Threads.cpp revision 8a397d583a4f4cf24ad88facaf2fd33990cfb811
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#include <mediautils/BatteryNotifier.h> 52 53#include <powermanager/PowerManager.h> 54 55#include <common_time/cc_helper.h> 56#include <common_time/local_clock.h> 57 58#include "AudioFlinger.h" 59#include "AudioMixer.h" 60#include "BufferProviders.h" 61#include "FastMixer.h" 62#include "FastCapture.h" 63#include "ServiceUtilities.h" 64#include "mediautils/SchedulingPolicyService.h" 65 66#ifdef ADD_BATTERY_DATA 67#include <media/IMediaPlayerService.h> 68#include <media/IMediaDeathNotifier.h> 69#endif 70 71#ifdef DEBUG_CPU_USAGE 72#include <cpustats/CentralTendencyStatistics.h> 73#include <cpustats/ThreadCpuUsage.h> 74#endif 75 76// ---------------------------------------------------------------------------- 77 78// Note: the following macro is used for extremely verbose logging message. In 79// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 80// 0; but one side effect of this is to turn all LOGV's as well. Some messages 81// are so verbose that we want to suppress them even when we have ALOG_ASSERT 82// turned on. Do not uncomment the #def below unless you really know what you 83// are doing and want to see all of the extremely verbose messages. 84//#define VERY_VERY_VERBOSE_LOGGING 85#ifdef VERY_VERY_VERBOSE_LOGGING 86#define ALOGVV ALOGV 87#else 88#define ALOGVV(a...) do { } while(0) 89#endif 90 91// TODO: Move these macro/inlines to a header file. 92#define max(a, b) ((a) > (b) ? (a) : (b)) 93template <typename T> 94static inline T min(const T& a, const T& b) 95{ 96 return a < b ? a : b; 97} 98 99#ifndef ARRAY_SIZE 100#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 101#endif 102 103namespace android { 104 105// retry counts for buffer fill timeout 106// 50 * ~20msecs = 1 second 107static const int8_t kMaxTrackRetries = 50; 108static const int8_t kMaxTrackStartupRetries = 50; 109// allow less retry attempts on direct output thread. 110// direct outputs can be a scarce resource in audio hardware and should 111// be released as quickly as possible. 112static const int8_t kMaxTrackRetriesDirect = 2; 113 114// don't warn about blocked writes or record buffer overflows more often than this 115static const nsecs_t kWarningThrottleNs = seconds(5); 116 117// RecordThread loop sleep time upon application overrun or audio HAL read error 118static const int kRecordThreadSleepUs = 5000; 119 120// maximum time to wait in sendConfigEvent_l() for a status to be received 121static const nsecs_t kConfigEventTimeoutNs = seconds(2); 122 123// minimum sleep time for the mixer thread loop when tracks are active but in underrun 124static const uint32_t kMinThreadSleepTimeUs = 5000; 125// maximum divider applied to the active sleep time in the mixer thread loop 126static const uint32_t kMaxThreadSleepTimeShift = 2; 127 128// minimum normal sink buffer size, expressed in milliseconds rather than frames 129// FIXME This should be based on experimentally observed scheduling jitter 130static const uint32_t kMinNormalSinkBufferSizeMs = 20; 131// maximum normal sink buffer size 132static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 133 134// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 135// FIXME This should be based on experimentally observed scheduling jitter 136static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 137 138// Offloaded output thread standby delay: allows track transition without going to standby 139static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 140 141// Whether to use fast mixer 142static const enum { 143 FastMixer_Never, // never initialize or use: for debugging only 144 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 145 // normal mixer multiplier is 1 146 FastMixer_Static, // initialize if needed, then use all the time if initialized, 147 // multiplier is calculated based on min & max normal mixer buffer size 148 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 149 // multiplier is calculated based on min & max normal mixer buffer size 150 // FIXME for FastMixer_Dynamic: 151 // Supporting this option will require fixing HALs that can't handle large writes. 152 // For example, one HAL implementation returns an error from a large write, 153 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 154 // We could either fix the HAL implementations, or provide a wrapper that breaks 155 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 156} kUseFastMixer = FastMixer_Static; 157 158// Whether to use fast capture 159static const enum { 160 FastCapture_Never, // never initialize or use: for debugging only 161 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 162 FastCapture_Static, // initialize if needed, then use all the time if initialized 163} kUseFastCapture = FastCapture_Static; 164 165// Priorities for requestPriority 166static const int kPriorityAudioApp = 2; 167static const int kPriorityFastMixer = 3; 168static const int kPriorityFastCapture = 3; 169 170// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 171// for the track. The client then sub-divides this into smaller buffers for its use. 172// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 173// So for now we just assume that client is double-buffered for fast tracks. 174// FIXME It would be better for client to tell AudioFlinger the value of N, 175// so AudioFlinger could allocate the right amount of memory. 176// See the client's minBufCount and mNotificationFramesAct calculations for details. 177 178// This is the default value, if not specified by property. 179static const int kFastTrackMultiplier = 2; 180 181// The minimum and maximum allowed values 182static const int kFastTrackMultiplierMin = 1; 183static const int kFastTrackMultiplierMax = 2; 184 185// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 186static int sFastTrackMultiplier = kFastTrackMultiplier; 187 188// See Thread::readOnlyHeap(). 189// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 190// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 191// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 192static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 193 194// ---------------------------------------------------------------------------- 195 196static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 197 198static void sFastTrackMultiplierInit() 199{ 200 char value[PROPERTY_VALUE_MAX]; 201 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 202 char *endptr; 203 unsigned long ul = strtoul(value, &endptr, 0); 204 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 205 sFastTrackMultiplier = (int) ul; 206 } 207 } 208} 209 210// ---------------------------------------------------------------------------- 211 212#ifdef ADD_BATTERY_DATA 213// To collect the amplifier usage 214static void addBatteryData(uint32_t params) { 215 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 216 if (service == NULL) { 217 // it already logged 218 return; 219 } 220 221 service->addBatteryData(params); 222} 223#endif 224 225 226// ---------------------------------------------------------------------------- 227// CPU Stats 228// ---------------------------------------------------------------------------- 229 230class CpuStats { 231public: 232 CpuStats(); 233 void sample(const String8 &title); 234#ifdef DEBUG_CPU_USAGE 235private: 236 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 237 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 238 239 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 240 241 int mCpuNum; // thread's current CPU number 242 int mCpukHz; // frequency of thread's current CPU in kHz 243#endif 244}; 245 246CpuStats::CpuStats() 247#ifdef DEBUG_CPU_USAGE 248 : mCpuNum(-1), mCpukHz(-1) 249#endif 250{ 251} 252 253void CpuStats::sample(const String8 &title 254#ifndef DEBUG_CPU_USAGE 255 __unused 256#endif 257 ) { 258#ifdef DEBUG_CPU_USAGE 259 // get current thread's delta CPU time in wall clock ns 260 double wcNs; 261 bool valid = mCpuUsage.sampleAndEnable(wcNs); 262 263 // record sample for wall clock statistics 264 if (valid) { 265 mWcStats.sample(wcNs); 266 } 267 268 // get the current CPU number 269 int cpuNum = sched_getcpu(); 270 271 // get the current CPU frequency in kHz 272 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 273 274 // check if either CPU number or frequency changed 275 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 276 mCpuNum = cpuNum; 277 mCpukHz = cpukHz; 278 // ignore sample for purposes of cycles 279 valid = false; 280 } 281 282 // if no change in CPU number or frequency, then record sample for cycle statistics 283 if (valid && mCpukHz > 0) { 284 double cycles = wcNs * cpukHz * 0.000001; 285 mHzStats.sample(cycles); 286 } 287 288 unsigned n = mWcStats.n(); 289 // mCpuUsage.elapsed() is expensive, so don't call it every loop 290 if ((n & 127) == 1) { 291 long long elapsed = mCpuUsage.elapsed(); 292 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 293 double perLoop = elapsed / (double) n; 294 double perLoop100 = perLoop * 0.01; 295 double perLoop1k = perLoop * 0.001; 296 double mean = mWcStats.mean(); 297 double stddev = mWcStats.stddev(); 298 double minimum = mWcStats.minimum(); 299 double maximum = mWcStats.maximum(); 300 double meanCycles = mHzStats.mean(); 301 double stddevCycles = mHzStats.stddev(); 302 double minCycles = mHzStats.minimum(); 303 double maxCycles = mHzStats.maximum(); 304 mCpuUsage.resetElapsed(); 305 mWcStats.reset(); 306 mHzStats.reset(); 307 ALOGD("CPU usage for %s over past %.1f secs\n" 308 " (%u mixer loops at %.1f mean ms per loop):\n" 309 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 310 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 311 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 312 title.string(), 313 elapsed * .000000001, n, perLoop * .000001, 314 mean * .001, 315 stddev * .001, 316 minimum * .001, 317 maximum * .001, 318 mean / perLoop100, 319 stddev / perLoop100, 320 minimum / perLoop100, 321 maximum / perLoop100, 322 meanCycles / perLoop1k, 323 stddevCycles / perLoop1k, 324 minCycles / perLoop1k, 325 maxCycles / perLoop1k); 326 327 } 328 } 329#endif 330}; 331 332// ---------------------------------------------------------------------------- 333// ThreadBase 334// ---------------------------------------------------------------------------- 335 336// static 337const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 338{ 339 switch (type) { 340 case MIXER: 341 return "MIXER"; 342 case DIRECT: 343 return "DIRECT"; 344 case DUPLICATING: 345 return "DUPLICATING"; 346 case RECORD: 347 return "RECORD"; 348 case OFFLOAD: 349 return "OFFLOAD"; 350 default: 351 return "unknown"; 352 } 353} 354 355String8 devicesToString(audio_devices_t devices) 356{ 357 static const struct mapping { 358 audio_devices_t mDevices; 359 const char * mString; 360 } mappingsOut[] = { 361 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 362 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 363 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 364 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 365 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 366 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 367 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 368 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 369 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 370 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 371 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 372 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 373 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 374 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 375 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 376 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 377 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 378 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 379 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 380 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 381 {AUDIO_DEVICE_OUT_FM, "FM"}, 382 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 383 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 384 {AUDIO_DEVICE_OUT_IP, "IP"}, 385 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 386 }, mappingsIn[] = { 387 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 388 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 389 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 390 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 391 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 392 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 393 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 394 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 395 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 396 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 397 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 398 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 399 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 400 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 401 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 402 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 403 {AUDIO_DEVICE_IN_LINE, "LINE"}, 404 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 405 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 406 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 407 {AUDIO_DEVICE_IN_IP, "IP"}, 408 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 409 }; 410 String8 result; 411 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 412 const mapping *entry; 413 if (devices & AUDIO_DEVICE_BIT_IN) { 414 devices &= ~AUDIO_DEVICE_BIT_IN; 415 entry = mappingsIn; 416 } else { 417 entry = mappingsOut; 418 } 419 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 420 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 421 if (devices & entry->mDevices) { 422 if (!result.isEmpty()) { 423 result.append("|"); 424 } 425 result.append(entry->mString); 426 } 427 } 428 if (devices & ~allDevices) { 429 if (!result.isEmpty()) { 430 result.append("|"); 431 } 432 result.appendFormat("0x%X", devices & ~allDevices); 433 } 434 if (result.isEmpty()) { 435 result.append(entry->mString); 436 } 437 return result; 438} 439 440String8 inputFlagsToString(audio_input_flags_t flags) 441{ 442 static const struct mapping { 443 audio_input_flags_t mFlag; 444 const char * mString; 445 } mappings[] = { 446 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 447 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 448 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 449 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 450 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 451 }; 452 String8 result; 453 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 454 const mapping *entry; 455 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 456 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 457 if (flags & entry->mFlag) { 458 if (!result.isEmpty()) { 459 result.append("|"); 460 } 461 result.append(entry->mString); 462 } 463 } 464 if (flags & ~allFlags) { 465 if (!result.isEmpty()) { 466 result.append("|"); 467 } 468 result.appendFormat("0x%X", flags & ~allFlags); 469 } 470 if (result.isEmpty()) { 471 result.append(entry->mString); 472 } 473 return result; 474} 475 476String8 outputFlagsToString(audio_output_flags_t flags) 477{ 478 static const struct mapping { 479 audio_output_flags_t mFlag; 480 const char * mString; 481 } mappings[] = { 482 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 483 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 484 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 485 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 486 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 487 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 488 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 489 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 490 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 491 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 492 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 493 }; 494 String8 result; 495 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 496 const mapping *entry; 497 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 498 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 499 if (flags & entry->mFlag) { 500 if (!result.isEmpty()) { 501 result.append("|"); 502 } 503 result.append(entry->mString); 504 } 505 } 506 if (flags & ~allFlags) { 507 if (!result.isEmpty()) { 508 result.append("|"); 509 } 510 result.appendFormat("0x%X", flags & ~allFlags); 511 } 512 if (result.isEmpty()) { 513 result.append(entry->mString); 514 } 515 return result; 516} 517 518const char *sourceToString(audio_source_t source) 519{ 520 switch (source) { 521 case AUDIO_SOURCE_DEFAULT: return "default"; 522 case AUDIO_SOURCE_MIC: return "mic"; 523 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 524 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 525 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 526 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 527 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 528 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 529 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 530 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 531 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 532 case AUDIO_SOURCE_HOTWORD: return "hotword"; 533 default: return "unknown"; 534 } 535} 536 537AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 538 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 539 : Thread(false /*canCallJava*/), 540 mType(type), 541 mAudioFlinger(audioFlinger), 542 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 543 // are set by PlaybackThread::readOutputParameters_l() or 544 // RecordThread::readInputParameters_l() 545 //FIXME: mStandby should be true here. Is this some kind of hack? 546 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 547 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 548 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 549 // mName will be set by concrete (non-virtual) subclass 550 mDeathRecipient(new PMDeathRecipient(this)), 551 mSystemReady(systemReady), 552 mNotifiedBatteryStart(false) 553{ 554 memset(&mPatch, 0, sizeof(struct audio_patch)); 555} 556 557AudioFlinger::ThreadBase::~ThreadBase() 558{ 559 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 560 mConfigEvents.clear(); 561 562 // do not lock the mutex in destructor 563 releaseWakeLock_l(); 564 if (mPowerManager != 0) { 565 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 566 binder->unlinkToDeath(mDeathRecipient); 567 } 568} 569 570status_t AudioFlinger::ThreadBase::readyToRun() 571{ 572 status_t status = initCheck(); 573 if (status == NO_ERROR) { 574 ALOGI("AudioFlinger's thread %p ready to run", this); 575 } else { 576 ALOGE("No working audio driver found."); 577 } 578 return status; 579} 580 581void AudioFlinger::ThreadBase::exit() 582{ 583 ALOGV("ThreadBase::exit"); 584 // do any cleanup required for exit to succeed 585 preExit(); 586 { 587 // This lock prevents the following race in thread (uniprocessor for illustration): 588 // if (!exitPending()) { 589 // // context switch from here to exit() 590 // // exit() calls requestExit(), what exitPending() observes 591 // // exit() calls signal(), which is dropped since no waiters 592 // // context switch back from exit() to here 593 // mWaitWorkCV.wait(...); 594 // // now thread is hung 595 // } 596 AutoMutex lock(mLock); 597 requestExit(); 598 mWaitWorkCV.broadcast(); 599 } 600 // When Thread::requestExitAndWait is made virtual and this method is renamed to 601 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 602 requestExitAndWait(); 603} 604 605status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 606{ 607 status_t status; 608 609 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 610 Mutex::Autolock _l(mLock); 611 612 return sendSetParameterConfigEvent_l(keyValuePairs); 613} 614 615// sendConfigEvent_l() must be called with ThreadBase::mLock held 616// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 617status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 618{ 619 status_t status = NO_ERROR; 620 621 if (event->mRequiresSystemReady && !mSystemReady) { 622 event->mWaitStatus = false; 623 mPendingConfigEvents.add(event); 624 return status; 625 } 626 mConfigEvents.add(event); 627 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 628 mWaitWorkCV.signal(); 629 mLock.unlock(); 630 { 631 Mutex::Autolock _l(event->mLock); 632 while (event->mWaitStatus) { 633 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 634 event->mStatus = TIMED_OUT; 635 event->mWaitStatus = false; 636 } 637 } 638 status = event->mStatus; 639 } 640 mLock.lock(); 641 return status; 642} 643 644void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 645{ 646 Mutex::Autolock _l(mLock); 647 sendIoConfigEvent_l(event, pid); 648} 649 650// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 651void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 652{ 653 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 654 sendConfigEvent_l(configEvent); 655} 656 657void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 658{ 659 Mutex::Autolock _l(mLock); 660 sendPrioConfigEvent_l(pid, tid, prio); 661} 662 663// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 664void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 665{ 666 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 667 sendConfigEvent_l(configEvent); 668} 669 670// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 671status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 672{ 673 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 674 return sendConfigEvent_l(configEvent); 675} 676 677status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 678 const struct audio_patch *patch, 679 audio_patch_handle_t *handle) 680{ 681 Mutex::Autolock _l(mLock); 682 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 683 status_t status = sendConfigEvent_l(configEvent); 684 if (status == NO_ERROR) { 685 CreateAudioPatchConfigEventData *data = 686 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 687 *handle = data->mHandle; 688 } 689 return status; 690} 691 692status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 693 const audio_patch_handle_t handle) 694{ 695 Mutex::Autolock _l(mLock); 696 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 697 return sendConfigEvent_l(configEvent); 698} 699 700 701// post condition: mConfigEvents.isEmpty() 702void AudioFlinger::ThreadBase::processConfigEvents_l() 703{ 704 bool configChanged = false; 705 706 while (!mConfigEvents.isEmpty()) { 707 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 708 sp<ConfigEvent> event = mConfigEvents[0]; 709 mConfigEvents.removeAt(0); 710 switch (event->mType) { 711 case CFG_EVENT_PRIO: { 712 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 713 // FIXME Need to understand why this has to be done asynchronously 714 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 715 true /*asynchronous*/); 716 if (err != 0) { 717 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 718 data->mPrio, data->mPid, data->mTid, err); 719 } 720 } break; 721 case CFG_EVENT_IO: { 722 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 723 ioConfigChanged(data->mEvent, data->mPid); 724 } break; 725 case CFG_EVENT_SET_PARAMETER: { 726 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 727 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 728 configChanged = true; 729 } 730 } break; 731 case CFG_EVENT_CREATE_AUDIO_PATCH: { 732 CreateAudioPatchConfigEventData *data = 733 (CreateAudioPatchConfigEventData *)event->mData.get(); 734 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 735 } break; 736 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 737 ReleaseAudioPatchConfigEventData *data = 738 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 739 event->mStatus = releaseAudioPatch_l(data->mHandle); 740 } break; 741 default: 742 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 743 break; 744 } 745 { 746 Mutex::Autolock _l(event->mLock); 747 if (event->mWaitStatus) { 748 event->mWaitStatus = false; 749 event->mCond.signal(); 750 } 751 } 752 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 753 } 754 755 if (configChanged) { 756 cacheParameters_l(); 757 } 758} 759 760String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 761 String8 s; 762 const audio_channel_representation_t representation = 763 audio_channel_mask_get_representation(mask); 764 765 switch (representation) { 766 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 767 if (output) { 768 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 769 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 770 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 771 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 772 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 773 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 774 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 775 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 776 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 777 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 778 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 779 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 780 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 781 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 782 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 783 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 784 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 785 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 786 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 787 } else { 788 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 789 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 790 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 791 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 792 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 793 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 794 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 795 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 796 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 797 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 798 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 799 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 800 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 801 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 802 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 803 } 804 const int len = s.length(); 805 if (len > 2) { 806 char *str = s.lockBuffer(len); // needed? 