 1351 1352void AudioFlinger::ThreadBase::lockEffectChains_l( 1353 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1354{ 1355 effectChains = mEffectChains; 1356 for (size_t i = 0; i < mEffectChains.size(); i++) { 1357 mEffectChains[i]->lock(); 1358 } 1359} 1360 1361void AudioFlinger::ThreadBase::unlockEffectChains( 1362 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1363{ 1364 for (size_t i = 0; i < effectChains.size(); i++) { 1365 effectChains[i]->unlock(); 1366 } 1367} 1368 1369sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1370{ 1371 Mutex::Autolock _l(mLock); 1372 return getEffectChain_l(sessionId); 1373} 1374 1375sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1376{ 1377 size_t size = mEffectChains.size(); 1378 for (size_t i = 0; i < size; i++) { 1379 if (mEffectChains[i]->sessionId() == sessionId) { 1380 return mEffectChains[i]; 1381 } 1382 } 1383 return 0; 1384} 1385 1386void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1387{ 1388 Mutex::Autolock _l(mLock); 1389 size_t size = mEffectChains.size(); 1390 for (size_t i = 0; i < size; i++) { 1391 mEffectChains[i]->setMode_l(mode); 1392 } 1393} 1394 1395void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1396{ 1397 config->type = AUDIO_PORT_TYPE_MIX; 1398 config->ext.mix.handle = mId; 1399 config->sample_rate = mSampleRate; 1400 config->format = mFormat; 1401 config->channel_mask = mChannelMask; 1402 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1403 AUDIO_PORT_CONFIG_FORMAT; 1404} 1405 1406void AudioFlinger::ThreadBase::systemReady() 1407{ 1408 Mutex::Autolock _l(mLock); 1409 if (mSystemReady) { 1410 return; 1411 } 1412 mSystemReady = true; 1413 1414 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1415 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1416 } 1417 mPendingConfigEvents.clear(); 1418} 1419 1420 1421// ---------------------------------------------------------------------------- 1422// Playback 1423// ---------------------------------------------------------------------------- 1424 1425AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1426 AudioStreamOut* output, 1427 audio_io_handle_t id, 1428 audio_devices_t device, 1429 type_t type, 1430 bool systemReady) 1431 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1432 mNormalFrameCount(0), mSinkBuffer(NULL), 1433 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1434 mMixerBuffer(NULL), 1435 mMixerBufferSize(0), 1436 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1437 mMixerBufferValid(false), 1438 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1439 mEffectBuffer(NULL), 1440 mEffectBufferSize(0), 1441 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1442 mEffectBufferValid(false), 1443 mSuspended(0), mBytesWritten(0), 1444 mActiveTracksGeneration(0), 1445 // mStreamTypes[] initialized in constructor body 1446 mOutput(output), 1447 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1448 mMixerStatus(MIXER_IDLE), 1449 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1450 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1451 mBytesRemaining(0), 1452 mCurrentWriteLength(0), 1453 mUseAsyncWrite(false), 1454 mWriteAckSequence(0), 1455 mDrainSequence(0), 1456 mSignalPending(false), 1457 mScreenState(AudioFlinger::mScreenState), 1458 // index 0 is reserved for normal mixer's submix 1459 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1460 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1461 // mLatchD, mLatchQ, 1462 mLatchDValid(false), mLatchQValid(false) 1463{ 1464 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1465 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1466 1467 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1468 // it would be safer to explicitly pass initial masterVolume/masterMute as 1469 // parameter. 1470 // 1471 // If the HAL we are using has support for master volume or master mute, 1472 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1473 // and the mute set to false). 1474 mMasterVolume = audioFlinger->masterVolume_l(); 1475 mMasterMute = audioFlinger->masterMute_l(); 1476 if (mOutput && mOutput->audioHwDev) { 1477 if (mOutput->audioHwDev->canSetMasterVolume()) { 1478 mMasterVolume = 1.0; 1479 } 1480 1481 if (mOutput->audioHwDev->canSetMasterMute()) { 1482 mMasterMute = false; 1483 } 1484 } 1485 1486 readOutputParameters_l(); 1487 1488 // ++ operator does not compile 1489 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1490 stream = (audio_stream_type_t) (stream + 1)) { 1491 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1492 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1493 } 1494} 1495 1496AudioFlinger::PlaybackThread::~PlaybackThread() 1497{ 1498 mAudioFlinger->unregisterWriter(mNBLogWriter); 1499 free(mSinkBuffer); 1500 free(mMixerBuffer); 1501 free(mEffectBuffer); 1502} 1503 1504void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1505{ 1506 dumpInternals(fd, args); 1507 dumpTracks(fd, args); 1508 dumpEffectChains(fd, args); 1509} 1510 1511void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1512{ 1513 const size_t SIZE = 256; 1514 char buffer[SIZE]; 1515 String8 result; 1516 1517 result.appendFormat(" Stream volumes in dB: "); 1518 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1519 const stream_type_t *st = &mStreamTypes[i]; 1520 if (i > 0) { 1521 result.appendFormat(", "); 1522 } 1523 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1524 if (st->mute) { 1525 result.append("M"); 1526 } 1527 } 1528 result.append("\n"); 1529 write(fd, result.string(), result.length()); 1530 result.clear(); 1531 1532 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1533 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1534 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1535 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1536 1537 size_t numtracks = mTracks.size(); 1538 size_t numactive = mActiveTracks.size(); 1539 dprintf(fd, " %d Tracks", numtracks); 1540 size_t numactiveseen = 0; 1541 if (numtracks) { 1542 dprintf(fd, " of which %d are active\n", numactive); 1543 Track::appendDumpHeader(result); 1544 for (size_t i = 0; i < numtracks; ++i) { 1545 sp<Track> track = mTracks[i]; 1546 if (track != 0) { 1547 bool active = mActiveTracks.indexOf(track) >= 0; 1548 if (active) { 1549 numactiveseen++; 1550 } 1551 track->dump(buffer, SIZE, active); 1552 result.append(buffer); 1553 } 1554 } 1555 } else { 1556 result.append("\n"); 1557 } 1558 if (numactiveseen != numactive) { 1559 // some tracks in the active list were not in the tracks list 1560 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1561 " not in the track list\n"); 1562 result.append(buffer); 1563 Track::appendDumpHeader(result); 1564 for (size_t i = 0; i < numactive; ++i) { 1565 sp<Track> track = mActiveTracks[i].promote(); 1566 if (track != 0 && mTracks.indexOf(track) < 0) { 1567 track->dump(buffer, SIZE, true); 1568 result.append(buffer); 1569 } 1570 } 1571 } 1572 1573 write(fd, result.string(), result.size()); 1574} 1575 1576void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1577{ 1578 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1579 1580 dumpBase(fd, args); 1581 1582 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1583 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1584 dprintf(fd, " Total writes: %d\n", mNumWrites); 1585 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1586 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1587 dprintf(fd, " Suspend count: %d\n", mSuspended); 1588 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1589 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1590 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1591 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1592 AudioStreamOut *output = mOutput; 1593 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1594 String8 flagsAsString = outputFlagsToString(flags); 1595 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1596} 1597 1598// Thread virtuals 1599 1600void AudioFlinger::PlaybackThread::onFirstRef() 1601{ 1602 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1603} 1604 1605// ThreadBase virtuals 1606void AudioFlinger::PlaybackThread::preExit() 1607{ 1608 ALOGV(" preExit()"); 1609 // FIXME this is using hard-coded strings but in the future, this functionality will be 1610 // converted to use audio HAL extensions required to support tunneling 1611 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1612} 1613 1614// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1615sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1616 const sp<AudioFlinger::Client>& client, 1617 audio_stream_type_t streamType, 1618 uint32_t sampleRate, 1619 audio_format_t format, 1620 audio_channel_mask_t channelMask, 1621 size_t *pFrameCount, 1622 const sp<IMemory>& sharedBuffer, 1623 int sessionId, 1624 IAudioFlinger::track_flags_t *flags, 1625 pid_t tid, 1626 int uid, 1627 status_t *status) 1628{ 1629 size_t frameCount = *pFrameCount; 1630 sp<Track> track; 1631 status_t lStatus; 1632 1633 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1634 1635 // client expresses a preference for FAST, but we get the final say 1636 if (*flags & IAudioFlinger::TRACK_FAST) { 1637 if ( 1638 // not timed 1639 (!isTimed) && 1640 // either of these use cases: 1641 ( 1642 // use case 1: shared buffer with any frame count 1643 ( 1644 (sharedBuffer != 0) 1645 ) || 1646 // use case 2: frame count is default or at least as large as HAL 1647 ( 1648 // we formerly checked for a callback handler (non-0 tid), 1649 // but that is no longer required for TRANSFER_OBTAIN mode 1650 ((frameCount == 0) || 1651 (frameCount >= mFrameCount)) 1652 ) 1653 ) && 1654 // PCM data 1655 audio_is_linear_pcm(format) && 1656 // TODO: extract as a data library function that checks that a computationally 1657 // expensive downmixer is not required: isFastOutputChannelConversion() 1658 (channelMask == mChannelMask || 1659 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1660 (channelMask == AUDIO_CHANNEL_OUT_MONO 1661 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1662 // hardware sample rate 1663 (sampleRate == mSampleRate) && 1664 // normal mixer has an associated fast mixer 1665 hasFastMixer() && 1666 // there are sufficient fast track slots available 1667 (mFastTrackAvailMask != 0) 1668 // FIXME test that MixerThread for this fast track has a capable output HAL 1669 // FIXME add a permission test also? 1670 ) { 1671 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1672 if (frameCount == 0) { 1673 // read the fast track multiplier property the first time it is needed 1674 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1675 if (ok != 0) { 1676 ALOGE("%s pthread_once failed: %d", __func__, ok); 1677 } 1678 frameCount = mFrameCount * sFastTrackMultiplier; 1679 } 1680 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1681 frameCount, mFrameCount); 1682 } else { 1683 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1684 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1685 "sampleRate=%u mSampleRate=%u " 1686 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1687 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1688 audio_is_linear_pcm(format), 1689 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1690 *flags &= ~IAudioFlinger::TRACK_FAST; 1691 } 1692 } 1693 // For normal PCM streaming tracks, update minimum frame count. 1694 // For compatibility with AudioTrack calculation, buffer depth is forced 1695 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1696 // This is probably too conservative, but legacy application code may depend on it. 1697 // If you change this calculation, also review the start threshold which is related. 1698 if (!(*flags & IAudioFlinger::TRACK_FAST) 1699 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1700 // this must match AudioTrack.cpp calculateMinFrameCount(). 1701 // TODO: Move to a common library 1702 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1703 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1704 if (minBufCount < 2) { 1705 minBufCount = 2; 1706 } 1707 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1708 // or the client should compute and pass in a larger buffer request. 1709 size_t minFrameCount = 1710 minBufCount * sourceFramesNeededWithTimestretch( 1711 sampleRate, mNormalFrameCount, 1712 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1713 if (frameCount < minFrameCount) { // including frameCount == 0 1714 frameCount = minFrameCount; 1715 } 1716 } 1717 *pFrameCount = frameCount; 1718 1719 switch (mType) { 1720 1721 case DIRECT: 1722 if (audio_is_linear_pcm(format)) { 1723 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1724 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1725 "for output %p with format %#x", 1726 sampleRate, format, channelMask, mOutput, mFormat); 1727 lStatus = BAD_VALUE; 1728 goto Exit; 1729 } 1730 } 1731 break; 1732 1733 case OFFLOAD: 1734 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1735 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1736 "for output %p with format %#x", 1737 sampleRate, format, channelMask, mOutput, mFormat); 1738 lStatus = BAD_VALUE; 1739 goto Exit; 1740 } 1741 break; 1742 1743 default: 1744 if (!audio_is_linear_pcm(format)) { 1745 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1746 "for output %p with format %#x", 1747 format, mOutput, mFormat); 1748 lStatus = BAD_VALUE; 1749 goto Exit; 1750 } 1751 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1752 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1753 lStatus = BAD_VALUE; 1754 goto Exit; 1755 } 1756 break; 1757 1758 } 1759 1760 lStatus = initCheck(); 1761 if (lStatus != NO_ERROR) { 1762 ALOGE("createTrack_l() audio driver not initialized"); 1763 goto Exit; 1764 } 1765 1766 { // scope for mLock 1767 Mutex::Autolock _l(mLock); 1768 1769 // all tracks in same audio session must share the same routing strategy otherwise 1770 // conflicts will happen when tracks are moved from one output to another by audio policy 1771 // manager 1772 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1773 for (size_t i = 0; i < mTracks.size(); ++i) { 1774 sp<Track> t = mTracks[i]; 1775 if (t != 0 && t->isExternalTrack()) { 1776 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1777 if (sessionId == t->sessionId() && strategy != actual) { 1778 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1779 strategy, actual); 1780 lStatus = BAD_VALUE; 1781 goto Exit; 1782 } 1783 } 1784 } 1785 1786 if (!isTimed) { 1787 track = new Track(this, client, streamType, sampleRate, format, 1788 channelMask, frameCount, NULL, sharedBuffer, 1789 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1790 } else { 1791 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1792 channelMask, frameCount, sharedBuffer, sessionId, uid); 1793 } 1794 1795 // new Track always returns non-NULL, 1796 // but TimedTrack::create() is a factory that could fail by returning NULL 1797 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1798 if (lStatus != NO_ERROR) { 1799 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1800 // track must be cleared from the caller as the caller has the AF lock 1801 goto Exit; 1802 } 1803 mTracks.add(track); 1804 1805 sp<EffectChain> chain = getEffectChain_l(sessionId); 1806 if (chain != 0) { 1807 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1808 track->setMainBuffer(chain->inBuffer()); 1809 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1810 chain->incTrackCnt(); 1811 } 1812 1813 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1814 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1815 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1816 // so ask activity manager to do this on our behalf 1817 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1818 } 1819 } 1820 1821 lStatus = NO_ERROR; 1822 1823Exit: 1824 *status = lStatus; 1825 return track; 1826} 1827 1828uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1829{ 1830 return latency; 1831} 1832 1833uint32_t AudioFlinger::PlaybackThread::latency() const 1834{ 1835 Mutex::Autolock _l(mLock); 1836 return latency_l(); 1837} 1838uint32_t AudioFlinger::PlaybackThread::latency_l() const 1839{ 1840 if (initCheck() == NO_ERROR) { 1841 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1842 } else { 1843 return 0; 1844 } 1845} 1846 1847void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1848{ 1849 Mutex::Autolock _l(mLock); 1850 // Don't apply master volume in SW if our HAL can do it for us. 1851 if (mOutput && mOutput->audioHwDev && 1852 mOutput->audioHwDev->canSetMasterVolume()) { 1853 mMasterVolume = 1.0; 1854 } else { 1855 mMasterVolume = value; 1856 } 1857} 1858 1859void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1860{ 1861 Mutex::Autolock _l(mLock); 1862 // Don't apply master mute in SW if our HAL can do it for us. 1863 if (mOutput && mOutput->audioHwDev && 1864 mOutput->audioHwDev->canSetMasterMute()) { 1865 mMasterMute = false; 1866 } else { 1867 mMasterMute = muted; 1868 } 1869} 1870 1871void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1872{ 1873 Mutex::Autolock _l(mLock); 1874 mStreamTypes[stream].volume = value; 1875 broadcast_l(); 1876} 1877 1878void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1879{ 1880 Mutex::Autolock _l(mLock); 1881 mStreamTypes[stream].mute = muted; 1882 broadcast_l(); 1883} 1884 1885float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1886{ 1887 Mutex::Autolock _l(mLock); 1888 return mStreamTypes[stream].volume; 1889} 1890 1891// addTrack_l() must be called with ThreadBase::mLock held 1892status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1893{ 1894 status_t status = ALREADY_EXISTS; 1895 1896 // set retry count for buffer fill 1897 track->mRetryCount = kMaxTrackStartupRetries; 1898 if (mActiveTracks.indexOf(track) < 0) { 1899 // the track is newly added, make sure it fills up all its 1900 // buffers before playing. This is to ensure the client will 1901 // effectively get the latency it requested. 1902 if (track->isExternalTrack()) { 1903 TrackBase::track_state state = track->mState; 1904 mLock.unlock(); 1905 status = AudioSystem::startOutput(mId, track->streamType(), 1906 (audio_session_t)track->sessionId()); 1907 mLock.lock(); 1908 // abort track was stopped/paused while we released the lock 1909 if (state != track->mState) { 1910 if (status == NO_ERROR) { 1911 mLock.unlock(); 1912 AudioSystem::stopOutput(mId, track->streamType(), 1913 (audio_session_t)track->sessionId()); 1914 mLock.lock(); 1915 } 1916 return INVALID_OPERATION; 1917 } 1918 // abort if start is rejected by audio policy manager 1919 if (status != NO_ERROR) { 1920 return PERMISSION_DENIED; 1921 } 1922#ifdef ADD_BATTERY_DATA 1923 // to track the speaker usage 1924 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1925#endif 1926 } 1927 1928 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1929 track->mResetDone = false; 1930 track->mPresentationCompleteFrames = 0; 1931 mActiveTracks.add(track); 1932 mWakeLockUids.add(track->uid()); 1933 mActiveTracksGeneration++; 1934 mLatestActiveTrack = track; 1935 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1936 if (chain != 0) { 1937 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1938 track->sessionId()); 1939 chain->incActiveTrackCnt(); 1940 } 1941 1942 status = NO_ERROR; 1943 } 1944 1945 onAddNewTrack_l(); 1946 return status; 1947} 1948 1949bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1950{ 1951 track->terminate(); 1952 // active tracks are removed by threadLoop() 1953 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1954 track->mState = TrackBase::STOPPED; 1955 if (!trackActive) { 1956 removeTrack_l(track); 1957 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1958 track->mState = TrackBase::STOPPING_1; 1959 } 1960 1961 return trackActive; 1962} 1963 1964void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1965{ 1966 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1967 mTracks.remove(track); 1968 deleteTrackName_l(track->name()); 1969 // redundant as track is about to be destroyed, for dumpsys only 1970 track->mName = -1; 1971 if (track->isFastTrack()) { 1972 int index = track->mFastIndex; 1973 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1974 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1975 mFastTrackAvailMask |= 1 << index; 1976 // redundant as track is about to be destroyed, for dumpsys only 1977 track->mFastIndex = -1; 1978 } 1979 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1980 if (chain != 0) { 1981 chain->decTrackCnt(); 1982 } 1983} 1984 1985void AudioFlinger::PlaybackThread::broadcast_l() 1986{ 1987 // Thread could be blocked waiting for async 1988 // so signal it to handle state changes immediately 1989 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1990 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1991 mSignalPending = true; 1992 mWaitWorkCV.broadcast(); 1993} 1994 1995String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1996{ 1997 Mutex::Autolock _l(mLock); 1998 if (initCheck() != NO_ERROR) { 1999 return String8(); 2000 } 2001 2002 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2003 const String8 out_s8(s); 2004 free(s); 2005 return out_s8; 2006} 2007 2008void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2009 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2010 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2011 2012 desc->mIoHandle = mId; 2013 2014 switch (event) { 2015 case AUDIO_OUTPUT_OPENED: 2016 case AUDIO_OUTPUT_CONFIG_CHANGED: 2017 desc->mPatch = mPatch; 2018 desc->mChannelMask = mChannelMask; 2019 desc->mSamplingRate = mSampleRate; 2020 desc->mFormat = mFormat; 2021 desc->mFrameCount = mNormalFrameCount; // FIXME see 2022 // AudioFlinger::frameCount(audio_io_handle_t) 2023 desc->mLatency = latency_l(); 2024 break; 2025 2026 case AUDIO_OUTPUT_CLOSED: 2027 default: 2028 break; 2029 } 2030 mAudioFlinger->ioConfigChanged(event, desc, pid); 2031} 2032 2033void AudioFlinger::PlaybackThread::writeCallback() 2034{ 2035 ALOG_ASSERT(mCallbackThread != 0); 2036 mCallbackThread->resetWriteBlocked(); 2037} 2038 2039void AudioFlinger::PlaybackThread::drainCallback() 2040{ 2041 ALOG_ASSERT(mCallbackThread != 0); 2042 mCallbackThread->resetDraining(); 2043} 2044 2045void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2046{ 2047 Mutex::Autolock _l(mLock); 2048 // reject out of sequence requests 2049 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2050 mWriteAckSequence &= ~1; 2051 mWaitWorkCV.signal(); 2052 } 2053} 2054 2055void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2056{ 2057 Mutex::Autolock _l(mLock); 2058 // reject out of sequence requests 2059 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2060 mDrainSequence &= ~1; 2061 mWaitWorkCV.signal(); 2062 } 2063} 2064 2065// static 2066int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2067 void *param __unused, 2068 void *cookie) 2069{ 2070 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2071 ALOGV("asyncCallback() event %d", event); 2072 switch (event) { 2073 case STREAM_CBK_EVENT_WRITE_READY: 2074 me->writeCallback(); 2075 break; 2076 case STREAM_CBK_EVENT_DRAIN_READY: 2077 me->drainCallback(); 2078 break; 2079 default: 2080 ALOGW("asyncCallback() unknown event %d", event); 2081 break; 2082 } 2083 return 0; 2084} 2085 2086void AudioFlinger::PlaybackThread::readOutputParameters_l() 2087{ 2088 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2089 mSampleRate = mOutput->getSampleRate(); 2090 mChannelMask = mOutput->getChannelMask(); 2091 if (!audio_is_output_channel(mChannelMask)) { 2092 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2093 } 2094 if ((mType == MIXER || mType == DUPLICATING) 2095 && !isValidPcmSinkChannelMask(mChannelMask)) { 2096 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2097 mChannelMask); 2098 } 2099 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2100 2101 // Get actual HAL format. 