807 s.unlockBuffer(len - 2); // remove trailing ", " 808 } 809 return s; 810 } 811 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 812 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 813 return s; 814 default: 815 s.appendFormat("unknown mask, representation:%d bits:%#x", 816 representation, audio_channel_mask_get_bits(mask)); 817 return s; 818 } 819} 820 821void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 822{ 823 const size_t SIZE = 256; 824 char buffer[SIZE]; 825 String8 result; 826 827 bool locked = AudioFlinger::dumpTryLock(mLock); 828 if (!locked) { 829 dprintf(fd, "thread %p may be deadlocked\n", this); 830 } 831 832 dprintf(fd, " Thread name: %s\n", mThreadName); 833 dprintf(fd, " I/O handle: %d\n", mId); 834 dprintf(fd, " TID: %d\n", getTid()); 835 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 836 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 837 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 838 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 839 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 840 dprintf(fd, " Channel count: %u\n", mChannelCount); 841 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 842 channelMaskToString(mChannelMask, mType != RECORD).string()); 843 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 844 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 845 dprintf(fd, " Pending config events:"); 846 size_t numConfig = mConfigEvents.size(); 847 if (numConfig) { 848 for (size_t i = 0; i < numConfig; i++) { 849 mConfigEvents[i]->dump(buffer, SIZE); 850 dprintf(fd, "\n %s", buffer); 851 } 852 dprintf(fd, "\n"); 853 } else { 854 dprintf(fd, " none\n"); 855 } 856 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 857 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 858 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 859 860 if (locked) { 861 mLock.unlock(); 862 } 863} 864 865void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 866{ 867 const size_t SIZE = 256; 868 char buffer[SIZE]; 869 String8 result; 870 871 size_t numEffectChains = mEffectChains.size(); 872 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 873 write(fd, buffer, strlen(buffer)); 874 875 for (size_t i = 0; i < numEffectChains; ++i) { 876 sp<EffectChain> chain = mEffectChains[i]; 877 if (chain != 0) { 878 chain->dump(fd, args); 879 } 880 } 881} 882 883void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 884{ 885 Mutex::Autolock _l(mLock); 886 acquireWakeLock_l(uid); 887} 888 889String16 AudioFlinger::ThreadBase::getWakeLockTag() 890{ 891 switch (mType) { 892 case MIXER: 893 return String16("AudioMix"); 894 case DIRECT: 895 return String16("AudioDirectOut"); 896 case DUPLICATING: 897 return String16("AudioDup"); 898 case RECORD: 899 return String16("AudioIn"); 900 case OFFLOAD: 901 return String16("AudioOffload"); 902 default: 903 ALOG_ASSERT(false); 904 return String16("AudioUnknown"); 905 } 906} 907 908void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 909{ 910 getPowerManager_l(); 911 if (mPowerManager != 0) { 912 sp<IBinder> binder = new BBinder(); 913 status_t status; 914 if (uid >= 0) { 915 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 916 binder, 917 getWakeLockTag(), 918 String16("audioserver"), 919 uid, 920 true /* FIXME force oneway contrary to .aidl */); 921 } else { 922 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 923 binder, 924 getWakeLockTag(), 925 String16("audioserver"), 926 true /* FIXME force oneway contrary to .aidl */); 927 } 928 if (status == NO_ERROR) { 929 mWakeLockToken = binder; 930 } 931 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 932 } 933 934 if (!mNotifiedBatteryStart) { 935 BatteryNotifier::getInstance().noteStartAudio(); 936 mNotifiedBatteryStart = true; 937 } 938} 939 940void AudioFlinger::ThreadBase::releaseWakeLock() 941{ 942 Mutex::Autolock _l(mLock); 943 releaseWakeLock_l(); 944} 945 946void AudioFlinger::ThreadBase::releaseWakeLock_l() 947{ 948 if (mWakeLockToken != 0) { 949 ALOGV("releaseWakeLock_l() %s", mThreadName); 950 if (mPowerManager != 0) { 951 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 952 true /* FIXME force oneway contrary to .aidl */); 953 } 954 mWakeLockToken.clear(); 955 } 956 957 if (mNotifiedBatteryStart) { 958 BatteryNotifier::getInstance().noteStopAudio(); 959 mNotifiedBatteryStart = false; 960 } 961} 962 963void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 964 Mutex::Autolock _l(mLock); 965 updateWakeLockUids_l(uids); 966} 967 968void AudioFlinger::ThreadBase::getPowerManager_l() { 969 if (mSystemReady && mPowerManager == 0) { 970 // use checkService() to avoid blocking if power service is not up yet 971 sp<IBinder> binder = 972 defaultServiceManager()->checkService(String16("power")); 973 if (binder == 0) { 974 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 975 } else { 976 mPowerManager = interface_cast<IPowerManager>(binder); 977 binder->linkToDeath(mDeathRecipient); 978 } 979 } 980} 981 982void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 983 getPowerManager_l(); 984 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 985 if (mSystemReady) { 986 ALOGE("no wake lock to update, but system ready!"); 987 } else { 988 ALOGW("no wake lock to update, system not ready yet"); 989 } 990 return; 991 } 992 if (mPowerManager != 0) { 993 sp<IBinder> binder = new BBinder(); 994 status_t status; 995 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 996 true /* FIXME force oneway contrary to .aidl */); 997 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 998 } 999} 1000 1001void AudioFlinger::ThreadBase::clearPowerManager() 1002{ 1003 Mutex::Autolock _l(mLock); 1004 releaseWakeLock_l(); 1005 mPowerManager.clear(); 1006} 1007 1008void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1009{ 1010 sp<ThreadBase> thread = mThread.promote(); 1011 if (thread != 0) { 1012 thread->clearPowerManager(); 1013 } 1014 ALOGW("power manager service died !!!"); 1015} 1016 1017void AudioFlinger::ThreadBase::setEffectSuspended( 1018 const effect_uuid_t *type, bool suspend, int sessionId) 1019{ 1020 Mutex::Autolock _l(mLock); 1021 setEffectSuspended_l(type, suspend, sessionId); 1022} 1023 1024void AudioFlinger::ThreadBase::setEffectSuspended_l( 1025 const effect_uuid_t *type, bool suspend, int sessionId) 1026{ 1027 sp<EffectChain> chain = getEffectChain_l(sessionId); 1028 if (chain != 0) { 1029 if (type != NULL) { 1030 chain->setEffectSuspended_l(type, suspend); 1031 } else { 1032 chain->setEffectSuspendedAll_l(suspend); 1033 } 1034 } 1035 1036 updateSuspendedSessions_l(type, suspend, sessionId); 1037} 1038 1039void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1040{ 1041 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1042 if (index < 0) { 1043 return; 1044 } 1045 1046 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1047 mSuspendedSessions.valueAt(index); 1048 1049 for (size_t i = 0; i < sessionEffects.size(); i++) { 1050 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1051 for (int j = 0; j < desc->mRefCount; j++) { 1052 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1053 chain->setEffectSuspendedAll_l(true); 1054 } else { 1055 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1056 desc->mType.timeLow); 1057 chain->setEffectSuspended_l(&desc->mType, true); 1058 } 1059 } 1060 } 1061} 1062 1063void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1064 bool suspend, 1065 int sessionId) 1066{ 1067 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1068 1069 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1070 1071 if (suspend) { 1072 if (index >= 0) { 1073 sessionEffects = mSuspendedSessions.valueAt(index); 1074 } else { 1075 mSuspendedSessions.add(sessionId, sessionEffects); 1076 } 1077 } else { 1078 if (index < 0) { 1079 return; 1080 } 1081 sessionEffects = mSuspendedSessions.valueAt(index); 1082 } 1083 1084 1085 int key = EffectChain::kKeyForSuspendAll; 1086 if (type != NULL) { 1087 key = type->timeLow; 1088 } 1089 index = sessionEffects.indexOfKey(key); 1090 1091 sp<SuspendedSessionDesc> desc; 1092 if (suspend) { 1093 if (index >= 0) { 1094 desc = sessionEffects.valueAt(index); 1095 } else { 1096 desc = new SuspendedSessionDesc(); 1097 if (type != NULL) { 1098 desc->mType = *type; 1099 } 1100 sessionEffects.add(key, desc); 1101 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1102 } 1103 desc->mRefCount++; 1104 } else { 1105 if (index < 0) { 1106 return; 1107 } 1108 desc = sessionEffects.valueAt(index); 1109 if (--desc->mRefCount == 0) { 1110 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1111 sessionEffects.removeItemsAt(index); 1112 if (sessionEffects.isEmpty()) { 1113 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1114 sessionId); 1115 mSuspendedSessions.removeItem(sessionId); 1116 } 1117 } 1118 } 1119 if (!sessionEffects.isEmpty()) { 1120 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1121 } 1122} 1123 1124void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1125 bool enabled, 1126 int sessionId) 1127{ 1128 Mutex::Autolock _l(mLock); 1129 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1130} 1131 1132void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1133 bool enabled, 1134 int sessionId) 1135{ 1136 if (mType != RECORD) { 1137 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1138 // another session. This gives the priority to well behaved effect control panels 1139 // and applications not using global effects. 1140 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1141 // global effects 1142 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1143 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1144 } 1145 } 1146 1147 sp<EffectChain> chain = getEffectChain_l(sessionId); 1148 if (chain != 0) { 1149 chain->checkSuspendOnEffectEnabled(effect, enabled); 1150 } 1151} 1152 1153// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1154sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1155 const sp<AudioFlinger::Client>& client, 1156 const sp<IEffectClient>& effectClient, 1157 int32_t priority, 1158 int sessionId, 1159 effect_descriptor_t *desc, 1160 int *enabled, 1161 status_t *status) 1162{ 1163 sp<EffectModule> effect; 1164 sp<EffectHandle> handle; 1165 status_t lStatus; 1166 sp<EffectChain> chain; 1167 bool chainCreated = false; 1168 bool effectCreated = false; 1169 bool effectRegistered = false; 1170 1171 lStatus = initCheck(); 1172 if (lStatus != NO_ERROR) { 1173 ALOGW("createEffect_l() Audio driver not initialized."); 1174 goto Exit; 1175 } 1176 1177 // Reject any effect on Direct output threads for now, since the format of 1178 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1179 if (mType == DIRECT) { 1180 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1181 desc->name, mThreadName); 1182 lStatus = BAD_VALUE; 1183 goto Exit; 1184 } 1185 1186 // Reject any effect on mixer or duplicating multichannel sinks. 1187 // TODO: fix both format and multichannel issues with effects. 1188 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1189 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1190 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1191 lStatus = BAD_VALUE; 1192 goto Exit; 1193 } 1194 1195 // Allow global effects only on offloaded and mixer threads 1196 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1197 switch (mType) { 1198 case MIXER: 1199 case OFFLOAD: 1200 break; 1201 case DIRECT: 1202 case DUPLICATING: 1203 case RECORD: 1204 default: 1205 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1206 desc->name, mThreadName); 1207 lStatus = BAD_VALUE; 1208 goto Exit; 1209 } 1210 } 1211 1212 // Only Pre processor effects are allowed on input threads and only on input threads 1213 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1214 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1215 desc->name, desc->flags, mType); 1216 lStatus = BAD_VALUE; 1217 goto Exit; 1218 } 1219 1220 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1221 1222 { // scope for mLock 1223 Mutex::Autolock _l(mLock); 1224 1225 // check for existing effect chain with the requested audio session 1226 chain = getEffectChain_l(sessionId); 1227 if (chain == 0) { 1228 // create a new chain for this session 1229 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1230 chain = new EffectChain(this, sessionId); 1231 addEffectChain_l(chain); 1232 chain->setStrategy(getStrategyForSession_l(sessionId)); 1233 chainCreated = true; 1234 } else { 1235 effect = chain->getEffectFromDesc_l(desc); 1236 } 1237 1238 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1239 1240 if (effect == 0) { 1241 int id = mAudioFlinger->nextUniqueId(); 1242 // Check CPU and memory usage 1243 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1244 if (lStatus != NO_ERROR) { 1245 goto Exit; 1246 } 1247 effectRegistered = true; 1248 // create a new effect module if none present in the chain 1249 effect = new EffectModule(this, chain, desc, id, sessionId); 1250 lStatus = effect->status(); 1251 if (lStatus != NO_ERROR) { 1252 goto Exit; 1253 } 1254 effect->setOffloaded(mType == OFFLOAD, mId); 1255 1256 lStatus = chain->addEffect_l(effect); 1257 if (lStatus != NO_ERROR) { 1258 goto Exit; 1259 } 1260 effectCreated = true; 1261 1262 effect->setDevice(mOutDevice); 1263 effect->setDevice(mInDevice); 1264 effect->setMode(mAudioFlinger->getMode()); 1265 effect->setAudioSource(mAudioSource); 1266 } 1267 // create effect handle and connect it to effect module 1268 handle = new EffectHandle(effect, client, effectClient, priority); 1269 lStatus = handle->initCheck(); 1270 if (lStatus == OK) { 1271 lStatus = effect->addHandle(handle.get()); 1272 } 1273 if (enabled != NULL) { 1274 *enabled = (int)effect->isEnabled(); 1275 } 1276 } 1277 1278Exit: 1279 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1280 Mutex::Autolock _l(mLock); 1281 if (effectCreated) { 1282 chain->removeEffect_l(effect); 1283 } 1284 if (effectRegistered) { 1285 AudioSystem::unregisterEffect(effect->id()); 1286 } 1287 if (chainCreated) { 1288 removeEffectChain_l(chain); 1289 } 1290 handle.clear(); 1291 } 1292 1293 *status = lStatus; 1294 return handle; 1295} 1296 1297sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1298{ 1299 Mutex::Autolock _l(mLock); 1300 return getEffect_l(sessionId, effectId); 1301} 1302 1303sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1304{ 1305 sp<EffectChain> chain = getEffectChain_l(sessionId); 1306 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1307} 1308 1309// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1310// PlaybackThread::mLock held 1311status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1312{ 1313 // check for existing effect chain with the requested audio session 1314 int sessionId = effect->sessionId(); 1315 sp<EffectChain> chain = getEffectChain_l(sessionId); 1316 bool chainCreated = false; 1317 1318 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1319 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1320 this, effect->desc().name, effect->desc().flags); 1321 1322 if (chain == 0) { 1323 // create a new chain for this session 1324 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1325 chain = new EffectChain(this, sessionId); 1326 addEffectChain_l(chain); 1327 chain->setStrategy(getStrategyForSession_l(sessionId)); 1328 chainCreated = true; 1329 } 1330 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1331 1332 if (chain->getEffectFromId_l(effect->id()) != 0) { 1333 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1334 this, effect->desc().name, chain.get()); 1335 return BAD_VALUE; 1336 } 1337 1338 effect->setOffloaded(mType == OFFLOAD, mId); 1339 1340 status_t status = chain->addEffect_l(effect); 1341 if (status != NO_ERROR) { 1342 if (chainCreated) { 1343 removeEffectChain_l(chain); 1344 } 1345 return status; 1346 } 1347 1348 effect->setDevice(mOutDevice); 1349 effect->setDevice(mInDevice); 1350 effect->setMode(mAudioFlinger->getMode()); 1351 effect->setAudioSource(mAudioSource); 1352 return NO_ERROR; 1353} 1354 1355void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1356 1357 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1358 effect_descriptor_t desc = effect->desc(); 1359 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1360 detachAuxEffect_l(effect->id()); 1361 } 1362 1363 sp<EffectChain> chain = effect->chain().promote(); 1364 if (chain != 0) { 1365 // remove effect chain if removing last effect 1366 if (chain->removeEffect_l(effect) == 0) { 1367 removeEffectChain_l(chain); 1368 } 1369 } else { 1370 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1371 } 1372} 1373 1374void AudioFlinger::ThreadBase::lockEffectChains_l( 1375 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1376{ 1377 effectChains = mEffectChains; 1378 for (size_t i = 0; i < mEffectChains.size(); i++) { 1379 mEffectChains[i]->lock(); 1380 } 1381} 1382 1383void AudioFlinger::ThreadBase::unlockEffectChains( 1384 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1385{ 1386 for (size_t i = 0; i < effectChains.size(); i++) { 1387 effectChains[i]->unlock(); 1388 } 1389} 1390 1391sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1392{ 1393 Mutex::Autolock _l(mLock); 1394 return getEffectChain_l(sessionId); 1395} 1396 1397sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1398{ 1399 size_t size = mEffectChains.size(); 1400 for (size_t i = 0; i < size; i++) { 1401 if (mEffectChains[i]->sessionId() == sessionId) { 1402 return mEffectChains[i]; 1403 } 1404 } 1405 return 0; 1406} 1407 1408void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1409{ 1410 Mutex::Autolock _l(mLock); 1411 size_t size = mEffectChains.size(); 1412 for (size_t i = 0; i < size; i++) { 1413 mEffectChains[i]->setMode_l(mode); 1414 } 1415} 1416 1417void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1418{ 1419 config->type = AUDIO_PORT_TYPE_MIX; 1420 config->ext.mix.handle = mId; 1421 config->sample_rate = mSampleRate; 1422 config->format = mFormat; 1423 config->channel_mask = mChannelMask; 1424 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1425 AUDIO_PORT_CONFIG_FORMAT; 1426} 1427 1428void AudioFlinger::ThreadBase::systemReady() 1429{ 1430 Mutex::Autolock _l(mLock); 1431 if (mSystemReady) { 1432 return; 1433 } 1434 mSystemReady = true; 1435 1436 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1437 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1438 } 1439 mPendingConfigEvents.clear(); 1440} 1441 1442 1443// ---------------------------------------------------------------------------- 1444// Playback 1445// ---------------------------------------------------------------------------- 1446 1447AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1448 AudioStreamOut* output, 1449 audio_io_handle_t id, 1450 audio_devices_t device, 1451 type_t type, 1452 bool systemReady) 1453 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1454 mNormalFrameCount(0), mSinkBuffer(NULL), 1455 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1456 mMixerBuffer(NULL), 1457 mMixerBufferSize(0), 1458 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1459 mMixerBufferValid(false), 1460 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1461 mEffectBuffer(NULL), 1462 mEffectBufferSize(0), 1463 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1464 mEffectBufferValid(false), 1465 mSuspended(0), mBytesWritten(0), 1466 mActiveTracksGeneration(0), 1467 // mStreamTypes[] initialized in constructor body 1468 mOutput(output), 1469 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1470 mMixerStatus(MIXER_IDLE), 1471 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1472 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1473 mBytesRemaining(0), 1474 mCurrentWriteLength(0), 1475 mUseAsyncWrite(false), 1476 mWriteAckSequence(0), 1477 mDrainSequence(0), 1478 mSignalPending(false), 1479 mScreenState(AudioFlinger::mScreenState), 1480 // index 0 is reserved for normal mixer's submix 1481 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1482 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1483 // mLatchD, mLatchQ, 1484 mLatchDValid(false), mLatchQValid(false) 1485{ 1486 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1487 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1488 1489 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1490 // it would be safer to explicitly pass initial masterVolume/masterMute as 1491 // parameter. 1492 // 1493 // If the HAL we are using has support for master volume or master mute, 1494 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1495 // and the mute set to false). 1496 mMasterVolume = audioFlinger->masterVolume_l(); 1497 mMasterMute = audioFlinger->masterMute_l(); 1498 if (mOutput && mOutput->audioHwDev) { 1499 if (mOutput->audioHwDev->canSetMasterVolume()) { 1500 mMasterVolume = 1.0; 1501 } 1502 1503 if (mOutput->audioHwDev->canSetMasterMute()) { 1504 mMasterMute = false; 1505 } 1506 } 1507 1508 readOutputParameters_l(); 1509 1510 // ++ operator does not compile 1511 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1512 stream = (audio_stream_type_t) (stream + 1)) { 1513 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1514 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1515 } 1516} 1517 1518AudioFlinger::PlaybackThread::~PlaybackThread() 1519{ 1520 mAudioFlinger->unregisterWriter(mNBLogWriter); 1521 free(mSinkBuffer); 1522 free(mMixerBuffer); 1523 free(mEffectBuffer); 1524} 1525 1526void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1527{ 1528 dumpInternals(fd, args); 1529 dumpTracks(fd, args); 1530 dumpEffectChains(fd, args); 1531} 1532 1533void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1534{ 1535 const size_t SIZE = 256; 1536 char buffer[SIZE]; 1537 String8 result; 1538 1539 result.appendFormat(" Stream volumes in dB: "); 1540 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1541 const stream_type_t *st = &mStreamTypes[i]; 1542 if (i > 0) { 1543 result.appendFormat(", "); 1544 } 1545 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1546 if (st->mute) { 1547 result.append("M"); 1548 } 1549 } 1550 result.append("\n"); 1551 write(fd, result.string(), result.length()); 1552 result.clear(); 1553 1554 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1555 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1556 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1557 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1558 1559 size_t numtracks = mTracks.size(); 1560 size_t numactive = mActiveTracks.size(); 1561 dprintf(fd, " %d Tracks", numtracks); 1562 size_t numactiveseen = 0; 1563 if (numtracks) { 1564 dprintf(fd, " of which %d are active\n", numactive); 1565 Track::appendDumpHeader(result); 1566 for (size_t i = 0; i < numtracks; ++i) { 1567 sp<Track> track = mTracks[i]; 1568 if (track != 0) { 1569 bool active = mActiveTracks.indexOf(track) >= 0; 1570 if (active) { 1571 numactiveseen++; 1572 } 1573 track->dump(buffer, SIZE, active); 1574 result.append(buffer); 1575 } 1576 } 1577 } else { 1578 result.append("\n"); 1579 } 1580 if (numactiveseen != numactive) { 1581 // some tracks in the active list were not in the tracks list 1582 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1583 " not in the track list\n"); 1584 result.append(buffer); 1585 Track::appendDumpHeader(result); 1586 for (size_t i = 0; i < numactive; ++i) { 1587 sp<Track> track = mActiveTracks[i].promote(); 1588 if (track != 0 && mTracks.indexOf(track) < 0) { 1589 track->dump(buffer, SIZE, true); 1590 result.append(buffer); 1591 } 1592 } 1593 } 1594 1595 write(fd, result.string(), result.size()); 1596} 1597 1598void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1599{ 1600 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1601 1602 dumpBase(fd, args); 1603 1604 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1605 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1606 dprintf(fd, " Total writes: %d\n", mNumWrites); 1607 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1608 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1609 dprintf(fd, " Suspend count: %d\n", mSuspended); 1610 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1611 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1612 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1613 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1614 AudioStreamOut *output = mOutput; 1615 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1616 String8 flagsAsString = outputFlagsToString(flags); 1617 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1618} 1619 1620// Thread virtuals 1621 1622void AudioFlinger::PlaybackThread::onFirstRef() 1623{ 1624 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1625} 1626 1627// ThreadBase virtuals 1628void AudioFlinger::PlaybackThread::preExit() 1629{ 1630 ALOGV(" preExit()"); 1631 // FIXME this is using hard-coded strings but in the future, this functionality will be 1632 // converted to use audio HAL extensions required to support tunneling 1633 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1634} 1635 1636// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1637sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1638 const sp<AudioFlinger::Client>& client, 1639 audio_stream_type_t streamType, 1640 uint32_t sampleRate, 1641 audio_format_t format, 1642 audio_channel_mask_t channelMask, 1643 size_t *pFrameCount, 1644 const sp<IMemory>& sharedBuffer, 1645 int sessionId, 1646 IAudioFlinger::track_flags_t *flags, 1647 pid_t tid, 1648 int uid, 1649 status_t *status) 1650{ 1651 size_t frameCount = *pFrameCount; 1652 sp<Track> track; 1653 status_t lStatus; 1654 1655 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1656 1657 // client expresses a preference for FAST, but we get the final say 1658 if (*flags & IAudioFlinger::TRACK_FAST) { 1659 if ( 1660 // not timed 1661 (!isTimed) && 1662 // either of these use cases: 1663 ( 1664 // use case 1: shared buffer with any frame count 1665 ( 1666 (sharedBuffer != 0) 1667 ) || 1668 // use case 2: frame count is default or at least as large as HAL 1669 ( 1670 // we formerly checked for a callback handler (non-0 tid), 1671 // but that is no longer required for TRANSFER_OBTAIN mode 1672 ((frameCount == 0) || 1673 (frameCount >= mFrameCount)) 1674 ) 1675 ) && 1676 // PCM data 1677 audio_is_linear_pcm(format) && 1678 // TODO: extract as a data library function that checks that a computationally 1679 // expensive downmixer is not required: isFastOutputChannelConversion() 1680 (channelMask == mChannelMask || 1681 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1682 (channelMask == AUDIO_CHANNEL_OUT_MONO 1683 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1684 // hardware sample rate 1685 (sampleRate == mSampleRate) && 1686 // normal mixer has an associated fast mixer 1687 hasFastMixer() && 1688 // there are sufficient fast track slots available 1689 (mFastTrackAvailMask != 0) 1690 // FIXME test that MixerThread for this fast track has a capable output HAL 1691 // FIXME add a permission test also? 1692 ) { 1693 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1694 if (frameCount == 0) { 1695 // read the fast track multiplier property the first time it is needed 1696 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1697 if (ok != 0) { 1698 ALOGE("%s pthread_once failed: %d", __func__, ok); 1699 } 1700 frameCount = mFrameCount * sFastTrackMultiplier; 1701 } 1702 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1703 frameCount, mFrameCount); 1704 } else { 1705 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1706 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1707 "sampleRate=%u mSampleRate=%u " 1708 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1709 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1710 audio_is_linear_pcm(format), 1711 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1712 *flags &= ~IAudioFlinger::TRACK_FAST; 1713 } 1714 } 1715 // For normal PCM streaming tracks, update minimum frame count. 1716 // For compatibility with AudioTrack calculation, buffer depth is forced 1717 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1718 // This is probably too conservative, but legacy application code may depend on it. 1719 // If you change this calculation, also review the start threshold which is related. 1720 if (!(*flags & IAudioFlinger::TRACK_FAST) 1721 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1722 // this must match AudioTrack.cpp calculateMinFrameCount(). 1723 // TODO: Move to a common library 1724 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1725 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1726 if (minBufCount < 2) { 1727 minBufCount = 2; 1728 } 1729 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1730 // or the client should compute and pass in a larger buffer request. 