2102 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2103 // Get format from the shim, which will be different than the HAL format 2104 // if playing compressed audio over HDMI passthrough. 2105 mFormat = mOutput->getFormat(); 2106 if (!audio_is_valid_format(mFormat)) { 2107 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2108 } 2109 if ((mType == MIXER || mType == DUPLICATING) 2110 && !isValidPcmSinkFormat(mFormat)) { 2111 LOG_FATAL("HAL format %#x not supported for mixed output", 2112 mFormat); 2113 } 2114 mFrameSize = mOutput->getFrameSize(); 2115 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2116 mFrameCount = mBufferSize / mFrameSize; 2117 if (mFrameCount & 15) { 2118 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2119 mFrameCount); 2120 } 2121 2122 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2123 (mOutput->stream->set_callback != NULL)) { 2124 if (mOutput->stream->set_callback(mOutput->stream, 2125 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2126 mUseAsyncWrite = true; 2127 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2128 } 2129 } 2130 2131 mHwSupportsPause = false; 2132 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2133 if (mOutput->stream->pause != NULL) { 2134 if (mOutput->stream->resume != NULL) { 2135 mHwSupportsPause = true; 2136 } else { 2137 ALOGW("direct output implements pause but not resume"); 2138 } 2139 } else if (mOutput->stream->resume != NULL) { 2140 ALOGW("direct output implements resume but not pause"); 2141 } 2142 } 2143 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2144 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2145 } 2146 2147 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2148 // For best precision, we use float instead of the associated output 2149 // device format (typically PCM 16 bit). 2150 2151 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2152 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2153 mBufferSize = mFrameSize * mFrameCount; 2154 2155 // TODO: We currently use the associated output device channel mask and sample rate. 2156 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2157 // (if a valid mask) to avoid premature downmix. 2158 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2159 // instead of the output device sample rate to avoid loss of high frequency information. 2160 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2161 } 2162 2163 // Calculate size of normal sink buffer relative to the HAL output buffer size 2164 double multiplier = 1.0; 2165 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2166 kUseFastMixer == FastMixer_Dynamic)) { 2167 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2168 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2169 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2170 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2171 maxNormalFrameCount = maxNormalFrameCount & ~15; 2172 if (maxNormalFrameCount < minNormalFrameCount) { 2173 maxNormalFrameCount = minNormalFrameCount; 2174 } 2175 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2176 if (multiplier <= 1.0) { 2177 multiplier = 1.0; 2178 } else if (multiplier <= 2.0) { 2179 if (2 * mFrameCount <= maxNormalFrameCount) { 2180 multiplier = 2.0; 2181 } else { 2182 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2183 } 2184 } else { 2185 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2186 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2187 // track, but we sometimes have to do this to satisfy the maximum frame count 2188 // constraint) 2189 // FIXME this rounding up should not be done if no HAL SRC 2190 uint32_t truncMult = (uint32_t) multiplier; 2191 if ((truncMult & 1)) { 2192 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2193 ++truncMult; 2194 } 2195 } 2196 multiplier = (double) truncMult; 2197 } 2198 } 2199 mNormalFrameCount = multiplier * mFrameCount; 2200 // round up to nearest 16 frames to satisfy AudioMixer 2201 if (mType == MIXER || mType == DUPLICATING) { 2202 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2203 } 2204 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2205 mNormalFrameCount); 2206 2207 // Check if we want to throttle the processing to no more than 2x normal rate 2208 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2209 mThreadThrottleTimeMs = 0; 2210 mThreadThrottleEndMs = 0; 2211 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2212 2213 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2214 // Originally this was int16_t[] array, need to remove legacy implications. 2215 free(mSinkBuffer); 2216 mSinkBuffer = NULL; 2217 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2218 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2219 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2220 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2221 2222 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2223 // drives the output. 2224 free(mMixerBuffer); 2225 mMixerBuffer = NULL; 2226 if (mMixerBufferEnabled) { 2227 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2228 mMixerBufferSize = mNormalFrameCount * mChannelCount 2229 * audio_bytes_per_sample(mMixerBufferFormat); 2230 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2231 } 2232 free(mEffectBuffer); 2233 mEffectBuffer = NULL; 2234 if (mEffectBufferEnabled) { 2235 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2236 mEffectBufferSize = mNormalFrameCount * mChannelCount 2237 * audio_bytes_per_sample(mEffectBufferFormat); 2238 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2239 } 2240 2241 // force reconfiguration of effect chains and engines to take new buffer size and audio 2242 // parameters into account 2243 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2244 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2245 // matter. 2246 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2247 Vector< sp<EffectChain> > effectChains = mEffectChains; 2248 for (size_t i = 0; i < effectChains.size(); i ++) { 2249 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2250 } 2251} 2252 2253 2254status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2255{ 2256 if (halFrames == NULL || dspFrames == NULL) { 2257 return BAD_VALUE; 2258 } 2259 Mutex::Autolock _l(mLock); 2260 if (initCheck() != NO_ERROR) { 2261 return INVALID_OPERATION; 2262 } 2263 size_t framesWritten = mBytesWritten / mFrameSize; 2264 *halFrames = framesWritten; 2265 2266 if (isSuspended()) { 2267 // return an estimation of rendered frames when the output is suspended 2268 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2269 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2270 return NO_ERROR; 2271 } else { 2272 status_t status; 2273 uint32_t frames; 2274 status = mOutput->getRenderPosition(&frames); 2275 *dspFrames = (size_t)frames; 2276 return status; 2277 } 2278} 2279 2280uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2281{ 2282 Mutex::Autolock _l(mLock); 2283 uint32_t result = 0; 2284 if (getEffectChain_l(sessionId) != 0) { 2285 result = EFFECT_SESSION; 2286 } 2287 2288 for (size_t i = 0; i < mTracks.size(); ++i) { 2289 sp<Track> track = mTracks[i]; 2290 if (sessionId == track->sessionId() && !track->isInvalid()) { 2291 result |= TRACK_SESSION; 2292 break; 2293 } 2294 } 2295 2296 return result; 2297} 2298 2299uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2300{ 2301 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2302 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2303 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2304 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2305 } 2306 for (size_t i = 0; i < mTracks.size(); i++) { 2307 sp<Track> track = mTracks[i]; 2308 if (sessionId == track->sessionId() && !track->isInvalid()) { 2309 return AudioSystem::getStrategyForStream(track->streamType()); 2310 } 2311 } 2312 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2313} 2314 2315 2316AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2317{ 2318 Mutex::Autolock _l(mLock); 2319 return mOutput; 2320} 2321 2322AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2323{ 2324 Mutex::Autolock _l(mLock); 2325 AudioStreamOut *output = mOutput; 2326 mOutput = NULL; 2327 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2328 // must push a NULL and wait for ack 2329 mOutputSink.clear(); 2330 mPipeSink.clear(); 2331 mNormalSink.clear(); 2332 return output; 2333} 2334 2335// this method must always be called either with ThreadBase mLock held or inside the thread loop 2336audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2337{ 2338 if (mOutput == NULL) { 2339 return NULL; 2340 } 2341 return &mOutput->stream->common; 2342} 2343 2344uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2345{ 2346 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2347} 2348 2349status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2350{ 2351 if (!isValidSyncEvent(event)) { 2352 return BAD_VALUE; 2353 } 2354 2355 Mutex::Autolock _l(mLock); 2356 2357 for (size_t i = 0; i < mTracks.size(); ++i) { 2358 sp<Track> track = mTracks[i]; 2359 if (event->triggerSession() == track->sessionId()) { 2360 (void) track->setSyncEvent(event); 2361 return NO_ERROR; 2362 } 2363 } 2364 2365 return NAME_NOT_FOUND; 2366} 2367 2368bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2369{ 2370 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2371} 2372 2373void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2374 const Vector< sp<Track> >& tracksToRemove) 2375{ 2376 size_t count = tracksToRemove.size(); 2377 if (count > 0) { 2378 for (size_t i = 0 ; i < count ; i++) { 2379 const sp<Track>& track = tracksToRemove.itemAt(i); 2380 if (track->isExternalTrack()) { 2381 AudioSystem::stopOutput(mId, track->streamType(), 2382 (audio_session_t)track->sessionId()); 2383#ifdef ADD_BATTERY_DATA 2384 // to track the speaker usage 2385 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2386#endif 2387 if (track->isTerminated()) { 2388 AudioSystem::releaseOutput(mId, track->streamType(), 2389 (audio_session_t)track->sessionId()); 2390 } 2391 } 2392 } 2393 } 2394} 2395 2396void AudioFlinger::PlaybackThread::checkSilentMode_l() 2397{ 2398 if (!mMasterMute) { 2399 char value[PROPERTY_VALUE_MAX]; 2400 if (property_get("ro.audio.silent", value, "0") > 0) { 2401 char *endptr; 2402 unsigned long ul = strtoul(value, &endptr, 0); 2403 if (*endptr == '\0' && ul != 0) { 2404 ALOGD("Silence is golden"); 2405 // The setprop command will not allow a property to be changed after 2406 // the first time it is set, so we don't have to worry about un-muting. 2407 setMasterMute_l(true); 2408 } 2409 } 2410 } 2411} 2412 2413// shared by MIXER and DIRECT, overridden by DUPLICATING 2414ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2415{ 2416 // FIXME rewrite to reduce number of system calls 2417 mLastWriteTime = systemTime(); 2418 mInWrite = true; 2419 ssize_t bytesWritten; 2420 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2421 2422 // If an NBAIO sink is present, use it to write the normal mixer's submix 2423 if (mNormalSink != 0) { 2424 2425 const size_t count = mBytesRemaining / mFrameSize; 2426 2427 ATRACE_BEGIN("write"); 2428 // update the setpoint when AudioFlinger::mScreenState changes 2429 uint32_t screenState = AudioFlinger::mScreenState; 2430 if (screenState != mScreenState) { 2431 mScreenState = screenState; 2432 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2433 if (pipe != NULL) { 2434 pipe->setAvgFrames((mScreenState & 1) ? 2435 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2436 } 2437 } 2438 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2439 ATRACE_END(); 2440 if (framesWritten > 0) { 2441 bytesWritten = framesWritten * mFrameSize; 2442 } else { 2443 bytesWritten = framesWritten; 2444 } 2445 mLatchDValid = false; 2446 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2447 if (status == NO_ERROR) { 2448 size_t totalFramesWritten = mNormalSink->framesWritten(); 2449 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2450 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2451 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2452 mLatchDValid = true; 2453 } 2454 } 2455 // otherwise use the HAL / AudioStreamOut directly 2456 } else { 2457 // Direct output and offload threads 2458 2459 if (mUseAsyncWrite) { 2460 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2461 mWriteAckSequence += 2; 2462 mWriteAckSequence |= 1; 2463 ALOG_ASSERT(mCallbackThread != 0); 2464 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2465 } 2466 // FIXME We should have an implementation of timestamps for direct output threads. 2467 // They are used e.g for multichannel PCM playback over HDMI. 2468 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2469 if (mUseAsyncWrite && 2470 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2471 // do not wait for async callback in case of error of full write 2472 mWriteAckSequence &= ~1; 2473 ALOG_ASSERT(mCallbackThread != 0); 2474 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2475 } 2476 } 2477 2478 mNumWrites++; 2479 mInWrite = false; 2480 mStandby = false; 2481 return bytesWritten; 2482} 2483 2484void AudioFlinger::PlaybackThread::threadLoop_drain() 2485{ 2486 if (mOutput->stream->drain) { 2487 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2488 if (mUseAsyncWrite) { 2489 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2490 mDrainSequence |= 1; 2491 ALOG_ASSERT(mCallbackThread != 0); 2492 mCallbackThread->setDraining(mDrainSequence); 2493 } 2494 mOutput->stream->drain(mOutput->stream, 2495 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2496 : AUDIO_DRAIN_ALL); 2497 } 2498} 2499 2500void AudioFlinger::PlaybackThread::threadLoop_exit() 2501{ 2502 { 2503 Mutex::Autolock _l(mLock); 2504 for (size_t i = 0; i < mTracks.size(); i++) { 2505 sp<Track> track = mTracks[i]; 2506 track->invalidate(); 2507 } 2508 } 2509} 2510 2511/* 2512The derived values that are cached: 2513 - mSinkBufferSize from frame count * frame size 2514 - mActiveSleepTimeUs from activeSleepTimeUs() 2515 - mIdleSleepTimeUs from idleSleepTimeUs() 2516 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) 2517 - maxPeriod from frame count and sample rate (MIXER only) 2518 2519The parameters that affect these derived values are: 2520 - frame count 2521 - frame size 2522 - sample rate 2523 - device type: A2DP or not 2524 - device latency 2525 - format: PCM or not 2526 - active sleep time 2527 - idle sleep time 2528*/ 2529 2530void AudioFlinger::PlaybackThread::cacheParameters_l() 2531{ 2532 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2533 mActiveSleepTimeUs = activeSleepTimeUs(); 2534 mIdleSleepTimeUs = idleSleepTimeUs(); 2535} 2536 2537void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2538{ 2539 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2540 this, streamType, mTracks.size()); 2541 Mutex::Autolock _l(mLock); 2542 2543 size_t size = mTracks.size(); 2544 for (size_t i = 0; i < size; i++) { 2545 sp<Track> t = mTracks[i]; 2546 if (t->streamType() == streamType) { 2547 t->invalidate(); 2548 } 2549 } 2550} 2551 2552status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2553{ 2554 int session = chain->sessionId(); 2555 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2556 ? mEffectBuffer : mSinkBuffer); 2557 bool ownsBuffer = false; 2558 2559 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2560 if (session > 0) { 2561 // Only one effect chain can be present in direct output thread and it uses 2562 // the sink buffer as input 2563 if (mType != DIRECT) { 2564 size_t numSamples = mNormalFrameCount * mChannelCount; 2565 buffer = new int16_t[numSamples]; 2566 memset(buffer, 0, numSamples * sizeof(int16_t)); 2567 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2568 ownsBuffer = true; 2569 } 2570 2571 // Attach all tracks with same session ID to this chain. 2572 for (size_t i = 0; i < mTracks.size(); ++i) { 2573 sp<Track> track = mTracks[i]; 2574 if (session == track->sessionId()) { 2575 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2576 buffer); 2577 track->setMainBuffer(buffer); 2578 chain->incTrackCnt(); 2579 } 2580 } 2581 2582 // indicate all active tracks in the chain 2583 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2584 sp<Track> track = mActiveTracks[i].promote(); 2585 if (track == 0) { 2586 continue; 2587 } 2588 if (session == track->sessionId()) { 2589 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2590 chain->incActiveTrackCnt(); 2591 } 2592 } 2593 } 2594 chain->setThread(this); 2595 chain->setInBuffer(buffer, ownsBuffer); 2596 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2597 ? mEffectBuffer : mSinkBuffer)); 2598 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2599 // chains list in order to be processed last as it contains output stage effects 2600 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2601 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2602 // after track specific effects and before output stage 2603 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2604 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2605 // Effect chain for other sessions are inserted at beginning of effect 2606 // chains list to be processed before output mix effects. Relative order between other 2607 // sessions is not important 2608 size_t size = mEffectChains.size(); 2609 size_t i = 0; 2610 for (i = 0; i < size; i++) { 2611 if (mEffectChains[i]->sessionId() < session) { 2612 break; 2613 } 2614 } 2615 mEffectChains.insertAt(chain, i); 2616 checkSuspendOnAddEffectChain_l(chain); 2617 2618 return NO_ERROR; 2619} 2620 2621size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2622{ 2623 int session = chain->sessionId(); 2624 2625 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2626 2627 for (size_t i = 0; i < mEffectChains.size(); i++) { 2628 if (chain == mEffectChains[i]) { 2629 mEffectChains.removeAt(i); 2630 // detach all active tracks from the chain 2631 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2632 sp<Track> track = mActiveTracks[i].promote(); 2633 if (track == 0) { 2634 continue; 2635 } 2636 if (session == track->sessionId()) { 2637 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2638 chain.get(), session); 2639 chain->decActiveTrackCnt(); 2640 } 2641 } 2642 2643 // detach all tracks with same session ID from this chain 2644 for (size_t i = 0; i < mTracks.size(); ++i) { 2645 sp<Track> track = mTracks[i]; 2646 if (session == track->sessionId()) { 2647 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2648 chain->decTrackCnt(); 2649 } 2650 } 2651 break; 2652 } 2653 } 2654 return mEffectChains.size(); 2655} 2656 2657status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2658 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2659{ 2660 Mutex::Autolock _l(mLock); 2661 return attachAuxEffect_l(track, EffectId); 2662} 2663 2664status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2665 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2666{ 2667 status_t status = NO_ERROR; 2668 2669 if (EffectId == 0) { 2670 track->setAuxBuffer(0, NULL); 2671 } else { 2672 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2673 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2674 if (effect != 0) { 2675 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2676 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2677 } else { 2678 status = INVALID_OPERATION; 2679 } 2680 } else { 2681 status = BAD_VALUE; 2682 } 2683 } 2684 return status; 2685} 2686 2687void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2688{ 2689 for (size_t i = 0; i < mTracks.size(); ++i) { 2690 sp<Track> track = mTracks[i]; 2691 if (track->auxEffectId() == effectId) { 2692 attachAuxEffect_l(track, 0); 2693 } 2694 } 2695} 2696 2697bool AudioFlinger::PlaybackThread::threadLoop() 2698{ 2699 Vector< sp<Track> > tracksToRemove; 2700 2701 mStandbyTimeNs = systemTime(); 2702 2703 // MIXER 2704 nsecs_t lastWarning = 0; 2705 2706 // DUPLICATING 2707 // FIXME could this be made local to while loop? 2708 writeFrames = 0; 2709 2710 int lastGeneration = 0; 2711 2712 cacheParameters_l(); 2713 mSleepTimeUs = mIdleSleepTimeUs; 2714 2715 if (mType == MIXER) { 2716 sleepTimeShift = 0; 2717 } 2718 2719 CpuStats cpuStats; 2720 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2721 2722 acquireWakeLock(); 2723 2724 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2725 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2726 // and then that string will be logged at the next convenient opportunity. 2727 const char *logString = NULL; 2728 2729 checkSilentMode_l(); 2730 2731 while (!exitPending()) 2732 { 2733 cpuStats.sample(myName); 2734 2735 Vector< sp<EffectChain> > effectChains; 2736 2737 { // scope for mLock 2738 2739 Mutex::Autolock _l(mLock); 2740 2741 processConfigEvents_l(); 2742 2743 if (logString != NULL) { 2744 mNBLogWriter->logTimestamp(); 2745 mNBLogWriter->log(logString); 2746 logString = NULL; 2747 } 2748 2749 // Gather the framesReleased counters for all active tracks, 2750 // and latch them atomically with the timestamp. 2751 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2752 mLatchD.mFramesReleased.clear(); 2753 size_t size = mActiveTracks.size(); 2754 for (size_t i = 0; i < size; i++) { 2755 sp<Track> t = mActiveTracks[i].promote(); 2756 if (t != 0) { 2757 mLatchD.mFramesReleased.add(t.get(), 2758 t->mAudioTrackServerProxy->framesReleased()); 2759 } 2760 } 2761 if (mLatchDValid) { 2762 mLatchQ = mLatchD; 2763 mLatchDValid = false; 2764 mLatchQValid = true; 2765 } 2766 2767 saveOutputTracks(); 2768 if (mSignalPending) { 2769 // A signal was raised while we were unlocked 2770 mSignalPending = false; 2771 } else if (waitingAsyncCallback_l()) { 2772 if (exitPending()) { 2773 break; 2774 } 2775 bool released = false; 2776 // The following works around a bug in the offload driver. Ideally we would release 2777 // the wake lock every time, but that causes the last offload buffer(s) to be 2778 // dropped while the device is on battery, so we need to hold a wake lock during 2779 // the drain phase. 2780 if (mBytesRemaining && !