1731 size_t minFrameCount = 1732 minBufCount * sourceFramesNeededWithTimestretch( 1733 sampleRate, mNormalFrameCount, 1734 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1735 if (frameCount < minFrameCount) { // including frameCount == 0 1736 frameCount = minFrameCount; 1737 } 1738 } 1739 *pFrameCount = frameCount; 1740 1741 switch (mType) { 1742 1743 case DIRECT: 1744 if (audio_is_linear_pcm(format)) { 1745 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1746 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1747 "for output %p with format %#x", 1748 sampleRate, format, channelMask, mOutput, mFormat); 1749 lStatus = BAD_VALUE; 1750 goto Exit; 1751 } 1752 } 1753 break; 1754 1755 case OFFLOAD: 1756 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1757 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1758 "for output %p with format %#x", 1759 sampleRate, format, channelMask, mOutput, mFormat); 1760 lStatus = BAD_VALUE; 1761 goto Exit; 1762 } 1763 break; 1764 1765 default: 1766 if (!audio_is_linear_pcm(format)) { 1767 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1768 "for output %p with format %#x", 1769 format, mOutput, mFormat); 1770 lStatus = BAD_VALUE; 1771 goto Exit; 1772 } 1773 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1774 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1775 lStatus = BAD_VALUE; 1776 goto Exit; 1777 } 1778 break; 1779 1780 } 1781 1782 lStatus = initCheck(); 1783 if (lStatus != NO_ERROR) { 1784 ALOGE("createTrack_l() audio driver not initialized"); 1785 goto Exit; 1786 } 1787 1788 { // scope for mLock 1789 Mutex::Autolock _l(mLock); 1790 1791 // all tracks in same audio session must share the same routing strategy otherwise 1792 // conflicts will happen when tracks are moved from one output to another by audio policy 1793 // manager 1794 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1795 for (size_t i = 0; i < mTracks.size(); ++i) { 1796 sp<Track> t = mTracks[i]; 1797 if (t != 0 && t->isExternalTrack()) { 1798 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1799 if (sessionId == t->sessionId() && strategy != actual) { 1800 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1801 strategy, actual); 1802 lStatus = BAD_VALUE; 1803 goto Exit; 1804 } 1805 } 1806 } 1807 1808 if (!isTimed) { 1809 track = new Track(this, client, streamType, sampleRate, format, 1810 channelMask, frameCount, NULL, sharedBuffer, 1811 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1812 } else { 1813 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1814 channelMask, frameCount, sharedBuffer, sessionId, uid); 1815 } 1816 1817 // new Track always returns non-NULL, 1818 // but TimedTrack::create() is a factory that could fail by returning NULL 1819 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1820 if (lStatus != NO_ERROR) { 1821 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1822 // track must be cleared from the caller as the caller has the AF lock 1823 goto Exit; 1824 } 1825 mTracks.add(track); 1826 1827 sp<EffectChain> chain = getEffectChain_l(sessionId); 1828 if (chain != 0) { 1829 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1830 track->setMainBuffer(chain->inBuffer()); 1831 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1832 chain->incTrackCnt(); 1833 } 1834 1835 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1836 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1837 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1838 // so ask activity manager to do this on our behalf 1839 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1840 } 1841 } 1842 1843 lStatus = NO_ERROR; 1844 1845Exit: 1846 *status = lStatus; 1847 return track; 1848} 1849 1850uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1851{ 1852 return latency; 1853} 1854 1855uint32_t AudioFlinger::PlaybackThread::latency() const 1856{ 1857 Mutex::Autolock _l(mLock); 1858 return latency_l(); 1859} 1860uint32_t AudioFlinger::PlaybackThread::latency_l() const 1861{ 1862 if (initCheck() == NO_ERROR) { 1863 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1864 } else { 1865 return 0; 1866 } 1867} 1868 1869void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1870{ 1871 Mutex::Autolock _l(mLock); 1872 // Don't apply master volume in SW if our HAL can do it for us. 1873 if (mOutput && mOutput->audioHwDev && 1874 mOutput->audioHwDev->canSetMasterVolume()) { 1875 mMasterVolume = 1.0; 1876 } else { 1877 mMasterVolume = value; 1878 } 1879} 1880 1881void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1882{ 1883 Mutex::Autolock _l(mLock); 1884 // Don't apply master mute in SW if our HAL can do it for us. 1885 if (mOutput && mOutput->audioHwDev && 1886 mOutput->audioHwDev->canSetMasterMute()) { 1887 mMasterMute = false; 1888 } else { 1889 mMasterMute = muted; 1890 } 1891} 1892 1893void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1894{ 1895 Mutex::Autolock _l(mLock); 1896 mStreamTypes[stream].volume = value; 1897 broadcast_l(); 1898} 1899 1900void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1901{ 1902 Mutex::Autolock _l(mLock); 1903 mStreamTypes[stream].mute = muted; 1904 broadcast_l(); 1905} 1906 1907float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1908{ 1909 Mutex::Autolock _l(mLock); 1910 return mStreamTypes[stream].volume; 1911} 1912 1913// addTrack_l() must be called with ThreadBase::mLock held 1914status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1915{ 1916 status_t status = ALREADY_EXISTS; 1917 1918 // set retry count for buffer fill 1919 track->mRetryCount = kMaxTrackStartupRetries; 1920 if (mActiveTracks.indexOf(track) < 0) { 1921 // the track is newly added, make sure it fills up all its 1922 // buffers before playing. This is to ensure the client will 1923 // effectively get the latency it requested. 1924 if (track->isExternalTrack()) { 1925 TrackBase::track_state state = track->mState; 1926 mLock.unlock(); 1927 status = AudioSystem::startOutput(mId, track->streamType(), 1928 (audio_session_t)track->sessionId()); 1929 mLock.lock(); 1930 // abort track was stopped/paused while we released the lock 1931 if (state != track->mState) { 1932 if (status == NO_ERROR) { 1933 mLock.unlock(); 1934 AudioSystem::stopOutput(mId, track->streamType(), 1935 (audio_session_t)track->sessionId()); 1936 mLock.lock(); 1937 } 1938 return INVALID_OPERATION; 1939 } 1940 // abort if start is rejected by audio policy manager 1941 if (status != NO_ERROR) { 1942 return PERMISSION_DENIED; 1943 } 1944#ifdef ADD_BATTERY_DATA 1945 // to track the speaker usage 1946 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1947#endif 1948 } 1949 1950 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1951 track->mResetDone = false; 1952 track->mPresentationCompleteFrames = 0; 1953 mActiveTracks.add(track); 1954 mWakeLockUids.add(track->uid()); 1955 mActiveTracksGeneration++; 1956 mLatestActiveTrack = track; 1957 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1958 if (chain != 0) { 1959 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1960 track->sessionId()); 1961 chain->incActiveTrackCnt(); 1962 } 1963 1964 status = NO_ERROR; 1965 } 1966 1967 onAddNewTrack_l(); 1968 return status; 1969} 1970 1971bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1972{ 1973 track->terminate(); 1974 // active tracks are removed by threadLoop() 1975 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1976 track->mState = TrackBase::STOPPED; 1977 if (!trackActive) { 1978 removeTrack_l(track); 1979 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1980 track->mState = TrackBase::STOPPING_1; 1981 } 1982 1983 return trackActive; 1984} 1985 1986void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1987{ 1988 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1989 mTracks.remove(track); 1990 deleteTrackName_l(track->name()); 1991 // redundant as track is about to be destroyed, for dumpsys only 1992 track->mName = -1; 1993 if (track->isFastTrack()) { 1994 int index = track->mFastIndex; 1995 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1996 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1997 mFastTrackAvailMask |= 1 << index; 1998 // redundant as track is about to be destroyed, for dumpsys only 1999 track->mFastIndex = -1; 2000 } 2001 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2002 if (chain != 0) { 2003 chain->decTrackCnt(); 2004 } 2005} 2006 2007void AudioFlinger::PlaybackThread::broadcast_l() 2008{ 2009 // Thread could be blocked waiting for async 2010 // so signal it to handle state changes immediately 2011 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2012 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2013 mSignalPending = true; 2014 mWaitWorkCV.broadcast(); 2015} 2016 2017String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2018{ 2019 Mutex::Autolock _l(mLock); 2020 if (initCheck() != NO_ERROR) { 2021 return String8(); 2022 } 2023 2024 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2025 const String8 out_s8(s); 2026 free(s); 2027 return out_s8; 2028} 2029 2030void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2031 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2032 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2033 2034 desc->mIoHandle = mId; 2035 2036 switch (event) { 2037 case AUDIO_OUTPUT_OPENED: 2038 case AUDIO_OUTPUT_CONFIG_CHANGED: 2039 desc->mPatch = mPatch; 2040 desc->mChannelMask = mChannelMask; 2041 desc->mSamplingRate = mSampleRate; 2042 desc->mFormat = mFormat; 2043 desc->mFrameCount = mNormalFrameCount; // FIXME see 2044 // AudioFlinger::frameCount(audio_io_handle_t) 2045 desc->mLatency = latency_l(); 2046 break; 2047 2048 case AUDIO_OUTPUT_CLOSED: 2049 default: 2050 break; 2051 } 2052 mAudioFlinger->ioConfigChanged(event, desc, pid); 2053} 2054 2055void AudioFlinger::PlaybackThread::writeCallback() 2056{ 2057 ALOG_ASSERT(mCallbackThread != 0); 2058 mCallbackThread->resetWriteBlocked(); 2059} 2060 2061void AudioFlinger::PlaybackThread::drainCallback() 2062{ 2063 ALOG_ASSERT(mCallbackThread != 0); 2064 mCallbackThread->resetDraining(); 2065} 2066 2067void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2068{ 2069 Mutex::Autolock _l(mLock); 2070 // reject out of sequence requests 2071 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2072 mWriteAckSequence &= ~1; 2073 mWaitWorkCV.signal(); 2074 } 2075} 2076 2077void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2078{ 2079 Mutex::Autolock _l(mLock); 2080 // reject out of sequence requests 2081 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2082 mDrainSequence &= ~1; 2083 mWaitWorkCV.signal(); 2084 } 2085} 2086 2087// static 2088int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2089 void *param __unused, 2090 void *cookie) 2091{ 2092 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2093 ALOGV("asyncCallback() event %d", event); 2094 switch (event) { 2095 case STREAM_CBK_EVENT_WRITE_READY: 2096 me->writeCallback(); 2097 break; 2098 case STREAM_CBK_EVENT_DRAIN_READY: 2099 me->drainCallback(); 2100 break; 2101 default: 2102 ALOGW("asyncCallback() unknown event %d", event); 2103 break; 2104 } 2105 return 0; 2106} 2107 2108void AudioFlinger::PlaybackThread::readOutputParameters_l() 2109{ 2110 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2111 mSampleRate = mOutput->getSampleRate(); 2112 mChannelMask = mOutput->getChannelMask(); 2113 if (!audio_is_output_channel(mChannelMask)) { 2114 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2115 } 2116 if ((mType == MIXER || mType == DUPLICATING) 2117 && !isValidPcmSinkChannelMask(mChannelMask)) { 2118 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2119 mChannelMask); 2120 } 2121 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2122 2123 // Get actual HAL format. 2124 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2125 // Get format from the shim, which will be different than the HAL format 2126 // if playing compressed audio over HDMI passthrough. 2127 mFormat = mOutput->getFormat(); 2128 if (!audio_is_valid_format(mFormat)) { 2129 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2130 } 2131 if ((mType == MIXER || mType == DUPLICATING) 2132 && !isValidPcmSinkFormat(mFormat)) { 2133 LOG_FATAL("HAL format %#x not supported for mixed output", 2134 mFormat); 2135 } 2136 mFrameSize = mOutput->getFrameSize(); 2137 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2138 mFrameCount = mBufferSize / mFrameSize; 2139 if (mFrameCount & 15) { 2140 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2141 mFrameCount); 2142 } 2143 2144 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2145 (mOutput->stream->set_callback != NULL)) { 2146 if (mOutput->stream->set_callback(mOutput->stream, 2147 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2148 mUseAsyncWrite = true; 2149 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2150 } 2151 } 2152 2153 mHwSupportsPause = false; 2154 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2155 if (mOutput->stream->pause != NULL) { 2156 if (mOutput->stream->resume != NULL) { 2157 mHwSupportsPause = true; 2158 } else { 2159 ALOGW("direct output implements pause but not resume"); 2160 } 2161 } else if (mOutput->stream->resume != NULL) { 2162 ALOGW("direct output implements resume but not pause"); 2163 } 2164 } 2165 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2166 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2167 } 2168 2169 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2170 // For best precision, we use float instead of the associated output 2171 // device format (typically PCM 16 bit). 2172 2173 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2174 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2175 mBufferSize = mFrameSize * mFrameCount; 2176 2177 // TODO: We currently use the associated output device channel mask and sample rate. 2178 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2179 // (if a valid mask) to avoid premature downmix. 2180 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2181 // instead of the output device sample rate to avoid loss of high frequency information. 2182 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2183 } 2184 2185 // Calculate size of normal sink buffer relative to the HAL output buffer size 2186 double multiplier = 1.0; 2187 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2188 kUseFastMixer == FastMixer_Dynamic)) { 2189 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2190 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2191 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2192 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2193 maxNormalFrameCount = maxNormalFrameCount & ~15; 2194 if (maxNormalFrameCount < minNormalFrameCount) { 2195 maxNormalFrameCount = minNormalFrameCount; 2196 } 2197 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2198 if (multiplier <= 1.0) { 2199 multiplier = 1.0; 2200 } else if (multiplier <= 2.0) { 2201 if (2 * mFrameCount <= maxNormalFrameCount) { 2202 multiplier = 2.0; 2203 } else { 2204 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2205 } 2206 } else { 2207 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2208 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2209 // track, but we sometimes have to do this to satisfy the maximum frame count 2210 // constraint) 2211 // FIXME this rounding up should not be done if no HAL SRC 2212 uint32_t truncMult = (uint32_t) multiplier; 2213 if ((truncMult & 1)) { 2214 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2215 ++truncMult; 2216 } 2217 } 2218 multiplier = (double) truncMult; 2219 } 2220 } 2221 mNormalFrameCount = multiplier * mFrameCount; 2222 // round up to nearest 16 frames to satisfy AudioMixer 2223 if (mType == MIXER || mType == DUPLICATING) { 2224 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2225 } 2226 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2227 mNormalFrameCount); 2228 2229 // Check if we want to throttle the processing to no more than 2x normal rate 2230 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2231 mThreadThrottleTimeMs = 0; 2232 mThreadThrottleEndMs = 0; 2233 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2234 2235 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2236 // Originally this was int16_t[] array, need to remove legacy implications. 2237 free(mSinkBuffer); 2238 mSinkBuffer = NULL; 2239 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2240 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2241 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2242 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2243 2244 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2245 // drives the output. 2246 free(mMixerBuffer); 2247 mMixerBuffer = NULL; 2248 if (mMixerBufferEnabled) { 2249 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2250 mMixerBufferSize = mNormalFrameCount * mChannelCount 2251 * audio_bytes_per_sample(mMixerBufferFormat); 2252 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2253 } 2254 free(mEffectBuffer); 2255 mEffectBuffer = NULL; 2256 if (mEffectBufferEnabled) { 2257 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2258 mEffectBufferSize = mNormalFrameCount * mChannelCount 2259 * audio_bytes_per_sample(mEffectBufferFormat); 2260 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2261 } 2262 2263 // force reconfiguration of effect chains and engines to take new buffer size and audio 2264 // parameters into account 2265 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2266 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2267 // matter. 2268 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2269 Vector< sp<EffectChain> > effectChains = mEffectChains; 2270 for (size_t i = 0; i < effectChains.size(); i ++) { 2271 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2272 } 2273} 2274 2275 2276status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2277{ 2278 if (halFrames == NULL || dspFrames == NULL) { 2279 return BAD_VALUE; 2280 } 2281 Mutex::Autolock _l(mLock); 2282 if (initCheck() != NO_ERROR) { 2283 return INVALID_OPERATION; 2284 } 2285 size_t framesWritten = mBytesWritten / mFrameSize; 2286 *halFrames = framesWritten; 2287 2288 if (isSuspended()) { 2289 // return an estimation of rendered frames when the output is suspended 2290 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2291 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2292 return NO_ERROR; 2293 } else { 2294 status_t status; 2295 uint32_t frames; 2296 status = mOutput->getRenderPosition(&frames); 2297 *dspFrames = (size_t)frames; 2298 return status; 2299 } 2300} 2301 2302uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2303{ 2304 Mutex::Autolock _l(mLock); 2305 uint32_t result = 0; 2306 if (getEffectChain_l(sessionId) != 0) { 2307 result = EFFECT_SESSION; 2308 } 2309 2310 for (size_t i = 0; i < mTracks.size(); ++i) { 2311 sp<Track> track = mTracks[i]; 2312 if (sessionId == track->sessionId() && !track->isInvalid()) { 2313 result |= TRACK_SESSION; 2314 break; 2315 } 2316 } 2317 2318 return result; 2319} 2320 2321uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2322{ 2323 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2324 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2325 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2326 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2327 } 2328 for (size_t i = 0; i < mTracks.size(); i++) { 2329 sp<Track> track = mTracks[i]; 2330 if (sessionId == track->sessionId() && !track->isInvalid()) { 2331 return AudioSystem::getStrategyForStream(track->streamType()); 2332 } 2333 } 2334 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2335} 2336 2337 2338AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2339{ 2340 Mutex::Autolock _l(mLock); 2341 return mOutput; 2342} 2343 2344AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2345{ 2346 Mutex::Autolock _l(mLock); 2347 AudioStreamOut *output = mOutput; 2348 mOutput = NULL; 2349 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2350 // must push a NULL and wait for ack 2351 mOutputSink.clear(); 2352 mPipeSink.clear(); 2353 mNormalSink.clear(); 2354 return output; 2355} 2356 2357// this method must always be called either with ThreadBase mLock held or inside the thread loop 2358audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2359{ 2360 if (mOutput == NULL) { 2361 return NULL; 2362 } 2363 return &mOutput->stream->common; 2364} 2365 2366uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2367{ 2368 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2369} 2370 2371status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2372{ 2373 if (!isValidSyncEvent(event)) { 2374 return BAD_VALUE; 2375 } 2376 2377 Mutex::Autolock _l(mLock); 2378 2379 for (size_t i = 0; i < mTracks.size(); ++i) { 2380 sp<Track> track = mTracks[i]; 2381 if (event->triggerSession() == track->sessionId()) { 2382 (void) track->setSyncEvent(event); 2383 return NO_ERROR; 2384 } 2385 } 2386 2387 return NAME_NOT_FOUND; 2388} 2389 2390bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2391{ 2392 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2393} 2394 2395void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2396 const Vector< sp<Track> >& tracksToRemove) 2397{ 2398 size_t count = tracksToRemove.size(); 2399 if (count > 0) { 2400 for (size_t i = 0 ; i < count ; i++) { 2401 const sp<Track>& track = tracksToRemove.itemAt(i); 2402 if (track->isExternalTrack()) { 2403 AudioSystem::stopOutput(mId, track->streamType(), 2404 (audio_session_t)track->sessionId()); 2405#ifdef ADD_BATTERY_DATA 2406 // to track the speaker usage 2407 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2408#endif 2409 if (track->isTerminated()) { 2410 AudioSystem::releaseOutput(mId, track->streamType(), 2411 (audio_session_t)track->sessionId()); 2412 } 2413 } 2414 } 2415 } 2416} 2417 2418void AudioFlinger::PlaybackThread::checkSilentMode_l() 2419{ 2420 if (!mMasterMute) { 2421 char value[PROPERTY_VALUE_MAX]; 2422 if (property_get("ro.audio.silent", value, "0") > 0) { 2423 char *endptr; 2424 unsigned long ul = strtoul(value, &endptr, 0); 2425 if (*endptr == '\0' && ul != 0) { 2426 ALOGD("Silence is golden"); 2427 // The setprop command will not allow a property to be changed after 2428 // the first time it is set, so we don't have to worry about un-muting. 2429 setMasterMute_l(true); 2430 } 2431 } 2432 } 2433} 2434 2435// shared by MIXER and DIRECT, overridden by DUPLICATING 2436ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2437{ 2438 // FIXME rewrite to reduce number of system calls 2439 mLastWriteTime = systemTime(); 2440 mInWrite = true; 2441 ssize_t bytesWritten; 2442 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2443 2444 // If an NBAIO sink is present, use it to write the normal mixer's submix 2445 if (mNormalSink != 0) { 2446 2447 const size_t count = mBytesRemaining / mFrameSize; 2448 2449 ATRACE_BEGIN("write"); 2450 // update the setpoint when AudioFlinger::mScreenState changes 2451 uint32_t screenState = AudioFlinger::mScreenState; 2452 if (screenState != mScreenState) { 2453 mScreenState = screenState; 2454 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2455 if (pipe != NULL) { 2456 pipe->setAvgFrames((mScreenState & 1) ? 2457 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2458 } 2459 } 2460 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2461 ATRACE_END(); 2462 if (framesWritten > 0) { 2463 bytesWritten = framesWritten * mFrameSize; 2464 } else { 2465 bytesWritten = framesWritten; 2466 } 2467 mLatchDValid = false; 2468 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2469 if (status == NO_ERROR) { 2470 size_t totalFramesWritten = mNormalSink->framesWritten(); 2471 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2472 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2473 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2474 mLatchDValid = true; 2475 } 2476 } 2477 // otherwise use the HAL / AudioStreamOut directly 2478 } else { 2479 // Direct output and offload threads 2480 2481 if (mUseAsyncWrite) { 2482 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2483 mWriteAckSequence += 2; 2484 mWriteAckSequence |= 1; 2485 ALOG_ASSERT(mCallbackThread != 0); 2486 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2487 } 2488 // FIXME We should have an implementation of timestamps for direct output threads. 2489 // They are used e.g for multichannel PCM playback over HDMI. 2490 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2491 if (mUseAsyncWrite && 2492 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2493 // do not wait for async callback in case of error of full write 2494 mWriteAckSequence &= ~1; 2495 ALOG_ASSERT(mCallbackThread != 0); 2496 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2497 } 2498 } 2499 2500 mNumWrites++; 2501 mInWrite = false; 2502 mStandby = false; 2503 return bytesWritten; 2504} 2505 2506void AudioFlinger::PlaybackThread::threadLoop_drain() 2507{ 2508 if (mOutput->stream->drain) { 2509 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2510 if (mUseAsyncWrite) { 2511 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2512 mDrainSequence |= 1; 2513 ALOG_ASSERT(mCallbackThread != 0); 2514 mCallbackThread->setDraining(mDrainSequence); 2515 } 2516 mOutput->stream->drain(mOutput->stream, 2517 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2518 : AUDIO_DRAIN_ALL); 2519 } 2520} 2521 2522void AudioFlinger::PlaybackThread::threadLoop_exit() 2523{ 2524 { 2525 Mutex::Autolock _l(mLock); 2526 for (size_t i = 0; i < mTracks.size(); i++) { 2527 sp<Track> track = mTracks[i]; 2528 track->invalidate(); 2529 } 2530 } 2531} 2532 2533/* 2534The derived values that are cached: 2535 - mSinkBufferSize from frame count * frame size 2536 - mActiveSleepTimeUs from activeSleepTimeUs() 2537 - mIdleSleepTimeUs from idleSleepTimeUs() 2538 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) 2539 - maxPeriod from frame count and sample rate (MIXER only) 2540 2541The parameters that affect these derived values are: 2542 - frame count 2543 - frame size 2544 - sample rate 2545 - device type: A2DP or not 2546 - device latency 2547 - format: PCM or not 2548 - active sleep time 2549 - idle sleep time 2550*/ 2551 2552void AudioFlinger::PlaybackThread::cacheParameters_l() 2553{ 2554 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2555 mActiveSleepTimeUs = activeSleepTimeUs(); 2556 mIdleSleepTimeUs = idleSleepTimeUs(); 2557} 2558 2559void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2560{ 2561 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2562 this, streamType, mTracks.size()); 2563 Mutex::Autolock _l(mLock); 2564 2565 size_t size = mTracks.size(); 2566 for (size_t i = 0; i < size; i++) { 2567 sp<Track> t = mTracks[i]; 2568 if (t->streamType() == streamType && t->isExternalTrack()) { 2569 t->invalidate(); 2570 } 2571 } 2572} 2573 2574status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2575{ 2576 int session = chain->sessionId(); 2577 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2578 ? mEffectBuffer : mSinkBuffer); 2579 bool ownsBuffer = false; 2580 2581 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2582 if (session > 0) { 2583 // Only one effect chain can be present in direct output thread and it uses 2584 // the sink buffer as input 2585 if (mType != DIRECT) { 2586 size_t numSamples = mNormalFrameCount * mChannelCount; 2587 buffer = new int16_t[numSamples]; 2588 memset(buffer, 0, numSamples * sizeof(int16_t)); 2589 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2590 ownsBuffer = true; 2591 } 2592 2593 // Attach all tracks with same session ID to this chain. 2594 for (size_t i = 0; i < mTracks.size(); ++i) { 2595 sp<Track> track = mTracks[i]; 2596 if (session == track->sessionId()) { 2597 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2598 buffer); 2599 track->setMainBuffer(buffer); 2600 chain->incTrackCnt(); 2601 } 2602 } 2603 2604 // indicate all active tracks in the chain 2605 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2606 sp<Track> track = mActiveTracks[i].promote(); 2607 if (track == 0) { 2608 continue; 2609 } 2610 if (session == track->sessionId()) { 2611 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2612 chain->incActiveTrackCnt(); 2613 } 2614 } 2615 } 2616 chain->setThread(this); 2617 chain->setInBuffer(buffer, ownsBuffer); 2618 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2619 ? mEffectBuffer : mSinkBuffer)); 2620 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2621 // chains list in order to be processed last as it contains output stage effects 2622 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2623 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2624 // after track specific effects and before output stage 2625 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2626 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2627 // Effect chain for other sessions are inserted at beginning of effect 2628 // chains list to be processed before output mix effects. Relative order between other 2629 // sessions is not important 2630 size_t size = mEffectChains.size(); 2631 size_t i = 0; 2632 for (i = 0; i < size; i++) { 2633 if (mEffectChains[i]->sessionId() < session) { 2634 break; 2635 } 2636 } 2637 mEffectChains.insertAt(chain, i); 2638 checkSuspendOnAddEffectChain_l(chain); 2639 2640 return NO_ERROR; 2641} 2642 2643size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2644{ 2645 int session = chain->sessionId(); 2646 2647 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2648 2649 for (size_t i = 0; i < mEffectChains.size(); i++) { 2650 if (chain == mEffectChains[i]) { 2651 mEffectChains.removeAt(i); 2652 // detach all active tracks from the chain 2653 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2654 sp<Track> track = mActiveTracks[i].promote(); 2655 if (track == 0) { 2656 continue; 2657 } 2658 if (session == track->sessionId()) { 2659 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2660 chain.get(), session); 2661 chain->decActiveTrackCnt(); 2662 } 2663 } 2664 2665 // detach all tracks with same session ID from this chain 2666 for (size_t i = 0; i < mTracks.size(); ++i) { 2667 sp<Track> track = mTracks[i]; 2668 if (session == track->sessionId()) { 2669 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2670 chain->decTrackCnt(); 2671 } 2672 } 2673 break; 2674 } 2675 } 2676 return mEffectChains.size(); 2677} 2678 2679status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2680 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2681{ 2682 Mutex::Autolock _l(mLock); 2683 return attachAuxEffect_l(track, EffectId); 2684} 2685 2686status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2687 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2688{ 2689 status_t status = NO_ERROR; 2690 2691 if (EffectId == 0) { 2692 track->setAuxBuffer(0, NULL); 2693 } else { 2694 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2695 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2696 if (effect != 0) { 2697 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2698 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2699 } else { 2700 status = INVALID_OPERATION; 2701 } 2702 } else { 2703 status = BAD_VALUE; 2704 } 2705 } 2706 return status; 2707} 2708 2709void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2710{ 2711 for (size_t i = 0; i < mTracks.size(); ++i) { 2712 sp<Track> track = mTracks[i]; 2713 if (track->auxEffectId() == effectId) { 2714 attachAuxEffect_l(track, 0); 2715 } 2716 } 2717} 2718 2719bool AudioFlinger::PlaybackThread::threadLoop() 2720{ 2721 Vector< sp<Track> > tracksToRemove; 2722 2723 mStandbyTimeNs = systemTime(); 2724 2725 // MIXER 2726 nsecs_t lastWarning = 0; 2727 2728 // DUPLICATING 2729 // FIXME could this be made local to while loop? 2730 writeFrames = 0; 2731 2732 int lastGeneration = 0; 2733 2734 cacheParameters_l(); 2735 mSleepTimeUs = mIdleSleepTimeUs; 2736 2737 if (mType == MIXER) { 2738 sleepTimeShift = 0; 2739 } 2740 2741 CpuStats cpuStats; 2742 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2743 2744 acquireWakeLock(); 2745 2746 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2747 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2748 // and then that string will be logged at the next convenient opportunity. 2749 const char *logString = NULL; 2750 2751 checkSilentMode_l(); 2752 2753 while (!exitPending()) 2754 { 2755 cpuStats.sample(myName); 2756 2757 Vector< sp<EffectChain> > effectChains; 2758 2759 { // scope for mLock 2760 2761 Mutex::Autolock _l(mLock); 2762 2763 processConfigEvents_l(); 2764 2765 if (logString != NULL) { 2766 mNBLogWriter->logTimestamp(); 2767 mNBLogWriter->log(logString); 2768 logString = NULL; 2769 } 2770 2771 // Gather the framesReleased counters for all active tracks, 2772 // and latch them atomically with the timestamp. 2773 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2774 mLatchD.mFramesReleased.clear(); 2775 size_t size = mActiveTracks.size(); 2776 for (size_t i = 0; i < size; i++) { 2777 sp<Track> t = mActiveTracks[i].promote(); 2778 if (t != 0) { 2779 mLatchD.mFramesReleased.add(t.get(), 2780 t->mAudioTrackServerProxy->framesReleased()); 2781 } 2782 } 2783 if (mLatchDValid) { 2784 mLatchQ = mLatchD; 2785 mLatchDValid = false; 2786 mLatchQValid = true; 2787 } 2788 2789 saveOutputTracks(); 2790 if (mSignalPending) { 2791 // A signal was raised while we were unlocked 2792 mSignalPending = false; 2793 } else if (waitingAsyncCallback_l()) { 2794 if (exitPending()) { 2795 break; 2796 } 2797 bool released = false; 2798 // The following works around a bug in the offload driver. Ideally we would release 2799 // the wake lock every time, but that causes the last offload buffer(s) to be 2800 // dropped while the device is on battery, so we need to hold a wake lock during 2801 // the drain phase. 2802 if (mBytesRemaining && !(mDrainSequence & 1)) { 2803 releaseWakeLock_l(); 2804 released = true; 2805 } 2806 mWakeLockUids.clear(); 2807 mActiveTracksGeneration++; 2808 ALOGV("wait async completion"); 2809 mWaitWorkCV.wait(mLock); 2810 ALOGV("async completion/wake"); 2811 if (released) { 2812 acquireWakeLock_l(); 2813 } 2814 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2815 mSleepTimeUs = 0; 2816 2817 continue; 2818 } 2819 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2820 isSuspended()) { 2821 // put audio hardware into standby after short delay 2822 if (shouldStandby_l()) { 2823 2824 threadLoop_standby(); 2825 2826 mStandby = true; 2827 } 2828 2829 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2830 // we're about to wait, flush the binder command buffer 2831 IPCThreadState::self()->flushCommands(); 2832 2833 clearOutputTracks(); 2834 2835 if (exitPending()) { 2836 break; 2837 } 2838 2839 releaseWakeLock_l(); 2840 mWakeLockUids.clear(); 2841 mActiveTracksGeneration++; 2842 // wait until we have something to do... 2843 ALOGV("%s going to sleep", myName.string()); 2844 mWaitWorkCV.wait(mLock); 2845 ALOGV("%s waking up", myName.string()); 2846 acquireWakeLock_l(); 2847 2848 mMixerStatus = MIXER_IDLE; 2849 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2850 mBytesWritten = 0; 2851 mBytesRemaining = 0; 2852 checkSilentMode_l(); 2853 2854 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2855 mSleepTimeUs = mIdleSleepTimeUs; 2856 if (mType == MIXER) { 2857 sleepTimeShift = 0; 2858 } 2859 2860 continue; 2861 } 2862 } 2863 // mMixerStatusIgnoringFastTracks is also updated internally 2864 mMixerStatus = prepareTracks_l(&tracksToRemove); 2865 2866 // compare with previously applied list 2867 if (lastGeneration != mActiveTracksGeneration) { 2868 // update wakelock 2869 updateWakeLockUids_l(mWakeLockUids); 2870 lastGeneration = mActiveTracksGeneration; 2871 } 2872 2873 // prevent any changes in effect chain list and in each effect chain 2874 // during mixing and effect process as the audio buffers could be deleted 2875 // or modified if an effect is created or deleted 2876 lockEffectChains_l(effectChains); 2877 } // mLock scope ends 2878 2879 if (mBytesRemaining == 0) { 2880 mCurrentWriteLength = 0; 2881 if (mMixerStatus == MIXER_TRACKS_READY) { 2882 // threadLoop_mix() sets mCurrentWriteLength 2883 threadLoop_mix(); 2884 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2885 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2886 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2887 // must be written to HAL 2888 threadLoop_sleepTime(); 2889 if (mSleepTimeUs == 0) { 2890 mCurrentWriteLength = mSinkBufferSize; 2891 } 2892 } 2893 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2894 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2895 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2896 // or mSinkBuffer (if there are no effects). 2897 // 2898 // This is done pre-effects computation; if effects change to 2899 // support higher precision, this needs to move. 2900 // 2901 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2902 // TODO use mSleepTimeUs == 0 as an additional condition. 2903 if (mMixerBufferValid) { 2904 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2905 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2906 2907 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2908 mNormalFrameCount * mChannelCount); 2909 } 2910 2911 mBytesRemaining = mCurrentWriteLength; 2912 if (isSuspended()) { 2913 mSleepTimeUs = suspendSleepTimeUs(); 2914 // simulate write to HAL when suspended 2915 mBytesWritten += mSinkBufferSize; 2916 mBytesRemaining = 0; 2917 } 2918 2919 // only process effects if we're going to write 2920 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 2921 for (size_t i = 0; i < effectChains.size(); i ++) { 2922 effectChains[i]->process_l(); 2923 } 2924 } 2925 } 2926 // Process effect chains for offloaded thread even if no audio 2927 // was read from audio track: process only updates effect state 2928 // and thus does have to be synchronized with audio writes but may have 2929 // to be called while waiting for async write callback 2930 if (mType == OFFLOAD) { 2931 for (size_t i = 0; i < effectChains.size(); i ++) { 2932 effectChains[i]->process_l(); 2933 } 2934 } 2935 2936 // Only if the Effects buffer is enabled and there is data in the 2937 // Effects buffer (buffer valid), we need to 2938 // copy into the sink buffer. 2939 // TODO use mSleepTimeUs == 0 as an additional condition. 2940 if (mEffectBufferValid) { 2941 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2942 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2943 mNormalFrameCount * mChannelCount); 2944 } 2945 2946 // enable changes in effect chain 2947 unlockEffectChains(effectChains); 2948 2949 if (!waitingAsyncCallback()) { 2950 // mSleepTimeUs == 0 means we must write to audio hardware 2951 if (mSleepTimeUs == 0) { 2952 ssize_t ret = 0; 2953 if (mBytesRemaining) { 2954 ret = threadLoop_write(); 2955 if (ret < 0) { 2956 mBytesRemaining = 0; 2957 } else { 2958 mBytesWritten += ret; 2959 mBytesRemaining -= ret; 2960 } 2961 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2962 (mMixerStatus == MIXER_DRAIN_ALL)) { 2963 threadLoop_drain(); 2964 } 2965 if (mType == MIXER && !mStandby) { 2966 // write blocked detection 2967 nsecs_t now = systemTime(); 2968 nsecs_t delta = now - mLastWriteTime; 2969 if (delta > maxPeriod) { 2970 mNumDelayedWrites++; 2971 if ((now - lastWarning) > kWarningThrottleNs) { 2972 ATRACE_NAME("underrun"); 2973 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2974 ns2ms(delta), mNumDelayedWrites, this); 2975 lastWarning = now; 2976 } 2977 } 2978 2979 if (mThreadThrottle 2980 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 2981 && ret > 0) { // we wrote something 2982 // Limit MixerThread data processing to no more than twice the 2983 // expected processing rate. 2984 // 2985 // This helps prevent underruns with NuPlayer and other applications 2986 // which may set up buffers that are close to the minimum size, or use 2987 // deep buffers, and rely on a double-buffering sleep strategy to fill. 2988 // 2989 // The throttle smooths out sudden large data drains from the device, 2990 // e.g. when it comes out of standby, which often causes problems with 2991 // (1) mixer threads without a fast mixer (which has its own warm-up) 2992 // (2) minimum buffer sized tracks (even if the track is full, 2993 // the app won't fill fast enough to handle the sudden draw). 2994 2995 const int32_t deltaMs = delta / 1000000; 2996 const int32_t throttleMs = mHalfBufferMs - deltaMs; 2997 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 2998 usleep(throttleMs * 1000); 2999 // notify of throttle start on verbose log 3000 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3001 "mixer(%p) throttle begin:" 3002 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3003 this, ret, deltaMs, throttleMs); 3004 mThreadThrottleTimeMs += throttleMs; 3005 } else { 3006 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3007 if (diff > 0) { 3008 // notify of throttle end on debug log 3009 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3010 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3011 } 3012 } 3013 } 3014 } 3015 3016 } else { 3017 ATRACE_BEGIN("sleep"); 3018 usleep(mSleepTimeUs); 3019 ATRACE_END(); 3020 } 3021 } 3022 3023 // Finally let go of removed track(s), without the lock held 3024 // since we can't guarantee the destructors won't acquire that 3025 // same lock. This will also mutate and push a new fast mixer state. 3026 threadLoop_removeTracks(tracksToRemove); 3027 tracksToRemove.clear(); 3028 3029 // FIXME I don't understand the need for this here; 3030 // it was in the original code but maybe the 3031 // assignment in saveOutputTracks() makes this unnecessary? 3032 clearOutputTracks(); 3033 3034 // Effect chains will be actually deleted here if they were removed from 3035 // mEffectChains list during mixing or effects processing 3036 effectChains.clear(); 3037 3038 // FIXME Note that the above .clear() is no longer necessary since effectChains 3039 // is now local to this block, but will keep it for now (at least until merge done). 3040 } 3041 3042 threadLoop_exit(); 3043 3044 if (!mStandby) { 3045 threadLoop_standby(); 3046 mStandby = true; 3047 } 3048 3049 releaseWakeLock(); 3050 mWakeLockUids.clear(); 3051 mActiveTracksGeneration++; 3052 3053 ALOGV("Thread %p type %d exiting", this, mType); 3054 return false; 3055} 3056 3057// removeTracks_l() must be called with ThreadBase::mLock held 3058void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3059{ 3060 size_t count = tracksToRemove.size(); 3061 if (count > 0) { 3062 for (size_t i=0 ; i<count ; i++) { 3063 const sp<Track>& track = tracksToRemove.itemAt(i); 3064 mActiveTracks.remove(track); 3065 mWakeLockUids.remove(track->uid()); 3066 mActiveTracksGeneration++; 3067 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3068 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3069 if (chain != 0) { 3070 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3071 track->sessionId()); 3072 chain->decActiveTrackCnt(); 3073 } 3074 if (track->isTerminated()) { 3075 removeTrack_l(track); 3076 } 3077 } 3078 } 3079 3080} 3081 3082status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3083{ 3084 if (mNormalSink != 0) { 3085 return mNormalSink->getTimestamp(timestamp); 3086 } 3087 if ((mType == OFFLOAD || mType == DIRECT) 3088 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3089 uint64_t position64; 3090 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3091 if (ret == 0) { 3092 timestamp.mPosition = (uint32_t)position64; 3093 return NO_ERROR; 3094 } 3095 } 3096 return INVALID_OPERATION; 3097} 3098 3099status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3100 audio_patch_handle_t *handle) 3101{ 3102 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3103 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3104 if (mFastMixer != 0) { 3105 FastMixerStateQueue *sq = mFastMixer->sq(); 3106 FastMixerState *state = sq->begin(); 3107 if (!(state->mCommand & FastMixerState::IDLE)) { 3108 previousCommand = state->mCommand; 3109 state->mCommand = FastMixerState::HOT_IDLE; 3110 sq->end(); 3111 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3112 } else { 3113 sq->end(false /*didModify*/); 3114 } 3115 } 3116 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3117 3118 if (!(previousCommand & FastMixerState::IDLE)) { 3119 ALOG_ASSERT(mFastMixer != 0); 3120 FastMixerStateQueue *sq = mFastMixer->sq(); 3121 FastMixerState *state = sq->begin(); 3122 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3123 state->mCommand = previousCommand; 3124 sq->end(); 3125 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3126 } 3127 3128 return status; 3129} 3130 3131status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3132 audio_patch_handle_t *handle) 3133{ 3134 status_t status = NO_ERROR; 3135 3136 // store new device and send to effects 3137 audio_devices_t type = AUDIO_DEVICE_NONE; 3138 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3139 type |= patch->sinks[i].ext.device.type; 3140 } 3141 3142#ifdef ADD_BATTERY_DATA 3143 // when changing the audio output device, call addBatteryData to notify 3144 // the change 3145 if (mOutDevice != type) { 3146 uint32_t params = 0; 3147 // check whether speaker is on 3148 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3149 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3150 } 3151 3152 audio_devices_t deviceWithoutSpeaker 3153 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3154 // check if any other device (except speaker) is on 3155 if (type & deviceWithoutSpeaker) { 3156 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3157 } 3158 3159 if (params != 0) { 3160 addBatteryData(params); 3161 } 3162 } 3163#endif 3164 3165 for (size_t i = 0; i < mEffectChains.size(); i++) { 3166 mEffectChains[i]->setDevice_l(type); 3167 } 3168 3169 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3170 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3171 bool configChanged = mPrevOutDevice != type; 3172 mOutDevice = type; 3173 mPatch = *patch; 3174 3175 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3176 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3177 status = hwDevice->create_audio_patch(hwDevice, 3178 patch->num_sources, 3179 patch->sources, 3180 patch->num_sinks, 3181 patch->sinks, 3182 handle); 3183 } else { 3184 char *address; 3185 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3186 //FIXME: we only support address on first sink with HAL version < 3.0 3187 address = audio_device_address_to_parameter( 3188 patch->sinks[0].ext.device.type, 3189 patch->sinks[0].ext.device.address); 3190 } else { 3191 address = (char *)calloc(1, 1); 3192 } 3193 AudioParameter param = AudioParameter(String8(address)); 3194 free(address); 3195 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3196 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3197 param.toString().string()); 3198 *handle = AUDIO_PATCH_HANDLE_NONE; 3199 } 3200 if (configChanged) { 3201 mPrevOutDevice = type; 3202 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3203 } 3204 return status; 3205} 3206 3207status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3208{ 3209 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3210 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3211 if (mFastMixer != 0) { 3212 FastMixerStateQueue *sq = mFastMixer->sq(); 3213 FastMixerState *state = sq->begin(); 3214 if (!(state->mCommand & FastMixerState::IDLE)) { 3215 previousCommand = state->mCommand; 3216 state->mCommand = FastMixerState::HOT_IDLE; 3217 sq->end(); 3218 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3219 } else { 3220 sq->end(false /*didModify*/); 3221 } 3222 } 3223 3224 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3225 3226 if (!(previousCommand & FastMixerState::IDLE)) { 3227 ALOG_ASSERT(mFastMixer != 0); 3228 FastMixerStateQueue *sq = mFastMixer->sq(); 3229 FastMixerState *state = sq->begin(); 3230 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3231 state->mCommand = previousCommand; 3232 sq->end(); 3233 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3234 } 3235 3236 return status; 3237} 3238 3239status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3240{ 3241 status_t status = NO_ERROR; 3242 3243 mOutDevice = AUDIO_DEVICE_NONE; 3244 3245 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3246 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3247 status = hwDevice->release_audio_patch(hwDevice, handle); 3248 } else { 3249 AudioParameter param; 3250 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3251 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3252 param.toString().string()); 3253 } 3254 return status; 3255} 3256 3257void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3258{ 3259 Mutex::Autolock _l(mLock); 3260 mTracks.add(track); 3261} 3262 3263void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3264{ 3265 Mutex::Autolock _l(mLock); 3266 destroyTrack_l(track); 3267} 3268 3269void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3270{ 3271 ThreadBase::getAudioPortConfig(config); 3272 config->role = AUDIO_PORT_ROLE_SOURCE; 3273 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3274 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3275} 3276 3277// ---------------------------------------------------------------------------- 3278 3279AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3280 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3281 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3282 // mAudioMixer below 3283 // mFastMixer below 3284 mFastMixerFutex(0) 3285 // mOutputSink below 3286 // mPipeSink below 3287 // mNormalSink below 3288{ 3289 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3290 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3291 "mFrameCount=%d, mNormalFrameCount=%d", 3292 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3293 mNormalFrameCount); 3294 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3295 3296 if (type == DUPLICATING) { 3297 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3298 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3299 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3300 return; 3301 } 3302 // create an NBAIO sink for the HAL output stream, and negotiate 3303 mOutputSink = new AudioStreamOutSink(output->stream); 3304 size_t numCounterOffers = 0; 3305 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3306 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3307 ALOG_ASSERT(index == 0); 3308 3309 // initialize fast mixer depending on configuration 3310 bool initFastMixer; 3311 switch (kUseFastMixer) { 3312 case FastMixer_Never: 3313 initFastMixer = false; 3314 break; 3315 case FastMixer_Always: 3316 initFastMixer = true; 3317 break; 3318 case FastMixer_Static: 3319 case FastMixer_Dynamic: 3320 initFastMixer = mFrameCount < mNormalFrameCount; 3321 break; 3322 } 3323 if (initFastMixer) { 3324 audio_format_t fastMixerFormat; 3325 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3326 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3327 } else { 3328 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3329 } 3330 if (mFormat != fastMixerFormat) { 3331 // change our Sink format to accept our intermediate precision 3332 mFormat = fastMixerFormat; 3333 free(mSinkBuffer); 3334 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3335 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3336 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3337 } 3338 3339 // create a MonoPipe to connect our submix to FastMixer 3340 NBAIO_Format format = mOutputSink->format(); 3341 NBAIO_Format origformat = format; 3342 // adjust format to match that of the Fast Mixer 3343 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3344 format.mFormat = fastMixerFormat; 3345 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3346 3347 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3348 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3349 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3350 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3351 const NBAIO_Format offers[1] = {format}; 3352 size_t numCounterOffers = 0; 3353 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3354 ALOG_ASSERT(index == 0); 3355 monoPipe->setAvgFrames((mScreenState & 1) ? 3356 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3357 mPipeSink = monoPipe; 3358 3359#ifdef TEE_SINK 3360 if (mTeeSinkOutputEnabled) { 3361 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3362 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3363 const NBAIO_Format offers2[1] = {origformat}; 3364 numCounterOffers = 0; 3365 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3366 ALOG_ASSERT(index == 0); 3367 mTeeSink = teeSink; 3368 PipeReader *teeSource = new PipeReader(*teeSink); 3369 numCounterOffers = 0; 3370 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3371 ALOG_ASSERT(index == 0); 3372 mTeeSource = teeSource; 3373 } 3374#endif 3375 3376 // create fast mixer and configure it initially with just one fast track for our submix 3377 mFastMixer = new FastMixer(); 3378 FastMixerStateQueue *sq = mFastMixer->sq(); 3379#ifdef STATE_QUEUE_DUMP 3380 sq->setObserverDump(&mStateQueueObserverDump); 3381 sq->setMutatorDump(&mStateQueueMutatorDump); 3382#endif 3383 FastMixerState *state = sq->begin(); 3384 FastTrack *fastTrack = &state->mFastTracks[0]; 3385 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3386 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3387 fastTrack->mVolumeProvider = NULL; 3388 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3389 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3390 fastTrack->mGeneration++; 3391 state->mFastTracksGen++; 3392 state->mTrackMask = 1; 3393 // fast mixer will use the HAL output sink 3394 state->mOutputSink = mOutputSink.get(); 3395 state->mOutputSinkGen++; 3396 state->mFrameCount = mFrameCount; 3397 state->mCommand = FastMixerState::COLD_IDLE; 3398 // already done in constructor initialization list 3399 //mFastMixerFutex = 0; 3400 state->mColdFutexAddr = &mFastMixerFutex; 3401 state->mColdGen++; 3402 state->mDumpState = &mFastMixerDumpState; 3403#ifdef TEE_SINK 3404 state->mTeeSink = mTeeSink.get(); 3405#endif 3406 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3407 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3408 sq->end(); 3409 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3410 3411 // start the fast mixer 3412 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3413 pid_t tid = mFastMixer->getTid(); 3414 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3415 3416#ifdef AUDIO_WATCHDOG 3417 // create and start the watchdog 3418 mAudioWatchdog = new AudioWatchdog(); 3419 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3420 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3421 tid = mAudioWatchdog->getTid(); 3422 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3423#endif 3424 3425 } 3426 3427 switch (kUseFastMixer) { 3428 case FastMixer_Never: 3429 case FastMixer_Dynamic: 3430 mNormalSink = mOutputSink; 3431 break; 3432 case FastMixer_Always: 3433 mNormalSink = mPipeSink; 3434 break; 3435 case FastMixer_Static: 3436 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3437 break; 3438 } 3439} 3440 3441AudioFlinger::MixerThread::~MixerThread() 3442{ 3443 if (mFastMixer != 0) { 3444 FastMixerStateQueue *sq = mFastMixer->sq(); 3445 FastMixerState *state = sq->begin(); 3446 if (state->mCommand == FastMixerState::COLD_IDLE) { 3447 int32_t old = android_atomic_inc(&mFastMixerFutex); 3448 if (old == -1) { 3449 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3450 } 3451 } 3452 state->mCommand = FastMixerState::EXIT; 3453 sq->end(); 3454 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3455 mFastMixer->join(); 3456 // Though the fast mixer thread has exited, it's state queue is still valid. 3457 // We'll use that extract the final state which contains one remaining fast track 3458 // corresponding to our sub-mix. 3459 state = sq->begin(); 3460 ALOG_ASSERT(state->mTrackMask == 1); 3461 FastTrack *fastTrack = &state->mFastTracks[0]; 3462 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3463 delete fastTrack->mBufferProvider; 3464 sq->end(false /*didModify*/); 3465 mFastMixer.clear(); 3466#ifdef AUDIO_WATCHDOG 3467 if (mAudioWatchdog != 0) { 3468 mAudioWatchdog->requestExit(); 3469 mAudioWatchdog->requestExitAndWait(); 3470 mAudioWatchdog.clear(); 3471 } 3472#endif 3473 } 3474 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3475 delete mAudioMixer; 3476} 3477 3478 3479uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3480{ 3481 if (mFastMixer != 0) { 3482 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3483 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3484 } 3485 return latency; 3486} 3487 3488 3489void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3490{ 3491 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3492} 3493 3494ssize_t AudioFlinger::MixerThread::threadLoop_write() 3495{ 3496 // FIXME we should only do one push per cycle; confirm this is true 3497 // Start the fast mixer if it's not already running 3498 if (mFastMixer != 0) { 3499 FastMixerStateQueue *sq = mFastMixer->sq(); 3500 FastMixerState *state = sq->begin(); 3501 if (state->mCommand != FastMixerState::MIX_WRITE && 3502 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3503 if (state->mCommand == FastMixerState::COLD_IDLE) { 3504 3505 // FIXME workaround for first HAL write being CPU bound on some devices 3506 ATRACE_BEGIN("write"); 3507 mOutput->write((char *)mSinkBuffer, 0); 3508 ATRACE_END(); 3509 3510 int32_t old = android_atomic_inc(&mFastMixerFutex); 3511 if (old == -1) { 3512 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3513 } 3514#ifdef AUDIO_WATCHDOG 3515 if (mAudioWatchdog != 0) { 3516 mAudioWatchdog->resume(); 3517 } 3518#endif 3519 } 3520 state->mCommand = FastMixerState::MIX_WRITE; 3521#ifdef FAST_THREAD_STATISTICS 3522 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3523 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3524#endif 3525 sq->end(); 3526 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3527 if (kUseFastMixer == FastMixer_Dynamic) { 3528 mNormalSink = mPipeSink; 3529 } 3530 } else { 3531 sq->end(false /*didModify*/); 3532 } 3533 } 3534 return PlaybackThread::threadLoop_write(); 3535} 3536 3537void AudioFlinger::MixerThread::threadLoop_standby() 3538{ 3539 // Idle the fast mixer if it's currently running 3540 if (mFastMixer != 0) { 3541 FastMixerStateQueue *sq = mFastMixer->sq(); 3542 FastMixerState *state = sq->begin(); 3543 if (!(state->mCommand & FastMixerState::IDLE)) { 3544 state->mCommand = FastMixerState::COLD_IDLE; 3545 state->mColdFutexAddr = &mFastMixerFutex; 3546 state->mColdGen++; 3547 mFastMixerFutex = 0; 3548 sq->end(); 3549 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3550 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3551 if (kUseFastMixer == FastMixer_Dynamic) { 3552 mNormalSink = mOutputSink; 3553 } 3554#ifdef AUDIO_WATCHDOG 3555 if (mAudioWatchdog != 0) { 3556 mAudioWatchdog->pause(); 3557 } 3558#endif 3559 } else { 3560 sq->end(false /*didModify*/); 3561 } 3562 } 3563 PlaybackThread::threadLoop_standby(); 3564} 3565 3566bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3567{ 3568 return false; 3569} 3570 3571bool AudioFlinger::PlaybackThread::shouldStandby_l() 3572{ 3573 return !mStandby; 3574} 3575 3576bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3577{ 3578 Mutex::Autolock _l(mLock); 3579 return waitingAsyncCallback_l(); 3580} 3581 3582// shared by MIXER and DIRECT, overridden by DUPLICATING 3583void AudioFlinger::PlaybackThread::threadLoop_standby() 3584{ 3585 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3586 mOutput->standby(); 3587 if (mUseAsyncWrite != 0) { 3588 // discard any pending drain or write ack by incrementing sequence 3589 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3590 mDrainSequence = (mDrainSequence + 2) & ~1; 3591 ALOG_ASSERT(mCallbackThread != 0); 3592 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3593 mCallbackThread->setDraining(mDrainSequence); 3594 } 3595 mHwPaused = false; 3596} 3597 3598void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3599{ 3600 ALOGV("signal playback thread"); 3601 broadcast_l(); 3602} 3603 3604void AudioFlinger::MixerThread::threadLoop_mix() 3605{ 3606 // obtain the presentation timestamp of the next output buffer 3607 int64_t pts; 3608 status_t status = INVALID_OPERATION; 3609 3610 if (mNormalSink != 0) { 3611 status = mNormalSink->getNextWriteTimestamp(&pts); 3612 } else { 3613 status = mOutputSink->getNextWriteTimestamp(&pts); 3614 } 3615 3616 if (status != NO_ERROR) { 3617 pts = AudioBufferProvider::kInvalidPTS; 3618 } 3619 3620 // mix buffers... 3621 mAudioMixer->process(pts); 3622 mCurrentWriteLength = mSinkBufferSize; 3623 // increase sleep time progressively when application underrun condition clears. 