(mDrainSequence & 1)) { 2781 releaseWakeLock_l(); 2782 released = true; 2783 } 2784 mWakeLockUids.clear(); 2785 mActiveTracksGeneration++; 2786 ALOGV("wait async completion"); 2787 mWaitWorkCV.wait(mLock); 2788 ALOGV("async completion/wake"); 2789 if (released) { 2790 acquireWakeLock_l(); 2791 } 2792 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2793 mSleepTimeUs = 0; 2794 2795 continue; 2796 } 2797 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2798 isSuspended()) { 2799 // put audio hardware into standby after short delay 2800 if (shouldStandby_l()) { 2801 2802 threadLoop_standby(); 2803 2804 mStandby = true; 2805 } 2806 2807 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2808 // we're about to wait, flush the binder command buffer 2809 IPCThreadState::self()->flushCommands(); 2810 2811 clearOutputTracks(); 2812 2813 if (exitPending()) { 2814 break; 2815 } 2816 2817 releaseWakeLock_l(); 2818 mWakeLockUids.clear(); 2819 mActiveTracksGeneration++; 2820 // wait until we have something to do... 2821 ALOGV("%s going to sleep", myName.string()); 2822 mWaitWorkCV.wait(mLock); 2823 ALOGV("%s waking up", myName.string()); 2824 acquireWakeLock_l(); 2825 2826 mMixerStatus = MIXER_IDLE; 2827 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2828 mBytesWritten = 0; 2829 mBytesRemaining = 0; 2830 checkSilentMode_l(); 2831 2832 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2833 mSleepTimeUs = mIdleSleepTimeUs; 2834 if (mType == MIXER) { 2835 sleepTimeShift = 0; 2836 } 2837 2838 continue; 2839 } 2840 } 2841 // mMixerStatusIgnoringFastTracks is also updated internally 2842 mMixerStatus = prepareTracks_l(&tracksToRemove); 2843 2844 // compare with previously applied list 2845 if (lastGeneration != mActiveTracksGeneration) { 2846 // update wakelock 2847 updateWakeLockUids_l(mWakeLockUids); 2848 lastGeneration = mActiveTracksGeneration; 2849 } 2850 2851 // prevent any changes in effect chain list and in each effect chain 2852 // during mixing and effect process as the audio buffers could be deleted 2853 // or modified if an effect is created or deleted 2854 lockEffectChains_l(effectChains); 2855 } // mLock scope ends 2856 2857 if (mBytesRemaining == 0) { 2858 mCurrentWriteLength = 0; 2859 if (mMixerStatus == MIXER_TRACKS_READY) { 2860 // threadLoop_mix() sets mCurrentWriteLength 2861 threadLoop_mix(); 2862 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2863 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2864 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2865 // must be written to HAL 2866 threadLoop_sleepTime(); 2867 if (mSleepTimeUs == 0) { 2868 mCurrentWriteLength = mSinkBufferSize; 2869 } 2870 } 2871 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2872 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2873 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2874 // or mSinkBuffer (if there are no effects). 2875 // 2876 // This is done pre-effects computation; if effects change to 2877 // support higher precision, this needs to move. 2878 // 2879 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2880 // TODO use mSleepTimeUs == 0 as an additional condition. 2881 if (mMixerBufferValid) { 2882 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2883 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2884 2885 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2886 mNormalFrameCount * mChannelCount); 2887 } 2888 2889 mBytesRemaining = mCurrentWriteLength; 2890 if (isSuspended()) { 2891 mSleepTimeUs = suspendSleepTimeUs(); 2892 // simulate write to HAL when suspended 2893 mBytesWritten += mSinkBufferSize; 2894 mBytesRemaining = 0; 2895 } 2896 2897 // only process effects if we're going to write 2898 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 2899 for (size_t i = 0; i < effectChains.size(); i ++) { 2900 effectChains[i]->process_l(); 2901 } 2902 } 2903 } 2904 // Process effect chains for offloaded thread even if no audio 2905 // was read from audio track: process only updates effect state 2906 // and thus does have to be synchronized with audio writes but may have 2907 // to be called while waiting for async write callback 2908 if (mType == OFFLOAD) { 2909 for (size_t i = 0; i < effectChains.size(); i ++) { 2910 effectChains[i]->process_l(); 2911 } 2912 } 2913 2914 // Only if the Effects buffer is enabled and there is data in the 2915 // Effects buffer (buffer valid), we need to 2916 // copy into the sink buffer. 2917 // TODO use mSleepTimeUs == 0 as an additional condition. 2918 if (mEffectBufferValid) { 2919 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2920 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2921 mNormalFrameCount * mChannelCount); 2922 } 2923 2924 // enable changes in effect chain 2925 unlockEffectChains(effectChains); 2926 2927 if (!waitingAsyncCallback()) { 2928 // mSleepTimeUs == 0 means we must write to audio hardware 2929 if (mSleepTimeUs == 0) { 2930 ssize_t ret = 0; 2931 if (mBytesRemaining) { 2932 ret = threadLoop_write(); 2933 if (ret < 0) { 2934 mBytesRemaining = 0; 2935 } else { 2936 mBytesWritten += ret; 2937 mBytesRemaining -= ret; 2938 } 2939 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2940 (mMixerStatus == MIXER_DRAIN_ALL)) { 2941 threadLoop_drain(); 2942 } 2943 if (mType == MIXER && !mStandby) { 2944 // write blocked detection 2945 nsecs_t now = systemTime(); 2946 nsecs_t delta = now - mLastWriteTime; 2947 if (delta > maxPeriod) { 2948 mNumDelayedWrites++; 2949 if ((now - lastWarning) > kWarningThrottleNs) { 2950 ATRACE_NAME("underrun"); 2951 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2952 ns2ms(delta), mNumDelayedWrites, this); 2953 lastWarning = now; 2954 } 2955 } 2956 2957 if (mThreadThrottle 2958 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 2959 && ret > 0) { // we wrote something 2960 // Limit MixerThread data processing to no more than twice the 2961 // expected processing rate. 2962 // 2963 // This helps prevent underruns with NuPlayer and other applications 2964 // which may set up buffers that are close to the minimum size, or use 2965 // deep buffers, and rely on a double-buffering sleep strategy to fill. 2966 // 2967 // The throttle smooths out sudden large data drains from the device, 2968 // e.g. when it comes out of standby, which often causes problems with 2969 // (1) mixer threads without a fast mixer (which has its own warm-up) 2970 // (2) minimum buffer sized tracks (even if the track is full, 2971 // the app won't fill fast enough to handle the sudden draw). 2972 2973 const int32_t deltaMs = delta / 1000000; 2974 const int32_t throttleMs = mHalfBufferMs - deltaMs; 2975 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 2976 usleep(throttleMs * 1000); 2977 // notify of throttle start on verbose log 2978 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 2979 "mixer(%p) throttle begin:" 2980 " ret(%zd) deltaMs(%d) requires sleep %d ms", 2981 this, ret, deltaMs, throttleMs); 2982 mThreadThrottleTimeMs += throttleMs; 2983 } else { 2984 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 2985 if (diff > 0) { 2986 // notify of throttle end on debug log 2987 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 2988 mThreadThrottleEndMs = mThreadThrottleTimeMs; 2989 } 2990 } 2991 } 2992 } 2993 2994 } else { 2995 ATRACE_BEGIN("sleep"); 2996 usleep(mSleepTimeUs); 2997 ATRACE_END(); 2998 } 2999 } 3000 3001 // Finally let go of removed track(s), without the lock held 3002 // since we can't guarantee the destructors won't acquire that 3003 // same lock. This will also mutate and push a new fast mixer state. 3004 threadLoop_removeTracks(tracksToRemove); 3005 tracksToRemove.clear(); 3006 3007 // FIXME I don't understand the need for this here; 3008 // it was in the original code but maybe the 3009 // assignment in saveOutputTracks() makes this unnecessary? 3010 clearOutputTracks(); 3011 3012 // Effect chains will be actually deleted here if they were removed from 3013 // mEffectChains list during mixing or effects processing 3014 effectChains.clear(); 3015 3016 // FIXME Note that the above .clear() is no longer necessary since effectChains 3017 // is now local to this block, but will keep it for now (at least until merge done). 3018 } 3019 3020 threadLoop_exit(); 3021 3022 if (!mStandby) { 3023 threadLoop_standby(); 3024 mStandby = true; 3025 } 3026 3027 releaseWakeLock(); 3028 mWakeLockUids.clear(); 3029 mActiveTracksGeneration++; 3030 3031 ALOGV("Thread %p type %d exiting", this, mType); 3032 return false; 3033} 3034 3035// removeTracks_l() must be called with ThreadBase::mLock held 3036void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3037{ 3038 size_t count = tracksToRemove.size(); 3039 if (count > 0) { 3040 for (size_t i=0 ; i<count ; i++) { 3041 const sp<Track>& track = tracksToRemove.itemAt(i); 3042 mActiveTracks.remove(track); 3043 mWakeLockUids.remove(track->uid()); 3044 mActiveTracksGeneration++; 3045 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3046 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3047 if (chain != 0) { 3048 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3049 track->sessionId()); 3050 chain->decActiveTrackCnt(); 3051 } 3052 if (track->isTerminated()) { 3053 removeTrack_l(track); 3054 } 3055 } 3056 } 3057 3058} 3059 3060status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3061{ 3062 if (mNormalSink != 0) { 3063 return mNormalSink->getTimestamp(timestamp); 3064 } 3065 if ((mType == OFFLOAD || mType == DIRECT) 3066 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3067 uint64_t position64; 3068 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3069 if (ret == 0) { 3070 timestamp.mPosition = (uint32_t)position64; 3071 return NO_ERROR; 3072 } 3073 } 3074 return INVALID_OPERATION; 3075} 3076 3077status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3078 audio_patch_handle_t *handle) 3079{ 3080 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3081 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3082 if (mFastMixer != 0) { 3083 FastMixerStateQueue *sq = mFastMixer->sq(); 3084 FastMixerState *state = sq->begin(); 3085 if (!(state->mCommand & FastMixerState::IDLE)) { 3086 previousCommand = state->mCommand; 3087 state->mCommand = FastMixerState::HOT_IDLE; 3088 sq->end(); 3089 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3090 } else { 3091 sq->end(false /*didModify*/); 3092 } 3093 } 3094 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3095 3096 if (!(previousCommand & FastMixerState::IDLE)) { 3097 ALOG_ASSERT(mFastMixer != 0); 3098 FastMixerStateQueue *sq = mFastMixer->sq(); 3099 FastMixerState *state = sq->begin(); 3100 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3101 state->mCommand = previousCommand; 3102 sq->end(); 3103 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3104 } 3105 3106 return status; 3107} 3108 3109status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3110 audio_patch_handle_t *handle) 3111{ 3112 status_t status = NO_ERROR; 3113 3114 // store new device and send to effects 3115 audio_devices_t type = AUDIO_DEVICE_NONE; 3116 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3117 type |= patch->sinks[i].ext.device.type; 3118 } 3119 3120#ifdef ADD_BATTERY_DATA 3121 // when changing the audio output device, call addBatteryData to notify 3122 // the change 3123 if (mOutDevice != type) { 3124 uint32_t params = 0; 3125 // check whether speaker is on 3126 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3127 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3128 } 3129 3130 audio_devices_t deviceWithoutSpeaker 3131 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3132 // check if any other device (except speaker) is on 3133 if (type & deviceWithoutSpeaker) { 3134 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3135 } 3136 3137 if (params != 0) { 3138 addBatteryData(params); 3139 } 3140 } 3141#endif 3142 3143 for (size_t i = 0; i < mEffectChains.size(); i++) { 3144 mEffectChains[i]->setDevice_l(type); 3145 } 3146 3147 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3148 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3149 bool configChanged = mPrevOutDevice != type; 3150 mOutDevice = type; 3151 mPatch = *patch; 3152 3153 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3154 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3155 status = hwDevice->create_audio_patch(hwDevice, 3156 patch->num_sources, 3157 patch->sources, 3158 patch->num_sinks, 3159 patch->sinks, 3160 handle); 3161 } else { 3162 char *address; 3163 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3164 //FIXME: we only support address on first sink with HAL version < 3.0 3165 address = audio_device_address_to_parameter( 3166 patch->sinks[0].ext.device.type, 3167 patch->sinks[0].ext.device.address); 3168 } else { 3169 address = (char *)calloc(1, 1); 3170 } 3171 AudioParameter param = AudioParameter(String8(address)); 3172 free(address); 3173 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3174 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3175 param.toString().string()); 3176 *handle = AUDIO_PATCH_HANDLE_NONE; 3177 } 3178 if (configChanged) { 3179 mPrevOutDevice = type; 3180 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3181 } 3182 return status; 3183} 3184 3185status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3186{ 3187 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3188 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3189 if (mFastMixer != 0) { 3190 FastMixerStateQueue *sq = mFastMixer->sq(); 3191 FastMixerState *state = sq->begin(); 3192 if (!(state->mCommand & FastMixerState::IDLE)) { 3193 previousCommand = state->mCommand; 3194 state->mCommand = FastMixerState::HOT_IDLE; 3195 sq->end(); 3196 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3197 } else { 3198 sq->end(false /*didModify*/); 3199 } 3200 } 3201 3202 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3203 3204 if (!(previousCommand & FastMixerState::IDLE)) { 3205 ALOG_ASSERT(mFastMixer != 0); 3206 FastMixerStateQueue *sq = mFastMixer->sq(); 3207 FastMixerState *state = sq->begin(); 3208 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3209 state->mCommand = previousCommand; 3210 sq->end(); 3211 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3212 } 3213 3214 return status; 3215} 3216 3217status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3218{ 3219 status_t status = NO_ERROR; 3220 3221 mOutDevice = AUDIO_DEVICE_NONE; 3222 3223 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3224 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3225 status = hwDevice->release_audio_patch(hwDevice, handle); 3226 } else { 3227 AudioParameter param; 3228 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3229 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3230 param.toString().string()); 3231 } 3232 return status; 3233} 3234 3235void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3236{ 3237 Mutex::Autolock _l(mLock); 3238 mTracks.add(track); 3239} 3240 3241void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3242{ 3243 Mutex::Autolock _l(mLock); 3244 destroyTrack_l(track); 3245} 3246 3247void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3248{ 3249 ThreadBase::getAudioPortConfig(config); 3250 config->role = AUDIO_PORT_ROLE_SOURCE; 3251 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3252 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3253} 3254 3255// ---------------------------------------------------------------------------- 3256 3257AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3258 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3259 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3260 // mAudioMixer below 3261 // mFastMixer below 3262 mFastMixerFutex(0) 3263 // mOutputSink below 3264 // mPipeSink below 3265 // mNormalSink below 3266{ 3267 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3268 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3269 "mFrameCount=%d, mNormalFrameCount=%d", 3270 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3271 mNormalFrameCount); 3272 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3273 3274 if (type == DUPLICATING) { 3275 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3276 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3277 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3278 return; 3279 } 3280 // create an NBAIO sink for the HAL output stream, and negotiate 3281 mOutputSink = new AudioStreamOutSink(output->stream); 3282 size_t numCounterOffers = 0; 3283 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3284 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3285 ALOG_ASSERT(index == 0); 3286 3287 // initialize fast mixer depending on configuration 3288 bool initFastMixer; 3289 switch (kUseFastMixer) { 3290 case FastMixer_Never: 3291 initFastMixer = false; 3292 break; 3293 case FastMixer_Always: 3294 initFastMixer = true; 3295 break; 3296 case FastMixer_Static: 3297 case FastMixer_Dynamic: 3298 initFastMixer = mFrameCount < mNormalFrameCount; 3299 break; 3300 } 3301 if (initFastMixer) { 3302 audio_format_t fastMixerFormat; 3303 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3304 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3305 } else { 3306 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3307 } 3308 if (mFormat != fastMixerFormat) { 3309 // change our Sink format to accept our intermediate precision 3310 mFormat = fastMixerFormat; 3311 free(mSinkBuffer); 3312 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3313 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3314 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3315 } 3316 3317 // create a MonoPipe to connect our submix to FastMixer 3318 NBAIO_Format format = mOutputSink->format(); 3319 NBAIO_Format origformat = format; 3320 // adjust format to match that of the Fast Mixer 3321 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3322 format.mFormat = fastMixerFormat; 3323 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3324 3325 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3326 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3327 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3328 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3329 const NBAIO_Format offers[1] = {format}; 3330 size_t numCounterOffers = 0; 3331 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3332 ALOG_ASSERT(index == 0); 3333 monoPipe->setAvgFrames((mScreenState & 1) ? 3334 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3335 mPipeSink = monoPipe; 3336 3337#ifdef TEE_SINK 3338 if (mTeeSinkOutputEnabled) { 3339 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3340 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3341 const NBAIO_Format offers2[1] = {origformat}; 3342 numCounterOffers = 0; 3343 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3344 ALOG_ASSERT(index == 0); 3345 mTeeSink = teeSink; 3346 PipeReader *teeSource = new PipeReader(*teeSink); 3347 numCounterOffers = 0; 3348 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3349 ALOG_ASSERT(index == 0); 3350 mTeeSource = teeSource; 3351 } 3352#endif 3353 3354 // create fast mixer and configure it initially with just one fast track for our submix 3355 mFastMixer = new FastMixer(); 3356 FastMixerStateQueue *sq = mFastMixer->sq(); 3357#ifdef STATE_QUEUE_DUMP 3358 sq->setObserverDump(&mStateQueueObserverDump); 3359 sq->setMutatorDump(&mStateQueueMutatorDump); 3360#endif 3361 FastMixerState *state = sq->begin(); 3362 FastTrack *fastTrack = &state->mFastTracks[0]; 3363 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3364 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3365 fastTrack->mVolumeProvider = NULL; 3366 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3367 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3368 fastTrack->mGeneration++; 3369 state->mFastTracksGen++; 3370 state->mTrackMask = 1; 3371 // fast mixer will use the HAL output sink 3372 state->mOutputSink = mOutputSink.get(); 3373 state->mOutputSinkGen++; 3374 state->mFrameCount = mFrameCount; 3375 state->mCommand = FastMixerState::COLD_IDLE; 3376 // already done in constructor initialization list 3377 //mFastMixerFutex = 0; 3378 state->mColdFutexAddr = &mFastMixerFutex; 3379 state->mColdGen++; 3380 state->mDumpState = &mFastMixerDumpState; 3381#ifdef TEE_SINK 3382 state->mTeeSink = mTeeSink.get(); 3383#endif 3384 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3385 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3386 sq->end(); 3387 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3388 3389 // start the fast mixer 3390 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3391 pid_t tid = mFastMixer->getTid(); 3392 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3393 3394#ifdef AUDIO_WATCHDOG 3395 // create and start the watchdog 3396 mAudioWatchdog = new AudioWatchdog(); 3397 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3398 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3399 tid = mAudioWatchdog->getTid(); 3400 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3401#endif 3402 3403 } 3404 3405 switch (kUseFastMixer) { 3406 case FastMixer_Never: 3407 case FastMixer_Dynamic: 3408 mNormalSink = mOutputSink; 3409 break; 3410 case FastMixer_Always: 3411 mNormalSink = mPipeSink; 3412 break; 3413 case FastMixer_Static: 3414 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3415 break; 3416 } 3417} 3418 3419AudioFlinger::MixerThread::~MixerThread() 3420{ 3421 if (mFastMixer != 0) { 3422 FastMixerStateQueue *sq = mFastMixer->sq(); 3423 FastMixerState *state = sq->begin(); 3424 if (state->mCommand == FastMixerState::COLD_IDLE) { 3425 int32_t old = android_atomic_inc(&mFastMixerFutex); 3426 if (old == -1) { 3427 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3428 } 3429 } 3430 state->mCommand = FastMixerState::EXIT; 3431 sq->end(); 3432 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3433 mFastMixer->join(); 3434 // Though the fast mixer thread has exited, it's state queue is still valid. 3435 // We'll use that extract the final state which contains one remaining fast track 3436 // corresponding to our sub-mix. 3437 state = sq->begin(); 3438 ALOG_ASSERT(state->mTrackMask == 1); 3439 FastTrack *fastTrack = &state->mFastTracks[0]; 3440 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3441 delete fastTrack->mBufferProvider; 3442 sq->end(false /*didModify*/); 3443 mFastMixer.clear(); 3444#ifdef AUDIO_WATCHDOG 3445 if (mAudioWatchdog != 0) { 3446 mAudioWatchdog->requestExit(); 3447 mAudioWatchdog->requestExitAndWait(); 3448 mAudioWatchdog.clear(); 3449 } 3450#endif 3451 } 3452 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3453 delete mAudioMixer; 3454} 3455 3456 3457uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3458{ 3459 if (mFastMixer != 0) { 3460 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3461 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3462 } 3463 return latency; 3464} 3465 3466 3467void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3468{ 3469 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3470} 3471 3472ssize_t AudioFlinger::MixerThread::threadLoop_write() 3473{ 3474 // FIXME we should only do one push per cycle; confirm this is true 3475 // Start the fast mixer if it's not already running 3476 if (mFastMixer != 0) { 3477 FastMixerStateQueue *sq = mFastMixer->sq(); 3478 FastMixerState *state = sq->begin(); 3479 if (state->mCommand != FastMixerState::MIX_WRITE && 3480 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3481 if (state->mCommand == FastMixerState::COLD_IDLE) { 3482 3483 // FIXME workaround for first HAL write being CPU bound on some devices 3484 ATRACE_BEGIN("write"); 3485 mOutput->write((char *)mSinkBuffer, 0); 3486 ATRACE_END(); 3487 3488 int32_t old = android_atomic_inc(&mFastMixerFutex); 3489 if (old == -1) { 3490 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3491 } 3492#ifdef AUDIO_WATCHDOG 3493 if (mAudioWatchdog != 0) { 3494 mAudioWatchdog->resume(); 3495 } 3496#endif 3497 } 3498 state->mCommand = FastMixerState::MIX_WRITE; 3499#ifdef FAST_THREAD_STATISTICS 3500 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3501 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3502#endif 3503 sq->end(); 3504 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3505 if (kUseFastMixer == FastMixer_Dynamic) { 3506 mNormalSink = mPipeSink; 3507 } 3508 } else { 3509 sq->end(false /*didModify*/); 3510 } 3511 } 3512 return PlaybackThread::threadLoop_write(); 3513} 3514 3515void AudioFlinger::MixerThread::threadLoop_standby() 3516{ 3517 // Idle the fast mixer if it's currently running 3518 if (mFastMixer != 0) { 3519 FastMixerStateQueue *sq = mFastMixer->sq(); 3520 FastMixerState *state = sq->begin(); 3521 if (!