3624 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3625 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3626 // such that we would underrun the audio HAL. 3627 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3628 sleepTimeShift--; 3629 } 3630 mSleepTimeUs = 0; 3631 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3632 //TODO: delay standby when effects have a tail 3633 3634} 3635 3636void AudioFlinger::MixerThread::threadLoop_sleepTime() 3637{ 3638 // If no tracks are ready, sleep once for the duration of an output 3639 // buffer size, then write 0s to the output 3640 if (mSleepTimeUs == 0) { 3641 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3642 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3643 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3644 mSleepTimeUs = kMinThreadSleepTimeUs; 3645 } 3646 // reduce sleep time in case of consecutive application underruns to avoid 3647 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3648 // duration we would end up writing less data than needed by the audio HAL if 3649 // the condition persists. 3650 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3651 sleepTimeShift++; 3652 } 3653 } else { 3654 mSleepTimeUs = mIdleSleepTimeUs; 3655 } 3656 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3657 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3658 // before effects processing or output. 3659 if (mMixerBufferValid) { 3660 memset(mMixerBuffer, 0, mMixerBufferSize); 3661 } else { 3662 memset(mSinkBuffer, 0, mSinkBufferSize); 3663 } 3664 mSleepTimeUs = 0; 3665 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3666 "anticipated start"); 3667 } 3668 // TODO add standby time extension fct of effect tail 3669} 3670 3671// prepareTracks_l() must be called with ThreadBase::mLock held 3672AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3673 Vector< sp<Track> > *tracksToRemove) 3674{ 3675 3676 mixer_state mixerStatus = MIXER_IDLE; 3677 // find out which tracks need to be processed 3678 size_t count = mActiveTracks.size(); 3679 size_t mixedTracks = 0; 3680 size_t tracksWithEffect = 0; 3681 // counts only _active_ fast tracks 3682 size_t fastTracks = 0; 3683 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3684 3685 float masterVolume = mMasterVolume; 3686 bool masterMute = mMasterMute; 3687 3688 if (masterMute) { 3689 masterVolume = 0; 3690 } 3691 // Delegate master volume control to effect in output mix effect chain if needed 3692 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3693 if (chain != 0) { 3694 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3695 chain->setVolume_l(&v, &v); 3696 masterVolume = (float)((v + (1 << 23)) >> 24); 3697 chain.clear(); 3698 } 3699 3700 // prepare a new state to push 3701 FastMixerStateQueue *sq = NULL; 3702 FastMixerState *state = NULL; 3703 bool didModify = false; 3704 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3705 if (mFastMixer != 0) { 3706 sq = mFastMixer->sq(); 3707 state = sq->begin(); 3708 } 3709 3710 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3711 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3712 3713 for (size_t i=0 ; i<count ; i++) { 3714 const sp<Track> t = mActiveTracks[i].promote(); 3715 if (t == 0) { 3716 continue; 3717 } 3718 3719 // this const just means the local variable doesn't change 3720 Track* const track = t.get(); 3721 3722 // process fast tracks 3723 if (track->isFastTrack()) { 3724 3725 // It's theoretically possible (though unlikely) for a fast track to be created 3726 // and then removed within the same normal mix cycle. This is not a problem, as 3727 // the track never becomes active so it's fast mixer slot is never touched. 3728 // The converse, of removing an (active) track and then creating a new track 3729 // at the identical fast mixer slot within the same normal mix cycle, 3730 // is impossible because the slot isn't marked available until the end of each cycle. 3731 int j = track->mFastIndex; 3732 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3733 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3734 FastTrack *fastTrack = &state->mFastTracks[j]; 3735 3736 // Determine whether the track is currently in underrun condition, 3737 // and whether it had a recent underrun. 3738 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3739 FastTrackUnderruns underruns = ftDump->mUnderruns; 3740 uint32_t recentFull = (underruns.mBitFields.mFull - 3741 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3742 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3743 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3744 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3745 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3746 uint32_t recentUnderruns = recentPartial + recentEmpty; 3747 track->mObservedUnderruns = underruns; 3748 // don't count underruns that occur while stopping or pausing 3749 // or stopped which can occur when flush() is called while active 3750 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3751 recentUnderruns > 0) { 3752 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3753 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3754 } 3755 3756 // This is similar to the state machine for normal tracks, 3757 // with a few modifications for fast tracks. 3758 bool isActive = true; 3759 switch (track->mState) { 3760 case TrackBase::STOPPING_1: 3761 // track stays active in STOPPING_1 state until first underrun 3762 if (recentUnderruns > 0 || track->isTerminated()) { 3763 track->mState = TrackBase::STOPPING_2; 3764 } 3765 break; 3766 case TrackBase::PAUSING: 3767 // ramp down is not yet implemented 3768 track->setPaused(); 3769 break; 3770 case TrackBase::RESUMING: 3771 // ramp up is not yet implemented 3772 track->mState = TrackBase::ACTIVE; 3773 break; 3774 case TrackBase::ACTIVE: 3775 if (recentFull > 0 || recentPartial > 0) { 3776 // track has provided at least some frames recently: reset retry count 3777 track->mRetryCount = kMaxTrackRetries; 3778 } 3779 if (recentUnderruns == 0) { 3780 // no recent underruns: stay active 3781 break; 3782 } 3783 // there has recently been an underrun of some kind 3784 if (track->sharedBuffer() == 0) { 3785 // were any of the recent underruns "empty" (no frames available)? 3786 if (recentEmpty == 0) { 3787 // no, then ignore the partial underruns as they are allowed indefinitely 3788 break; 3789 } 3790 // there has recently been an "empty" underrun: decrement the retry counter 3791 if (--(track->mRetryCount) > 0) { 3792 break; 3793 } 3794 // indicate to client process that the track was disabled because of underrun; 3795 // it will then automatically call start() when data is available 3796 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3797 // remove from active list, but state remains ACTIVE [confusing but true] 3798 isActive = false; 3799 break; 3800 } 3801 // fall through 3802 case TrackBase::STOPPING_2: 3803 case TrackBase::PAUSED: 3804 case TrackBase::STOPPED: 3805 case TrackBase::FLUSHED: // flush() while active 3806 // Check for presentation complete if track is inactive 3807 // We have consumed all the buffers of this track. 3808 // This would be incomplete if we auto-paused on underrun 3809 { 3810 size_t audioHALFrames = 3811 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3812 size_t framesWritten = mBytesWritten / mFrameSize; 3813 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3814 // track stays in active list until presentation is complete 3815 break; 3816 } 3817 } 3818 if (track->isStopping_2()) { 3819 track->mState = TrackBase::STOPPED; 3820 } 3821 if (track->isStopped()) { 3822 // Can't reset directly, as fast mixer is still polling this track 3823 // track->reset(); 3824 // So instead mark this track as needing to be reset after push with ack 3825 resetMask |= 1 << i; 3826 } 3827 isActive = false; 3828 break; 3829 case TrackBase::IDLE: 3830 default: 3831 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3832 } 3833 3834 if (isActive) { 3835 // was it previously inactive? 3836 if (!(state->mTrackMask & (1 << j))) { 3837 ExtendedAudioBufferProvider *eabp = track; 3838 VolumeProvider *vp = track; 3839 fastTrack->mBufferProvider = eabp; 3840 fastTrack->mVolumeProvider = vp; 3841 fastTrack->mChannelMask = track->mChannelMask; 3842 fastTrack->mFormat = track->mFormat; 3843 fastTrack->mGeneration++; 3844 state->mTrackMask |= 1 << j; 3845 didModify = true; 3846 // no acknowledgement required for newly active tracks 3847 } 3848 // cache the combined master volume and stream type volume for fast mixer; this 3849 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3850 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3851 ++fastTracks; 3852 } else { 3853 // was it previously active? 3854 if (state->mTrackMask & (1 << j)) { 3855 fastTrack->mBufferProvider = NULL; 3856 fastTrack->mGeneration++; 3857 state->mTrackMask &= ~(1 << j); 3858 didModify = true; 3859 // If any fast tracks were removed, we must wait for acknowledgement 3860 // because we're about to decrement the last sp<> on those tracks. 3861 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3862 } else { 3863 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3864 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 3865 j, track->mState, state->mTrackMask, recentUnderruns, 3866 track->sharedBuffer() != 0); 3867 } 3868 tracksToRemove->add(track); 3869 // Avoids a misleading display in dumpsys 3870 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3871 } 3872 continue; 3873 } 3874 3875 { // local variable scope to avoid goto warning 3876 3877 audio_track_cblk_t* cblk = track->cblk(); 3878 3879 // The first time a track is added we wait 3880 // for all its buffers to be filled before processing it 3881 int name = track->name(); 3882 // make sure that we have enough frames to mix one full buffer. 3883 // enforce this condition only once to enable draining the buffer in case the client 3884 // app does not call stop() and relies on underrun to stop: 3885 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3886 // during last round 3887 size_t desiredFrames; 3888 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3889 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3890 3891 desiredFrames = sourceFramesNeededWithTimestretch( 3892 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3893 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3894 // add frames already consumed but not yet released by the resampler 3895 // because mAudioTrackServerProxy->framesReady() will include these frames 3896 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3897 3898 uint32_t minFrames = 1; 3899 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3900 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3901 minFrames = desiredFrames; 3902 } 3903 3904 size_t framesReady = track->framesReady(); 3905 if (ATRACE_ENABLED()) { 3906 // I wish we had formatted trace names 3907 char traceName[16]; 3908 strcpy(traceName, "nRdy"); 3909 int name = track->name(); 3910 if (AudioMixer::TRACK0 <= name && 3911 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3912 name -= AudioMixer::TRACK0; 3913 traceName[4] = (name / 10) + '0'; 3914 traceName[5] = (name % 10) + '0'; 3915 } else { 3916 traceName[4] = '?'; 3917 traceName[5] = '?'; 3918 } 3919 traceName[6] = '\0'; 3920 ATRACE_INT(traceName, framesReady); 3921 } 3922 if ((framesReady >= minFrames) && track->isReady() && 3923 !track->isPaused() && !track->isTerminated()) 3924 { 3925 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3926 3927 mixedTracks++; 3928 3929 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3930 // there is an effect chain connected to the track 3931 chain.clear(); 3932 if (track->mainBuffer() != mSinkBuffer && 3933 track->mainBuffer() != mMixerBuffer) { 3934 if (mEffectBufferEnabled) { 3935 mEffectBufferValid = true; // Later can set directly. 3936 } 3937 chain = getEffectChain_l(track->sessionId()); 3938 // Delegate volume control to effect in track effect chain if needed 3939 if (chain != 0) { 3940 tracksWithEffect++; 3941 } else { 3942 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3943 "session %d", 3944 name, track->sessionId()); 3945 } 3946 } 3947 3948 3949 int param = AudioMixer::VOLUME; 3950 if (track->mFillingUpStatus == Track::FS_FILLED) { 3951 // no ramp for the first volume setting 3952 track->mFillingUpStatus = Track::FS_ACTIVE; 3953 if (track->mState == TrackBase::RESUMING) { 3954 track->mState = TrackBase::ACTIVE; 3955 param = AudioMixer::RAMP_VOLUME; 3956 } 3957 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3958 // FIXME should not make a decision based on mServer 3959 } else if (cblk->mServer != 0) { 3960 // If the track is stopped before the first frame was mixed, 3961 // do not apply ramp 3962 param = AudioMixer::RAMP_VOLUME; 3963 } 3964 3965 // compute volume for this track 3966 uint32_t vl, vr; // in U8.24 integer format 3967 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3968 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3969 vl = vr = 0; 3970 vlf = vrf = vaf = 0.; 3971 if (track->isPausing()) { 3972 track->setPaused(); 3973 } 3974 } else { 3975 3976 // read original volumes with volume control 3977 float typeVolume = mStreamTypes[track->streamType()].volume; 3978 float v = masterVolume * typeVolume; 3979 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3980 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3981 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3982 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3983 // track volumes come from shared memory, so can't be trusted and must be clamped 3984 if (vlf > GAIN_FLOAT_UNITY) { 3985 ALOGV("Track left volume out of range: %.3g", vlf); 3986 vlf = GAIN_FLOAT_UNITY; 3987 } 3988 if (vrf > GAIN_FLOAT_UNITY) { 3989 ALOGV("Track right volume out of range: %.3g", vrf); 3990 vrf = GAIN_FLOAT_UNITY; 3991 } 3992 // now apply the master volume and stream type volume 3993 vlf *= v; 3994 vrf *= v; 3995 // assuming master volume and stream type volume each go up to 1.0, 3996 // then derive vl and vr as U8.24 versions for the effect chain 3997 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3998 vl = (uint32_t) (scaleto8_24 * vlf); 3999 vr = (uint32_t) (scaleto8_24 * vrf); 4000 // vl and vr are now in U8.24 format 4001 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4002 // send level comes from shared memory and so may be corrupt 4003 if (sendLevel > MAX_GAIN_INT) { 4004 ALOGV("Track send level out of range: %04X", sendLevel); 4005 sendLevel = MAX_GAIN_INT; 4006 } 4007 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4008 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4009 } 4010 4011 // Delegate volume control to effect in track effect chain if needed 4012 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4013 // Do not ramp volume if volume is controlled by effect 4014 param = AudioMixer::VOLUME; 4015 // Update remaining floating point volume levels 4016 vlf = (float)vl / (1 << 24); 4017 vrf = (float)vr / (1 << 24); 4018 track->mHasVolumeController = true; 4019 } else { 4020 // force no volume ramp when volume controller was just disabled or removed 4021 // from effect chain to avoid volume spike 4022 if (track->mHasVolumeController) { 4023 param = AudioMixer::VOLUME; 4024 } 4025 track->mHasVolumeController = false; 4026 } 4027 4028 // XXX: these things DON'T need to be done each time 4029 mAudioMixer->setBufferProvider(name, track); 4030 mAudioMixer->enable(name); 4031 4032 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4033 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4034 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4035 mAudioMixer->setParameter( 4036 name, 4037 AudioMixer::TRACK, 4038 AudioMixer::FORMAT, (void *)track->format()); 4039 mAudioMixer->setParameter( 4040 name, 4041 AudioMixer::TRACK, 4042 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4043 mAudioMixer->setParameter( 4044 name, 4045 AudioMixer::TRACK, 4046 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4047 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4048 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4049 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4050 if (reqSampleRate == 0) { 4051 reqSampleRate = mSampleRate; 4052 } else if (reqSampleRate > maxSampleRate) { 4053 reqSampleRate = maxSampleRate; 4054 } 4055 mAudioMixer->setParameter( 4056 name, 4057 AudioMixer::RESAMPLE, 4058 AudioMixer::SAMPLE_RATE, 4059 (void *)(uintptr_t)reqSampleRate); 4060 4061 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4062 mAudioMixer->setParameter( 4063 name, 4064 AudioMixer::TIMESTRETCH, 4065 AudioMixer::PLAYBACK_RATE, 4066 &playbackRate); 4067 4068 /* 4069 * Select the appropriate output buffer for the track. 4070 * 4071 * Tracks with effects go into their own effects chain buffer 4072 * and from there into either mEffectBuffer or mSinkBuffer. 4073 * 4074 * Other tracks can use mMixerBuffer for higher precision 4075 * channel accumulation. If this buffer is enabled 4076 * (mMixerBufferEnabled true), then selected tracks will accumulate 4077 * into it. 4078 * 4079 */ 4080 if (mMixerBufferEnabled 4081 && (track->mainBuffer() == mSinkBuffer 4082 || track->mainBuffer() == mMixerBuffer)) { 4083 mAudioMixer->setParameter( 4084 name, 4085 AudioMixer::TRACK, 4086 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4087 mAudioMixer->setParameter( 4088 name, 4089 AudioMixer::TRACK, 4090 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4091 // TODO: override track->mainBuffer()? 4092 mMixerBufferValid = true; 4093 } else { 4094 mAudioMixer->setParameter( 4095 name, 4096 AudioMixer::TRACK, 4097 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4098 mAudioMixer->setParameter( 4099 name, 4100 AudioMixer::TRACK, 4101 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4102 } 4103 mAudioMixer->setParameter( 4104 name, 4105 AudioMixer::TRACK, 4106 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4107 4108 // reset retry count 4109 track->mRetryCount = kMaxTrackRetries; 4110 4111 // If one track is ready, set the mixer ready if: 4112 // - the mixer was not ready during previous round OR 4113 // - no other track is not ready 4114 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4115 mixerStatus != MIXER_TRACKS_ENABLED) { 4116 mixerStatus = MIXER_TRACKS_READY; 4117 } 4118 } else { 4119 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4120 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4121 track, framesReady, desiredFrames); 4122 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4123 } 4124 // clear effect chain input buffer if an active track underruns to avoid sending 4125 // previous audio buffer again to effects 4126 chain = getEffectChain_l(track->sessionId()); 4127 if (chain != 0) { 4128 chain->clearInputBuffer(); 4129 } 4130 4131 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4132 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4133 track->isStopped() || track->isPaused()) { 4134 // We have consumed all the buffers of this track. 4135 // Remove it from the list of active tracks. 4136 // TODO: use actual buffer filling status instead of latency when available from 4137 // audio HAL 4138 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4139 size_t framesWritten = mBytesWritten / mFrameSize; 4140 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4141 if (track->isStopped()) { 4142 track->reset(); 4143 } 4144 tracksToRemove->add(track); 4145 } 4146 } else { 4147 // No buffers for this track. Give it a few chances to 4148 // fill a buffer, then remove it from active list. 4149 if (--(track->mRetryCount) <= 0) { 4150 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4151 tracksToRemove->add(track); 4152 // indicate to client process that the track was disabled because of underrun; 4153 // it will then automatically call start() when data is available 4154 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4155 // If one track is not ready, mark the mixer also not ready if: 4156 // - the mixer was ready during previous round OR 4157 // - no other track is ready 4158 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4159 mixerStatus != MIXER_TRACKS_READY) { 4160 mixerStatus = MIXER_TRACKS_ENABLED; 4161 } 4162 } 4163 mAudioMixer->disable(name); 4164 } 4165 4166 } // local variable scope to avoid goto warning 4167track_is_ready: ; 4168 4169 } 4170 4171 // Push the new FastMixer state if necessary 4172 bool pauseAudioWatchdog = false; 4173 if (didModify) { 4174 state->mFastTracksGen++; 4175 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4176 if (kUseFastMixer == FastMixer_Dynamic && 4177 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4178 state->mCommand = FastMixerState::COLD_IDLE; 4179 state->mColdFutexAddr = &mFastMixerFutex; 4180 state->mColdGen++; 4181 mFastMixerFutex = 0; 4182 if (kUseFastMixer == FastMixer_Dynamic) { 4183 mNormalSink = mOutputSink; 4184 } 4185 // If we go into cold idle, need to wait for acknowledgement 4186 // so that fast mixer stops doing I/O. 4187 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4188 pauseAudioWatchdog = true; 4189 } 4190 } 4191 if (sq != NULL) { 4192 sq->end(didModify); 4193 sq->push(block); 4194 } 4195#ifdef AUDIO_WATCHDOG 4196 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4197 mAudioWatchdog->pause(); 4198 } 4199#endif 4200 4201 // Now perform the deferred reset on fast tracks that have stopped 4202 while (resetMask != 0) { 4203 size_t i = __builtin_ctz(resetMask); 4204 ALOG_ASSERT(i < count); 4205 resetMask &= ~(1 << i); 4206 sp<Track> t = mActiveTracks[i].promote(); 4207 if (t == 0) { 4208 continue; 4209 } 4210 Track* track = t.get(); 4211 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4212 track->reset(); 4213 } 4214 4215 // remove all the tracks that need to be... 4216 removeTracks_l(*tracksToRemove); 4217 4218 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4219 mEffectBufferValid = true; 4220 } 4221 4222 if (mEffectBufferValid) { 4223 // as long as there are effects we should clear the effects buffer, to avoid 4224 // passing a non-clean buffer to the effect chain 4225 memset(mEffectBuffer, 0, mEffectBufferSize); 4226 } 4227 // sink or mix buffer must be cleared if all tracks are connected to an 4228 // effect chain as in this case the mixer will not write to the sink or mix buffer 4229 // and track effects will accumulate into it 4230 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4231 (mixedTracks == 0 && fastTracks > 0))) { 4232 // FIXME as a performance optimization, should remember previous zero status 4233 if (mMixerBufferValid) { 4234 memset(mMixerBuffer, 0, mMixerBufferSize); 4235 // TODO: In testing, mSinkBuffer below need not be cleared because 4236 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4237 // after mixing. 4238 // 4239 // To enforce this guarantee: 4240 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4241 // (mixedTracks == 0 && fastTracks > 0)) 4242 // must imply MIXER_TRACKS_READY. 4243 // Later, we may clear buffers regardless, and skip much of this logic. 4244 } 4245 // FIXME as a performance optimization, should remember previous zero status 4246 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4247 } 4248 4249 // if any fast tracks, then status is ready 4250 mMixerStatusIgnoringFastTracks = mixerStatus; 4251 if (fastTracks > 0) { 4252 mixerStatus = MIXER_TRACKS_READY; 4253 } 4254 return mixerStatus; 4255} 4256 4257// getTrackName_l() must be called with ThreadBase::mLock held 4258int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4259 audio_format_t format, int sessionId) 4260{ 4261 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4262} 4263 4264// deleteTrackName_l() must be called with ThreadBase::mLock held 4265void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4266{ 4267 ALOGV("remove track (%d) and delete from mixer", name); 4268 mAudioMixer->deleteTrackName(name); 4269} 4270 4271// checkForNewParameter_l() must be called with ThreadBase::mLock held 4272bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4273 status_t& status) 4274{ 4275 bool reconfig = false; 4276 4277 status = NO_ERROR; 4278 4279 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4280 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4281 if (mFastMixer != 0) { 4282 FastMixerStateQueue *sq = mFastMixer->sq(); 4283 FastMixerState *state = sq->begin(); 4284 if (!(state->mCommand & FastMixerState::IDLE)) { 4285 previousCommand = state->mCommand; 4286 state->mCommand = FastMixerState::HOT_IDLE; 4287 sq->end(); 4288 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4289 } else { 4290 sq->end(false /*didModify*/); 4291 } 4292 } 4293 4294 AudioParameter param = AudioParameter(keyValuePair); 4295 int value; 4296 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4297 reconfig = true; 4298 } 4299 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4300 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4301 status = BAD_VALUE; 4302 } else { 4303 // no need to save value, since it's constant 4304 reconfig = true; 4305 } 4306 } 4307 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4308 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4309 status = BAD_VALUE; 4310 } else { 4311 // no need to save value, since it's constant 4312 reconfig = true; 4313 } 4314 } 4315 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4316 // do not accept frame count changes if tracks are open as the track buffer 4317 // size depends on frame count and correct behavior would not be guaranteed 4318 // if frame count is changed after track creation 4319 if (!mTracks.isEmpty()) { 4320 status = INVALID_OPERATION; 4321 } else { 4322 reconfig = true; 4323 } 4324 } 4325 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4326#ifdef ADD_BATTERY_DATA 4327 // when changing the audio output device, call addBatteryData to notify 4328 // the change 4329 if (mOutDevice != value) { 4330 uint32_t params = 0; 4331 // check whether speaker is on 4332 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4333 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4334 } 4335 4336 audio_devices_t deviceWithoutSpeaker 4337 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4338 // check if any other device (except speaker) is on 4339 if (value & deviceWithoutSpeaker) { 4340 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4341 } 4342 4343 if (params != 0) { 4344 addBatteryData(params); 4345 } 4346 } 4347#endif 4348 4349 // forward device change to effects that have requested to be 4350 // aware of attached audio device. 4351 if (value != AUDIO_DEVICE_NONE) { 4352 mOutDevice = value; 4353 for (size_t i = 0; i < mEffectChains.size(); i++) { 4354 mEffectChains[i]->setDevice_l(mOutDevice); 4355 } 4356 } 4357 } 4358 4359 if (status == NO_ERROR) { 4360 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4361 keyValuePair.string()); 4362 if (!mStandby && status == INVALID_OPERATION) { 4363 mOutput->standby(); 4364 mStandby = true; 4365 mBytesWritten = 0; 4366 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4367 keyValuePair.string()); 4368 } 4369 if (status == NO_ERROR && reconfig) { 4370 readOutputParameters_l(); 4371 delete mAudioMixer; 4372 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4373 for (size_t i = 0; i < mTracks.size() ; i++) { 4374 int name = getTrackName_l(mTracks[i]->mChannelMask, 4375 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4376 if (name < 0) { 4377 break; 4378 } 4379 mTracks[i]->mName = name; 4380 } 4381 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4382 } 4383 } 4384 4385 if (!(previousCommand & FastMixerState::IDLE)) { 4386 ALOG_ASSERT(mFastMixer != 0); 4387 FastMixerStateQueue *sq = mFastMixer->sq(); 4388 FastMixerState *state = sq->begin(); 4389 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4390 state->mCommand = previousCommand; 4391 sq->end(); 4392 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4393 } 4394 4395 return reconfig; 4396} 4397 4398 4399void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4400{ 4401 const size_t SIZE = 256; 4402 char buffer[SIZE]; 4403 String8 result; 4404 4405 PlaybackThread::dumpInternals(fd, args); 4406 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4407 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4408 4409 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4410 // while we are dumping it. It may be inconsistent, but it won't mutate! 4411 // This is a large object so we place it on the heap. 4412 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4413 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4414 copy->dump(fd); 4415 delete copy; 4416 4417#ifdef STATE_QUEUE_DUMP 4418 // Similar for state queue 4419 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4420 observerCopy.dump(fd); 4421 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4422 mutatorCopy.dump(fd); 4423#endif 4424 4425#ifdef TEE_SINK 4426 // Write the tee output to a .wav file 4427 dumpTee(fd, mTeeSource, mId); 4428#endif 4429 4430#ifdef AUDIO_WATCHDOG 4431 if (mAudioWatchdog != 0) { 4432 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4433 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4434 wdCopy.dump(fd); 4435 } 4436#endif 4437} 4438 4439uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4440{ 4441 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4442} 4443 4444uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4445{ 4446 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4447} 4448 4449void AudioFlinger::MixerThread::cacheParameters_l() 4450{ 4451 PlaybackThread::cacheParameters_l(); 4452 4453 // FIXME: Relaxed timing because of a certain device that can't meet latency 4454 // Should be reduced to 2x after the vendor fixes the driver issue 4455 // increase threshold again due to low power audio mode. The way this warning 4456 // threshold is calculated and its usefulness should be reconsidered anyway. 4457 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4458} 4459 4460// ---------------------------------------------------------------------------- 4461 4462AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4463 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4464 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4465 // mLeftVolFloat, mRightVolFloat 4466{ 4467} 4468 4469AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4470 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4471 ThreadBase::type_t type, bool systemReady) 4472 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4473 // mLeftVolFloat, mRightVolFloat 4474{ 4475} 4476 4477AudioFlinger::DirectOutputThread::~DirectOutputThread() 4478{ 4479} 4480 4481void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4482{ 4483 audio_track_cblk_t* cblk = track->cblk(); 4484 float left, right; 4485 4486 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4487 left = right = 0; 4488 } else { 4489 float typeVolume = mStreamTypes[track->streamType()].volume; 4490 float v = mMasterVolume * typeVolume; 4491 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4492 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4493 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4494 if (left > GAIN_FLOAT_UNITY) { 4495 left = GAIN_FLOAT_UNITY; 4496 } 4497 left *= v; 4498 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4499 if (right > GAIN_FLOAT_UNITY) { 4500 right = GAIN_FLOAT_UNITY; 4501 } 4502 right *= v; 4503 } 4504 4505 if (lastTrack) { 4506 if (left != mLeftVolFloat || right != mRightVolFloat) { 4507 mLeftVolFloat = left; 4508 mRightVolFloat = right; 4509 4510 // Convert volumes from float to 8.24 4511 uint32_t vl = (uint32_t)(left * (1 << 24)); 4512 uint32_t vr = (uint32_t)(right * (1 << 24)); 4513 4514 // Delegate volume control to effect in track effect chain if needed 4515 // only one effect chain can be present on DirectOutputThread, so if 4516 // there is one, the track is connected to it 4517 if (!mEffectChains.isEmpty()) { 4518 mEffectChains[0]->setVolume_l(&vl, &vr); 4519 left = (float)vl / (1 << 24); 4520 right = (float)vr / (1 << 24); 4521 } 4522 if (mOutput->stream->set_volume) { 4523 mOutput->stream->set_volume(mOutput->stream, left, right); 4524 } 4525 } 4526 } 4527} 4528 4529void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4530{ 4531 sp<Track> previousTrack = mPreviousTrack.promote(); 4532 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4533 4534 if (previousTrack != 0 && latestTrack != 0) { 4535 if (mType == DIRECT) { 4536 if (previousTrack.get() != latestTrack.get()) { 4537 mFlushPending = true; 4538 } 4539 } else /* mType == OFFLOAD */ { 4540 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4541 mFlushPending = true; 4542 } 4543 } 4544 } 4545 PlaybackThread::onAddNewTrack_l(); 4546} 4547 4548AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4549 Vector< sp<Track> > *tracksToRemove 4550) 4551{ 4552 size_t count = mActiveTracks.size(); 4553 mixer_state mixerStatus = MIXER_IDLE; 4554 bool doHwPause = false; 4555 bool doHwResume = false; 4556 4557 // find out which tracks need to be processed 4558 for (size_t i = 0; i < count; i++) { 4559 sp<Track> t = mActiveTracks[i].promote(); 4560 // The track died recently 4561 if (t == 0) { 4562 continue; 4563 } 4564 4565 if (t->isInvalid()) { 4566 ALOGW("An invalidated track shouldn't be in active list"); 4567 tracksToRemove->add(t); 4568 continue; 4569 } 4570 4571 Track* const track = t.get(); 4572 audio_track_cblk_t* cblk = track->cblk(); 4573 // Only consider last track started for volume and mixer state control. 4574 // In theory an older track could underrun and restart after the new one starts 4575 // but as we only care about the transition phase between two tracks on a 4576 // direct output, it is not a problem to ignore the underrun case. 4577 sp<Track> l = mLatestActiveTrack.promote(); 4578 bool last = l.get() == track; 4579 4580 if (track->isPausing()) { 4581 track->setPaused(); 4582 if (mHwSupportsPause && last && !mHwPaused) { 4583 doHwPause = true; 4584 mHwPaused = true; 4585 } 4586 tracksToRemove->add(track); 4587 } else if (track->isFlushPending()) { 4588 track->flushAck(); 4589 if (last) { 4590 mFlushPending = true; 4591 } 4592 } else if (track->isResumePending()) { 4593 track->resumeAck(); 4594 if (last && mHwPaused) { 4595 doHwResume = true; 4596 mHwPaused = false; 4597 } 4598 } 4599 4600 // The first time a track is added we wait 4601 // for all its buffers to be filled before processing it. 4602 // Allow draining the buffer in case the client 4603 // app does not call stop() and relies on underrun to stop: 4604 // hence the test on (track->mRetryCount > 1). 4605 // If retryCount<=1 then track is about to underrun and be removed. 4606 // Do not use a high threshold for compressed audio. 4607 uint32_t minFrames; 4608 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4609 && (track->mRetryCount > 1) && audio_is_linear_pcm(mFormat)) { 4610 minFrames = mNormalFrameCount; 4611 } else { 4612 minFrames = 1; 4613 } 4614 4615 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4616 !track->isStopping_2() && !track->isStopped()) 4617 { 4618 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4619 4620 if (track->mFillingUpStatus == Track::FS_FILLED) { 4621 track->mFillingUpStatus = Track::FS_ACTIVE; 4622 // make sure processVolume_l() will apply new volume even if 0 4623 mLeftVolFloat = mRightVolFloat = -1.0; 4624 if (!mHwSupportsPause) { 4625 track->resumeAck(); 4626 } 4627 } 4628 4629 // compute volume for this track 4630 processVolume_l(track, last); 4631 if (last) { 4632 sp<Track> previousTrack = mPreviousTrack.promote(); 4633 if (previousTrack != 0) { 4634 if (track != previousTrack.get()) { 4635 // Flush any data still being written from last track 4636 mBytesRemaining = 0; 4637 // Invalidate previous track to force a seek when resuming. 4638 previousTrack->invalidate(); 4639 } 4640 } 4641 mPreviousTrack = track; 4642 4643 // reset retry count 4644 track->mRetryCount = kMaxTrackRetriesDirect; 4645 mActiveTrack = t; 4646 mixerStatus = MIXER_TRACKS_READY; 4647 if (mHwPaused) { 4648 doHwResume = true; 4649 mHwPaused = false; 4650 } 4651 } 4652 } else { 4653 // clear effect chain input buffer if the last active track started underruns 4654 // to avoid sending previous audio buffer again to effects 4655 if (!mEffectChains.isEmpty() && last) { 4656 mEffectChains[0]->clearInputBuffer(); 4657 } 4658 if (track->isStopping_1()) { 4659 track->mState = TrackBase::STOPPING_2; 4660 if (last && mHwPaused) { 4661 doHwResume = true; 4662 mHwPaused = false; 4663 } 4664 } 4665 if ((track->sharedBuffer() != 0) || track->isStopped() || 4666 track->isStopping_2() || track->isPaused()) { 4667 // We have consumed all the buffers of this track. 4668 // Remove it from the list of active tracks. 4669 size_t audioHALFrames; 4670 if (audio_is_linear_pcm(mFormat)) { 4671 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4672 } else { 4673 audioHALFrames = 0; 4674 } 4675 4676 size_t framesWritten = mBytesWritten / mFrameSize; 4677 if (mStandby || !last || 4678 track->presentationComplete(framesWritten, audioHALFrames)) { 4679 if (track->isStopping_2()) { 4680 track->mState = TrackBase::STOPPED; 4681 } 4682 if (track->isStopped()) { 4683 track->reset(); 4684 } 4685 tracksToRemove->add(track); 4686 } 4687 } else { 4688 // No buffers for this track. Give it a few chances to 4689 // fill a buffer, then remove it from active list. 4690 // Only consider last track started for mixer state control 4691 if (--(track->mRetryCount) <= 0) { 4692 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4693 tracksToRemove->add(track); 4694 // indicate to client process that the track was disabled because of underrun; 4695 // it will then automatically call start() when data is available 4696 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4697 } else if (last) { 4698 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4699 "minFrames = %u, mFormat = %#x", 4700 track->framesReady(), minFrames, mFormat); 4701 mixerStatus = MIXER_TRACKS_ENABLED; 4702 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4703 doHwPause = true; 4704 mHwPaused = true; 4705 } 4706 } 4707 } 4708 } 4709 } 4710 4711 // if an active track did not command a flush, check for pending flush on stopped tracks 4712 if (!mFlushPending) { 4713 for (size_t i = 0; i < mTracks.size(); i++) { 4714 if (mTracks[i]->isFlushPending()) { 4715 mTracks[i]->flushAck(); 4716 mFlushPending = true; 4717 } 4718 } 4719 } 4720 4721 // make sure the pause/flush/resume sequence is executed in the right order. 4722 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4723 // before flush and then resume HW. This can happen in case of pause/flush/resume 4724 // if resume is received before pause is executed. 4725 if (mHwSupportsPause && !mStandby && 4726 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4727 mOutput->stream->pause(mOutput->stream); 4728 } 4729 if (mFlushPending) { 4730 flushHw_l(); 4731 } 4732 if (mHwSupportsPause && !mStandby && doHwResume) { 4733 mOutput->stream->resume(mOutput->stream); 4734 } 4735 // remove all the tracks that need to be... 4736 removeTracks_l(*tracksToRemove); 4737 4738 return mixerStatus; 4739} 4740 4741void AudioFlinger::DirectOutputThread::threadLoop_mix() 4742{ 4743 size_t frameCount = mFrameCount; 4744 int8_t *curBuf = (int8_t *)mSinkBuffer; 4745 // output audio to hardware 4746 while (frameCount) { 4747 AudioBufferProvider::Buffer buffer; 4748 buffer.frameCount = frameCount; 4749 status_t status = mActiveTrack->getNextBuffer(&buffer); 4750 if (status != NO_ERROR || buffer.raw == NULL) { 4751 memset(curBuf, 0, frameCount * mFrameSize); 4752 break; 4753 } 4754 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4755 frameCount -= buffer.frameCount; 4756 curBuf += buffer.frameCount * mFrameSize; 4757 mActiveTrack->releaseBuffer(&buffer); 4758 } 4759 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4760 mSleepTimeUs = 0; 4761 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4762 mActiveTrack.clear(); 4763} 4764 4765void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4766{ 4767 // do not write to HAL when paused 4768 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4769 mSleepTimeUs = mIdleSleepTimeUs; 4770 return; 4771 } 4772 if (mSleepTimeUs == 0) { 4773 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4774 mSleepTimeUs = mActiveSleepTimeUs; 4775 } else { 4776 mSleepTimeUs = mIdleSleepTimeUs; 4777 } 4778 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4779 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4780 mSleepTimeUs = 0; 4781 } 4782} 4783 4784void AudioFlinger::DirectOutputThread::threadLoop_exit() 4785{ 4786 { 4787 Mutex::Autolock _l(mLock); 4788 for (size_t i = 0; i < mTracks.size(); i++) { 4789 if (mTracks[i]->isFlushPending()) { 4790 mTracks[i]->flushAck(); 4791 mFlushPending = true; 4792 } 4793 } 4794 if (mFlushPending) { 4795 flushHw_l(); 4796 } 4797 } 4798 PlaybackThread::threadLoop_exit(); 4799} 4800 4801// must be called with thread mutex locked 4802bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4803{ 4804 bool trackPaused = false; 4805 bool trackStopped = false; 4806 4807 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4808 // after a timeout and we will enter standby then. 4809 if (mTracks.size() > 0) { 4810 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4811 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4812 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4813 } 4814 4815 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4816} 4817 4818// getTrackName_l() must be called with ThreadBase::mLock held 4819int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4820 audio_format_t format __unused, int sessionId __unused) 4821{ 4822 return 0; 4823} 4824 4825// deleteTrackName_l() must be called with ThreadBase::mLock held 4826void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4827{ 4828} 4829 4830// checkForNewParameter_l() must be called with ThreadBase::mLock held 4831bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4832 status_t& status) 4833{ 4834 bool reconfig = false; 4835 4836 status = NO_ERROR; 4837 4838 AudioParameter param = AudioParameter(keyValuePair); 4839 int value; 4840 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4841 // forward device change to effects that have requested to be 4842 // aware of attached audio device. 4843 if (value != AUDIO_DEVICE_NONE) { 4844 mOutDevice = value; 4845 for (size_t i = 0; i < mEffectChains.size(); i++) { 4846 mEffectChains[i]->setDevice_l(mOutDevice); 4847 } 4848 } 4849 } 4850 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4851 // do not accept frame count changes if tracks are open as the track buffer 4852 // size depends on frame count and correct behavior would not be garantied 4853 // if frame count is changed after track creation 4854 if (!mTracks.isEmpty()) { 4855 status = INVALID_OPERATION; 4856 } else { 4857 reconfig = true; 4858 } 4859 } 4860 if (status == NO_ERROR) { 4861 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4862 keyValuePair.string()); 4863 if (!mStandby && status == INVALID_OPERATION) { 4864 mOutput->standby(); 4865 mStandby = true; 4866 mBytesWritten = 0; 4867 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4868 keyValuePair.string()); 4869 } 4870 if (status == NO_ERROR && reconfig) { 4871 readOutputParameters_l(); 4872 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4873 } 4874 } 4875 4876 return reconfig; 4877} 4878 4879uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4880{ 4881 uint32_t time; 4882 if (audio_is_linear_pcm(mFormat)) { 4883 time = PlaybackThread::activeSleepTimeUs(); 4884 } else { 4885 time = 10000; 4886 } 4887 return time; 4888} 4889 4890uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4891{ 4892 uint32_t time; 4893 if (audio_is_linear_pcm(mFormat)) { 4894 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4895 } else { 4896 time = 10000; 4897 } 4898 return time; 4899} 4900 4901uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4902{ 4903 uint32_t time; 4904 if (audio_is_linear_pcm(mFormat)) { 4905 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4906 } else { 4907 time = 10000; 4908 } 4909 return time; 4910} 4911 4912void AudioFlinger::DirectOutputThread::cacheParameters_l() 4913{ 4914 PlaybackThread::cacheParameters_l(); 4915 4916 // use shorter standby delay as on normal output to release 4917 // hardware resources as soon as possible 4918 // no delay on outputs with HW A/V sync 4919 if (usesHwAvSync()) { 4920 mStandbyDelayNs = 0; 4921 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 4922 mStandbyDelayNs = kOffloadStandbyDelayNs; 4923 } else { 4924 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 4925 } 4926} 4927 4928void AudioFlinger::DirectOutputThread::flushHw_l() 4929{ 4930 mOutput->flush(); 4931 mHwPaused = false; 4932 mFlushPending = false; 4933} 4934 4935// ---------------------------------------------------------------------------- 4936 4937AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4938 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4939 : Thread(false /*canCallJava*/), 4940 mPlaybackThread(playbackThread), 4941 mWriteAckSequence(0), 4942 mDrainSequence(0) 4943{ 4944} 4945 4946AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4947{ 4948} 4949 4950void AudioFlinger::AsyncCallbackThread::onFirstRef() 4951{ 4952 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4953} 4954 4955bool AudioFlinger::AsyncCallbackThread::threadLoop() 4956{ 4957 while (!exitPending()) { 4958 uint32_t writeAckSequence; 4959 uint32_t drainSequence; 4960 4961 { 4962 Mutex::Autolock _l(mLock); 4963 while (!((mWriteAckSequence & 1) || 4964 (mDrainSequence & 1) || 4965 exitPending())) { 4966 mWaitWorkCV.wait(mLock); 4967 } 4968 4969 if (exitPending()) { 4970 break; 4971 } 4972 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4973 mWriteAckSequence, mDrainSequence); 4974 writeAckSequence = mWriteAckSequence; 4975 mWriteAckSequence &= ~1; 4976 drainSequence = mDrainSequence; 4977 mDrainSequence &= ~1; 4978 } 4979 { 4980 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4981 if (playbackThread != 0) { 4982 if (writeAckSequence & 1) { 4983 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4984 } 4985 if (drainSequence & 1) { 4986 playbackThread->resetDraining(drainSequence >> 1); 4987 } 4988 } 4989 } 4990 } 4991 return false; 4992} 4993 4994void AudioFlinger::AsyncCallbackThread::exit() 4995{ 4996 ALOGV("AsyncCallbackThread::exit"); 4997 Mutex::Autolock _l(mLock); 4998 requestExit(); 4999 mWaitWorkCV.broadcast(); 5000} 5001 5002void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5003{ 5004 Mutex::Autolock _l(mLock); 5005 // bit 0 is cleared 5006 mWriteAckSequence = sequence << 1; 5007} 5008 5009void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5010{ 5011 Mutex::Autolock _l(mLock); 5012 // ignore unexpected callbacks 5013 if (mWriteAckSequence & 2) { 5014 mWriteAckSequence |= 1; 5015 mWaitWorkCV.signal(); 5016 } 5017} 5018 5019void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5020{ 5021 Mutex::Autolock _l(mLock); 5022 // bit 0 is cleared 5023 mDrainSequence = sequence << 1; 5024} 5025 5026void AudioFlinger::AsyncCallbackThread::resetDraining() 5027{ 5028 Mutex::Autolock _l(mLock); 5029 // ignore unexpected callbacks 5030 if (mDrainSequence & 2) { 5031 mDrainSequence |= 1; 5032 mWaitWorkCV.signal(); 5033 } 5034} 5035 5036 5037// ---------------------------------------------------------------------------- 5038AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5039 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5040 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5041 mPausedBytesRemaining(0) 5042{ 5043 //FIXME: mStandby should be set to true by ThreadBase constructor 5044 mStandby = true; 5045} 5046 5047void AudioFlinger::OffloadThread::threadLoop_exit() 5048{ 5049 if (mFlushPending || mHwPaused) { 5050 // If a flush is pending or track was paused, just discard buffered data 5051 flushHw_l(); 5052 } else { 5053 mMixerStatus = MIXER_DRAIN_ALL; 5054 threadLoop_drain(); 5055 } 5056 if (mUseAsyncWrite) { 5057 ALOG_ASSERT(mCallbackThread != 0); 5058 mCallbackThread->exit(); 5059 } 5060 PlaybackThread::threadLoop_exit(); 5061} 5062 5063AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5064 Vector< sp<Track> > *tracksToRemove 5065) 5066{ 5067 size_t count = mActiveTracks.size(); 5068 5069 mixer_state mixerStatus = MIXER_IDLE; 5070 bool doHwPause = false; 5071 bool doHwResume = false; 5072 5073 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5074 5075 // find out which tracks need to be processed 5076 for (size_t i = 0; i < count; i++) { 5077 sp<Track> t = mActiveTracks[i].promote(); 5078 // The track died recently 5079 if (t == 0) { 5080 continue; 5081 } 5082 Track* const track = t.get(); 5083 audio_track_cblk_t* cblk = track->cblk(); 5084 // Only consider last track started for volume and mixer state control. 5085 // In theory an older track could underrun and restart after the new one starts 5086 // but as we only care about the transition phase between two tracks on a 5087 // direct output, it is not a problem to ignore the underrun case. 5088 sp<Track> l = mLatestActiveTrack.promote(); 5089 bool last = l.get() == track; 5090 5091 if (track->isInvalid()) { 5092 ALOGW("An invalidated track shouldn't be in active list"); 5093 tracksToRemove->add(track); 5094 continue; 5095 } 5096 5097 if (track->mState == TrackBase::IDLE) { 5098 ALOGW("An idle track shouldn't be in active list"); 5099 continue; 5100 } 5101 5102 if (track->isPausing()) { 5103 track->setPaused(); 5104 if (last) { 5105 if (mHwSupportsPause && !mHwPaused) { 5106 doHwPause = true; 5107 mHwPaused = true; 5108 } 5109 // If we were part way through writing the mixbuffer to 5110 // the HAL we must save this until we resume 5111 // BUG - this will be wrong if a different track is made active, 5112 // in that case we want to discard the pending data in the 5113 // mixbuffer and tell the client to present it again when the 5114 // track is resumed 5115 mPausedWriteLength = mCurrentWriteLength; 5116 mPausedBytesRemaining = mBytesRemaining; 5117 mBytesRemaining = 0; // stop writing 5118 } 5119 tracksToRemove->add(track); 5120 } else if (track->isFlushPending()) { 5121 track->flushAck(); 5122 if (last) { 5123 mFlushPending = true; 5124 } 5125 } else if (track->isResumePending()){ 5126 track->resumeAck(); 5127 if (last) { 5128 if (mPausedBytesRemaining) { 5129 // Need to continue write that was interrupted 5130 mCurrentWriteLength = mPausedWriteLength; 5131 mBytesRemaining = mPausedBytesRemaining; 5132 mPausedBytesRemaining = 0; 5133 } 5134 if (mHwPaused) { 5135 doHwResume = true; 5136 mHwPaused = false; 5137 // threadLoop_mix() will handle the case that we need to 5138 // resume an interrupted write 5139 } 5140 // enable write to audio HAL 5141 mSleepTimeUs = 0; 5142 5143 // Do not handle new data in this iteration even if track->framesReady() 5144 mixerStatus = MIXER_TRACKS_ENABLED; 5145 } 5146 } else if (track->framesReady() && track->isReady() && 5147 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5148 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5149 if (track->mFillingUpStatus == Track::FS_FILLED) { 5150 track->mFillingUpStatus = Track::FS_ACTIVE; 5151 // make sure processVolume_l() will apply new volume even if 0 5152 mLeftVolFloat = mRightVolFloat = -1.0; 5153 } 5154 5155 if (last) { 5156 sp<Track> previousTrack = mPreviousTrack.promote(); 5157 if (previousTrack != 0) { 5158 if (track != previousTrack.get()) { 5159 // Flush any data still being written from last track 5160 mBytesRemaining = 0; 5161 if (mPausedBytesRemaining) { 5162 // Last track was paused so we also need to flush saved 5163 // mixbuffer state and invalidate track so that it will 5164 // re-submit that unwritten data when it is next resumed 5165 mPausedBytesRemaining = 0; 5166 // Invalidate is a bit drastic - would be more efficient 5167 // to have a flag to tell client that some of the 5168 // previously written data was lost 5169 previousTrack->invalidate(); 5170 } 5171 // flush data already sent to the DSP if changing audio session as audio 5172 // comes from a different source. Also invalidate previous track to force a 5173 // seek when resuming. 5174 if (previousTrack->sessionId() != track->sessionId()) { 5175 previousTrack->invalidate(); 5176 } 5177 } 5178 } 5179 mPreviousTrack = track; 5180 // reset retry count 5181 track->mRetryCount = kMaxTrackRetriesOffload; 5182 mActiveTrack = t; 5183 mixerStatus = MIXER_TRACKS_READY; 5184 } 5185 } else { 5186 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5187 if (track->isStopping_1()) { 5188 // Hardware buffer can hold a large amount of audio so we must 5189 // wait for all current track's data to drain before we say 5190 // that the track is stopped. 5191 if (mBytesRemaining == 0) { 5192 // Only start draining when all data in mixbuffer 5193 // has been written 5194 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5195 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5196 // do not drain if no data was ever sent to HAL (mStandby == true) 5197 if (last && !mStandby) { 5198 // do not modify drain sequence if we are already draining. This happens 5199 // when resuming from pause after drain. 5200 if ((mDrainSequence & 1) == 0) { 5201 mSleepTimeUs = 0; 5202 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5203 mixerStatus = MIXER_DRAIN_TRACK; 5204 mDrainSequence += 2; 5205 } 5206 if (mHwPaused) { 5207 // It is possible to move from PAUSED to STOPPING_1 without 5208 // a resume so we must ensure hardware is running 5209 doHwResume = true; 5210 mHwPaused = false; 5211 } 5212 } 5213 } 5214 } else if (track->isStopping_2()) { 5215 // Drain has completed or we are in standby, signal presentation complete 5216 if (!(mDrainSequence & 1) || !last || mStandby) { 5217 track->mState = TrackBase::STOPPED; 5218 size_t audioHALFrames = 5219 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5220 size_t framesWritten = 5221 mBytesWritten / mOutput->getFrameSize(); 5222 track->presentationComplete(framesWritten, audioHALFrames); 5223 track->reset(); 5224 tracksToRemove->add(track); 5225 } 5226 } else { 5227 // No buffers for this track. Give it a few chances to 5228 // fill a buffer, then remove it from active list. 5229 if (--(track->mRetryCount) <= 0) { 5230 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5231 track->name()); 5232 tracksToRemove->add(track); 5233 // indicate to client process that the track was disabled because of underrun; 5234 // it will then automatically call start() when data is available 5235 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5236 } else if (last){ 5237 mixerStatus = MIXER_TRACKS_ENABLED; 5238 } 5239 } 5240 } 5241 // compute volume for this track 5242 processVolume_l(track, last); 5243 } 5244 5245 // make sure the pause/flush/resume sequence is executed in the right order. 5246 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5247 // before flush and then resume HW. This can happen in case of pause/flush/resume 5248 // if resume is received before pause is executed. 5249 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5250 mOutput->stream->pause(mOutput->stream); 5251 } 5252 if (mFlushPending) { 5253 flushHw_l(); 5254 } 5255 if (!mStandby && doHwResume) { 5256 mOutput->stream->resume(mOutput->stream); 5257 } 5258 5259 // remove all the tracks that need to be... 5260 removeTracks_l(*tracksToRemove); 5261 5262 return mixerStatus; 5263} 5264 5265// must be called with thread mutex locked 5266bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5267{ 5268 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5269 mWriteAckSequence, mDrainSequence); 5270 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5271 return true; 5272 } 5273 return false; 5274} 5275 5276bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5277{ 5278 Mutex::Autolock _l(mLock); 5279 return waitingAsyncCallback_l(); 5280} 5281 5282void AudioFlinger::OffloadThread::flushHw_l() 5283{ 5284 DirectOutputThread::flushHw_l(); 5285 // Flush anything still waiting in the mixbuffer 5286 mCurrentWriteLength = 0; 5287 mBytesRemaining = 0; 5288 mPausedWriteLength = 0; 5289 mPausedBytesRemaining = 0; 5290 5291 if (mUseAsyncWrite) { 5292 // discard any pending drain or write ack by incrementing sequence 5293 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5294 mDrainSequence = (mDrainSequence + 2) & ~1; 5295 ALOG_ASSERT(mCallbackThread != 0); 5296 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5297 mCallbackThread->setDraining(mDrainSequence); 5298 } 5299} 5300 5301// ---------------------------------------------------------------------------- 5302 5303AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5304 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5305 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5306 systemReady, DUPLICATING), 5307 mWaitTimeMs(UINT_MAX) 5308{ 5309 addOutputTrack(mainThread); 5310} 5311 5312AudioFlinger::DuplicatingThread::~DuplicatingThread() 5313{ 5314 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5315 mOutputTracks[i]->destroy(); 5316 } 5317} 5318 5319void AudioFlinger::DuplicatingThread::threadLoop_mix() 5320{ 5321 // mix buffers... 5322 if (outputsReady(outputTracks)) { 5323 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5324 } else { 5325 if (mMixerBufferValid) { 5326 memset(mMixerBuffer, 0, mMixerBufferSize); 5327 } else { 5328 memset(mSinkBuffer, 0, mSinkBufferSize); 5329 } 5330 } 5331 mSleepTimeUs = 0; 5332 writeFrames = mNormalFrameCount; 5333 mCurrentWriteLength = mSinkBufferSize; 5334 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5335} 5336 5337void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5338{ 5339 if (mSleepTimeUs == 0) { 5340 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5341 mSleepTimeUs = mActiveSleepTimeUs; 5342 } else { 5343 mSleepTimeUs = mIdleSleepTimeUs; 5344 } 5345 } else if (mBytesWritten != 0) { 5346 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5347 writeFrames = mNormalFrameCount; 5348 memset(mSinkBuffer, 0, mSinkBufferSize); 5349 } else { 5350 // flush remaining overflow buffers in output tracks 5351 writeFrames = 0; 5352 } 5353 mSleepTimeUs = 0; 5354 } 5355} 5356 5357ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5358{ 5359 for (size_t i = 0; i < outputTracks.size(); i++) { 5360 outputTracks[i]->write(mSinkBuffer, writeFrames); 5361 } 5362 mStandby = false; 5363 return (ssize_t)mSinkBufferSize; 5364} 5365 5366void AudioFlinger::DuplicatingThread::threadLoop_standby() 5367{ 5368 // DuplicatingThread implements standby by stopping all tracks 5369 for (size_t i = 0; i < outputTracks.size(); i++) { 5370 outputTracks[i]->stop(); 5371 } 5372} 5373 5374void AudioFlinger::DuplicatingThread::saveOutputTracks() 5375{ 5376 outputTracks = mOutputTracks; 5377} 5378 5379void AudioFlinger::DuplicatingThread::clearOutputTracks() 5380{ 5381 outputTracks.clear(); 5382} 5383 5384void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5385{ 5386 Mutex::Autolock _l(mLock); 5387 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5388 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5389 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5390 const size_t frameCount = 5391 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5392 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5393 // from different OutputTracks and their associated MixerThreads (e.g. one may 5394 // nearly empty and the other may be dropping data). 