(state->mCommand & FastMixerState::IDLE)) { 3522 state->mCommand = FastMixerState::COLD_IDLE; 3523 state->mColdFutexAddr = &mFastMixerFutex; 3524 state->mColdGen++; 3525 mFastMixerFutex = 0; 3526 sq->end(); 3527 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3528 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3529 if (kUseFastMixer == FastMixer_Dynamic) { 3530 mNormalSink = mOutputSink; 3531 } 3532#ifdef AUDIO_WATCHDOG 3533 if (mAudioWatchdog != 0) { 3534 mAudioWatchdog->pause(); 3535 } 3536#endif 3537 } else { 3538 sq->end(false /*didModify*/); 3539 } 3540 } 3541 PlaybackThread::threadLoop_standby(); 3542} 3543 3544bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3545{ 3546 return false; 3547} 3548 3549bool AudioFlinger::PlaybackThread::shouldStandby_l() 3550{ 3551 return !mStandby; 3552} 3553 3554bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3555{ 3556 Mutex::Autolock _l(mLock); 3557 return waitingAsyncCallback_l(); 3558} 3559 3560// shared by MIXER and DIRECT, overridden by DUPLICATING 3561void AudioFlinger::PlaybackThread::threadLoop_standby() 3562{ 3563 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3564 mOutput->standby(); 3565 if (mUseAsyncWrite != 0) { 3566 // discard any pending drain or write ack by incrementing sequence 3567 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3568 mDrainSequence = (mDrainSequence + 2) & ~1; 3569 ALOG_ASSERT(mCallbackThread != 0); 3570 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3571 mCallbackThread->setDraining(mDrainSequence); 3572 } 3573 mHwPaused = false; 3574} 3575 3576void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3577{ 3578 ALOGV("signal playback thread"); 3579 broadcast_l(); 3580} 3581 3582void AudioFlinger::MixerThread::threadLoop_mix() 3583{ 3584 // obtain the presentation timestamp of the next output buffer 3585 int64_t pts; 3586 status_t status = INVALID_OPERATION; 3587 3588 if (mNormalSink != 0) { 3589 status = mNormalSink->getNextWriteTimestamp(&pts); 3590 } else { 3591 status = mOutputSink->getNextWriteTimestamp(&pts); 3592 } 3593 3594 if (status != NO_ERROR) { 3595 pts = AudioBufferProvider::kInvalidPTS; 3596 } 3597 3598 // mix buffers... 3599 mAudioMixer->process(pts); 3600 mCurrentWriteLength = mSinkBufferSize; 3601 // increase sleep time progressively when application underrun condition clears. 3602 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3603 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3604 // such that we would underrun the audio HAL. 3605 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3606 sleepTimeShift--; 3607 } 3608 mSleepTimeUs = 0; 3609 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3610 //TODO: delay standby when effects have a tail 3611 3612} 3613 3614void AudioFlinger::MixerThread::threadLoop_sleepTime() 3615{ 3616 // If no tracks are ready, sleep once for the duration of an output 3617 // buffer size, then write 0s to the output 3618 if (mSleepTimeUs == 0) { 3619 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3620 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3621 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3622 mSleepTimeUs = kMinThreadSleepTimeUs; 3623 } 3624 // reduce sleep time in case of consecutive application underruns to avoid 3625 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3626 // duration we would end up writing less data than needed by the audio HAL if 3627 // the condition persists. 3628 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3629 sleepTimeShift++; 3630 } 3631 } else { 3632 mSleepTimeUs = mIdleSleepTimeUs; 3633 } 3634 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3635 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3636 // before effects processing or output. 3637 if (mMixerBufferValid) { 3638 memset(mMixerBuffer, 0, mMixerBufferSize); 3639 } else { 3640 memset(mSinkBuffer, 0, mSinkBufferSize); 3641 } 3642 mSleepTimeUs = 0; 3643 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3644 "anticipated start"); 3645 } 3646 // TODO add standby time extension fct of effect tail 3647} 3648 3649// prepareTracks_l() must be called with ThreadBase::mLock held 3650AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3651 Vector< sp<Track> > *tracksToRemove) 3652{ 3653 3654 mixer_state mixerStatus = MIXER_IDLE; 3655 // find out which tracks need to be processed 3656 size_t count = mActiveTracks.size(); 3657 size_t mixedTracks = 0; 3658 size_t tracksWithEffect = 0; 3659 // counts only _active_ fast tracks 3660 size_t fastTracks = 0; 3661 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3662 3663 float masterVolume = mMasterVolume; 3664 bool masterMute = mMasterMute; 3665 3666 if (masterMute) { 3667 masterVolume = 0; 3668 } 3669 // Delegate master volume control to effect in output mix effect chain if needed 3670 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3671 if (chain != 0) { 3672 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3673 chain->setVolume_l(&v, &v); 3674 masterVolume = (float)((v + (1 << 23)) >> 24); 3675 chain.clear(); 3676 } 3677 3678 // prepare a new state to push 3679 FastMixerStateQueue *sq = NULL; 3680 FastMixerState *state = NULL; 3681 bool didModify = false; 3682 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3683 if (mFastMixer != 0) { 3684 sq = mFastMixer->sq(); 3685 state = sq->begin(); 3686 } 3687 3688 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3689 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3690 3691 for (size_t i=0 ; i<count ; i++) { 3692 const sp<Track> t = mActiveTracks[i].promote(); 3693 if (t == 0) { 3694 continue; 3695 } 3696 3697 // this const just means the local variable doesn't change 3698 Track* const track = t.get(); 3699 3700 // process fast tracks 3701 if (track->isFastTrack()) { 3702 3703 // It's theoretically possible (though unlikely) for a fast track to be created 3704 // and then removed within the same normal mix cycle. This is not a problem, as 3705 // the track never becomes active so it's fast mixer slot is never touched. 3706 // The converse, of removing an (active) track and then creating a new track 3707 // at the identical fast mixer slot within the same normal mix cycle, 3708 // is impossible because the slot isn't marked available until the end of each cycle. 3709 int j = track->mFastIndex; 3710 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3711 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3712 FastTrack *fastTrack = &state->mFastTracks[j]; 3713 3714 // Determine whether the track is currently in underrun condition, 3715 // and whether it had a recent underrun. 3716 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3717 FastTrackUnderruns underruns = ftDump->mUnderruns; 3718 uint32_t recentFull = (underruns.mBitFields.mFull - 3719 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3720 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3721 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3722 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3723 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3724 uint32_t recentUnderruns = recentPartial + recentEmpty; 3725 track->mObservedUnderruns = underruns; 3726 // don't count underruns that occur while stopping or pausing 3727 // or stopped which can occur when flush() is called while active 3728 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3729 recentUnderruns > 0) { 3730 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3731 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3732 } 3733 3734 // This is similar to the state machine for normal tracks, 3735 // with a few modifications for fast tracks. 3736 bool isActive = true; 3737 switch (track->mState) { 3738 case TrackBase::STOPPING_1: 3739 // track stays active in STOPPING_1 state until first underrun 3740 if (recentUnderruns > 0 || track->isTerminated()) { 3741 track->mState = TrackBase::STOPPING_2; 3742 } 3743 break; 3744 case TrackBase::PAUSING: 3745 // ramp down is not yet implemented 3746 track->setPaused(); 3747 break; 3748 case TrackBase::RESUMING: 3749 // ramp up is not yet implemented 3750 track->mState = TrackBase::ACTIVE; 3751 break; 3752 case TrackBase::ACTIVE: 3753 if (recentFull > 0 || recentPartial > 0) { 3754 // track has provided at least some frames recently: reset retry count 3755 track->mRetryCount = kMaxTrackRetries; 3756 } 3757 if (recentUnderruns == 0) { 3758 // no recent underruns: stay active 3759 break; 3760 } 3761 // there has recently been an underrun of some kind 3762 if (track->sharedBuffer() == 0) { 3763 // were any of the recent underruns "empty" (no frames available)? 3764 if (recentEmpty == 0) { 3765 // no, then ignore the partial underruns as they are allowed indefinitely 3766 break; 3767 } 3768 // there has recently been an "empty" underrun: decrement the retry counter 3769 if (--(track->mRetryCount) > 0) { 3770 break; 3771 } 3772 // indicate to client process that the track was disabled because of underrun; 3773 // it will then automatically call start() when data is available 3774 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3775 // remove from active list, but state remains ACTIVE [confusing but true] 3776 isActive = false; 3777 break; 3778 } 3779 // fall through 3780 case TrackBase::STOPPING_2: 3781 case TrackBase::PAUSED: 3782 case TrackBase::STOPPED: 3783 case TrackBase::FLUSHED: // flush() while active 3784 // Check for presentation complete if track is inactive 3785 // We have consumed all the buffers of this track. 3786 // This would be incomplete if we auto-paused on underrun 3787 { 3788 size_t audioHALFrames = 3789 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3790 size_t framesWritten = mBytesWritten / mFrameSize; 3791 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3792 // track stays in active list until presentation is complete 3793 break; 3794 } 3795 } 3796 if (track->isStopping_2()) { 3797 track->mState = TrackBase::STOPPED; 3798 } 3799 if (track->isStopped()) { 3800 // Can't reset directly, as fast mixer is still polling this track 3801 // track->reset(); 3802 // So instead mark this track as needing to be reset after push with ack 3803 resetMask |= 1 << i; 3804 } 3805 isActive = false; 3806 break; 3807 case TrackBase::IDLE: 3808 default: 3809 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3810 } 3811 3812 if (isActive) { 3813 // was it previously inactive? 3814 if (!(state->mTrackMask & (1 << j))) { 3815 ExtendedAudioBufferProvider *eabp = track; 3816 VolumeProvider *vp = track; 3817 fastTrack->mBufferProvider = eabp; 3818 fastTrack->mVolumeProvider = vp; 3819 fastTrack->mChannelMask = track->mChannelMask; 3820 fastTrack->mFormat = track->mFormat; 3821 fastTrack->mGeneration++; 3822 state->mTrackMask |= 1 << j; 3823 didModify = true; 3824 // no acknowledgement required for newly active tracks 3825 } 3826 // cache the combined master volume and stream type volume for fast mixer; this 3827 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3828 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3829 ++fastTracks; 3830 } else { 3831 // was it previously active? 3832 if (state->mTrackMask & (1 << j)) { 3833 fastTrack->mBufferProvider = NULL; 3834 fastTrack->mGeneration++; 3835 state->mTrackMask &= ~(1 << j); 3836 didModify = true; 3837 // If any fast tracks were removed, we must wait for acknowledgement 3838 // because we're about to decrement the last sp<> on those tracks. 3839 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3840 } else { 3841 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3842 } 3843 tracksToRemove->add(track); 3844 // Avoids a misleading display in dumpsys 3845 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3846 } 3847 continue; 3848 } 3849 3850 { // local variable scope to avoid goto warning 3851 3852 audio_track_cblk_t* cblk = track->cblk(); 3853 3854 // The first time a track is added we wait 3855 // for all its buffers to be filled before processing it 3856 int name = track->name(); 3857 // make sure that we have enough frames to mix one full buffer. 3858 // enforce this condition only once to enable draining the buffer in case the client 3859 // app does not call stop() and relies on underrun to stop: 3860 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3861 // during last round 3862 size_t desiredFrames; 3863 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3864 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3865 3866 desiredFrames = sourceFramesNeededWithTimestretch( 3867 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3868 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3869 // add frames already consumed but not yet released by the resampler 3870 // because mAudioTrackServerProxy->framesReady() will include these frames 3871 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3872 3873 uint32_t minFrames = 1; 3874 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3875 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3876 minFrames = desiredFrames; 3877 } 3878 3879 size_t framesReady = track->framesReady(); 3880 if (ATRACE_ENABLED()) { 3881 // I wish we had formatted trace names 3882 char traceName[16]; 3883 strcpy(traceName, "nRdy"); 3884 int name = track->name(); 3885 if (AudioMixer::TRACK0 <= name && 3886 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3887 name -= AudioMixer::TRACK0; 3888 traceName[4] = (name / 10) + '0'; 3889 traceName[5] = (name % 10) + '0'; 3890 } else { 3891 traceName[4] = '?'; 3892 traceName[5] = '?'; 3893 } 3894 traceName[6] = '\0'; 3895 ATRACE_INT(traceName, framesReady); 3896 } 3897 if ((framesReady >= minFrames) && track->isReady() && 3898 !track->isPaused() && !track->isTerminated()) 3899 { 3900 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3901 3902 mixedTracks++; 3903 3904 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3905 // there is an effect chain connected to the track 3906 chain.clear(); 3907 if (track->mainBuffer() != mSinkBuffer && 3908 track->mainBuffer() != mMixerBuffer) { 3909 if (mEffectBufferEnabled) { 3910 mEffectBufferValid = true; // Later can set directly. 3911 } 3912 chain = getEffectChain_l(track->sessionId()); 3913 // Delegate volume control to effect in track effect chain if needed 3914 if (chain != 0) { 3915 tracksWithEffect++; 3916 } else { 3917 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3918 "session %d", 3919 name, track->sessionId()); 3920 } 3921 } 3922 3923 3924 int param = AudioMixer::VOLUME; 3925 if (track->mFillingUpStatus == Track::FS_FILLED) { 3926 // no ramp for the first volume setting 3927 track->mFillingUpStatus = Track::FS_ACTIVE; 3928 if (track->mState == TrackBase::RESUMING) { 3929 track->mState = TrackBase::ACTIVE; 3930 param = AudioMixer::RAMP_VOLUME; 3931 } 3932 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3933 // FIXME should not make a decision based on mServer 3934 } else if (cblk->mServer != 0) { 3935 // If the track is stopped before the first frame was mixed, 3936 // do not apply ramp 3937 param = AudioMixer::RAMP_VOLUME; 3938 } 3939 3940 // compute volume for this track 3941 uint32_t vl, vr; // in U8.24 integer format 3942 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3943 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3944 vl = vr = 0; 3945 vlf = vrf = vaf = 0.; 3946 if (track->isPausing()) { 3947 track->setPaused(); 3948 } 3949 } else { 3950 3951 // read original volumes with volume control 3952 float typeVolume = mStreamTypes[track->streamType()].volume; 3953 float v = masterVolume * typeVolume; 3954 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3955 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3956 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3957 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3958 // track volumes come from shared memory, so can't be trusted and must be clamped 3959 if (vlf > GAIN_FLOAT_UNITY) { 3960 ALOGV("Track left volume out of range: %.3g", vlf); 3961 vlf = GAIN_FLOAT_UNITY; 3962 } 3963 if (vrf > GAIN_FLOAT_UNITY) { 3964 ALOGV("Track right volume out of range: %.3g", vrf); 3965 vrf = GAIN_FLOAT_UNITY; 3966 } 3967 // now apply the master volume and stream type volume 3968 vlf *= v; 3969 vrf *= v; 3970 // assuming master volume and stream type volume each go up to 1.0, 3971 // then derive vl and vr as U8.24 versions for the effect chain 3972 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3973 vl = (uint32_t) (scaleto8_24 * vlf); 3974 vr = (uint32_t) (scaleto8_24 * vrf); 3975 // vl and vr are now in U8.24 format 3976 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3977 // send level comes from shared memory and so may be corrupt 3978 if (sendLevel > MAX_GAIN_INT) { 3979 ALOGV("Track send level out of range: %04X", sendLevel); 3980 sendLevel = MAX_GAIN_INT; 3981 } 3982 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3983 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3984 } 3985 3986 // Delegate volume control to effect in track effect chain if needed 3987 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3988 // Do not ramp volume if volume is controlled by effect 3989 param = AudioMixer::VOLUME; 3990 // Update remaining floating point volume levels 3991 vlf = (float)vl / (1 << 24); 3992 vrf = (float)vr / (1 << 24); 3993 track->mHasVolumeController = true; 3994 } else { 3995 // force no volume ramp when volume controller was just disabled or removed 3996 // from effect chain to avoid volume spike 3997 if (track->mHasVolumeController) { 3998 param = AudioMixer::VOLUME; 3999 } 4000 track->mHasVolumeController = false; 4001 } 4002 4003 // XXX: these things DON'T need to be done each time 4004 mAudioMixer->setBufferProvider(name, track); 4005 mAudioMixer->enable(name); 4006 4007 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4008 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4009 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4010 mAudioMixer->setParameter( 4011 name, 4012 AudioMixer::TRACK, 4013 AudioMixer::FORMAT, (void *)track->format()); 4014 mAudioMixer->setParameter( 4015 name, 4016 AudioMixer::TRACK, 4017 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4018 mAudioMixer->setParameter( 4019 name, 4020 AudioMixer::TRACK, 4021 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4022 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4023 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4024 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4025 if (reqSampleRate == 0) { 4026 reqSampleRate = mSampleRate; 4027 } else if (reqSampleRate > maxSampleRate) { 4028 reqSampleRate = maxSampleRate; 4029 } 4030 mAudioMixer->setParameter( 4031 name, 4032 AudioMixer::RESAMPLE, 4033 AudioMixer::SAMPLE_RATE, 4034 (void *)(uintptr_t)reqSampleRate); 4035 4036 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4037 mAudioMixer->setParameter( 4038 name, 4039 AudioMixer::TIMESTRETCH, 4040 AudioMixer::PLAYBACK_RATE, 4041 &playbackRate); 4042 4043 /* 4044 * Select the appropriate output buffer for the track. 4045 * 4046 * Tracks with effects go into their own effects chain buffer 4047 * and from there into either mEffectBuffer or mSinkBuffer. 4048 * 4049 * Other tracks can use mMixerBuffer for higher precision 4050 * channel accumulation. If this buffer is enabled 4051 * (mMixerBufferEnabled true), then selected tracks will accumulate 4052 * into it. 4053 * 4054 */ 4055 if (mMixerBufferEnabled 4056 && (track->mainBuffer() == mSinkBuffer 4057 || track->mainBuffer() == mMixerBuffer)) { 4058 mAudioMixer->setParameter( 4059 name, 4060 AudioMixer::TRACK, 4061 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4062 mAudioMixer->setParameter( 4063 name, 4064 AudioMixer::TRACK, 4065 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4066 // TODO: override track->mainBuffer()? 4067 mMixerBufferValid = true; 4068 } else { 4069 mAudioMixer->setParameter( 4070 name, 4071 AudioMixer::TRACK, 4072 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4073 mAudioMixer->setParameter( 4074 name, 4075 AudioMixer::TRACK, 4076 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4077 } 4078 mAudioMixer->setParameter( 4079 name, 4080 AudioMixer::TRACK, 4081 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4082 4083 // reset retry count 4084 track->mRetryCount = kMaxTrackRetries; 4085 4086 // If one track is ready, set the mixer ready if: 4087 // - the mixer was not ready during previous round OR 4088 // - no other track is not ready 4089 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4090 mixerStatus != MIXER_TRACKS_ENABLED) { 4091 mixerStatus = MIXER_TRACKS_READY; 4092 } 4093 } else { 4094 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4095 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4096 track, framesReady, desiredFrames); 4097 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4098 } 4099 // clear effect chain input buffer if an active track underruns to avoid sending 4100 // previous audio buffer again to effects 4101 chain = getEffectChain_l(track->sessionId()); 4102 if (chain != 0) { 4103 chain->clearInputBuffer(); 4104 } 4105 4106 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4107 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4108 track->isStopped() || track->isPaused()) { 4109 // We have consumed all the buffers of this track. 4110 // Remove it from the list of active tracks. 4111 // TODO: use actual buffer filling status instead of latency when available from 4112 // audio HAL 4113 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4114 size_t framesWritten = mBytesWritten / mFrameSize; 4115 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4116 if (track->isStopped()) { 4117 track->reset(); 4118 } 4119 tracksToRemove->add(track); 4120 } 4121 } else { 4122 // No buffers for this track. Give it a few chances to 4123 // fill a buffer, then remove it from active list. 4124 if (--(track->mRetryCount) <= 0) { 4125 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4126 tracksToRemove->add(track); 4127 // indicate to client process that the track was disabled because of underrun; 4128 // it will then automatically call start() when data is available 4129 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4130 // If one track is not ready, mark the mixer also not ready if: 4131 // - the mixer was ready during previous round OR 4132 // - no other track is ready 4133 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4134 mixerStatus != MIXER_TRACKS_READY) { 4135 mixerStatus = MIXER_TRACKS_ENABLED; 4136 } 4137 } 4138 mAudioMixer->disable(name); 4139 } 4140 4141 } // local variable scope to avoid goto warning 4142track_is_ready: ; 4143 4144 } 4145 4146 // Push the new FastMixer state if necessary 4147 bool pauseAudioWatchdog = false; 4148 if (didModify) { 4149 state->mFastTracksGen++; 4150 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4151 if (kUseFastMixer == FastMixer_Dynamic && 4152 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4153 state->mCommand = FastMixerState::COLD_IDLE; 4154 state->mColdFutexAddr = &mFastMixerFutex; 4155 state->mColdGen++; 4156 mFastMixerFutex = 0; 4157 if (kUseFastMixer == FastMixer_Dynamic) { 4158 mNormalSink = mOutputSink; 4159 } 4160 // If we go into cold idle, need to wait for acknowledgement 4161 // so that fast mixer stops doing I/O. 4162 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4163 pauseAudioWatchdog = true; 4164 } 4165 } 4166 if (sq != NULL) { 4167 sq->end(didModify); 4168 sq->push(block); 4169 } 4170#ifdef AUDIO_WATCHDOG 4171 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4172 mAudioWatchdog->pause(); 4173 } 4174#endif 4175 4176 // Now perform the deferred reset on fast tracks that have stopped 4177 while (resetMask != 0) { 4178 size_t i = __builtin_ctz(resetMask); 4179 ALOG_ASSERT(i < count); 4180 resetMask &= ~(1 << i); 4181 sp<Track> t = mActiveTracks[i].promote(); 4182 if (t == 0) { 4183 continue; 4184 } 4185 Track* track = t.get(); 4186 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4187 track->reset(); 4188 } 4189 4190 // remove all the tracks that need to be... 4191 removeTracks_l(*tracksToRemove); 4192 4193 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4194 mEffectBufferValid = true; 4195 } 4196 4197 if (mEffectBufferValid) { 4198 // as long as there are effects we should clear the effects buffer, to avoid 4199 // passing a non-clean buffer to the effect chain 4200 memset(mEffectBuffer, 0, mEffectBufferSize); 4201 } 4202 // sink or mix buffer must be cleared if all tracks are connected to an 4203 // effect chain as in this case the mixer will not write to the sink or mix buffer 4204 // and track effects will accumulate into it 4205 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4206 (mixedTracks == 0 && fastTracks > 0))) { 4207 // FIXME as a performance optimization, should remember previous zero status 4208 if (mMixerBufferValid) { 4209 memset(mMixerBuffer, 0, mMixerBufferSize); 4210 // TODO: In testing, mSinkBuffer below need not be cleared because 4211 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4212 // after mixing. 