5395 5396 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5397 this, 5398 mSampleRate, 5399 mFormat, 5400 mChannelMask, 5401 frameCount, 5402 IPCThreadState::self()->getCallingUid()); 5403 if (outputTrack->cblk() != NULL) { 5404 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5405 mOutputTracks.add(outputTrack); 5406 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5407 updateWaitTime_l(); 5408 } 5409} 5410 5411void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5412{ 5413 Mutex::Autolock _l(mLock); 5414 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5415 if (mOutputTracks[i]->thread() == thread) { 5416 mOutputTracks[i]->destroy(); 5417 mOutputTracks.removeAt(i); 5418 updateWaitTime_l(); 5419 if (thread->getOutput() == mOutput) { 5420 mOutput = NULL; 5421 } 5422 return; 5423 } 5424 } 5425 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5426} 5427 5428// caller must hold mLock 5429void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5430{ 5431 mWaitTimeMs = UINT_MAX; 5432 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5433 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5434 if (strong != 0) { 5435 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5436 if (waitTimeMs < mWaitTimeMs) { 5437 mWaitTimeMs = waitTimeMs; 5438 } 5439 } 5440 } 5441} 5442 5443 5444bool AudioFlinger::DuplicatingThread::outputsReady( 5445 const SortedVector< sp<OutputTrack> > &outputTracks) 5446{ 5447 for (size_t i = 0; i < outputTracks.size(); i++) { 5448 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5449 if (thread == 0) { 5450 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5451 outputTracks[i].get()); 5452 return false; 5453 } 5454 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5455 // see note at standby() declaration 5456 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5457 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5458 thread.get()); 5459 return false; 5460 } 5461 } 5462 return true; 5463} 5464 5465uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5466{ 5467 return (mWaitTimeMs * 1000) / 2; 5468} 5469 5470void AudioFlinger::DuplicatingThread::cacheParameters_l() 5471{ 5472 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5473 updateWaitTime_l(); 5474 5475 MixerThread::cacheParameters_l(); 5476} 5477 5478// ---------------------------------------------------------------------------- 5479// Record 5480// ---------------------------------------------------------------------------- 5481 5482AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5483 AudioStreamIn *input, 5484 audio_io_handle_t id, 5485 audio_devices_t outDevice, 5486 audio_devices_t inDevice, 5487 bool systemReady 5488#ifdef TEE_SINK 5489 , const sp<NBAIO_Sink>& teeSink 5490#endif 5491 ) : 5492 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5493 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5494 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5495 mRsmpInRear(0) 5496#ifdef TEE_SINK 5497 , mTeeSink(teeSink) 5498#endif 5499 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5500 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5501 // mFastCapture below 5502 , mFastCaptureFutex(0) 5503 // mInputSource 5504 // mPipeSink 5505 // mPipeSource 5506 , mPipeFramesP2(0) 5507 // mPipeMemory 5508 // mFastCaptureNBLogWriter 5509 , mFastTrackAvail(false) 5510{ 5511 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5512 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5513 5514 readInputParameters_l(); 5515 5516 // create an NBAIO source for the HAL input stream, and negotiate 5517 mInputSource = new AudioStreamInSource(input->stream); 5518 size_t numCounterOffers = 0; 5519 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5520 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5521 ALOG_ASSERT(index == 0); 5522 5523 // initialize fast capture depending on configuration 5524 bool initFastCapture; 5525 switch (kUseFastCapture) { 5526 case FastCapture_Never: 5527 initFastCapture = false; 5528 break; 5529 case FastCapture_Always: 5530 initFastCapture = true; 5531 break; 5532 case FastCapture_Static: 5533 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5534 break; 5535 // case FastCapture_Dynamic: 5536 } 5537 5538 if (initFastCapture) { 5539 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5540 NBAIO_Format format = mInputSource->format(); 5541 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5542 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5543 void *pipeBuffer; 5544 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5545 sp<IMemory> pipeMemory; 5546 if ((roHeap == 0) || 5547 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5548 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5549 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5550 goto failed; 5551 } 5552 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5553 memset(pipeBuffer, 0, pipeSize); 5554 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5555 const NBAIO_Format offers[1] = {format}; 5556 size_t numCounterOffers = 0; 5557 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5558 ALOG_ASSERT(index == 0); 5559 mPipeSink = pipe; 5560 PipeReader *pipeReader = new PipeReader(*pipe); 5561 numCounterOffers = 0; 5562 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5563 ALOG_ASSERT(index == 0); 5564 mPipeSource = pipeReader; 5565 mPipeFramesP2 = pipeFramesP2; 5566 mPipeMemory = pipeMemory; 5567 5568 // create fast capture 5569 mFastCapture = new FastCapture(); 5570 FastCaptureStateQueue *sq = mFastCapture->sq(); 5571#ifdef STATE_QUEUE_DUMP 5572 // FIXME 5573#endif 5574 FastCaptureState *state = sq->begin(); 5575 state->mCblk = NULL; 5576 state->mInputSource = mInputSource.get(); 5577 state->mInputSourceGen++; 5578 state->mPipeSink = pipe; 5579 state->mPipeSinkGen++; 5580 state->mFrameCount = mFrameCount; 5581 state->mCommand = FastCaptureState::COLD_IDLE; 5582 // already done in constructor initialization list 5583 //mFastCaptureFutex = 0; 5584 state->mColdFutexAddr = &mFastCaptureFutex; 5585 state->mColdGen++; 5586 state->mDumpState = &mFastCaptureDumpState; 5587#ifdef TEE_SINK 5588 // FIXME 5589#endif 5590 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5591 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5592 sq->end(); 5593 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5594 5595 // start the fast capture 5596 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5597 pid_t tid = mFastCapture->getTid(); 5598 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5599#ifdef AUDIO_WATCHDOG 5600 // FIXME 5601#endif 5602 5603 mFastTrackAvail = true; 5604 } 5605failed: ; 5606 5607 // FIXME mNormalSource 5608} 5609 5610AudioFlinger::RecordThread::~RecordThread() 5611{ 5612 if (mFastCapture != 0) { 5613 FastCaptureStateQueue *sq = mFastCapture->sq(); 5614 FastCaptureState *state = sq->begin(); 5615 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5616 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5617 if (old == -1) { 5618 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5619 } 5620 } 5621 state->mCommand = FastCaptureState::EXIT; 5622 sq->end(); 5623 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5624 mFastCapture->join(); 5625 mFastCapture.clear(); 5626 } 5627 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5628 mAudioFlinger->unregisterWriter(mNBLogWriter); 5629 free(mRsmpInBuffer); 5630} 5631 5632void AudioFlinger::RecordThread::onFirstRef() 5633{ 5634 run(mThreadName, PRIORITY_URGENT_AUDIO); 5635} 5636 5637bool AudioFlinger::RecordThread::threadLoop() 5638{ 5639 nsecs_t lastWarning = 0; 5640 5641 inputStandBy(); 5642 5643reacquire_wakelock: 5644 sp<RecordTrack> activeTrack; 5645 int activeTracksGen; 5646 { 5647 Mutex::Autolock _l(mLock); 5648 size_t size = mActiveTracks.size(); 5649 activeTracksGen = mActiveTracksGen; 5650 if (size > 0) { 5651 // FIXME an arbitrary choice 5652 activeTrack = mActiveTracks[0]; 5653 acquireWakeLock_l(activeTrack->uid()); 5654 if (size > 1) { 5655 SortedVector<int> tmp; 5656 for (size_t i = 0; i < size; i++) { 5657 tmp.add(mActiveTracks[i]->uid()); 5658 } 5659 updateWakeLockUids_l(tmp); 5660 } 5661 } else { 5662 acquireWakeLock_l(-1); 5663 } 5664 } 5665 5666 // used to request a deferred sleep, to be executed later while mutex is unlocked 5667 uint32_t sleepUs = 0; 5668 5669 // loop while there is work to do 5670 for (;;) { 5671 Vector< sp<EffectChain> > effectChains; 5672 5673 // sleep with mutex unlocked 5674 if (sleepUs > 0) { 5675 ATRACE_BEGIN("sleep"); 5676 usleep(sleepUs); 5677 ATRACE_END(); 5678 sleepUs = 0; 5679 } 5680 5681 // activeTracks accumulates a copy of a subset of mActiveTracks 5682 Vector< sp<RecordTrack> > activeTracks; 5683 5684 // reference to the (first and only) active fast track 5685 sp<RecordTrack> fastTrack; 5686 5687 // reference to a fast track which is about to be removed 5688 sp<RecordTrack> fastTrackToRemove; 5689 5690 { // scope for mLock 5691 Mutex::Autolock _l(mLock); 5692 5693 processConfigEvents_l(); 5694 5695 // check exitPending here because checkForNewParameters_l() and 5696 // checkForNewParameters_l() can temporarily release mLock 5697 if (exitPending()) { 5698 break; 5699 } 5700 5701 // if no active track(s), then standby and release wakelock 5702 size_t size = mActiveTracks.size(); 5703 if (size == 0) { 5704 standbyIfNotAlreadyInStandby(); 5705 // exitPending() can't become true here 5706 releaseWakeLock_l(); 5707 ALOGV("RecordThread: loop stopping"); 5708 // go to sleep 5709 mWaitWorkCV.wait(mLock); 5710 ALOGV("RecordThread: loop starting"); 5711 goto reacquire_wakelock; 5712 } 5713 5714 if (mActiveTracksGen != activeTracksGen) { 5715 activeTracksGen = mActiveTracksGen; 5716 SortedVector<int> tmp; 5717 for (size_t i = 0; i < size; i++) { 5718 tmp.add(mActiveTracks[i]->uid()); 5719 } 5720 updateWakeLockUids_l(tmp); 5721 } 5722 5723 bool doBroadcast = false; 5724 for (size_t i = 0; i < size; ) { 5725 5726 activeTrack = mActiveTracks[i]; 5727 if (activeTrack->isTerminated()) { 5728 if (activeTrack->isFastTrack()) { 5729 ALOG_ASSERT(fastTrackToRemove == 0); 5730 fastTrackToRemove = activeTrack; 5731 } 5732 removeTrack_l(activeTrack); 5733 mActiveTracks.remove(activeTrack); 5734 mActiveTracksGen++; 5735 size--; 5736 continue; 5737 } 5738 5739 TrackBase::track_state activeTrackState = activeTrack->mState; 5740 switch (activeTrackState) { 5741 5742 case TrackBase::PAUSING: 5743 mActiveTracks.remove(activeTrack); 5744 mActiveTracksGen++; 5745 doBroadcast = true; 5746 size--; 5747 continue; 5748 5749 case TrackBase::STARTING_1: 5750 sleepUs = 10000; 5751 i++; 5752 continue; 5753 5754 case TrackBase::STARTING_2: 5755 doBroadcast = true; 5756 mStandby = false; 5757 activeTrack->mState = TrackBase::ACTIVE; 5758 break; 5759 5760 case TrackBase::ACTIVE: 5761 break; 5762 5763 case TrackBase::IDLE: 5764 i++; 5765 continue; 5766 5767 default: 5768 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5769 } 5770 5771 activeTracks.add(activeTrack); 5772 i++; 5773 5774 if (activeTrack->isFastTrack()) { 5775 ALOG_ASSERT(!mFastTrackAvail); 5776 ALOG_ASSERT(fastTrack == 0); 5777 fastTrack = activeTrack; 5778 } 5779 } 5780 if (doBroadcast) { 5781 mStartStopCond.broadcast(); 5782 } 5783 5784 // sleep if there are no active tracks to process 5785 if (activeTracks.size() == 0) { 5786 if (sleepUs == 0) { 5787 sleepUs = kRecordThreadSleepUs; 5788 } 5789 continue; 5790 } 5791 sleepUs = 0; 5792 5793 lockEffectChains_l(effectChains); 5794 } 5795 5796 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5797 5798 size_t size = effectChains.size(); 5799 for (size_t i = 0; i < size; i++) { 5800 // thread mutex is not locked, but effect chain is locked 5801 effectChains[i]->process_l(); 5802 } 5803 5804 // Push a new fast capture state if fast capture is not already running, or cblk change 5805 if (mFastCapture != 0) { 5806 FastCaptureStateQueue *sq = mFastCapture->sq(); 5807 FastCaptureState *state = sq->begin(); 5808 bool didModify = false; 5809 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5810 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5811 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5812 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5813 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5814 if (old == -1) { 5815 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5816 } 5817 } 5818 state->mCommand = FastCaptureState::READ_WRITE; 5819#if 0 // FIXME 5820 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5821 FastThreadDumpState::kSamplingNforLowRamDevice : 5822 FastThreadDumpState::kSamplingN); 5823#endif 5824 didModify = true; 5825 } 5826 audio_track_cblk_t *cblkOld = state->mCblk; 5827 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5828 if (cblkNew != cblkOld) { 5829 state->mCblk = cblkNew; 5830 // block until acked if removing a fast track 5831 if (cblkOld != NULL) { 5832 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5833 } 5834 didModify = true; 5835 } 5836 sq->end(didModify); 5837 if (didModify) { 5838 sq->push(block); 5839#if 0 5840 if (kUseFastCapture == FastCapture_Dynamic) { 5841 mNormalSource = mPipeSource; 5842 } 5843#endif 5844 } 5845 } 5846 5847 // now run the fast track destructor with thread mutex unlocked 5848 fastTrackToRemove.clear(); 5849 5850 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5851 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5852 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5853 // If destination is non-contiguous, first read past the nominal end of buffer, then 5854 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5855 5856 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5857 ssize_t framesRead; 5858 5859 // If an NBAIO source is present, use it to read the normal capture's data 5860 if (mPipeSource != 0) { 5861 size_t framesToRead = mBufferSize / mFrameSize; 5862 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5863 framesToRead, AudioBufferProvider::kInvalidPTS); 5864 if (framesRead == 0) { 5865 // since pipe is non-blocking, simulate blocking input 5866 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5867 } 5868 // otherwise use the HAL / AudioStreamIn directly 5869 } else { 5870 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5871 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5872 if (bytesRead < 0) { 5873 framesRead = bytesRead; 5874 } else { 5875 framesRead = bytesRead / mFrameSize; 5876 } 5877 } 5878 5879 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5880 ALOGE("read failed: framesRead=%d", framesRead); 5881 // Force input into standby so that it tries to recover at next read attempt 5882 inputStandBy(); 5883 sleepUs = kRecordThreadSleepUs; 5884 } 5885 if (framesRead <= 0) { 5886 goto unlock; 5887 } 5888 ALOG_ASSERT(framesRead > 0); 5889 5890 if (mTeeSink != 0) { 5891 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5892 } 5893 // If destination is non-contiguous, we now correct for reading past end of buffer. 5894 { 5895 size_t part1 = mRsmpInFramesP2 - rear; 5896 if ((size_t) framesRead > part1) { 5897 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5898 (framesRead - part1) * mFrameSize); 5899 } 5900 } 5901 rear = mRsmpInRear += framesRead; 5902 5903 size = activeTracks.size(); 5904 // loop over each active track 5905 for (size_t i = 0; i < size; i++) { 5906 activeTrack = activeTracks[i]; 5907 5908 // skip fast tracks, as those are handled directly by FastCapture 5909 if (activeTrack->isFastTrack()) { 5910 continue; 5911 } 5912 5913 // TODO: This code probably should be moved to RecordTrack. 5914 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5915 5916 enum { 5917 OVERRUN_UNKNOWN, 5918 OVERRUN_TRUE, 5919 OVERRUN_FALSE 5920 } overrun = OVERRUN_UNKNOWN; 5921 5922 // loop over getNextBuffer to handle circular sink 5923 for (;;) { 5924 5925 activeTrack->mSink.frameCount = ~0; 5926 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5927 size_t framesOut = activeTrack->mSink.frameCount; 5928 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5929 5930 // check available frames and handle overrun conditions 5931 // if the record track isn't draining fast enough. 5932 bool hasOverrun; 5933 size_t framesIn; 5934 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5935 if (hasOverrun) { 5936 overrun = OVERRUN_TRUE; 5937 } 5938 if (framesOut == 0 || framesIn == 0) { 5939 break; 5940 } 5941 5942 // Don't allow framesOut to be larger than what is possible with resampling 5943 // from framesIn. 5944 // This isn't strictly necessary but helps limit buffer resizing in 5945 // RecordBufferConverter. TODO: remove when no longer needed. 5946 framesOut = min(framesOut, 5947 destinationFramesPossible( 5948 framesIn, mSampleRate, activeTrack->mSampleRate)); 5949 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5950 framesOut = activeTrack->mRecordBufferConverter->convert( 5951 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5952 5953 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5954 overrun = OVERRUN_FALSE; 5955 } 5956 5957 if (activeTrack->mFramesToDrop == 0) { 5958 if (framesOut > 0) { 5959 activeTrack->mSink.frameCount = framesOut; 5960 activeTrack->releaseBuffer(&activeTrack->mSink); 5961 } 5962 } else { 5963 // FIXME could do a partial drop of framesOut 5964 if (activeTrack->mFramesToDrop > 0) { 5965 activeTrack->mFramesToDrop -= framesOut; 5966 if (activeTrack->mFramesToDrop <= 0) { 5967 activeTrack->clearSyncStartEvent(); 5968 } 5969 } else { 5970 activeTrack->mFramesToDrop += framesOut; 5971 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5972 activeTrack->mSyncStartEvent->isCancelled()) { 5973 ALOGW("Synced record %s, session %d, trigger session %d", 5974 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5975 activeTrack->sessionId(), 5976 (activeTrack->mSyncStartEvent != 0) ? 5977 activeTrack->mSyncStartEvent->triggerSession() : 0); 5978 activeTrack->clearSyncStartEvent(); 5979 } 5980 } 5981 } 5982 5983 if (framesOut == 0) { 5984 break; 5985 } 5986 } 5987 5988 switch (overrun) { 5989 case OVERRUN_TRUE: 5990 // client isn't retrieving buffers fast enough 5991 if (!activeTrack->setOverflow()) { 5992 nsecs_t now = systemTime(); 5993 // FIXME should lastWarning per track? 5994 if ((now - lastWarning) > kWarningThrottleNs) { 5995 ALOGW("RecordThread: buffer overflow"); 5996 lastWarning = now; 5997 } 5998 } 5999 break; 6000 case OVERRUN_FALSE: 6001 activeTrack->clearOverflow(); 6002 break; 6003 case OVERRUN_UNKNOWN: 6004 break; 6005 } 6006 6007 } 6008 6009unlock: 6010 // enable changes in effect chain 6011 unlockEffectChains(effectChains); 6012 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6013 } 6014 6015 standbyIfNotAlreadyInStandby(); 6016 6017 { 6018 Mutex::Autolock _l(mLock); 6019 for (size_t i = 0; i < mTracks.size(); i++) { 6020 sp<RecordTrack> track = mTracks[i]; 6021 track->invalidate(); 6022 } 6023 mActiveTracks.clear(); 6024 mActiveTracksGen++; 6025 mStartStopCond.broadcast(); 6026 } 6027 6028 releaseWakeLock(); 6029 6030 ALOGV("RecordThread %p exiting", this); 6031 return false; 6032} 6033 6034void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6035{ 6036 if (!mStandby) { 6037 inputStandBy(); 6038 mStandby = true; 6039 } 6040} 6041 6042void AudioFlinger::RecordThread::inputStandBy() 6043{ 6044 // Idle the fast capture if it's currently running 6045 if (mFastCapture != 0) { 6046 FastCaptureStateQueue *sq = mFastCapture->sq(); 6047 FastCaptureState *state = sq->begin(); 6048 if (!(state->mCommand & FastCaptureState::IDLE)) { 6049 state->mCommand = FastCaptureState::COLD_IDLE; 6050 state->mColdFutexAddr = &mFastCaptureFutex; 6051 state->mColdGen++; 6052 mFastCaptureFutex = 0; 6053 sq->end(); 6054 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6055 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6056#if 0 6057 if (kUseFastCapture == FastCapture_Dynamic) { 6058 // FIXME 6059 } 6060#endif 6061#ifdef AUDIO_WATCHDOG 6062 // FIXME 6063#endif 6064 } else { 6065 sq->end(false /*didModify*/); 6066 } 6067 } 6068 mInput->stream->common.standby(&mInput->stream->common); 6069} 6070 6071// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6072sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6073 const sp<AudioFlinger::Client>& client, 6074 uint32_t sampleRate, 6075 audio_format_t format, 6076 audio_channel_mask_t channelMask, 6077 size_t *pFrameCount, 6078 int sessionId, 6079 size_t *notificationFrames, 6080 int uid, 6081 IAudioFlinger::track_flags_t *flags, 6082 pid_t tid, 6083 status_t *status) 6084{ 6085 size_t frameCount = *pFrameCount; 6086 sp<RecordTrack> track; 6087 status_t lStatus; 6088 6089 // client expresses a preference for FAST, but we get the final say 6090 if (*flags & IAudioFlinger::TRACK_FAST) { 6091 if ( 6092 // we formerly checked for a callback handler (non-0 tid), 6093 // but that is no longer required for TRANSFER_OBTAIN mode 6094 // 6095 // frame count is not specified, or is exactly the pipe depth 6096 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6097 // PCM data 6098 audio_is_linear_pcm(format) && 6099 // native format 6100 (format == mFormat) && 6101 // native channel mask 6102 (channelMask == mChannelMask) && 6103 // native hardware sample rate 6104 (sampleRate == mSampleRate) && 6105 // record thread has an associated fast capture 6106 hasFastCapture() && 6107 // there are sufficient fast track slots available 6108 mFastTrackAvail 6109 ) { 6110 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6111 frameCount, mFrameCount); 6112 } else { 6113 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6114 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6115 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6116 frameCount, mFrameCount, mPipeFramesP2, 6117 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6118 hasFastCapture(), tid, mFastTrackAvail); 6119 *flags &= ~IAudioFlinger::TRACK_FAST; 6120 } 6121 } 6122 6123 // compute track buffer size in frames, and suggest the notification frame count 6124 if (*flags & IAudioFlinger::TRACK_FAST) { 6125 // fast track: frame count is exactly the pipe depth 6126 frameCount = mPipeFramesP2; 6127 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6128 *notificationFrames = mFrameCount; 6129 } else { 6130 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6131 // or 20 ms if there is a fast capture 6132 // TODO This could be a roundupRatio inline, and const 6133 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6134 * sampleRate + mSampleRate - 1) / mSampleRate; 6135 // minimum number of notification periods is at least kMinNotifications, 6136 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6137 static const size_t kMinNotifications = 3; 6138 static const uint32_t kMinMs = 30; 6139 // TODO This could be a roundupRatio inline 6140 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6141 // TODO This could be a roundupRatio inline 6142 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6143 maxNotificationFrames; 6144 const size_t minFrameCount = maxNotificationFrames * 6145 max(kMinNotifications, minNotificationsByMs); 6146 frameCount = max(frameCount, minFrameCount); 6147 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6148 *notificationFrames = maxNotificationFrames; 6149 } 6150 } 6151 *pFrameCount = frameCount; 6152 6153 lStatus = initCheck(); 6154 if (lStatus != NO_ERROR) { 6155 ALOGE("createRecordTrack_l() audio driver not initialized"); 6156 goto Exit; 6157 } 6158 6159 { // scope for mLock 6160 Mutex::Autolock _l(mLock); 6161 6162 track = new RecordTrack(this, client, sampleRate, 6163 format, channelMask, frameCount, NULL, sessionId, uid, 6164 *flags, TrackBase::TYPE_DEFAULT); 6165 6166 lStatus = track->initCheck(); 6167 if (lStatus != NO_ERROR) { 6168 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6169 // track must be cleared from the caller as the caller has the AF lock 6170 goto Exit; 6171 } 6172 mTracks.add(track); 6173 6174 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6175 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6176 mAudioFlinger->btNrecIsOff(); 6177 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6178 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6179 6180 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6181 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6182 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6183 // so ask activity manager to do this on our behalf 6184 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6185 } 6186 } 6187 6188 lStatus = NO_ERROR; 6189 6190Exit: 6191 *status = lStatus; 6192 return track; 6193} 6194 6195status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6196 AudioSystem::sync_event_t event, 6197 int triggerSession) 6198{ 6199 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6200 sp<ThreadBase> strongMe = this; 6201 status_t status = NO_ERROR; 6202 6203 if (event == AudioSystem::SYNC_EVENT_NONE) { 6204 recordTrack->clearSyncStartEvent(); 6205 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6206 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6207 triggerSession, 6208 recordTrack->sessionId(), 6209 syncStartEventCallback, 6210 recordTrack); 6211 // Sync event can be cancelled by the trigger session if the track is not in a 6212 // compatible state in which case we start record immediately 6213 if (recordTrack->mSyncStartEvent->isCancelled()) { 6214 recordTrack->clearSyncStartEvent(); 6215 } else { 6216 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6217 recordTrack->mFramesToDrop = - 6218 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6219 } 6220 } 6221 6222 { 6223 // This section is a rendezvous between binder thread executing start() and RecordThread 6224 AutoMutex lock(mLock); 6225 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6226 if (recordTrack->mState == TrackBase::PAUSING) { 6227 ALOGV("active record track PAUSING -> ACTIVE"); 6228 recordTrack->mState = TrackBase::ACTIVE; 6229 } else { 6230 ALOGV("active record track state %d", recordTrack->mState); 6231 } 6232 return status; 6233 } 6234 6235 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6236 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6237 // or using a separate command thread 6238 recordTrack->mState = TrackBase::STARTING_1; 6239 mActiveTracks.add(recordTrack); 6240 mActiveTracksGen++; 6241 status_t status = NO_ERROR; 6242 if (recordTrack->isExternalTrack()) { 6243 mLock.unlock(); 6244 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6245 mLock.lock(); 6246 // FIXME should verify that recordTrack is still in mActiveTracks 6247 if (status != NO_ERROR) { 6248 mActiveTracks.remove(recordTrack); 6249 mActiveTracksGen++; 6250 recordTrack->clearSyncStartEvent(); 6251 ALOGV("RecordThread::start error %d", status); 6252 return status; 6253 } 6254 } 6255 // Catch up with current buffer indices if thread is already running. 6256 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6257 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6258 // see previously buffered data before it called start(), but with greater risk of overrun. 