4213 // 4214 // To enforce this guarantee: 4215 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4216 // (mixedTracks == 0 && fastTracks > 0)) 4217 // must imply MIXER_TRACKS_READY. 4218 // Later, we may clear buffers regardless, and skip much of this logic. 4219 } 4220 // FIXME as a performance optimization, should remember previous zero status 4221 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4222 } 4223 4224 // if any fast tracks, then status is ready 4225 mMixerStatusIgnoringFastTracks = mixerStatus; 4226 if (fastTracks > 0) { 4227 mixerStatus = MIXER_TRACKS_READY; 4228 } 4229 return mixerStatus; 4230} 4231 4232// getTrackName_l() must be called with ThreadBase::mLock held 4233int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4234 audio_format_t format, int sessionId) 4235{ 4236 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4237} 4238 4239// deleteTrackName_l() must be called with ThreadBase::mLock held 4240void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4241{ 4242 ALOGV("remove track (%d) and delete from mixer", name); 4243 mAudioMixer->deleteTrackName(name); 4244} 4245 4246// checkForNewParameter_l() must be called with ThreadBase::mLock held 4247bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4248 status_t& status) 4249{ 4250 bool reconfig = false; 4251 4252 status = NO_ERROR; 4253 4254 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4255 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4256 if (mFastMixer != 0) { 4257 FastMixerStateQueue *sq = mFastMixer->sq(); 4258 FastMixerState *state = sq->begin(); 4259 if (!(state->mCommand & FastMixerState::IDLE)) { 4260 previousCommand = state->mCommand; 4261 state->mCommand = FastMixerState::HOT_IDLE; 4262 sq->end(); 4263 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4264 } else { 4265 sq->end(false /*didModify*/); 4266 } 4267 } 4268 4269 AudioParameter param = AudioParameter(keyValuePair); 4270 int value; 4271 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4272 reconfig = true; 4273 } 4274 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4275 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4276 status = BAD_VALUE; 4277 } else { 4278 // no need to save value, since it's constant 4279 reconfig = true; 4280 } 4281 } 4282 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4283 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4284 status = BAD_VALUE; 4285 } else { 4286 // no need to save value, since it's constant 4287 reconfig = true; 4288 } 4289 } 4290 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4291 // do not accept frame count changes if tracks are open as the track buffer 4292 // size depends on frame count and correct behavior would not be guaranteed 4293 // if frame count is changed after track creation 4294 if (!mTracks.isEmpty()) { 4295 status = INVALID_OPERATION; 4296 } else { 4297 reconfig = true; 4298 } 4299 } 4300 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4301#ifdef ADD_BATTERY_DATA 4302 // when changing the audio output device, call addBatteryData to notify 4303 // the change 4304 if (mOutDevice != value) { 4305 uint32_t params = 0; 4306 // check whether speaker is on 4307 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4308 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4309 } 4310 4311 audio_devices_t deviceWithoutSpeaker 4312 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4313 // check if any other device (except speaker) is on 4314 if (value & deviceWithoutSpeaker) { 4315 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4316 } 4317 4318 if (params != 0) { 4319 addBatteryData(params); 4320 } 4321 } 4322#endif 4323 4324 // forward device change to effects that have requested to be 4325 // aware of attached audio device. 4326 if (value != AUDIO_DEVICE_NONE) { 4327 mOutDevice = value; 4328 for (size_t i = 0; i < mEffectChains.size(); i++) { 4329 mEffectChains[i]->setDevice_l(mOutDevice); 4330 } 4331 } 4332 } 4333 4334 if (status == NO_ERROR) { 4335 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4336 keyValuePair.string()); 4337 if (!mStandby && status == INVALID_OPERATION) { 4338 mOutput->standby(); 4339 mStandby = true; 4340 mBytesWritten = 0; 4341 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4342 keyValuePair.string()); 4343 } 4344 if (status == NO_ERROR && reconfig) { 4345 readOutputParameters_l(); 4346 delete mAudioMixer; 4347 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4348 for (size_t i = 0; i < mTracks.size() ; i++) { 4349 int name = getTrackName_l(mTracks[i]->mChannelMask, 4350 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4351 if (name < 0) { 4352 break; 4353 } 4354 mTracks[i]->mName = name; 4355 } 4356 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4357 } 4358 } 4359 4360 if (!(previousCommand & FastMixerState::IDLE)) { 4361 ALOG_ASSERT(mFastMixer != 0); 4362 FastMixerStateQueue *sq = mFastMixer->sq(); 4363 FastMixerState *state = sq->begin(); 4364 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4365 state->mCommand = previousCommand; 4366 sq->end(); 4367 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4368 } 4369 4370 return reconfig; 4371} 4372 4373 4374void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4375{ 4376 const size_t SIZE = 256; 4377 char buffer[SIZE]; 4378 String8 result; 4379 4380 PlaybackThread::dumpInternals(fd, args); 4381 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4382 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4383 4384 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4385 const FastMixerDumpState copy(mFastMixerDumpState); 4386 copy.dump(fd); 4387 4388#ifdef STATE_QUEUE_DUMP 4389 // Similar for state queue 4390 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4391 observerCopy.dump(fd); 4392 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4393 mutatorCopy.dump(fd); 4394#endif 4395 4396#ifdef TEE_SINK 4397 // Write the tee output to a .wav file 4398 dumpTee(fd, mTeeSource, mId); 4399#endif 4400 4401#ifdef AUDIO_WATCHDOG 4402 if (mAudioWatchdog != 0) { 4403 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4404 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4405 wdCopy.dump(fd); 4406 } 4407#endif 4408} 4409 4410uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4411{ 4412 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4413} 4414 4415uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4416{ 4417 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4418} 4419 4420void AudioFlinger::MixerThread::cacheParameters_l() 4421{ 4422 PlaybackThread::cacheParameters_l(); 4423 4424 // FIXME: Relaxed timing because of a certain device that can't meet latency 4425 // Should be reduced to 2x after the vendor fixes the driver issue 4426 // increase threshold again due to low power audio mode. The way this warning 4427 // threshold is calculated and its usefulness should be reconsidered anyway. 4428 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4429} 4430 4431// ---------------------------------------------------------------------------- 4432 4433AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4434 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4435 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4436 // mLeftVolFloat, mRightVolFloat 4437{ 4438} 4439 4440AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4441 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4442 ThreadBase::type_t type, bool systemReady) 4443 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4444 // mLeftVolFloat, mRightVolFloat 4445{ 4446} 4447 4448AudioFlinger::DirectOutputThread::~DirectOutputThread() 4449{ 4450} 4451 4452void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4453{ 4454 audio_track_cblk_t* cblk = track->cblk(); 4455 float left, right; 4456 4457 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4458 left = right = 0; 4459 } else { 4460 float typeVolume = mStreamTypes[track->streamType()].volume; 4461 float v = mMasterVolume * typeVolume; 4462 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4463 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4464 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4465 if (left > GAIN_FLOAT_UNITY) { 4466 left = GAIN_FLOAT_UNITY; 4467 } 4468 left *= v; 4469 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4470 if (right > GAIN_FLOAT_UNITY) { 4471 right = GAIN_FLOAT_UNITY; 4472 } 4473 right *= v; 4474 } 4475 4476 if (lastTrack) { 4477 if (left != mLeftVolFloat || right != mRightVolFloat) { 4478 mLeftVolFloat = left; 4479 mRightVolFloat = right; 4480 4481 // Convert volumes from float to 8.24 4482 uint32_t vl = (uint32_t)(left * (1 << 24)); 4483 uint32_t vr = (uint32_t)(right * (1 << 24)); 4484 4485 // Delegate volume control to effect in track effect chain if needed 4486 // only one effect chain can be present on DirectOutputThread, so if 4487 // there is one, the track is connected to it 4488 if (!mEffectChains.isEmpty()) { 4489 mEffectChains[0]->setVolume_l(&vl, &vr); 4490 left = (float)vl / (1 << 24); 4491 right = (float)vr / (1 << 24); 4492 } 4493 if (mOutput->stream->set_volume) { 4494 mOutput->stream->set_volume(mOutput->stream, left, right); 4495 } 4496 } 4497 } 4498} 4499 4500void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4501{ 4502 sp<Track> previousTrack = mPreviousTrack.promote(); 4503 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4504 4505 if (previousTrack != 0 && latestTrack != 0) { 4506 if (mType == DIRECT) { 4507 if (previousTrack.get() != latestTrack.get()) { 4508 mFlushPending = true; 4509 } 4510 } else /* mType == OFFLOAD */ { 4511 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4512 mFlushPending = true; 4513 } 4514 } 4515 } 4516 PlaybackThread::onAddNewTrack_l(); 4517} 4518 4519AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4520 Vector< sp<Track> > *tracksToRemove 4521) 4522{ 4523 size_t count = mActiveTracks.size(); 4524 mixer_state mixerStatus = MIXER_IDLE; 4525 bool doHwPause = false; 4526 bool doHwResume = false; 4527 4528 // find out which tracks need to be processed 4529 for (size_t i = 0; i < count; i++) { 4530 sp<Track> t = mActiveTracks[i].promote(); 4531 // The track died recently 4532 if (t == 0) { 4533 continue; 4534 } 4535 4536 if (t->isInvalid()) { 4537 ALOGW("An invalidated track shouldn't be in active list"); 4538 tracksToRemove->add(t); 4539 continue; 4540 } 4541 4542 Track* const track = t.get(); 4543 audio_track_cblk_t* cblk = track->cblk(); 4544 // Only consider last track started for volume and mixer state control. 4545 // In theory an older track could underrun and restart after the new one starts 4546 // but as we only care about the transition phase between two tracks on a 4547 // direct output, it is not a problem to ignore the underrun case. 4548 sp<Track> l = mLatestActiveTrack.promote(); 4549 bool last = l.get() == track; 4550 4551 if (track->isPausing()) { 4552 track->setPaused(); 4553 if (mHwSupportsPause && last && !mHwPaused) { 4554 doHwPause = true; 4555 mHwPaused = true; 4556 } 4557 tracksToRemove->add(track); 4558 } else if (track->isFlushPending()) { 4559 track->flushAck(); 4560 if (last) { 4561 mFlushPending = true; 4562 } 4563 } else if (track->isResumePending()) { 4564 track->resumeAck(); 4565 if (last && mHwPaused) { 4566 doHwResume = true; 4567 mHwPaused = false; 4568 } 4569 } 4570 4571 // The first time a track is added we wait 4572 // for all its buffers to be filled before processing it. 4573 // Allow draining the buffer in case the client 4574 // app does not call stop() and relies on underrun to stop: 4575 // hence the test on (track->mRetryCount > 1). 4576 // If retryCount<=1 then track is about to underrun and be removed. 4577 // Do not use a high threshold for compressed audio. 4578 uint32_t minFrames; 4579 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4580 && (track->mRetryCount > 1) && audio_is_linear_pcm(mFormat)) { 4581 minFrames = mNormalFrameCount; 4582 } else { 4583 minFrames = 1; 4584 } 4585 4586 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4587 !track->isStopping_2() && !track->isStopped()) 4588 { 4589 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4590 4591 if (track->mFillingUpStatus == Track::FS_FILLED) { 4592 track->mFillingUpStatus = Track::FS_ACTIVE; 4593 // make sure processVolume_l() will apply new volume even if 0 4594 mLeftVolFloat = mRightVolFloat = -1.0; 4595 if (!mHwSupportsPause) { 4596 track->resumeAck(); 4597 } 4598 } 4599 4600 // compute volume for this track 4601 processVolume_l(track, last); 4602 if (last) { 4603 sp<Track> previousTrack = mPreviousTrack.promote(); 4604 if (previousTrack != 0) { 4605 if (track != previousTrack.get()) { 4606 // Flush any data still being written from last track 4607 mBytesRemaining = 0; 4608 // Invalidate previous track to force a seek when resuming. 4609 previousTrack->invalidate(); 4610 } 4611 } 4612 mPreviousTrack = track; 4613 4614 // reset retry count 4615 track->mRetryCount = kMaxTrackRetriesDirect; 4616 mActiveTrack = t; 4617 mixerStatus = MIXER_TRACKS_READY; 4618 if (mHwPaused) { 4619 doHwResume = true; 4620 mHwPaused = false; 4621 } 4622 } 4623 } else { 4624 // clear effect chain input buffer if the last active track started underruns 4625 // to avoid sending previous audio buffer again to effects 4626 if (!mEffectChains.isEmpty() && last) { 4627 mEffectChains[0]->clearInputBuffer(); 4628 } 4629 if (track->isStopping_1()) { 4630 track->mState = TrackBase::STOPPING_2; 4631 if (last && mHwPaused) { 4632 doHwResume = true; 4633 mHwPaused = false; 4634 } 4635 } 4636 if ((track->sharedBuffer() != 0) || track->isStopped() || 4637 track->isStopping_2() || track->isPaused()) { 4638 // We have consumed all the buffers of this track. 4639 // Remove it from the list of active tracks. 4640 size_t audioHALFrames; 4641 if (audio_is_linear_pcm(mFormat)) { 4642 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4643 } else { 4644 audioHALFrames = 0; 4645 } 4646 4647 size_t framesWritten = mBytesWritten / mFrameSize; 4648 if (mStandby || !last || 4649 track->presentationComplete(framesWritten, audioHALFrames)) { 4650 if (track->isStopping_2()) { 4651 track->mState = TrackBase::STOPPED; 4652 } 4653 if (track->isStopped()) { 4654 track->reset(); 4655 } 4656 tracksToRemove->add(track); 4657 } 4658 } else { 4659 // No buffers for this track. Give it a few chances to 4660 // fill a buffer, then remove it from active list. 4661 // Only consider last track started for mixer state control 4662 if (--(track->mRetryCount) <= 0) { 4663 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4664 tracksToRemove->add(track); 4665 // indicate to client process that the track was disabled because of underrun; 4666 // it will then automatically call start() when data is available 4667 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4668 } else if (last) { 4669 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4670 "minFrames = %u, mFormat = %#x", 4671 track->framesReady(), minFrames, mFormat); 4672 mixerStatus = MIXER_TRACKS_ENABLED; 4673 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4674 doHwPause = true; 4675 mHwPaused = true; 4676 } 4677 } 4678 } 4679 } 4680 } 4681 4682 // if an active track did not command a flush, check for pending flush on stopped tracks 4683 if (!mFlushPending) { 4684 for (size_t i = 0; i < mTracks.size(); i++) { 4685 if (mTracks[i]->isFlushPending()) { 4686 mTracks[i]->flushAck(); 4687 mFlushPending = true; 4688 } 4689 } 4690 } 4691 4692 // make sure the pause/flush/resume sequence is executed in the right order. 4693 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4694 // before flush and then resume HW. This can happen in case of pause/flush/resume 4695 // if resume is received before pause is executed. 4696 if (mHwSupportsPause && !mStandby && 4697 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4698 mOutput->stream->pause(mOutput->stream); 4699 } 4700 if (mFlushPending) { 4701 flushHw_l(); 4702 } 4703 if (mHwSupportsPause && !mStandby && doHwResume) { 4704 mOutput->stream->resume(mOutput->stream); 4705 } 4706 // remove all the tracks that need to be... 4707 removeTracks_l(*tracksToRemove); 4708 4709 return mixerStatus; 4710} 4711 4712void AudioFlinger::DirectOutputThread::threadLoop_mix() 4713{ 4714 size_t frameCount = mFrameCount; 4715 int8_t *curBuf = (int8_t *)mSinkBuffer; 4716 // output audio to hardware 4717 while (frameCount) { 4718 AudioBufferProvider::Buffer buffer; 4719 buffer.frameCount = frameCount; 4720 status_t status = mActiveTrack->getNextBuffer(&buffer); 4721 if (status != NO_ERROR || buffer.raw == NULL) { 4722 memset(curBuf, 0, frameCount * mFrameSize); 4723 break; 4724 } 4725 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4726 frameCount -= buffer.frameCount; 4727 curBuf += buffer.frameCount * mFrameSize; 4728 mActiveTrack->releaseBuffer(&buffer); 4729 } 4730 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4731 mSleepTimeUs = 0; 4732 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4733 mActiveTrack.clear(); 4734} 4735 4736void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4737{ 4738 // do not write to HAL when paused 4739 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4740 mSleepTimeUs = mIdleSleepTimeUs; 4741 return; 4742 } 4743 if (mSleepTimeUs == 0) { 4744 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4745 mSleepTimeUs = mActiveSleepTimeUs; 4746 } else { 4747 mSleepTimeUs = mIdleSleepTimeUs; 4748 } 4749 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4750 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4751 mSleepTimeUs = 0; 4752 } 4753} 4754 4755void AudioFlinger::DirectOutputThread::threadLoop_exit() 4756{ 4757 { 4758 Mutex::Autolock _l(mLock); 4759 for (size_t i = 0; i < mTracks.size(); i++) { 4760 if (mTracks[i]->isFlushPending()) { 4761 mTracks[i]->flushAck(); 4762 mFlushPending = true; 4763 } 4764 } 4765 if (mFlushPending) { 4766 flushHw_l(); 4767 } 4768 } 4769 PlaybackThread::threadLoop_exit(); 4770} 4771 4772// must be called with thread mutex locked 4773bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4774{ 4775 bool trackPaused = false; 4776 bool trackStopped = false; 4777 4778 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4779 // after a timeout and we will enter standby then. 4780 if (mTracks.size() > 0) { 4781 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4782 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4783 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4784 } 4785 4786 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4787} 4788 4789// getTrackName_l() must be called with ThreadBase::mLock held 4790int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4791 audio_format_t format __unused, int sessionId __unused) 4792{ 4793 return 0; 4794} 4795 4796// deleteTrackName_l() must be called with ThreadBase::mLock held 4797void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4798{ 4799} 4800 4801// checkForNewParameter_l() must be called with ThreadBase::mLock held 4802bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4803 status_t& status) 4804{ 4805 bool reconfig = false; 4806 4807 status = NO_ERROR; 4808 4809 AudioParameter param = AudioParameter(keyValuePair); 4810 int value; 4811 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4812 // forward device change to effects that have requested to be 4813 // aware of attached audio device. 4814 if (value != AUDIO_DEVICE_NONE) { 4815 mOutDevice = value; 4816 for (size_t i = 0; i < mEffectChains.size(); i++) { 4817 mEffectChains[i]->setDevice_l(mOutDevice); 4818 } 4819 } 4820 } 4821 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4822 // do not accept frame count changes if tracks are open as the track buffer 4823 // size depends on frame count and correct behavior would not be garantied 4824 // if frame count is changed after track creation 4825 if (!mTracks.isEmpty()) { 4826 status = INVALID_OPERATION; 4827 } else { 4828 reconfig = true; 4829 } 4830 } 4831 if (status == NO_ERROR) { 4832 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4833 keyValuePair.string()); 4834 if (!mStandby && status == INVALID_OPERATION) { 4835 mOutput->standby(); 4836 mStandby = true; 4837 mBytesWritten = 0; 4838 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4839 keyValuePair.string()); 4840 } 4841 if (status == NO_ERROR && reconfig) { 4842 readOutputParameters_l(); 4843 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4844 } 4845 } 4846 4847 return reconfig; 4848} 4849 4850uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4851{ 4852 uint32_t time; 4853 if (audio_is_linear_pcm(mFormat)) { 4854 time = PlaybackThread::activeSleepTimeUs(); 4855 } else { 4856 time = 10000; 4857 } 4858 return time; 4859} 4860 4861uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4862{ 4863 uint32_t time; 4864 if (audio_is_linear_pcm(mFormat)) { 4865 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4866 } else { 4867 time = 10000; 4868 } 4869 return time; 4870} 4871 4872uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4873{ 4874 uint32_t time; 4875 if (audio_is_linear_pcm(mFormat)) { 4876 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4877 } else { 4878 time = 10000; 4879 } 4880 return time; 4881} 4882 4883void AudioFlinger::DirectOutputThread::cacheParameters_l() 4884{ 4885 PlaybackThread::cacheParameters_l(); 4886 4887 // use shorter standby delay as on normal output to release 4888 // hardware resources as soon as possible 4889 // no delay on outputs with HW A/V sync 4890 if (usesHwAvSync()) { 4891 mStandbyDelayNs = 0; 4892 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 4893 mStandbyDelayNs = kOffloadStandbyDelayNs; 4894 } else { 4895 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 4896 } 4897} 4898 4899void AudioFlinger::DirectOutputThread::flushHw_l() 4900{ 4901 mOutput->flush(); 4902 mHwPaused = false; 4903 mFlushPending = false; 4904} 4905 4906// ---------------------------------------------------------------------------- 4907 4908AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4909 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4910 : Thread(false /*canCallJava*/), 4911 mPlaybackThread(playbackThread), 4912 mWriteAckSequence(0), 4913 mDrainSequence(0) 4914{ 4915} 4916 4917AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4918{ 4919} 4920 4921void AudioFlinger::AsyncCallbackThread::onFirstRef() 4922{ 4923 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4924} 4925 4926bool AudioFlinger::AsyncCallbackThread::threadLoop() 4927{ 4928 while (!exitPending()) { 4929 uint32_t writeAckSequence; 4930 uint32_t drainSequence; 4931 4932 { 4933 Mutex::Autolock _l(mLock); 4934 while (!((mWriteAckSequence & 1) || 4935 (mDrainSequence & 1) || 4936 exitPending())) { 4937 mWaitWorkCV.wait(mLock); 4938 } 4939 4940 if (exitPending()) { 4941 break; 4942 } 4943 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4944 mWriteAckSequence, mDrainSequence); 4945 writeAckSequence = mWriteAckSequence; 4946 mWriteAckSequence &= ~1; 4947 drainSequence = mDrainSequence; 4948 mDrainSequence &= ~1; 4949 } 4950 { 4951 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4952 if (playbackThread != 0) { 4953 if (writeAckSequence & 1) { 4954 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4955 } 4956 if (drainSequence & 1) { 4957 playbackThread->resetDraining(drainSequence >> 1); 4958 } 4959 } 4960 } 4961 } 4962 return false; 4963} 4964 4965void AudioFlinger::AsyncCallbackThread::exit() 4966{ 4967 ALOGV("AsyncCallbackThread::exit"); 4968 Mutex::Autolock _l(mLock); 4969 requestExit(); 4970 mWaitWorkCV.broadcast(); 4971} 4972 4973void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4974{ 4975 Mutex::Autolock _l(mLock); 4976 // bit 0 is cleared 4977 mWriteAckSequence = sequence << 1; 4978} 4979 4980void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4981{ 4982 Mutex::Autolock _l(mLock); 4983 // ignore unexpected callbacks 4984 if (mWriteAckSequence & 2) { 4985 mWriteAckSequence |= 1; 4986 mWaitWorkCV.signal(); 4987 } 4988} 4989 4990void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4991{ 4992 Mutex::Autolock _l(mLock); 4993 // bit 0 is cleared 4994 mDrainSequence = sequence << 1; 4995} 4996 4997void AudioFlinger::AsyncCallbackThread::resetDraining() 4998{ 4999 Mutex::Autolock _l(mLock); 5000 // ignore unexpected callbacks 5001 if (mDrainSequence & 2) { 5002 mDrainSequence |= 1; 5003 mWaitWorkCV.signal(); 5004 } 5005} 5006 5007 5008// ---------------------------------------------------------------------------- 5009AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5010 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5011 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5012 mPausedBytesRemaining(0) 5013{ 5014 //FIXME: mStandby should be set to true by ThreadBase constructor 5015 mStandby = true; 5016} 5017 5018void AudioFlinger::OffloadThread::threadLoop_exit() 5019{ 5020 if (mFlushPending || mHwPaused) { 5021 // If a flush is pending or track was paused, just discard buffered data 5022 flushHw_l(); 5023 } else { 5024 mMixerStatus = MIXER_DRAIN_ALL; 5025 threadLoop_drain(); 5026 } 5027 if (mUseAsyncWrite) { 5028 ALOG_ASSERT(mCallbackThread != 0); 5029 mCallbackThread->exit(); 5030 } 5031 PlaybackThread::threadLoop_exit(); 5032} 5033 5034AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5035 Vector< sp<Track> > *tracksToRemove 5036) 5037{ 5038 size_t count = mActiveTracks.size(); 5039 5040 mixer_state mixerStatus = MIXER_IDLE; 5041 bool doHwPause = false; 5042 bool doHwResume = false; 5043 5044 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5045 5046 // find out which tracks need to be processed 5047 for (size_t i = 0; i < count; i++) { 5048 sp<Track> t = mActiveTracks[i].promote(); 5049 // The track died recently 5050 if (t == 0) { 5051 continue; 5052 } 5053 Track* const track = t.get(); 5054 audio_track_cblk_t* cblk = track->cblk(); 5055 // Only consider last track started for volume and mixer state control. 5056 // In theory an older track could underrun and restart after the new one starts 5057 // but as we only care about the transition phase between two tracks on a 5058 // direct output, it is not a problem to ignore the underrun case. 5059 sp<Track> l = mLatestActiveTrack.promote(); 5060 bool last = l.get() == track; 5061 5062 if (track->isInvalid()) { 5063 ALOGW("An invalidated track shouldn't be in active list"); 5064 tracksToRemove->add(track); 5065 continue; 5066 } 5067 5068 if (track->mState == TrackBase::IDLE) { 5069 ALOGW("An idle track shouldn't be in active list"); 5070 continue; 5071 } 5072 5073 if (track->isPausing()) { 5074 track->setPaused(); 5075 if (last) { 5076 if (mHwSupportsPause && !mHwPaused) { 5077 doHwPause = true; 5078 mHwPaused = true; 5079 } 5080 // If we were part way through writing the mixbuffer to 5081 // the HAL we must save this until we resume 5082 // BUG - this will be wrong if a different track is made active, 5083 // in that case we want to discard the pending data in the 5084 // mixbuffer and tell the client to present it again when the 5085 // track is resumed 5086 mPausedWriteLength = mCurrentWriteLength; 5087 mPausedBytesRemaining = mBytesRemaining; 5088 mBytesRemaining = 0; // stop writing 5089 } 5090 tracksToRemove->add(track); 5091 } else if (track->isFlushPending()) { 5092 track->flushAck(); 5093 if (last) { 5094 mFlushPending = true; 5095 } 5096 } else if (track->isResumePending()){ 5097 track->resumeAck(); 5098 if (last) { 5099 if (mPausedBytesRemaining) { 5100 // Need to continue write that was interrupted 5101 mCurrentWriteLength = mPausedWriteLength; 5102 mBytesRemaining = mPausedBytesRemaining; 5103 mPausedBytesRemaining = 0; 5104 } 5105 if (mHwPaused) { 5106 doHwResume = true; 5107 mHwPaused = false; 5108 // threadLoop_mix() will handle the case that we need to 5109 // resume an interrupted write 5110 } 5111 // enable write to audio HAL 5112 mSleepTimeUs = 0; 5113 5114 // Do not handle new data in this iteration even if track->framesReady() 5115 mixerStatus = MIXER_TRACKS_ENABLED; 5116 } 5117 } else if (track->framesReady() && track->isReady() && 5118 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5119 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5120 if (track->mFillingUpStatus == Track::FS_FILLED) { 5121 track->mFillingUpStatus = Track::FS_ACTIVE; 5122 // make sure processVolume_l() will apply new volume even if 0 5123 mLeftVolFloat = mRightVolFloat = -1.0; 5124 } 5125 5126 if (last) { 5127 sp<Track> previousTrack = mPreviousTrack.promote(); 5128 if (previousTrack != 0) { 5129 if (track != previousTrack.get()) { 5130 // Flush any data still being written from last track 5131 mBytesRemaining = 0; 5132 if (mPausedBytesRemaining) { 5133 // Last track was paused so we also need to flush saved 5134 // mixbuffer state and invalidate track so that it will 5135 // re-submit that unwritten data when it is next resumed 5136 mPausedBytesRemaining = 0; 5137 // Invalidate is a bit drastic - would be more efficient 5138 // to have a flag to tell client that some of the 5139 // previously written data was lost 5140 previousTrack->invalidate(); 5141 } 5142 // flush data already sent to the DSP if changing audio session as audio 5143 // comes from a different source. Also invalidate previous track to force a 5144 // seek when resuming. 5145 if (previousTrack->sessionId() != track->sessionId()) { 5146 previousTrack->invalidate(); 5147 } 5148 } 5149 } 5150 mPreviousTrack = track; 5151 // reset retry count 5152 track->mRetryCount = kMaxTrackRetriesOffload; 5153 mActiveTrack = t; 5154 mixerStatus = MIXER_TRACKS_READY; 5155 } 5156 } else { 5157 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5158 if (track->isStopping_1()) { 5159 // Hardware buffer can hold a large amount of audio so we must 5160 // wait for all current track's data to drain before we say 5161 // that the track is stopped. 5162 if (mBytesRemaining == 0) { 5163 // Only start draining when all data in mixbuffer 5164 // has been written 5165 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5166 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5167 // do not drain if no data was ever sent to HAL (mStandby == true) 5168 if (last && !mStandby) { 5169 // do not modify drain sequence if we are already draining. This happens 5170 // when resuming from pause after drain. 5171 if ((mDrainSequence & 1) == 0) { 5172 mSleepTimeUs = 0; 5173 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5174 mixerStatus = MIXER_DRAIN_TRACK; 5175 mDrainSequence += 2; 5176 } 5177 if (mHwPaused) { 5178 // It is possible to move from PAUSED to STOPPING_1 without 5179 // a resume so we must ensure hardware is running 5180 doHwResume = true; 5181 mHwPaused = false; 5182 } 5183 } 5184 } 5185 } else if (track->isStopping_2()) { 5186 // Drain has completed or we are in standby, signal presentation complete 5187 if (!(mDrainSequence & 1) || !last || mStandby) { 5188 track->mState = TrackBase::STOPPED; 5189 size_t audioHALFrames = 5190 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5191 size_t framesWritten = 5192 mBytesWritten / mOutput->getFrameSize(); 5193 track->presentationComplete(framesWritten, audioHALFrames); 5194 track->reset(); 5195 tracksToRemove->add(track); 5196 } 5197 } else { 5198 // No buffers for this track. Give it a few chances to 5199 // fill a buffer, then remove it from active list. 5200 if (--(track->mRetryCount) <= 0) { 5201 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5202 track->name()); 5203 tracksToRemove->add(track); 5204 // indicate to client process that the track was disabled because of underrun; 5205 // it will then automatically call start() when data is available 5206 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5207 } else if (last){ 5208 mixerStatus = MIXER_TRACKS_ENABLED; 5209 } 5210 } 5211 } 5212 // compute volume for this track 5213 processVolume_l(track, last); 5214 } 5215 5216 // make sure the pause/flush/resume sequence is executed in the right order. 5217 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5218 // before flush and then resume HW. This can happen in case of pause/flush/resume 5219 // if resume is received before pause is executed. 5220 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5221 mOutput->stream->pause(mOutput->stream); 5222 } 5223 if (mFlushPending) { 5224 flushHw_l(); 5225 } 5226 if (!mStandby && doHwResume) { 5227 mOutput->stream->resume(mOutput->stream); 5228 } 5229 5230 // remove all the tracks that need to be... 5231 removeTracks_l(*tracksToRemove); 5232 5233 return mixerStatus; 5234} 5235 5236// must be called with thread mutex locked 5237bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5238{ 5239 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5240 mWriteAckSequence, mDrainSequence); 5241 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5242 return true; 5243 } 5244 return false; 5245} 5246 5247bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5248{ 5249 Mutex::Autolock _l(mLock); 5250 return waitingAsyncCallback_l(); 5251} 5252 5253void AudioFlinger::OffloadThread::flushHw_l() 5254{ 5255 DirectOutputThread::flushHw_l(); 5256 // Flush anything still waiting in the mixbuffer 5257 mCurrentWriteLength = 0; 5258 mBytesRemaining = 0; 5259 mPausedWriteLength = 0; 5260 mPausedBytesRemaining = 0; 5261 5262 if (mUseAsyncWrite) { 5263 // discard any pending drain or write ack by incrementing sequence 5264 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5265 mDrainSequence = (mDrainSequence + 2) & ~1; 5266 ALOG_ASSERT(mCallbackThread != 0); 5267 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5268 mCallbackThread->setDraining(mDrainSequence); 5269 } 5270} 5271 5272// ---------------------------------------------------------------------------- 5273 5274AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5275 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5276 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5277 systemReady, DUPLICATING), 5278 mWaitTimeMs(UINT_MAX) 5279{ 5280 addOutputTrack(mainThread); 5281} 5282 5283AudioFlinger::DuplicatingThread::~DuplicatingThread() 5284{ 5285 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5286 mOutputTracks[i]->destroy(); 5287 } 5288} 5289 5290void AudioFlinger::DuplicatingThread::threadLoop_mix() 5291{ 5292 // mix buffers... 5293 if (outputsReady(outputTracks)) { 5294 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5295 } else { 5296 if (mMixerBufferValid) { 5297 memset(mMixerBuffer, 0, mMixerBufferSize); 5298 } else { 5299 memset(mSinkBuffer, 0, mSinkBufferSize); 5300 } 5301 } 5302 mSleepTimeUs = 0; 5303 writeFrames = mNormalFrameCount; 5304 mCurrentWriteLength = mSinkBufferSize; 5305 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5306} 5307 5308void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5309{ 5310 if (mSleepTimeUs == 0) { 5311 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5312 mSleepTimeUs = mActiveSleepTimeUs; 5313 } else { 5314 mSleepTimeUs = mIdleSleepTimeUs; 5315 } 5316 } else if (mBytesWritten != 0) { 5317 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5318 writeFrames = mNormalFrameCount; 5319 memset(mSinkBuffer, 0, mSinkBufferSize); 5320 } else { 5321 // flush remaining overflow buffers in output tracks 5322 writeFrames = 0; 5323 } 5324 mSleepTimeUs = 0; 5325 } 5326} 5327 5328ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5329{ 5330 for (size_t i = 0; i < outputTracks.size(); i++) { 5331 outputTracks[i]->write(mSinkBuffer, writeFrames); 5332 } 5333 mStandby = false; 5334 return (ssize_t)mSinkBufferSize; 5335} 5336 5337void AudioFlinger::DuplicatingThread::threadLoop_standby() 5338{ 5339 // DuplicatingThread implements standby by stopping all tracks 5340 for (size_t i = 0; i < outputTracks.size(); i++) { 5341 outputTracks[i]->stop(); 5342 } 5343} 5344 5345void AudioFlinger::DuplicatingThread::saveOutputTracks() 5346{ 5347 outputTracks = mOutputTracks; 5348} 5349 5350void AudioFlinger::DuplicatingThread::clearOutputTracks() 5351{ 5352 outputTracks.clear(); 5353} 5354 5355void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5356{ 5357 Mutex::Autolock _l(mLock); 5358 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5359 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5360 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5361 const size_t frameCount = 5362 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5363 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5364 // from different OutputTracks and their associated MixerThreads (e.g. one may 5365 // nearly empty and the other may be dropping data). 5366 5367 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5368 this, 5369 mSampleRate, 5370 mFormat, 5371 mChannelMask, 5372 frameCount, 5373 IPCThreadState::self()->getCallingUid()); 5374 if (outputTrack->cblk() != NULL) { 5375 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5376 mOutputTracks.add(outputTrack); 5377 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5378 updateWaitTime_l(); 5379 } 5380} 5381 5382void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5383{ 5384 Mutex::Autolock _l(mLock); 5385 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5386 if (mOutputTracks[i]->thread() == thread) { 5387 mOutputTracks[i]->destroy(); 5388 mOutputTracks.removeAt(i); 5389 updateWaitTime_l(); 5390 if (thread->getOutput() == mOutput) { 5391 mOutput = NULL; 5392 } 5393 return; 5394 } 5395 } 5396 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5397} 5398 5399// caller must hold mLock 5400void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5401{ 5402 mWaitTimeMs = UINT_MAX; 5403 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5404 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5405 if (strong != 0) { 5406 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5407 if (waitTimeMs < mWaitTimeMs) { 5408 mWaitTimeMs = waitTimeMs; 5409 } 5410 } 5411 } 5412} 5413 5414 5415bool AudioFlinger::DuplicatingThread::outputsReady( 5416 const SortedVector< sp<OutputTrack> > &outputTracks) 5417{ 5418 for (size_t i = 0; i < outputTracks.size(); i++) { 5419 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5420 if (thread == 0) { 5421 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5422 outputTracks[i].get()); 5423 return false; 5424 } 5425 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5426 // see note at standby() declaration 5427 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5428 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5429 thread.get()); 5430 return false; 5431 } 5432 } 5433 return true; 5434} 5435 5436uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5437{ 5438 return (mWaitTimeMs * 1000) / 2; 5439} 5440 5441void AudioFlinger::DuplicatingThread::cacheParameters_l() 5442{ 5443 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5444 updateWaitTime_l(); 5445 5446 MixerThread::cacheParameters_l(); 5447} 5448 5449// ---------------------------------------------------------------------------- 5450// Record 5451// ---------------------------------------------------------------------------- 5452 5453AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5454 AudioStreamIn *input, 5455 audio_io_handle_t id, 5456 audio_devices_t outDevice, 5457 audio_devices_t inDevice, 5458 bool systemReady 5459#ifdef TEE_SINK 5460 , const sp<NBAIO_Sink>& teeSink 5461#endif 5462 ) : 5463 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5464 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5465 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5466 mRsmpInRear(0) 5467#ifdef TEE_SINK 5468 , mTeeSink(teeSink) 5469#endif 5470 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5471 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5472 // mFastCapture below 5473 , mFastCaptureFutex(0) 5474 // mInputSource 5475 // mPipeSink 5476 // mPipeSource 5477 , mPipeFramesP2(0) 5478 // mPipeMemory 5479 // mFastCaptureNBLogWriter 5480 , mFastTrackAvail(false) 5481{ 5482 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5483 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5484 5485 readInputParameters_l(); 5486 5487 // create an NBAIO source for the HAL input stream, and negotiate 5488 mInputSource = new AudioStreamInSource(input->stream); 5489 size_t numCounterOffers = 0; 5490 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5491 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5492 ALOG_ASSERT(index == 0); 5493 5494 // initialize fast capture depending on configuration 5495 bool initFastCapture; 5496 switch (kUseFastCapture) { 5497 case FastCapture_Never: 5498 initFastCapture = false; 5499 break; 5500 case FastCapture_Always: 5501 initFastCapture = true; 5502 break; 5503 case FastCapture_Static: 5504 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5505 break; 5506 // case FastCapture_Dynamic: 5507 } 5508 5509 if (initFastCapture) { 5510 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5511 NBAIO_Format format = mInputSource->format(); 5512 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5513 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5514 void *pipeBuffer; 5515 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5516 sp<IMemory> pipeMemory; 5517 if ((roHeap == 0) || 5518 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5519 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5520 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5521 goto failed; 5522 } 5523 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5524 memset(pipeBuffer, 0, pipeSize); 5525 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5526 const NBAIO_Format offers[1] = {format}; 5527 size_t numCounterOffers = 0; 5528 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5529 ALOG_ASSERT(index == 0); 5530 mPipeSink = pipe; 5531 PipeReader *pipeReader = new PipeReader(*pipe); 5532 numCounterOffers = 0; 5533 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5534 ALOG_ASSERT(index == 0); 5535 mPipeSource = pipeReader; 5536 mPipeFramesP2 = pipeFramesP2; 5537 mPipeMemory = pipeMemory; 5538 5539 // create fast capture 5540 mFastCapture = new FastCapture(); 5541 FastCaptureStateQueue *sq = mFastCapture->sq(); 5542#ifdef STATE_QUEUE_DUMP 5543 // FIXME 5544#endif 5545 FastCaptureState *state = sq->begin(); 5546 state->mCblk = NULL; 5547 state->mInputSource = mInputSource.get(); 5548 state->mInputSourceGen++; 5549 state->mPipeSink = pipe; 5550 state->mPipeSinkGen++; 5551 state->mFrameCount = mFrameCount; 5552 state->mCommand = FastCaptureState::COLD_IDLE; 5553 // already done in constructor initialization list 5554 //mFastCaptureFutex = 0; 5555 state->mColdFutexAddr = &mFastCaptureFutex; 5556 state->mColdGen++; 5557 state->mDumpState = &mFastCaptureDumpState; 5558#ifdef TEE_SINK 5559 // FIXME 5560#endif 5561 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5562 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5563 sq->end(); 5564 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5565 5566 // start the fast capture 5567 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5568 pid_t tid = mFastCapture->getTid(); 5569 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5570#ifdef AUDIO_WATCHDOG 5571 // FIXME 5572#endif 5573 5574 mFastTrackAvail = true; 5575 } 5576failed: ; 5577 5578 // FIXME mNormalSource 5579} 5580 5581AudioFlinger::RecordThread::~RecordThread() 5582{ 5583 if (mFastCapture != 0) { 5584 FastCaptureStateQueue *sq = mFastCapture->sq(); 5585 FastCaptureState *state = sq->begin(); 5586 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5587 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5588 if (old == -1) { 5589 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5590 } 5591 } 5592 state->mCommand = FastCaptureState::EXIT; 5593 sq->end(); 5594 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5595 mFastCapture->join(); 5596 mFastCapture.clear(); 5597 } 5598 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5599 mAudioFlinger->unregisterWriter(mNBLogWriter); 5600 free(mRsmpInBuffer); 5601} 5602 5603void AudioFlinger::RecordThread::onFirstRef() 5604{ 5605 run(mThreadName, PRIORITY_URGENT_AUDIO); 5606} 5607 5608bool AudioFlinger::RecordThread::threadLoop() 5609{ 5610 nsecs_t lastWarning = 0; 5611 5612 inputStandBy(); 5613 5614reacquire_wakelock: 5615 sp<RecordTrack> activeTrack; 5616 int activeTracksGen; 5617 { 5618 Mutex::Autolock _l(mLock); 5619 size_t size = mActiveTracks.size(); 5620 activeTracksGen = mActiveTracksGen; 5621 if (size > 0) { 5622 // FIXME an arbitrary choice 5623 activeTrack = mActiveTracks[0]; 5624 acquireWakeLock_l(activeTrack->uid()); 5625 if (size > 1) { 5626 SortedVector<int> tmp; 5627 for (size_t i = 0; i < size; i++) { 5628 tmp.add(mActiveTracks[i]->uid()); 5629 } 5630 updateWakeLockUids_l(tmp); 5631 } 5632 } else { 5633 acquireWakeLock_l(-1); 5634 } 5635 } 5636 5637 // used to request a deferred sleep, to be executed later while mutex is unlocked 5638 uint32_t sleepUs = 0; 5639 5640 // loop while there is work to do 5641 for (;;) { 5642 Vector< sp<EffectChain> > effectChains; 5643 5644 // sleep with mutex unlocked 5645 if (sleepUs > 0) { 5646 ATRACE_BEGIN("sleep"); 5647 usleep(sleepUs); 5648 ATRACE_END(); 5649 sleepUs = 0; 5650 } 5651 5652 // activeTracks accumulates a copy of a subset of mActiveTracks 5653 Vector< sp<RecordTrack> > activeTracks; 5654 5655 // reference to the (first and only) active fast track 5656 sp<RecordTrack> fastTrack; 5657 5658 // reference to a fast track which is about to be removed 5659 sp<RecordTrack> fastTrackToRemove; 5660 5661 { // scope for mLock 5662 Mutex::Autolock _l(mLock); 5663 5664 processConfigEvents_l(); 5665 5666 // check exitPending here because checkForNewParameters_l() and 5667 // checkForNewParameters_l() can temporarily release mLock 5668 if (exitPending()) { 5669 break; 5670 } 5671 5672 // if no active track(s), then standby and release wakelock 5673 size_t size = mActiveTracks.size(); 5674 if (size == 0) { 5675 standbyIfNotAlreadyInStandby(); 5676 // exitPending() can't become true here 5677 releaseWakeLock_l(); 5678 ALOGV("RecordThread: loop stopping"); 5679 // go to sleep 5680 mWaitWorkCV.wait(mLock); 5681 ALOGV("RecordThread: loop starting"); 5682 goto reacquire_wakelock; 5683 } 5684 5685 if (mActiveTracksGen != activeTracksGen) { 5686 activeTracksGen = mActiveTracksGen; 5687 SortedVector<int> tmp; 5688 for (size_t i = 0; i < size; i++) { 5689 tmp.add(mActiveTracks[i]->uid()); 5690 } 5691 updateWakeLockUids_l(tmp); 