6259 6260 recordTrack->mResamplerBufferProvider->reset(); 6261 // clear any converter state as new data will be discontinuous 6262 recordTrack->mRecordBufferConverter->reset(); 6263 recordTrack->mState = TrackBase::STARTING_2; 6264 // signal thread to start 6265 mWaitWorkCV.broadcast(); 6266 if (mActiveTracks.indexOf(recordTrack) < 0) { 6267 ALOGV("Record failed to start"); 6268 status = BAD_VALUE; 6269 goto startError; 6270 } 6271 return status; 6272 } 6273 6274startError: 6275 if (recordTrack->isExternalTrack()) { 6276 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6277 } 6278 recordTrack->clearSyncStartEvent(); 6279 // FIXME I wonder why we do not reset the state here? 6280 return status; 6281} 6282 6283void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6284{ 6285 sp<SyncEvent> strongEvent = event.promote(); 6286 6287 if (strongEvent != 0) { 6288 sp<RefBase> ptr = strongEvent->cookie().promote(); 6289 if (ptr != 0) { 6290 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6291 recordTrack->handleSyncStartEvent(strongEvent); 6292 } 6293 } 6294} 6295 6296bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6297 ALOGV("RecordThread::stop"); 6298 AutoMutex _l(mLock); 6299 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6300 return false; 6301 } 6302 // note that threadLoop may still be processing the track at this point [without lock] 6303 recordTrack->mState = TrackBase::PAUSING; 6304 // do not wait for mStartStopCond if exiting 6305 if (exitPending()) { 6306 return true; 6307 } 6308 // FIXME incorrect usage of wait: no explicit predicate or loop 6309 mStartStopCond.wait(mLock); 6310 // if we have been restarted, recordTrack is in mActiveTracks here 6311 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6312 ALOGV("Record stopped OK"); 6313 return true; 6314 } 6315 return false; 6316} 6317 6318bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6319{ 6320 return false; 6321} 6322 6323status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6324{ 6325#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6326 if (!isValidSyncEvent(event)) { 6327 return BAD_VALUE; 6328 } 6329 6330 int eventSession = event->triggerSession(); 6331 status_t ret = NAME_NOT_FOUND; 6332 6333 Mutex::Autolock _l(mLock); 6334 6335 for (size_t i = 0; i < mTracks.size(); i++) { 6336 sp<RecordTrack> track = mTracks[i]; 6337 if (eventSession == track->sessionId()) { 6338 (void) track->setSyncEvent(event); 6339 ret = NO_ERROR; 6340 } 6341 } 6342 return ret; 6343#else 6344 return BAD_VALUE; 6345#endif 6346} 6347 6348// destroyTrack_l() must be called with ThreadBase::mLock held 6349void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6350{ 6351 track->terminate(); 6352 track->mState = TrackBase::STOPPED; 6353 // active tracks are removed by threadLoop() 6354 if (mActiveTracks.indexOf(track) < 0) { 6355 removeTrack_l(track); 6356 } 6357} 6358 6359void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6360{ 6361 mTracks.remove(track); 6362 // need anything related to effects here? 6363 if (track->isFastTrack()) { 6364 ALOG_ASSERT(!mFastTrackAvail); 6365 mFastTrackAvail = true; 6366 } 6367} 6368 6369void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6370{ 6371 dumpInternals(fd, args); 6372 dumpTracks(fd, args); 6373 dumpEffectChains(fd, args); 6374} 6375 6376void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6377{ 6378 dprintf(fd, "\nInput thread %p:\n", this); 6379 6380 dumpBase(fd, args); 6381 6382 if (mActiveTracks.size() == 0) { 6383 dprintf(fd, " No active record clients\n"); 6384 } 6385 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6386 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6387 6388 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6389 // while we are dumping it. It may be inconsistent, but it won't mutate! 6390 // This is a large object so we place it on the heap. 6391 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6392 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6393 copy->dump(fd); 6394 delete copy; 6395} 6396 6397void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6398{ 6399 const size_t SIZE = 256; 6400 char buffer[SIZE]; 6401 String8 result; 6402 6403 size_t numtracks = mTracks.size(); 6404 size_t numactive = mActiveTracks.size(); 6405 size_t numactiveseen = 0; 6406 dprintf(fd, " %d Tracks", numtracks); 6407 if (numtracks) { 6408 dprintf(fd, " of which %d are active\n", numactive); 6409 RecordTrack::appendDumpHeader(result); 6410 for (size_t i = 0; i < numtracks ; ++i) { 6411 sp<RecordTrack> track = mTracks[i]; 6412 if (track != 0) { 6413 bool active = mActiveTracks.indexOf(track) >= 0; 6414 if (active) { 6415 numactiveseen++; 6416 } 6417 track->dump(buffer, SIZE, active); 6418 result.append(buffer); 6419 } 6420 } 6421 } else { 6422 dprintf(fd, "\n"); 6423 } 6424 6425 if (numactiveseen != numactive) { 6426 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6427 " not in the track list\n"); 6428 result.append(buffer); 6429 RecordTrack::appendDumpHeader(result); 6430 for (size_t i = 0; i < numactive; ++i) { 6431 sp<RecordTrack> track = mActiveTracks[i]; 6432 if (mTracks.indexOf(track) < 0) { 6433 track->dump(buffer, SIZE, true); 6434 result.append(buffer); 6435 } 6436 } 6437 6438 } 6439 write(fd, result.string(), result.size()); 6440} 6441 6442 6443void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6444{ 6445 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6446 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6447 mRsmpInFront = recordThread->mRsmpInRear; 6448 mRsmpInUnrel = 0; 6449} 6450 6451void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6452 size_t *framesAvailable, bool *hasOverrun) 6453{ 6454 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6455 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6456 const int32_t rear = recordThread->mRsmpInRear; 6457 const int32_t front = mRsmpInFront; 6458 const ssize_t filled = rear - front; 6459 6460 size_t framesIn; 6461 bool overrun = false; 6462 if (filled < 0) { 6463 // should not happen, but treat like a massive overrun and re-sync 6464 framesIn = 0; 6465 mRsmpInFront = rear; 6466 overrun = true; 6467 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6468 framesIn = (size_t) filled; 6469 } else { 6470 // client is not keeping up with server, but give it latest data 6471 framesIn = recordThread->mRsmpInFrames; 6472 mRsmpInFront = /* front = */ rear - framesIn; 6473 overrun = true; 6474 } 6475 if (framesAvailable != NULL) { 6476 *framesAvailable = framesIn; 6477 } 6478 if (hasOverrun != NULL) { 6479 *hasOverrun = overrun; 6480 } 6481} 6482 6483// AudioBufferProvider interface 6484status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6485 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6486{ 6487 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6488 if (threadBase == 0) { 6489 buffer->frameCount = 0; 6490 buffer->raw = NULL; 6491 return NOT_ENOUGH_DATA; 6492 } 6493 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6494 int32_t rear = recordThread->mRsmpInRear; 6495 int32_t front = mRsmpInFront; 6496 ssize_t filled = rear - front; 6497 // FIXME should not be P2 (don't want to increase latency) 6498 // FIXME if client not keeping up, discard 6499 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6500 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6501 front &= recordThread->mRsmpInFramesP2 - 1; 6502 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6503 if (part1 > (size_t) filled) { 6504 part1 = filled; 6505 } 6506 size_t ask = buffer->frameCount; 6507 ALOG_ASSERT(ask > 0); 6508 if (part1 > ask) { 6509 part1 = ask; 6510 } 6511 if (part1 == 0) { 6512 // out of data is fine since the resampler will return a short-count. 6513 buffer->raw = NULL; 6514 buffer->frameCount = 0; 6515 mRsmpInUnrel = 0; 6516 return NOT_ENOUGH_DATA; 6517 } 6518 6519 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6520 buffer->frameCount = part1; 6521 mRsmpInUnrel = part1; 6522 return NO_ERROR; 6523} 6524 6525// AudioBufferProvider interface 6526void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6527 AudioBufferProvider::Buffer* buffer) 6528{ 6529 size_t stepCount = buffer->frameCount; 6530 if (stepCount == 0) { 6531 return; 6532 } 6533 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6534 mRsmpInUnrel -= stepCount; 6535 mRsmpInFront += stepCount; 6536 buffer->raw = NULL; 6537 buffer->frameCount = 0; 6538} 6539 6540AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6541 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6542 uint32_t srcSampleRate, 6543 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6544 uint32_t dstSampleRate) : 6545 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6546 // mSrcFormat 6547 // mSrcSampleRate 6548 // mDstChannelMask 6549 // mDstFormat 6550 // mDstSampleRate 6551 // mSrcChannelCount 6552 // mDstChannelCount 6553 // mDstFrameSize 6554 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6555 mResampler(NULL), 6556 mIsLegacyDownmix(false), 6557 mIsLegacyUpmix(false), 6558 mRequiresFloat(false), 6559 mInputConverterProvider(NULL) 6560{ 6561 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6562 dstChannelMask, dstFormat, dstSampleRate); 6563} 6564 6565AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6566 free(mBuf); 6567 delete mResampler; 6568 delete mInputConverterProvider; 6569} 6570 6571size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6572 AudioBufferProvider *provider, size_t frames) 6573{ 6574 if (mInputConverterProvider != NULL) { 6575 mInputConverterProvider->setBufferProvider(provider); 6576 provider = mInputConverterProvider; 6577 } 6578 6579 if (mResampler == NULL) { 6580 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6581 mSrcSampleRate, mSrcFormat, mDstFormat); 6582 6583 AudioBufferProvider::Buffer buffer; 6584 for (size_t i = frames; i > 0; ) { 6585 buffer.frameCount = i; 6586 status_t status = provider->getNextBuffer(&buffer, 0); 6587 if (status != OK || buffer.frameCount == 0) { 6588 frames -= i; // cannot fill request. 6589 break; 6590 } 6591 // format convert to destination buffer 6592 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6593 6594 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6595 i -= buffer.frameCount; 6596 provider->releaseBuffer(&buffer); 6597 } 6598 } else { 6599 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6600 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6601 6602 // reallocate buffer if needed 6603 if (mBufFrameSize != 0 && mBufFrames < frames) { 6604 free(mBuf); 6605 mBufFrames = frames; 6606 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6607 } 6608 // resampler accumulates, but we only have one source track 6609 memset(mBuf, 0, frames * mBufFrameSize); 6610 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6611 // format convert to destination buffer 6612 convertResampler(dst, mBuf, frames); 6613 } 6614 return frames; 6615} 6616 6617status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6618 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6619 uint32_t srcSampleRate, 6620 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6621 uint32_t dstSampleRate) 6622{ 6623 // quick evaluation if there is any change. 6624 if (mSrcFormat == srcFormat 6625 && mSrcChannelMask == srcChannelMask 6626 && mSrcSampleRate == srcSampleRate 6627 && mDstFormat == dstFormat 6628 && mDstChannelMask == dstChannelMask 6629 && mDstSampleRate == dstSampleRate) { 6630 return NO_ERROR; 6631 } 6632 6633 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6634 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6635 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6636 const bool valid = 6637 audio_is_input_channel(srcChannelMask) 6638 && audio_is_input_channel(dstChannelMask) 6639 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6640 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6641 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6642 ; // no upsampling checks for now 6643 if (!valid) { 6644 return BAD_VALUE; 6645 } 6646 6647 mSrcFormat = srcFormat; 6648 mSrcChannelMask = srcChannelMask; 6649 mSrcSampleRate = srcSampleRate; 6650 mDstFormat = dstFormat; 6651 mDstChannelMask = dstChannelMask; 6652 mDstSampleRate = dstSampleRate; 6653 6654 // compute derived parameters 6655 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6656 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6657 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6658 6659 // do we need to resample? 6660 delete mResampler; 6661 mResampler = NULL; 6662 if (mSrcSampleRate != mDstSampleRate) { 6663 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6664 mSrcChannelCount, mDstSampleRate); 6665 mResampler->setSampleRate(mSrcSampleRate); 6666 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6667 } 6668 6669 // are we running legacy channel conversion modes? 6670 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6671 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6672 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6673 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6674 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6675 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6676 6677 // do we need to process in float? 6678 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6679 6680 // do we need a staging buffer to convert for destination (we can still optimize this)? 6681 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6682 if (mResampler != NULL) { 6683 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6684 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6685 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6686 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6687 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6688 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6689 } else { 6690 mBufFrameSize = 0; 6691 } 6692 mBufFrames = 0; // force the buffer to be resized. 6693 6694 // do we need an input converter buffer provider to give us float? 6695 delete mInputConverterProvider; 6696 mInputConverterProvider = NULL; 6697 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6698 mInputConverterProvider = new ReformatBufferProvider( 6699 audio_channel_count_from_in_mask(mSrcChannelMask), 6700 mSrcFormat, 6701 AUDIO_FORMAT_PCM_FLOAT, 6702 256 /* provider buffer frame count */); 6703 } 6704 6705 // do we need a remixer to do channel mask conversion 6706 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6707 (void) memcpy_by_index_array_initialization_from_channel_mask( 6708 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6709 } 6710 return NO_ERROR; 6711} 6712 6713void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6714 void *dst, const void *src, size_t frames) 6715{ 6716 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6717 if (mBufFrameSize != 0 && mBufFrames < frames) { 6718 free(mBuf); 6719 mBufFrames = frames; 6720 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6721 } 6722 // do we need to do legacy upmix and downmix? 6723 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6724 void *dstBuf = mBuf != NULL ? mBuf : dst; 6725 if (mIsLegacyUpmix) { 6726 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6727 (const float *)src, frames); 6728 } else /*mIsLegacyDownmix */ { 6729 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6730 (const float *)src, frames); 6731 } 6732 if (mBuf != NULL) { 6733 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6734 frames * mDstChannelCount); 6735 } 6736 return; 6737 } 6738 // do we need to do channel mask conversion? 6739 if (mSrcChannelMask != mDstChannelMask) { 6740 void *dstBuf = mBuf != NULL ? mBuf : dst; 6741 memcpy_by_index_array(dstBuf, mDstChannelCount, 6742 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6743 if (dstBuf == dst) { 6744 return; // format is the same 6745 } 6746 } 6747 // convert to destination buffer 6748 const void *convertBuf = mBuf != NULL ? mBuf : src; 6749 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6750 frames * mDstChannelCount); 6751} 6752 6753void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6754 void *dst, /*not-a-const*/ void *src, size_t frames) 6755{ 6756 // src buffer format is ALWAYS float when entering this routine 6757 if (mIsLegacyUpmix) { 6758 ; // mono to stereo already handled by resampler 6759 } else if (mIsLegacyDownmix 6760 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6761 // the resampler outputs stereo for mono input channel (a feature?) 6762 // must convert to mono 6763 downmix_to_mono_float_from_stereo_float((float *)src, 6764 (const float *)src, frames); 6765 } else if (mSrcChannelMask != mDstChannelMask) { 6766 // convert to mono channel again for channel mask conversion (could be skipped 6767 // with further optimization). 6768 if (mSrcChannelCount == 1) { 6769 downmix_to_mono_float_from_stereo_float((float *)src, 6770 (const float *)src, frames); 6771 } 6772 // convert to destination format (in place, OK as float is larger than other types) 6773 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6774 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6775 frames * mSrcChannelCount); 6776 } 6777 // channel convert and save to dst 6778 memcpy_by_index_array(dst, mDstChannelCount, 6779 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6780 return; 6781 } 6782 // convert to destination format and save to dst 6783 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6784 frames * mDstChannelCount); 6785} 6786 6787bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6788 status_t& status) 6789{ 6790 bool reconfig = false; 6791 6792 status = NO_ERROR; 6793 6794 audio_format_t reqFormat = mFormat; 6795 uint32_t samplingRate = mSampleRate; 6796 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6797 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6798 6799 AudioParameter param = AudioParameter(keyValuePair); 6800 int value; 6801 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6802 // channel count change can be requested. Do we mandate the first client defines the 6803 // HAL sampling rate and channel count or do we allow changes on the fly? 6804 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6805 samplingRate = value; 6806 reconfig = true; 6807 } 6808 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6809 if (!audio_is_linear_pcm((audio_format_t) value)) { 6810 status = BAD_VALUE; 6811 } else { 6812 reqFormat = (audio_format_t) value; 6813 reconfig = true; 6814 } 6815 } 6816 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6817 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6818 if (!audio_is_input_channel(mask) || 6819 audio_channel_count_from_in_mask(mask) > FCC_8) { 6820 status = BAD_VALUE; 6821 } else { 6822 channelMask = mask; 6823 reconfig = true; 6824 } 6825 } 6826 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6827 // do not accept frame count changes if tracks are open as the track buffer 6828 // size depends on frame count and correct behavior would not be guaranteed 6829 // if frame count is changed after track creation 6830 if (mActiveTracks.size() > 0) { 6831 status = INVALID_OPERATION; 6832 } else { 6833 reconfig = true; 6834 } 6835 } 6836 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6837 // forward device change to effects that have requested to be 6838 // aware of attached audio device. 6839 for (size_t i = 0; i < mEffectChains.size(); i++) { 6840 mEffectChains[i]->setDevice_l(value); 6841 } 6842 6843 // store input device and output device but do not forward output device to audio HAL. 6844 // Note that status is ignored by the caller for output device 6845 // (see AudioFlinger::setParameters() 6846 if (audio_is_output_devices(value)) { 6847 mOutDevice = value; 6848 status = BAD_VALUE; 6849 } else { 6850 mInDevice = value; 6851 if (value != AUDIO_DEVICE_NONE) { 6852 mPrevInDevice = value; 6853 } 6854 // disable AEC and NS if the device is a BT SCO headset supporting those 6855 // pre processings 6856 if (mTracks.size() > 0) { 6857 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6858 mAudioFlinger->btNrecIsOff(); 6859 for (size_t i = 0; i < mTracks.size(); i++) { 6860 sp<RecordTrack> track = mTracks[i]; 6861 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6862 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6863 } 6864 } 6865 } 6866 } 6867 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6868 mAudioSource != (audio_source_t)value) { 6869 // forward device change to effects that have requested to be 6870 // aware of attached audio device. 6871 for (size_t i = 0; i < mEffectChains.size(); i++) { 6872 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6873 } 6874 mAudioSource = (audio_source_t)value; 6875 } 6876 6877 if (status == NO_ERROR) { 6878 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6879 keyValuePair.string()); 6880 if (status == INVALID_OPERATION) { 6881 inputStandBy(); 6882 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6883 keyValuePair.string()); 6884 } 6885 if (reconfig) { 6886 if (status == BAD_VALUE && 6887 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6888 audio_is_linear_pcm(reqFormat) && 6889 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6890 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6891 audio_channel_count_from_in_mask( 6892 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6893 status = NO_ERROR; 6894 } 6895 if (status == NO_ERROR) { 6896 readInputParameters_l(); 6897 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6898 } 6899 } 6900 } 6901 6902 return reconfig; 6903} 6904 6905String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6906{ 6907 Mutex::Autolock _l(mLock); 6908 if (initCheck() != NO_ERROR) { 6909 return String8(); 6910 } 6911 6912 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6913 const String8 out_s8(s); 6914 free(s); 6915 return out_s8; 6916} 6917 6918void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 6919 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6920 6921 desc->mIoHandle = mId; 6922 6923 switch (event) { 6924 case AUDIO_INPUT_OPENED: 6925 case AUDIO_INPUT_CONFIG_CHANGED: 6926 desc->mPatch = mPatch; 6927 desc->mChannelMask = mChannelMask; 6928 desc->mSamplingRate = mSampleRate; 6929 desc->mFormat = mFormat; 6930 desc->mFrameCount = mFrameCount; 6931 desc->mLatency = 0; 6932 break; 6933 6934 case AUDIO_INPUT_CLOSED: 6935 default: 6936 break; 6937 } 6938 mAudioFlinger->ioConfigChanged(event, desc, pid); 6939} 6940 6941void AudioFlinger::RecordThread::readInputParameters_l() 6942{ 6943 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6944 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6945 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6946 if (mChannelCount > FCC_8) { 6947 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6948 } 6949 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6950 mFormat = mHALFormat; 6951 if (!audio_is_linear_pcm(mFormat)) { 6952 ALOGE("HAL format %#x is not linear pcm", mFormat); 6953 } 6954 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6955 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6956 mFrameCount = mBufferSize / mFrameSize; 6957 // This is the formula for calculating the temporary buffer size. 6958 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6959 // 1 full output buffer, regardless of the alignment of the available input. 6960 // The value is somewhat arbitrary, and could probably be even larger. 6961 // A larger value should allow more old data to be read after a track calls start(), 6962 // without increasing latency. 6963 // 6964 // Note this is independent of the maximum downsampling ratio permitted for capture. 6965 mRsmpInFrames = mFrameCount * 7; 6966 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6967 free(mRsmpInBuffer); 6968 mRsmpInBuffer = NULL; 6969 6970 // TODO optimize audio capture buffer sizes ... 6971 // Here we calculate the size of the sliding buffer used as a source 6972 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6973 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6974 // be better to have it derived from the pipe depth in the long term. 6975 // The current value is higher than necessary. However it should not add to latency. 6976 6977 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6978 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 6979 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 6980 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 6981 6982 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6983 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6984} 6985 6986uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6987{ 6988 Mutex::Autolock _l(mLock); 6989 if (initCheck() != NO_ERROR) { 6990 return 0; 6991 } 6992 6993 return mInput->stream->get_input_frames_lost(mInput->stream); 6994} 6995 6996uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6997{ 6998 Mutex::Autolock _l(mLock); 6999 uint32_t result = 0; 7000 if (getEffectChain_l(sessionId) != 0) { 7001 result = EFFECT_SESSION; 7002 } 7003 7004 for (size_t i = 0; i < mTracks.size(); ++i) { 7005 if (sessionId == mTracks[i]->sessionId()) { 7006 result |= TRACK_SESSION; 7007 break; 7008 } 7009 } 7010 7011 return result; 7012} 7013 7014KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 7015{ 7016 KeyedVector<int, bool> ids; 7017 Mutex::Autolock _l(mLock); 7018 for (size_t j = 0; j < mTracks.size(); ++j) { 7019 sp<RecordThread::RecordTrack> track = mTracks[j]; 7020 int sessionId = track->sessionId(); 7021 if (ids.indexOfKey(sessionId) < 0) { 7022 ids.add(sessionId, true); 7023 } 7024 } 7025 return ids; 7026} 7027 7028AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7029{ 7030 Mutex::Autolock _l(mLock); 7031 AudioStreamIn *input = mInput; 7032 mInput = NULL; 7033 return input; 7034} 7035 7036// this method must always be called either with ThreadBase mLock held or inside the thread loop 7037audio_stream_t* AudioFlinger::RecordThread::stream() const 7038{ 7039 if (mInput == NULL) { 7040 return NULL; 7041 } 7042 return &mInput->stream->common; 7043} 7044 7045status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7046{ 7047 // only one chain per input thread 7048 if (mEffectChains.size() != 0) { 7049 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7050 return INVALID_OPERATION; 7051 } 7052 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7053 chain->setThread(this); 7054 chain->setInBuffer(NULL); 7055 chain->setOutBuffer(NULL); 7056 7057 checkSuspendOnAddEffectChain_l(chain); 7058 7059 // make sure enabled pre processing effects state is communicated to the HAL as we 7060 // just moved them to a new input stream. 7061 chain->syncHalEffectsState(); 7062 7063 mEffectChains.add(chain); 7064 7065 return NO_ERROR; 7066} 7067 7068size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7069{ 7070 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7071 ALOGW_IF(mEffectChains.size() != 1, 7072 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7073 chain.get(), mEffectChains.size(), this); 7074 if (mEffectChains.size() == 1) { 7075 mEffectChains.removeAt(0); 7076 } 7077 return 0; 7078} 7079 7080status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7081 audio_patch_handle_t *handle) 7082{ 7083 status_t status = NO_ERROR; 7084 7085 // store new device and send to effects 7086 mInDevice = patch->sources[0].ext.device.type; 7087 mPatch = *patch; 7088 for (size_t i = 0; i < mEffectChains.size(); i++) { 7089 mEffectChains[i]->setDevice_l(mInDevice); 7090 } 7091 7092 // disable AEC and NS if the device is a BT SCO headset supporting those 7093 // pre processings 7094 if (mTracks.size() > 0) { 7095 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7096 mAudioFlinger->btNrecIsOff(); 7097 for (size_t i = 0; i < mTracks.size(); i++) { 7098 sp<RecordTrack> track = mTracks[i]; 7099 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7100 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7101 } 7102 } 7103 7104 // store new source and send to effects 7105 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7106 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7107 for (size_t i = 0; i < mEffectChains.size(); i++) { 7108 mEffectChains[i]->setAudioSource_l(mAudioSource); 7109 } 7110 } 7111 7112 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7113 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7114 status = hwDevice->create_audio_patch(hwDevice, 7115 patch->num_sources, 7116 patch->sources, 7117 patch->num_sinks, 7118 patch->sinks, 7119 handle); 7120 } else { 7121 char *address; 7122 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7123 address = audio_device_address_to_parameter( 7124 patch->sources[0].ext.device.type, 7125 patch->sources[0].ext.device.address); 7126 } else { 7127 address = (char *)calloc(1, 1); 7128 } 7129 AudioParameter param = AudioParameter(String8(address)); 7130 free(address); 7131 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7132 (int)patch->sources[0].ext.device.type); 7133 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7134 (int)patch->sinks[0].ext.mix.usecase.source); 7135 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7136 param.toString().string()); 7137 *handle = AUDIO_PATCH_HANDLE_NONE; 7138 } 7139 7140 if (mInDevice != mPrevInDevice) { 7141 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7142 mPrevInDevice = mInDevice; 7143 } 7144 7145 return status; 7146} 7147 7148status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7149{ 7150 status_t status = NO_ERROR; 7151 7152 mInDevice = AUDIO_DEVICE_NONE; 7153 7154 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7155 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7156 status = hwDevice->release_audio_patch(hwDevice, handle); 7157 } else { 7158 AudioParameter param; 7159 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7160 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7161 param.toString().string()); 7162 } 7163 return status; 7164} 7165 7166void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7167{ 7168 Mutex::Autolock _l(mLock); 7169 mTracks.add(record); 7170} 7171 7172void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7173{ 7174 Mutex::Autolock _l(mLock); 7175 destroyTrack_l(record); 7176} 7177 7178void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7179{ 7180 ThreadBase::getAudioPortConfig(config); 7181 config->role = AUDIO_PORT_ROLE_SINK; 7182 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7183 config->ext.mix.usecase.source = mAudioSource; 7184} 7185 7186} // namespace android 7187