5692 } 5693 5694 bool doBroadcast = false; 5695 for (size_t i = 0; i < size; ) { 5696 5697 activeTrack = mActiveTracks[i]; 5698 if (activeTrack->isTerminated()) { 5699 if (activeTrack->isFastTrack()) { 5700 ALOG_ASSERT(fastTrackToRemove == 0); 5701 fastTrackToRemove = activeTrack; 5702 } 5703 removeTrack_l(activeTrack); 5704 mActiveTracks.remove(activeTrack); 5705 mActiveTracksGen++; 5706 size--; 5707 continue; 5708 } 5709 5710 TrackBase::track_state activeTrackState = activeTrack->mState; 5711 switch (activeTrackState) { 5712 5713 case TrackBase::PAUSING: 5714 mActiveTracks.remove(activeTrack); 5715 mActiveTracksGen++; 5716 doBroadcast = true; 5717 size--; 5718 continue; 5719 5720 case TrackBase::STARTING_1: 5721 sleepUs = 10000; 5722 i++; 5723 continue; 5724 5725 case TrackBase::STARTING_2: 5726 doBroadcast = true; 5727 mStandby = false; 5728 activeTrack->mState = TrackBase::ACTIVE; 5729 break; 5730 5731 case TrackBase::ACTIVE: 5732 break; 5733 5734 case TrackBase::IDLE: 5735 i++; 5736 continue; 5737 5738 default: 5739 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5740 } 5741 5742 activeTracks.add(activeTrack); 5743 i++; 5744 5745 if (activeTrack->isFastTrack()) { 5746 ALOG_ASSERT(!mFastTrackAvail); 5747 ALOG_ASSERT(fastTrack == 0); 5748 fastTrack = activeTrack; 5749 } 5750 } 5751 if (doBroadcast) { 5752 mStartStopCond.broadcast(); 5753 } 5754 5755 // sleep if there are no active tracks to process 5756 if (activeTracks.size() == 0) { 5757 if (sleepUs == 0) { 5758 sleepUs = kRecordThreadSleepUs; 5759 } 5760 continue; 5761 } 5762 sleepUs = 0; 5763 5764 lockEffectChains_l(effectChains); 5765 } 5766 5767 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5768 5769 size_t size = effectChains.size(); 5770 for (size_t i = 0; i < size; i++) { 5771 // thread mutex is not locked, but effect chain is locked 5772 effectChains[i]->process_l(); 5773 } 5774 5775 // Push a new fast capture state if fast capture is not already running, or cblk change 5776 if (mFastCapture != 0) { 5777 FastCaptureStateQueue *sq = mFastCapture->sq(); 5778 FastCaptureState *state = sq->begin(); 5779 bool didModify = false; 5780 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5781 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5782 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5783 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5784 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5785 if (old == -1) { 5786 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5787 } 5788 } 5789 state->mCommand = FastCaptureState::READ_WRITE; 5790#if 0 // FIXME 5791 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5792 FastThreadDumpState::kSamplingNforLowRamDevice : 5793 FastThreadDumpState::kSamplingN); 5794#endif 5795 didModify = true; 5796 } 5797 audio_track_cblk_t *cblkOld = state->mCblk; 5798 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5799 if (cblkNew != cblkOld) { 5800 state->mCblk = cblkNew; 5801 // block until acked if removing a fast track 5802 if (cblkOld != NULL) { 5803 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5804 } 5805 didModify = true; 5806 } 5807 sq->end(didModify); 5808 if (didModify) { 5809 sq->push(block); 5810#if 0 5811 if (kUseFastCapture == FastCapture_Dynamic) { 5812 mNormalSource = mPipeSource; 5813 } 5814#endif 5815 } 5816 } 5817 5818 // now run the fast track destructor with thread mutex unlocked 5819 fastTrackToRemove.clear(); 5820 5821 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5822 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5823 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5824 // If destination is non-contiguous, first read past the nominal end of buffer, then 5825 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5826 5827 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5828 ssize_t framesRead; 5829 5830 // If an NBAIO source is present, use it to read the normal capture's data 5831 if (mPipeSource != 0) { 5832 size_t framesToRead = mBufferSize / mFrameSize; 5833 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5834 framesToRead, AudioBufferProvider::kInvalidPTS); 5835 if (framesRead == 0) { 5836 // since pipe is non-blocking, simulate blocking input 5837 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5838 } 5839 // otherwise use the HAL / AudioStreamIn directly 5840 } else { 5841 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5842 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5843 if (bytesRead < 0) { 5844 framesRead = bytesRead; 5845 } else { 5846 framesRead = bytesRead / mFrameSize; 5847 } 5848 } 5849 5850 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5851 ALOGE("read failed: framesRead=%d", framesRead); 5852 // Force input into standby so that it tries to recover at next read attempt 5853 inputStandBy(); 5854 sleepUs = kRecordThreadSleepUs; 5855 } 5856 if (framesRead <= 0) { 5857 goto unlock; 5858 } 5859 ALOG_ASSERT(framesRead > 0); 5860 5861 if (mTeeSink != 0) { 5862 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5863 } 5864 // If destination is non-contiguous, we now correct for reading past end of buffer. 5865 { 5866 size_t part1 = mRsmpInFramesP2 - rear; 5867 if ((size_t) framesRead > part1) { 5868 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5869 (framesRead - part1) * mFrameSize); 5870 } 5871 } 5872 rear = mRsmpInRear += framesRead; 5873 5874 size = activeTracks.size(); 5875 // loop over each active track 5876 for (size_t i = 0; i < size; i++) { 5877 activeTrack = activeTracks[i]; 5878 5879 // skip fast tracks, as those are handled directly by FastCapture 5880 if (activeTrack->isFastTrack()) { 5881 continue; 5882 } 5883 5884 // TODO: This code probably should be moved to RecordTrack. 5885 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5886 5887 enum { 5888 OVERRUN_UNKNOWN, 5889 OVERRUN_TRUE, 5890 OVERRUN_FALSE 5891 } overrun = OVERRUN_UNKNOWN; 5892 5893 // loop over getNextBuffer to handle circular sink 5894 for (;;) { 5895 5896 activeTrack->mSink.frameCount = ~0; 5897 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5898 size_t framesOut = activeTrack->mSink.frameCount; 5899 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5900 5901 // check available frames and handle overrun conditions 5902 // if the record track isn't draining fast enough. 5903 bool hasOverrun; 5904 size_t framesIn; 5905 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5906 if (hasOverrun) { 5907 overrun = OVERRUN_TRUE; 5908 } 5909 if (framesOut == 0 || framesIn == 0) { 5910 break; 5911 } 5912 5913 // Don't allow framesOut to be larger than what is possible with resampling 5914 // from framesIn. 5915 // This isn't strictly necessary but helps limit buffer resizing in 5916 // RecordBufferConverter. TODO: remove when no longer needed. 5917 framesOut = min(framesOut, 5918 destinationFramesPossible( 5919 framesIn, mSampleRate, activeTrack->mSampleRate)); 5920 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5921 framesOut = activeTrack->mRecordBufferConverter->convert( 5922 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5923 5924 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5925 overrun = OVERRUN_FALSE; 5926 } 5927 5928 if (activeTrack->mFramesToDrop == 0) { 5929 if (framesOut > 0) { 5930 activeTrack->mSink.frameCount = framesOut; 5931 activeTrack->releaseBuffer(&activeTrack->mSink); 5932 } 5933 } else { 5934 // FIXME could do a partial drop of framesOut 5935 if (activeTrack->mFramesToDrop > 0) { 5936 activeTrack->mFramesToDrop -= framesOut; 5937 if (activeTrack->mFramesToDrop <= 0) { 5938 activeTrack->clearSyncStartEvent(); 5939 } 5940 } else { 5941 activeTrack->mFramesToDrop += framesOut; 5942 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5943 activeTrack->mSyncStartEvent->isCancelled()) { 5944 ALOGW("Synced record %s, session %d, trigger session %d", 5945 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5946 activeTrack->sessionId(), 5947 (activeTrack->mSyncStartEvent != 0) ? 5948 activeTrack->mSyncStartEvent->triggerSession() : 0); 5949 activeTrack->clearSyncStartEvent(); 5950 } 5951 } 5952 } 5953 5954 if (framesOut == 0) { 5955 break; 5956 } 5957 } 5958 5959 switch (overrun) { 5960 case OVERRUN_TRUE: 5961 // client isn't retrieving buffers fast enough 5962 if (!activeTrack->setOverflow()) { 5963 nsecs_t now = systemTime(); 5964 // FIXME should lastWarning per track? 5965 if ((now - lastWarning) > kWarningThrottleNs) { 5966 ALOGW("RecordThread: buffer overflow"); 5967 lastWarning = now; 5968 } 5969 } 5970 break; 5971 case OVERRUN_FALSE: 5972 activeTrack->clearOverflow(); 5973 break; 5974 case OVERRUN_UNKNOWN: 5975 break; 5976 } 5977 5978 } 5979 5980unlock: 5981 // enable changes in effect chain 5982 unlockEffectChains(effectChains); 5983 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5984 } 5985 5986 standbyIfNotAlreadyInStandby(); 5987 5988 { 5989 Mutex::Autolock _l(mLock); 5990 for (size_t i = 0; i < mTracks.size(); i++) { 5991 sp<RecordTrack> track = mTracks[i]; 5992 track->invalidate(); 5993 } 5994 mActiveTracks.clear(); 5995 mActiveTracksGen++; 5996 mStartStopCond.broadcast(); 5997 } 5998 5999 releaseWakeLock(); 6000 6001 ALOGV("RecordThread %p exiting", this); 6002 return false; 6003} 6004 6005void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6006{ 6007 if (!mStandby) { 6008 inputStandBy(); 6009 mStandby = true; 6010 } 6011} 6012 6013void AudioFlinger::RecordThread::inputStandBy() 6014{ 6015 // Idle the fast capture if it's currently running 6016 if (mFastCapture != 0) { 6017 FastCaptureStateQueue *sq = mFastCapture->sq(); 6018 FastCaptureState *state = sq->begin(); 6019 if (!(state->mCommand & FastCaptureState::IDLE)) { 6020 state->mCommand = FastCaptureState::COLD_IDLE; 6021 state->mColdFutexAddr = &mFastCaptureFutex; 6022 state->mColdGen++; 6023 mFastCaptureFutex = 0; 6024 sq->end(); 6025 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6026 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6027#if 0 6028 if (kUseFastCapture == FastCapture_Dynamic) { 6029 // FIXME 6030 } 6031#endif 6032#ifdef AUDIO_WATCHDOG 6033 // FIXME 6034#endif 6035 } else { 6036 sq->end(false /*didModify*/); 6037 } 6038 } 6039 mInput->stream->common.standby(&mInput->stream->common); 6040} 6041 6042// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6043sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6044 const sp<AudioFlinger::Client>& client, 6045 uint32_t sampleRate, 6046 audio_format_t format, 6047 audio_channel_mask_t channelMask, 6048 size_t *pFrameCount, 6049 int sessionId, 6050 size_t *notificationFrames, 6051 int uid, 6052 IAudioFlinger::track_flags_t *flags, 6053 pid_t tid, 6054 status_t *status) 6055{ 6056 size_t frameCount = *pFrameCount; 6057 sp<RecordTrack> track; 6058 status_t lStatus; 6059 6060 // client expresses a preference for FAST, but we get the final say 6061 if (*flags & IAudioFlinger::TRACK_FAST) { 6062 if ( 6063 // we formerly checked for a callback handler (non-0 tid), 6064 // but that is no longer required for TRANSFER_OBTAIN mode 6065 // 6066 // frame count is not specified, or is exactly the pipe depth 6067 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6068 // PCM data 6069 audio_is_linear_pcm(format) && 6070 // native format 6071 (format == mFormat) && 6072 // native channel mask 6073 (channelMask == mChannelMask) && 6074 // native hardware sample rate 6075 (sampleRate == mSampleRate) && 6076 // record thread has an associated fast capture 6077 hasFastCapture() && 6078 // there are sufficient fast track slots available 6079 mFastTrackAvail 6080 ) { 6081 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6082 frameCount, mFrameCount); 6083 } else { 6084 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6085 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6086 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6087 frameCount, mFrameCount, mPipeFramesP2, 6088 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6089 hasFastCapture(), tid, mFastTrackAvail); 6090 *flags &= ~IAudioFlinger::TRACK_FAST; 6091 } 6092 } 6093 6094 // compute track buffer size in frames, and suggest the notification frame count 6095 if (*flags & IAudioFlinger::TRACK_FAST) { 6096 // fast track: frame count is exactly the pipe depth 6097 frameCount = mPipeFramesP2; 6098 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6099 *notificationFrames = mFrameCount; 6100 } else { 6101 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6102 // or 20 ms if there is a fast capture 6103 // TODO This could be a roundupRatio inline, and const 6104 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6105 * sampleRate + mSampleRate - 1) / mSampleRate; 6106 // minimum number of notification periods is at least kMinNotifications, 6107 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6108 static const size_t kMinNotifications = 3; 6109 static const uint32_t kMinMs = 30; 6110 // TODO This could be a roundupRatio inline 6111 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6112 // TODO This could be a roundupRatio inline 6113 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6114 maxNotificationFrames; 6115 const size_t minFrameCount = maxNotificationFrames * 6116 max(kMinNotifications, minNotificationsByMs); 6117 frameCount = max(frameCount, minFrameCount); 6118 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6119 *notificationFrames = maxNotificationFrames; 6120 } 6121 } 6122 *pFrameCount = frameCount; 6123 6124 lStatus = initCheck(); 6125 if (lStatus != NO_ERROR) { 6126 ALOGE("createRecordTrack_l() audio driver not initialized"); 6127 goto Exit; 6128 } 6129 6130 { // scope for mLock 6131 Mutex::Autolock _l(mLock); 6132 6133 track = new RecordTrack(this, client, sampleRate, 6134 format, channelMask, frameCount, NULL, sessionId, uid, 6135 *flags, TrackBase::TYPE_DEFAULT); 6136 6137 lStatus = track->initCheck(); 6138 if (lStatus != NO_ERROR) { 6139 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6140 // track must be cleared from the caller as the caller has the AF lock 6141 goto Exit; 6142 } 6143 mTracks.add(track); 6144 6145 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6146 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6147 mAudioFlinger->btNrecIsOff(); 6148 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6149 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6150 6151 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6152 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6153 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6154 // so ask activity manager to do this on our behalf 6155 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6156 } 6157 } 6158 6159 lStatus = NO_ERROR; 6160 6161Exit: 6162 *status = lStatus; 6163 return track; 6164} 6165 6166status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6167 AudioSystem::sync_event_t event, 6168 int triggerSession) 6169{ 6170 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6171 sp<ThreadBase> strongMe = this; 6172 status_t status = NO_ERROR; 6173 6174 if (event == AudioSystem::SYNC_EVENT_NONE) { 6175 recordTrack->clearSyncStartEvent(); 6176 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6177 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6178 triggerSession, 6179 recordTrack->sessionId(), 6180 syncStartEventCallback, 6181 recordTrack); 6182 // Sync event can be cancelled by the trigger session if the track is not in a 6183 // compatible state in which case we start record immediately 6184 if (recordTrack->mSyncStartEvent->isCancelled()) { 6185 recordTrack->clearSyncStartEvent(); 6186 } else { 6187 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6188 recordTrack->mFramesToDrop = - 6189 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6190 } 6191 } 6192 6193 { 6194 // This section is a rendezvous between binder thread executing start() and RecordThread 6195 AutoMutex lock(mLock); 6196 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6197 if (recordTrack->mState == TrackBase::PAUSING) { 6198 ALOGV("active record track PAUSING -> ACTIVE"); 6199 recordTrack->mState = TrackBase::ACTIVE; 6200 } else { 6201 ALOGV("active record track state %d", recordTrack->mState); 6202 } 6203 return status; 6204 } 6205 6206 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6207 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6208 // or using a separate command thread 6209 recordTrack->mState = TrackBase::STARTING_1; 6210 mActiveTracks.add(recordTrack); 6211 mActiveTracksGen++; 6212 status_t status = NO_ERROR; 6213 if (recordTrack->isExternalTrack()) { 6214 mLock.unlock(); 6215 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6216 mLock.lock(); 6217 // FIXME should verify that recordTrack is still in mActiveTracks 6218 if (status != NO_ERROR) { 6219 mActiveTracks.remove(recordTrack); 6220 mActiveTracksGen++; 6221 recordTrack->clearSyncStartEvent(); 6222 ALOGV("RecordThread::start error %d", status); 6223 return status; 6224 } 6225 } 6226 // Catch up with current buffer indices if thread is already running. 6227 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6228 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6229 // see previously buffered data before it called start(), but with greater risk of overrun. 6230 6231 recordTrack->mResamplerBufferProvider->reset(); 6232 // clear any converter state as new data will be discontinuous 6233 recordTrack->mRecordBufferConverter->reset(); 6234 recordTrack->mState = TrackBase::STARTING_2; 6235 // signal thread to start 6236 mWaitWorkCV.broadcast(); 6237 if (mActiveTracks.indexOf(recordTrack) < 0) { 6238 ALOGV("Record failed to start"); 6239 status = BAD_VALUE; 6240 goto startError; 6241 } 6242 return status; 6243 } 6244 6245startError: 6246 if (recordTrack->isExternalTrack()) { 6247 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6248 } 6249 recordTrack->clearSyncStartEvent(); 6250 // FIXME I wonder why we do not reset the state here? 6251 return status; 6252} 6253 6254void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6255{ 6256 sp<SyncEvent> strongEvent = event.promote(); 6257 6258 if (strongEvent != 0) { 6259 sp<RefBase> ptr = strongEvent->cookie().promote(); 6260 if (ptr != 0) { 6261 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6262 recordTrack->handleSyncStartEvent(strongEvent); 6263 } 6264 } 6265} 6266 6267bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6268 ALOGV("RecordThread::stop"); 6269 AutoMutex _l(mLock); 6270 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6271 return false; 6272 } 6273 // note that threadLoop may still be processing the track at this point [without lock] 6274 recordTrack->mState = TrackBase::PAUSING; 6275 // do not wait for mStartStopCond if exiting 6276 if (exitPending()) { 6277 return true; 6278 } 6279 // FIXME incorrect usage of wait: no explicit predicate or loop 6280 mStartStopCond.wait(mLock); 6281 // if we have been restarted, recordTrack is in mActiveTracks here 6282 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6283 ALOGV("Record stopped OK"); 6284 return true; 6285 } 6286 return false; 6287} 6288 6289bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6290{ 6291 return false; 6292} 6293 6294status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6295{ 6296#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6297 if (!isValidSyncEvent(event)) { 6298 return BAD_VALUE; 6299 } 6300 6301 int eventSession = event->triggerSession(); 6302 status_t ret = NAME_NOT_FOUND; 6303 6304 Mutex::Autolock _l(mLock); 6305 6306 for (size_t i = 0; i < mTracks.size(); i++) { 6307 sp<RecordTrack> track = mTracks[i]; 6308 if (eventSession == track->sessionId()) { 6309 (void) track->setSyncEvent(event); 6310 ret = NO_ERROR; 6311 } 6312 } 6313 return ret; 6314#else 6315 return BAD_VALUE; 6316#endif 6317} 6318 6319// destroyTrack_l() must be called with ThreadBase::mLock held 6320void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6321{ 6322 track->terminate(); 6323 track->mState = TrackBase::STOPPED; 6324 // active tracks are removed by threadLoop() 6325 if (mActiveTracks.indexOf(track) < 0) { 6326 removeTrack_l(track); 6327 } 6328} 6329 6330void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6331{ 6332 mTracks.remove(track); 6333 // need anything related to effects here? 6334 if (track->isFastTrack()) { 6335 ALOG_ASSERT(!mFastTrackAvail); 6336 mFastTrackAvail = true; 6337 } 6338} 6339 6340void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6341{ 6342 dumpInternals(fd, args); 6343 dumpTracks(fd, args); 6344 dumpEffectChains(fd, args); 6345} 6346 6347void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6348{ 6349 dprintf(fd, "\nInput thread %p:\n", this); 6350 6351 dumpBase(fd, args); 6352 6353 if (mActiveTracks.size() == 0) { 6354 dprintf(fd, " No active record clients\n"); 6355 } 6356 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6357 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6358 6359 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6360 const FastCaptureDumpState copy(mFastCaptureDumpState); 6361 copy.dump(fd); 6362} 6363 6364void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6365{ 6366 const size_t SIZE = 256; 6367 char buffer[SIZE]; 6368 String8 result; 6369 6370 size_t numtracks = mTracks.size(); 6371 size_t numactive = mActiveTracks.size(); 6372 size_t numactiveseen = 0; 6373 dprintf(fd, " %d Tracks", numtracks); 6374 if (numtracks) { 6375 dprintf(fd, " of which %d are active\n", numactive); 6376 RecordTrack::appendDumpHeader(result); 6377 for (size_t i = 0; i < numtracks ; ++i) { 6378 sp<RecordTrack> track = mTracks[i]; 6379 if (track != 0) { 6380 bool active = mActiveTracks.indexOf(track) >= 0; 6381 if (active) { 6382 numactiveseen++; 6383 } 6384 track->dump(buffer, SIZE, active); 6385 result.append(buffer); 6386 } 6387 } 6388 } else { 6389 dprintf(fd, "\n"); 6390 } 6391 6392 if (numactiveseen != numactive) { 6393 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6394 " not in the track list\n"); 6395 result.append(buffer); 6396 RecordTrack::appendDumpHeader(result); 6397 for (size_t i = 0; i < numactive; ++i) { 6398 sp<RecordTrack> track = mActiveTracks[i]; 6399 if (mTracks.indexOf(track) < 0) { 6400 track->dump(buffer, SIZE, true); 6401 result.append(buffer); 6402 } 6403 } 6404 6405 } 6406 write(fd, result.string(), result.size()); 6407} 6408 6409 6410void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6411{ 6412 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6413 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6414 mRsmpInFront = recordThread->mRsmpInRear; 6415 mRsmpInUnrel = 0; 6416} 6417 6418void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6419 size_t *framesAvailable, bool *hasOverrun) 6420{ 6421 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6422 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6423 const int32_t rear = recordThread->mRsmpInRear; 6424 const int32_t front = mRsmpInFront; 6425 const ssize_t filled = rear - front; 6426 6427 size_t framesIn; 6428 bool overrun = false; 6429 if (filled < 0) { 6430 // should not happen, but treat like a massive overrun and re-sync 6431 framesIn = 0; 6432 mRsmpInFront = rear; 6433 overrun = true; 6434 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6435 framesIn = (size_t) filled; 6436 } else { 6437 // client is not keeping up with server, but give it latest data 6438 framesIn = recordThread->mRsmpInFrames; 6439 mRsmpInFront = /* front = */ rear - framesIn; 6440 overrun = true; 6441 } 6442 if (framesAvailable != NULL) { 6443 *framesAvailable = framesIn; 6444 } 6445 if (hasOverrun != NULL) { 6446 *hasOverrun = overrun; 6447 } 6448} 6449 6450// AudioBufferProvider interface 6451status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6452 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6453{ 6454 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6455 if (threadBase == 0) { 6456 buffer->frameCount = 0; 6457 buffer->raw = NULL; 6458 return NOT_ENOUGH_DATA; 6459 } 6460 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6461 int32_t rear = recordThread->mRsmpInRear; 6462 int32_t front = mRsmpInFront; 6463 ssize_t filled = rear - front; 6464 // FIXME should not be P2 (don't want to increase latency) 6465 // FIXME if client not keeping up, discard 6466 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6467 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6468 front &= recordThread->mRsmpInFramesP2 - 1; 6469 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6470 if (part1 > (size_t) filled) { 6471 part1 = filled; 6472 } 6473 size_t ask = buffer->frameCount; 6474 ALOG_ASSERT(ask > 0); 6475 if (part1 > ask) { 6476 part1 = ask; 6477 } 6478 if (part1 == 0) { 6479 // out of data is fine since the resampler will return a short-count. 6480 buffer->raw = NULL; 6481 buffer->frameCount = 0; 6482 mRsmpInUnrel = 0; 6483 return NOT_ENOUGH_DATA; 6484 } 6485 6486 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6487 buffer->frameCount = part1; 6488 mRsmpInUnrel = part1; 6489 return NO_ERROR; 6490} 6491 6492// AudioBufferProvider interface 6493void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6494 AudioBufferProvider::Buffer* buffer) 6495{ 6496 size_t stepCount = buffer->frameCount; 6497 if (stepCount == 0) { 6498 return; 6499 } 6500 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6501 mRsmpInUnrel -= stepCount; 6502 mRsmpInFront += stepCount; 6503 buffer->raw = NULL; 6504 buffer->frameCount = 0; 6505} 6506 6507AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6508 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6509 uint32_t srcSampleRate, 6510 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6511 uint32_t dstSampleRate) : 6512 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6513 // mSrcFormat 6514 // mSrcSampleRate 6515 // mDstChannelMask 6516 // mDstFormat 6517 // mDstSampleRate 6518 // mSrcChannelCount 6519 // mDstChannelCount 6520 // mDstFrameSize 6521 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6522 mResampler(NULL), 6523 mIsLegacyDownmix(false), 6524 mIsLegacyUpmix(false), 6525 mRequiresFloat(false), 6526 mInputConverterProvider(NULL) 6527{ 6528 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6529 dstChannelMask, dstFormat, dstSampleRate); 6530} 6531 6532AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6533 free(mBuf); 6534 delete mResampler; 6535 delete mInputConverterProvider; 6536} 6537 6538size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6539 AudioBufferProvider *provider, size_t frames) 6540{ 6541 if (mInputConverterProvider != NULL) { 6542 mInputConverterProvider->setBufferProvider(provider); 6543 provider = mInputConverterProvider; 6544 } 6545 6546 if (mResampler == NULL) { 6547 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6548 mSrcSampleRate, mSrcFormat, mDstFormat); 6549 6550 AudioBufferProvider::Buffer buffer; 6551 for (size_t i = frames; i > 0; ) { 6552 buffer.frameCount = i; 6553 status_t status = provider->getNextBuffer(&buffer, 0); 6554 if (status != OK || buffer.frameCount == 0) { 6555 frames -= i; // cannot fill request. 6556 break; 6557 } 6558 // format convert to destination buffer 6559 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6560 6561 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6562 i -= buffer.frameCount; 6563 provider->releaseBuffer(&buffer); 6564 } 6565 } else { 6566 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6567 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6568 6569 // reallocate buffer if needed 6570 if (mBufFrameSize != 0 && mBufFrames < frames) { 6571 free(mBuf); 6572 mBufFrames = frames; 6573 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6574 } 6575 // resampler accumulates, but we only have one source track 6576 memset(mBuf, 0, frames * mBufFrameSize); 6577 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6578 // format convert to destination buffer 6579 convertResampler(dst, mBuf, frames); 6580 } 6581 return frames; 6582} 6583 6584status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6585 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6586 uint32_t srcSampleRate, 6587 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6588 uint32_t dstSampleRate) 6589{ 6590 // quick evaluation if there is any change. 6591 if (mSrcFormat == srcFormat 6592 && mSrcChannelMask == srcChannelMask 6593 && mSrcSampleRate == srcSampleRate 6594 && mDstFormat == dstFormat 6595 && mDstChannelMask == dstChannelMask 6596 && mDstSampleRate == dstSampleRate) { 6597 return NO_ERROR; 6598 } 6599 6600 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6601 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6602 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6603 const bool valid = 6604 audio_is_input_channel(srcChannelMask) 6605 && audio_is_input_channel(dstChannelMask) 6606 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6607 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6608 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6609 ; // no upsampling checks for now 6610 if (!valid) { 6611 return BAD_VALUE; 6612 } 6613 6614 mSrcFormat = srcFormat; 6615 mSrcChannelMask = srcChannelMask; 6616 mSrcSampleRate = srcSampleRate; 6617 mDstFormat = dstFormat; 6618 mDstChannelMask = dstChannelMask; 6619 mDstSampleRate = dstSampleRate; 6620 6621 // compute derived parameters 6622 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6623 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6624 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6625 6626 // do we need to resample? 6627 delete mResampler; 6628 mResampler = NULL; 6629 if (mSrcSampleRate != mDstSampleRate) { 6630 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6631 mSrcChannelCount, mDstSampleRate); 6632 mResampler->setSampleRate(mSrcSampleRate); 6633 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6634 } 6635 6636 // are we running legacy channel conversion modes? 6637 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6638 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6639 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6640 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6641 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6642 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6643 6644 // do we need to process in float? 6645 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6646 6647 // do we need a staging buffer to convert for destination (we can still optimize this)? 6648 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6649 if (mResampler != NULL) { 6650 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6651 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6652 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6653 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6654 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6655 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6656 } else { 6657 mBufFrameSize = 0; 6658 } 6659 mBufFrames = 0; // force the buffer to be resized. 6660 6661 // do we need an input converter buffer provider to give us float? 6662 delete mInputConverterProvider; 6663 mInputConverterProvider = NULL; 6664 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6665 mInputConverterProvider = new ReformatBufferProvider( 6666 audio_channel_count_from_in_mask(mSrcChannelMask), 6667 mSrcFormat, 6668 AUDIO_FORMAT_PCM_FLOAT, 6669 256 /* provider buffer frame count */); 6670 } 6671 6672 // do we need a remixer to do channel mask conversion 6673 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6674 (void) memcpy_by_index_array_initialization_from_channel_mask( 6675 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6676 } 6677 return NO_ERROR; 6678} 6679 6680void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6681 void *dst, const void *src, size_t frames) 6682{ 6683 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6684 if (mBufFrameSize != 0 && mBufFrames < frames) { 6685 free(mBuf); 6686 mBufFrames = frames; 6687 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6688 } 6689 // do we need to do legacy upmix and downmix? 6690 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6691 void *dstBuf = mBuf != NULL ? mBuf : dst; 6692 if (mIsLegacyUpmix) { 6693 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6694 (const float *)src, frames); 6695 } else /*mIsLegacyDownmix */ { 6696 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6697 (const float *)src, frames); 6698 } 6699 if (mBuf != NULL) { 6700 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6701 frames * mDstChannelCount); 6702 } 6703 return; 6704 } 6705 // do we need to do channel mask conversion? 6706 if (mSrcChannelMask != mDstChannelMask) { 6707 void *dstBuf = mBuf != NULL ? mBuf : dst; 6708 memcpy_by_index_array(dstBuf, mDstChannelCount, 6709 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6710 if (dstBuf == dst) { 6711 return; // format is the same 6712 } 6713 } 6714 // convert to destination buffer 6715 const void *convertBuf = mBuf != NULL ? mBuf : src; 6716 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6717 frames * mDstChannelCount); 6718} 6719 6720void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6721 void *dst, /*not-a-const*/ void *src, size_t frames) 6722{ 6723 // src buffer format is ALWAYS float when entering this routine 6724 if (mIsLegacyUpmix) { 6725 ; // mono to stereo already handled by resampler 6726 } else if (mIsLegacyDownmix 6727 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6728 // the resampler outputs stereo for mono input channel (a feature?) 6729 // must convert to mono 6730 downmix_to_mono_float_from_stereo_float((float *)src, 6731 (const float *)src, frames); 6732 } else if (mSrcChannelMask != mDstChannelMask) { 6733 // convert to mono channel again for channel mask conversion (could be skipped 6734 // with further optimization). 6735 if (mSrcChannelCount == 1) { 6736 downmix_to_mono_float_from_stereo_float((float *)src, 6737 (const float *)src, frames); 6738 } 6739 // convert to destination format (in place, OK as float is larger than other types) 6740 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6741 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6742 frames * mSrcChannelCount); 6743 } 6744 // channel convert and save to dst 6745 memcpy_by_index_array(dst, mDstChannelCount, 6746 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6747 return; 6748 } 6749 // convert to destination format and save to dst 6750 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6751 frames * mDstChannelCount); 6752} 6753 6754bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6755 status_t& status) 6756{ 6757 bool reconfig = false; 6758 6759 status = NO_ERROR; 6760 6761 audio_format_t reqFormat = mFormat; 6762 uint32_t samplingRate = mSampleRate; 6763 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6764 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6765 6766 AudioParameter param = AudioParameter(keyValuePair); 6767 int value; 6768 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6769 // channel count change can be requested. Do we mandate the first client defines the 6770 // HAL sampling rate and channel count or do we allow changes on the fly? 6771 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6772 samplingRate = value; 6773 reconfig = true; 6774 } 6775 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6776 if (!audio_is_linear_pcm((audio_format_t) value)) { 6777 status = BAD_VALUE; 6778 } else { 6779 reqFormat = (audio_format_t) value; 6780 reconfig = true; 6781 } 6782 } 6783 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6784 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6785 if (!audio_is_input_channel(mask) || 6786 audio_channel_count_from_in_mask(mask) > FCC_8) { 6787 status = BAD_VALUE; 6788 } else { 6789 channelMask = mask; 6790 reconfig = true; 6791 } 6792 } 6793 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6794 // do not accept frame count changes if tracks are open as the track buffer 6795 // size depends on frame count and correct behavior would not be guaranteed 6796 // if frame count is changed after track creation 6797 if (mActiveTracks.size() > 0) { 6798 status = INVALID_OPERATION; 6799 } else { 6800 reconfig = true; 6801 } 6802 } 6803 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6804 // forward device change to effects that have requested to be 6805 // aware of attached audio device. 6806 for (size_t i = 0; i < mEffectChains.size(); i++) { 6807 mEffectChains[i]->setDevice_l(value); 6808 } 6809 6810 // store input device and output device but do not forward output device to audio HAL. 6811 // Note that status is ignored by the caller for output device 6812 // (see AudioFlinger::setParameters() 6813 if (audio_is_output_devices(value)) { 6814 mOutDevice = value; 6815 status = BAD_VALUE; 6816 } else { 6817 mInDevice = value; 6818 if (value != AUDIO_DEVICE_NONE) { 6819 mPrevInDevice = value; 6820 } 6821 // disable AEC and NS if the device is a BT SCO headset supporting those 6822 // pre processings 6823 if (mTracks.size() > 0) { 6824 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6825 mAudioFlinger->btNrecIsOff(); 6826 for (size_t i = 0; i < mTracks.size(); i++) { 6827 sp<RecordTrack> track = mTracks[i]; 6828 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6829 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6830 } 6831 } 6832 } 6833 } 6834 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6835 mAudioSource != (audio_source_t)value) { 6836 // forward device change to effects that have requested to be 6837 // aware of attached audio device. 6838 for (size_t i = 0; i < mEffectChains.size(); i++) { 6839 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6840 } 6841 mAudioSource = (audio_source_t)value; 6842 } 6843 6844 if (status == NO_ERROR) { 6845 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6846 keyValuePair.string()); 6847 if (status == INVALID_OPERATION) { 6848 inputStandBy(); 6849 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6850 keyValuePair.string()); 6851 } 6852 if (reconfig) { 6853 if (status == BAD_VALUE && 6854 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6855 audio_is_linear_pcm(reqFormat) && 6856 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6857 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6858 audio_channel_count_from_in_mask( 6859 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6860 status = NO_ERROR; 6861 } 6862 if (status == NO_ERROR) { 6863 readInputParameters_l(); 6864 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6865 } 6866 } 6867 } 6868 6869 return reconfig; 6870} 6871 6872String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6873{ 6874 Mutex::Autolock _l(mLock); 6875 if (initCheck() != NO_ERROR) { 6876 return String8(); 6877 } 6878 6879 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6880 const String8 out_s8(s); 6881 free(s); 6882 return out_s8; 6883} 6884 6885void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 6886 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6887 6888 desc->mIoHandle = mId; 6889 6890 switch (event) { 6891 case AUDIO_INPUT_OPENED: 6892 case AUDIO_INPUT_CONFIG_CHANGED: 6893 desc->mPatch = mPatch; 6894 desc->mChannelMask = mChannelMask; 6895 desc->mSamplingRate = mSampleRate; 6896 desc->mFormat = mFormat; 6897 desc->mFrameCount = mFrameCount; 6898 desc->mLatency = 0; 6899 break; 6900 6901 case AUDIO_INPUT_CLOSED: 6902 default: 6903 break; 6904 } 6905 mAudioFlinger->ioConfigChanged(event, desc, pid); 6906} 6907 6908void AudioFlinger::RecordThread::readInputParameters_l() 6909{ 6910 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6911 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6912 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6913 if (mChannelCount > FCC_8) { 6914 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6915 } 6916 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6917 mFormat = mHALFormat; 6918 if (!audio_is_linear_pcm(mFormat)) { 6919 ALOGE("HAL format %#x is not linear pcm", mFormat); 6920 } 6921 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6922 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6923 mFrameCount = mBufferSize / mFrameSize; 6924 // This is the formula for calculating the temporary buffer size. 6925 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6926 // 1 full output buffer, regardless of the alignment of the available input. 6927 // The value is somewhat arbitrary, and could probably be even larger. 6928 // A larger value should allow more old data to be read after a track calls start(), 6929 // without increasing latency. 6930 // 6931 // Note this is independent of the maximum downsampling ratio permitted for capture. 6932 mRsmpInFrames = mFrameCount * 7; 6933 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6934 free(mRsmpInBuffer); 6935 6936 // TODO optimize audio capture buffer sizes ... 6937 // Here we calculate the size of the sliding buffer used as a source 6938 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6939 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6940 // be better to have it derived from the pipe depth in the long term. 6941 // The current value is higher than necessary. However it should not add to latency. 6942 6943 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6944 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6945 6946 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6947 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6948} 6949 6950uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6951{ 6952 Mutex::Autolock _l(mLock); 6953 if (initCheck() != NO_ERROR) { 6954 return 0; 6955 } 6956 6957 return mInput->stream->get_input_frames_lost(mInput->stream); 6958} 6959 6960uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6961{ 6962 Mutex::Autolock _l(mLock); 6963 uint32_t result = 0; 6964 if (getEffectChain_l(sessionId) != 0) { 6965 result = EFFECT_SESSION; 6966 } 6967 6968 for (size_t i = 0; i < mTracks.size(); ++i) { 6969 if (sessionId == mTracks[i]->sessionId()) { 6970 result |= TRACK_SESSION; 6971 break; 6972 } 6973 } 6974 6975 return result; 6976} 6977 6978KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6979{ 6980 KeyedVector<int, bool> ids; 6981 Mutex::Autolock _l(mLock); 6982 for (size_t j = 0; j < mTracks.size(); ++j) { 6983 sp<RecordThread::RecordTrack> track = mTracks[j]; 6984 int sessionId = track->sessionId(); 6985 if (ids.indexOfKey(sessionId) < 0) { 6986 ids.add(sessionId, true); 6987 } 6988 } 6989 return ids; 6990} 6991 6992AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6993{ 6994 Mutex::Autolock _l(mLock); 6995 AudioStreamIn *input = mInput; 6996 mInput = NULL; 6997 return input; 6998} 6999 7000// this method must always be called either with ThreadBase mLock held or inside the thread loop 7001audio_stream_t* AudioFlinger::RecordThread::stream() const 7002{ 7003 if (mInput == NULL) { 7004 return NULL; 7005 } 7006 return &mInput->stream->common; 7007} 7008 7009status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7010{ 7011 // only one chain per input thread 7012 if (mEffectChains.size() != 0) { 7013 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7014 return INVALID_OPERATION; 7015 } 7016 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7017 chain->setThread(this); 7018 chain->setInBuffer(NULL); 7019 chain->setOutBuffer(NULL); 7020 7021 checkSuspendOnAddEffectChain_l(chain); 7022 7023 // make sure enabled pre processing effects state is communicated to the HAL as we 7024 // just moved them to a new input stream. 7025 chain->syncHalEffectsState(); 7026 7027 mEffectChains.add(chain); 7028 7029 return NO_ERROR; 7030} 7031 7032size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7033{ 7034 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7035 ALOGW_IF(mEffectChains.size() != 1, 7036 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7037 chain.get(), mEffectChains.size(), this); 7038 if (mEffectChains.size() == 1) { 7039 mEffectChains.removeAt(0); 7040 } 7041 return 0; 7042} 7043 7044status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7045 audio_patch_handle_t *handle) 7046{ 7047 status_t status = NO_ERROR; 7048 7049 // store new device and send to effects 7050 mInDevice = patch->sources[0].ext.device.type; 7051 mPatch = *patch; 7052 for (size_t i = 0; i < mEffectChains.size(); i++) { 7053 mEffectChains[i]->setDevice_l(mInDevice); 7054 } 7055 7056 // disable AEC and NS if the device is a BT SCO headset supporting those 7057 // pre processings 7058 if (mTracks.size() > 0) { 7059 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7060 mAudioFlinger->btNrecIsOff(); 7061 for (size_t i = 0; i < mTracks.size(); i++) { 7062 sp<RecordTrack> track = mTracks[i]; 7063 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7064 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7065 } 7066 } 7067 7068 // store new source and send to effects 7069 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7070 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7071 for (size_t i = 0; i < mEffectChains.size(); i++) { 7072 mEffectChains[i]->setAudioSource_l(mAudioSource); 7073 } 7074 } 7075 7076 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7077 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7078 status = hwDevice->create_audio_patch(hwDevice, 7079 patch->num_sources, 7080 patch->sources, 7081 patch->num_sinks, 7082 patch->sinks, 7083 handle); 7084 } else { 7085 char *address; 7086 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7087 address = audio_device_address_to_parameter( 7088 patch->sources[0].ext.device.type, 7089 patch->sources[0].ext.device.address); 7090 } else { 7091 address = (char *)calloc(1, 1); 7092 } 7093 AudioParameter param = AudioParameter(String8(address)); 7094 free(address); 7095 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7096 (int)patch->sources[0].ext.device.type); 7097 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7098 (int)patch->sinks[0].ext.mix.usecase.source); 7099 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7100 param.toString().string()); 7101 *handle = AUDIO_PATCH_HANDLE_NONE; 7102 } 7103 7104 if (mInDevice != mPrevInDevice) { 7105 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7106 mPrevInDevice = mInDevice; 7107 } 7108 7109 return status; 7110} 7111 7112status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7113{ 7114 status_t status = NO_ERROR; 7115 7116 mInDevice = AUDIO_DEVICE_NONE; 7117 7118 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7119 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7120 status = hwDevice->release_audio_patch(hwDevice, handle); 7121 } else { 7122 AudioParameter param; 7123 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7124 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7125 param.toString().string()); 7126 } 7127 return status; 7128} 7129 7130void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7131{ 7132 Mutex::Autolock _l(mLock); 7133 mTracks.add(record); 7134} 7135 7136void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7137{ 7138 Mutex::Autolock _l(mLock); 7139 destroyTrack_l(record); 7140} 7141 7142void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7143{ 7144 ThreadBase::getAudioPortConfig(config); 7145 config->role = AUDIO_PORT_ROLE_SINK; 7146 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7147 config->ext.mix.usecase.source = mAudioSource; 7148} 7149 7150} // namespace android 7151